diff options
Diffstat (limited to 'sound')
188 files changed, 16249 insertions, 3796 deletions
diff --git a/sound/Kconfig b/sound/Kconfig index 4247406160e..a37bee094eb 100644 --- a/sound/Kconfig +++ b/sound/Kconfig @@ -1,11 +1,9 @@ # sound/Config.in # -menu "Sound" - depends on HAS_IOMEM - -config SOUND +menuconfig SOUND tristate "Sound card support" + depends on HAS_IOMEM help If you have a sound card in your computer, i.e. if it can say more than an occasional beep, say Y. Be sure to have all the information @@ -28,22 +26,22 @@ config SOUND and read <file:Documentation/sound/oss/README.modules>; the module will be called soundcore. +if SOUND + source "sound/oss/dmasound/Kconfig" if !M68K -menu "Advanced Linux Sound Architecture" - depends on SOUND!=n - -config SND +menuconfig SND tristate "Advanced Linux Sound Architecture" - depends on SOUND help Say 'Y' or 'M' to enable ALSA (Advanced Linux Sound Architecture), the new base sound system. For more information, see <http://www.alsa-project.org/> +if SND + source "sound/core/Kconfig" source "sound/drivers/Kconfig" @@ -58,9 +56,7 @@ source "sound/aoa/Kconfig" source "sound/arm/Kconfig" -if SPI source "sound/spi/Kconfig" -endif source "sound/mips/Kconfig" @@ -80,22 +76,20 @@ source "sound/parisc/Kconfig" source "sound/soc/Kconfig" -endmenu +endif # SND -menu "Open Sound System" - depends on SOUND!=n - -config SOUND_PRIME +menuconfig SOUND_PRIME tristate "Open Sound System (DEPRECATED)" - depends on SOUND help Say 'Y' or 'M' to enable Open Sound System drivers. +if SOUND_PRIME + source "sound/oss/Kconfig" -endmenu +endif # SOUND_PRIME -endif +endif # !M68K config AC97_BUS tristate @@ -105,4 +99,4 @@ config AC97_BUS sound although they're sharing the AC97 bus. Concerned drivers should "select" this. -endmenu +endif # SOUND diff --git a/sound/aoa/Kconfig b/sound/aoa/Kconfig index 5d5813cec4c..c081e18b954 100644 --- a/sound/aoa/Kconfig +++ b/sound/aoa/Kconfig @@ -1,18 +1,17 @@ -menu "Apple Onboard Audio driver" - depends on SND!=n && PPC_PMAC - -config SND_AOA +menuconfig SND_AOA tristate "Apple Onboard Audio driver" - depends on SND + depends on PPC_PMAC select SND_PCM ---help--- This option enables the new driver for the various Apple Onboard Audio components. +if SND_AOA + source "sound/aoa/fabrics/Kconfig" source "sound/aoa/codecs/Kconfig" source "sound/aoa/soundbus/Kconfig" -endmenu +endif # SND_AOA diff --git a/sound/aoa/codecs/Kconfig b/sound/aoa/codecs/Kconfig index d5fbd6016e9..808eb11ebac 100644 --- a/sound/aoa/codecs/Kconfig +++ b/sound/aoa/codecs/Kconfig @@ -1,6 +1,5 @@ config SND_AOA_ONYX tristate "support Onyx chip" - depends on SND_AOA select I2C select I2C_POWERMAC ---help--- @@ -10,7 +9,6 @@ config SND_AOA_ONYX #config SND_AOA_TOPAZ # tristate "support Topaz chips" -# depends on SND_AOA # ---help--- # This option enables support for the Topaz (CS84xx) # codec chips found in the latest Apple machines, @@ -19,7 +17,6 @@ config SND_AOA_ONYX config SND_AOA_TAS tristate "support TAS chips" - depends on SND_AOA select I2C select I2C_POWERMAC ---help--- @@ -29,7 +26,6 @@ config SND_AOA_TAS config SND_AOA_TOONIE tristate "support Toonie chip" - depends on SND_AOA ---help--- This option enables support for the toonie codec found in the Mac Mini. If you have a Mac Mini and diff --git a/sound/aoa/fabrics/Kconfig b/sound/aoa/fabrics/Kconfig index 50d7021ff67..3ca475a886b 100644 --- a/sound/aoa/fabrics/Kconfig +++ b/sound/aoa/fabrics/Kconfig @@ -1,6 +1,5 @@ config SND_AOA_FABRIC_LAYOUT tristate "layout-id fabric" - depends on SND_AOA select SND_AOA_SOUNDBUS select SND_AOA_SOUNDBUS_I2S ---help--- diff --git a/sound/aoa/soundbus/Kconfig b/sound/aoa/soundbus/Kconfig index 7368b7ddfe0..839d1137b9b 100644 --- a/sound/aoa/soundbus/Kconfig +++ b/sound/aoa/soundbus/Kconfig @@ -1,6 +1,5 @@ config SND_AOA_SOUNDBUS tristate "Apple Soundbus support" - depends on SOUND select SND_PCM ---help--- This option enables the generic driver for the soundbus diff --git a/sound/arm/Kconfig b/sound/arm/Kconfig index 2e4a5e0d16d..351e19ea378 100644 --- a/sound/arm/Kconfig +++ b/sound/arm/Kconfig @@ -1,11 +1,19 @@ # ALSA ARM drivers -menu "ALSA ARM devices" - depends on SND!=n && ARM +menuconfig SND_ARM + bool "ARM sound devices" + depends on ARM + default y + help + Support for sound devices specific to ARM architectures. + Drivers that are implemented on ASoC can be found in + "ALSA for SoC audio support" section. + +if SND_ARM config SND_SA11XX_UDA1341 tristate "SA11xx UDA1341TS driver (iPaq H3600)" - depends on ARCH_SA1100 && SND && L3 + depends on ARCH_SA1100 && L3 select SND_PCM help Say Y here if you have a Compaq iPaq H3x00 handheld computer @@ -16,7 +24,7 @@ config SND_SA11XX_UDA1341 config SND_ARMAACI tristate "ARM PrimeCell PL041 AC Link support" - depends on SND && ARM_AMBA + depends on ARM_AMBA select SND_PCM select SND_AC97_CODEC @@ -26,11 +34,12 @@ config SND_PXA2XX_PCM config SND_PXA2XX_AC97 tristate "AC97 driver for the Intel PXA2xx chip" - depends on ARCH_PXA && SND + depends on ARCH_PXA select SND_PXA2XX_PCM select SND_AC97_CODEC help Say Y or M if you want to support any AC97 codec attached to the PXA2xx AC97 interface. -endmenu +endif # SND_ARM + diff --git a/sound/arm/sa11xx-uda1341.c b/sound/arm/sa11xx-uda1341.c index 0eff33ca0f7..faeddf3eced 100644 --- a/sound/arm/sa11xx-uda1341.c +++ b/sound/arm/sa11xx-uda1341.c @@ -21,8 +21,6 @@ * merged HAL layer (patches from Brian) */ -/* $Id: sa11xx-uda1341.c,v 1.27 2005/12/07 09:13:42 cladisch Exp $ */ - /*************************************************************************************************** * * To understand what Alsa Drivers should be doing look at "Writing an Alsa Driver" by Takashi Iwai diff --git a/sound/core/Kconfig b/sound/core/Kconfig index a8d71c6c8e7..335d45ecde6 100644 --- a/sound/core/Kconfig +++ b/sound/core/Kconfig @@ -1,24 +1,19 @@ # ALSA soundcard-configuration config SND_TIMER tristate - depends on SND config SND_PCM tristate select SND_TIMER - depends on SND config SND_HWDEP tristate - depends on SND config SND_RAWMIDI tristate - depends on SND config SND_SEQUENCER tristate "Sequencer support" - depends on SND select SND_TIMER help Say Y or M to enable MIDI sequencer and router support. This @@ -44,11 +39,9 @@ config SND_SEQ_DUMMY config SND_OSSEMUL bool - depends on SND config SND_MIXER_OSS tristate "OSS Mixer API" - depends on SND select SND_OSSEMUL help To enable OSS mixer API emulation (/dev/mixer*), say Y here @@ -61,7 +54,6 @@ config SND_MIXER_OSS config SND_PCM_OSS tristate "OSS PCM (digital audio) API" - depends on SND select SND_OSSEMUL select SND_PCM help @@ -84,7 +76,7 @@ config SND_PCM_OSS_PLUGINS config SND_SEQUENCER_OSS bool "OSS Sequencer API" - depends on SND && SND_SEQUENCER + depends on SND_SEQUENCER select SND_OSSEMUL help Say Y here to enable OSS sequencer emulation (both @@ -98,7 +90,7 @@ config SND_SEQUENCER_OSS config SND_RTCTIMER tristate "RTC Timer support" - depends on SND && RTC + depends on RTC select SND_TIMER help Say Y here to enable RTC timer support for ALSA. ALSA uses @@ -123,7 +115,6 @@ config SND_SEQ_RTCTIMER_DEFAULT config SND_DYNAMIC_MINORS bool "Dynamic device file minor numbers" - depends on SND help If you say Y here, the minor numbers of ALSA device files in /dev/snd/ are allocated dynamically. This allows you to have @@ -134,7 +125,6 @@ config SND_DYNAMIC_MINORS config SND_SUPPORT_OLD_API bool "Support old ALSA API" - depends on SND default y help Say Y here to support the obsolete ALSA PCM API (ver.0.9.0 rc3 @@ -142,7 +132,7 @@ config SND_SUPPORT_OLD_API config SND_VERBOSE_PROCFS bool "Verbose procfs contents" - depends on SND && PROC_FS + depends on PROC_FS default y help Say Y here to include code for verbose procfs contents (provides @@ -151,7 +141,6 @@ config SND_VERBOSE_PROCFS config SND_VERBOSE_PRINTK bool "Verbose printk" - depends on SND help Say Y here to enable verbose log messages. These messages will help to identify source file and position containing @@ -161,16 +150,17 @@ config SND_VERBOSE_PRINTK config SND_DEBUG bool "Debug" - depends on SND help Say Y here to enable ALSA debug code. -config SND_DEBUG_DETECT - bool "Debug detection" +config SND_DEBUG_VERBOSE + bool "More verbose debug" depends on SND_DEBUG help - Say Y here to enable extra-verbose log messages printed when - detecting devices. + Say Y here to enable extra-verbose debugging messages. + + Let me repeat: it enables EXTRA-VERBOSE DEBUGGING messages. + So, say Y only if you are ready to be annoyed. config SND_PCM_XRUN_DEBUG bool "Enable PCM ring buffer overrun/underrun debugging" @@ -184,4 +174,3 @@ config SND_PCM_XRUN_DEBUG config SND_VMASTER bool - depends on SND diff --git a/sound/core/control.c b/sound/core/control.c index 01a1a5af47b..281b2e2ef0e 100644 --- a/sound/core/control.c +++ b/sound/core/control.c @@ -684,7 +684,8 @@ static int snd_ctl_elem_info_user(struct snd_ctl_file *ctl, return result; } -int snd_ctl_elem_read(struct snd_card *card, struct snd_ctl_elem_value *control) +static int snd_ctl_elem_read(struct snd_card *card, + struct snd_ctl_elem_value *control) { struct snd_kcontrol *kctl; struct snd_kcontrol_volatile *vd; @@ -734,8 +735,8 @@ static int snd_ctl_elem_read_user(struct snd_card *card, return result; } -int snd_ctl_elem_write(struct snd_card *card, struct snd_ctl_file *file, - struct snd_ctl_elem_value *control) +static int snd_ctl_elem_write(struct snd_card *card, struct snd_ctl_file *file, + struct snd_ctl_elem_value *control) { struct snd_kcontrol *kctl; struct snd_kcontrol_volatile *vd; diff --git a/sound/core/init.c b/sound/core/init.c index ac057341613..5c254d498ae 100644 --- a/sound/core/init.c +++ b/sound/core/init.c @@ -46,17 +46,24 @@ static char *slots[SNDRV_CARDS]; module_param_array(slots, charp, NULL, 0444); MODULE_PARM_DESC(slots, "Module names assigned to the slots."); -/* return non-zero if the given index is already reserved for another +/* return non-zero if the given index is reserved for the given * module via slots option */ -static int module_slot_mismatch(struct module *module, int idx) +static int module_slot_match(struct module *module, int idx) { + int match = 1; #ifdef MODULE - char *s1, *s2; + const char *s1, *s2; + if (!module || !module->name || !slots[idx]) return 0; - s1 = slots[idx]; - s2 = module->name; + + s1 = module->name; + s2 = slots[idx]; + if (*s2 == '!') { + match = 0; /* negative match */ + s2++; + } /* compare module name strings * hyphens are handled as equivalent with underscore */ @@ -68,12 +75,12 @@ static int module_slot_mismatch(struct module *module, int idx) if (c2 == '-') c2 = '_'; if (c1 != c2) - return 1; + return !match; if (!c1) break; } -#endif - return 0; +#endif /* MODULE */ + return match; } #if defined(CONFIG_SND_MIXER_OSS) || defined(CONFIG_SND_MIXER_OSS_MODULE) @@ -129,7 +136,7 @@ struct snd_card *snd_card_new(int idx, const char *xid, struct module *module, int extra_size) { struct snd_card *card; - int err; + int err, idx2; if (extra_size < 0) extra_size = 0; @@ -144,35 +151,41 @@ struct snd_card *snd_card_new(int idx, const char *xid, err = 0; mutex_lock(&snd_card_mutex); if (idx < 0) { - int idx2; for (idx2 = 0; idx2 < SNDRV_CARDS; idx2++) /* idx == -1 == 0xffff means: take any free slot */ if (~snd_cards_lock & idx & 1<<idx2) { - if (module_slot_mismatch(module, idx2)) - continue; - idx = idx2; - if (idx >= snd_ecards_limit) - snd_ecards_limit = idx + 1; - break; + if (module_slot_match(module, idx2)) { + idx = idx2; + break; + } + } + } + if (idx < 0) { + for (idx2 = 0; idx2 < SNDRV_CARDS; idx2++) + /* idx == -1 == 0xffff means: take any free slot */ + if (~snd_cards_lock & idx & 1<<idx2) { + if (!slots[idx2] || !*slots[idx2]) { + idx = idx2; + break; + } } - } else { - if (idx < snd_ecards_limit) { - if (snd_cards_lock & (1 << idx)) - err = -EBUSY; /* invalid */ - } else { - if (idx < SNDRV_CARDS) - snd_ecards_limit = idx + 1; /* increase the limit */ - else - err = -ENODEV; - } } - if (idx < 0 || err < 0) { + if (idx < 0) + err = -ENODEV; + else if (idx < snd_ecards_limit) { + if (snd_cards_lock & (1 << idx)) + err = -EBUSY; /* invalid */ + } else if (idx >= SNDRV_CARDS) + err = -ENODEV; + if (err < 0) { mutex_unlock(&snd_card_mutex); snd_printk(KERN_ERR "cannot find the slot for index %d (range 0-%i), error: %d\n", idx, snd_ecards_limit - 1, err); goto __error; } snd_cards_lock |= 1 << idx; /* lock it */ + if (idx >= snd_ecards_limit) + snd_ecards_limit = idx + 1; /* increase the limit */ mutex_unlock(&snd_card_mutex); card->number = idx; card->module = module; diff --git a/sound/core/memalloc.c b/sound/core/memalloc.c index 23b7bc02728..f5d6d8d1297 100644 --- a/sound/core/memalloc.c +++ b/sound/core/memalloc.c @@ -80,68 +80,6 @@ struct snd_mem_list { #endif /* - * Hacks - */ - -#if defined(__i386__) -/* - * A hack to allocate large buffers via dma_alloc_coherent() - * - * since dma_alloc_coherent always tries GFP_DMA when the requested - * pci memory region is below 32bit, it happens quite often that even - * 2 order of pages cannot be allocated. - * - * so in the following, we allocate at first without dma_mask, so that - * allocation will be done without GFP_DMA. if the area doesn't match - * with the requested region, then realloate with the original dma_mask - * again. - * - * Really, we want to move this type of thing into dma_alloc_coherent() - * so dma_mask doesn't have to be messed with. - */ - -static void *snd_dma_hack_alloc_coherent(struct device *dev, size_t size, - dma_addr_t *dma_handle, - gfp_t flags) -{ - void *ret; - u64 dma_mask, coherent_dma_mask; - - if (dev == NULL || !dev->dma_mask) - return dma_alloc_coherent(dev, size, dma_handle, flags); - dma_mask = *dev->dma_mask; - coherent_dma_mask = dev->coherent_dma_mask; - *dev->dma_mask = 0xffffffff; /* do without masking */ - dev->coherent_dma_mask = 0xffffffff; /* do without masking */ - ret = dma_alloc_coherent(dev, size, dma_handle, flags); - *dev->dma_mask = dma_mask; /* restore */ - dev->coherent_dma_mask = coherent_dma_mask; /* restore */ - if (ret) { - /* obtained address is out of range? */ - if (((unsigned long)*dma_handle + size - 1) & ~dma_mask) { - /* reallocate with the proper mask */ - dma_free_coherent(dev, size, ret, *dma_handle); - ret = dma_alloc_coherent(dev, size, dma_handle, flags); - } - } else { - /* wish to success now with the proper mask... */ - if (dma_mask != 0xffffffffUL) { - /* allocation with GFP_ATOMIC to avoid the long stall */ - flags &= ~GFP_KERNEL; - flags |= GFP_ATOMIC; - ret = dma_alloc_coherent(dev, size, dma_handle, flags); - } - } - return ret; -} - -/* redefine dma_alloc_coherent for some architectures */ -#undef dma_alloc_coherent -#define dma_alloc_coherent snd_dma_hack_alloc_coherent - -#endif /* arch */ - -/* * * Generic memory allocators * diff --git a/sound/core/seq/seq_clientmgr.c b/sound/core/seq/seq_clientmgr.c index 47cfa5186e3..7a1545d2d95 100644 --- a/sound/core/seq/seq_clientmgr.c +++ b/sound/core/seq/seq_clientmgr.c @@ -148,7 +148,7 @@ struct snd_seq_client *snd_seq_client_use_ptr(int clientid) return NULL; } spin_unlock_irqrestore(&clients_lock, flags); -#ifdef CONFIG_KMOD +#ifdef CONFIG_MODULES if (!in_interrupt()) { static char client_requested[SNDRV_SEQ_GLOBAL_CLIENTS]; static char card_requested[SNDRV_CARDS]; diff --git a/sound/core/seq/seq_device.c b/sound/core/seq/seq_device.c index 2f00ad28a2b..05410e536a4 100644 --- a/sound/core/seq/seq_device.c +++ b/sound/core/seq/seq_device.c @@ -124,7 +124,7 @@ static void snd_seq_device_info(struct snd_info_entry *entry, * load all registered drivers (called from seq_clientmgr.c) */ -#ifdef CONFIG_KMOD +#ifdef CONFIG_MODULES /* avoid auto-loading during module_init() */ static int snd_seq_in_init; void snd_seq_autoload_lock(void) @@ -140,7 +140,7 @@ void snd_seq_autoload_unlock(void) void snd_seq_device_load_drivers(void) { -#ifdef CONFIG_KMOD +#ifdef CONFIG_MODULES struct ops_list *ops; /* Calling request_module during module_init() @@ -566,7 +566,5 @@ EXPORT_SYMBOL(snd_seq_device_load_drivers); EXPORT_SYMBOL(snd_seq_device_new); EXPORT_SYMBOL(snd_seq_device_register_driver); EXPORT_SYMBOL(snd_seq_device_unregister_driver); -#ifdef CONFIG_KMOD EXPORT_SYMBOL(snd_seq_autoload_lock); EXPORT_SYMBOL(snd_seq_autoload_unlock); -#endif diff --git a/sound/core/sound.c b/sound/core/sound.c index 62e057a9465..1003ae375d4 100644 --- a/sound/core/sound.c +++ b/sound/core/sound.c @@ -61,14 +61,14 @@ EXPORT_SYMBOL(snd_ecards_limit); static struct snd_minor *snd_minors[SNDRV_OS_MINORS]; static DEFINE_MUTEX(sound_mutex); -#ifdef CONFIG_KMOD +#ifdef CONFIG_MODULES /** * snd_request_card - try to load the card module * @card: the card number * * Tries to load the module "snd-card-X" for the given card number - * via KMOD. Returns immediately if already loaded. + * via request_module. Returns immediately if already loaded. */ void snd_request_card(int card) { @@ -93,7 +93,7 @@ static void snd_request_other(int minor) request_module(str); } -#endif /* request_module support */ +#endif /* modular kernel */ /** * snd_lookup_minor_data - get user data of a registered device @@ -133,7 +133,7 @@ static int __snd_open(struct inode *inode, struct file *file) return -ENODEV; mptr = snd_minors[minor]; if (mptr == NULL) { -#ifdef CONFIG_KMOD +#ifdef CONFIG_MODULES int dev = SNDRV_MINOR_DEVICE(minor); if (dev == SNDRV_MINOR_CONTROL) { /* /dev/aloadC? */ diff --git a/sound/core/timer.c b/sound/core/timer.c index 9d8184a2c2d..0af337efc64 100644 --- a/sound/core/timer.c +++ b/sound/core/timer.c @@ -146,7 +146,7 @@ static struct snd_timer *snd_timer_find(struct snd_timer_id *tid) return NULL; } -#ifdef CONFIG_KMOD +#ifdef CONFIG_MODULES static void snd_timer_request(struct snd_timer_id *tid) { @@ -259,8 +259,8 @@ int snd_timer_open(struct snd_timer_instance **ti, /* open a master instance */ mutex_lock(®ister_mutex); timer = snd_timer_find(tid); -#ifdef CONFIG_KMOD - if (timer == NULL) { +#ifdef CONFIG_MODULES + if (!timer) { mutex_unlock(®ister_mutex); snd_timer_request(tid); mutex_lock(®ister_mutex); diff --git a/sound/drivers/Kconfig b/sound/drivers/Kconfig index 602b58e3b55..255fd18b9ae 100644 --- a/sound/drivers/Kconfig +++ b/sound/drivers/Kconfig @@ -1,15 +1,41 @@ -# ALSA generic drivers +config SND_MPU401_UART + tristate + select SND_RAWMIDI -menu "Generic devices" - depends on SND!=n +config SND_OPL3_LIB + tristate + select SND_TIMER + select SND_HWDEP +config SND_OPL4_LIB + tristate + select SND_TIMER + select SND_HWDEP + +config SND_VX_LIB + tristate + select SND_HWDEP + select SND_PCM + +config SND_AC97_CODEC + tristate + select SND_PCM + select AC97_BUS + select SND_VMASTER + +menuconfig SND_DRIVERS + bool "Generic sound devices" + default y + help + Support for generic sound devices. + +if SND_DRIVERS config SND_PCSP tristate "PC-Speaker support (READ HELP!)" depends on PCSPKR_PLATFORM && X86_PC && HIGH_RES_TIMERS depends on INPUT depends on EXPERIMENTAL - depends on SND select SND_PCM help If you don't have a sound card in your computer, you can include a @@ -35,33 +61,8 @@ config SND_PCSP Say M if you don't. Say Y only if you really know what you do. -config SND_MPU401_UART - tristate - select SND_RAWMIDI - -config SND_OPL3_LIB - tristate - select SND_TIMER - select SND_HWDEP - -config SND_OPL4_LIB - tristate - select SND_TIMER - select SND_HWDEP - -config SND_VX_LIB - tristate - select SND_HWDEP - select SND_PCM - -config SND_AC97_CODEC - tristate - select SND_PCM - select AC97_BUS - config SND_DUMMY tristate "Dummy (/dev/null) soundcard" - depends on SND select SND_PCM help Say Y here to include the dummy driver. This driver does @@ -90,7 +91,6 @@ config SND_VIRMIDI config SND_MTPAV tristate "MOTU MidiTimePiece AV multiport MIDI" - depends on SND select SND_RAWMIDI help To use a MOTU MidiTimePiece AV multiport MIDI adapter @@ -102,7 +102,7 @@ config SND_MTPAV config SND_MTS64 tristate "ESI Miditerminal 4140 driver" - depends on SND && PARPORT + depends on PARPORT select SND_RAWMIDI help The ESI Miditerminal 4140 is a 4 In 4 Out MIDI Interface with @@ -115,7 +115,6 @@ config SND_MTS64 config SND_SERIAL_U16550 tristate "UART16550 serial MIDI driver" - depends on SND select SND_RAWMIDI help To include support for MIDI serial port interfaces, say Y here @@ -131,7 +130,6 @@ config SND_SERIAL_U16550 config SND_MPU401 tristate "Generic MPU-401 UART driver" - depends on SND select SND_MPU401_UART help Say Y here to include support for MIDI ports compatible with @@ -142,7 +140,7 @@ config SND_MPU401 config SND_PORTMAN2X4 tristate "Portman 2x4 driver" - depends on SND && PARPORT + depends on PARPORT select SND_RAWMIDI help Say Y here to include support for Midiman Portman 2x4 parallel @@ -153,7 +151,7 @@ config SND_PORTMAN2X4 config SND_ML403_AC97CR tristate "Xilinx ML403 AC97 Controller Reference" - depends on SND && XILINX_VIRTEX + depends on XILINX_VIRTEX select SND_AC97_CODEC help Say Y here to include support for the @@ -163,4 +161,25 @@ config SND_ML403_AC97CR To compile this driver as a module, choose M here: the module will be called snd-ml403_ac97cr. -endmenu +config SND_AC97_POWER_SAVE + bool "AC97 Power-Saving Mode" + depends on SND_AC97_CODEC && EXPERIMENTAL + default n + help + Say Y here to enable the aggressive power-saving support of + AC97 codecs. In this mode, the power-mode is dynamically + controlled at each open/close. + + The mode is activated by passing power_save=1 option to + snd-ac97-codec driver. You can toggle it dynamically over + sysfs, too. + +config SND_AC97_POWER_SAVE_DEFAULT + int "Default time-out for AC97 power-save mode" + depends on SND_AC97_POWER_SAVE + default 0 + help + The default time-out value in seconds for AC97 automatic + power-save mode. 0 means to disable the power-save mode. + +endif # SND_DRIVERS diff --git a/sound/drivers/vx/vx_hwdep.c b/sound/drivers/vx/vx_hwdep.c index 1dfe6948e6f..efd22e92bce 100644 --- a/sound/drivers/vx/vx_hwdep.c +++ b/sound/drivers/vx/vx_hwdep.c @@ -183,7 +183,7 @@ static int vx_hwdep_dsp_load(struct snd_hwdep *hw, kfree(fw); return -ENOMEM; } - if (copy_from_user(fw->data, dsp->image, dsp->length)) { + if (copy_from_user((void *)fw->data, dsp->image, dsp->length)) { free_fw(fw); return -EFAULT; } diff --git a/sound/i2c/cs8427.c b/sound/i2c/cs8427.c index e57e9cbe6a0..9c3d361accf 100644 --- a/sound/i2c/cs8427.c +++ b/sound/i2c/cs8427.c @@ -23,6 +23,7 @@ #include <linux/slab.h> #include <linux/delay.h> #include <linux/init.h> +#include <asm/unaligned.h> #include <sound/core.h> #include <sound/control.h> #include <sound/pcm.h> @@ -264,10 +265,7 @@ int snd_cs8427_create(struct snd_i2c_bus *bus, goto __fail; } /* write default channel status bytes */ - buf[0] = ((unsigned char)(SNDRV_PCM_DEFAULT_CON_SPDIF >> 0)); - buf[1] = ((unsigned char)(SNDRV_PCM_DEFAULT_CON_SPDIF >> 8)); - buf[2] = ((unsigned char)(SNDRV_PCM_DEFAULT_CON_SPDIF >> 16)); - buf[3] = ((unsigned char)(SNDRV_PCM_DEFAULT_CON_SPDIF >> 24)); + put_unaligned_le32(SNDRV_PCM_DEFAULT_CON_SPDIF, buf); memset(buf + 4, 0, 24 - 4); if (snd_cs8427_send_corudata(device, 0, buf, 24) < 0) goto __fail; diff --git a/sound/i2c/l3/uda1341.c b/sound/i2c/l3/uda1341.c index bfa5d2c3608..1f4942ea141 100644 --- a/sound/i2c/l3/uda1341.c +++ b/sound/i2c/l3/uda1341.c @@ -17,8 +17,6 @@ * 2002-05-12 Tomas Kasparek another code cleanup */ -/* $Id: uda1341.c,v 1.18 2005/11/17 14:17:21 tiwai Exp $ */ - #include <linux/module.h> #include <linux/init.h> #include <linux/types.h> diff --git a/sound/isa/Kconfig b/sound/isa/Kconfig index 2639a6ab8f2..25347a25d63 100644 --- a/sound/isa/Kconfig +++ b/sound/isa/Kconfig @@ -21,12 +21,17 @@ config SND_SB16_DSP select SND_PCM select SND_SB_COMMON -menu "ISA devices" - depends on SND!=n && ISA && ISA_DMA_API +menuconfig SND_ISA + bool "ISA sound devices" + depends on ISA && ISA_DMA_API + default y + help + Support for sound devices connected via the ISA bus. + +if SND_ISA config SND_ADLIB tristate "AdLib FM card" - depends on SND select SND_OPL3_LIB help Say Y here to include support for AdLib FM cards. @@ -36,7 +41,7 @@ config SND_ADLIB config SND_AD1816A tristate "Analog Devices SoundPort AD1816A" - depends on SND && PNP && ISA + depends on PNP select ISAPNP select SND_OPL3_LIB select SND_MPU401_UART @@ -50,7 +55,6 @@ config SND_AD1816A config SND_AD1848 tristate "Generic AD1848/CS4248 driver" - depends on SND select SND_AD1848_LIB help Say Y here to include support for AD1848 (Analog Devices) or @@ -64,7 +68,7 @@ config SND_AD1848 config SND_ALS100 tristate "Avance Logic ALS100/ALS120" - depends on SND && PNP && ISA + depends on PNP select ISAPNP select SND_OPL3_LIB select SND_MPU401_UART @@ -78,7 +82,7 @@ config SND_ALS100 config SND_AZT2320 tristate "Aztech Systems AZT2320" - depends on SND && PNP && ISA + depends on PNP select ISAPNP select SND_OPL3_LIB select SND_MPU401_UART @@ -92,7 +96,6 @@ config SND_AZT2320 config SND_CMI8330 tristate "C-Media CMI8330" - depends on SND select SND_AD1848_LIB select SND_SB16_DSP help @@ -104,7 +107,6 @@ config SND_CMI8330 config SND_CS4231 tristate "Generic Cirrus Logic CS4231 driver" - depends on SND select SND_MPU401_UART select SND_CS4231_LIB help @@ -116,7 +118,6 @@ config SND_CS4231 config SND_CS4232 tristate "Generic Cirrus Logic CS4232 driver" - depends on SND select SND_OPL3_LIB select SND_MPU401_UART select SND_CS4231_LIB @@ -129,7 +130,6 @@ config SND_CS4232 config SND_CS4236 tristate "Generic Cirrus Logic CS4236+ driver" - depends on SND select SND_OPL3_LIB select SND_MPU401_UART select SND_CS4231_LIB @@ -142,7 +142,7 @@ config SND_CS4236 config SND_DT019X tristate "Diamond Technologies DT-019X, Avance Logic ALS-007" - depends on SND && PNP && ISA + depends on PNP select ISAPNP select SND_OPL3_LIB select SND_MPU401_UART @@ -156,7 +156,7 @@ config SND_DT019X config SND_ES968 tristate "Generic ESS ES968 driver" - depends on SND && PNP && ISA + depends on PNP select ISAPNP select SND_MPU401_UART select SND_SB8_DSP @@ -168,7 +168,6 @@ config SND_ES968 config SND_ES1688 tristate "Generic ESS ES688/ES1688 driver" - depends on SND select SND_OPL3_LIB select SND_MPU401_UART select SND_PCM @@ -181,7 +180,6 @@ config SND_ES1688 config SND_ES18XX tristate "Generic ESS ES18xx driver" - depends on SND select SND_OPL3_LIB select SND_MPU401_UART select SND_PCM @@ -193,7 +191,7 @@ config SND_ES18XX config SND_SC6000 tristate "Gallant SC-6000, Audio Excel DSP 16" - depends on SND && HAS_IOPORT + depends on HAS_IOPORT select SND_AD1848_LIB select SND_OPL3_LIB select SND_MPU401_UART @@ -204,15 +202,10 @@ config SND_SC6000 To compile this driver as a module, choose M here: the module will be called snd-sc6000. -config SND_GUS_SYNTH - tristate - config SND_GUSCLASSIC tristate "Gravis UltraSound Classic" - depends on SND select SND_RAWMIDI select SND_PCM - select SND_GUS_SYNTH help Say Y here to include support for Gravis UltraSound Classic soundcards. @@ -222,11 +215,9 @@ config SND_GUSCLASSIC config SND_GUSEXTREME tristate "Gravis UltraSound Extreme" - depends on SND select SND_HWDEP select SND_MPU401_UART select SND_PCM - select SND_GUS_SYNTH help Say Y here to include support for Gravis UltraSound Extreme soundcards. @@ -236,10 +227,8 @@ config SND_GUSEXTREME config SND_GUSMAX tristate "Gravis UltraSound MAX" - depends on SND select SND_RAWMIDI select SND_CS4231_LIB - select SND_GUS_SYNTH help Say Y here to include support for Gravis UltraSound MAX soundcards. @@ -249,10 +238,9 @@ config SND_GUSMAX config SND_INTERWAVE tristate "AMD InterWave, Gravis UltraSound PnP" - depends on SND && PNP && ISA + depends on PNP select SND_RAWMIDI select SND_CS4231_LIB - select SND_GUS_SYNTH help Say Y here to include support for AMD InterWave based soundcards (Gravis UltraSound Plug & Play, STB SoundRage32, @@ -263,10 +251,9 @@ config SND_INTERWAVE config SND_INTERWAVE_STB tristate "AMD InterWave + TEA6330T (UltraSound 32-Pro)" - depends on SND && PNP && ISA + depends on PNP select SND_RAWMIDI select SND_CS4231_LIB - select SND_GUS_SYNTH help Say Y here to include support for AMD InterWave based soundcards with a TEA6330T bass and treble regulator @@ -277,7 +264,6 @@ config SND_INTERWAVE_STB config SND_OPL3SA2 tristate "Yamaha OPL3-SA2/SA3" - depends on SND select SND_OPL3_LIB select SND_MPU401_UART select SND_CS4231_LIB @@ -290,7 +276,6 @@ config SND_OPL3SA2 config SND_OPTI92X_AD1848 tristate "OPTi 82C92x - AD1848" - depends on SND select SND_OPL3_LIB select SND_OPL4_LIB select SND_MPU401_UART @@ -304,7 +289,6 @@ config SND_OPTI92X_AD1848 config SND_OPTI92X_CS4231 tristate "OPTi 82C92x - CS4231" - depends on SND select SND_OPL3_LIB select SND_OPL4_LIB select SND_MPU401_UART @@ -318,10 +302,9 @@ config SND_OPTI92X_CS4231 config SND_OPTI93X tristate "OPTi 82C93x" - depends on SND select SND_OPL3_LIB select SND_MPU401_UART - select SND_PCM + select SND_CS4231_LIB help Say Y here to include support for soundcards based on Opti 82C93x chips. @@ -331,7 +314,6 @@ config SND_OPTI93X config SND_MIRO tristate "Miro miroSOUND PCM1pro/PCM12/PCM20radio driver" - depends on SND select SND_OPL4_LIB select SND_CS4231_LIB select SND_MPU401_UART @@ -345,7 +327,6 @@ config SND_MIRO config SND_SB8 tristate "Sound Blaster 1.0/2.0/Pro (8-bit)" - depends on SND select SND_OPL3_LIB select SND_RAWMIDI select SND_SB8_DSP @@ -358,7 +339,6 @@ config SND_SB8 config SND_SB16 tristate "Sound Blaster 16 (PnP)" - depends on SND select SND_OPL3_LIB select SND_MPU401_UART select SND_SB16_DSP @@ -371,7 +351,6 @@ config SND_SB16 config SND_SBAWE tristate "Sound Blaster AWE (32,64) (PnP)" - depends on SND select SND_OPL3_LIB select SND_MPU401_UART select SND_SB16_DSP @@ -402,7 +381,6 @@ config SND_SB16_CSP_FIRMWARE_IN_KERNEL config SND_SGALAXY tristate "Aztech Sound Galaxy" - depends on SND select SND_AD1848_LIB help Say Y here to include support for Aztech Sound Galaxy @@ -413,7 +391,6 @@ config SND_SGALAXY config SND_SSCAPE tristate "Ensoniq SoundScape PnP driver" - depends on SND select SND_HWDEP select SND_MPU401_UART select SND_CS4231_LIB @@ -426,7 +403,6 @@ config SND_SSCAPE config SND_WAVEFRONT tristate "Turtle Beach Maui,Tropez,Tropez+ (Wavefront)" - depends on SND select FW_LOADER select SND_OPL3_LIB select SND_MPU401_UART @@ -448,4 +424,5 @@ config SND_WAVEFRONT_FIRMWARE_IN_KERNEL you need to install the firmware files from the alsa-firmware package. -endmenu +endif # SND_ISA + diff --git a/sound/isa/cs423x/cs4231_lib.c b/sound/isa/cs423x/cs4231_lib.c index 0aa8649e5c7..521db705d17 100644 --- a/sound/isa/cs423x/cs4231_lib.c +++ b/sound/isa/cs423x/cs4231_lib.c @@ -119,6 +119,42 @@ static unsigned char snd_cs4231_original_image[32] = 0x00, /* 1f/31 - cbrl */ }; +static unsigned char snd_opti93x_original_image[32] = +{ + 0x00, /* 00/00 - l_mixout_outctrl */ + 0x00, /* 01/01 - r_mixout_outctrl */ + 0x88, /* 02/02 - l_cd_inctrl */ + 0x88, /* 03/03 - r_cd_inctrl */ + 0x88, /* 04/04 - l_a1/fm_inctrl */ + 0x88, /* 05/05 - r_a1/fm_inctrl */ + 0x80, /* 06/06 - l_dac_inctrl */ + 0x80, /* 07/07 - r_dac_inctrl */ + 0x00, /* 08/08 - ply_dataform_reg */ + 0x00, /* 09/09 - if_conf */ + 0x00, /* 0a/10 - pin_ctrl */ + 0x00, /* 0b/11 - err_init_reg */ + 0x0a, /* 0c/12 - id_reg */ + 0x00, /* 0d/13 - reserved */ + 0x00, /* 0e/14 - ply_upcount_reg */ + 0x00, /* 0f/15 - ply_lowcount_reg */ + 0x88, /* 10/16 - reserved/l_a1_inctrl */ + 0x88, /* 11/17 - reserved/r_a1_inctrl */ + 0x88, /* 12/18 - l_line_inctrl */ + 0x88, /* 13/19 - r_line_inctrl */ + 0x88, /* 14/20 - l_mic_inctrl */ + 0x88, /* 15/21 - r_mic_inctrl */ + 0x80, /* 16/22 - l_out_outctrl */ + 0x80, /* 17/23 - r_out_outctrl */ + 0x00, /* 18/24 - reserved */ + 0x00, /* 19/25 - reserved */ + 0x00, /* 1a/26 - reserved */ + 0x00, /* 1b/27 - reserved */ + 0x00, /* 1c/28 - cap_dataform_reg */ + 0x00, /* 1d/29 - reserved */ + 0x00, /* 1e/30 - cap_upcount_reg */ + 0x00 /* 1f/31 - cap_lowcount_reg */ +}; + /* * Basic I/O functions */ @@ -895,7 +931,7 @@ static int snd_cs4231_capture_prepare(struct snd_pcm_substream *substream) return 0; } -static void snd_cs4231_overrange(struct snd_cs4231 *chip) +void snd_cs4231_overrange(struct snd_cs4231 *chip) { unsigned long flags; unsigned char res; @@ -1054,8 +1090,11 @@ static int snd_cs4231_probe(struct snd_cs4231 *chip) chip->image[CS4231_IFACE_CTRL] = (chip->image[CS4231_IFACE_CTRL] & ~CS4231_SINGLE_DMA) | (chip->single_dma ? CS4231_SINGLE_DMA : 0); - chip->image[CS4231_ALT_FEATURE_1] = 0x80; - chip->image[CS4231_ALT_FEATURE_2] = chip->hardware == CS4231_HW_INTERWAVE ? 0xc2 : 0x01; + if (chip->hardware != CS4231_HW_OPTI93X) { + chip->image[CS4231_ALT_FEATURE_1] = 0x80; + chip->image[CS4231_ALT_FEATURE_2] = + chip->hardware == CS4231_HW_INTERWAVE ? 0xc2 : 0x01; + } ptr = (unsigned char *) &chip->image; snd_cs4231_mce_down(chip); spin_lock_irqsave(&chip->reg_lock, flags); @@ -1376,6 +1415,7 @@ const char *snd_cs4231_chip_id(struct snd_cs4231 *chip) case CS4231_HW_INTERWAVE: return "AMD InterWave"; case CS4231_HW_OPL3SA2: return chip->card->shortname; case CS4231_HW_AD1845: return "AD1845"; + case CS4231_HW_OPTI93X: return "OPTi 93x"; default: return "???"; } } @@ -1401,8 +1441,13 @@ static int snd_cs4231_new(struct snd_card *card, chip->rate_constraint = snd_cs4231_xrate; chip->set_playback_format = snd_cs4231_playback_format; chip->set_capture_format = snd_cs4231_capture_format; - memcpy(&chip->image, &snd_cs4231_original_image, sizeof(snd_cs4231_original_image)); - + if (chip->hardware == CS4231_HW_OPTI93X) + memcpy(&chip->image, &snd_opti93x_original_image, + sizeof(snd_opti93x_original_image)); + else + memcpy(&chip->image, &snd_cs4231_original_image, + sizeof(snd_cs4231_original_image)); + *rchip = chip; return 0; } @@ -1790,6 +1835,48 @@ CS4231_SINGLE("Loopback Capture Switch", 0, CS4231_LOOPBACK, 0, 1, 0), CS4231_SINGLE("Loopback Capture Volume", 0, CS4231_LOOPBACK, 2, 63, 1) }; +static struct snd_kcontrol_new snd_opti93x_controls[] = { +CS4231_DOUBLE("Master Playback Switch", 0, + OPTi93X_OUT_LEFT, OPTi93X_OUT_RIGHT, 7, 7, 1, 1), +CS4231_DOUBLE("Master Playback Volume", 0, + OPTi93X_OUT_LEFT, OPTi93X_OUT_RIGHT, 1, 1, 31, 1), +CS4231_DOUBLE("PCM Playback Switch", 0, + CS4231_LEFT_OUTPUT, CS4231_RIGHT_OUTPUT, 7, 7, 1, 1), +CS4231_DOUBLE("PCM Playback Volume", 0, + CS4231_LEFT_OUTPUT, CS4231_RIGHT_OUTPUT, 0, 0, 31, 1), +CS4231_DOUBLE("FM Playback Switch", 0, + CS4231_AUX2_LEFT_INPUT, CS4231_AUX2_RIGHT_INPUT, 7, 7, 1, 1), +CS4231_DOUBLE("FM Playback Volume", 0, + CS4231_AUX2_LEFT_INPUT, CS4231_AUX2_RIGHT_INPUT, 1, 1, 15, 1), +CS4231_DOUBLE("Line Playback Switch", 0, + CS4231_LEFT_LINE_IN, CS4231_RIGHT_LINE_IN, 7, 7, 1, 1), +CS4231_DOUBLE("Line Playback Volume", 0, + CS4231_LEFT_LINE_IN, CS4231_RIGHT_LINE_IN, 0, 0, 15, 1), +CS4231_DOUBLE("Mic Playback Switch", 0, + OPTi93X_MIC_LEFT_INPUT, OPTi93X_MIC_RIGHT_INPUT, 7, 7, 1, 1), +CS4231_DOUBLE("Mic Playback Volume", 0, + OPTi93X_MIC_LEFT_INPUT, OPTi93X_MIC_RIGHT_INPUT, 1, 1, 15, 1), +CS4231_DOUBLE("Mic Boost", 0, + CS4231_LEFT_INPUT, CS4231_RIGHT_INPUT, 5, 5, 1, 0), +CS4231_DOUBLE("CD Playback Switch", 0, + CS4231_AUX1_LEFT_INPUT, CS4231_AUX1_RIGHT_INPUT, 7, 7, 1, 1), +CS4231_DOUBLE("CD Playback Volume", 0, + CS4231_AUX1_LEFT_INPUT, CS4231_AUX1_RIGHT_INPUT, 1, 1, 15, 1), +CS4231_DOUBLE("Aux Playback Switch", 0, + OPTi931_AUX_LEFT_INPUT, OPTi931_AUX_RIGHT_INPUT, 7, 7, 1, 1), +CS4231_DOUBLE("Aux Playback Volume", 0, + OPTi931_AUX_LEFT_INPUT, OPTi931_AUX_RIGHT_INPUT, 1, 1, 15, 1), +CS4231_DOUBLE("Capture Volume", 0, + CS4231_LEFT_INPUT, CS4231_RIGHT_INPUT, 0, 0, 15, 0), +{ + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "Capture Source", + .info = snd_cs4231_info_mux, + .get = snd_cs4231_get_mux, + .put = snd_cs4231_put_mux, +} +}; + int snd_cs4231_mixer(struct snd_cs4231 *chip) { struct snd_card *card; @@ -1802,10 +1889,22 @@ int snd_cs4231_mixer(struct snd_cs4231 *chip) strcpy(card->mixername, chip->pcm->name); - for (idx = 0; idx < ARRAY_SIZE(snd_cs4231_controls); idx++) { - if ((err = snd_ctl_add(card, snd_ctl_new1(&snd_cs4231_controls[idx], chip))) < 0) - return err; - } + if (chip->hardware == CS4231_HW_OPTI93X) + for (idx = 0; idx < ARRAY_SIZE(snd_opti93x_controls); idx++) { + err = snd_ctl_add(card, + snd_ctl_new1(&snd_opti93x_controls[idx], + chip)); + if (err < 0) + return err; + } + else + for (idx = 0; idx < ARRAY_SIZE(snd_cs4231_controls); idx++) { + err = snd_ctl_add(card, + snd_ctl_new1(&snd_cs4231_controls[idx], + chip)); + if (err < 0) + return err; + } return 0; } @@ -1815,6 +1914,7 @@ EXPORT_SYMBOL(snd_cs4236_ext_out); EXPORT_SYMBOL(snd_cs4236_ext_in); EXPORT_SYMBOL(snd_cs4231_mce_up); EXPORT_SYMBOL(snd_cs4231_mce_down); +EXPORT_SYMBOL(snd_cs4231_overrange); EXPORT_SYMBOL(snd_cs4231_interrupt); EXPORT_SYMBOL(snd_cs4231_chip_id); EXPORT_SYMBOL(snd_cs4231_create); diff --git a/sound/isa/opti9xx/opti92x-ad1848.c b/sound/isa/opti9xx/opti92x-ad1848.c index fe1afc13a01..41c047e665e 100644 --- a/sound/isa/opti9xx/opti92x-ad1848.c +++ b/sound/isa/opti9xx/opti92x-ad1848.c @@ -33,15 +33,10 @@ #include <asm/io.h> #include <asm/dma.h> #include <sound/core.h> -#ifdef CS4231 +#if defined(CS4231) || defined(OPTi93X) #include <sound/cs4231.h> #else -#ifndef OPTi93X #include <sound/ad1848.h> -#else -#include <sound/control.h> -#include <sound/pcm.h> -#endif /* OPTi93X */ #endif /* CS4231 */ #include <sound/mpu401.h> #include <sound/opl3.h> @@ -109,7 +104,6 @@ module_param(dma2, int, 0444); MODULE_PARM_DESC(dma2, "2nd dma # for opti9xx driver."); #endif /* CS4231 || OPTi93X */ -#define OPTi9XX_HW_DETECT 0 #define OPTi9XX_HW_82C928 1 #define OPTi9XX_HW_82C929 2 #define OPTi9XX_HW_82C924 3 @@ -123,105 +117,12 @@ MODULE_PARM_DESC(dma2, "2nd dma # for opti9xx driver."); #ifdef OPTi93X -#define OPTi93X_INDEX 0x00 -#define OPTi93X_DATA 0x01 #define OPTi93X_STATUS 0x02 -#define OPTi93X_DDATA 0x03 #define OPTi93X_PORT(chip, r) ((chip)->port + OPTi93X_##r) -#define OPTi93X_MIXOUT_LEFT 0x00 -#define OPTi93X_MIXOUT_RIGHT 0x01 -#define OPTi93X_CD_LEFT_INPUT 0x02 -#define OPTi93X_CD_RIGHT_INPUT 0x03 -#define OPTi930_AUX_LEFT_INPUT 0x04 -#define OPTi930_AUX_RIGHT_INPUT 0x05 -#define OPTi931_FM_LEFT_INPUT 0x04 -#define OPTi931_FM_RIGHT_INPUT 0x05 -#define OPTi93X_DAC_LEFT 0x06 -#define OPTi93X_DAC_RIGHT 0x07 -#define OPTi93X_PLAY_FORMAT 0x08 -#define OPTi93X_IFACE_CONF 0x09 -#define OPTi93X_PIN_CTRL 0x0a -#define OPTi93X_ERR_INIT 0x0b -#define OPTi93X_ID 0x0c -#define OPTi93X_PLAY_UPR_CNT 0x0e -#define OPTi93X_PLAY_LWR_CNT 0x0f -#define OPTi931_AUX_LEFT_INPUT 0x10 -#define OPTi931_AUX_RIGHT_INPUT 0x11 -#define OPTi93X_LINE_LEFT_INPUT 0x12 -#define OPTi93X_LINE_RIGHT_INPUT 0x13 -#define OPTi93X_MIC_LEFT_INPUT 0x14 -#define OPTi93X_MIC_RIGHT_INPUT 0x15 -#define OPTi93X_OUT_LEFT 0x16 -#define OPTi93X_OUT_RIGHT 0x17 -#define OPTi93X_CAPT_FORMAT 0x1c -#define OPTi93X_CAPT_UPR_CNT 0x1e -#define OPTi93X_CAPT_LWR_CNT 0x1f - -#define OPTi93X_TRD 0x20 -#define OPTi93X_MCE 0x40 -#define OPTi93X_INIT 0x80 - -#define OPTi93X_MIXOUT_MIC_GAIN 0x20 -#define OPTi93X_MIXOUT_LINE 0x00 -#define OPTi93X_MIXOUT_CD 0x40 -#define OPTi93X_MIXOUT_MIC 0x80 -#define OPTi93X_MIXOUT_MIXER 0xc0 - -#define OPTi93X_STEREO 0x10 -#define OPTi93X_LINEAR_8 0x00 -#define OPTi93X_ULAW_8 0x20 -#define OPTi93X_LINEAR_16_LIT 0x40 -#define OPTi93X_ALAW_8 0x60 -#define OPTi93X_ADPCM_16 0xa0 -#define OPTi93X_LINEAR_16_BIG 0xc0 - -#define OPTi93X_CAPTURE_PIO 0x80 -#define OPTi93X_PLAYBACK_PIO 0x40 -#define OPTi93X_AUTOCALIB 0x08 -#define OPTi93X_SINGLE_DMA 0x04 -#define OPTi93X_CAPTURE_ENABLE 0x02 -#define OPTi93X_PLAYBACK_ENABLE 0x01 - -#define OPTi93X_IRQ_ENABLE 0x02 - -#define OPTi93X_DMA_REQUEST 0x10 -#define OPTi93X_CALIB_IN_PROGRESS 0x20 - #define OPTi93X_IRQ_PLAYBACK 0x04 #define OPTi93X_IRQ_CAPTURE 0x08 - -struct snd_opti93x { - unsigned long port; - struct resource *res_port; - int irq; - int dma1; - int dma2; - - struct snd_opti9xx *chip; - unsigned short hardware; - unsigned char image[32]; - - unsigned char mce_bit; - unsigned short mode; - int mute; - - spinlock_t lock; - - struct snd_card *card; - struct snd_pcm *pcm; - struct snd_pcm_substream *playback_substream; - struct snd_pcm_substream *capture_substream; - unsigned int p_dma_size; - unsigned int c_dma_size; -}; - -#define OPTi93X_MODE_NONE 0x00 -#define OPTi93X_MODE_PLAY 0x01 -#define OPTi93X_MODE_CAPTURE 0x02 -#define OPTi93X_MODE_OPEN (OPTi93X_MODE_PLAY | OPTi93X_MODE_CAPTURE) - #endif /* OPTi93X */ struct snd_opti9xx { @@ -234,6 +135,7 @@ struct snd_opti9xx { unsigned long mc_base_size; #ifdef OPTi93X unsigned long mc_indir_index; + struct snd_cs4231 *codec; #endif /* OPTi93X */ unsigned long pwd_reg; @@ -491,16 +393,9 @@ static int __devinit snd_opti9xx_configure(struct snd_opti9xx *chip) break; #else /* OPTi93X */ - case OPTi9XX_HW_82C930: case OPTi9XX_HW_82C931: case OPTi9XX_HW_82C933: - snd_opti9xx_write_mask(chip, OPTi9XX_MC_REG(6), 0x02, 0x03); - snd_opti9xx_write_mask(chip, OPTi9XX_MC_REG(3), 0x00, 0xff); - snd_opti9xx_write_mask(chip, OPTi9XX_MC_REG(4), 0x10 | - (chip->hardware == OPTi9XX_HW_82C930 ? 0x00 : 0x04), - 0x34); - snd_opti9xx_write_mask(chip, OPTi9XX_MC_REG(5), 0x20, 0xbf); - /* + /* * The BTC 1817DW has QS1000 wavetable which is connected * to the serial digital input of the OPTI931. */ @@ -510,6 +405,13 @@ static int __devinit snd_opti9xx_configure(struct snd_opti9xx *chip) * or digital input signal. */ snd_opti9xx_write_mask(chip, OPTi9XX_MC_REG(26), 0x01, 0x01); + case OPTi9XX_HW_82C930: /* FALL THROUGH */ + snd_opti9xx_write_mask(chip, OPTi9XX_MC_REG(6), 0x02, 0x03); + snd_opti9xx_write_mask(chip, OPTi9XX_MC_REG(3), 0x00, 0xff); + snd_opti9xx_write_mask(chip, OPTi9XX_MC_REG(4), 0x10 | + (chip->hardware == OPTi9XX_HW_82C930 ? 0x00 : 0x04), + 0x34); + snd_opti9xx_write_mask(chip, OPTi9XX_MC_REG(5), 0x20, 0xbf); break; #endif /* OPTi93X */ @@ -654,979 +556,23 @@ __skip_mpu: #ifdef OPTi93X -static unsigned char snd_opti93x_default_image[32] = -{ - 0x00, /* 00/00 - l_mixout_outctrl */ - 0x00, /* 01/01 - r_mixout_outctrl */ - 0x88, /* 02/02 - l_cd_inctrl */ - 0x88, /* 03/03 - r_cd_inctrl */ - 0x88, /* 04/04 - l_a1/fm_inctrl */ - 0x88, /* 05/05 - r_a1/fm_inctrl */ - 0x80, /* 06/06 - l_dac_inctrl */ - 0x80, /* 07/07 - r_dac_inctrl */ - 0x00, /* 08/08 - ply_dataform_reg */ - 0x00, /* 09/09 - if_conf */ - 0x00, /* 0a/10 - pin_ctrl */ - 0x00, /* 0b/11 - err_init_reg */ - 0x0a, /* 0c/12 - id_reg */ - 0x00, /* 0d/13 - reserved */ - 0x00, /* 0e/14 - ply_upcount_reg */ - 0x00, /* 0f/15 - ply_lowcount_reg */ - 0x88, /* 10/16 - reserved/l_a1_inctrl */ - 0x88, /* 11/17 - reserved/r_a1_inctrl */ - 0x88, /* 12/18 - l_line_inctrl */ - 0x88, /* 13/19 - r_line_inctrl */ - 0x88, /* 14/20 - l_mic_inctrl */ - 0x88, /* 15/21 - r_mic_inctrl */ - 0x80, /* 16/22 - l_out_outctrl */ - 0x80, /* 17/23 - r_out_outctrl */ - 0x00, /* 18/24 - reserved */ - 0x00, /* 19/25 - reserved */ - 0x00, /* 1a/26 - reserved */ - 0x00, /* 1b/27 - reserved */ - 0x00, /* 1c/28 - cap_dataform_reg */ - 0x00, /* 1d/29 - reserved */ - 0x00, /* 1e/30 - cap_upcount_reg */ - 0x00 /* 1f/31 - cap_lowcount_reg */ -}; - - -static int snd_opti93x_busy_wait(struct snd_opti93x *chip) -{ - int timeout; - - for (timeout = 250; timeout-- > 0; udelay(10)) - if (!(inb(OPTi93X_PORT(chip, INDEX)) & OPTi93X_INIT)) - return 0; - - snd_printk("chip still busy.\n"); - return -EBUSY; -} - -static unsigned char snd_opti93x_in(struct snd_opti93x *chip, unsigned char reg) -{ - snd_opti93x_busy_wait(chip); - outb(chip->mce_bit | (reg & 0x1f), OPTi93X_PORT(chip, INDEX)); - return inb(OPTi93X_PORT(chip, DATA)); -} - -static void snd_opti93x_out(struct snd_opti93x *chip, unsigned char reg, - unsigned char value) -{ - snd_opti93x_busy_wait(chip); - outb(chip->mce_bit | (reg & 0x1f), OPTi93X_PORT(chip, INDEX)); - outb(value, OPTi93X_PORT(chip, DATA)); -} - -static void snd_opti93x_out_image(struct snd_opti93x *chip, unsigned char reg, - unsigned char value) -{ - snd_opti93x_out(chip, reg, chip->image[reg] = value); -} - -static void snd_opti93x_out_mask(struct snd_opti93x *chip, unsigned char reg, - unsigned char mask, unsigned char value) -{ - snd_opti93x_out_image(chip, reg, - (chip->image[reg] & ~mask) | (value & mask)); -} - - -static void snd_opti93x_mce_up(struct snd_opti93x *chip) -{ - snd_opti93x_busy_wait(chip); - - chip->mce_bit = OPTi93X_MCE; - if (!(inb(OPTi93X_PORT(chip, INDEX)) & OPTi93X_MCE)) - outb(chip->mce_bit, OPTi93X_PORT(chip, INDEX)); -} - -static void snd_opti93x_mce_down(struct snd_opti93x *chip) -{ - snd_opti93x_busy_wait(chip); - - chip->mce_bit = 0; - if (inb(OPTi93X_PORT(chip, INDEX)) & OPTi93X_MCE) - outb(chip->mce_bit, OPTi93X_PORT(chip, INDEX)); -} - -#define snd_opti93x_mute_reg(chip, reg, mute) \ - snd_opti93x_out(chip, reg, mute ? 0x80 : chip->image[reg]); - -static void snd_opti93x_mute(struct snd_opti93x *chip, int mute) -{ - mute = mute ? 1 : 0; - if (chip->mute == mute) - return; - - chip->mute = mute; - - snd_opti93x_mute_reg(chip, OPTi93X_CD_LEFT_INPUT, mute); - snd_opti93x_mute_reg(chip, OPTi93X_CD_RIGHT_INPUT, mute); - switch (chip->hardware) { - case OPTi9XX_HW_82C930: - snd_opti93x_mute_reg(chip, OPTi930_AUX_LEFT_INPUT, mute); - snd_opti93x_mute_reg(chip, OPTi930_AUX_RIGHT_INPUT, mute); - break; - case OPTi9XX_HW_82C931: - case OPTi9XX_HW_82C933: - snd_opti93x_mute_reg(chip, OPTi931_FM_LEFT_INPUT, mute); - snd_opti93x_mute_reg(chip, OPTi931_FM_RIGHT_INPUT, mute); - snd_opti93x_mute_reg(chip, OPTi931_AUX_LEFT_INPUT, mute); - snd_opti93x_mute_reg(chip, OPTi931_AUX_RIGHT_INPUT, mute); - } - snd_opti93x_mute_reg(chip, OPTi93X_DAC_LEFT, mute); - snd_opti93x_mute_reg(chip, OPTi93X_DAC_RIGHT, mute); - snd_opti93x_mute_reg(chip, OPTi93X_LINE_LEFT_INPUT, mute); - snd_opti93x_mute_reg(chip, OPTi93X_LINE_RIGHT_INPUT, mute); - snd_opti93x_mute_reg(chip, OPTi93X_MIC_LEFT_INPUT, mute); - snd_opti93x_mute_reg(chip, OPTi93X_MIC_RIGHT_INPUT, mute); - snd_opti93x_mute_reg(chip, OPTi93X_OUT_LEFT, mute); - snd_opti93x_mute_reg(chip, OPTi93X_OUT_RIGHT, mute); -} - - -static unsigned int snd_opti93x_get_count(unsigned char format, - unsigned int size) -{ - switch (format & 0xe0) { - case OPTi93X_LINEAR_16_LIT: - case OPTi93X_LINEAR_16_BIG: - size >>= 1; - break; - case OPTi93X_ADPCM_16: - return size >> 2; - } - return (format & OPTi93X_STEREO) ? (size >> 1) : size; -} - -static unsigned int rates[] = { 5512, 6615, 8000, 9600, 11025, 16000, - 18900, 22050, 27428, 32000, 33075, 37800, - 44100, 48000 }; -#define RATES ARRAY_SIZE(rates) - -static struct snd_pcm_hw_constraint_list hw_constraints_rates = { - .count = RATES, - .list = rates, - .mask = 0, -}; - -static unsigned char bits[] = { 0x01, 0x0f, 0x00, 0x0e, 0x03, 0x02, - 0x05, 0x07, 0x04, 0x06, 0x0d, 0x09, - 0x0b, 0x0c}; - -static unsigned char snd_opti93x_get_freq(unsigned int rate) -{ - unsigned int i; - - for (i = 0; i < RATES; i++) { - if (rate == rates[i]) - return bits[i]; - } - snd_BUG(); - return bits[RATES-1]; -} - -static unsigned char snd_opti93x_get_format(struct snd_opti93x *chip, - unsigned int format, int channels) -{ - unsigned char retval = OPTi93X_LINEAR_8; - - switch (format) { - case SNDRV_PCM_FORMAT_MU_LAW: - retval = OPTi93X_ULAW_8; - break; - case SNDRV_PCM_FORMAT_A_LAW: - retval = OPTi93X_ALAW_8; - break; - case SNDRV_PCM_FORMAT_S16_LE: - retval = OPTi93X_LINEAR_16_LIT; - break; - case SNDRV_PCM_FORMAT_S16_BE: - retval = OPTi93X_LINEAR_16_BIG; - break; - case SNDRV_PCM_FORMAT_IMA_ADPCM: - retval = OPTi93X_ADPCM_16; - } - return (channels > 1) ? (retval | OPTi93X_STEREO) : retval; -} - - -static void snd_opti93x_playback_format(struct snd_opti93x *chip, unsigned char fmt) -{ - unsigned char mask; - - snd_opti93x_mute(chip, 1); - - snd_opti93x_mce_up(chip); - mask = (chip->mode & OPTi93X_MODE_CAPTURE) ? 0xf0 : 0xff; - snd_opti93x_out_mask(chip, OPTi93X_PLAY_FORMAT, mask, fmt); - snd_opti93x_mce_down(chip); - - snd_opti93x_mute(chip, 0); -} - -static void snd_opti93x_capture_format(struct snd_opti93x *chip, unsigned char fmt) -{ - snd_opti93x_mute(chip, 1); - - snd_opti93x_mce_up(chip); - if (!(chip->mode & OPTi93X_MODE_PLAY)) - snd_opti93x_out_mask(chip, OPTi93X_PLAY_FORMAT, 0x0f, fmt); - else - fmt = chip->image[OPTi93X_PLAY_FORMAT] & 0xf0; - snd_opti93x_out_image(chip, OPTi93X_CAPT_FORMAT, fmt); - snd_opti93x_mce_down(chip); - - snd_opti93x_mute(chip, 0); -} - - -static int snd_opti93x_open(struct snd_opti93x *chip, unsigned int mode) -{ - unsigned long flags; - - spin_lock_irqsave(&chip->lock, flags); - - if (chip->mode & mode) { - spin_unlock_irqrestore(&chip->lock, flags); - return -EAGAIN; - } - - if (!(chip->mode & OPTi93X_MODE_OPEN)) { - outb(0x00, OPTi93X_PORT(chip, STATUS)); - snd_opti93x_out_mask(chip, OPTi93X_PIN_CTRL, - OPTi93X_IRQ_ENABLE, OPTi93X_IRQ_ENABLE); - chip->mode = mode; - } - else - chip->mode |= mode; - - spin_unlock_irqrestore(&chip->lock, flags); - return 0; -} - -static void snd_opti93x_close(struct snd_opti93x *chip, unsigned int mode) -{ - unsigned long flags; - - spin_lock_irqsave(&chip->lock, flags); - - chip->mode &= ~mode; - if (chip->mode & OPTi93X_MODE_OPEN) { - spin_unlock_irqrestore(&chip->lock, flags); - return; - } - - snd_opti93x_mute(chip, 1); - - outb(0, OPTi93X_PORT(chip, STATUS)); - snd_opti93x_out_mask(chip, OPTi93X_PIN_CTRL, OPTi93X_IRQ_ENABLE, - ~OPTi93X_IRQ_ENABLE); - - snd_opti93x_mce_up(chip); - snd_opti93x_out_image(chip, OPTi93X_IFACE_CONF, 0x00); - snd_opti93x_mce_down(chip); - chip->mode = 0; - - snd_opti93x_mute(chip, 0); - spin_unlock_irqrestore(&chip->lock, flags); -} - -static int snd_opti93x_trigger(struct snd_pcm_substream *substream, - unsigned char what, int cmd) -{ - struct snd_opti93x *chip = snd_pcm_substream_chip(substream); - - switch (cmd) { - case SNDRV_PCM_TRIGGER_START: - case SNDRV_PCM_TRIGGER_STOP: - { - unsigned int what = 0; - struct snd_pcm_substream *s; - snd_pcm_group_for_each_entry(s, substream) { - if (s == chip->playback_substream) { - what |= OPTi93X_PLAYBACK_ENABLE; - snd_pcm_trigger_done(s, substream); - } else if (s == chip->capture_substream) { - what |= OPTi93X_CAPTURE_ENABLE; - snd_pcm_trigger_done(s, substream); - } - } - spin_lock(&chip->lock); - if (cmd == SNDRV_PCM_TRIGGER_START) { - snd_opti93x_out_mask(chip, OPTi93X_IFACE_CONF, what, what); - if (what & OPTi93X_CAPTURE_ENABLE) - udelay(50); - } else - snd_opti93x_out_mask(chip, OPTi93X_IFACE_CONF, what, 0x00); - spin_unlock(&chip->lock); - break; - } - default: - return -EINVAL; - } - return 0; -} - -static int snd_opti93x_playback_trigger(struct snd_pcm_substream *substream, int cmd) -{ - return snd_opti93x_trigger(substream, - OPTi93X_PLAYBACK_ENABLE, cmd); -} - -static int snd_opti93x_capture_trigger(struct snd_pcm_substream *substream, int cmd) -{ - return snd_opti93x_trigger(substream, - OPTi93X_CAPTURE_ENABLE, cmd); -} - -static int snd_opti93x_hw_params(struct snd_pcm_substream *substream, - struct snd_pcm_hw_params *hw_params) -{ - return snd_pcm_lib_malloc_pages(substream, params_buffer_bytes(hw_params)); -} - - -static int snd_opti93x_hw_free(struct snd_pcm_substream *substream) -{ - snd_pcm_lib_free_pages(substream); - return 0; -} - - -static int snd_opti93x_playback_prepare(struct snd_pcm_substream *substream) -{ - struct snd_opti93x *chip = snd_pcm_substream_chip(substream); - struct snd_pcm_runtime *runtime = substream->runtime; - unsigned long flags; - unsigned char format; - unsigned int count = snd_pcm_lib_period_bytes(substream); - unsigned int size = snd_pcm_lib_buffer_bytes(substream); - - spin_lock_irqsave(&chip->lock, flags); - - chip->p_dma_size = size; - snd_opti93x_out_mask(chip, OPTi93X_IFACE_CONF, - OPTi93X_PLAYBACK_ENABLE | OPTi93X_PLAYBACK_PIO, - ~(OPTi93X_PLAYBACK_ENABLE | OPTi93X_PLAYBACK_PIO)); - - snd_dma_program(chip->dma1, runtime->dma_addr, size, - DMA_MODE_WRITE | DMA_AUTOINIT); - - format = snd_opti93x_get_freq(runtime->rate); - format |= snd_opti93x_get_format(chip, runtime->format, - runtime->channels); - snd_opti93x_playback_format(chip, format); - format = chip->image[OPTi93X_PLAY_FORMAT]; - - count = snd_opti93x_get_count(format, count) - 1; - snd_opti93x_out_image(chip, OPTi93X_PLAY_LWR_CNT, count); - snd_opti93x_out_image(chip, OPTi93X_PLAY_UPR_CNT, count >> 8); - - spin_unlock_irqrestore(&chip->lock, flags); - return 0; -} - -static int snd_opti93x_capture_prepare(struct snd_pcm_substream *substream) -{ - struct snd_opti93x *chip = snd_pcm_substream_chip(substream); - struct snd_pcm_runtime *runtime = substream->runtime; - unsigned long flags; - unsigned char format; - unsigned int count = snd_pcm_lib_period_bytes(substream); - unsigned int size = snd_pcm_lib_buffer_bytes(substream); - - spin_lock_irqsave(&chip->lock, flags); - - chip->c_dma_size = size; - snd_opti93x_out_mask(chip, OPTi93X_IFACE_CONF, - OPTi93X_CAPTURE_ENABLE | OPTi93X_CAPTURE_PIO, 0); - - snd_dma_program(chip->dma2, runtime->dma_addr, size, - DMA_MODE_READ | DMA_AUTOINIT); - - format = snd_opti93x_get_freq(runtime->rate); - format |= snd_opti93x_get_format(chip, runtime->format, - runtime->channels); - snd_opti93x_capture_format(chip, format); - format = chip->image[OPTi93X_CAPT_FORMAT]; - - count = snd_opti93x_get_count(format, count) - 1; - snd_opti93x_out_image(chip, OPTi93X_CAPT_LWR_CNT, count); - snd_opti93x_out_image(chip, OPTi93X_CAPT_UPR_CNT, count >> 8); - - spin_unlock_irqrestore(&chip->lock, flags); - return 0; -} - -static snd_pcm_uframes_t snd_opti93x_playback_pointer(struct snd_pcm_substream *substream) -{ - struct snd_opti93x *chip = snd_pcm_substream_chip(substream); - size_t ptr; - - if (!(chip->image[OPTi93X_IFACE_CONF] & OPTi93X_PLAYBACK_ENABLE)) - return 0; - - ptr = snd_dma_pointer(chip->dma1, chip->p_dma_size); - return bytes_to_frames(substream->runtime, ptr); -} - -static snd_pcm_uframes_t snd_opti93x_capture_pointer(struct snd_pcm_substream *substream) -{ - struct snd_opti93x *chip = snd_pcm_substream_chip(substream); - size_t ptr; - - if (!(chip->image[OPTi93X_IFACE_CONF] & OPTi93X_CAPTURE_ENABLE)) - return 0; - - ptr = snd_dma_pointer(chip->dma2, chip->c_dma_size); - return bytes_to_frames(substream->runtime, ptr); -} - - -static void snd_opti93x_overrange(struct snd_opti93x *chip) -{ - unsigned long flags; - - spin_lock_irqsave(&chip->lock, flags); - - if (snd_opti93x_in(chip, OPTi93X_ERR_INIT) & (0x08 | 0x02)) - chip->capture_substream->runtime->overrange++; - - spin_unlock_irqrestore(&chip->lock, flags); -} - static irqreturn_t snd_opti93x_interrupt(int irq, void *dev_id) { - struct snd_opti93x *codec = dev_id; + struct snd_cs4231 *codec = dev_id; + struct snd_opti9xx *chip = codec->card->private_data; unsigned char status; - status = snd_opti9xx_read(codec->chip, OPTi9XX_MC_REG(11)); + status = snd_opti9xx_read(chip, OPTi9XX_MC_REG(11)); if ((status & OPTi93X_IRQ_PLAYBACK) && codec->playback_substream) snd_pcm_period_elapsed(codec->playback_substream); if ((status & OPTi93X_IRQ_CAPTURE) && codec->capture_substream) { - snd_opti93x_overrange(codec); + snd_cs4231_overrange(codec); snd_pcm_period_elapsed(codec->capture_substream); } outb(0x00, OPTi93X_PORT(codec, STATUS)); return IRQ_HANDLED; } - -static struct snd_pcm_hardware snd_opti93x_playback = { - .info = (SNDRV_PCM_INFO_MMAP | SNDRV_PCM_INFO_INTERLEAVED | - SNDRV_PCM_INFO_MMAP_VALID | SNDRV_PCM_INFO_SYNC_START), - .formats = (SNDRV_PCM_FMTBIT_MU_LAW | SNDRV_PCM_FMTBIT_A_LAW | SNDRV_PCM_FMTBIT_IMA_ADPCM | - SNDRV_PCM_FMTBIT_U8 | SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S16_BE), - .rates = SNDRV_PCM_RATE_KNOT | SNDRV_PCM_RATE_8000_48000, - .rate_min = 5512, - .rate_max = 48000, - .channels_min = 1, - .channels_max = 2, - .buffer_bytes_max = (128*1024), - .period_bytes_min = 64, - .period_bytes_max = (128*1024), - .periods_min = 1, - .periods_max = 1024, - .fifo_size = 0, -}; - -static struct snd_pcm_hardware snd_opti93x_capture = { - .info = (SNDRV_PCM_INFO_MMAP | SNDRV_PCM_INFO_INTERLEAVED | - SNDRV_PCM_INFO_MMAP_VALID | SNDRV_PCM_INFO_SYNC_START), - .formats = (SNDRV_PCM_FMTBIT_MU_LAW | SNDRV_PCM_FMTBIT_A_LAW | SNDRV_PCM_FMTBIT_IMA_ADPCM | - SNDRV_PCM_FMTBIT_U8 | SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S16_BE), - .rates = SNDRV_PCM_RATE_8000_48000, - .rate_min = 5512, - .rate_max = 48000, - .channels_min = 1, - .channels_max = 2, - .buffer_bytes_max = (128*1024), - .period_bytes_min = 64, - .period_bytes_max = (128*1024), - .periods_min = 1, - .periods_max = 1024, - .fifo_size = 0, -}; - -static int snd_opti93x_playback_open(struct snd_pcm_substream *substream) -{ - int error; - struct snd_opti93x *chip = snd_pcm_substream_chip(substream); - struct snd_pcm_runtime *runtime = substream->runtime; - - if ((error = snd_opti93x_open(chip, OPTi93X_MODE_PLAY)) < 0) - return error; - snd_pcm_set_sync(substream); - chip->playback_substream = substream; - runtime->hw = snd_opti93x_playback; - snd_pcm_limit_isa_dma_size(chip->dma1, &runtime->hw.buffer_bytes_max); - snd_pcm_hw_constraint_list(runtime, 0, SNDRV_PCM_HW_PARAM_RATE, &hw_constraints_rates); - return error; -} - -static int snd_opti93x_capture_open(struct snd_pcm_substream *substream) -{ - int error; - struct snd_opti93x *chip = snd_pcm_substream_chip(substream); - struct snd_pcm_runtime *runtime = substream->runtime; - - if ((error = snd_opti93x_open(chip, OPTi93X_MODE_CAPTURE)) < 0) - return error; - runtime->hw = snd_opti93x_capture; - snd_pcm_set_sync(substream); - chip->capture_substream = substream; - snd_pcm_limit_isa_dma_size(chip->dma2, &runtime->hw.buffer_bytes_max); - snd_pcm_hw_constraint_list(runtime, 0, SNDRV_PCM_HW_PARAM_RATE, &hw_constraints_rates); - return error; -} - -static int snd_opti93x_playback_close(struct snd_pcm_substream *substream) -{ - struct snd_opti93x *chip = snd_pcm_substream_chip(substream); - - chip->playback_substream = NULL; - snd_opti93x_close(chip, OPTi93X_MODE_PLAY); - return 0; -} - -static int snd_opti93x_capture_close(struct snd_pcm_substream *substream) -{ - struct snd_opti93x *chip = snd_pcm_substream_chip(substream); - - chip->capture_substream = NULL; - snd_opti93x_close(chip, OPTi93X_MODE_CAPTURE); - return 0; -} - - -static void snd_opti93x_init(struct snd_opti93x *chip) -{ - unsigned long flags; - int i; - - spin_lock_irqsave(&chip->lock, flags); - snd_opti93x_mce_up(chip); - - for (i = 0; i < 32; i++) - snd_opti93x_out_image(chip, i, snd_opti93x_default_image[i]); - - snd_opti93x_mce_down(chip); - spin_unlock_irqrestore(&chip->lock, flags); -} - -static int snd_opti93x_probe(struct snd_opti93x *chip) -{ - unsigned long flags; - unsigned char val; - - spin_lock_irqsave(&chip->lock, flags); - val = snd_opti93x_in(chip, OPTi93X_ID) & 0x0f; - spin_unlock_irqrestore(&chip->lock, flags); - - return (val == 0x0a) ? 0 : -ENODEV; -} - -static int snd_opti93x_free(struct snd_opti93x *chip) -{ - release_and_free_resource(chip->res_port); - if (chip->dma1 >= 0) { - disable_dma(chip->dma1); - free_dma(chip->dma1); - } - if (chip->dma2 >= 0) { - disable_dma(chip->dma2); - free_dma(chip->dma2); - } - if (chip->irq >= 0) { - free_irq(chip->irq, chip); - } - kfree(chip); - return 0; -} - -static int snd_opti93x_dev_free(struct snd_device *device) -{ - struct snd_opti93x *chip = device->device_data; - return snd_opti93x_free(chip); -} - -static const char *snd_opti93x_chip_id(struct snd_opti93x *codec) -{ - switch (codec->hardware) { - case OPTi9XX_HW_82C930: return "82C930"; - case OPTi9XX_HW_82C931: return "82C931"; - case OPTi9XX_HW_82C933: return "82C933"; - default: return "???"; - } -} - -static int snd_opti93x_create(struct snd_card *card, struct snd_opti9xx *chip, - int dma1, int dma2, - struct snd_opti93x **rcodec) -{ - static struct snd_device_ops ops = { - .dev_free = snd_opti93x_dev_free, - }; - int error; - struct snd_opti93x *codec; - - *rcodec = NULL; - codec = kzalloc(sizeof(*codec), GFP_KERNEL); - if (codec == NULL) - return -ENOMEM; - codec->irq = -1; - codec->dma1 = -1; - codec->dma2 = -1; - - if ((codec->res_port = request_region(chip->wss_base + 4, 4, "OPTI93x CODEC")) == NULL) { - snd_printk(KERN_ERR "opti9xx: can't grab port 0x%lx\n", chip->wss_base + 4); - snd_opti93x_free(codec); - return -EBUSY; - } - if (request_dma(dma1, "OPTI93x - 1")) { - snd_printk(KERN_ERR "opti9xx: can't grab DMA1 %d\n", dma1); - snd_opti93x_free(codec); - return -EBUSY; - } - codec->dma1 = chip->dma1; - if (request_dma(dma2, "OPTI93x - 2")) { - snd_printk(KERN_ERR "opti9xx: can't grab DMA2 %d\n", dma2); - snd_opti93x_free(codec); - return -EBUSY; - } - codec->dma2 = chip->dma2; - - if (request_irq(chip->irq, snd_opti93x_interrupt, IRQF_DISABLED, DEV_NAME" - WSS", codec)) { - snd_printk(KERN_ERR "opti9xx: can't grab IRQ %d\n", chip->irq); - snd_opti93x_free(codec); - return -EBUSY; - } - - codec->card = card; - codec->port = chip->wss_base + 4; - codec->irq = chip->irq; - - spin_lock_init(&codec->lock); - codec->hardware = chip->hardware; - codec->chip = chip; - - if ((error = snd_opti93x_probe(codec))) { - snd_opti93x_free(codec); - return error; - } - - snd_opti93x_init(codec); - - /* Register device */ - if ((error = snd_device_new(card, SNDRV_DEV_LOWLEVEL, codec, &ops)) < 0) { - snd_opti93x_free(codec); - return error; - } - - *rcodec = codec; - return 0; -} - -static struct snd_pcm_ops snd_opti93x_playback_ops = { - .open = snd_opti93x_playback_open, - .close = snd_opti93x_playback_close, - .ioctl = snd_pcm_lib_ioctl, - .hw_params = snd_opti93x_hw_params, - .hw_free = snd_opti93x_hw_free, - .prepare = snd_opti93x_playback_prepare, - .trigger = snd_opti93x_playback_trigger, - .pointer = snd_opti93x_playback_pointer, -}; - -static struct snd_pcm_ops snd_opti93x_capture_ops = { - .open = snd_opti93x_capture_open, - .close = snd_opti93x_capture_close, - .ioctl = snd_pcm_lib_ioctl, - .hw_params = snd_opti93x_hw_params, - .hw_free = snd_opti93x_hw_free, - .prepare = snd_opti93x_capture_prepare, - .trigger = snd_opti93x_capture_trigger, - .pointer = snd_opti93x_capture_pointer, -}; - -static int snd_opti93x_pcm(struct snd_opti93x *codec, int device, struct snd_pcm **rpcm) -{ - int error; - struct snd_pcm *pcm; - - if ((error = snd_pcm_new(codec->card, "OPTi 82C93X", device, 1, 1, &pcm)) < 0) - return error; - - snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_PLAYBACK, &snd_opti93x_playback_ops); - snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_CAPTURE, &snd_opti93x_capture_ops); - - pcm->private_data = codec; - pcm->info_flags = SNDRV_PCM_INFO_JOINT_DUPLEX; - - strcpy(pcm->name, snd_opti93x_chip_id(codec)); - - snd_pcm_lib_preallocate_pages_for_all(pcm, SNDRV_DMA_TYPE_DEV, - snd_dma_isa_data(), - 64*1024, codec->dma1 > 3 || codec->dma2 > 3 ? 128*1024 : 64*1024); - - codec->pcm = pcm; - if (rpcm) - *rpcm = pcm; - return 0; -} - -/* - * MIXER part - */ - -static int snd_opti93x_info_mux(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo) -{ - static char *texts[4] = { - "Line1", "Aux", "Mic", "Mix" - }; - - uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED; - uinfo->count = 2; - uinfo->value.enumerated.items = 4; - if (uinfo->value.enumerated.item > 3) - uinfo->value.enumerated.item = 3; - strcpy(uinfo->value.enumerated.name, texts[uinfo->value.enumerated.item]); - return 0; -} - -static int snd_opti93x_get_mux(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) -{ - struct snd_opti93x *chip = snd_kcontrol_chip(kcontrol); - unsigned long flags; - - spin_lock_irqsave(&chip->lock, flags); - ucontrol->value.enumerated.item[0] = (chip->image[OPTi93X_MIXOUT_LEFT] & OPTi93X_MIXOUT_MIXER) >> 6; - ucontrol->value.enumerated.item[1] = (chip->image[OPTi93X_MIXOUT_RIGHT] & OPTi93X_MIXOUT_MIXER) >> 6; - spin_unlock_irqrestore(&chip->lock, flags); - return 0; -} - -static int snd_opti93x_put_mux(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) -{ - struct snd_opti93x *chip = snd_kcontrol_chip(kcontrol); - unsigned long flags; - unsigned short left, right; - int change; - - if (ucontrol->value.enumerated.item[0] > 3 || - ucontrol->value.enumerated.item[1] > 3) - return -EINVAL; - left = ucontrol->value.enumerated.item[0] << 6; - right = ucontrol->value.enumerated.item[1] << 6; - spin_lock_irqsave(&chip->lock, flags); - left = (chip->image[OPTi93X_MIXOUT_LEFT] & ~OPTi93X_MIXOUT_MIXER) | left; - right = (chip->image[OPTi93X_MIXOUT_RIGHT] & ~OPTi93X_MIXOUT_MIXER) | right; - change = left != chip->image[OPTi93X_MIXOUT_LEFT] || - right != chip->image[OPTi93X_MIXOUT_RIGHT]; - snd_opti93x_out_image(chip, OPTi93X_MIXOUT_LEFT, left); - snd_opti93x_out_image(chip, OPTi93X_MIXOUT_RIGHT, right); - spin_unlock_irqrestore(&chip->lock, flags); - return change; -} - -#if 0 - -#define OPTi93X_SINGLE(xname, xindex, reg, shift, mask, invert) \ -{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, .index = xindex, \ - .info = snd_opti93x_info_single, \ - .get = snd_opti93x_get_single, .put = snd_opti93x_put_single, \ - .private_value = reg | (shift << 8) | (mask << 16) | (invert << 24) } - -static int snd_opti93x_info_single(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo) -{ - int mask = (kcontrol->private_value >> 16) & 0xff; - - uinfo->type = mask == 1 ? SNDRV_CTL_ELEM_TYPE_BOOLEAN : SNDRV_CTL_ELEM_TYPE_INTEGER; - uinfo->count = 1; - uinfo->value.integer.min = 0; - uinfo->value.integer.max = mask; - return 0; -} - -static int snd_opti93x_get_single(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) -{ - struct snd_opti93x *chip = snd_kcontrol_chip(kcontrol); - unsigned long flags; - int reg = kcontrol->private_value & 0xff; - int shift = (kcontrol->private_value >> 8) & 0xff; - int mask = (kcontrol->private_value >> 16) & 0xff; - int invert = (kcontrol->private_value >> 24) & 0xff; - - spin_lock_irqsave(&chip->lock, flags); - ucontrol->value.integer.value[0] = (chip->image[reg] >> shift) & mask; - spin_unlock_irqrestore(&chip->lock, flags); - if (invert) - ucontrol->value.integer.value[0] = mask - ucontrol->value.integer.value[0]; - return 0; -} - -static int snd_opti93x_put_single(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) -{ - struct snd_opti93x *chip = snd_kcontrol_chip(kcontrol); - unsigned long flags; - int reg = kcontrol->private_value & 0xff; - int shift = (kcontrol->private_value >> 8) & 0xff; - int mask = (kcontrol->private_value >> 16) & 0xff; - int invert = (kcontrol->private_value >> 24) & 0xff; - int change; - unsigned short val; - - val = (ucontrol->value.integer.value[0] & mask); - if (invert) - val = mask - val; - val <<= shift; - spin_lock_irqsave(&chip->lock, flags); - val = (chip->image[reg] & ~(mask << shift)) | val; - change = val != chip->image[reg]; - snd_opti93x_out(chip, reg, val); - spin_unlock_irqrestore(&chip->lock, flags); - return change; -} - -#endif /* single */ - -#define OPTi93X_DOUBLE(xname, xindex, left_reg, right_reg, shift_left, shift_right, mask, invert) \ -{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, .index = xindex, \ - .info = snd_opti93x_info_double, \ - .get = snd_opti93x_get_double, .put = snd_opti93x_put_double, \ - .private_value = left_reg | (right_reg << 8) | (shift_left << 16) | (shift_right << 19) | (mask << 24) | (invert << 22) } - -#define OPTi93X_DOUBLE_INVERT_INVERT(xctl) \ - do { xctl.private_value ^= 22; } while (0) -#define OPTi93X_DOUBLE_CHANGE_REGS(xctl, left_reg, right_reg) \ - do { xctl.private_value &= ~0x0000ffff; \ - xctl.private_value |= left_reg | (right_reg << 8); } while (0) - -static int snd_opti93x_info_double(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo) -{ - int mask = (kcontrol->private_value >> 24) & 0xff; - - uinfo->type = mask == 1 ? SNDRV_CTL_ELEM_TYPE_BOOLEAN : SNDRV_CTL_ELEM_TYPE_INTEGER; - uinfo->count = 2; - uinfo->value.integer.min = 0; - uinfo->value.integer.max = mask; - return 0; -} - -static int snd_opti93x_get_double(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) -{ - struct snd_opti93x *chip = snd_kcontrol_chip(kcontrol); - unsigned long flags; - int left_reg = kcontrol->private_value & 0xff; - int right_reg = (kcontrol->private_value >> 8) & 0xff; - int shift_left = (kcontrol->private_value >> 16) & 0x07; - int shift_right = (kcontrol->private_value >> 19) & 0x07; - int mask = (kcontrol->private_value >> 24) & 0xff; - int invert = (kcontrol->private_value >> 22) & 1; - - spin_lock_irqsave(&chip->lock, flags); - ucontrol->value.integer.value[0] = (chip->image[left_reg] >> shift_left) & mask; - ucontrol->value.integer.value[1] = (chip->image[right_reg] >> shift_right) & mask; - spin_unlock_irqrestore(&chip->lock, flags); - if (invert) { - ucontrol->value.integer.value[0] = mask - ucontrol->value.integer.value[0]; - ucontrol->value.integer.value[1] = mask - ucontrol->value.integer.value[1]; - } - return 0; -} - -static int snd_opti93x_put_double(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) -{ - struct snd_opti93x *chip = snd_kcontrol_chip(kcontrol); - unsigned long flags; - int left_reg = kcontrol->private_value & 0xff; - int right_reg = (kcontrol->private_value >> 8) & 0xff; - int shift_left = (kcontrol->private_value >> 16) & 0x07; - int shift_right = (kcontrol->private_value >> 19) & 0x07; - int mask = (kcontrol->private_value >> 24) & 0xff; - int invert = (kcontrol->private_value >> 22) & 1; - int change; - unsigned short val1, val2; - - val1 = ucontrol->value.integer.value[0] & mask; - val2 = ucontrol->value.integer.value[1] & mask; - if (invert) { - val1 = mask - val1; - val2 = mask - val2; - } - val1 <<= shift_left; - val2 <<= shift_right; - spin_lock_irqsave(&chip->lock, flags); - val1 = (chip->image[left_reg] & ~(mask << shift_left)) | val1; - val2 = (chip->image[right_reg] & ~(mask << shift_right)) | val2; - change = val1 != chip->image[left_reg] || val2 != chip->image[right_reg]; - snd_opti93x_out_image(chip, left_reg, val1); - snd_opti93x_out_image(chip, right_reg, val2); - spin_unlock_irqrestore(&chip->lock, flags); - return change; -} - -static struct snd_kcontrol_new snd_opti93x_controls[] __devinitdata = { -OPTi93X_DOUBLE("Master Playback Switch", 0, OPTi93X_OUT_LEFT, OPTi93X_OUT_RIGHT, 7, 7, 1, 1), -OPTi93X_DOUBLE("Master Playback Volume", 0, OPTi93X_OUT_LEFT, OPTi93X_OUT_RIGHT, 1, 1, 31, 1), -OPTi93X_DOUBLE("PCM Playback Switch", 0, OPTi93X_DAC_LEFT, OPTi93X_DAC_RIGHT, 7, 7, 1, 1), -OPTi93X_DOUBLE("PCM Playback Volume", 0, OPTi93X_DAC_LEFT, OPTi93X_DAC_RIGHT, 0, 0, 31, 1), -OPTi93X_DOUBLE("FM Playback Switch", 0, OPTi931_FM_LEFT_INPUT, OPTi931_FM_RIGHT_INPUT, 7, 7, 1, 1), -OPTi93X_DOUBLE("FM Playback Volume", 0, OPTi931_FM_LEFT_INPUT, OPTi931_FM_RIGHT_INPUT, 1, 1, 15, 1), -OPTi93X_DOUBLE("Line Playback Switch", 0, OPTi93X_LINE_LEFT_INPUT, OPTi93X_LINE_RIGHT_INPUT, 7, 7, 1, 1), -OPTi93X_DOUBLE("Line Playback Volume", 0, OPTi93X_LINE_LEFT_INPUT, OPTi93X_LINE_RIGHT_INPUT, 1, 1, 15, 1), -OPTi93X_DOUBLE("Mic Playback Switch", 0, OPTi93X_MIC_LEFT_INPUT, OPTi93X_MIC_RIGHT_INPUT, 7, 7, 1, 1), -OPTi93X_DOUBLE("Mic Playback Volume", 0, OPTi93X_MIC_LEFT_INPUT, OPTi93X_MIC_RIGHT_INPUT, 1, 1, 15, 1), -OPTi93X_DOUBLE("Mic Boost", 0, OPTi93X_MIXOUT_LEFT, OPTi93X_MIXOUT_RIGHT, 5, 5, 1, 1), -OPTi93X_DOUBLE("CD Playback Switch", 0, OPTi93X_CD_LEFT_INPUT, OPTi93X_CD_RIGHT_INPUT, 7, 7, 1, 1), -OPTi93X_DOUBLE("CD Playback Volume", 0, OPTi93X_CD_LEFT_INPUT, OPTi93X_CD_RIGHT_INPUT, 1, 1, 15, 1), -OPTi93X_DOUBLE("Aux Playback Switch", 0, OPTi931_AUX_LEFT_INPUT, OPTi931_AUX_RIGHT_INPUT, 7, 7, 1, 1), -OPTi93X_DOUBLE("Aux Playback Volume", 0, OPTi931_AUX_LEFT_INPUT, OPTi931_AUX_RIGHT_INPUT, 1, 1, 15, 1), -OPTi93X_DOUBLE("Capture Volume", 0, OPTi93X_MIXOUT_LEFT, OPTi93X_MIXOUT_RIGHT, 0, 0, 15, 0), -{ - .iface = SNDRV_CTL_ELEM_IFACE_MIXER, - .name = "Capture Source", - .info = snd_opti93x_info_mux, - .get = snd_opti93x_get_mux, - .put = snd_opti93x_put_mux, -} -}; - -static int __devinit snd_opti93x_mixer(struct snd_opti93x *chip) -{ - struct snd_card *card; - struct snd_kcontrol_new knew; - int err; - unsigned int idx; - - snd_assert(chip != NULL && chip->card != NULL, return -EINVAL); - - card = chip->card; - - strcpy(card->mixername, snd_opti93x_chip_id(chip)); - - for (idx = 0; idx < ARRAY_SIZE(snd_opti93x_controls); idx++) { - knew = snd_opti93x_controls[idx]; - if (chip->hardware == OPTi9XX_HW_82C930) { - if (strstr(knew.name, "FM")) /* skip FM controls */ - continue; - else if (strcmp(knew.name, "Mic Playback Volume")) - OPTi93X_DOUBLE_INVERT_INVERT(knew); - else if (strstr(knew.name, "Aux")) - OPTi93X_DOUBLE_CHANGE_REGS(knew, OPTi930_AUX_LEFT_INPUT, OPTi930_AUX_RIGHT_INPUT); - else if (strcmp(knew.name, "PCM Playback Volume")) - OPTi93X_DOUBLE_INVERT_INVERT(knew); - else if (strcmp(knew.name, "Master Playback Volume")) - OPTi93X_DOUBLE_INVERT_INVERT(knew); - } - if ((err = snd_ctl_add(card, snd_ctl_new1(&snd_opti93x_controls[idx], chip))) < 0) - return err; - } - return 0; -} - #endif /* OPTi93X */ static int __devinit snd_card_opti9xx_detect(struct snd_card *card, @@ -1739,8 +685,16 @@ static void snd_card_opti9xx_free(struct snd_card *card) { struct snd_opti9xx *chip = card->private_data; - if (chip) + if (chip) { +#ifdef OPTi93X + struct snd_cs4231 *codec = chip->codec; + if (codec->irq > 0) { + disable_irq(codec->irq); + free_irq(codec->irq, codec); + } +#endif release_and_free_resource(chip->res_mc_base); + } } static int __devinit snd_opti9xx_probe(struct snd_card *card) @@ -1748,11 +702,11 @@ static int __devinit snd_opti9xx_probe(struct snd_card *card) static long possible_ports[] = {0x530, 0xe80, 0xf40, 0x604, -1}; int error; struct snd_opti9xx *chip = card->private_data; -#if defined(OPTi93X) - struct snd_opti93x *codec; -#elif defined(CS4231) +#if defined(CS4231) || defined(OPTi93X) struct snd_cs4231 *codec; +#ifdef CS4231 struct snd_timer *timer; +#endif #else struct snd_ad1848 *codec; #endif @@ -1784,26 +738,34 @@ static int __devinit snd_opti9xx_probe(struct snd_card *card) if ((error = snd_opti9xx_configure(chip))) return error; -#if defined(OPTi93X) - if ((error = snd_opti93x_create(card, chip, chip->dma1, chip->dma2, &codec))) - return error; - if ((error = snd_opti93x_pcm(codec, 0, &pcm)) < 0) - return error; - if ((error = snd_opti93x_mixer(codec)) < 0) - return error; -#elif defined(CS4231) +#if defined(CS4231) || defined(OPTi93X) if ((error = snd_cs4231_create(card, chip->wss_base + 4, -1, chip->irq, chip->dma1, chip->dma2, - CS4231_HW_DETECT, - 0, +#ifdef CS4231 + CS4231_HW_DETECT, 0, +#else /* OPTi93x */ + CS4231_HW_OPTI93X, CS4231_HWSHARE_IRQ, +#endif &codec)) < 0) return error; +#ifdef OPTi93X + chip->codec = codec; +#endif if ((error = snd_cs4231_pcm(codec, 0, &pcm)) < 0) return error; if ((error = snd_cs4231_mixer(codec)) < 0) return error; +#ifdef CS4231 if ((error = snd_cs4231_timer(codec, 0, &timer)) < 0) return error; +#else /* OPTI93X */ + error = request_irq(chip->irq, snd_opti93x_interrupt, + IRQF_DISABLED, DEV_NAME" - WSS", codec); + if (error < 0) { + snd_printk(KERN_ERR "opti9xx: can't grab IRQ %d\n", chip->irq); + return error; + } +#endif #else if ((error = snd_ad1848_create(card, chip->wss_base + 4, chip->irq, chip->dma1, diff --git a/sound/isa/sb/Makefile b/sound/isa/sb/Makefile index c9d1c986d70..1098a56b2f4 100644 --- a/sound/isa/sb/Makefile +++ b/sound/isa/sb/Makefile @@ -34,5 +34,3 @@ ifeq ($(CONFIG_SND_SB16_CSP),y) obj-$(CONFIG_SND_SBAWE) += snd-sb16-csp.o endif obj-$(call sequencer,$(CONFIG_SND_SBAWE)) += snd-emu8000-synth.o - -obj-m := $(sort $(obj-m)) diff --git a/sound/isa/wavefront/wavefront_synth.c b/sound/isa/wavefront/wavefront_synth.c index 95eeca16335..0bb9b925660 100644 --- a/sound/isa/wavefront/wavefront_synth.c +++ b/sound/isa/wavefront/wavefront_synth.c @@ -1939,7 +1939,7 @@ static int __devinit wavefront_download_firmware (snd_wavefront_t *dev, char *path) { - unsigned char *buf; + const unsigned char *buf; int len, err; int section_cnt_downloaded = 0; const struct firmware *firmware; diff --git a/sound/mips/Kconfig b/sound/mips/Kconfig index 531f8ba96a7..a9823fad85c 100644 --- a/sound/mips/Kconfig +++ b/sound/mips/Kconfig @@ -1,15 +1,34 @@ # ALSA MIPS drivers -menu "ALSA MIPS devices" - depends on SND!=n && MIPS +menuconfig SND_MIPS + bool "MIPS sound devices" + depends on MIPS + default y + help + Support for sound devices of MIPS architectures. + +if SND_MIPS + +config SND_SGI_O2 + tristate "SGI O2 Audio" + depends on SGI_IP32 + help + Sound support for the SGI O2 Workstation. + +config SND_SGI_HAL2 + tristate "SGI HAL2 Audio" + depends on SGI_HAS_HAL2 + help + Sound support for the SGI Indy and Indigo2 Workstation. + config SND_AU1X00 tristate "Au1x00 AC97 Port Driver" - depends on (SOC_AU1000 || SOC_AU1100 || SOC_AU1500) && SND + depends on SOC_AU1000 || SOC_AU1100 || SOC_AU1500 select SND_PCM select SND_AC97_CODEC help ALSA Sound driver for the Au1x00's AC97 port. -endmenu +endif # SND_MIPS diff --git a/sound/mips/Makefile b/sound/mips/Makefile index 47afed971fb..861ec0a574b 100644 --- a/sound/mips/Makefile +++ b/sound/mips/Makefile @@ -3,6 +3,10 @@ # snd-au1x00-objs := au1x00.o +snd-sgi-o2-objs := sgio2audio.o ad1843.o +snd-sgi-hal2-objs := hal2.o # Toplevel Module Dependency obj-$(CONFIG_SND_AU1X00) += snd-au1x00.o +obj-$(CONFIG_SND_SGI_O2) += snd-sgi-o2.o +obj-$(CONFIG_SND_SGI_HAL2) += snd-sgi-hal2.o diff --git a/sound/mips/ad1843.c b/sound/mips/ad1843.c new file mode 100644 index 00000000000..c624510ec37 --- /dev/null +++ b/sound/mips/ad1843.c @@ -0,0 +1,561 @@ +/* + * AD1843 low level driver + * + * Copyright 2003 Vivien Chappelier <vivien.chappelier@linux-mips.org> + * Copyright 2008 Thomas Bogendoerfer <tsbogend@alpha.franken.de> + * + * inspired from vwsnd.c (SGI VW audio driver) + * Copyright 1999 Silicon Graphics, Inc. All rights reserved. + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License as published by + * the Free Software Foundation; either version 2 of the License, or + * (at your option) any later version. + * + * This program is distributed in the hope that it will be useful, + * but WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the + * GNU General Public License for more details. + * + * You should have received a copy of the GNU General Public License + * along with this program; if not, write to the Free Software + * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA + * + */ + +#include <linux/init.h> +#include <linux/sched.h> +#include <linux/errno.h> +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/ad1843.h> + +/* + * AD1843 bitfield definitions. All are named as in the AD1843 data + * sheet, with ad1843_ prepended and individual bit numbers removed. + * + * E.g., bits LSS0 through LSS2 become ad1843_LSS. + * + * Only the bitfields we need are defined. + */ + +struct ad1843_bitfield { + char reg; + char lo_bit; + char nbits; +}; + +static const struct ad1843_bitfield + ad1843_PDNO = { 0, 14, 1 }, /* Converter Power-Down Flag */ + ad1843_INIT = { 0, 15, 1 }, /* Clock Initialization Flag */ + ad1843_RIG = { 2, 0, 4 }, /* Right ADC Input Gain */ + ad1843_RMGE = { 2, 4, 1 }, /* Right ADC Mic Gain Enable */ + ad1843_RSS = { 2, 5, 3 }, /* Right ADC Source Select */ + ad1843_LIG = { 2, 8, 4 }, /* Left ADC Input Gain */ + ad1843_LMGE = { 2, 12, 1 }, /* Left ADC Mic Gain Enable */ + ad1843_LSS = { 2, 13, 3 }, /* Left ADC Source Select */ + ad1843_RD2M = { 3, 0, 5 }, /* Right DAC 2 Mix Gain/Atten */ + ad1843_RD2MM = { 3, 7, 1 }, /* Right DAC 2 Mix Mute */ + ad1843_LD2M = { 3, 8, 5 }, /* Left DAC 2 Mix Gain/Atten */ + ad1843_LD2MM = { 3, 15, 1 }, /* Left DAC 2 Mix Mute */ + ad1843_RX1M = { 4, 0, 5 }, /* Right Aux 1 Mix Gain/Atten */ + ad1843_RX1MM = { 4, 7, 1 }, /* Right Aux 1 Mix Mute */ + ad1843_LX1M = { 4, 8, 5 }, /* Left Aux 1 Mix Gain/Atten */ + ad1843_LX1MM = { 4, 15, 1 }, /* Left Aux 1 Mix Mute */ + ad1843_RX2M = { 5, 0, 5 }, /* Right Aux 2 Mix Gain/Atten */ + ad1843_RX2MM = { 5, 7, 1 }, /* Right Aux 2 Mix Mute */ + ad1843_LX2M = { 5, 8, 5 }, /* Left Aux 2 Mix Gain/Atten */ + ad1843_LX2MM = { 5, 15, 1 }, /* Left Aux 2 Mix Mute */ + ad1843_RMCM = { 7, 0, 5 }, /* Right Mic Mix Gain/Atten */ + ad1843_RMCMM = { 7, 7, 1 }, /* Right Mic Mix Mute */ + ad1843_LMCM = { 7, 8, 5 }, /* Left Mic Mix Gain/Atten */ + ad1843_LMCMM = { 7, 15, 1 }, /* Left Mic Mix Mute */ + ad1843_HPOS = { 8, 4, 1 }, /* Headphone Output Voltage Swing */ + ad1843_HPOM = { 8, 5, 1 }, /* Headphone Output Mute */ + ad1843_MPOM = { 8, 6, 1 }, /* Mono Output Mute */ + ad1843_RDA1G = { 9, 0, 6 }, /* Right DAC1 Analog/Digital Gain */ + ad1843_RDA1GM = { 9, 7, 1 }, /* Right DAC1 Analog Mute */ + ad1843_LDA1G = { 9, 8, 6 }, /* Left DAC1 Analog/Digital Gain */ + ad1843_LDA1GM = { 9, 15, 1 }, /* Left DAC1 Analog Mute */ + ad1843_RDA2G = { 10, 0, 6 }, /* Right DAC2 Analog/Digital Gain */ + ad1843_RDA2GM = { 10, 7, 1 }, /* Right DAC2 Analog Mute */ + ad1843_LDA2G = { 10, 8, 6 }, /* Left DAC2 Analog/Digital Gain */ + ad1843_LDA2GM = { 10, 15, 1 }, /* Left DAC2 Analog Mute */ + ad1843_RDA1AM = { 11, 7, 1 }, /* Right DAC1 Digital Mute */ + ad1843_LDA1AM = { 11, 15, 1 }, /* Left DAC1 Digital Mute */ + ad1843_RDA2AM = { 12, 7, 1 }, /* Right DAC2 Digital Mute */ + ad1843_LDA2AM = { 12, 15, 1 }, /* Left DAC2 Digital Mute */ + ad1843_ADLC = { 15, 0, 2 }, /* ADC Left Sample Rate Source */ + ad1843_ADRC = { 15, 2, 2 }, /* ADC Right Sample Rate Source */ + ad1843_DA1C = { 15, 8, 2 }, /* DAC1 Sample Rate Source */ + ad1843_DA2C = { 15, 10, 2 }, /* DAC2 Sample Rate Source */ + ad1843_C1C = { 17, 0, 16 }, /* Clock 1 Sample Rate Select */ + ad1843_C2C = { 20, 0, 16 }, /* Clock 2 Sample Rate Select */ + ad1843_C3C = { 23, 0, 16 }, /* Clock 3 Sample Rate Select */ + ad1843_DAADL = { 25, 4, 2 }, /* Digital ADC Left Source Select */ + ad1843_DAADR = { 25, 6, 2 }, /* Digital ADC Right Source Select */ + ad1843_DAMIX = { 25, 14, 1 }, /* DAC Digital Mix Enable */ + ad1843_DRSFLT = { 25, 15, 1 }, /* Digital Reampler Filter Mode */ + ad1843_ADLF = { 26, 0, 2 }, /* ADC Left Channel Data Format */ + ad1843_ADRF = { 26, 2, 2 }, /* ADC Right Channel Data Format */ + ad1843_ADTLK = { 26, 4, 1 }, /* ADC Transmit Lock Mode Select */ + ad1843_SCF = { 26, 7, 1 }, /* SCLK Frequency Select */ + ad1843_DA1F = { 26, 8, 2 }, /* DAC1 Data Format Select */ + ad1843_DA2F = { 26, 10, 2 }, /* DAC2 Data Format Select */ + ad1843_DA1SM = { 26, 14, 1 }, /* DAC1 Stereo/Mono Mode Select */ + ad1843_DA2SM = { 26, 15, 1 }, /* DAC2 Stereo/Mono Mode Select */ + ad1843_ADLEN = { 27, 0, 1 }, /* ADC Left Channel Enable */ + ad1843_ADREN = { 27, 1, 1 }, /* ADC Right Channel Enable */ + ad1843_AAMEN = { 27, 4, 1 }, /* Analog to Analog Mix Enable */ + ad1843_ANAEN = { 27, 7, 1 }, /* Analog Channel Enable */ + ad1843_DA1EN = { 27, 8, 1 }, /* DAC1 Enable */ + ad1843_DA2EN = { 27, 9, 1 }, /* DAC2 Enable */ + ad1843_DDMEN = { 27, 12, 1 }, /* DAC2 to DAC1 Mix Enable */ + ad1843_C1EN = { 28, 11, 1 }, /* Clock Generator 1 Enable */ + ad1843_C2EN = { 28, 12, 1 }, /* Clock Generator 2 Enable */ + ad1843_C3EN = { 28, 13, 1 }, /* Clock Generator 3 Enable */ + ad1843_PDNI = { 28, 15, 1 }; /* Converter Power Down */ + +/* + * The various registers of the AD1843 use three different formats for + * specifying gain. The ad1843_gain structure parameterizes the + * formats. + */ + +struct ad1843_gain { + int negative; /* nonzero if gain is negative. */ + const struct ad1843_bitfield *lfield; + const struct ad1843_bitfield *rfield; + const struct ad1843_bitfield *lmute; + const struct ad1843_bitfield *rmute; +}; + +static const struct ad1843_gain ad1843_gain_RECLEV = { + .negative = 0, + .lfield = &ad1843_LIG, + .rfield = &ad1843_RIG +}; +static const struct ad1843_gain ad1843_gain_LINE = { + .negative = 1, + .lfield = &ad1843_LX1M, + .rfield = &ad1843_RX1M, + .lmute = &ad1843_LX1MM, + .rmute = &ad1843_RX1MM +}; +static const struct ad1843_gain ad1843_gain_LINE_2 = { + .negative = 1, + .lfield = &ad1843_LDA2G, + .rfield = &ad1843_RDA2G, + .lmute = &ad1843_LDA2GM, + .rmute = &ad1843_RDA2GM +}; +static const struct ad1843_gain ad1843_gain_MIC = { + .negative = 1, + .lfield = &ad1843_LMCM, + .rfield = &ad1843_RMCM, + .lmute = &ad1843_LMCMM, + .rmute = &ad1843_RMCMM +}; +static const struct ad1843_gain ad1843_gain_PCM_0 = { + .negative = 1, + .lfield = &ad1843_LDA1G, + .rfield = &ad1843_RDA1G, + .lmute = &ad1843_LDA1GM, + .rmute = &ad1843_RDA1GM +}; +static const struct ad1843_gain ad1843_gain_PCM_1 = { + .negative = 1, + .lfield = &ad1843_LD2M, + .rfield = &ad1843_RD2M, + .lmute = &ad1843_LD2MM, + .rmute = &ad1843_RD2MM +}; + +static const struct ad1843_gain *ad1843_gain[AD1843_GAIN_SIZE] = +{ + &ad1843_gain_RECLEV, + &ad1843_gain_LINE, + &ad1843_gain_LINE_2, + &ad1843_gain_MIC, + &ad1843_gain_PCM_0, + &ad1843_gain_PCM_1, +}; + +/* read the current value of an AD1843 bitfield. */ + +static int ad1843_read_bits(struct snd_ad1843 *ad1843, + const struct ad1843_bitfield *field) +{ + int w; + + w = ad1843->read(ad1843->chip, field->reg); + return w >> field->lo_bit & ((1 << field->nbits) - 1); +} + +/* + * write a new value to an AD1843 bitfield and return the old value. + */ + +static int ad1843_write_bits(struct snd_ad1843 *ad1843, + const struct ad1843_bitfield *field, + int newval) +{ + int w, mask, oldval, newbits; + + w = ad1843->read(ad1843->chip, field->reg); + mask = ((1 << field->nbits) - 1) << field->lo_bit; + oldval = (w & mask) >> field->lo_bit; + newbits = (newval << field->lo_bit) & mask; + w = (w & ~mask) | newbits; + ad1843->write(ad1843->chip, field->reg, w); + + return oldval; +} + +/* + * ad1843_read_multi reads multiple bitfields from the same AD1843 + * register. It uses a single read cycle to do it. (Reading the + * ad1843 requires 256 bit times at 12.288 MHz, or nearly 20 + * microseconds.) + * + * Called like this. + * + * ad1843_read_multi(ad1843, nfields, + * &ad1843_FIELD1, &val1, + * &ad1843_FIELD2, &val2, ...); + */ + +static void ad1843_read_multi(struct snd_ad1843 *ad1843, int argcount, ...) +{ + va_list ap; + const struct ad1843_bitfield *fp; + int w = 0, mask, *value, reg = -1; + + va_start(ap, argcount); + while (--argcount >= 0) { + fp = va_arg(ap, const struct ad1843_bitfield *); + value = va_arg(ap, int *); + if (reg == -1) { + reg = fp->reg; + w = ad1843->read(ad1843->chip, reg); + } + + mask = (1 << fp->nbits) - 1; + *value = w >> fp->lo_bit & mask; + } + va_end(ap); +} + +/* + * ad1843_write_multi stores multiple bitfields into the same AD1843 + * register. It uses one read and one write cycle to do it. + * + * Called like this. + * + * ad1843_write_multi(ad1843, nfields, + * &ad1843_FIELD1, val1, + * &ad1843_FIELF2, val2, ...); + */ + +static void ad1843_write_multi(struct snd_ad1843 *ad1843, int argcount, ...) +{ + va_list ap; + int reg; + const struct ad1843_bitfield *fp; + int value; + int w, m, mask, bits; + + mask = 0; + bits = 0; + reg = -1; + + va_start(ap, argcount); + while (--argcount >= 0) { + fp = va_arg(ap, const struct ad1843_bitfield *); + value = va_arg(ap, int); + if (reg == -1) + reg = fp->reg; + else + BUG_ON(reg != fp->reg); + m = ((1 << fp->nbits) - 1) << fp->lo_bit; + mask |= m; + bits |= (value << fp->lo_bit) & m; + } + va_end(ap); + + if (~mask & 0xFFFF) + w = ad1843->read(ad1843->chip, reg); + else + w = 0; + w = (w & ~mask) | bits; + ad1843->write(ad1843->chip, reg, w); +} + +int ad1843_get_gain_max(struct snd_ad1843 *ad1843, int id) +{ + const struct ad1843_gain *gp = ad1843_gain[id]; + int ret; + + ret = (1 << gp->lfield->nbits); + if (!gp->lmute) + ret -= 1; + return ret; +} + +/* + * ad1843_get_gain reads the specified register and extracts the gain value + * using the supplied gain type. + */ + +int ad1843_get_gain(struct snd_ad1843 *ad1843, int id) +{ + int lg, rg, lm, rm; + const struct ad1843_gain *gp = ad1843_gain[id]; + unsigned short mask = (1 << gp->lfield->nbits) - 1; + + ad1843_read_multi(ad1843, 2, gp->lfield, &lg, gp->rfield, &rg); + if (gp->negative) { + lg = mask - lg; + rg = mask - rg; + } + if (gp->lmute) { + ad1843_read_multi(ad1843, 2, gp->lmute, &lm, gp->rmute, &rm); + if (lm) + lg = 0; + if (rm) + rg = 0; + } + return lg << 0 | rg << 8; +} + +/* + * Set an audio channel's gain. + * + * Returns the new gain, which may be lower than the old gain. + */ + +int ad1843_set_gain(struct snd_ad1843 *ad1843, int id, int newval) +{ + const struct ad1843_gain *gp = ad1843_gain[id]; + unsigned short mask = (1 << gp->lfield->nbits) - 1; + + int lg = (newval >> 0) & mask; + int rg = (newval >> 8) & mask; + int lm = (lg == 0) ? 1 : 0; + int rm = (rg == 0) ? 1 : 0; + + if (gp->negative) { + lg = mask - lg; + rg = mask - rg; + } + if (gp->lmute) + ad1843_write_multi(ad1843, 2, gp->lmute, lm, gp->rmute, rm); + ad1843_write_multi(ad1843, 2, gp->lfield, lg, gp->rfield, rg); + return ad1843_get_gain(ad1843, id); +} + +/* Returns the current recording source */ + +int ad1843_get_recsrc(struct snd_ad1843 *ad1843) +{ + int val = ad1843_read_bits(ad1843, &ad1843_LSS); + + if (val < 0 || val > 2) { + val = 2; + ad1843_write_multi(ad1843, 2, + &ad1843_LSS, val, &ad1843_RSS, val); + } + return val; +} + +/* + * Set recording source. + * + * Returns newsrc on success, -errno on failure. + */ + +int ad1843_set_recsrc(struct snd_ad1843 *ad1843, int newsrc) +{ + if (newsrc < 0 || newsrc > 2) + return -EINVAL; + + ad1843_write_multi(ad1843, 2, &ad1843_LSS, newsrc, &ad1843_RSS, newsrc); + return newsrc; +} + +/* Setup ad1843 for D/A conversion. */ + +void ad1843_setup_dac(struct snd_ad1843 *ad1843, + unsigned int id, + unsigned int framerate, + snd_pcm_format_t fmt, + unsigned int channels) +{ + int ad_fmt = 0, ad_mode = 0; + + switch (fmt) { + case SNDRV_PCM_FORMAT_S8: + ad_fmt = 0; + break; + case SNDRV_PCM_FORMAT_U8: + ad_fmt = 0; + break; + case SNDRV_PCM_FORMAT_S16_LE: + ad_fmt = 1; + break; + case SNDRV_PCM_FORMAT_MU_LAW: + ad_fmt = 2; + break; + case SNDRV_PCM_FORMAT_A_LAW: + ad_fmt = 3; + break; + default: + break; + } + + switch (channels) { + case 2: + ad_mode = 0; + break; + case 1: + ad_mode = 1; + break; + default: + break; + } + + if (id) { + ad1843_write_bits(ad1843, &ad1843_C2C, framerate); + ad1843_write_multi(ad1843, 2, + &ad1843_DA2SM, ad_mode, + &ad1843_DA2F, ad_fmt); + } else { + ad1843_write_bits(ad1843, &ad1843_C1C, framerate); + ad1843_write_multi(ad1843, 2, + &ad1843_DA1SM, ad_mode, + &ad1843_DA1F, ad_fmt); + } +} + +void ad1843_shutdown_dac(struct snd_ad1843 *ad1843, unsigned int id) +{ + if (id) + ad1843_write_bits(ad1843, &ad1843_DA2F, 1); + else + ad1843_write_bits(ad1843, &ad1843_DA1F, 1); +} + +void ad1843_setup_adc(struct snd_ad1843 *ad1843, + unsigned int framerate, + snd_pcm_format_t fmt, + unsigned int channels) +{ + int da_fmt = 0; + + switch (fmt) { + case SNDRV_PCM_FORMAT_S8: da_fmt = 0; break; + case SNDRV_PCM_FORMAT_U8: da_fmt = 0; break; + case SNDRV_PCM_FORMAT_S16_LE: da_fmt = 1; break; + case SNDRV_PCM_FORMAT_MU_LAW: da_fmt = 2; break; + case SNDRV_PCM_FORMAT_A_LAW: da_fmt = 3; break; + default: break; + } + + ad1843_write_bits(ad1843, &ad1843_C3C, framerate); + ad1843_write_multi(ad1843, 2, + &ad1843_ADLF, da_fmt, &ad1843_ADRF, da_fmt); +} + +void ad1843_shutdown_adc(struct snd_ad1843 *ad1843) +{ + /* nothing to do */ +} + +/* + * Fully initialize the ad1843. As described in the AD1843 data + * sheet, section "START-UP SEQUENCE". The numbered comments are + * subsection headings from the data sheet. See the data sheet, pages + * 52-54, for more info. + * + * return 0 on success, -errno on failure. */ + +int ad1843_init(struct snd_ad1843 *ad1843) +{ + unsigned long later; + + if (ad1843_read_bits(ad1843, &ad1843_INIT) != 0) { + printk(KERN_ERR "ad1843: AD1843 won't initialize\n"); + return -EIO; + } + + ad1843_write_bits(ad1843, &ad1843_SCF, 1); + + /* 4. Put the conversion resources into standby. */ + ad1843_write_bits(ad1843, &ad1843_PDNI, 0); + later = jiffies + msecs_to_jiffies(500); + + while (ad1843_read_bits(ad1843, &ad1843_PDNO)) { + if (time_after(jiffies, later)) { + printk(KERN_ERR + "ad1843: AD1843 won't power up\n"); + return -EIO; + } + schedule_timeout_interruptible(5); + } + + /* 5. Power up the clock generators and enable clock output pins. */ + ad1843_write_multi(ad1843, 3, + &ad1843_C1EN, 1, + &ad1843_C2EN, 1, + &ad1843_C3EN, 1); + + /* 6. Configure conversion resources while they are in standby. */ + + /* DAC1/2 use clock 1/2 as source, ADC uses clock 3. Always. */ + ad1843_write_multi(ad1843, 4, + &ad1843_DA1C, 1, + &ad1843_DA2C, 2, + &ad1843_ADLC, 3, + &ad1843_ADRC, 3); + + /* 7. Enable conversion resources. */ + ad1843_write_bits(ad1843, &ad1843_ADTLK, 1); + ad1843_write_multi(ad1843, 7, + &ad1843_ANAEN, 1, + &ad1843_AAMEN, 1, + &ad1843_DA1EN, 1, + &ad1843_DA2EN, 1, + &ad1843_DDMEN, 1, + &ad1843_ADLEN, 1, + &ad1843_ADREN, 1); + + /* 8. Configure conversion resources while they are enabled. */ + + /* set gain to 0 for all channels */ + ad1843_set_gain(ad1843, AD1843_GAIN_RECLEV, 0); + ad1843_set_gain(ad1843, AD1843_GAIN_LINE, 0); + ad1843_set_gain(ad1843, AD1843_GAIN_LINE_2, 0); + ad1843_set_gain(ad1843, AD1843_GAIN_MIC, 0); + ad1843_set_gain(ad1843, AD1843_GAIN_PCM_0, 0); + ad1843_set_gain(ad1843, AD1843_GAIN_PCM_1, 0); + + /* Unmute all channels. */ + /* DAC1 */ + ad1843_write_multi(ad1843, 2, &ad1843_LDA1GM, 0, &ad1843_RDA1GM, 0); + /* DAC2 */ + ad1843_write_multi(ad1843, 2, &ad1843_LDA2GM, 0, &ad1843_RDA2GM, 0); + + /* Set default recording source to Line In and set + * mic gain to +20 dB. + */ + ad1843_set_recsrc(ad1843, 2); + ad1843_write_multi(ad1843, 2, &ad1843_LMGE, 1, &ad1843_RMGE, 1); + + /* Set Speaker Out level to +/- 4V and unmute it. */ + ad1843_write_multi(ad1843, 3, + &ad1843_HPOS, 1, + &ad1843_HPOM, 0, + &ad1843_MPOM, 0); + + return 0; +} diff --git a/sound/mips/hal2.c b/sound/mips/hal2.c new file mode 100644 index 00000000000..db495be0186 --- /dev/null +++ b/sound/mips/hal2.c @@ -0,0 +1,947 @@ +/* + * Driver for A2 audio system used in SGI machines + * Copyright (c) 2008 Thomas Bogendoerfer <tsbogend@alpha.fanken.de> + * + * Based on OSS code from Ladislav Michl <ladis@linux-mips.org>, which + * was based on code from Ulf Carlsson + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. + * + * This program is distributed in the hope that it will be useful, + * but WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the + * GNU General Public License for more details. + * + * You should have received a copy of the GNU General Public License + * along with this program; if not, write to the Free Software + * Foundation, Inc., 675 Mass Ave, Cambridge, MA 02139, USA. + * + */ +#include <linux/kernel.h> +#include <linux/init.h> +#include <linux/interrupt.h> +#include <linux/dma-mapping.h> +#include <linux/platform_device.h> +#include <linux/io.h> + +#include <asm/sgi/hpc3.h> +#include <asm/sgi/ip22.h> + +#include <sound/core.h> +#include <sound/control.h> +#include <sound/pcm.h> +#include <sound/pcm-indirect.h> +#include <sound/initval.h> + +#include "hal2.h" + +static int index = SNDRV_DEFAULT_IDX1; /* Index 0-MAX */ +static char *id = SNDRV_DEFAULT_STR1; /* ID for this card */ + +module_param(index, int, 0444); +MODULE_PARM_DESC(index, "Index value for SGI HAL2 soundcard."); +module_param(id, charp, 0444); +MODULE_PARM_DESC(id, "ID string for SGI HAL2 soundcard."); +MODULE_DESCRIPTION("ALSA driver for SGI HAL2 audio"); +MODULE_AUTHOR("Thomas Bogendoerfer"); +MODULE_LICENSE("GPL"); + + +#define H2_BLOCK_SIZE 1024 +#define H2_BUF_SIZE 16384 + +struct hal2_pbus { + struct hpc3_pbus_dmacregs *pbus; + int pbusnr; + unsigned int ctrl; /* Current state of pbus->pbdma_ctrl */ +}; + +struct hal2_desc { + struct hpc_dma_desc desc; + u32 pad; /* padding */ +}; + +struct hal2_codec { + struct snd_pcm_indirect pcm_indirect; + struct snd_pcm_substream *substream; + + unsigned char *buffer; + dma_addr_t buffer_dma; + struct hal2_desc *desc; + dma_addr_t desc_dma; + int desc_count; + struct hal2_pbus pbus; + int voices; /* mono/stereo */ + unsigned int sample_rate; + unsigned int master; /* Master frequency */ + unsigned short mod; /* MOD value */ + unsigned short inc; /* INC value */ +}; + +#define H2_MIX_OUTPUT_ATT 0 +#define H2_MIX_INPUT_GAIN 1 + +struct snd_hal2 { + struct snd_card *card; + + struct hal2_ctl_regs *ctl_regs; /* HAL2 ctl registers */ + struct hal2_aes_regs *aes_regs; /* HAL2 aes registers */ + struct hal2_vol_regs *vol_regs; /* HAL2 vol registers */ + struct hal2_syn_regs *syn_regs; /* HAL2 syn registers */ + + struct hal2_codec dac; + struct hal2_codec adc; +}; + +#define H2_INDIRECT_WAIT(regs) while (hal2_read(®s->isr) & H2_ISR_TSTATUS); + +#define H2_READ_ADDR(addr) (addr | (1<<7)) +#define H2_WRITE_ADDR(addr) (addr) + +static inline u32 hal2_read(u32 *reg) +{ + return __raw_readl(reg); +} + +static inline void hal2_write(u32 val, u32 *reg) +{ + __raw_writel(val, reg); +} + + +static u32 hal2_i_read32(struct snd_hal2 *hal2, u16 addr) +{ + u32 ret; + struct hal2_ctl_regs *regs = hal2->ctl_regs; + + hal2_write(H2_READ_ADDR(addr), ®s->iar); + H2_INDIRECT_WAIT(regs); + ret = hal2_read(®s->idr0) & 0xffff; + hal2_write(H2_READ_ADDR(addr) | 0x1, ®s->iar); + H2_INDIRECT_WAIT(regs); + ret |= (hal2_read(®s->idr0) & 0xffff) << 16; + return ret; +} + +static void hal2_i_write16(struct snd_hal2 *hal2, u16 addr, u16 val) +{ + struct hal2_ctl_regs *regs = hal2->ctl_regs; + + hal2_write(val, ®s->idr0); + hal2_write(0, ®s->idr1); + hal2_write(0, ®s->idr2); + hal2_write(0, ®s->idr3); + hal2_write(H2_WRITE_ADDR(addr), ®s->iar); + H2_INDIRECT_WAIT(regs); +} + +static void hal2_i_write32(struct snd_hal2 *hal2, u16 addr, u32 val) +{ + struct hal2_ctl_regs *regs = hal2->ctl_regs; + + hal2_write(val & 0xffff, ®s->idr0); + hal2_write(val >> 16, ®s->idr1); + hal2_write(0, ®s->idr2); + hal2_write(0, ®s->idr3); + hal2_write(H2_WRITE_ADDR(addr), ®s->iar); + H2_INDIRECT_WAIT(regs); +} + +static void hal2_i_setbit16(struct snd_hal2 *hal2, u16 addr, u16 bit) +{ + struct hal2_ctl_regs *regs = hal2->ctl_regs; + + hal2_write(H2_READ_ADDR(addr), ®s->iar); + H2_INDIRECT_WAIT(regs); + hal2_write((hal2_read(®s->idr0) & 0xffff) | bit, ®s->idr0); + hal2_write(0, ®s->idr1); + hal2_write(0, ®s->idr2); + hal2_write(0, ®s->idr3); + hal2_write(H2_WRITE_ADDR(addr), ®s->iar); + H2_INDIRECT_WAIT(regs); +} + +static void hal2_i_clearbit16(struct snd_hal2 *hal2, u16 addr, u16 bit) +{ + struct hal2_ctl_regs *regs = hal2->ctl_regs; + + hal2_write(H2_READ_ADDR(addr), ®s->iar); + H2_INDIRECT_WAIT(regs); + hal2_write((hal2_read(®s->idr0) & 0xffff) & ~bit, ®s->idr0); + hal2_write(0, ®s->idr1); + hal2_write(0, ®s->idr2); + hal2_write(0, ®s->idr3); + hal2_write(H2_WRITE_ADDR(addr), ®s->iar); + H2_INDIRECT_WAIT(regs); +} + +static int hal2_gain_info(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_info *uinfo) +{ + uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER; + uinfo->count = 2; + uinfo->value.integer.min = 0; + switch ((int)kcontrol->private_value) { + case H2_MIX_OUTPUT_ATT: + uinfo->value.integer.max = 31; + break; + case H2_MIX_INPUT_GAIN: + uinfo->value.integer.max = 15; + break; + } + return 0; +} + +static int hal2_gain_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_hal2 *hal2 = snd_kcontrol_chip(kcontrol); + u32 tmp; + int l, r; + + switch ((int)kcontrol->private_value) { + case H2_MIX_OUTPUT_ATT: + tmp = hal2_i_read32(hal2, H2I_DAC_C2); + if (tmp & H2I_C2_MUTE) { + l = 0; + r = 0; + } else { + l = 31 - ((tmp >> H2I_C2_L_ATT_SHIFT) & 31); + r = 31 - ((tmp >> H2I_C2_R_ATT_SHIFT) & 31); + } + break; + case H2_MIX_INPUT_GAIN: + tmp = hal2_i_read32(hal2, H2I_ADC_C2); + l = (tmp >> H2I_C2_L_GAIN_SHIFT) & 15; + r = (tmp >> H2I_C2_R_GAIN_SHIFT) & 15; + break; + } + ucontrol->value.integer.value[0] = l; + ucontrol->value.integer.value[1] = r; + + return 0; +} + +static int hal2_gain_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_hal2 *hal2 = snd_kcontrol_chip(kcontrol); + u32 old, new; + int l, r; + + l = ucontrol->value.integer.value[0]; + r = ucontrol->value.integer.value[1]; + + switch ((int)kcontrol->private_value) { + case H2_MIX_OUTPUT_ATT: + old = hal2_i_read32(hal2, H2I_DAC_C2); + new = old & ~(H2I_C2_L_ATT_M | H2I_C2_R_ATT_M | H2I_C2_MUTE); + if (l | r) { + l = 31 - l; + r = 31 - r; + new |= (l << H2I_C2_L_ATT_SHIFT); + new |= (r << H2I_C2_R_ATT_SHIFT); + } else + new |= H2I_C2_L_ATT_M | H2I_C2_R_ATT_M | H2I_C2_MUTE; + hal2_i_write32(hal2, H2I_DAC_C2, new); + break; + case H2_MIX_INPUT_GAIN: + old = hal2_i_read32(hal2, H2I_ADC_C2); + new = old & ~(H2I_C2_L_GAIN_M | H2I_C2_R_GAIN_M); + new |= (l << H2I_C2_L_GAIN_SHIFT); + new |= (r << H2I_C2_R_GAIN_SHIFT); + hal2_i_write32(hal2, H2I_ADC_C2, new); + break; + } + return old != new; +} + +static struct snd_kcontrol_new hal2_ctrl_headphone __devinitdata = { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "Headphone Playback Volume", + .access = SNDRV_CTL_ELEM_ACCESS_READWRITE, + .private_value = H2_MIX_OUTPUT_ATT, + .info = hal2_gain_info, + .get = hal2_gain_get, + .put = hal2_gain_put, +}; + +static struct snd_kcontrol_new hal2_ctrl_mic __devinitdata = { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "Mic Capture Volume", + .access = SNDRV_CTL_ELEM_ACCESS_READWRITE, + .private_value = H2_MIX_INPUT_GAIN, + .info = hal2_gain_info, + .get = hal2_gain_get, + .put = hal2_gain_put, +}; + +static int __devinit hal2_mixer_create(struct snd_hal2 *hal2) +{ + int err; + + /* mute DAC */ + hal2_i_write32(hal2, H2I_DAC_C2, + H2I_C2_L_ATT_M | H2I_C2_R_ATT_M | H2I_C2_MUTE); + /* mute ADC */ + hal2_i_write32(hal2, H2I_ADC_C2, 0); + + err = snd_ctl_add(hal2->card, + snd_ctl_new1(&hal2_ctrl_headphone, hal2)); + if (err < 0) + return err; + + err = snd_ctl_add(hal2->card, + snd_ctl_new1(&hal2_ctrl_mic, hal2)); + if (err < 0) + return err; + + return 0; +} + +static irqreturn_t hal2_interrupt(int irq, void *dev_id) +{ + struct snd_hal2 *hal2 = dev_id; + irqreturn_t ret = IRQ_NONE; + + /* decide what caused this interrupt */ + if (hal2->dac.pbus.pbus->pbdma_ctrl & HPC3_PDMACTRL_INT) { + snd_pcm_period_elapsed(hal2->dac.substream); + ret = IRQ_HANDLED; + } + if (hal2->adc.pbus.pbus->pbdma_ctrl & HPC3_PDMACTRL_INT) { + snd_pcm_period_elapsed(hal2->adc.substream); + ret = IRQ_HANDLED; + } + return ret; +} + +static int hal2_compute_rate(struct hal2_codec *codec, unsigned int rate) +{ + unsigned short mod; + + if (44100 % rate < 48000 % rate) { + mod = 4 * 44100 / rate; + codec->master = 44100; + } else { + mod = 4 * 48000 / rate; + codec->master = 48000; + } + + codec->inc = 4; + codec->mod = mod; + rate = 4 * codec->master / mod; + + return rate; +} + +static void hal2_set_dac_rate(struct snd_hal2 *hal2) +{ + unsigned int master = hal2->dac.master; + int inc = hal2->dac.inc; + int mod = hal2->dac.mod; + + hal2_i_write16(hal2, H2I_BRES1_C1, (master == 44100) ? 1 : 0); + hal2_i_write32(hal2, H2I_BRES1_C2, + ((0xffff & (inc - mod - 1)) << 16) | inc); +} + +static void hal2_set_adc_rate(struct snd_hal2 *hal2) +{ + unsigned int master = hal2->adc.master; + int inc = hal2->adc.inc; + int mod = hal2->adc.mod; + + hal2_i_write16(hal2, H2I_BRES2_C1, (master == 44100) ? 1 : 0); + hal2_i_write32(hal2, H2I_BRES2_C2, + ((0xffff & (inc - mod - 1)) << 16) | inc); +} + +static void hal2_setup_dac(struct snd_hal2 *hal2) +{ + unsigned int fifobeg, fifoend, highwater, sample_size; + struct hal2_pbus *pbus = &hal2->dac.pbus; + + /* Now we set up some PBUS information. The PBUS needs information about + * what portion of the fifo it will use. If it's receiving or + * transmitting, and finally whether the stream is little endian or big + * endian. The information is written later, on the start call. + */ + sample_size = 2 * hal2->dac.voices; + /* Fifo should be set to hold exactly four samples. Highwater mark + * should be set to two samples. */ + highwater = (sample_size * 2) >> 1; /* halfwords */ + fifobeg = 0; /* playback is first */ + fifoend = (sample_size * 4) >> 3; /* doublewords */ + pbus->ctrl = HPC3_PDMACTRL_RT | HPC3_PDMACTRL_LD | + (highwater << 8) | (fifobeg << 16) | (fifoend << 24); + /* We disable everything before we do anything at all */ + pbus->pbus->pbdma_ctrl = HPC3_PDMACTRL_LD; + hal2_i_clearbit16(hal2, H2I_DMA_PORT_EN, H2I_DMA_PORT_EN_CODECTX); + /* Setup the HAL2 for playback */ + hal2_set_dac_rate(hal2); + /* Set endianess */ + hal2_i_clearbit16(hal2, H2I_DMA_END, H2I_DMA_END_CODECTX); + /* Set DMA bus */ + hal2_i_setbit16(hal2, H2I_DMA_DRV, (1 << pbus->pbusnr)); + /* We are using 1st Bresenham clock generator for playback */ + hal2_i_write16(hal2, H2I_DAC_C1, (pbus->pbusnr << H2I_C1_DMA_SHIFT) + | (1 << H2I_C1_CLKID_SHIFT) + | (hal2->dac.voices << H2I_C1_DATAT_SHIFT)); +} + +static void hal2_setup_adc(struct snd_hal2 *hal2) +{ + unsigned int fifobeg, fifoend, highwater, sample_size; + struct hal2_pbus *pbus = &hal2->adc.pbus; + + sample_size = 2 * hal2->adc.voices; + highwater = (sample_size * 2) >> 1; /* halfwords */ + fifobeg = (4 * 4) >> 3; /* record is second */ + fifoend = (4 * 4 + sample_size * 4) >> 3; /* doublewords */ + pbus->ctrl = HPC3_PDMACTRL_RT | HPC3_PDMACTRL_RCV | HPC3_PDMACTRL_LD | + (highwater << 8) | (fifobeg << 16) | (fifoend << 24); + pbus->pbus->pbdma_ctrl = HPC3_PDMACTRL_LD; + hal2_i_clearbit16(hal2, H2I_DMA_PORT_EN, H2I_DMA_PORT_EN_CODECR); + /* Setup the HAL2 for record */ + hal2_set_adc_rate(hal2); + /* Set endianess */ + hal2_i_clearbit16(hal2, H2I_DMA_END, H2I_DMA_END_CODECR); + /* Set DMA bus */ + hal2_i_setbit16(hal2, H2I_DMA_DRV, (1 << pbus->pbusnr)); + /* We are using 2nd Bresenham clock generator for record */ + hal2_i_write16(hal2, H2I_ADC_C1, (pbus->pbusnr << H2I_C1_DMA_SHIFT) + | (2 << H2I_C1_CLKID_SHIFT) + | (hal2->adc.voices << H2I_C1_DATAT_SHIFT)); +} + +static void hal2_start_dac(struct snd_hal2 *hal2) +{ + struct hal2_pbus *pbus = &hal2->dac.pbus; + + pbus->pbus->pbdma_dptr = hal2->dac.desc_dma; + pbus->pbus->pbdma_ctrl = pbus->ctrl | HPC3_PDMACTRL_ACT; + /* enable DAC */ + hal2_i_setbit16(hal2, H2I_DMA_PORT_EN, H2I_DMA_PORT_EN_CODECTX); +} + +static void hal2_start_adc(struct snd_hal2 *hal2) +{ + struct hal2_pbus *pbus = &hal2->adc.pbus; + + pbus->pbus->pbdma_dptr = hal2->adc.desc_dma; + pbus->pbus->pbdma_ctrl = pbus->ctrl | HPC3_PDMACTRL_ACT; + /* enable ADC */ + hal2_i_setbit16(hal2, H2I_DMA_PORT_EN, H2I_DMA_PORT_EN_CODECR); +} + +static inline void hal2_stop_dac(struct snd_hal2 *hal2) +{ + hal2->dac.pbus.pbus->pbdma_ctrl = HPC3_PDMACTRL_LD; + /* The HAL2 itself may remain enabled safely */ +} + +static inline void hal2_stop_adc(struct snd_hal2 *hal2) +{ + hal2->adc.pbus.pbus->pbdma_ctrl = HPC3_PDMACTRL_LD; +} + +static int hal2_alloc_dmabuf(struct hal2_codec *codec) +{ + struct hal2_desc *desc; + dma_addr_t desc_dma, buffer_dma; + int count = H2_BUF_SIZE / H2_BLOCK_SIZE; + int i; + + codec->buffer = dma_alloc_noncoherent(NULL, H2_BUF_SIZE, + &buffer_dma, GFP_KERNEL); + if (!codec->buffer) + return -ENOMEM; + desc = dma_alloc_noncoherent(NULL, count * sizeof(struct hal2_desc), + &desc_dma, GFP_KERNEL); + if (!desc) { + dma_free_noncoherent(NULL, H2_BUF_SIZE, + codec->buffer, buffer_dma); + return -ENOMEM; + } + codec->buffer_dma = buffer_dma; + codec->desc_dma = desc_dma; + codec->desc = desc; + for (i = 0; i < count; i++) { + desc->desc.pbuf = buffer_dma + i * H2_BLOCK_SIZE; + desc->desc.cntinfo = HPCDMA_XIE | H2_BLOCK_SIZE; + desc->desc.pnext = (i == count - 1) ? + desc_dma : desc_dma + (i + 1) * sizeof(struct hal2_desc); + desc++; + } + dma_cache_sync(NULL, codec->desc, count * sizeof(struct hal2_desc), + DMA_TO_DEVICE); + codec->desc_count = count; + return 0; +} + +static void hal2_free_dmabuf(struct hal2_codec *codec) +{ + dma_free_noncoherent(NULL, codec->desc_count * sizeof(struct hal2_desc), + codec->desc, codec->desc_dma); + dma_free_noncoherent(NULL, H2_BUF_SIZE, codec->buffer, + codec->buffer_dma); +} + +static struct snd_pcm_hardware hal2_pcm_hw = { + .info = (SNDRV_PCM_INFO_MMAP | + SNDRV_PCM_INFO_MMAP_VALID | + SNDRV_PCM_INFO_INTERLEAVED | + SNDRV_PCM_INFO_BLOCK_TRANSFER), + .formats = SNDRV_PCM_FMTBIT_S16_BE, + .rates = SNDRV_PCM_RATE_8000_48000, + .rate_min = 8000, + .rate_max = 48000, + .channels_min = 2, + .channels_max = 2, + .buffer_bytes_max = 65536, + .period_bytes_min = 1024, + .period_bytes_max = 65536, + .periods_min = 2, + .periods_max = 1024, +}; + +static int hal2_pcm_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params) +{ + int err; + + err = snd_pcm_lib_malloc_pages(substream, params_buffer_bytes(params)); + if (err < 0) + return err; + + return 0; +} + +static int hal2_pcm_hw_free(struct snd_pcm_substream *substream) +{ + return snd_pcm_lib_free_pages(substream); +} + +static int hal2_playback_open(struct snd_pcm_substream *substream) +{ + struct snd_pcm_runtime *runtime = substream->runtime; + struct snd_hal2 *hal2 = snd_pcm_substream_chip(substream); + int err; + + runtime->hw = hal2_pcm_hw; + + err = hal2_alloc_dmabuf(&hal2->dac); + if (err) + return err; + return 0; +} + +static int hal2_playback_close(struct snd_pcm_substream *substream) +{ + struct snd_hal2 *hal2 = snd_pcm_substream_chip(substream); + + hal2_free_dmabuf(&hal2->dac); + return 0; +} + +static int hal2_playback_prepare(struct snd_pcm_substream *substream) +{ + struct snd_hal2 *hal2 = snd_pcm_substream_chip(substream); + struct snd_pcm_runtime *runtime = substream->runtime; + struct hal2_codec *dac = &hal2->dac; + + dac->voices = runtime->channels; + dac->sample_rate = hal2_compute_rate(dac, runtime->rate); + memset(&dac->pcm_indirect, 0, sizeof(dac->pcm_indirect)); + dac->pcm_indirect.hw_buffer_size = H2_BUF_SIZE; + dac->pcm_indirect.sw_buffer_size = snd_pcm_lib_buffer_bytes(substream); + dac->substream = substream; + hal2_setup_dac(hal2); + return 0; +} + +static int hal2_playback_trigger(struct snd_pcm_substream *substream, int cmd) +{ + struct snd_hal2 *hal2 = snd_pcm_substream_chip(substream); + + switch (cmd) { + case SNDRV_PCM_TRIGGER_START: + hal2->dac.pcm_indirect.hw_io = hal2->dac.buffer_dma; + hal2->dac.pcm_indirect.hw_data = 0; + substream->ops->ack(substream); + hal2_start_dac(hal2); + break; + case SNDRV_PCM_TRIGGER_STOP: + hal2_stop_dac(hal2); + break; + default: + return -EINVAL; + } + return 0; +} + +static snd_pcm_uframes_t +hal2_playback_pointer(struct snd_pcm_substream *substream) +{ + struct snd_hal2 *hal2 = snd_pcm_substream_chip(substream); + struct hal2_codec *dac = &hal2->dac; + + return snd_pcm_indirect_playback_pointer(substream, &dac->pcm_indirect, + dac->pbus.pbus->pbdma_bptr); +} + +static void hal2_playback_transfer(struct snd_pcm_substream *substream, + struct snd_pcm_indirect *rec, size_t bytes) +{ + struct snd_hal2 *hal2 = snd_pcm_substream_chip(substream); + unsigned char *buf = hal2->dac.buffer + rec->hw_data; + + memcpy(buf, substream->runtime->dma_area + rec->sw_data, bytes); + dma_cache_sync(NULL, buf, bytes, DMA_TO_DEVICE); + +} + +static int hal2_playback_ack(struct snd_pcm_substream *substream) +{ + struct snd_hal2 *hal2 = snd_pcm_substream_chip(substream); + struct hal2_codec *dac = &hal2->dac; + + dac->pcm_indirect.hw_queue_size = H2_BUF_SIZE / 2; + snd_pcm_indirect_playback_transfer(substream, + &dac->pcm_indirect, + hal2_playback_transfer); + return 0; +} + +static int hal2_capture_open(struct snd_pcm_substream *substream) +{ + struct snd_pcm_runtime *runtime = substream->runtime; + struct snd_hal2 *hal2 = snd_pcm_substream_chip(substream); + struct hal2_codec *adc = &hal2->adc; + int err; + + runtime->hw = hal2_pcm_hw; + + err = hal2_alloc_dmabuf(adc); + if (err) + return err; + return 0; +} + +static int hal2_capture_close(struct snd_pcm_substream *substream) +{ + struct snd_hal2 *hal2 = snd_pcm_substream_chip(substream); + + hal2_free_dmabuf(&hal2->adc); + return 0; +} + +static int hal2_capture_prepare(struct snd_pcm_substream *substream) +{ + struct snd_hal2 *hal2 = snd_pcm_substream_chip(substream); + struct snd_pcm_runtime *runtime = substream->runtime; + struct hal2_codec *adc = &hal2->adc; + + adc->voices = runtime->channels; + adc->sample_rate = hal2_compute_rate(adc, runtime->rate); + memset(&adc->pcm_indirect, 0, sizeof(adc->pcm_indirect)); + adc->pcm_indirect.hw_buffer_size = H2_BUF_SIZE; + adc->pcm_indirect.hw_queue_size = H2_BUF_SIZE / 2; + adc->pcm_indirect.sw_buffer_size = snd_pcm_lib_buffer_bytes(substream); + adc->substream = substream; + hal2_setup_adc(hal2); + return 0; +} + +static int hal2_capture_trigger(struct snd_pcm_substream *substream, int cmd) +{ + struct snd_hal2 *hal2 = snd_pcm_substream_chip(substream); + + switch (cmd) { + case SNDRV_PCM_TRIGGER_START: + hal2->adc.pcm_indirect.hw_io = hal2->adc.buffer_dma; + hal2->adc.pcm_indirect.hw_data = 0; + printk(KERN_DEBUG "buffer_dma %x\n", hal2->adc.buffer_dma); + hal2_start_adc(hal2); + break; + case SNDRV_PCM_TRIGGER_STOP: + hal2_stop_adc(hal2); + break; + default: + return -EINVAL; + } + return 0; +} + +static snd_pcm_uframes_t +hal2_capture_pointer(struct snd_pcm_substream *substream) +{ + struct snd_hal2 *hal2 = snd_pcm_substream_chip(substream); + struct hal2_codec *adc = &hal2->adc; + + return snd_pcm_indirect_capture_pointer(substream, &adc->pcm_indirect, + adc->pbus.pbus->pbdma_bptr); +} + +static void hal2_capture_transfer(struct snd_pcm_substream *substream, + struct snd_pcm_indirect *rec, size_t bytes) +{ + struct snd_hal2 *hal2 = snd_pcm_substream_chip(substream); + unsigned char *buf = hal2->adc.buffer + rec->hw_data; + + dma_cache_sync(NULL, buf, bytes, DMA_FROM_DEVICE); + memcpy(substream->runtime->dma_area + rec->sw_data, buf, bytes); +} + +static int hal2_capture_ack(struct snd_pcm_substream *substream) +{ + struct snd_hal2 *hal2 = snd_pcm_substream_chip(substream); + struct hal2_codec *adc = &hal2->adc; + + snd_pcm_indirect_capture_transfer(substream, + &adc->pcm_indirect, + hal2_capture_transfer); + return 0; +} + +static struct snd_pcm_ops hal2_playback_ops = { + .open = hal2_playback_open, + .close = hal2_playback_close, + .ioctl = snd_pcm_lib_ioctl, + .hw_params = hal2_pcm_hw_params, + .hw_free = hal2_pcm_hw_free, + .prepare = hal2_playback_prepare, + .trigger = hal2_playback_trigger, + .pointer = hal2_playback_pointer, + .ack = hal2_playback_ack, +}; + +static struct snd_pcm_ops hal2_capture_ops = { + .open = hal2_capture_open, + .close = hal2_capture_close, + .ioctl = snd_pcm_lib_ioctl, + .hw_params = hal2_pcm_hw_params, + .hw_free = hal2_pcm_hw_free, + .prepare = hal2_capture_prepare, + .trigger = hal2_capture_trigger, + .pointer = hal2_capture_pointer, + .ack = hal2_capture_ack, +}; + +static int __devinit hal2_pcm_create(struct snd_hal2 *hal2) +{ + struct snd_pcm *pcm; + int err; + + /* create first pcm device with one outputs and one input */ + err = snd_pcm_new(hal2->card, "SGI HAL2 Audio", 0, 1, 1, &pcm); + if (err < 0) + return err; + + pcm->private_data = hal2; + strcpy(pcm->name, "SGI HAL2"); + + /* set operators */ + snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_PLAYBACK, + &hal2_playback_ops); + snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_CAPTURE, + &hal2_capture_ops); + snd_pcm_lib_preallocate_pages_for_all(pcm, SNDRV_DMA_TYPE_CONTINUOUS, + snd_dma_continuous_data(GFP_KERNEL), + 0, 1024 * 1024); + + return 0; +} + +static int hal2_dev_free(struct snd_device *device) +{ + struct snd_hal2 *hal2 = device->device_data; + + free_irq(SGI_HPCDMA_IRQ, hal2); + kfree(hal2); + return 0; +} + +static struct snd_device_ops hal2_ops = { + .dev_free = hal2_dev_free, +}; + +static void hal2_init_codec(struct hal2_codec *codec, struct hpc3_regs *hpc3, + int index) +{ + codec->pbus.pbusnr = index; + codec->pbus.pbus = &hpc3->pbdma[index]; +} + +static int hal2_detect(struct snd_hal2 *hal2) +{ + unsigned short board, major, minor; + unsigned short rev; + + /* reset HAL2 */ + hal2_write(0, &hal2->ctl_regs->isr); + + /* release reset */ + hal2_write(H2_ISR_GLOBAL_RESET_N | H2_ISR_CODEC_RESET_N, + &hal2->ctl_regs->isr); + + + hal2_i_write16(hal2, H2I_RELAY_C, H2I_RELAY_C_STATE); + rev = hal2_read(&hal2->ctl_regs->rev); + if (rev & H2_REV_AUDIO_PRESENT) + return -ENODEV; + + board = (rev & H2_REV_BOARD_M) >> 12; + major = (rev & H2_REV_MAJOR_CHIP_M) >> 4; + minor = (rev & H2_REV_MINOR_CHIP_M); + + printk(KERN_INFO "SGI HAL2 revision %i.%i.%i\n", + board, major, minor); + + return 0; +} + +static int hal2_create(struct snd_card *card, struct snd_hal2 **rchip) +{ + struct snd_hal2 *hal2; + struct hpc3_regs *hpc3 = hpc3c0; + int err; + + hal2 = kzalloc(sizeof(struct snd_hal2), GFP_KERNEL); + if (!hal2) + return -ENOMEM; + + hal2->card = card; + + if (request_irq(SGI_HPCDMA_IRQ, hal2_interrupt, IRQF_SHARED, + "SGI HAL2", hal2)) { + printk(KERN_ERR "HAL2: Can't get irq %d\n", SGI_HPCDMA_IRQ); + kfree(hal2); + return -EAGAIN; + } + + hal2->ctl_regs = (struct hal2_ctl_regs *)hpc3->pbus_extregs[0]; + hal2->aes_regs = (struct hal2_aes_regs *)hpc3->pbus_extregs[1]; + hal2->vol_regs = (struct hal2_vol_regs *)hpc3->pbus_extregs[2]; + hal2->syn_regs = (struct hal2_syn_regs *)hpc3->pbus_extregs[3]; + + if (hal2_detect(hal2) < 0) { + kfree(hal2); + return -ENODEV; + } + + hal2_init_codec(&hal2->dac, hpc3, 0); + hal2_init_codec(&hal2->adc, hpc3, 1); + + /* + * All DMA channel interfaces in HAL2 are designed to operate with + * PBUS programmed for 2 cycles in D3, 2 cycles in D4 and 2 cycles + * in D5. HAL2 is a 16-bit device which can accept both big and little + * endian format. It assumes that even address bytes are on high + * portion of PBUS (15:8) and assumes that HPC3 is programmed to + * accept a live (unsynchronized) version of P_DREQ_N from HAL2. + */ +#define HAL2_PBUS_DMACFG ((0 << HPC3_DMACFG_D3R_SHIFT) | \ + (2 << HPC3_DMACFG_D4R_SHIFT) | \ + (2 << HPC3_DMACFG_D5R_SHIFT) | \ + (0 << HPC3_DMACFG_D3W_SHIFT) | \ + (2 << HPC3_DMACFG_D4W_SHIFT) | \ + (2 << HPC3_DMACFG_D5W_SHIFT) | \ + HPC3_DMACFG_DS16 | \ + HPC3_DMACFG_EVENHI | \ + HPC3_DMACFG_RTIME | \ + (8 << HPC3_DMACFG_BURST_SHIFT) | \ + HPC3_DMACFG_DRQLIVE) + /* + * Ignore what's mentioned in the specification and write value which + * works in The Real World (TM) + */ + hpc3->pbus_dmacfg[hal2->dac.pbus.pbusnr][0] = 0x8208844; + hpc3->pbus_dmacfg[hal2->adc.pbus.pbusnr][0] = 0x8208844; + + err = snd_device_new(card, SNDRV_DEV_LOWLEVEL, hal2, &hal2_ops); + if (err < 0) { + free_irq(SGI_HPCDMA_IRQ, hal2); + kfree(hal2); + return err; + } + *rchip = hal2; + return 0; +} + +static int __devinit hal2_probe(struct platform_device *pdev) +{ + struct snd_card *card; + struct snd_hal2 *chip; + int err; + + card = snd_card_new(index, id, THIS_MODULE, 0); + if (card == NULL) + return -ENOMEM; + + err = hal2_create(card, &chip); + if (err < 0) { + snd_card_free(card); + return err; + } + snd_card_set_dev(card, &pdev->dev); + + err = hal2_pcm_create(chip); + if (err < 0) { + snd_card_free(card); + return err; + } + err = hal2_mixer_create(chip); + if (err < 0) { + snd_card_free(card); + return err; + } + + strcpy(card->driver, "SGI HAL2 Audio"); + strcpy(card->shortname, "SGI HAL2 Audio"); + sprintf(card->longname, "%s irq %i", + card->shortname, + SGI_HPCDMA_IRQ); + + err = snd_card_register(card); + if (err < 0) { + snd_card_free(card); + return err; + } + platform_set_drvdata(pdev, card); + return 0; +} + +static int __exit hal2_remove(struct platform_device *pdev) +{ + struct snd_card *card = platform_get_drvdata(pdev); + + snd_card_free(card); + platform_set_drvdata(pdev, NULL); + return 0; +} + +static struct platform_driver hal2_driver = { + .probe = hal2_probe, + .remove = __devexit_p(hal2_remove), + .driver = { + .name = "sgihal2", + .owner = THIS_MODULE, + } +}; + +static int __init alsa_card_hal2_init(void) +{ + return platform_driver_register(&hal2_driver); +} + +static void __exit alsa_card_hal2_exit(void) +{ + platform_driver_unregister(&hal2_driver); +} + +module_init(alsa_card_hal2_init); +module_exit(alsa_card_hal2_exit); diff --git a/sound/mips/hal2.h b/sound/mips/hal2.h new file mode 100644 index 00000000000..f19828bc64e --- /dev/null +++ b/sound/mips/hal2.h @@ -0,0 +1,245 @@ +#ifndef __HAL2_H +#define __HAL2_H + +/* + * Driver for HAL2 sound processors + * Copyright (c) 1999 Ulf Carlsson <ulfc@bun.falkenberg.se> + * Copyright (c) 2001, 2002, 2003 Ladislav Michl <ladis@linux-mips.org> + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. + * + * This program is distributed in the hope that it will be useful, + * but WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the + * GNU General Public License for more details. + * + * You should have received a copy of the GNU General Public License + * along with this program; if not, write to the Free Software + * Foundation, Inc., 675 Mass Ave, Cambridge, MA 02139, USA. + * + */ + +#include <linux/types.h> + +/* Indirect status register */ + +#define H2_ISR_TSTATUS 0x01 /* RO: transaction status 1=busy */ +#define H2_ISR_USTATUS 0x02 /* RO: utime status bit 1=armed */ +#define H2_ISR_QUAD_MODE 0x04 /* codec mode 0=indigo 1=quad */ +#define H2_ISR_GLOBAL_RESET_N 0x08 /* chip global reset 0=reset */ +#define H2_ISR_CODEC_RESET_N 0x10 /* codec/synth reset 0=reset */ + +/* Revision register */ + +#define H2_REV_AUDIO_PRESENT 0x8000 /* RO: audio present 0=present */ +#define H2_REV_BOARD_M 0x7000 /* RO: bits 14:12, board revision */ +#define H2_REV_MAJOR_CHIP_M 0x00F0 /* RO: bits 7:4, major chip revision */ +#define H2_REV_MINOR_CHIP_M 0x000F /* RO: bits 3:0, minor chip revision */ + +/* Indirect address register */ + +/* + * Address of indirect internal register to be accessed. A write to this + * register initiates read or write access to the indirect registers in the + * HAL2. Note that there af four indirect data registers for write access to + * registers larger than 16 byte. + */ + +#define H2_IAR_TYPE_M 0xF000 /* bits 15:12, type of functional */ + /* block the register resides in */ + /* 1=DMA Port */ + /* 9=Global DMA Control */ + /* 2=Bresenham */ + /* 3=Unix Timer */ +#define H2_IAR_NUM_M 0x0F00 /* bits 11:8 instance of the */ + /* blockin which the indirect */ + /* register resides */ + /* If IAR_TYPE_M=DMA Port: */ + /* 1=Synth In */ + /* 2=AES In */ + /* 3=AES Out */ + /* 4=DAC Out */ + /* 5=ADC Out */ + /* 6=Synth Control */ + /* If IAR_TYPE_M=Global DMA Control: */ + /* 1=Control */ + /* If IAR_TYPE_M=Bresenham: */ + /* 1=Bresenham Clock Gen 1 */ + /* 2=Bresenham Clock Gen 2 */ + /* 3=Bresenham Clock Gen 3 */ + /* If IAR_TYPE_M=Unix Timer: */ + /* 1=Unix Timer */ +#define H2_IAR_ACCESS_SELECT 0x0080 /* 1=read 0=write */ +#define H2_IAR_PARAM 0x000C /* Parameter Select */ +#define H2_IAR_RB_INDEX_M 0x0003 /* Read Back Index */ + /* 00:word0 */ + /* 01:word1 */ + /* 10:word2 */ + /* 11:word3 */ +/* + * HAL2 internal addressing + * + * The HAL2 has "indirect registers" (idr) which are accessed by writing to the + * Indirect Data registers. Write the address to the Indirect Address register + * to transfer the data. + * + * We define the H2IR_* to the read address and H2IW_* to the write address and + * H2I_* to be fields in whatever register is referred to. + * + * When we write to indirect registers which are larger than one word (16 bit) + * we have to fill more than one indirect register before writing. When we read + * back however we have to read several times, each time with different Read + * Back Indexes (there are defs for doing this easily). + */ + +/* + * Relay Control + */ +#define H2I_RELAY_C 0x9100 +#define H2I_RELAY_C_STATE 0x01 /* state of RELAY pin signal */ + +/* DMA port enable */ + +#define H2I_DMA_PORT_EN 0x9104 +#define H2I_DMA_PORT_EN_SY_IN 0x01 /* Synth_in DMA port */ +#define H2I_DMA_PORT_EN_AESRX 0x02 /* AES receiver DMA port */ +#define H2I_DMA_PORT_EN_AESTX 0x04 /* AES transmitter DMA port */ +#define H2I_DMA_PORT_EN_CODECTX 0x08 /* CODEC transmit DMA port */ +#define H2I_DMA_PORT_EN_CODECR 0x10 /* CODEC receive DMA port */ + +#define H2I_DMA_END 0x9108 /* global dma endian select */ +#define H2I_DMA_END_SY_IN 0x01 /* Synth_in DMA port */ +#define H2I_DMA_END_AESRX 0x02 /* AES receiver DMA port */ +#define H2I_DMA_END_AESTX 0x04 /* AES transmitter DMA port */ +#define H2I_DMA_END_CODECTX 0x08 /* CODEC transmit DMA port */ +#define H2I_DMA_END_CODECR 0x10 /* CODEC receive DMA port */ + /* 0=b_end 1=l_end */ + +#define H2I_DMA_DRV 0x910C /* global PBUS DMA enable */ + +#define H2I_SYNTH_C 0x1104 /* Synth DMA control */ + +#define H2I_AESRX_C 0x1204 /* AES RX dma control */ + +#define H2I_C_TS_EN 0x20 /* Timestamp enable */ +#define H2I_C_TS_FRMT 0x40 /* Timestamp format */ +#define H2I_C_NAUDIO 0x80 /* Sign extend */ + +/* AESRX CTL, 16 bit */ + +#define H2I_AESTX_C 0x1304 /* AES TX DMA control */ +#define H2I_AESTX_C_CLKID_SHIFT 3 /* Bresenham Clock Gen 1-3 */ +#define H2I_AESTX_C_CLKID_M 0x18 +#define H2I_AESTX_C_DATAT_SHIFT 8 /* 1=mono 2=stereo (3=quad) */ +#define H2I_AESTX_C_DATAT_M 0x300 + +/* CODEC registers */ + +#define H2I_DAC_C1 0x1404 /* DAC DMA control, 16 bit */ +#define H2I_DAC_C2 0x1408 /* DAC DMA control, 32 bit */ +#define H2I_ADC_C1 0x1504 /* ADC DMA control, 16 bit */ +#define H2I_ADC_C2 0x1508 /* ADC DMA control, 32 bit */ + +/* Bits in CTL1 register */ + +#define H2I_C1_DMA_SHIFT 0 /* DMA channel */ +#define H2I_C1_DMA_M 0x7 +#define H2I_C1_CLKID_SHIFT 3 /* Bresenham Clock Gen 1-3 */ +#define H2I_C1_CLKID_M 0x18 +#define H2I_C1_DATAT_SHIFT 8 /* 1=mono 2=stereo (3=quad) */ +#define H2I_C1_DATAT_M 0x300 + +/* Bits in CTL2 register */ + +#define H2I_C2_R_GAIN_SHIFT 0 /* right a/d input gain */ +#define H2I_C2_R_GAIN_M 0xf +#define H2I_C2_L_GAIN_SHIFT 4 /* left a/d input gain */ +#define H2I_C2_L_GAIN_M 0xf0 +#define H2I_C2_R_SEL 0x100 /* right input select */ +#define H2I_C2_L_SEL 0x200 /* left input select */ +#define H2I_C2_MUTE 0x400 /* mute */ +#define H2I_C2_DO1 0x00010000 /* digital output port bit 0 */ +#define H2I_C2_DO2 0x00020000 /* digital output port bit 1 */ +#define H2I_C2_R_ATT_SHIFT 18 /* right d/a output - */ +#define H2I_C2_R_ATT_M 0x007c0000 /* attenuation */ +#define H2I_C2_L_ATT_SHIFT 23 /* left d/a output - */ +#define H2I_C2_L_ATT_M 0x0f800000 /* attenuation */ + +#define H2I_SYNTH_MAP_C 0x1104 /* synth dma handshake ctrl */ + +/* Clock generator CTL 1, 16 bit */ + +#define H2I_BRES1_C1 0x2104 +#define H2I_BRES2_C1 0x2204 +#define H2I_BRES3_C1 0x2304 + +#define H2I_BRES_C1_SHIFT 0 /* 0=48.0 1=44.1 2=aes_rx */ +#define H2I_BRES_C1_M 0x03 + +/* Clock generator CTL 2, 32 bit */ + +#define H2I_BRES1_C2 0x2108 +#define H2I_BRES2_C2 0x2208 +#define H2I_BRES3_C2 0x2308 + +#define H2I_BRES_C2_INC_SHIFT 0 /* increment value */ +#define H2I_BRES_C2_INC_M 0xffff +#define H2I_BRES_C2_MOD_SHIFT 16 /* modcontrol value */ +#define H2I_BRES_C2_MOD_M 0xffff0000 /* modctrl=0xffff&(modinc-1) */ + +/* Unix timer, 64 bit */ + +#define H2I_UTIME 0x3104 +#define H2I_UTIME_0_LD 0xffff /* microseconds, LSB's */ +#define H2I_UTIME_1_LD0 0x0f /* microseconds, MSB's */ +#define H2I_UTIME_1_LD1 0xf0 /* tenths of microseconds */ +#define H2I_UTIME_2_LD 0xffff /* seconds, LSB's */ +#define H2I_UTIME_3_LD 0xffff /* seconds, MSB's */ + +struct hal2_ctl_regs { + u32 _unused0[4]; + u32 isr; /* 0x10 Status Register */ + u32 _unused1[3]; + u32 rev; /* 0x20 Revision Register */ + u32 _unused2[3]; + u32 iar; /* 0x30 Indirect Address Register */ + u32 _unused3[3]; + u32 idr0; /* 0x40 Indirect Data Register 0 */ + u32 _unused4[3]; + u32 idr1; /* 0x50 Indirect Data Register 1 */ + u32 _unused5[3]; + u32 idr2; /* 0x60 Indirect Data Register 2 */ + u32 _unused6[3]; + u32 idr3; /* 0x70 Indirect Data Register 3 */ +}; + +struct hal2_aes_regs { + u32 rx_stat[2]; /* Status registers */ + u32 rx_cr[2]; /* Control registers */ + u32 rx_ud[4]; /* User data window */ + u32 rx_st[24]; /* Channel status data */ + + u32 tx_stat[1]; /* Status register */ + u32 tx_cr[3]; /* Control registers */ + u32 tx_ud[4]; /* User data window */ + u32 tx_st[24]; /* Channel status data */ +}; + +struct hal2_vol_regs { + u32 right; /* Right volume */ + u32 left; /* Left volume */ +}; + +struct hal2_syn_regs { + u32 _unused0[2]; + u32 page; /* DOC Page register */ + u32 regsel; /* DOC Register selection */ + u32 dlow; /* DOC Data low */ + u32 dhigh; /* DOC Data high */ + u32 irq; /* IRQ Status */ + u32 dram; /* DRAM Access */ +}; + +#endif /* __HAL2_H */ diff --git a/sound/mips/sgio2audio.c b/sound/mips/sgio2audio.c new file mode 100644 index 00000000000..4c63504348d --- /dev/null +++ b/sound/mips/sgio2audio.c @@ -0,0 +1,1006 @@ +/* + * Sound driver for Silicon Graphics O2 Workstations A/V board audio. + * + * Copyright 2003 Vivien Chappelier <vivien.chappelier@linux-mips.org> + * Copyright 2008 Thomas Bogendoerfer <tsbogend@alpha.franken.de> + * Mxier part taken from mace_audio.c: + * Copyright 2007 Thorben Jändling <tj.trevelyan@gmail.com> + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License as published by + * the Free Software Foundation; either version 2 of the License, or + * (at your option) any later version. + * + * This program is distributed in the hope that it will be useful, + * but WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the + * GNU General Public License for more details. + * + * You should have received a copy of the GNU General Public License + * along with this program; if not, write to the Free Software + * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA + * + */ + +#include <linux/init.h> +#include <linux/delay.h> +#include <linux/spinlock.h> +#include <linux/gfp.h> +#include <linux/vmalloc.h> +#include <linux/interrupt.h> +#include <linux/dma-mapping.h> +#include <linux/platform_device.h> +#include <linux/io.h> + +#include <asm/ip32/ip32_ints.h> +#include <asm/ip32/mace.h> + +#include <sound/core.h> +#include <sound/control.h> +#include <sound/pcm.h> +#define SNDRV_GET_ID +#include <sound/initval.h> +#include <sound/ad1843.h> + + +MODULE_AUTHOR("Vivien Chappelier <vivien.chappelier@linux-mips.org>"); +MODULE_DESCRIPTION("SGI O2 Audio"); +MODULE_LICENSE("GPL"); +MODULE_SUPPORTED_DEVICE("{{Silicon Graphics, O2 Audio}}"); + +static int index = SNDRV_DEFAULT_IDX1; /* Index 0-MAX */ +static char *id = SNDRV_DEFAULT_STR1; /* ID for this card */ + +module_param(index, int, 0444); +MODULE_PARM_DESC(index, "Index value for SGI O2 soundcard."); +module_param(id, charp, 0444); +MODULE_PARM_DESC(id, "ID string for SGI O2 soundcard."); + + +#define AUDIO_CONTROL_RESET BIT(0) /* 1: reset audio interface */ +#define AUDIO_CONTROL_CODEC_PRESENT BIT(1) /* 1: codec detected */ + +#define CODEC_CONTROL_WORD_SHIFT 0 +#define CODEC_CONTROL_READ BIT(16) +#define CODEC_CONTROL_ADDRESS_SHIFT 17 + +#define CHANNEL_CONTROL_RESET BIT(10) /* 1: reset channel */ +#define CHANNEL_DMA_ENABLE BIT(9) /* 1: enable DMA transfer */ +#define CHANNEL_INT_THRESHOLD_DISABLED (0 << 5) /* interrupt disabled */ +#define CHANNEL_INT_THRESHOLD_25 (1 << 5) /* int on buffer >25% full */ +#define CHANNEL_INT_THRESHOLD_50 (2 << 5) /* int on buffer >50% full */ +#define CHANNEL_INT_THRESHOLD_75 (3 << 5) /* int on buffer >75% full */ +#define CHANNEL_INT_THRESHOLD_EMPTY (4 << 5) /* int on buffer empty */ +#define CHANNEL_INT_THRESHOLD_NOT_EMPTY (5 << 5) /* int on buffer !empty */ +#define CHANNEL_INT_THRESHOLD_FULL (6 << 5) /* int on buffer empty */ +#define CHANNEL_INT_THRESHOLD_NOT_FULL (7 << 5) /* int on buffer !empty */ + +#define CHANNEL_RING_SHIFT 12 +#define CHANNEL_RING_SIZE (1 << CHANNEL_RING_SHIFT) +#define CHANNEL_RING_MASK (CHANNEL_RING_SIZE - 1) + +#define CHANNEL_LEFT_SHIFT 40 +#define CHANNEL_RIGHT_SHIFT 8 + +struct snd_sgio2audio_chan { + int idx; + struct snd_pcm_substream *substream; + int pos; + snd_pcm_uframes_t size; + spinlock_t lock; +}; + +/* definition of the chip-specific record */ +struct snd_sgio2audio { + struct snd_card *card; + + /* codec */ + struct snd_ad1843 ad1843; + spinlock_t ad1843_lock; + + /* channels */ + struct snd_sgio2audio_chan channel[3]; + + /* resources */ + void *ring_base; + dma_addr_t ring_base_dma; +}; + +/* AD1843 access */ + +/* + * read_ad1843_reg returns the current contents of a 16 bit AD1843 register. + * + * Returns unsigned register value on success, -errno on failure. + */ +static int read_ad1843_reg(void *priv, int reg) +{ + struct snd_sgio2audio *chip = priv; + int val; + unsigned long flags; + + spin_lock_irqsave(&chip->ad1843_lock, flags); + + writeq((reg << CODEC_CONTROL_ADDRESS_SHIFT) | + CODEC_CONTROL_READ, &mace->perif.audio.codec_control); + wmb(); + val = readq(&mace->perif.audio.codec_control); /* flush bus */ + udelay(200); + + val = readq(&mace->perif.audio.codec_read); + + spin_unlock_irqrestore(&chip->ad1843_lock, flags); + return val; +} + +/* + * write_ad1843_reg writes the specified value to a 16 bit AD1843 register. + */ +static int write_ad1843_reg(void *priv, int reg, int word) +{ + struct snd_sgio2audio *chip = priv; + int val; + unsigned long flags; + + spin_lock_irqsave(&chip->ad1843_lock, flags); + + writeq((reg << CODEC_CONTROL_ADDRESS_SHIFT) | + (word << CODEC_CONTROL_WORD_SHIFT), + &mace->perif.audio.codec_control); + wmb(); + val = readq(&mace->perif.audio.codec_control); /* flush bus */ + udelay(200); + + spin_unlock_irqrestore(&chip->ad1843_lock, flags); + return 0; +} + +static int sgio2audio_gain_info(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_info *uinfo) +{ + struct snd_sgio2audio *chip = snd_kcontrol_chip(kcontrol); + + uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER; + uinfo->count = 2; + uinfo->value.integer.min = 0; + uinfo->value.integer.max = ad1843_get_gain_max(&chip->ad1843, + (int)kcontrol->private_value); + return 0; +} + +static int sgio2audio_gain_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_sgio2audio *chip = snd_kcontrol_chip(kcontrol); + int vol; + + vol = ad1843_get_gain(&chip->ad1843, (int)kcontrol->private_value); + + ucontrol->value.integer.value[0] = (vol >> 8) & 0xFF; + ucontrol->value.integer.value[1] = vol & 0xFF; + + return 0; +} + +static int sgio2audio_gain_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_sgio2audio *chip = snd_kcontrol_chip(kcontrol); + int newvol, oldvol; + + oldvol = ad1843_get_gain(&chip->ad1843, kcontrol->private_value); + newvol = (ucontrol->value.integer.value[0] << 8) | + ucontrol->value.integer.value[1]; + + newvol = ad1843_set_gain(&chip->ad1843, kcontrol->private_value, + newvol); + + return newvol != oldvol; +} + +static int sgio2audio_source_info(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_info *uinfo) +{ + static const char *texts[3] = { + "Cam Mic", "Mic", "Line" + }; + uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED; + uinfo->count = 1; + uinfo->value.enumerated.items = 3; + if (uinfo->value.enumerated.item >= 3) + uinfo->value.enumerated.item = 1; + strcpy(uinfo->value.enumerated.name, + texts[uinfo->value.enumerated.item]); + return 0; +} + +static int sgio2audio_source_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_sgio2audio *chip = snd_kcontrol_chip(kcontrol); + + ucontrol->value.enumerated.item[0] = ad1843_get_recsrc(&chip->ad1843); + return 0; +} + +static int sgio2audio_source_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_sgio2audio *chip = snd_kcontrol_chip(kcontrol); + int newsrc, oldsrc; + + oldsrc = ad1843_get_recsrc(&chip->ad1843); + newsrc = ad1843_set_recsrc(&chip->ad1843, + ucontrol->value.enumerated.item[0]); + + return newsrc != oldsrc; +} + +/* dac1/pcm0 mixer control */ +static struct snd_kcontrol_new sgio2audio_ctrl_pcm0 __devinitdata = { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "PCM Playback Volume", + .index = 0, + .access = SNDRV_CTL_ELEM_ACCESS_READWRITE, + .private_value = AD1843_GAIN_PCM_0, + .info = sgio2audio_gain_info, + .get = sgio2audio_gain_get, + .put = sgio2audio_gain_put, +}; + +/* dac2/pcm1 mixer control */ +static struct snd_kcontrol_new sgio2audio_ctrl_pcm1 __devinitdata = { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "PCM Playback Volume", + .index = 1, + .access = SNDRV_CTL_ELEM_ACCESS_READWRITE, + .private_value = AD1843_GAIN_PCM_1, + .info = sgio2audio_gain_info, + .get = sgio2audio_gain_get, + .put = sgio2audio_gain_put, +}; + +/* record level mixer control */ +static struct snd_kcontrol_new sgio2audio_ctrl_reclevel __devinitdata = { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "Capture Volume", + .access = SNDRV_CTL_ELEM_ACCESS_READWRITE, + .private_value = AD1843_GAIN_RECLEV, + .info = sgio2audio_gain_info, + .get = sgio2audio_gain_get, + .put = sgio2audio_gain_put, +}; + +/* record level source control */ +static struct snd_kcontrol_new sgio2audio_ctrl_recsource __devinitdata = { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "Capture Source", + .access = SNDRV_CTL_ELEM_ACCESS_READWRITE, + .info = sgio2audio_source_info, + .get = sgio2audio_source_get, + .put = sgio2audio_source_put, +}; + +/* line mixer control */ +static struct snd_kcontrol_new sgio2audio_ctrl_line __devinitdata = { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "Line Playback Volume", + .index = 0, + .access = SNDRV_CTL_ELEM_ACCESS_READWRITE, + .private_value = AD1843_GAIN_LINE, + .info = sgio2audio_gain_info, + .get = sgio2audio_gain_get, + .put = sgio2audio_gain_put, +}; + +/* cd mixer control */ +static struct snd_kcontrol_new sgio2audio_ctrl_cd __devinitdata = { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "Line Playback Volume", + .index = 1, + .access = SNDRV_CTL_ELEM_ACCESS_READWRITE, + .private_value = AD1843_GAIN_LINE_2, + .info = sgio2audio_gain_info, + .get = sgio2audio_gain_get, + .put = sgio2audio_gain_put, +}; + +/* mic mixer control */ +static struct snd_kcontrol_new sgio2audio_ctrl_mic __devinitdata = { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "Mic Playback Volume", + .access = SNDRV_CTL_ELEM_ACCESS_READWRITE, + .private_value = AD1843_GAIN_MIC, + .info = sgio2audio_gain_info, + .get = sgio2audio_gain_get, + .put = sgio2audio_gain_put, +}; + + +static int __devinit snd_sgio2audio_new_mixer(struct snd_sgio2audio *chip) +{ + int err; + + err = snd_ctl_add(chip->card, + snd_ctl_new1(&sgio2audio_ctrl_pcm0, chip)); + if (err < 0) + return err; + + err = snd_ctl_add(chip->card, + snd_ctl_new1(&sgio2audio_ctrl_pcm1, chip)); + if (err < 0) + return err; + + err = snd_ctl_add(chip->card, + snd_ctl_new1(&sgio2audio_ctrl_reclevel, chip)); + if (err < 0) + return err; + + err = snd_ctl_add(chip->card, + snd_ctl_new1(&sgio2audio_ctrl_recsource, chip)); + if (err < 0) + return err; + err = snd_ctl_add(chip->card, + snd_ctl_new1(&sgio2audio_ctrl_line, chip)); + if (err < 0) + return err; + + err = snd_ctl_add(chip->card, + snd_ctl_new1(&sgio2audio_ctrl_cd, chip)); + if (err < 0) + return err; + + err = snd_ctl_add(chip->card, + snd_ctl_new1(&sgio2audio_ctrl_mic, chip)); + if (err < 0) + return err; + + return 0; +} + +/* low-level audio interface DMA */ + +/* get data out of bounce buffer, count must be a multiple of 32 */ +/* returns 1 if a period has elapsed */ +static int snd_sgio2audio_dma_pull_frag(struct snd_sgio2audio *chip, + unsigned int ch, unsigned int count) +{ + int ret; + unsigned long src_base, src_pos, dst_mask; + unsigned char *dst_base; + int dst_pos; + u64 *src; + s16 *dst; + u64 x; + unsigned long flags; + struct snd_pcm_runtime *runtime = chip->channel[ch].substream->runtime; + + spin_lock_irqsave(&chip->channel[ch].lock, flags); + + src_base = (unsigned long) chip->ring_base | (ch << CHANNEL_RING_SHIFT); + src_pos = readq(&mace->perif.audio.chan[ch].read_ptr); + dst_base = runtime->dma_area; + dst_pos = chip->channel[ch].pos; + dst_mask = frames_to_bytes(runtime, runtime->buffer_size) - 1; + + /* check if a period has elapsed */ + chip->channel[ch].size += (count >> 3); /* in frames */ + ret = chip->channel[ch].size >= runtime->period_size; + chip->channel[ch].size %= runtime->period_size; + + while (count) { + src = (u64 *)(src_base + src_pos); + dst = (s16 *)(dst_base + dst_pos); + + x = *src; + dst[0] = (x >> CHANNEL_LEFT_SHIFT) & 0xffff; + dst[1] = (x >> CHANNEL_RIGHT_SHIFT) & 0xffff; + + src_pos = (src_pos + sizeof(u64)) & CHANNEL_RING_MASK; + dst_pos = (dst_pos + 2 * sizeof(s16)) & dst_mask; + count -= sizeof(u64); + } + + writeq(src_pos, &mace->perif.audio.chan[ch].read_ptr); /* in bytes */ + chip->channel[ch].pos = dst_pos; + + spin_unlock_irqrestore(&chip->channel[ch].lock, flags); + return ret; +} + +/* put some DMA data in bounce buffer, count must be a multiple of 32 */ +/* returns 1 if a period has elapsed */ +static int snd_sgio2audio_dma_push_frag(struct snd_sgio2audio *chip, + unsigned int ch, unsigned int count) +{ + int ret; + s64 l, r; + unsigned long dst_base, dst_pos, src_mask; + unsigned char *src_base; + int src_pos; + u64 *dst; + s16 *src; + unsigned long flags; + struct snd_pcm_runtime *runtime = chip->channel[ch].substream->runtime; + + spin_lock_irqsave(&chip->channel[ch].lock, flags); + + dst_base = (unsigned long)chip->ring_base | (ch << CHANNEL_RING_SHIFT); + dst_pos = readq(&mace->perif.audio.chan[ch].write_ptr); + src_base = runtime->dma_area; + src_pos = chip->channel[ch].pos; + src_mask = frames_to_bytes(runtime, runtime->buffer_size) - 1; + + /* check if a period has elapsed */ + chip->channel[ch].size += (count >> 3); /* in frames */ + ret = chip->channel[ch].size >= runtime->period_size; + chip->channel[ch].size %= runtime->period_size; + + while (count) { + src = (s16 *)(src_base + src_pos); + dst = (u64 *)(dst_base + dst_pos); + + l = src[0]; /* sign extend */ + r = src[1]; /* sign extend */ + + *dst = ((l & 0x00ffffff) << CHANNEL_LEFT_SHIFT) | + ((r & 0x00ffffff) << CHANNEL_RIGHT_SHIFT); + + dst_pos = (dst_pos + sizeof(u64)) & CHANNEL_RING_MASK; + src_pos = (src_pos + 2 * sizeof(s16)) & src_mask; + count -= sizeof(u64); + } + + writeq(dst_pos, &mace->perif.audio.chan[ch].write_ptr); /* in bytes */ + chip->channel[ch].pos = src_pos; + + spin_unlock_irqrestore(&chip->channel[ch].lock, flags); + return ret; +} + +static int snd_sgio2audio_dma_start(struct snd_pcm_substream *substream) +{ + struct snd_sgio2audio *chip = snd_pcm_substream_chip(substream); + struct snd_sgio2audio_chan *chan = substream->runtime->private_data; + int ch = chan->idx; + + /* reset DMA channel */ + writeq(CHANNEL_CONTROL_RESET, &mace->perif.audio.chan[ch].control); + udelay(10); + writeq(0, &mace->perif.audio.chan[ch].control); + + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { + /* push a full buffer */ + snd_sgio2audio_dma_push_frag(chip, ch, CHANNEL_RING_SIZE - 32); + } + /* set DMA to wake on 50% empty and enable interrupt */ + writeq(CHANNEL_DMA_ENABLE | CHANNEL_INT_THRESHOLD_50, + &mace->perif.audio.chan[ch].control); + return 0; +} + +static int snd_sgio2audio_dma_stop(struct snd_pcm_substream *substream) +{ + struct snd_sgio2audio_chan *chan = substream->runtime->private_data; + + writeq(0, &mace->perif.audio.chan[chan->idx].control); + return 0; +} + +static irqreturn_t snd_sgio2audio_dma_in_isr(int irq, void *dev_id) +{ + struct snd_sgio2audio_chan *chan = dev_id; + struct snd_pcm_substream *substream; + struct snd_sgio2audio *chip; + int count, ch; + + substream = chan->substream; + chip = snd_pcm_substream_chip(substream); + ch = chan->idx; + + /* empty the ring */ + count = CHANNEL_RING_SIZE - + readq(&mace->perif.audio.chan[ch].depth) - 32; + if (snd_sgio2audio_dma_pull_frag(chip, ch, count)) + snd_pcm_period_elapsed(substream); + + return IRQ_HANDLED; +} + +static irqreturn_t snd_sgio2audio_dma_out_isr(int irq, void *dev_id) +{ + struct snd_sgio2audio_chan *chan = dev_id; + struct snd_pcm_substream *substream; + struct snd_sgio2audio *chip; + int count, ch; + + substream = chan->substream; + chip = snd_pcm_substream_chip(substream); + ch = chan->idx; + /* fill the ring */ + count = CHANNEL_RING_SIZE - + readq(&mace->perif.audio.chan[ch].depth) - 32; + if (snd_sgio2audio_dma_push_frag(chip, ch, count)) + snd_pcm_period_elapsed(substream); + + return IRQ_HANDLED; +} + +static irqreturn_t snd_sgio2audio_error_isr(int irq, void *dev_id) +{ + struct snd_sgio2audio_chan *chan = dev_id; + struct snd_pcm_substream *substream; + + substream = chan->substream; + snd_sgio2audio_dma_stop(substream); + snd_sgio2audio_dma_start(substream); + return IRQ_HANDLED; +} + +/* PCM part */ +/* PCM hardware definition */ +static struct snd_pcm_hardware snd_sgio2audio_pcm_hw = { + .info = (SNDRV_PCM_INFO_MMAP | + SNDRV_PCM_INFO_MMAP_VALID | + SNDRV_PCM_INFO_INTERLEAVED | + SNDRV_PCM_INFO_BLOCK_TRANSFER), + .formats = SNDRV_PCM_FMTBIT_S16_BE, + .rates = SNDRV_PCM_RATE_8000_48000, + .rate_min = 8000, + .rate_max = 48000, + .channels_min = 2, + .channels_max = 2, + .buffer_bytes_max = 65536, + .period_bytes_min = 32768, + .period_bytes_max = 65536, + .periods_min = 1, + .periods_max = 1024, +}; + +/* PCM playback open callback */ +static int snd_sgio2audio_playback1_open(struct snd_pcm_substream *substream) +{ + struct snd_sgio2audio *chip = snd_pcm_substream_chip(substream); + struct snd_pcm_runtime *runtime = substream->runtime; + + runtime->hw = snd_sgio2audio_pcm_hw; + runtime->private_data = &chip->channel[1]; + return 0; +} + +static int snd_sgio2audio_playback2_open(struct snd_pcm_substream *substream) +{ + struct snd_sgio2audio *chip = snd_pcm_substream_chip(substream); + struct snd_pcm_runtime *runtime = substream->runtime; + + runtime->hw = snd_sgio2audio_pcm_hw; + runtime->private_data = &chip->channel[2]; + return 0; +} + +/* PCM capture open callback */ +static int snd_sgio2audio_capture_open(struct snd_pcm_substream *substream) +{ + struct snd_sgio2audio *chip = snd_pcm_substream_chip(substream); + struct snd_pcm_runtime *runtime = substream->runtime; + + runtime->hw = snd_sgio2audio_pcm_hw; + runtime->private_data = &chip->channel[0]; + return 0; +} + +/* PCM close callback */ +static int snd_sgio2audio_pcm_close(struct snd_pcm_substream *substream) +{ + struct snd_pcm_runtime *runtime = substream->runtime; + + runtime->private_data = NULL; + return 0; +} + + +/* hw_params callback */ +static int snd_sgio2audio_pcm_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *hw_params) +{ + struct snd_pcm_runtime *runtime = substream->runtime; + int size = params_buffer_bytes(hw_params); + + /* alloc virtual 'dma' area */ + if (runtime->dma_area) + vfree(runtime->dma_area); + runtime->dma_area = vmalloc(size); + if (runtime->dma_area == NULL) + return -ENOMEM; + runtime->dma_bytes = size; + return 0; +} + +/* hw_free callback */ +static int snd_sgio2audio_pcm_hw_free(struct snd_pcm_substream *substream) +{ + if (substream->runtime->dma_area) + vfree(substream->runtime->dma_area); + substream->runtime->dma_area = NULL; + return 0; +} + +/* prepare callback */ +static int snd_sgio2audio_pcm_prepare(struct snd_pcm_substream *substream) +{ + struct snd_sgio2audio *chip = snd_pcm_substream_chip(substream); + struct snd_pcm_runtime *runtime = substream->runtime; + struct snd_sgio2audio_chan *chan = substream->runtime->private_data; + int ch = chan->idx; + unsigned long flags; + + spin_lock_irqsave(&chip->channel[ch].lock, flags); + + /* Setup the pseudo-dma transfer pointers. */ + chip->channel[ch].pos = 0; + chip->channel[ch].size = 0; + chip->channel[ch].substream = substream; + + /* set AD1843 format */ + /* hardware format is always S16_LE */ + switch (substream->stream) { + case SNDRV_PCM_STREAM_PLAYBACK: + ad1843_setup_dac(&chip->ad1843, + ch - 1, + runtime->rate, + SNDRV_PCM_FORMAT_S16_LE, + runtime->channels); + break; + case SNDRV_PCM_STREAM_CAPTURE: + ad1843_setup_adc(&chip->ad1843, + runtime->rate, + SNDRV_PCM_FORMAT_S16_LE, + runtime->channels); + break; + } + spin_unlock_irqrestore(&chip->channel[ch].lock, flags); + return 0; +} + +/* trigger callback */ +static int snd_sgio2audio_pcm_trigger(struct snd_pcm_substream *substream, + int cmd) +{ + switch (cmd) { + case SNDRV_PCM_TRIGGER_START: + /* start the PCM engine */ + snd_sgio2audio_dma_start(substream); + break; + case SNDRV_PCM_TRIGGER_STOP: + /* stop the PCM engine */ + snd_sgio2audio_dma_stop(substream); + break; + default: + return -EINVAL; + } + return 0; +} + +/* pointer callback */ +static snd_pcm_uframes_t +snd_sgio2audio_pcm_pointer(struct snd_pcm_substream *substream) +{ + struct snd_sgio2audio *chip = snd_pcm_substream_chip(substream); + struct snd_sgio2audio_chan *chan = substream->runtime->private_data; + + /* get the current hardware pointer */ + return bytes_to_frames(substream->runtime, + chip->channel[chan->idx].pos); +} + +/* get the physical page pointer on the given offset */ +static struct page *snd_sgio2audio_page(struct snd_pcm_substream *substream, + unsigned long offset) +{ + return vmalloc_to_page(substream->runtime->dma_area + offset); +} + +/* operators */ +static struct snd_pcm_ops snd_sgio2audio_playback1_ops = { + .open = snd_sgio2audio_playback1_open, + .close = snd_sgio2audio_pcm_close, + .ioctl = snd_pcm_lib_ioctl, + .hw_params = snd_sgio2audio_pcm_hw_params, + .hw_free = snd_sgio2audio_pcm_hw_free, + .prepare = snd_sgio2audio_pcm_prepare, + .trigger = snd_sgio2audio_pcm_trigger, + .pointer = snd_sgio2audio_pcm_pointer, + .page = snd_sgio2audio_page, +}; + +static struct snd_pcm_ops snd_sgio2audio_playback2_ops = { + .open = snd_sgio2audio_playback2_open, + .close = snd_sgio2audio_pcm_close, + .ioctl = snd_pcm_lib_ioctl, + .hw_params = snd_sgio2audio_pcm_hw_params, + .hw_free = snd_sgio2audio_pcm_hw_free, + .prepare = snd_sgio2audio_pcm_prepare, + .trigger = snd_sgio2audio_pcm_trigger, + .pointer = snd_sgio2audio_pcm_pointer, + .page = snd_sgio2audio_page, +}; + +static struct snd_pcm_ops snd_sgio2audio_capture_ops = { + .open = snd_sgio2audio_capture_open, + .close = snd_sgio2audio_pcm_close, + .ioctl = snd_pcm_lib_ioctl, + .hw_params = snd_sgio2audio_pcm_hw_params, + .hw_free = snd_sgio2audio_pcm_hw_free, + .prepare = snd_sgio2audio_pcm_prepare, + .trigger = snd_sgio2audio_pcm_trigger, + .pointer = snd_sgio2audio_pcm_pointer, + .page = snd_sgio2audio_page, +}; + +/* + * definitions of capture are omitted here... + */ + +/* create a pcm device */ +static int __devinit snd_sgio2audio_new_pcm(struct snd_sgio2audio *chip) +{ + struct snd_pcm *pcm; + int err; + + /* create first pcm device with one outputs and one input */ + err = snd_pcm_new(chip->card, "SGI O2 Audio", 0, 1, 1, &pcm); + if (err < 0) + return err; + + pcm->private_data = chip; + strcpy(pcm->name, "SGI O2 DAC1"); + + /* set operators */ + snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_PLAYBACK, + &snd_sgio2audio_playback1_ops); + snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_CAPTURE, + &snd_sgio2audio_capture_ops); + + /* create second pcm device with one outputs and no input */ + err = snd_pcm_new(chip->card, "SGI O2 Audio", 1, 1, 0, &pcm); + if (err < 0) + return err; + + pcm->private_data = chip; + strcpy(pcm->name, "SGI O2 DAC2"); + + /* set operators */ + snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_PLAYBACK, + &snd_sgio2audio_playback2_ops); + + return 0; +} + +static struct { + int idx; + int irq; + irqreturn_t (*isr)(int, void *); + const char *desc; +} snd_sgio2_isr_table[] = { + { + .idx = 0, + .irq = MACEISA_AUDIO1_DMAT_IRQ, + .isr = snd_sgio2audio_dma_in_isr, + .desc = "Capture DMA Channel 0" + }, { + .idx = 0, + .irq = MACEISA_AUDIO1_OF_IRQ, + .isr = snd_sgio2audio_error_isr, + .desc = "Capture Overflow" + }, { + .idx = 1, + .irq = MACEISA_AUDIO2_DMAT_IRQ, + .isr = snd_sgio2audio_dma_out_isr, + .desc = "Playback DMA Channel 1" + }, { + .idx = 1, + .irq = MACEISA_AUDIO2_MERR_IRQ, + .isr = snd_sgio2audio_error_isr, + .desc = "Memory Error Channel 1" + }, { + .idx = 2, + .irq = MACEISA_AUDIO3_DMAT_IRQ, + .isr = snd_sgio2audio_dma_out_isr, + .desc = "Playback DMA Channel 2" + }, { + .idx = 2, + .irq = MACEISA_AUDIO3_MERR_IRQ, + .isr = snd_sgio2audio_error_isr, + .desc = "Memory Error Channel 2" + } +}; + +/* ALSA driver */ + +static int snd_sgio2audio_free(struct snd_sgio2audio *chip) +{ + int i; + + /* reset interface */ + writeq(AUDIO_CONTROL_RESET, &mace->perif.audio.control); + udelay(1); + writeq(0, &mace->perif.audio.control); + + /* release IRQ's */ + for (i = 0; i < ARRAY_SIZE(snd_sgio2_isr_table); i++) + free_irq(snd_sgio2_isr_table[i].irq, + &chip->channel[snd_sgio2_isr_table[i].idx]); + + dma_free_coherent(NULL, MACEISA_RINGBUFFERS_SIZE, + chip->ring_base, chip->ring_base_dma); + + /* release card data */ + kfree(chip); + return 0; +} + +static int snd_sgio2audio_dev_free(struct snd_device *device) +{ + struct snd_sgio2audio *chip = device->device_data; + + return snd_sgio2audio_free(chip); +} + +static struct snd_device_ops ops = { + .dev_free = snd_sgio2audio_dev_free, +}; + +static int __devinit snd_sgio2audio_create(struct snd_card *card, + struct snd_sgio2audio **rchip) +{ + struct snd_sgio2audio *chip; + int i, err; + + *rchip = NULL; + + /* check if a codec is attached to the interface */ + /* (Audio or Audio/Video board present) */ + if (!(readq(&mace->perif.audio.control) & AUDIO_CONTROL_CODEC_PRESENT)) + return -ENOENT; + + chip = kzalloc(sizeof(struct snd_sgio2audio), GFP_KERNEL); + if (chip == NULL) + return -ENOMEM; + + chip->card = card; + + chip->ring_base = dma_alloc_coherent(NULL, MACEISA_RINGBUFFERS_SIZE, + &chip->ring_base_dma, GFP_USER); + if (chip->ring_base == NULL) { + printk(KERN_ERR + "sgio2audio: could not allocate ring buffers\n"); + kfree(chip); + return -ENOMEM; + } + + spin_lock_init(&chip->ad1843_lock); + + /* initialize channels */ + for (i = 0; i < 3; i++) { + spin_lock_init(&chip->channel[i].lock); + chip->channel[i].idx = i; + } + + /* allocate IRQs */ + for (i = 0; i < ARRAY_SIZE(snd_sgio2_isr_table); i++) { + if (request_irq(snd_sgio2_isr_table[i].irq, + snd_sgio2_isr_table[i].isr, + 0, + snd_sgio2_isr_table[i].desc, + &chip->channel[snd_sgio2_isr_table[i].idx])) { + snd_sgio2audio_free(chip); + printk(KERN_ERR "sgio2audio: cannot allocate irq %d\n", + snd_sgio2_isr_table[i].irq); + return -EBUSY; + } + } + + /* reset the interface */ + writeq(AUDIO_CONTROL_RESET, &mace->perif.audio.control); + udelay(1); + writeq(0, &mace->perif.audio.control); + msleep_interruptible(1); /* give time to recover */ + + /* set ring base */ + writeq(chip->ring_base_dma, &mace->perif.ctrl.ringbase); + + /* attach the AD1843 codec */ + chip->ad1843.read = read_ad1843_reg; + chip->ad1843.write = write_ad1843_reg; + chip->ad1843.chip = chip; + + /* initialize the AD1843 codec */ + err = ad1843_init(&chip->ad1843); + if (err < 0) { + snd_sgio2audio_free(chip); + return err; + } + + err = snd_device_new(card, SNDRV_DEV_LOWLEVEL, chip, &ops); + if (err < 0) { + snd_sgio2audio_free(chip); + return err; + } + *rchip = chip; + return 0; +} + +static int __devinit snd_sgio2audio_probe(struct platform_device *pdev) +{ + struct snd_card *card; + struct snd_sgio2audio *chip; + int err; + + card = snd_card_new(index, id, THIS_MODULE, 0); + if (card == NULL) + return -ENOMEM; + + err = snd_sgio2audio_create(card, &chip); + if (err < 0) { + snd_card_free(card); + return err; + } + snd_card_set_dev(card, &pdev->dev); + + err = snd_sgio2audio_new_pcm(chip); + if (err < 0) { + snd_card_free(card); + return err; + } + err = snd_sgio2audio_new_mixer(chip); + if (err < 0) { + snd_card_free(card); + return err; + } + + strcpy(card->driver, "SGI O2 Audio"); + strcpy(card->shortname, "SGI O2 Audio"); + sprintf(card->longname, "%s irq %i-%i", + card->shortname, + MACEISA_AUDIO1_DMAT_IRQ, + MACEISA_AUDIO3_MERR_IRQ); + + err = snd_card_register(card); + if (err < 0) { + snd_card_free(card); + return err; + } + platform_set_drvdata(pdev, card); + return 0; +} + +static int __exit snd_sgio2audio_remove(struct platform_device *pdev) +{ + struct snd_card *card = platform_get_drvdata(pdev); + + snd_card_free(card); + platform_set_drvdata(pdev, NULL); + return 0; +} + +static struct platform_driver sgio2audio_driver = { + .probe = snd_sgio2audio_probe, + .remove = __devexit_p(snd_sgio2audio_remove), + .driver = { + .name = "sgio2audio", + .owner = THIS_MODULE, + } +}; + +static int __init alsa_card_sgio2audio_init(void) +{ + return platform_driver_register(&sgio2audio_driver); +} + +static void __exit alsa_card_sgio2audio_exit(void) +{ + platform_driver_unregister(&sgio2audio_driver); +} + +module_init(alsa_card_sgio2audio_init) +module_exit(alsa_card_sgio2audio_exit) diff --git a/sound/oss/Kconfig b/sound/oss/Kconfig index 3be2dc1025b..33940139844 100644 --- a/sound/oss/Kconfig +++ b/sound/oss/Kconfig @@ -7,7 +7,7 @@ config SOUND_BCM_CS4297A tristate "Crystal Sound CS4297a (for Swarm)" - depends on SOUND_PRIME && SIBYTE_SWARM + depends on SIBYTE_SWARM help The BCM91250A has a Crystal CS4297a on synchronous serial port B (in addition to the DB-9 serial port). Say Y or M @@ -17,7 +17,7 @@ config SOUND_BCM_CS4297A config SOUND_VWSND tristate "SGI Visual Workstation Sound" - depends on SOUND_PRIME && X86_VISWS + depends on X86_VISWS help Say Y or M if you have an SGI Visual Workstation and you want to be able to use its on-board audio. Read @@ -26,19 +26,18 @@ config SOUND_VWSND config SOUND_HAL2 tristate "SGI HAL2 sound (EXPERIMENTAL)" - depends on SOUND_PRIME && SGI_IP22 && EXPERIMENTAL + depends on SGI_IP22 && EXPERIMENTAL help Say Y or M if you have an SGI Indy or Indigo2 system and want to be able to use its on-board A2 audio system. config SOUND_AU1550_AC97 tristate "Au1550/Au1200 AC97 Sound" - select SND_AC97_CODEC - depends on SOUND_PRIME && (SOC_AU1550 || SOC_AU1200) + depends on SOC_AU1550 || SOC_AU1200 config SOUND_TRIDENT tristate "Trident 4DWave DX/NX, SiS 7018 or ALi 5451 PCI Audio Core" - depends on SOUND_PRIME && PCI + depends on PCI ---help--- Say Y or M if you have a PCI sound card utilizing the Trident 4DWave-DX/NX chipset or your mother board chipset has SiS 7018 @@ -79,7 +78,7 @@ config SOUND_TRIDENT config SOUND_MSNDCLAS tristate "Support for Turtle Beach MultiSound Classic, Tahiti, Monterey" - depends on SOUND_PRIME && (m || !STANDALONE) && ISA + depends on (m || !STANDALONE) && ISA help Say M here if you have a Turtle Beach MultiSound Classic, Tahiti or Monterey (not for the Pinnacle or Fiji). @@ -143,7 +142,7 @@ config MSNDCLAS_IO config SOUND_MSNDPIN tristate "Support for Turtle Beach MultiSound Pinnacle, Fiji" - depends on SOUND_PRIME && (m || !STANDALONE) && ISA + depends on (m || !STANDALONE) && ISA help Say M here if you have a Turtle Beach MultiSound Pinnacle or Fiji. See <file:Documentation/sound/oss/MultiSound> for important information @@ -229,7 +228,7 @@ config MSNDPIN_NONPNP configure the card's resources. comment "MSND Pinnacle DSP section will be configured to above parameters." - depends on SOUND_PRIME && SOUND_MSNDPIN=y && MSNDPIN_NONPNP + depends on SOUND_MSNDPIN=y && MSNDPIN_NONPNP config MSNDPIN_CFG hex "MSND Pinnacle config port 250,260,270" @@ -242,7 +241,7 @@ config MSNDPIN_CFG Mode". comment "Pinnacle-specific Device Configuration (0 disables)" - depends on SOUND_PRIME && SOUND_MSNDPIN=y && MSNDPIN_NONPNP + depends on SOUND_MSNDPIN=y && MSNDPIN_NONPNP config MSNDPIN_MPU_IO hex "MSND Pinnacle MPU I/O (e.g. 330)" @@ -294,7 +293,7 @@ config MSNDPIN_JOYSTICK_IO config MSND_FIFOSIZE int "MSND buffer size (kB)" - depends on SOUND_PRIME && (SOUND_MSNDPIN=y || SOUND_MSNDCLAS=y) + depends on SOUND_MSNDPIN=y || SOUND_MSNDCLAS=y default "128" help Configures the size of each audio buffer, in kilobytes, for @@ -302,9 +301,9 @@ config MSND_FIFOSIZE and Pinnacle). Larger values reduce the chance of data overruns at the expense of overall latency. If unsure, use the default. -config SOUND_OSS +menuconfig SOUND_OSS tristate "OSS sound modules" - depends on SOUND_PRIME && ISA_DMA_API && VIRT_TO_BUS + depends on ISA_DMA_API && VIRT_TO_BUS help OSS is the Open Sound System suite of sound card drivers. They make sound programming easier since they provide a common API. Say Y or @@ -312,16 +311,16 @@ config SOUND_OSS driver for your sound card above, then pick your driver from the list below. +if SOUND_OSS + config SOUND_TRACEINIT bool "Verbose initialisation" - depends on SOUND_OSS help Verbose soundcard initialization -- affects the format of autoprobe and initialization messages at boot time. config SOUND_DMAP bool "Persistent DMA buffers" - depends on SOUND_OSS ---help--- Linux can often have problems allocating DMA buffers for ISA sound cards on machines with more than 16MB of RAM. This is because ISA @@ -338,8 +337,6 @@ config SOUND_DMAP config SOUND_SSCAPE tristate "Ensoniq SoundScape support" - depends on SOUND_OSS - depends on VIRT_TO_BUS help Answer Y if you have a sound card based on the Ensoniq SoundScape chipset. Such cards are being manufactured at least by Ensoniq, Spea @@ -352,13 +349,11 @@ config SOUND_SSCAPE config SOUND_VMIDI tristate "Loopback MIDI device support" - depends on SOUND_OSS help Support for MIDI loopback on port 1 or 2. config SOUND_TRIX tristate "MediaTrix AudioTrix Pro support" - depends on SOUND_OSS help Answer Y if you have the AudioTriX Pro sound card manufactured by MediaTrix. @@ -382,7 +377,6 @@ config TRIX_BOOT_FILE config SOUND_MSS tristate "Microsoft Sound System support" - depends on SOUND_OSS ---help--- Again think carefully before answering Y to this question. It's safe to answer Y if you have the original Windows Sound System card @@ -414,7 +408,6 @@ config SOUND_MSS config SOUND_MPU401 tristate "MPU-401 support (NOT for SB16)" - depends on SOUND_OSS ---help--- Be careful with this question. The MPU401 interface is supported by all sound cards. However, some natively supported cards have their @@ -430,7 +423,6 @@ config SOUND_MPU401 config SOUND_PAS tristate "ProAudioSpectrum 16 support" - depends on SOUND_OSS ---help--- Answer Y only if you have a Pro Audio Spectrum 16, ProAudio Studio 16 or Logitech SoundMan 16 sound card. Answer N if you have some @@ -452,7 +444,6 @@ config PAS_JOYSTICK config SOUND_PSS tristate "PSS (AD1848, ADSP-2115, ESC614) support" - depends on SOUND_OSS help Answer Y or M if you have an Orchid SW32, Cardinal DSP16, Beethoven ADSP-16 or some other card based on the PSS chipset (AD1848 codec + @@ -495,7 +486,6 @@ config PSS_BOOT_FILE config SOUND_SB tristate "100% Sound Blaster compatibles (SB16/32/64, ESS, Jazz16) support" - depends on SOUND_OSS ---help--- Answer Y if you have an original Sound Blaster card made by Creative Labs or a 100% hardware compatible clone (like the Thunderboard or @@ -522,7 +512,6 @@ config SOUND_SB config SOUND_YM3812 tristate "Yamaha FM synthesizer (YM3812/OPL-3) support" - depends on SOUND_OSS ---help--- Answer Y if your card has a FM chip made by Yamaha (OPL2/OPL3/OPL4). Answering Y is usually a safe and recommended choice, however some @@ -538,7 +527,6 @@ config SOUND_YM3812 config SOUND_UART6850 tristate "6850 UART support" - depends on SOUND_OSS help This option enables support for MIDI interfaces based on the 6850 UART chip. This interface is rarely found on sound cards. It's safe @@ -549,7 +537,6 @@ config SOUND_UART6850 config SOUND_AEDSP16 tristate "Gallant Audio Cards (SC-6000 and SC-6600 based)" - depends on SOUND_OSS ---help--- Answer Y if you have a Gallant's Audio Excel DSP 16 card. This driver supports Audio Excel DSP 16 but not the III nor PnP versions @@ -630,14 +617,14 @@ endchoice config SOUND_VIDC tristate "VIDC 16-bit sound" - depends on ARM && (ARCH_ACORN || ARCH_CLPS7500) && SOUND_OSS + depends on ARM && (ARCH_ACORN || ARCH_CLPS7500) help 16-bit support for the VIDC onboard sound hardware found on Acorn machines. config SOUND_WAVEARTIST tristate "Netwinder WaveArtist" - depends on ARM && SOUND_OSS && ARCH_NETWINDER + depends on ARM && ARCH_NETWINDER help Say Y here to include support for the Rockwell WaveArtist sound system. This driver is mainly for the NetWinder. @@ -646,9 +633,11 @@ config SOUND_KAHLUA tristate "XpressAudio Sound Blaster emulation" depends on SOUND_SB +endif # SOUND_OSS + config SOUND_SH_DAC_AUDIO tristate "SuperH DAC audio support" - depends on SOUND_PRIME && CPU_SH3 + depends on CPU_SH3 config SOUND_SH_DAC_AUDIO_CHANNEL int "DAC channel" diff --git a/sound/oss/dmasound/dmasound_core.c b/sound/oss/dmasound/dmasound_core.c index a003c0ea930..95fc5c68175 100644 --- a/sound/oss/dmasound/dmasound_core.c +++ b/sound/oss/dmasound/dmasound_core.c @@ -211,10 +211,6 @@ static int state_unit = -1; static int irq_installed; #endif /* MODULE */ -/* software implemented recording volume! */ -uint software_input_volume = SW_INPUT_VOLUME_SCALE * SW_INPUT_VOLUME_DEFAULT; -EXPORT_SYMBOL(software_input_volume); - /* control over who can modify resources shared between play/record */ static mode_t shared_resource_owner; static int shared_resources_initialised; @@ -1188,7 +1184,7 @@ static struct { /* publish this function for use by low-level code, if required */ -char *get_afmt_string(int afmt) +static char *get_afmt_string(int afmt) { switch(afmt) { case AFMT_MU_LAW: @@ -1551,4 +1547,3 @@ EXPORT_SYMBOL(dmasound_catchRadius); EXPORT_SYMBOL(dmasound_ulaw2dma8); EXPORT_SYMBOL(dmasound_alaw2dma8); #endif -EXPORT_SYMBOL(get_afmt_string) ; diff --git a/sound/oss/dmasound/dmasound_paula.c b/sound/oss/dmasound/dmasound_paula.c index 202e8103dc4..06e9e88e4c0 100644 --- a/sound/oss/dmasound/dmasound_paula.c +++ b/sound/oss/dmasound/dmasound_paula.c @@ -710,7 +710,7 @@ static MACHINE machAmiga = { /*** Config & Setup **********************************************************/ -int __init dmasound_paula_init(void) +static int __init dmasound_paula_init(void) { int err; diff --git a/sound/oss/dmasound/dmasound_q40.c b/sound/oss/dmasound/dmasound_q40.c index b3379dd7ca5..1855b14d90c 100644 --- a/sound/oss/dmasound/dmasound_q40.c +++ b/sound/oss/dmasound/dmasound_q40.c @@ -611,7 +611,7 @@ static MACHINE machQ40 = { /*** Config & Setup **********************************************************/ -int __init dmasound_q40_init(void) +static int __init dmasound_q40_init(void) { if (MACH_IS_Q40) { dmasound.mach = machQ40; diff --git a/sound/oss/msnd.c b/sound/oss/msnd.c index ba38d620009..e4282d93a1a 100644 --- a/sound/oss/msnd.c +++ b/sound/oss/msnd.c @@ -20,8 +20,6 @@ * along with this program; if not, write to the Free Software * Foundation, Inc., 675 Mass Ave, Cambridge, MA 02139, USA. * - * $Id: msnd.c,v 1.17 1999/03/21 16:50:09 andrewtv Exp $ - * ********************************************************************/ #include <linux/module.h> diff --git a/sound/oss/msnd.h b/sound/oss/msnd.h index d0ca582c458..61b3955481c 100644 --- a/sound/oss/msnd.h +++ b/sound/oss/msnd.h @@ -24,8 +24,6 @@ * along with this program; if not, write to the Free Software * Foundation, Inc., 675 Mass Ave, Cambridge, MA 02139, USA. * - * $Id: msnd.h,v 1.36 1999/03/21 17:05:42 andrewtv Exp $ - * ********************************************************************/ #ifndef __MSND_H #define __MSND_H diff --git a/sound/oss/msnd_classic.h b/sound/oss/msnd_classic.h index 7ffea5267f9..1a17dde2f65 100644 --- a/sound/oss/msnd_classic.h +++ b/sound/oss/msnd_classic.h @@ -24,8 +24,6 @@ * along with this program; if not, write to the Free Software * Foundation, Inc., 675 Mass Ave, Cambridge, MA 02139, USA. * - * $Id: msnd_classic.h,v 1.10 1999/03/21 17:36:09 andrewtv Exp $ - * ********************************************************************/ #ifndef __MSND_CLASSIC_H #define __MSND_CLASSIC_H diff --git a/sound/oss/msnd_pinnacle.c b/sound/oss/msnd_pinnacle.c index f1f49ebf752..bf27e008f46 100644 --- a/sound/oss/msnd_pinnacle.c +++ b/sound/oss/msnd_pinnacle.c @@ -29,13 +29,8 @@ * along with this program; if not, write to the Free Software * Foundation, Inc., 675 Mass Ave, Cambridge, MA 02139, USA. * - * $Id: msnd_pinnacle.c,v 1.8 2000/12/30 00:33:21 sycamore Exp $ - * * 12-3-2000 Modified IO port validation Steve Sycamore * - * - * $$$: msnd_pinnacle.c,v 1.75 1999/03/21 16:50:09 andrewtv $$$ $ - * ********************************************************************/ #include <linux/kernel.h> diff --git a/sound/oss/msnd_pinnacle.h b/sound/oss/msnd_pinnacle.h index cce91148700..c18d66cbbe3 100644 --- a/sound/oss/msnd_pinnacle.h +++ b/sound/oss/msnd_pinnacle.h @@ -24,8 +24,6 @@ * along with this program; if not, write to the Free Software * Foundation, Inc., 675 Mass Ave, Cambridge, MA 02139, USA. * - * $Id: msnd_pinnacle.h,v 1.11 1999/03/21 17:36:09 andrewtv Exp $ - * ********************************************************************/ #ifndef __MSND_PINNACLE_H #define __MSND_PINNACLE_H diff --git a/sound/oss/vwsnd.c b/sound/oss/vwsnd.c index 2c5aaa58046..dcbb3f739e6 100644 --- a/sound/oss/vwsnd.c +++ b/sound/oss/vwsnd.c @@ -150,7 +150,7 @@ #include <linux/interrupt.h> #include <linux/mutex.h> -#include <asm/mach-visws/cobalt.h> +#include <asm/visws/cobalt.h> #include "sound_config.h" diff --git a/sound/parisc/Kconfig b/sound/parisc/Kconfig index a5a7f9d75d0..9b61d95010f 100644 --- a/sound/parisc/Kconfig +++ b/sound/parisc/Kconfig @@ -1,15 +1,20 @@ # ALSA PA-RISC drivers -menu "GSC devices" - depends on SND!=n && GSC +menuconfig SND_GSC + bool "GSC sound devices" + depends on GSC + default y + help + Support for GSC sound devices on PA-RISC architectures. + +if SND_GSC config SND_HARMONY tristate "Harmony/Vivace sound chip" - depends on SND select SND_PCM help Say 'Y' or 'M' to include support for the Harmony/Vivace sound chip found in most GSC-based PA-RISC workstations. It's frequently provided as part of the Lasi multi-function IC. -endmenu +endif # SND_GSC diff --git a/sound/pci/Kconfig b/sound/pci/Kconfig index 7e474210957..8fe5dac3942 100644 --- a/sound/pci/Kconfig +++ b/sound/pci/Kconfig @@ -1,11 +1,16 @@ # ALSA PCI drivers -menu "PCI devices" - depends on SND!=n && PCI +menuconfig SND_PCI + bool "PCI sound devices" + depends on PCI + default y + help + Support for sound devices connected via the PCI bus. + +if SND_PCI config SND_AD1889 tristate "Analog Devices AD1889" - depends on SND select SND_AC97_CODEC help Say Y here to include support for the integrated AC97 sound @@ -17,7 +22,6 @@ config SND_AD1889 config SND_ALS300 tristate "Avance Logic ALS300/ALS300+" - depends on SND select SND_PCM select SND_AC97_CODEC select SND_OPL3_LIB @@ -29,7 +33,7 @@ config SND_ALS300 config SND_ALS4000 tristate "Avance Logic ALS4000" - depends on SND && ISA_DMA_API + depends on ISA_DMA_API select SND_OPL3_LIB select SND_MPU401_UART select SND_PCM @@ -43,7 +47,6 @@ config SND_ALS4000 config SND_ALI5451 tristate "ALi M5451 PCI Audio Controller" - depends on SND select SND_MPU401_UART select SND_AC97_CODEC help @@ -57,7 +60,6 @@ config SND_ALI5451 config SND_ATIIXP tristate "ATI IXP AC97 Controller" - depends on SND select SND_AC97_CODEC help Say Y here to include support for the integrated AC97 sound @@ -69,7 +71,6 @@ config SND_ATIIXP config SND_ATIIXP_MODEM tristate "ATI IXP Modem" - depends on SND select SND_AC97_CODEC help Say Y here to include support for the integrated MC97 modem on @@ -80,7 +81,6 @@ config SND_ATIIXP_MODEM config SND_AU8810 tristate "Aureal Advantage" - depends on SND select SND_MPU401_UART select SND_AC97_CODEC help @@ -95,7 +95,6 @@ config SND_AU8810 config SND_AU8820 tristate "Aureal Vortex" - depends on SND select SND_MPU401_UART select SND_AC97_CODEC help @@ -109,7 +108,6 @@ config SND_AU8820 config SND_AU8830 tristate "Aureal Vortex 2" - depends on SND select SND_MPU401_UART select SND_AC97_CODEC help @@ -124,7 +122,6 @@ config SND_AU8830 config SND_AW2 tristate "Emagic Audiowerk 2" - depends on SND help Say Y here to include support for Emagic Audiowerk 2 soundcards. @@ -139,7 +136,7 @@ config SND_AW2 config SND_AZT3328 tristate "Aztech AZF3328 / PCI168 (EXPERIMENTAL)" - depends on SND && EXPERIMENTAL + depends on EXPERIMENTAL select SND_OPL3_LIB select SND_MPU401_UART select SND_PCM @@ -152,7 +149,6 @@ config SND_AZT3328 config SND_BT87X tristate "Bt87x Audio Capture" - depends on SND select SND_PCM help If you want to record audio from TV cards based on @@ -174,7 +170,6 @@ config SND_BT87X_OVERCLOCK config SND_CA0106 tristate "SB Audigy LS / Live 24bit" - depends on SND select SND_AC97_CODEC select SND_RAWMIDI select SND_VMASTER @@ -187,7 +182,6 @@ config SND_CA0106 config SND_CMIPCI tristate "C-Media 8338, 8738, 8768, 8770" - depends on SND select SND_OPL3_LIB select SND_MPU401_UART select SND_PCM @@ -201,13 +195,11 @@ config SND_CMIPCI config SND_OXYGEN_LIB tristate - depends on SND select SND_PCM select SND_MPU401_UART config SND_OXYGEN tristate "C-Media 8788 (Oxygen)" - depends on SND select SND_OXYGEN_LIB help Say Y here to include support for sound cards based on the @@ -225,7 +217,6 @@ config SND_OXYGEN config SND_CS4281 tristate "Cirrus Logic (Sound Fusion) CS4281" - depends on SND select SND_OPL3_LIB select SND_RAWMIDI select SND_AC97_CODEC @@ -237,7 +228,6 @@ config SND_CS4281 config SND_CS46XX tristate "Cirrus Logic (Sound Fusion) CS4280/CS461x/CS462x/CS463x" - depends on SND select SND_RAWMIDI select SND_AC97_CODEC help @@ -258,7 +248,7 @@ config SND_CS46XX_NEW_DSP config SND_CS5530 tristate "CS5530 Audio" - depends on SND && ISA_DMA_API + depends on ISA_DMA_API select SND_SB16_DSP help Say Y here to include support for audio on Cyrix/NatSemi CS5530 chips. @@ -268,7 +258,7 @@ config SND_CS5530 config SND_CS5535AUDIO tristate "CS5535/CS5536 Audio" - depends on SND && X86 && !X86_64 + depends on X86 && !X86_64 select SND_PCM select SND_AC97_CODEC help @@ -286,7 +276,6 @@ config SND_CS5535AUDIO config SND_DARLA20 tristate "(Echoaudio) Darla20" - depends on SND select FW_LOADER select SND_PCM help @@ -297,7 +286,6 @@ config SND_DARLA20 config SND_GINA20 tristate "(Echoaudio) Gina20" - depends on SND select FW_LOADER select SND_PCM help @@ -308,7 +296,6 @@ config SND_GINA20 config SND_LAYLA20 tristate "(Echoaudio) Layla20" - depends on SND select FW_LOADER select SND_RAWMIDI select SND_PCM @@ -320,7 +307,6 @@ config SND_LAYLA20 config SND_DARLA24 tristate "(Echoaudio) Darla24" - depends on SND select FW_LOADER select SND_PCM help @@ -331,7 +317,6 @@ config SND_DARLA24 config SND_GINA24 tristate "(Echoaudio) Gina24" - depends on SND select FW_LOADER select SND_PCM help @@ -342,7 +327,6 @@ config SND_GINA24 config SND_LAYLA24 tristate "(Echoaudio) Layla24" - depends on SND select FW_LOADER select SND_RAWMIDI select SND_PCM @@ -354,7 +338,6 @@ config SND_LAYLA24 config SND_MONA tristate "(Echoaudio) Mona" - depends on SND select FW_LOADER select SND_RAWMIDI select SND_PCM @@ -366,7 +349,6 @@ config SND_MONA config SND_MIA tristate "(Echoaudio) Mia" - depends on SND select FW_LOADER select SND_RAWMIDI select SND_PCM @@ -378,7 +360,6 @@ config SND_MIA config SND_ECHO3G tristate "(Echoaudio) 3G cards" - depends on SND select FW_LOADER select SND_RAWMIDI select SND_PCM @@ -390,7 +371,6 @@ config SND_ECHO3G config SND_INDIGO tristate "(Echoaudio) Indigo" - depends on SND select FW_LOADER select SND_PCM help @@ -401,7 +381,6 @@ config SND_INDIGO config SND_INDIGOIO tristate "(Echoaudio) Indigo IO" - depends on SND select FW_LOADER select SND_PCM help @@ -412,7 +391,6 @@ config SND_INDIGOIO config SND_INDIGODJ tristate "(Echoaudio) Indigo DJ" - depends on SND select FW_LOADER select SND_PCM help @@ -423,7 +401,6 @@ config SND_INDIGODJ config SND_EMU10K1 tristate "Emu10k1 (SB Live!, Audigy, E-mu APS)" - depends on SND select FW_LOADER select SND_HWDEP select SND_RAWMIDI @@ -441,7 +418,6 @@ config SND_EMU10K1 config SND_EMU10K1X tristate "Emu10k1X (Dell OEM Version)" - depends on SND select SND_AC97_CODEC select SND_RAWMIDI help @@ -453,7 +429,6 @@ config SND_EMU10K1X config SND_ENS1370 tristate "(Creative) Ensoniq AudioPCI 1370" - depends on SND select SND_RAWMIDI select SND_PCM help @@ -464,7 +439,6 @@ config SND_ENS1370 config SND_ENS1371 tristate "(Creative) Ensoniq AudioPCI 1371/1373" - depends on SND select SND_RAWMIDI select SND_AC97_CODEC help @@ -476,7 +450,6 @@ config SND_ENS1371 config SND_ES1938 tristate "ESS ES1938/1946/1969 (Solo-1)" - depends on SND select SND_OPL3_LIB select SND_MPU401_UART select SND_AC97_CODEC @@ -489,7 +462,6 @@ config SND_ES1938 config SND_ES1968 tristate "ESS ES1968/1978 (Maestro-1/2/2E)" - depends on SND select SND_MPU401_UART select SND_AC97_CODEC help @@ -501,7 +473,6 @@ config SND_ES1968 config SND_FM801 tristate "ForteMedia FM801" - depends on SND select SND_OPL3_LIB select SND_MPU401_UART select SND_AC97_CODEC @@ -528,7 +499,6 @@ config SND_FM801_TEA575X config SND_HDA_INTEL tristate "Intel HD Audio" - depends on SND select SND_PCM select SND_VMASTER help @@ -637,7 +607,6 @@ config SND_HDA_POWER_SAVE_DEFAULT config SND_HDSP tristate "RME Hammerfall DSP Audio" - depends on SND select SND_HWDEP select SND_RAWMIDI select SND_PCM @@ -650,7 +619,6 @@ config SND_HDSP config SND_HDSPM tristate "RME Hammerfall DSP MADI" - depends on SND select SND_HWDEP select SND_RAWMIDI select SND_PCM @@ -663,7 +631,6 @@ config SND_HDSPM config SND_HIFIER tristate "TempoTec HiFier Fantasia" - depends on SND select SND_OXYGEN_LIB help Say Y here to include support for the MediaTek/TempoTec HiFier @@ -674,7 +641,6 @@ config SND_HIFIER config SND_ICE1712 tristate "ICEnsemble ICE1712 (Envy24)" - depends on SND select SND_MPU401_UART select SND_AC97_CODEC help @@ -691,8 +657,7 @@ config SND_ICE1712 config SND_ICE1724 tristate "ICE/VT1724/1720 (Envy24HT/PT)" - depends on SND - select SND_MPU401_UART + select SND_RAWMIDI select SND_AC97_CODEC select SND_VMASTER help @@ -709,7 +674,6 @@ config SND_ICE1724 config SND_INTEL8X0 tristate "Intel/SiS/nVidia/AMD/ALi AC97 Controller" - depends on SND select SND_AC97_CODEC help Say Y here to include support for the integrated AC97 sound @@ -722,7 +686,6 @@ config SND_INTEL8X0 config SND_INTEL8X0M tristate "Intel/SiS/nVidia/AMD MC97 Modem" - depends on SND select SND_AC97_CODEC help Say Y here to include support for the integrated MC97 modem on @@ -733,7 +696,6 @@ config SND_INTEL8X0M config SND_KORG1212 tristate "Korg 1212 IO" - depends on SND select FW_LOADER if !SND_KORG1212_FIRMWARE_IN_KERNEL select SND_PCM help @@ -753,7 +715,6 @@ config SND_KORG1212_FIRMWARE_IN_KERNEL config SND_MAESTRO3 tristate "ESS Allegro/Maestro3" - depends on SND select FW_LOADER if !SND_MAESTRO3_FIRMWARE_IN_KERNEL select SND_AC97_CODEC help @@ -774,7 +735,6 @@ config SND_MAESTRO3_FIRMWARE_IN_KERNEL config SND_MIXART tristate "Digigram miXart" - depends on SND select SND_HWDEP select SND_PCM help @@ -786,7 +746,6 @@ config SND_MIXART config SND_NM256 tristate "NeoMagic NM256AV/ZX" - depends on SND select SND_AC97_CODEC help Say Y here to include support for NeoMagic NM256AV/ZX chips. @@ -796,7 +755,6 @@ config SND_NM256 config SND_PCXHR tristate "Digigram PCXHR" - depends on SND select SND_PCM select SND_HWDEP help @@ -807,7 +765,6 @@ config SND_PCXHR config SND_RIPTIDE tristate "Conexant Riptide" - depends on SND select FW_LOADER select SND_OPL3_LIB select SND_MPU401_UART @@ -820,7 +777,6 @@ config SND_RIPTIDE config SND_RME32 tristate "RME Digi32, 32/8, 32 PRO" - depends on SND select SND_PCM help Say Y to include support for RME Digi32, Digi32 PRO and @@ -832,7 +788,6 @@ config SND_RME32 config SND_RME96 tristate "RME Digi96, 96/8, 96/8 PRO" - depends on SND select SND_PCM help Say Y here to include support for RME Digi96, Digi96/8 and @@ -843,7 +798,6 @@ config SND_RME96 config SND_RME9652 tristate "RME Digi9652 (Hammerfall)" - depends on SND select SND_PCM help Say Y here to include support for RME Hammerfall (RME @@ -854,7 +808,7 @@ config SND_RME9652 config SND_SIS7019 tristate "SiS 7019 Audio Accelerator" - depends on SND && X86 && !X86_64 + depends on X86 && !X86_64 select SND_AC97_CODEC help Say Y here to include support for the SiS 7019 Audio Accelerator. @@ -864,7 +818,6 @@ config SND_SIS7019 config SND_SONICVIBES tristate "S3 SonicVibes" - depends on SND select SND_OPL3_LIB select SND_MPU401_UART select SND_AC97_CODEC @@ -877,7 +830,6 @@ config SND_SONICVIBES config SND_TRIDENT tristate "Trident 4D-Wave DX/NX; SiS 7018" - depends on SND select SND_MPU401_UART select SND_AC97_CODEC help @@ -889,7 +841,6 @@ config SND_TRIDENT config SND_VIA82XX tristate "VIA 82C686A/B, 8233/8235 AC97 Controller" - depends on SND select SND_MPU401_UART select SND_AC97_CODEC help @@ -901,7 +852,6 @@ config SND_VIA82XX config SND_VIA82XX_MODEM tristate "VIA 82C686A/B, 8233 based Modems" - depends on SND select SND_AC97_CODEC help Say Y here to include support for the integrated MC97 modem on @@ -912,7 +862,6 @@ config SND_VIA82XX_MODEM config SND_VIRTUOSO tristate "Asus Virtuoso 100/200 (Xonar)" - depends on SND select SND_OXYGEN_LIB help Say Y here to include support for sound cards based on the @@ -923,7 +872,6 @@ config SND_VIRTUOSO config SND_VX222 tristate "Digigram VX222" - depends on SND select SND_VX_LIB help Say Y here to include support for Digigram VX222 soundcards. @@ -933,7 +881,6 @@ config SND_VX222 config SND_YMFPCI tristate "Yamaha YMF724/740/744/754" - depends on SND select FW_LOADER if !SND_YMFPCI_FIRMWARE_IN_KERNEL select SND_OPL3_LIB select SND_MPU401_UART @@ -954,25 +901,4 @@ config SND_YMFPCI_FIRMWARE_IN_KERNEL for the YMFPCI driver. If you choose N here, you need to install the firmware files from the alsa-firmware package. -config SND_AC97_POWER_SAVE - bool "AC97 Power-Saving Mode" - depends on SND_AC97_CODEC && EXPERIMENTAL - default n - help - Say Y here to enable the aggressive power-saving support of - AC97 codecs. In this mode, the power-mode is dynamically - controlled at each open/close. - - The mode is activated by passing power_save=1 option to - snd-ac97-codec driver. You can toggle it dynamically over - sysfs, too. - -config SND_AC97_POWER_SAVE_DEFAULT - int "Default time-out for AC97 power-save mode" - depends on SND_AC97_POWER_SAVE - default 0 - help - The default time-out value in seconds for AC97 automatic - power-save mode. 0 means to disable the power-save mode. - -endmenu +endif # SND_PCI diff --git a/sound/pci/Makefile b/sound/pci/Makefile index 85ef14bc805..65b25d221cd 100644 --- a/sound/pci/Makefile +++ b/sound/pci/Makefile @@ -13,7 +13,7 @@ snd-bt87x-objs := bt87x.o snd-cmipci-objs := cmipci.o snd-cs4281-objs := cs4281.o snd-cs5530-objs := cs5530.o -snd-ens1370-objs := ens1370.o +snd-ens1370-objs := ens1370.o ak4531_codec.o snd-ens1371-objs := ens1371.o snd-es1938-objs := es1938.o snd-es1968-objs := es1968.o diff --git a/sound/pci/ac97/Makefile b/sound/pci/ac97/Makefile index 0be48b1a22d..41fa322f097 100644 --- a/sound/pci/ac97/Makefile +++ b/sound/pci/ac97/Makefile @@ -3,16 +3,8 @@ # Copyright (c) 2001 by Jaroslav Kysela <perex@perex.cz> # -snd-ac97-codec-objs := ac97_codec.o ac97_pcm.o - -ifneq ($(CONFIG_PROC_FS),) -snd-ac97-codec-objs += ac97_proc.o -endif - -snd-ak4531-codec-objs := ak4531_codec.o +snd-ac97-codec-y := ac97_codec.o ac97_pcm.o +snd-ac97-codec-$(CONFIG_PROC_FS) += ac97_proc.o # Toplevel Module Dependency obj-$(CONFIG_SND_AC97_CODEC) += snd-ac97-codec.o -obj-$(CONFIG_SND_ENS1370) += snd-ak4531-codec.o - -obj-m := $(sort $(obj-m)) diff --git a/sound/pci/ac97/ac97_codec.c b/sound/pci/ac97/ac97_codec.c index 45fd29017dd..07364c00768 100644 --- a/sound/pci/ac97/ac97_codec.c +++ b/sound/pci/ac97/ac97_codec.c @@ -49,8 +49,9 @@ MODULE_PARM_DESC(enable_loopback, "Enable AC97 ADC/DAC Loopback Control"); #ifdef CONFIG_SND_AC97_POWER_SAVE static int power_save = CONFIG_SND_AC97_POWER_SAVE_DEFAULT; -module_param(power_save, bool, 0644); -MODULE_PARM_DESC(power_save, "Enable AC97 power-saving control"); +module_param(power_save, int, 0644); +MODULE_PARM_DESC(power_save, "Automatic power-saving timeout " + "(in second, 0 = disable)."); #endif /* @@ -2294,9 +2295,11 @@ static void snd_ac97_powerdown(struct snd_ac97 *ac97) power |= AC97_PD_PR0 | AC97_PD_PR1; /* ADC & DAC powerdown */ snd_ac97_write(ac97, AC97_POWERDOWN, power); udelay(100); - power |= AC97_PD_PR2 | AC97_PD_PR3; /* Analog Mixer powerdown */ + power |= AC97_PD_PR2; /* Analog Mixer powerdown (Vref on) */ snd_ac97_write(ac97, AC97_POWERDOWN, power); if (ac97_is_power_save_mode(ac97)) { + power |= AC97_PD_PR3; /* Analog Mixer powerdown */ + snd_ac97_write(ac97, AC97_POWERDOWN, power); udelay(100); /* AC-link powerdown, internal Clk disable */ /* FIXME: this may cause click noises on some boards */ @@ -2362,7 +2365,7 @@ int snd_ac97_update_power(struct snd_ac97 *ac97, int reg, int powerup) * that open/close frequently) */ schedule_delayed_work(&ac97->power_work, - msecs_to_jiffies(2000)); + msecs_to_jiffies(power_save * 1000)); else { cancel_delayed_work(&ac97->power_work); update_power_regs(ac97); diff --git a/sound/pci/ac97/ac97_patch.c b/sound/pci/ac97/ac97_patch.c index 1292dcee072..0746e9ccc20 100644 --- a/sound/pci/ac97/ac97_patch.c +++ b/sound/pci/ac97/ac97_patch.c @@ -669,6 +669,7 @@ AC97_SINGLE("Mic 1 Volume", AC97_MIC, 8, 31, 1), AC97_SINGLE("Mic 2 Volume", AC97_MIC, 0, 31, 1), AC97_SINGLE("Mic 20dB Boost Switch", AC97_MIC, 7, 1, 0), +AC97_SINGLE("Master Left Inv Switch", AC97_MASTER, 6, 1, 0), AC97_SINGLE("Master ZC Switch", AC97_MASTER, 7, 1, 0), AC97_SINGLE("Headphone ZC Switch", AC97_HEADPHONE, 7, 1, 0), AC97_SINGLE("Mono ZC Switch", AC97_MASTER_MONO, 7, 1, 0), @@ -3352,8 +3353,66 @@ AC97_SINGLE("Downmix LFE and Center to Front", 0x5a, 12, 1, 0), AC97_SINGLE("Downmix Surround to Front", 0x5a, 11, 1, 0), }; +static const char *slave_vols_vt1616[] = { + "Front Playback Volume", + "Surround Playback Volume", + "Center Playback Volume", + "LFE Playback Volume", + NULL +}; + +static const char *slave_sws_vt1616[] = { + "Front Playback Switch", + "Surround Playback Switch", + "Center Playback Switch", + "LFE Playback Switch", + NULL +}; + +/* find a mixer control element with the given name */ +static struct snd_kcontrol *snd_ac97_find_mixer_ctl(struct snd_ac97 *ac97, + const char *name) +{ + struct snd_ctl_elem_id id; + memset(&id, 0, sizeof(id)); + id.iface = SNDRV_CTL_ELEM_IFACE_MIXER; + strcpy(id.name, name); + return snd_ctl_find_id(ac97->bus->card, &id); +} + +/* create a virtual master control and add slaves */ +int snd_ac97_add_vmaster(struct snd_ac97 *ac97, char *name, + const unsigned int *tlv, const char **slaves) +{ + struct snd_kcontrol *kctl; + const char **s; + int err; + + kctl = snd_ctl_make_virtual_master(name, tlv); + if (!kctl) + return -ENOMEM; + err = snd_ctl_add(ac97->bus->card, kctl); + if (err < 0) + return err; + + for (s = slaves; *s; s++) { + struct snd_kcontrol *sctl; + + sctl = snd_ac97_find_mixer_ctl(ac97, *s); + if (!sctl) { + snd_printdd("Cannot find slave %s, skipped\n", *s); + continue; + } + err = snd_ctl_add_slave(kctl, sctl); + if (err < 0) + return err; + } + return 0; +} + static int patch_vt1616_specific(struct snd_ac97 * ac97) { + struct snd_kcontrol *kctl; int err; if (snd_ac97_try_bit(ac97, 0x5a, 9)) @@ -3361,6 +3420,24 @@ static int patch_vt1616_specific(struct snd_ac97 * ac97) return err; if ((err = patch_build_controls(ac97, &snd_ac97_controls_vt1616[1], ARRAY_SIZE(snd_ac97_controls_vt1616) - 1)) < 0) return err; + + /* There is already a misnamed master switch. Rename it. */ + kctl = snd_ac97_find_mixer_ctl(ac97, "Master Playback Volume"); + if (!kctl) + return -EINVAL; + + snd_ac97_rename_vol_ctl(ac97, "Master Playback", "Front Playback"); + + err = snd_ac97_add_vmaster(ac97, "Master Playback Volume", + kctl->tlv.p, slave_vols_vt1616); + if (err < 0) + return err; + + err = snd_ac97_add_vmaster(ac97, "Master Playback Switch", + NULL, slave_sws_vt1616); + if (err < 0) + return err; + return 0; } @@ -3633,7 +3710,7 @@ static int patch_ucb1400(struct snd_ac97 * ac97) { ac97->build_ops = &patch_ucb1400_ops; /* enable headphone driver and smart low power mode by default */ - snd_ac97_write(ac97, 0x6a, 0x0050); - snd_ac97_write(ac97, 0x6c, 0x0030); + snd_ac97_write_cache(ac97, 0x6a, 0x0050); + snd_ac97_write_cache(ac97, 0x6c, 0x0030); return 0; } diff --git a/sound/pci/ac97/ak4531_codec.c b/sound/pci/ak4531_codec.c index c0c1633999e..33d37b1c42f 100644 --- a/sound/pci/ac97/ak4531_codec.c +++ b/sound/pci/ak4531_codec.c @@ -28,9 +28,11 @@ #include <sound/ak4531_codec.h> #include <sound/tlv.h> +/* MODULE_AUTHOR("Jaroslav Kysela <perex@perex.cz>"); MODULE_DESCRIPTION("Universal routines for AK4531 codec"); MODULE_LICENSE("GPL"); +*/ #ifdef CONFIG_PROC_FS static void snd_ak4531_proc_init(struct snd_card *card, struct snd_ak4531 *ak4531); @@ -270,7 +272,7 @@ static const DECLARE_TLV_DB_SCALE(db_scale_master, -6200, 200, 0); static const DECLARE_TLV_DB_SCALE(db_scale_mono, -2800, 400, 0); static const DECLARE_TLV_DB_SCALE(db_scale_input, -5000, 200, 0); -static struct snd_kcontrol_new snd_ak4531_controls[] = { +static struct snd_kcontrol_new snd_ak4531_controls[] __devinitdata = { AK4531_DOUBLE_TLV("Master Playback Switch", 0, AK4531_LMASTER, AK4531_RMASTER, 7, 7, 1, 1, @@ -379,8 +381,9 @@ static u8 snd_ak4531_initial_map[0x19 + 1] = { 0x01 /* 19: Mic Amp Setup */ }; -int snd_ak4531_mixer(struct snd_card *card, struct snd_ak4531 *_ak4531, - struct snd_ak4531 **rak4531) +int __devinit snd_ak4531_mixer(struct snd_card *card, + struct snd_ak4531 *_ak4531, + struct snd_ak4531 **rak4531) { unsigned int idx; int err; @@ -476,7 +479,8 @@ static void snd_ak4531_proc_read(struct snd_info_entry *entry, ak4531->regs[AK4531_MIC_GAIN] & 1 ? "+30dB" : "+0dB"); } -static void snd_ak4531_proc_init(struct snd_card *card, struct snd_ak4531 *ak4531) +static void __devinit +snd_ak4531_proc_init(struct snd_card *card, struct snd_ak4531 *ak4531) { struct snd_info_entry *entry; @@ -484,25 +488,3 @@ static void snd_ak4531_proc_init(struct snd_card *card, struct snd_ak4531 *ak453 snd_info_set_text_ops(entry, ak4531, snd_ak4531_proc_read); } #endif - -EXPORT_SYMBOL(snd_ak4531_mixer); -#ifdef CONFIG_PM -EXPORT_SYMBOL(snd_ak4531_suspend); -EXPORT_SYMBOL(snd_ak4531_resume); -#endif - -/* - * INIT part - */ - -static int __init alsa_ak4531_init(void) -{ - return 0; -} - -static void __exit alsa_ak4531_exit(void) -{ -} - -module_init(alsa_ak4531_init) -module_exit(alsa_ak4531_exit) diff --git a/sound/pci/au88x0/au88x0_game.c b/sound/pci/au88x0/au88x0_game.c index bc212f41a38..e291aa59742 100644 --- a/sound/pci/au88x0/au88x0_game.c +++ b/sound/pci/au88x0/au88x0_game.c @@ -1,6 +1,4 @@ /* - * $Id: au88x0_game.c,v 1.9 2003/09/22 03:51:28 mjander Exp $ - * * Manuel Jander. * * Based on the work of: diff --git a/sound/pci/azt3328.c b/sound/pci/azt3328.c index 5f63af6b88a..22f18f3cfbc 100644 --- a/sound/pci/azt3328.c +++ b/sound/pci/azt3328.c @@ -1,6 +1,6 @@ /* * azt3328.c - driver for Aztech AZF3328 based soundcards (e.g. PCI168). - * Copyright (C) 2002, 2005, 2006, 2007 by Andreas Mohr <andi AT lisas.de> + * Copyright (C) 2002, 2005 - 2008 by Andreas Mohr <andi AT lisas.de> * * Framework borrowed from Bart Hartgers's als4000.c. * Driver developed on PCI168 AP(W) version (PCI rev. 10, subsystem ID 1801), @@ -35,9 +35,20 @@ * (3 weeks' worth of evenings filled with driver work). * (and no, I did NOT go the easy way: to pick up a SB PCI128 for 9 Euros) * + * It is quite likely that the AZF3328 chip is the PCI cousin of the + * AZF3318 ("azt1020 pnp", "MM Pro 16") ISA chip, given very similar specs. + * * The AZF3328 chip (note: AZF3328, *not* AZT3328, that's just the driver name - * for compatibility reasons) has the following features: + * for compatibility reasons) from Azfin (joint-venture of Aztech and Fincitec, + * Fincitec acquired by National Semiconductor in 2002, together with the + * Fincitec-related company ARSmikro) has the following features: * + * - compatibility & compliance: + * - Microsoft PC 97 ("PC 97 Hardware Design Guide", + * http://www.microsoft.com/whdc/archive/pcguides.mspx) + * - Microsoft PC 98 Baseline Audio + * - MPU401 UART + * - Sound Blaster Emulation (DOS Box) * - builtin AC97 conformant codec (SNR over 80dB) * Note that "conformant" != "compliant"!! this chip's mixer register layout * *differs* from the standard AC97 layout: @@ -48,21 +59,28 @@ * addresses illegally. So far unfortunately it looks like the very flexible * ALSA AC97 support is still not enough to easily compensate for such a * grave layout violation despite all tweaks and quirks mechanisms it offers. - * - builtin genuine OPL3 + * - builtin genuine OPL3 - verified to work fine, 20080506 * - full duplex 16bit playback/record at independent sampling rate - * - MPU401 (+ legacy address support) FIXME: how to enable legacy addr?? + * - MPU401 (+ legacy address support, claimed by one official spec sheet) + * FIXME: how to enable legacy addr?? * - game port (legacy address support) - * - builtin 3D enhancement (said to be YAMAHA Ymersion) * - builtin DirectInput support, helps reduce CPU overhead (interrupt-driven - * features supported) + * features supported). - See common term "Digital Enhanced Game Port"... + * (probably DirectInput 3.0 spec - confirm) + * - builtin 3D enhancement (said to be YAMAHA Ymersion) * - built-in General DirectX timer having a 20 bits counter * with 1us resolution (see below!) - * - I2S serial port for external DAC + * - I2S serial output port for external DAC * - supports 33MHz PCI spec 2.1, PCI power management 1.0, compliant with ACPI * - supports hardware volume control * - single chip low cost solution (128 pin QFP) * - supports programmable Sub-vendor and Sub-system ID * required for Microsoft's logo compliance (FIXME: where?) + * At least the Trident 4D Wave DX has one bit somewhere + * to enable writes to PCI subsystem VID registers, that should be it. + * This might easily be in extended PCI reg space, since PCI168 also has + * some custom data starting at 0x80. What kind of config settings + * are located in our extended PCI space anyway?? * - PCI168 AP(W) card: power amplifier with 4 Watts/channel at 4 Ohms * * Note that this driver now is actually *better* than the Windows driver, @@ -74,6 +92,24 @@ * - "timidity -iAv -B2,8 -Os -EFreverb=0" * - "pmidi -p 128:0 jazz.mid" * + * OPL3 hardware playback testing, try something like: + * cat /proc/asound/hwdep + * and + * aconnect -o + * Then use + * sbiload -Dhw:x,y --opl3 /usr/share/sounds/opl3/std.o3 ......./drums.o3 + * where x,y is the xx-yy number as given in hwdep. + * Then try + * pmidi -p a:b jazz.mid + * where a:b is the client number plus 0 usually, as given by aconnect above. + * Oh, and make sure to unmute the FM mixer control (doh!) + * NOTE: power use during OPL3 playback is _VERY_ high (70W --> 90W!) + * despite no CPU activity, possibly due to hindering ACPI idling somehow. + * Shouldn't be a problem of the AZF3328 chip itself, I'd hope. + * Higher PCM / FM mixer levels seem to conflict (causes crackling), + * at least sometimes. Maybe even use with hardware sequencer timer above :) + * adplay/adplug-utils might soon offer hardware-based OPL3 playback, too. + * * Certain PCI versions of this card are susceptible to DMA traffic underruns * in some systems (resulting in sound crackling/clicking/popping), * probably because they don't have a DMA FIFO buffer or so. @@ -87,6 +123,8 @@ * better than a VIA, yet ironically I still get crackling, like many other * people with the same chipset. * Possible remedies: + * - use speaker (amplifier) output instead of headphone output + * (in case crackling is due to overloaded output clipping) * - plug card into a different PCI slot, preferrably one that isn't shared * too much (this helps a lot, but not completely!) * - get rid of PCI VGA card, use AGP instead @@ -94,18 +132,23 @@ * - fiddle with PCI latency settings (setpci -v -s BUSID latency_timer=XX) * Not too helpful. * - Disable ACPI/power management/"Auto Detect RAM/PCI Clk" in BIOS - * + * * BUGS - * - full-duplex might *still* be problematic, not fully tested recently + * - full-duplex might *still* be problematic, however a recent test was fine * - (non-bug) "Bass/Treble or 3D settings don't work" - they do get evaluated * if you set PCM output switch to "pre 3D" instead of "post 3D". * If this can't be set, then get a mixer application that Isn't Stupid (tm) * (e.g. kmix, gamix) - unfortunately several are!! - * + * - locking is not entirely clean, especially the audio stream activity + * ints --> may be racy + * - an _unconnected_ secondary joystick at the gameport will be reported + * to be "active" (floating values, not precisely -1) due to the way we need + * to read the Digital Enhanced Game Port. Not sure whether it is fixable. + * * TODO * - test MPU401 MIDI playback etc. - * - add some power micro-management (disable various units of the card - * as long as they're unused). However this requires I/O ports which I + * - add more power micro-management (disable various units of the card + * as long as they're unused). However this requires more I/O ports which I * haven't figured out yet and which thus might not even exist... * The standard suspend/resume functionality could probably make use of * some improvement, too... @@ -113,6 +156,7 @@ * - figure out some cleverly evil scheme to possibly make ALSA AC97 code * fully accept our quite incompatible ""AC97"" mixer and thus save some * code (but I'm not too optimistic that doing this is possible at all) + * - use MMIO (memory-mapped I/O)? Slightly faster access, e.g. for gameport. */ #include <asm/io.h> @@ -138,7 +182,7 @@ MODULE_LICENSE("GPL"); MODULE_SUPPORTED_DEVICE("{{Aztech,AZF3328}}"); #if defined(CONFIG_GAMEPORT) || (defined(MODULE) && defined(CONFIG_GAMEPORT_MODULE)) -#define SUPPORT_JOYSTICK 1 +#define SUPPORT_GAMEPORT 1 #endif #define DEBUG_MISC 0 @@ -147,13 +191,14 @@ MODULE_SUPPORTED_DEVICE("{{Aztech,AZF3328}}"); #define DEBUG_PLAY_REC 0 #define DEBUG_IO 0 #define DEBUG_TIMER 0 +#define DEBUG_GAME 0 #define MIXER_TESTING 0 #if DEBUG_MISC #define snd_azf3328_dbgmisc(format, args...) printk(KERN_ERR format, ##args) #else #define snd_azf3328_dbgmisc(format, args...) -#endif +#endif #if DEBUG_CALLS #define snd_azf3328_dbgcalls(format, args...) printk(format, ##args) @@ -163,25 +208,31 @@ MODULE_SUPPORTED_DEVICE("{{Aztech,AZF3328}}"); #define snd_azf3328_dbgcalls(format, args...) #define snd_azf3328_dbgcallenter() #define snd_azf3328_dbgcallleave() -#endif +#endif #if DEBUG_MIXER #define snd_azf3328_dbgmixer(format, args...) printk(format, ##args) #else #define snd_azf3328_dbgmixer(format, args...) -#endif +#endif #if DEBUG_PLAY_REC #define snd_azf3328_dbgplay(format, args...) printk(KERN_ERR format, ##args) #else #define snd_azf3328_dbgplay(format, args...) -#endif +#endif #if DEBUG_MISC #define snd_azf3328_dbgtimer(format, args...) printk(KERN_ERR format, ##args) #else #define snd_azf3328_dbgtimer(format, args...) -#endif +#endif + +#if DEBUG_GAME +#define snd_azf3328_dbggame(format, args...) printk(KERN_ERR format, ##args) +#else +#define snd_azf3328_dbggame(format, args...) +#endif static int index[SNDRV_CARDS] = SNDRV_DEFAULT_IDX; /* Index 0-MAX */ module_param_array(index, int, NULL, 0444); @@ -195,51 +246,62 @@ static int enable[SNDRV_CARDS] = SNDRV_DEFAULT_ENABLE_PNP; /* Enable this card * module_param_array(enable, bool, NULL, 0444); MODULE_PARM_DESC(enable, "Enable AZF3328 soundcard."); -#ifdef SUPPORT_JOYSTICK -static int joystick[SNDRV_CARDS]; -module_param_array(joystick, bool, NULL, 0444); -MODULE_PARM_DESC(joystick, "Enable joystick for AZF3328 soundcard."); -#endif - static int seqtimer_scaling = 128; module_param(seqtimer_scaling, int, 0444); MODULE_PARM_DESC(seqtimer_scaling, "Set 1024000Hz sequencer timer scale factor (lockup danger!). Default 128."); +struct snd_azf3328_audio_stream { + struct snd_pcm_substream *substream; + int enabled; + int running; + unsigned long portbase; +}; + +enum snd_azf3328_stream_index { + AZF_PLAYBACK = 0, + AZF_CAPTURE = 1, +}; + struct snd_azf3328 { /* often-used fields towards beginning, then grouped */ - unsigned long codec_port; - unsigned long io2_port; - unsigned long mpu_port; - unsigned long synth_port; - unsigned long mixer_port; + + unsigned long codec_io; /* usually 0xb000, size 128 */ + unsigned long game_io; /* usually 0xb400, size 8 */ + unsigned long mpu_io; /* usually 0xb800, size 4 */ + unsigned long opl3_io; /* usually 0xbc00, size 8 */ + unsigned long mixer_io; /* usually 0xc000, size 64 */ spinlock_t reg_lock; struct snd_timer *timer; - + struct snd_pcm *pcm; - struct snd_pcm_substream *playback_substream; - struct snd_pcm_substream *capture_substream; - unsigned int is_playing; - unsigned int is_recording; + struct snd_azf3328_audio_stream audio_stream[2]; struct snd_card *card; struct snd_rawmidi *rmidi; -#ifdef SUPPORT_JOYSTICK +#ifdef SUPPORT_GAMEPORT struct gameport *gameport; + int axes[4]; #endif struct pci_dev *pci; int irq; + /* register 0x6a is write-only, thus need to remember setting. + * If we need to add more registers here, then we might try to fold this + * into some transparent combined shadow register handling with + * CONFIG_PM register storage below, but that's slightly difficult. */ + u16 shadow_reg_codec_6AH; + #ifdef CONFIG_PM /* register value containers for power management * Note: not always full I/O range preserved (just like Win driver!) */ - u16 saved_regs_codec [AZF_IO_SIZE_CODEC_PM / 2]; - u16 saved_regs_io2 [AZF_IO_SIZE_IO2_PM / 2]; - u16 saved_regs_mpu [AZF_IO_SIZE_MPU_PM / 2]; - u16 saved_regs_synth[AZF_IO_SIZE_SYNTH_PM / 2]; + u16 saved_regs_codec[AZF_IO_SIZE_CODEC_PM / 2]; + u16 saved_regs_game [AZF_IO_SIZE_GAME_PM / 2]; + u16 saved_regs_mpu [AZF_IO_SIZE_MPU_PM / 2]; + u16 saved_regs_opl3 [AZF_IO_SIZE_OPL3_PM / 2]; u16 saved_regs_mixer[AZF_IO_SIZE_MIXER_PM / 2]; #endif }; @@ -252,126 +314,166 @@ static const struct pci_device_id snd_azf3328_ids[] = { MODULE_DEVICE_TABLE(pci, snd_azf3328_ids); + +static int +snd_azf3328_io_reg_setb(unsigned reg, u8 mask, int do_set) +{ + u8 prev = inb(reg), new; + + new = (do_set) ? (prev|mask) : (prev & ~mask); + /* we need to always write the new value no matter whether it differs + * or not, since some register bits don't indicate their setting */ + outb(new, reg); + if (new != prev) + return 1; + + return 0; +} + static inline void -snd_azf3328_codec_outb(const struct snd_azf3328 *chip, int reg, u8 value) +snd_azf3328_codec_outb(const struct snd_azf3328 *chip, unsigned reg, u8 value) { - outb(value, chip->codec_port + reg); + outb(value, chip->codec_io + reg); } static inline u8 -snd_azf3328_codec_inb(const struct snd_azf3328 *chip, int reg) +snd_azf3328_codec_inb(const struct snd_azf3328 *chip, unsigned reg) { - return inb(chip->codec_port + reg); + return inb(chip->codec_io + reg); } static inline void -snd_azf3328_codec_outw(const struct snd_azf3328 *chip, int reg, u16 value) +snd_azf3328_codec_outw(const struct snd_azf3328 *chip, unsigned reg, u16 value) { - outw(value, chip->codec_port + reg); + outw(value, chip->codec_io + reg); } static inline u16 -snd_azf3328_codec_inw(const struct snd_azf3328 *chip, int reg) +snd_azf3328_codec_inw(const struct snd_azf3328 *chip, unsigned reg) +{ + return inw(chip->codec_io + reg); +} + +static inline void +snd_azf3328_codec_outl(const struct snd_azf3328 *chip, unsigned reg, u32 value) +{ + outl(value, chip->codec_io + reg); +} + +static inline u32 +snd_azf3328_codec_inl(const struct snd_azf3328 *chip, unsigned reg) { - return inw(chip->codec_port + reg); + return inl(chip->codec_io + reg); } static inline void -snd_azf3328_codec_outl(const struct snd_azf3328 *chip, int reg, u32 value) +snd_azf3328_game_outb(const struct snd_azf3328 *chip, unsigned reg, u8 value) { - outl(value, chip->codec_port + reg); + outb(value, chip->game_io + reg); } static inline void -snd_azf3328_io2_outb(const struct snd_azf3328 *chip, int reg, u8 value) +snd_azf3328_game_outw(const struct snd_azf3328 *chip, unsigned reg, u16 value) { - outb(value, chip->io2_port + reg); + outw(value, chip->game_io + reg); } static inline u8 -snd_azf3328_io2_inb(const struct snd_azf3328 *chip, int reg) +snd_azf3328_game_inb(const struct snd_azf3328 *chip, unsigned reg) { - return inb(chip->io2_port + reg); + return inb(chip->game_io + reg); +} + +static inline u16 +snd_azf3328_game_inw(const struct snd_azf3328 *chip, unsigned reg) +{ + return inw(chip->game_io + reg); } static inline void -snd_azf3328_mixer_outw(const struct snd_azf3328 *chip, int reg, u16 value) +snd_azf3328_mixer_outw(const struct snd_azf3328 *chip, unsigned reg, u16 value) { - outw(value, chip->mixer_port + reg); + outw(value, chip->mixer_io + reg); } static inline u16 -snd_azf3328_mixer_inw(const struct snd_azf3328 *chip, int reg) +snd_azf3328_mixer_inw(const struct snd_azf3328 *chip, unsigned reg) { - return inw(chip->mixer_port + reg); + return inw(chip->mixer_io + reg); } -static void -snd_azf3328_mixer_set_mute(const struct snd_azf3328 *chip, int reg, int do_mute) +#define AZF_MUTE_BIT 0x80 + +static int +snd_azf3328_mixer_set_mute(const struct snd_azf3328 *chip, + unsigned reg, int do_mute +) { - unsigned long portbase = chip->mixer_port + reg + 1; - unsigned char oldval; + unsigned long portbase = chip->mixer_io + reg + 1; + int updated; /* the mute bit is on the *second* (i.e. right) register of a * left/right channel setting */ - oldval = inb(portbase); - if (do_mute) - oldval |= 0x80; - else - oldval &= ~0x80; - outb(oldval, portbase); + updated = snd_azf3328_io_reg_setb(portbase, AZF_MUTE_BIT, do_mute); + + /* indicate whether it was muted before */ + return (do_mute) ? !updated : updated; } static void -snd_azf3328_mixer_write_volume_gradually(const struct snd_azf3328 *chip, int reg, unsigned char dst_vol_left, unsigned char dst_vol_right, int chan_sel, int delay) +snd_azf3328_mixer_write_volume_gradually(const struct snd_azf3328 *chip, + unsigned reg, + unsigned char dst_vol_left, + unsigned char dst_vol_right, + int chan_sel, int delay +) { - unsigned long portbase = chip->mixer_port + reg; + unsigned long portbase = chip->mixer_io + reg; unsigned char curr_vol_left = 0, curr_vol_right = 0; - int left_done = 0, right_done = 0; - + int left_change = 0, right_change = 0; + snd_azf3328_dbgcallenter(); - if (chan_sel & SET_CHAN_LEFT) + + if (chan_sel & SET_CHAN_LEFT) { curr_vol_left = inb(portbase + 1); - else - left_done = 1; - if (chan_sel & SET_CHAN_RIGHT) + + /* take care of muting flag contained in left channel */ + if (curr_vol_left & AZF_MUTE_BIT) + dst_vol_left |= AZF_MUTE_BIT; + else + dst_vol_left &= ~AZF_MUTE_BIT; + + left_change = (curr_vol_left > dst_vol_left) ? -1 : 1; + } + + if (chan_sel & SET_CHAN_RIGHT) { curr_vol_right = inb(portbase + 0); - else - right_done = 1; - - /* take care of muting flag (0x80) contained in left channel */ - if (curr_vol_left & 0x80) - dst_vol_left |= 0x80; - else - dst_vol_left &= ~0x80; + + right_change = (curr_vol_right > dst_vol_right) ? -1 : 1; + } do { - if (!left_done) { - if (curr_vol_left > dst_vol_left) - curr_vol_left--; - else - if (curr_vol_left < dst_vol_left) - curr_vol_left++; - else - left_done = 1; - outb(curr_vol_left, portbase + 1); + if (left_change) { + if (curr_vol_left != dst_vol_left) { + curr_vol_left += left_change; + outb(curr_vol_left, portbase + 1); + } else + left_change = 0; } - if (!right_done) { - if (curr_vol_right > dst_vol_right) - curr_vol_right--; - else - if (curr_vol_right < dst_vol_right) - curr_vol_right++; - else - right_done = 1; + if (right_change) { + if (curr_vol_right != dst_vol_right) { + curr_vol_right += right_change; + /* during volume change, the right channel is crackling * somewhat more than the left channel, unfortunately. * This seems to be a hardware issue. */ - outb(curr_vol_right, portbase + 0); + outb(curr_vol_right, portbase + 0); + } else + right_change = 0; } if (delay) mdelay(delay); - } while ((!left_done) || (!right_done)); + } while ((left_change) || (right_change)); snd_azf3328_dbgcallleave(); } @@ -379,7 +481,7 @@ snd_azf3328_mixer_write_volume_gradually(const struct snd_azf3328 *chip, int reg * general mixer element */ struct azf3328_mixer_reg { - unsigned int reg; + unsigned reg; unsigned int lchan_shift, rchan_shift; unsigned int mask; unsigned int invert: 1; @@ -544,13 +646,14 @@ snd_azf3328_info_mixer_enum(struct snd_kcontrol *kcontrol, "Mix", "Mic" }; static const char * const texts3[] = { - "Mic", "CD", "Video", "Aux", + "Mic", "CD", "Video", "Aux", "Line", "Mix", "Mix Mono", "Phone" }; static const char * const texts4[] = { "pre 3D", "post 3D" }; struct azf3328_mixer_reg reg; + const char * const *p = NULL; snd_azf3328_mixer_reg_decode(®, kcontrol->private_value); uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED; @@ -561,18 +664,20 @@ snd_azf3328_info_mixer_enum(struct snd_kcontrol *kcontrol, if (reg.reg == IDX_MIXER_ADVCTL2) { switch(reg.lchan_shift) { case 8: /* modem out sel */ - strcpy(uinfo->value.enumerated.name, texts1[uinfo->value.enumerated.item]); + p = texts1; break; case 9: /* mono sel source */ - strcpy(uinfo->value.enumerated.name, texts2[uinfo->value.enumerated.item]); + p = texts2; break; case 15: /* PCM Out Path */ - strcpy(uinfo->value.enumerated.name, texts4[uinfo->value.enumerated.item]); + p = texts4; break; } } else - strcpy(uinfo->value.enumerated.name, texts3[uinfo->value.enumerated.item] -); + if (reg.reg == IDX_MIXER_REC_SELECT) + p = texts3; + + strcpy(uinfo->value.enumerated.name, p[uinfo->value.enumerated.item]); return 0; } @@ -583,7 +688,7 @@ snd_azf3328_get_mixer_enum(struct snd_kcontrol *kcontrol, struct snd_azf3328 *chip = snd_kcontrol_chip(kcontrol); struct azf3328_mixer_reg reg; unsigned short val; - + snd_azf3328_mixer_reg_decode(®, kcontrol->private_value); val = snd_azf3328_mixer_inw(chip, reg.reg); if (reg.reg == IDX_MIXER_REC_SELECT) { @@ -605,7 +710,7 @@ snd_azf3328_put_mixer_enum(struct snd_kcontrol *kcontrol, struct snd_azf3328 *chip = snd_kcontrol_chip(kcontrol); struct azf3328_mixer_reg reg; unsigned int oreg, nreg, val; - + snd_azf3328_mixer_reg_decode(®, kcontrol->private_value); oreg = snd_azf3328_mixer_inw(chip, reg.reg); val = oreg; @@ -631,9 +736,11 @@ snd_azf3328_put_mixer_enum(struct snd_kcontrol *kcontrol, static struct snd_kcontrol_new snd_azf3328_mixer_controls[] __devinitdata = { AZF3328_MIXER_SWITCH("Master Playback Switch", IDX_MIXER_PLAY_MASTER, 15, 1), AZF3328_MIXER_VOL_STEREO("Master Playback Volume", IDX_MIXER_PLAY_MASTER, 0x1f, 1), - AZF3328_MIXER_SWITCH("Wave Playback Switch", IDX_MIXER_WAVEOUT, 15, 1), - AZF3328_MIXER_VOL_STEREO("Wave Playback Volume", IDX_MIXER_WAVEOUT, 0x1f, 1), - AZF3328_MIXER_SWITCH("Wave 3D Bypass Playback Switch", IDX_MIXER_ADVCTL2, 7, 1), + AZF3328_MIXER_SWITCH("PCM Playback Switch", IDX_MIXER_WAVEOUT, 15, 1), + AZF3328_MIXER_VOL_STEREO("PCM Playback Volume", + IDX_MIXER_WAVEOUT, 0x1f, 1), + AZF3328_MIXER_SWITCH("PCM 3D Bypass Playback Switch", + IDX_MIXER_ADVCTL2, 7, 1), AZF3328_MIXER_SWITCH("FM Playback Switch", IDX_MIXER_FMSYNTH, 15, 1), AZF3328_MIXER_VOL_STEREO("FM Playback Volume", IDX_MIXER_FMSYNTH, 0x1f, 1), AZF3328_MIXER_SWITCH("CD Playback Switch", IDX_MIXER_CDAUDIO, 15, 1), @@ -717,15 +824,16 @@ snd_azf3328_mixer_new(struct snd_azf3328 *chip) snd_azf3328_mixer_outw(chip, IDX_MIXER_RESET, 0x0000); /* mute and zero volume channels */ - for (idx = 0; idx < ARRAY_SIZE(snd_azf3328_init_values); idx++) { + for (idx = 0; idx < ARRAY_SIZE(snd_azf3328_init_values); ++idx) { snd_azf3328_mixer_outw(chip, snd_azf3328_init_values[idx][0], snd_azf3328_init_values[idx][1]); } - + /* add mixer controls */ sw = snd_azf3328_mixer_controls; - for (idx = 0; idx < ARRAY_SIZE(snd_azf3328_mixer_controls); idx++, sw++) { + for (idx = 0; idx < ARRAY_SIZE(snd_azf3328_mixer_controls); + ++idx, ++sw) { if ((err = snd_ctl_add(chip->card, snd_ctl_new1(sw, chip))) < 0) return err; } @@ -757,9 +865,9 @@ snd_azf3328_hw_free(struct snd_pcm_substream *substream) } static void -snd_azf3328_setfmt(struct snd_azf3328 *chip, - unsigned int reg, - unsigned int bitrate, +snd_azf3328_codec_setfmt(struct snd_azf3328 *chip, + unsigned reg, + enum azf_freq_t bitrate, unsigned int format_width, unsigned int channels ) @@ -769,24 +877,25 @@ snd_azf3328_setfmt(struct snd_azf3328 *chip, snd_azf3328_dbgcallenter(); switch (bitrate) { - case 4000: val |= SOUNDFORMAT_FREQ_SUSPECTED_4000; break; - case 4800: val |= SOUNDFORMAT_FREQ_SUSPECTED_4800; break; - case 5512: val |= SOUNDFORMAT_FREQ_5510; break; /* the AZF3328 names it "5510" for some strange reason */ - case 6620: val |= SOUNDFORMAT_FREQ_6620; break; - case 8000: val |= SOUNDFORMAT_FREQ_8000; break; - case 9600: val |= SOUNDFORMAT_FREQ_9600; break; - case 11025: val |= SOUNDFORMAT_FREQ_11025; break; - case 13240: val |= SOUNDFORMAT_FREQ_SUSPECTED_13240; break; - case 16000: val |= SOUNDFORMAT_FREQ_16000; break; - case 22050: val |= SOUNDFORMAT_FREQ_22050; break; - case 32000: val |= SOUNDFORMAT_FREQ_32000; break; - case 44100: val |= SOUNDFORMAT_FREQ_44100; break; - case 48000: val |= SOUNDFORMAT_FREQ_48000; break; - case 66200: val |= SOUNDFORMAT_FREQ_SUSPECTED_66200; break; + case AZF_FREQ_4000: val |= SOUNDFORMAT_FREQ_SUSPECTED_4000; break; + case AZF_FREQ_4800: val |= SOUNDFORMAT_FREQ_SUSPECTED_4800; break; + case AZF_FREQ_5512: + /* the AZF3328 names it "5510" for some strange reason */ + val |= SOUNDFORMAT_FREQ_5510; break; + case AZF_FREQ_6620: val |= SOUNDFORMAT_FREQ_6620; break; + case AZF_FREQ_8000: val |= SOUNDFORMAT_FREQ_8000; break; + case AZF_FREQ_9600: val |= SOUNDFORMAT_FREQ_9600; break; + case AZF_FREQ_11025: val |= SOUNDFORMAT_FREQ_11025; break; + case AZF_FREQ_13240: val |= SOUNDFORMAT_FREQ_SUSPECTED_13240; break; + case AZF_FREQ_16000: val |= SOUNDFORMAT_FREQ_16000; break; + case AZF_FREQ_22050: val |= SOUNDFORMAT_FREQ_22050; break; + case AZF_FREQ_32000: val |= SOUNDFORMAT_FREQ_32000; break; default: snd_printk(KERN_WARNING "unknown bitrate %d, assuming 44.1kHz!\n", bitrate); - val |= SOUNDFORMAT_FREQ_44100; - break; + /* fall-through */ + case AZF_FREQ_44100: val |= SOUNDFORMAT_FREQ_44100; break; + case AZF_FREQ_48000: val |= SOUNDFORMAT_FREQ_48000; break; + case AZF_FREQ_66200: val |= SOUNDFORMAT_FREQ_SUSPECTED_66200; break; } /* val = 0xff07; 3m27.993s (65301Hz; -> 64000Hz???) hmm, 66120, 65967, 66123 */ /* val = 0xff09; 17m15.098s (13123,478Hz; -> 12000Hz???) hmm, 13237.2Hz? */ @@ -805,10 +914,10 @@ snd_azf3328_setfmt(struct snd_azf3328 *chip, val |= SOUNDFORMAT_FLAG_16BIT; spin_lock_irqsave(&chip->reg_lock, flags); - + /* set bitrate/format */ snd_azf3328_codec_outw(chip, reg, val); - + /* changing the bitrate/format settings switches off the * audio output with an annoying click in case of 8/16bit format change * (maybe shutting down DAC/ADC?), thus immediately @@ -830,31 +939,95 @@ snd_azf3328_setfmt(struct snd_azf3328 *chip, snd_azf3328_dbgcallleave(); } +static inline void +snd_azf3328_codec_setfmt_lowpower(struct snd_azf3328 *chip, + unsigned reg +) +{ + /* choose lowest frequency for low power consumption. + * While this will cause louder noise due to rather coarse frequency, + * it should never matter since output should always + * get disabled properly when idle anyway. */ + snd_azf3328_codec_setfmt(chip, reg, AZF_FREQ_4000, 8, 1); +} + +static void +snd_azf3328_codec_reg_6AH_update(struct snd_azf3328 *chip, + unsigned bitmask, + int enable +) +{ + if (enable) + chip->shadow_reg_codec_6AH &= ~bitmask; + else + chip->shadow_reg_codec_6AH |= bitmask; + snd_azf3328_dbgplay("6AH_update mask 0x%04x enable %d: val 0x%04x\n", + bitmask, enable, chip->shadow_reg_codec_6AH); + snd_azf3328_codec_outw(chip, IDX_IO_6AH, chip->shadow_reg_codec_6AH); +} + +static inline void +snd_azf3328_codec_enable(struct snd_azf3328 *chip, int enable) +{ + snd_azf3328_dbgplay("codec_enable %d\n", enable); + /* no idea what exactly is being done here, but I strongly assume it's + * PM related */ + snd_azf3328_codec_reg_6AH_update( + chip, IO_6A_PAUSE_PLAYBACK_BIT8, enable + ); +} + +static void +snd_azf3328_codec_activity(struct snd_azf3328 *chip, + enum snd_azf3328_stream_index stream_type, + int enable +) +{ + int need_change = (chip->audio_stream[stream_type].running != enable); + + snd_azf3328_dbgplay( + "codec_activity: type %d, enable %d, need_change %d\n", + stream_type, enable, need_change + ); + if (need_change) { + enum snd_azf3328_stream_index other = + (stream_type == AZF_PLAYBACK) ? + AZF_CAPTURE : AZF_PLAYBACK; + /* small check to prevent shutting down the other party + * in case it's active */ + if ((enable) || !(chip->audio_stream[other].running)) + snd_azf3328_codec_enable(chip, enable); + + /* ...and adjust clock, too + * (reduce noise and power consumption) */ + if (!enable) + snd_azf3328_codec_setfmt_lowpower( + chip, + chip->audio_stream[stream_type].portbase + + IDX_IO_PLAY_SOUNDFORMAT + ); + } + chip->audio_stream[stream_type].running = enable; +} + static void snd_azf3328_setdmaa(struct snd_azf3328 *chip, long unsigned int addr, unsigned int count, unsigned int size, - int do_recording) + enum snd_azf3328_stream_index stream_type +) { - unsigned long flags, portbase; - unsigned int is_running; - snd_azf3328_dbgcallenter(); - if (do_recording) { - /* access capture registers, i.e. skip playback reg section */ - portbase = chip->codec_port + 0x20; - is_running = chip->is_recording; - } else { - /* access the playback register section */ - portbase = chip->codec_port + 0x00; - is_running = chip->is_playing; - } + if (!chip->audio_stream[stream_type].running) { + /* AZF3328 uses a two buffer pointer DMA playback approach */ - /* AZF3328 uses a two buffer pointer DMA playback approach */ - if (!is_running) { - unsigned long addr_area2; - unsigned long count_areas, count_tmp; /* width 32bit -- overflow!! */ + unsigned long flags, portbase, addr_area2; + + /* width 32bit (prevent overflow): */ + unsigned long count_areas, count_tmp; + + portbase = chip->audio_stream[stream_type].portbase; count_areas = size/2; addr_area2 = addr+count_areas; count_areas--; /* max. index */ @@ -884,11 +1057,11 @@ snd_azf3328_playback_prepare(struct snd_pcm_substream *substream) snd_azf3328_dbgcallenter(); #if 0 - snd_azf3328_setfmt(chip, IDX_IO_PLAY_SOUNDFORMAT, + snd_azf3328_codec_setfmt(chip, IDX_IO_PLAY_SOUNDFORMAT, runtime->rate, snd_pcm_format_width(runtime->format), runtime->channels); - snd_azf3328_setdmaa(chip, runtime->dma_addr, count, size, 0); + snd_azf3328_setdmaa(chip, runtime->dma_addr, count, size, AZF_PLAYBACK); #endif snd_azf3328_dbgcallleave(); return 0; @@ -906,11 +1079,11 @@ snd_azf3328_capture_prepare(struct snd_pcm_substream *substream) snd_azf3328_dbgcallenter(); #if 0 - snd_azf3328_setfmt(chip, IDX_IO_REC_SOUNDFORMAT, + snd_azf3328_codec_setfmt(chip, IDX_IO_REC_SOUNDFORMAT, runtime->rate, snd_pcm_format_width(runtime->format), runtime->channels); - snd_azf3328_setdmaa(chip, runtime->dma_addr, count, size, 1); + snd_azf3328_setdmaa(chip, runtime->dma_addr, count, size, AZF_CAPTURE); #endif snd_azf3328_dbgcallleave(); return 0; @@ -923,6 +1096,7 @@ snd_azf3328_playback_trigger(struct snd_pcm_substream *substream, int cmd) struct snd_pcm_runtime *runtime = substream->runtime; int result = 0; unsigned int status1; + int previously_muted; snd_azf3328_dbgcalls("snd_azf3328_playback_trigger cmd %d\n", cmd); @@ -930,20 +1104,23 @@ snd_azf3328_playback_trigger(struct snd_pcm_substream *substream, int cmd) case SNDRV_PCM_TRIGGER_START: snd_azf3328_dbgplay("START PLAYBACK\n"); - /* mute WaveOut */ - snd_azf3328_mixer_set_mute(chip, IDX_MIXER_WAVEOUT, 1); + /* mute WaveOut (avoid clicking during setup) */ + previously_muted = + snd_azf3328_mixer_set_mute(chip, IDX_MIXER_WAVEOUT, 1); - snd_azf3328_setfmt(chip, IDX_IO_PLAY_SOUNDFORMAT, + snd_azf3328_codec_setfmt(chip, IDX_IO_PLAY_SOUNDFORMAT, runtime->rate, snd_pcm_format_width(runtime->format), runtime->channels); spin_lock(&chip->reg_lock); - /* stop playback */ + /* first, remember current value: */ status1 = snd_azf3328_codec_inw(chip, IDX_IO_PLAY_FLAGS); + + /* stop playback */ status1 &= ~DMA_RESUME; snd_azf3328_codec_outw(chip, IDX_IO_PLAY_FLAGS, status1); - + /* FIXME: clear interrupts or what??? */ snd_azf3328_codec_outw(chip, IDX_IO_PLAY_IRQTYPE, 0xffff); spin_unlock(&chip->reg_lock); @@ -951,7 +1128,7 @@ snd_azf3328_playback_trigger(struct snd_pcm_substream *substream, int cmd) snd_azf3328_setdmaa(chip, runtime->dma_addr, snd_pcm_lib_period_bytes(substream), snd_pcm_lib_buffer_bytes(substream), - 0); + AZF_PLAYBACK); spin_lock(&chip->reg_lock); #ifdef WIN9X @@ -978,30 +1155,35 @@ snd_azf3328_playback_trigger(struct snd_pcm_substream *substream, int cmd) DMA_SOMETHING_ELSE); #endif spin_unlock(&chip->reg_lock); + snd_azf3328_codec_activity(chip, AZF_PLAYBACK, 1); /* now unmute WaveOut */ - snd_azf3328_mixer_set_mute(chip, IDX_MIXER_WAVEOUT, 0); + if (!previously_muted) + snd_azf3328_mixer_set_mute(chip, IDX_MIXER_WAVEOUT, 0); - chip->is_playing = 1; snd_azf3328_dbgplay("STARTED PLAYBACK\n"); break; case SNDRV_PCM_TRIGGER_RESUME: snd_azf3328_dbgplay("RESUME PLAYBACK\n"); /* resume playback if we were active */ - if (chip->is_playing) + spin_lock(&chip->reg_lock); + if (chip->audio_stream[AZF_PLAYBACK].running) snd_azf3328_codec_outw(chip, IDX_IO_PLAY_FLAGS, snd_azf3328_codec_inw(chip, IDX_IO_PLAY_FLAGS) | DMA_RESUME); + spin_unlock(&chip->reg_lock); break; case SNDRV_PCM_TRIGGER_STOP: snd_azf3328_dbgplay("STOP PLAYBACK\n"); - /* mute WaveOut */ - snd_azf3328_mixer_set_mute(chip, IDX_MIXER_WAVEOUT, 1); + /* mute WaveOut (avoid clicking during setup) */ + previously_muted = + snd_azf3328_mixer_set_mute(chip, IDX_MIXER_WAVEOUT, 1); spin_lock(&chip->reg_lock); - /* stop playback */ + /* first, remember current value: */ status1 = snd_azf3328_codec_inw(chip, IDX_IO_PLAY_FLAGS); + /* stop playback */ status1 &= ~DMA_RESUME; snd_azf3328_codec_outw(chip, IDX_IO_PLAY_FLAGS, status1); @@ -1013,10 +1195,12 @@ snd_azf3328_playback_trigger(struct snd_pcm_substream *substream, int cmd) status1 &= ~DMA_PLAY_SOMETHING1; snd_azf3328_codec_outw(chip, IDX_IO_PLAY_FLAGS, status1); spin_unlock(&chip->reg_lock); - + snd_azf3328_codec_activity(chip, AZF_PLAYBACK, 0); + /* now unmute WaveOut */ - snd_azf3328_mixer_set_mute(chip, IDX_MIXER_WAVEOUT, 0); - chip->is_playing = 0; + if (!previously_muted) + snd_azf3328_mixer_set_mute(chip, IDX_MIXER_WAVEOUT, 0); + snd_azf3328_dbgplay("STOPPED PLAYBACK\n"); break; case SNDRV_PCM_TRIGGER_SUSPEND: @@ -1035,7 +1219,7 @@ snd_azf3328_playback_trigger(struct snd_pcm_substream *substream, int cmd) printk(KERN_ERR "FIXME: unknown trigger mode!\n"); return -EINVAL; } - + snd_azf3328_dbgcallleave(); return result; } @@ -1057,17 +1241,19 @@ snd_azf3328_capture_trigger(struct snd_pcm_substream *substream, int cmd) snd_azf3328_dbgplay("START CAPTURE\n"); - snd_azf3328_setfmt(chip, IDX_IO_REC_SOUNDFORMAT, + snd_azf3328_codec_setfmt(chip, IDX_IO_REC_SOUNDFORMAT, runtime->rate, snd_pcm_format_width(runtime->format), runtime->channels); spin_lock(&chip->reg_lock); - /* stop recording */ + /* first, remember current value: */ status1 = snd_azf3328_codec_inw(chip, IDX_IO_REC_FLAGS); + + /* stop recording */ status1 &= ~DMA_RESUME; snd_azf3328_codec_outw(chip, IDX_IO_REC_FLAGS, status1); - + /* FIXME: clear interrupts or what??? */ snd_azf3328_codec_outw(chip, IDX_IO_REC_IRQTYPE, 0xffff); spin_unlock(&chip->reg_lock); @@ -1075,7 +1261,7 @@ snd_azf3328_capture_trigger(struct snd_pcm_substream *substream, int cmd) snd_azf3328_setdmaa(chip, runtime->dma_addr, snd_pcm_lib_period_bytes(substream), snd_pcm_lib_buffer_bytes(substream), - 1); + AZF_CAPTURE); spin_lock(&chip->reg_lock); #ifdef WIN9X @@ -1102,24 +1288,27 @@ snd_azf3328_capture_trigger(struct snd_pcm_substream *substream, int cmd) DMA_SOMETHING_ELSE); #endif spin_unlock(&chip->reg_lock); + snd_azf3328_codec_activity(chip, AZF_CAPTURE, 1); - chip->is_recording = 1; snd_azf3328_dbgplay("STARTED CAPTURE\n"); break; case SNDRV_PCM_TRIGGER_RESUME: snd_azf3328_dbgplay("RESUME CAPTURE\n"); /* resume recording if we were active */ - if (chip->is_recording) + spin_lock(&chip->reg_lock); + if (chip->audio_stream[AZF_CAPTURE].running) snd_azf3328_codec_outw(chip, IDX_IO_REC_FLAGS, snd_azf3328_codec_inw(chip, IDX_IO_REC_FLAGS) | DMA_RESUME); + spin_unlock(&chip->reg_lock); break; case SNDRV_PCM_TRIGGER_STOP: snd_azf3328_dbgplay("STOP CAPTURE\n"); spin_lock(&chip->reg_lock); - /* stop recording */ + /* first, remember current value: */ status1 = snd_azf3328_codec_inw(chip, IDX_IO_REC_FLAGS); + /* stop recording */ status1 &= ~DMA_RESUME; snd_azf3328_codec_outw(chip, IDX_IO_REC_FLAGS, status1); @@ -1129,8 +1318,8 @@ snd_azf3328_capture_trigger(struct snd_pcm_substream *substream, int cmd) status1 &= ~DMA_PLAY_SOMETHING1; snd_azf3328_codec_outw(chip, IDX_IO_REC_FLAGS, status1); spin_unlock(&chip->reg_lock); - - chip->is_recording = 0; + snd_azf3328_codec_activity(chip, AZF_CAPTURE, 0); + snd_azf3328_dbgplay("STOPPED CAPTURE\n"); break; case SNDRV_PCM_TRIGGER_SUSPEND: @@ -1149,7 +1338,7 @@ snd_azf3328_capture_trigger(struct snd_pcm_substream *substream, int cmd) printk(KERN_ERR "FIXME: unknown trigger mode!\n"); return -EINVAL; } - + snd_azf3328_dbgcallleave(); return result; } @@ -1162,11 +1351,11 @@ snd_azf3328_playback_pointer(struct snd_pcm_substream *substream) snd_pcm_uframes_t frmres; #ifdef QUERY_HARDWARE - bufptr = inl(chip->codec_port+IDX_IO_PLAY_DMA_START_1); + bufptr = snd_azf3328_codec_inl(chip, IDX_IO_PLAY_DMA_START_1); #else bufptr = substream->runtime->dma_addr; #endif - result = inl(chip->codec_port+IDX_IO_PLAY_DMA_CURRPOS); + result = snd_azf3328_codec_inl(chip, IDX_IO_PLAY_DMA_CURRPOS); /* calculate offset */ result -= bufptr; @@ -1183,11 +1372,11 @@ snd_azf3328_capture_pointer(struct snd_pcm_substream *substream) snd_pcm_uframes_t frmres; #ifdef QUERY_HARDWARE - bufptr = inl(chip->codec_port+IDX_IO_REC_DMA_START_1); + bufptr = snd_azf3328_codec_inl(chip, IDX_IO_REC_DMA_START_1); #else bufptr = substream->runtime->dma_addr; #endif - result = inl(chip->codec_port+IDX_IO_REC_DMA_CURRPOS); + result = snd_azf3328_codec_inl(chip, IDX_IO_REC_DMA_CURRPOS); /* calculate offset */ result -= bufptr; @@ -1196,27 +1385,241 @@ snd_azf3328_capture_pointer(struct snd_pcm_substream *substream) return frmres; } +/******************************************************************/ + +#ifdef SUPPORT_GAMEPORT +static inline void +snd_azf3328_gameport_irq_enable(struct snd_azf3328 *chip, int enable) +{ + snd_azf3328_io_reg_setb( + chip->game_io+IDX_GAME_HWCONFIG, + GAME_HWCFG_IRQ_ENABLE, + enable + ); +} + +static inline void +snd_azf3328_gameport_legacy_address_enable(struct snd_azf3328 *chip, int enable) +{ + snd_azf3328_io_reg_setb( + chip->game_io+IDX_GAME_HWCONFIG, + GAME_HWCFG_LEGACY_ADDRESS_ENABLE, + enable + ); +} + +static inline void +snd_azf3328_gameport_axis_circuit_enable(struct snd_azf3328 *chip, int enable) +{ + snd_azf3328_codec_reg_6AH_update( + chip, IO_6A_SOMETHING2_GAMEPORT, enable + ); +} + +static inline void +snd_azf3328_gameport_interrupt(struct snd_azf3328 *chip) +{ + /* + * skeleton handler only + * (we do not want axis reading in interrupt handler - too much load!) + */ + snd_azf3328_dbggame("gameport irq\n"); + + /* this should ACK the gameport IRQ properly, hopefully. */ + snd_azf3328_game_inw(chip, IDX_GAME_AXIS_VALUE); +} + +static int +snd_azf3328_gameport_open(struct gameport *gameport, int mode) +{ + struct snd_azf3328 *chip = gameport_get_port_data(gameport); + int res; + + snd_azf3328_dbggame("gameport_open, mode %d\n", mode); + switch (mode) { + case GAMEPORT_MODE_COOKED: + case GAMEPORT_MODE_RAW: + res = 0; + break; + default: + res = -1; + break; + } + + snd_azf3328_gameport_axis_circuit_enable(chip, (res == 0)); + + return res; +} + +static void +snd_azf3328_gameport_close(struct gameport *gameport) +{ + struct snd_azf3328 *chip = gameport_get_port_data(gameport); + + snd_azf3328_dbggame("gameport_close\n"); + snd_azf3328_gameport_axis_circuit_enable(chip, 0); +} + +static int +snd_azf3328_gameport_cooked_read(struct gameport *gameport, + int *axes, + int *buttons +) +{ + struct snd_azf3328 *chip = gameport_get_port_data(gameport); + int i; + u8 val; + unsigned long flags; + + snd_assert(chip, return 0); + + spin_lock_irqsave(&chip->reg_lock, flags); + val = snd_azf3328_game_inb(chip, IDX_GAME_LEGACY_COMPATIBLE); + *buttons = (~(val) >> 4) & 0xf; + + /* ok, this one is a bit dirty: cooked_read is being polled by a timer, + * thus we're atomic and cannot actively wait in here + * (which would be useful for us since it probably would be better + * to trigger a measurement in here, then wait a short amount of + * time until it's finished, then read values of _this_ measurement). + * + * Thus we simply resort to reading values if they're available already + * and trigger the next measurement. + */ + + val = snd_azf3328_game_inb(chip, IDX_GAME_AXES_CONFIG); + if (val & GAME_AXES_SAMPLING_READY) { + for (i = 0; i < 4; ++i) { + /* configure the axis to read */ + val = (i << 4) | 0x0f; + snd_azf3328_game_outb(chip, IDX_GAME_AXES_CONFIG, val); + + chip->axes[i] = snd_azf3328_game_inw( + chip, IDX_GAME_AXIS_VALUE + ); + } + } + + /* trigger next axes sampling, to be evaluated the next time we + * enter this function */ + + /* for some very, very strange reason we cannot enable + * Measurement Ready monitoring for all axes here, + * at least not when only one joystick connected */ + val = 0x03; /* we're able to monitor axes 1 and 2 only */ + snd_azf3328_game_outb(chip, IDX_GAME_AXES_CONFIG, val); + + snd_azf3328_game_outw(chip, IDX_GAME_AXIS_VALUE, 0xffff); + spin_unlock_irqrestore(&chip->reg_lock, flags); + + for (i = 0; i < 4; i++) { + axes[i] = chip->axes[i]; + if (axes[i] == 0xffff) + axes[i] = -1; + } + + snd_azf3328_dbggame("cooked_read: axes %d %d %d %d buttons %d\n", + axes[0], axes[1], axes[2], axes[3], *buttons + ); + + return 0; +} + +static int __devinit +snd_azf3328_gameport(struct snd_azf3328 *chip, int dev) +{ + struct gameport *gp; + + chip->gameport = gp = gameport_allocate_port(); + if (!gp) { + printk(KERN_ERR "azt3328: cannot alloc memory for gameport\n"); + return -ENOMEM; + } + + gameport_set_name(gp, "AZF3328 Gameport"); + gameport_set_phys(gp, "pci%s/gameport0", pci_name(chip->pci)); + gameport_set_dev_parent(gp, &chip->pci->dev); + gp->io = chip->game_io; + gameport_set_port_data(gp, chip); + + gp->open = snd_azf3328_gameport_open; + gp->close = snd_azf3328_gameport_close; + gp->fuzz = 16; /* seems ok */ + gp->cooked_read = snd_azf3328_gameport_cooked_read; + + /* DISABLE legacy address: we don't need it! */ + snd_azf3328_gameport_legacy_address_enable(chip, 0); + + snd_azf3328_gameport_axis_circuit_enable(chip, 0); + + gameport_register_port(chip->gameport); + + return 0; +} + +static void +snd_azf3328_gameport_free(struct snd_azf3328 *chip) +{ + if (chip->gameport) { + gameport_unregister_port(chip->gameport); + chip->gameport = NULL; + } + snd_azf3328_gameport_irq_enable(chip, 0); +} +#else +static inline int +snd_azf3328_gameport(struct snd_azf3328 *chip, int dev) { return -ENOSYS; } +static inline void +snd_azf3328_gameport_free(struct snd_azf3328 *chip) { } +static inline void +snd_azf3328_gameport_interrupt(struct snd_azf3328 *chip) +{ + printk(KERN_WARNING "huh, game port IRQ occurred!?\n"); +} +#endif /* SUPPORT_GAMEPORT */ + +/******************************************************************/ + +static inline void +snd_azf3328_irq_log_unknown_type(u8 which) +{ + snd_azf3328_dbgplay( + "azt3328: unknown IRQ type (%x) occurred, please report!\n", + which + ); +} + static irqreturn_t snd_azf3328_interrupt(int irq, void *dev_id) { struct snd_azf3328 *chip = dev_id; u8 status, which; +#if DEBUG_PLAY_REC static unsigned long irq_count; +#endif status = snd_azf3328_codec_inb(chip, IDX_IO_IRQSTATUS); /* fast path out, to ease interrupt sharing */ - if (!(status & (IRQ_PLAYBACK|IRQ_RECORDING|IRQ_MPU401|IRQ_TIMER))) + if (!(status & + (IRQ_PLAYBACK|IRQ_RECORDING|IRQ_GAMEPORT|IRQ_MPU401|IRQ_TIMER) + )) return IRQ_NONE; /* must be interrupt for another device */ - snd_azf3328_dbgplay("Interrupt %ld!\nIDX_IO_PLAY_FLAGS %04x, IDX_IO_PLAY_IRQTYPE %04x, IDX_IO_IRQSTATUS %04x\n", - irq_count, - snd_azf3328_codec_inw(chip, IDX_IO_PLAY_FLAGS), - snd_azf3328_codec_inw(chip, IDX_IO_PLAY_IRQTYPE), - status); - + snd_azf3328_dbgplay( + "irq_count %ld! IDX_IO_PLAY_FLAGS %04x, " + "IDX_IO_PLAY_IRQTYPE %04x, IDX_IO_IRQSTATUS %04x\n", + irq_count++ /* debug-only */, + snd_azf3328_codec_inw(chip, IDX_IO_PLAY_FLAGS), + snd_azf3328_codec_inw(chip, IDX_IO_PLAY_IRQTYPE), + status + ); + if (status & IRQ_TIMER) { - /* snd_azf3328_dbgplay("timer %ld\n", inl(chip->codec_port+IDX_IO_TIMER_VALUE) & TIMER_VALUE_MASK); */ + /* snd_azf3328_dbgplay("timer %ld\n", + snd_azf3328_codec_inl(chip, IDX_IO_TIMER_VALUE) + & TIMER_VALUE_MASK + ); */ if (chip->timer) snd_timer_interrupt(chip->timer, chip->timer->sticks); /* ACK timer */ @@ -1232,15 +1635,20 @@ snd_azf3328_interrupt(int irq, void *dev_id) snd_azf3328_codec_outb(chip, IDX_IO_PLAY_IRQTYPE, which); spin_unlock(&chip->reg_lock); - if (chip->pcm && chip->playback_substream) { - snd_pcm_period_elapsed(chip->playback_substream); + if (chip->pcm && chip->audio_stream[AZF_PLAYBACK].substream) { + snd_pcm_period_elapsed( + chip->audio_stream[AZF_PLAYBACK].substream + ); snd_azf3328_dbgplay("PLAY period done (#%x), @ %x\n", which, - inl(chip->codec_port+IDX_IO_PLAY_DMA_CURRPOS)); + snd_azf3328_codec_inl( + chip, IDX_IO_PLAY_DMA_CURRPOS + ) + ); } else - snd_azf3328_dbgplay("azt3328: ouch, irq handler problem!\n"); + printk(KERN_WARNING "azt3328: irq handler problem!\n"); if (which & IRQ_PLAY_SOMETHING) - snd_azf3328_dbgplay("azt3328: unknown play IRQ type occurred, please report!\n"); + snd_azf3328_irq_log_unknown_type(which); } if (status & IRQ_RECORDING) { spin_lock(&chip->reg_lock); @@ -1249,16 +1657,23 @@ snd_azf3328_interrupt(int irq, void *dev_id) snd_azf3328_codec_outb(chip, IDX_IO_REC_IRQTYPE, which); spin_unlock(&chip->reg_lock); - if (chip->pcm && chip->capture_substream) { - snd_pcm_period_elapsed(chip->capture_substream); + if (chip->pcm && chip->audio_stream[AZF_CAPTURE].substream) { + snd_pcm_period_elapsed( + chip->audio_stream[AZF_CAPTURE].substream + ); snd_azf3328_dbgplay("REC period done (#%x), @ %x\n", which, - inl(chip->codec_port+IDX_IO_REC_DMA_CURRPOS)); + snd_azf3328_codec_inl( + chip, IDX_IO_REC_DMA_CURRPOS + ) + ); } else - snd_azf3328_dbgplay("azt3328: ouch, irq handler problem!\n"); + printk(KERN_WARNING "azt3328: irq handler problem!\n"); if (which & IRQ_REC_SOMETHING) - snd_azf3328_dbgplay("azt3328: unknown rec IRQ type occurred, please report!\n"); + snd_azf3328_irq_log_unknown_type(which); } + if (status & IRQ_GAMEPORT) + snd_azf3328_gameport_interrupt(chip); /* MPU401 has less critical IRQ requirements * than timer and playback/recording, right? */ if (status & IRQ_MPU401) { @@ -1268,7 +1683,6 @@ snd_azf3328_interrupt(int irq, void *dev_id) * If so, then I don't know how... */ snd_azf3328_dbgplay("azt3328: MPU401 IRQ\n"); } - irq_count++; return IRQ_HANDLED; } @@ -1287,8 +1701,8 @@ static const struct snd_pcm_hardware snd_azf3328_playback = .rates = SNDRV_PCM_RATE_5512 | SNDRV_PCM_RATE_8000_48000 | SNDRV_PCM_RATE_KNOT, - .rate_min = 4000, - .rate_max = 66200, + .rate_min = AZF_FREQ_4000, + .rate_max = AZF_FREQ_66200, .channels_min = 1, .channels_max = 2, .buffer_bytes_max = 65536, @@ -1315,8 +1729,8 @@ static const struct snd_pcm_hardware snd_azf3328_capture = .rates = SNDRV_PCM_RATE_5512 | SNDRV_PCM_RATE_8000_48000 | SNDRV_PCM_RATE_KNOT, - .rate_min = 4000, - .rate_max = 66200, + .rate_min = AZF_FREQ_4000, + .rate_max = AZF_FREQ_66200, .channels_min = 1, .channels_max = 2, .buffer_bytes_max = 65536, @@ -1329,10 +1743,24 @@ static const struct snd_pcm_hardware snd_azf3328_capture = static unsigned int snd_azf3328_fixed_rates[] = { - 4000, 4800, 5512, 6620, 8000, 9600, 11025, 13240, 16000, 22050, 32000, - 44100, 48000, 66200 }; + AZF_FREQ_4000, + AZF_FREQ_4800, + AZF_FREQ_5512, + AZF_FREQ_6620, + AZF_FREQ_8000, + AZF_FREQ_9600, + AZF_FREQ_11025, + AZF_FREQ_13240, + AZF_FREQ_16000, + AZF_FREQ_22050, + AZF_FREQ_32000, + AZF_FREQ_44100, + AZF_FREQ_48000, + AZF_FREQ_66200 +}; + static struct snd_pcm_hw_constraint_list snd_azf3328_hw_constraints_rates = { - .count = ARRAY_SIZE(snd_azf3328_fixed_rates), + .count = ARRAY_SIZE(snd_azf3328_fixed_rates), .list = snd_azf3328_fixed_rates, .mask = 0, }; @@ -1346,7 +1774,7 @@ snd_azf3328_playback_open(struct snd_pcm_substream *substream) struct snd_pcm_runtime *runtime = substream->runtime; snd_azf3328_dbgcallenter(); - chip->playback_substream = substream; + chip->audio_stream[AZF_PLAYBACK].substream = substream; runtime->hw = snd_azf3328_playback; snd_pcm_hw_constraint_list(runtime, 0, SNDRV_PCM_HW_PARAM_RATE, &snd_azf3328_hw_constraints_rates); @@ -1361,7 +1789,7 @@ snd_azf3328_capture_open(struct snd_pcm_substream *substream) struct snd_pcm_runtime *runtime = substream->runtime; snd_azf3328_dbgcallenter(); - chip->capture_substream = substream; + chip->audio_stream[AZF_CAPTURE].substream = substream; runtime->hw = snd_azf3328_capture; snd_pcm_hw_constraint_list(runtime, 0, SNDRV_PCM_HW_PARAM_RATE, &snd_azf3328_hw_constraints_rates); @@ -1375,7 +1803,7 @@ snd_azf3328_playback_close(struct snd_pcm_substream *substream) struct snd_azf3328 *chip = snd_pcm_substream_chip(substream); snd_azf3328_dbgcallenter(); - chip->playback_substream = NULL; + chip->audio_stream[AZF_PLAYBACK].substream = NULL; snd_azf3328_dbgcallleave(); return 0; } @@ -1386,7 +1814,7 @@ snd_azf3328_capture_close(struct snd_pcm_substream *substream) struct snd_azf3328 *chip = snd_pcm_substream_chip(substream); snd_azf3328_dbgcallenter(); - chip->capture_substream = NULL; + chip->audio_stream[AZF_CAPTURE].substream = NULL; snd_azf3328_dbgcallleave(); return 0; } @@ -1441,102 +1869,8 @@ snd_azf3328_pcm(struct snd_azf3328 *chip, int device) /******************************************************************/ -#ifdef SUPPORT_JOYSTICK -static int __devinit -snd_azf3328_config_joystick(struct snd_azf3328 *chip, int dev) -{ - struct gameport *gp; - struct resource *r; - - if (!joystick[dev]) - return -ENODEV; - - if (!(r = request_region(0x200, 8, "AZF3328 gameport"))) { - printk(KERN_WARNING "azt3328: cannot reserve joystick ports\n"); - return -EBUSY; - } - - chip->gameport = gp = gameport_allocate_port(); - if (!gp) { - printk(KERN_ERR "azt3328: cannot allocate memory for gameport\n"); - release_and_free_resource(r); - return -ENOMEM; - } - - gameport_set_name(gp, "AZF3328 Gameport"); - gameport_set_phys(gp, "pci%s/gameport0", pci_name(chip->pci)); - gameport_set_dev_parent(gp, &chip->pci->dev); - gp->io = 0x200; - gameport_set_port_data(gp, r); - - snd_azf3328_io2_outb(chip, IDX_IO2_LEGACY_ADDR, - snd_azf3328_io2_inb(chip, IDX_IO2_LEGACY_ADDR) | LEGACY_JOY); - - gameport_register_port(chip->gameport); - - return 0; -} - -static void -snd_azf3328_free_joystick(struct snd_azf3328 *chip) -{ - if (chip->gameport) { - struct resource *r = gameport_get_port_data(chip->gameport); - - gameport_unregister_port(chip->gameport); - chip->gameport = NULL; - /* disable gameport */ - snd_azf3328_io2_outb(chip, IDX_IO2_LEGACY_ADDR, - snd_azf3328_io2_inb(chip, IDX_IO2_LEGACY_ADDR) & ~LEGACY_JOY); - release_and_free_resource(r); - } -} -#else -static inline int -snd_azf3328_config_joystick(struct snd_azf3328 *chip, int dev) { return -ENOSYS; } -static inline void -snd_azf3328_free_joystick(struct snd_azf3328 *chip) { } -#endif - -/******************************************************************/ - -static int -snd_azf3328_free(struct snd_azf3328 *chip) -{ - if (chip->irq < 0) - goto __end_hw; - - /* reset (close) mixer */ - snd_azf3328_mixer_set_mute(chip, IDX_MIXER_PLAY_MASTER, 1); /* first mute master volume */ - snd_azf3328_mixer_outw(chip, IDX_MIXER_RESET, 0x0000); - - /* interrupt setup - mask everything (FIXME!) */ - /* well, at least we know how to disable the timer IRQ */ - snd_azf3328_codec_outb(chip, IDX_IO_TIMER_VALUE + 3, 0x00); - - if (chip->irq >= 0) - synchronize_irq(chip->irq); -__end_hw: - snd_azf3328_free_joystick(chip); - if (chip->irq >= 0) - free_irq(chip->irq, chip); - pci_release_regions(chip->pci); - pci_disable_device(chip->pci); - - kfree(chip); - return 0; -} - -static int -snd_azf3328_dev_free(struct snd_device *device) -{ - struct snd_azf3328 *chip = device->device_data; - return snd_azf3328_free(chip); -} - -/******************************************************************/ - -/*** NOTE: the physical timer resolution actually is 1024000 ticks per second, +/*** NOTE: the physical timer resolution actually is 1024000 ticks per second + *** (probably derived from main crystal via a divider of 24), *** but announcing those attributes to user-space would make programs *** configure the timer to a 1 tick value, resulting in an absolutely fatal *** timer IRQ storm. @@ -1564,7 +1898,7 @@ snd_azf3328_timer_start(struct snd_timer *timer) delay = 49; /* minimum time is 49 ticks */ } snd_azf3328_dbgtimer("setting timer countdown value %d, add COUNTDOWN|IRQ\n", delay); - delay |= TIMER_ENABLE_COUNTDOWN | TIMER_ENABLE_IRQ; + delay |= TIMER_COUNTDOWN_ENABLE | TIMER_IRQ_ENABLE; spin_lock_irqsave(&chip->reg_lock, flags); snd_azf3328_codec_outl(chip, IDX_IO_TIMER_VALUE, delay); spin_unlock_irqrestore(&chip->reg_lock, flags); @@ -1582,7 +1916,7 @@ snd_azf3328_timer_stop(struct snd_timer *timer) chip = snd_timer_chip(timer); spin_lock_irqsave(&chip->reg_lock, flags); /* disable timer countdown and interrupt */ - /* FIXME: should we write TIMER_ACK_IRQ here? */ + /* FIXME: should we write TIMER_IRQ_ACK here? */ snd_azf3328_codec_outb(chip, IDX_IO_TIMER_VALUE + 3, 0); spin_unlock_irqrestore(&chip->reg_lock, flags); snd_azf3328_dbgcallleave(); @@ -1626,9 +1960,10 @@ snd_azf3328_timer(struct snd_azf3328 *chip, int device) snd_azf3328_timer_hw.resolution *= seqtimer_scaling; snd_azf3328_timer_hw.ticks /= seqtimer_scaling; - if ((err = snd_timer_new(chip->card, "AZF3328", &tid, &timer)) < 0) { + + err = snd_timer_new(chip->card, "AZF3328", &tid, &timer); + if (err < 0) goto out; - } strcpy(timer->name, "AZF3328 timer"); timer->private_data = chip; @@ -1636,6 +1971,8 @@ snd_azf3328_timer(struct snd_azf3328 *chip, int device) chip->timer = timer; + snd_azf3328_timer_stop(timer); + err = 0; out: @@ -1645,10 +1982,44 @@ out: /******************************************************************/ +static int +snd_azf3328_free(struct snd_azf3328 *chip) +{ + if (chip->irq < 0) + goto __end_hw; + + /* reset (close) mixer: + * first mute master volume, then reset + */ + snd_azf3328_mixer_set_mute(chip, IDX_MIXER_PLAY_MASTER, 1); + snd_azf3328_mixer_outw(chip, IDX_MIXER_RESET, 0x0000); + + snd_azf3328_timer_stop(chip->timer); + snd_azf3328_gameport_free(chip); + + if (chip->irq >= 0) + synchronize_irq(chip->irq); +__end_hw: + if (chip->irq >= 0) + free_irq(chip->irq, chip); + pci_release_regions(chip->pci); + pci_disable_device(chip->pci); + + kfree(chip); + return 0; +} + +static int +snd_azf3328_dev_free(struct snd_device *device) +{ + struct snd_azf3328 *chip = device->device_data; + return snd_azf3328_free(chip); +} + #if 0 /* check whether a bit can be modified */ static void -snd_azf3328_test_bit(unsigned int reg, int bit) +snd_azf3328_test_bit(unsigned unsigned reg, int bit) { unsigned char val, valoff, valon; @@ -1659,42 +2030,74 @@ snd_azf3328_test_bit(unsigned int reg, int bit) outb(val|(1 << bit), reg); valon = inb(reg); - + outb(val, reg); - printk(KERN_ERR "reg %04x bit %d: %02x %02x %02x\n", reg, bit, val, valoff, valon); + printk(KERN_ERR "reg %04x bit %d: %02x %02x %02x\n", + reg, bit, val, valoff, valon + ); } #endif -#if DEBUG_MISC -static void +static inline void snd_azf3328_debug_show_ports(const struct snd_azf3328 *chip) { +#if DEBUG_MISC u16 tmp; - snd_azf3328_dbgmisc("codec_port 0x%lx, io2_port 0x%lx, mpu_port 0x%lx, synth_port 0x%lx, mixer_port 0x%lx, irq %d\n", chip->codec_port, chip->io2_port, chip->mpu_port, chip->synth_port, chip->mixer_port, chip->irq); - - snd_azf3328_dbgmisc("io2 %02x %02x %02x %02x %02x %02x\n", snd_azf3328_io2_inb(chip, 0), snd_azf3328_io2_inb(chip, 1), snd_azf3328_io2_inb(chip, 2), snd_azf3328_io2_inb(chip, 3), snd_azf3328_io2_inb(chip, 4), snd_azf3328_io2_inb(chip, 5)); - - for (tmp=0; tmp <= 0x01; tmp += 1) - snd_azf3328_dbgmisc("0x%02x: opl 0x%04x, mpu300 0x%04x, mpu310 0x%04x, mpu320 0x%04x, mpu330 0x%04x\n", tmp, inb(0x388 + tmp), inb(0x300 + tmp), inb(0x310 + tmp), inb(0x320 + tmp), inb(0x330 + tmp)); + snd_azf3328_dbgmisc( + "codec_io 0x%lx, game_io 0x%lx, mpu_io 0x%lx, " + "opl3_io 0x%lx, mixer_io 0x%lx, irq %d\n", + chip->codec_io, chip->game_io, chip->mpu_io, + chip->opl3_io, chip->mixer_io, chip->irq + ); + + snd_azf3328_dbgmisc("game %02x %02x %02x %02x %02x %02x\n", + snd_azf3328_game_inb(chip, 0), + snd_azf3328_game_inb(chip, 1), + snd_azf3328_game_inb(chip, 2), + snd_azf3328_game_inb(chip, 3), + snd_azf3328_game_inb(chip, 4), + snd_azf3328_game_inb(chip, 5) + ); + + for (tmp = 0; tmp < 0x07; tmp += 1) + snd_azf3328_dbgmisc("mpu_io 0x%04x\n", inb(chip->mpu_io + tmp)); + + for (tmp = 0; tmp <= 0x07; tmp += 1) + snd_azf3328_dbgmisc("0x%02x: game200 0x%04x, game208 0x%04x\n", + tmp, inb(0x200 + tmp), inb(0x208 + tmp)); + + for (tmp = 0; tmp <= 0x01; tmp += 1) + snd_azf3328_dbgmisc( + "0x%02x: mpu300 0x%04x, mpu310 0x%04x, mpu320 0x%04x, " + "mpu330 0x%04x opl388 0x%04x opl38c 0x%04x\n", + tmp, + inb(0x300 + tmp), + inb(0x310 + tmp), + inb(0x320 + tmp), + inb(0x330 + tmp), + inb(0x388 + tmp), + inb(0x38c + tmp) + ); for (tmp = 0; tmp < AZF_IO_SIZE_CODEC; tmp += 2) - snd_azf3328_dbgmisc("codec 0x%02x: 0x%04x\n", tmp, snd_azf3328_codec_inw(chip, tmp)); + snd_azf3328_dbgmisc("codec 0x%02x: 0x%04x\n", + tmp, snd_azf3328_codec_inw(chip, tmp) + ); for (tmp = 0; tmp < AZF_IO_SIZE_MIXER; tmp += 2) - snd_azf3328_dbgmisc("mixer 0x%02x: 0x%04x\n", tmp, snd_azf3328_mixer_inw(chip, tmp)); + snd_azf3328_dbgmisc("mixer 0x%02x: 0x%04x\n", + tmp, snd_azf3328_mixer_inw(chip, tmp) + ); +#endif /* DEBUG_MISC */ } -#else -static inline void -snd_azf3328_debug_show_ports(const struct snd_azf3328 *chip) {} -#endif static int __devinit snd_azf3328_create(struct snd_card *card, - struct pci_dev *pci, - unsigned long device_type, - struct snd_azf3328 ** rchip) + struct pci_dev *pci, + unsigned long device_type, + struct snd_azf3328 **rchip) { struct snd_azf3328 *chip; int err; @@ -1705,7 +2108,8 @@ snd_azf3328_create(struct snd_card *card, *rchip = NULL; - if ((err = pci_enable_device(pci)) < 0) + err = pci_enable_device(pci); + if (err < 0) return err; chip = kzalloc(sizeof(*chip), GFP_KERNEL); @@ -1721,20 +2125,25 @@ snd_azf3328_create(struct snd_card *card, /* check if we can restrict PCI DMA transfers to 24 bits */ if (pci_set_dma_mask(pci, DMA_24BIT_MASK) < 0 || pci_set_consistent_dma_mask(pci, DMA_24BIT_MASK) < 0) { - snd_printk(KERN_ERR "architecture does not support 24bit PCI busmaster DMA\n"); + snd_printk(KERN_ERR "architecture does not support " + "24bit PCI busmaster DMA\n" + ); err = -ENXIO; goto out_err; } - if ((err = pci_request_regions(pci, "Aztech AZF3328")) < 0) { + err = pci_request_regions(pci, "Aztech AZF3328"); + if (err < 0) goto out_err; - } - chip->codec_port = pci_resource_start(pci, 0); - chip->io2_port = pci_resource_start(pci, 1); - chip->mpu_port = pci_resource_start(pci, 2); - chip->synth_port = pci_resource_start(pci, 3); - chip->mixer_port = pci_resource_start(pci, 4); + chip->codec_io = pci_resource_start(pci, 0); + chip->game_io = pci_resource_start(pci, 1); + chip->mpu_io = pci_resource_start(pci, 2); + chip->opl3_io = pci_resource_start(pci, 3); + chip->mixer_io = pci_resource_start(pci, 4); + + chip->audio_stream[AZF_PLAYBACK].portbase = chip->codec_io + 0x00; + chip->audio_stream[AZF_CAPTURE].portbase = chip->codec_io + 0x20; if (request_irq(pci->irq, snd_azf3328_interrupt, IRQF_SHARED, card->shortname, chip)) { @@ -1747,29 +2156,29 @@ snd_azf3328_create(struct snd_card *card, synchronize_irq(chip->irq); snd_azf3328_debug_show_ports(chip); - - if ((err = snd_device_new(card, SNDRV_DEV_LOWLEVEL, chip, &ops)) < 0) { + + err = snd_device_new(card, SNDRV_DEV_LOWLEVEL, chip, &ops); + if (err < 0) goto out_err; - } /* create mixer interface & switches */ - if ((err = snd_azf3328_mixer_new(chip)) < 0) + err = snd_azf3328_mixer_new(chip); + if (err < 0) goto out_err; -#if 0 - /* set very low bitrate to reduce noise and power consumption? */ - snd_azf3328_setfmt(chip, IDX_IO_PLAY_SOUNDFORMAT, 5512, 8, 1); -#endif + /* shutdown codecs to save power */ + /* have snd_azf3328_codec_activity() act properly */ + chip->audio_stream[AZF_PLAYBACK].running = 1; + snd_azf3328_codec_activity(chip, AZF_PLAYBACK, 0); /* standard chip init stuff */ - /* default IRQ init value */ + /* default IRQ init value */ tmp = DMA_PLAY_SOMETHING2|DMA_EPILOGUE_SOMETHING|DMA_SOMETHING_ELSE; spin_lock_irq(&chip->reg_lock); snd_azf3328_codec_outb(chip, IDX_IO_PLAY_FLAGS, tmp); snd_azf3328_codec_outb(chip, IDX_IO_REC_FLAGS, tmp); snd_azf3328_codec_outb(chip, IDX_IO_SOMETHING_FLAGS, tmp); - snd_azf3328_codec_outb(chip, IDX_IO_TIMER_VALUE + 3, 0x00); /* disable timer */ spin_unlock_irq(&chip->reg_lock); snd_card_set_dev(card, &pci->dev); @@ -1805,52 +2214,61 @@ snd_azf3328_probe(struct pci_dev *pci, const struct pci_device_id *pci_id) return -ENOENT; } - card = snd_card_new(index[dev], id[dev], THIS_MODULE, 0 ); + card = snd_card_new(index[dev], id[dev], THIS_MODULE, 0); if (card == NULL) return -ENOMEM; strcpy(card->driver, "AZF3328"); strcpy(card->shortname, "Aztech AZF3328 (PCI168)"); - if ((err = snd_azf3328_create(card, pci, pci_id->driver_data, &chip)) < 0) { + err = snd_azf3328_create(card, pci, pci_id->driver_data, &chip); + if (err < 0) goto out_err; - } card->private_data = chip; - if ((err = snd_mpu401_uart_new( card, 0, MPU401_HW_MPU401, - chip->mpu_port, MPU401_INFO_INTEGRATED, - pci->irq, 0, &chip->rmidi)) < 0) { - snd_printk(KERN_ERR "azf3328: no MPU-401 device at 0x%lx?\n", chip->mpu_port); + err = snd_mpu401_uart_new( + card, 0, MPU401_HW_MPU401, chip->mpu_io, MPU401_INFO_INTEGRATED, + pci->irq, 0, &chip->rmidi + ); + if (err < 0) { + snd_printk(KERN_ERR "azf3328: no MPU-401 device at 0x%lx?\n", + chip->mpu_io + ); goto out_err; } - if ((err = snd_azf3328_timer(chip, 0)) < 0) { + err = snd_azf3328_timer(chip, 0); + if (err < 0) goto out_err; - } - if ((err = snd_azf3328_pcm(chip, 0)) < 0) { + err = snd_azf3328_pcm(chip, 0); + if (err < 0) goto out_err; - } - if (snd_opl3_create(card, chip->synth_port, chip->synth_port+2, + if (snd_opl3_create(card, chip->opl3_io, chip->opl3_io+2, OPL3_HW_AUTO, 1, &opl3) < 0) { snd_printk(KERN_ERR "azf3328: no OPL3 device at 0x%lx-0x%lx?\n", - chip->synth_port, chip->synth_port+2 ); + chip->opl3_io, chip->opl3_io+2 + ); } else { - if ((err = snd_opl3_hwdep_new(opl3, 0, 1, NULL)) < 0) { + /* need to use IDs 1, 2 since ID 0 is snd_azf3328_timer above */ + err = snd_opl3_timer_new(opl3, 1, 2); + if (err < 0) + goto out_err; + err = snd_opl3_hwdep_new(opl3, 0, 1, NULL); + if (err < 0) goto out_err; - } } opl3->private_data = chip; sprintf(card->longname, "%s at 0x%lx, irq %i", - card->shortname, chip->codec_port, chip->irq); + card->shortname, chip->codec_io, chip->irq); - if ((err = snd_card_register(card)) < 0) { + err = snd_card_register(card); + if (err < 0) goto out_err; - } #ifdef MODULE printk( @@ -1861,19 +2279,18 @@ snd_azf3328_probe(struct pci_dev *pci, const struct pci_device_id *pci_id) 1024000 / seqtimer_scaling, seqtimer_scaling); #endif - if (snd_azf3328_config_joystick(chip, dev) < 0) - snd_azf3328_io2_outb(chip, IDX_IO2_LEGACY_ADDR, - snd_azf3328_io2_inb(chip, IDX_IO2_LEGACY_ADDR) & ~LEGACY_JOY); + snd_azf3328_gameport(chip, dev); pci_set_drvdata(pci, card); dev++; err = 0; goto out; - + out_err: + snd_printk(KERN_ERR "azf3328: something failed, exiting\n"); snd_card_free(card); - + out: snd_azf3328_dbgcallleave(); return err; @@ -1894,27 +2311,31 @@ snd_azf3328_suspend(struct pci_dev *pci, pm_message_t state) { struct snd_card *card = pci_get_drvdata(pci); struct snd_azf3328 *chip = card->private_data; - int reg; + unsigned reg; snd_power_change_state(card, SNDRV_CTL_POWER_D3hot); - + snd_pcm_suspend_all(chip->pcm); - for (reg = 0; reg < AZF_IO_SIZE_MIXER_PM / 2; reg++) - chip->saved_regs_mixer[reg] = inw(chip->mixer_port + reg * 2); + for (reg = 0; reg < AZF_IO_SIZE_MIXER_PM / 2; ++reg) + chip->saved_regs_mixer[reg] = inw(chip->mixer_io + reg * 2); /* make sure to disable master volume etc. to prevent looping sound */ snd_azf3328_mixer_set_mute(chip, IDX_MIXER_PLAY_MASTER, 1); snd_azf3328_mixer_set_mute(chip, IDX_MIXER_WAVEOUT, 1); - - for (reg = 0; reg < AZF_IO_SIZE_CODEC_PM / 2; reg++) - chip->saved_regs_codec[reg] = inw(chip->codec_port + reg * 2); - for (reg = 0; reg < AZF_IO_SIZE_IO2_PM / 2; reg++) - chip->saved_regs_io2[reg] = inw(chip->io2_port + reg * 2); - for (reg = 0; reg < AZF_IO_SIZE_MPU_PM / 2; reg++) - chip->saved_regs_mpu[reg] = inw(chip->mpu_port + reg * 2); - for (reg = 0; reg < AZF_IO_SIZE_SYNTH_PM / 2; reg++) - chip->saved_regs_synth[reg] = inw(chip->synth_port + reg * 2); + + for (reg = 0; reg < AZF_IO_SIZE_CODEC_PM / 2; ++reg) + chip->saved_regs_codec[reg] = inw(chip->codec_io + reg * 2); + + /* manually store the one currently relevant write-only reg, too */ + chip->saved_regs_codec[IDX_IO_6AH / 2] = chip->shadow_reg_codec_6AH; + + for (reg = 0; reg < AZF_IO_SIZE_GAME_PM / 2; ++reg) + chip->saved_regs_game[reg] = inw(chip->game_io + reg * 2); + for (reg = 0; reg < AZF_IO_SIZE_MPU_PM / 2; ++reg) + chip->saved_regs_mpu[reg] = inw(chip->mpu_io + reg * 2); + for (reg = 0; reg < AZF_IO_SIZE_OPL3_PM / 2; ++reg) + chip->saved_regs_opl3[reg] = inw(chip->opl3_io + reg * 2); pci_disable_device(pci); pci_save_state(pci); @@ -1927,7 +2348,7 @@ snd_azf3328_resume(struct pci_dev *pci) { struct snd_card *card = pci_get_drvdata(pci); struct snd_azf3328 *chip = card->private_data; - int reg; + unsigned reg; pci_set_power_state(pci, PCI_D0); pci_restore_state(pci); @@ -1939,23 +2360,21 @@ snd_azf3328_resume(struct pci_dev *pci) } pci_set_master(pci); - for (reg = 0; reg < AZF_IO_SIZE_IO2_PM / 2; reg++) - outw(chip->saved_regs_io2[reg], chip->io2_port + reg * 2); - for (reg = 0; reg < AZF_IO_SIZE_MPU_PM / 2; reg++) - outw(chip->saved_regs_mpu[reg], chip->mpu_port + reg * 2); - for (reg = 0; reg < AZF_IO_SIZE_SYNTH_PM / 2; reg++) - outw(chip->saved_regs_synth[reg], chip->synth_port + reg * 2); - for (reg = 0; reg < AZF_IO_SIZE_MIXER_PM / 2; reg++) - outw(chip->saved_regs_mixer[reg], chip->mixer_port + reg * 2); - for (reg = 0; reg < AZF_IO_SIZE_CODEC_PM / 2; reg++) - outw(chip->saved_regs_codec[reg], chip->codec_port + reg * 2); + for (reg = 0; reg < AZF_IO_SIZE_GAME_PM / 2; ++reg) + outw(chip->saved_regs_game[reg], chip->game_io + reg * 2); + for (reg = 0; reg < AZF_IO_SIZE_MPU_PM / 2; ++reg) + outw(chip->saved_regs_mpu[reg], chip->mpu_io + reg * 2); + for (reg = 0; reg < AZF_IO_SIZE_OPL3_PM / 2; ++reg) + outw(chip->saved_regs_opl3[reg], chip->opl3_io + reg * 2); + for (reg = 0; reg < AZF_IO_SIZE_MIXER_PM / 2; ++reg) + outw(chip->saved_regs_mixer[reg], chip->mixer_io + reg * 2); + for (reg = 0; reg < AZF_IO_SIZE_CODEC_PM / 2; ++reg) + outw(chip->saved_regs_codec[reg], chip->codec_io + reg * 2); snd_power_change_state(card, SNDRV_CTL_POWER_D0); return 0; } -#endif - - +#endif /* CONFIG_PM */ static struct pci_driver driver = { diff --git a/sound/pci/azt3328.h b/sound/pci/azt3328.h index 679fa992e2b..7e3e8942d07 100644 --- a/sound/pci/azt3328.h +++ b/sound/pci/azt3328.h @@ -1,7 +1,8 @@ #ifndef __SOUND_AZT3328_H #define __SOUND_AZT3328_H -/* "PU" == "power-up value", as tested on PCI168 PCI rev. 10 */ +/* "PU" == "power-up value", as tested on PCI168 PCI rev. 10 + * "WRITE_ONLY" == register does not indicate actual bit values */ /*** main I/O area port indices ***/ /* (only 0x70 of 0x80 bytes saved/restored by Windows driver) */ @@ -54,7 +55,10 @@ #define SOUNDFORMAT_XTAL1 0x00 #define SOUNDFORMAT_XTAL2 0x01 /* all _SUSPECTED_ values are not used by Windows drivers, so we don't - * have any hard facts, only rough measurements */ + * have any hard facts, only rough measurements. + * All we know is that the crystal used on the board has 24.576MHz, + * like many soundcards (which results in the frequencies below when + * using certain divider values selected by the values below) */ #define SOUNDFORMAT_FREQ_SUSPECTED_4000 0x0c | SOUNDFORMAT_XTAL1 #define SOUNDFORMAT_FREQ_SUSPECTED_4800 0x0a | SOUNDFORMAT_XTAL1 #define SOUNDFORMAT_FREQ_5510 0x0c | SOUNDFORMAT_XTAL2 @@ -72,6 +76,26 @@ #define SOUNDFORMAT_FLAG_16BIT 0x0010 #define SOUNDFORMAT_FLAG_2CHANNELS 0x0020 +/* define frequency helpers, for maximum value safety */ +enum azf_freq_t { +#define AZF_FREQ(rate) AZF_FREQ_##rate = rate + AZF_FREQ(4000), + AZF_FREQ(4800), + AZF_FREQ(5512), + AZF_FREQ(6620), + AZF_FREQ(8000), + AZF_FREQ(9600), + AZF_FREQ(11025), + AZF_FREQ(13240), + AZF_FREQ(16000), + AZF_FREQ(22050), + AZF_FREQ(32000), + AZF_FREQ(44100), + AZF_FREQ(48000), + AZF_FREQ(66200), +#undef AZF_FREQ +} AZF_FREQUENCIES; + /** recording area (see also: playback bit flag definitions) **/ #define IDX_IO_REC_FLAGS 0x20 /* ??, PU:0x0000 */ #define IDX_IO_REC_IRQTYPE 0x22 /* ??, PU:0x0000 */ @@ -97,40 +121,171 @@ /** DirectX timer, main interrupt area (FIXME: and something else?) **/ #define IDX_IO_TIMER_VALUE 0x60 /* found this timer area by pure luck :-) */ - #define TIMER_VALUE_MASK 0x000fffffUL /* timer countdown value; triggers IRQ when timer is finished */ - #define TIMER_ENABLE_COUNTDOWN 0x01000000UL /* activate the timer countdown */ - #define TIMER_ENABLE_IRQ 0x02000000UL /* trigger timer IRQ on zero transition */ - #define TIMER_ACK_IRQ 0x04000000UL /* being set in IRQ handler in case port 0x00 (hmm, not port 0x64!?!?) had 0x0020 set upon IRQ handler */ + /* timer countdown value; triggers IRQ when timer is finished */ + #define TIMER_VALUE_MASK 0x000fffffUL + /* activate timer countdown */ + #define TIMER_COUNTDOWN_ENABLE 0x01000000UL + /* trigger timer IRQ on zero transition */ + #define TIMER_IRQ_ENABLE 0x02000000UL + /* being set in IRQ handler in case port 0x00 (hmm, not port 0x64!?!?) + * had 0x0020 set upon IRQ handler */ + #define TIMER_IRQ_ACK 0x04000000UL #define IDX_IO_IRQSTATUS 0x64 - #define IRQ_PLAYBACK 0x0001 - #define IRQ_RECORDING 0x0002 - #define IRQ_MPU401 0x0010 - #define IRQ_TIMER 0x0020 /* DirectX timer */ - #define IRQ_UNKNOWN1 0x0040 /* probably unused, or possibly I2S port? or gameport IRQ? */ - #define IRQ_UNKNOWN2 0x0080 /* probably unused, or possibly I2S port? or gameport IRQ? */ + /* some IRQ bit in here might also be used to signal a power-management timer + * timeout, to request shutdown of the chip (e.g. AD1815JS has such a thing). + * Some OPL3 hardware (e.g. in LM4560) has some special timer hardware which + * can trigger an OPL3 timer IRQ, so maybe there's such a thing as well... */ + + #define IRQ_PLAYBACK 0x0001 + #define IRQ_RECORDING 0x0002 + #define IRQ_UNKNOWN1 0x0004 /* most probably I2S port */ + #define IRQ_GAMEPORT 0x0008 /* Interrupt of Digital(ly) Enhanced Game Port */ + #define IRQ_MPU401 0x0010 + #define IRQ_TIMER 0x0020 /* DirectX timer */ + #define IRQ_UNKNOWN2 0x0040 /* probably unused, or possibly I2S port? */ + #define IRQ_UNKNOWN3 0x0080 /* probably unused, or possibly I2S port? */ #define IDX_IO_66H 0x66 /* writing 0xffff returns 0x0000 */ -#define IDX_IO_SOME_VALUE 0x68 /* this is set to e.g. 0x3ff or 0x300, and writable; maybe some buffer limit, but I couldn't find out more, PU:0x00ff */ -#define IDX_IO_6AH 0x6A /* this WORD can be set to have bits 0x0028 activated (FIXME: correct??); actually inhibits PCM playback!!! maybe power management?? */ - #define IO_6A_PAUSE_PLAYBACK 0x0200 /* bit 9; sure, this pauses playback, but what the heck is this really about?? */ -#define IDX_IO_6CH 0x6C -#define IDX_IO_6EH 0x6E /* writing 0xffff returns 0x83fe */ -/* further I/O indices not saved/restored, so probably not used */ + /* this is set to e.g. 0x3ff or 0x300, and writable; + * maybe some buffer limit, but I couldn't find out more, PU:0x00ff: */ +#define IDX_IO_SOME_VALUE 0x68 + #define IO_68_RANDOM_TOGGLE1 0x0100 /* toggles randomly */ + #define IO_68_RANDOM_TOGGLE2 0x0200 /* toggles randomly */ + /* umm, nope, behaviour of these bits changes depending on what we wrote + * to 0x6b!! + * And they change upon playback/stop, too: + * Writing a value to 0x68 will display this exact value during playback, + * too but when stopped it can fall back to a rather different + * seemingly random value). Hmm, possibly this is a register which + * has a remote shadow which needs proper device supply which only exists + * in case playback is active? Or is this driver-induced? + */ + +/* this WORD can be set to have bits 0x0028 activated (FIXME: correct??); + * actually inhibits PCM playback!!! maybe power management??: */ +#define IDX_IO_6AH 0x6A /* WRITE_ONLY! */ + /* bit 5: enabling this will activate permanent counting of bytes 2/3 + * at gameport I/O (0xb402/3) (equal values each) and cause + * gameport legacy I/O at 0x0200 to be _DISABLED_! + * Is this Digital Enhanced Game Port Enable??? Or maybe it's Testmode + * for Enhanced Digital Gameport (see 4D Wave DX card): */ + #define IO_6A_SOMETHING1_GAMEPORT 0x0020 + /* bit 8; sure, this _pauses_ playback (later resumes at same spot!), + * but what the heck is this really about??: */ + #define IO_6A_PAUSE_PLAYBACK_BIT8 0x0100 + /* bit 9; sure, this _pauses_ playback (later resumes at same spot!), + * but what the heck is this really about??: */ + #define IO_6A_PAUSE_PLAYBACK_BIT9 0x0200 + /* BIT8 and BIT9 are _NOT_ able to affect OPL3 MIDI playback, + * thus it suggests influence on PCM only!! + * However OTOH there seems to be no bit anywhere around here + * which is able to disable OPL3... */ + /* bit 10: enabling this actually changes values at legacy gameport + * I/O address (0x200); is this enabling of the Digital Enhanced Game Port??? + * Or maybe this simply switches off the NE558 circuit, since enabling this + * still lets us evaluate button states, but not axis states */ + #define IO_6A_SOMETHING2_GAMEPORT 0x0400 + /* writing 0x0300: causes quite some crackling during + * PC activity such as switching windows (PCI traffic?? + * --> FIFO/timing settings???) */ + /* writing 0x0100 plus/or 0x0200 inhibits playback */ + /* since the Windows .INF file has Flag_Enable_JoyStick and + * Flag_Enable_SB_DOS_Emulation directly together, it stands to reason + * that some other bit in this same register might be responsible + * for SB DOS Emulation activation (note that the file did NOT define + * a switch for OPL3!) */ +#define IDX_IO_6CH 0x6C /* unknown; fully read-writable */ +#define IDX_IO_6EH 0x6E + /* writing 0xffff returns 0x83fe (or 0x03fe only). + * writing 0x83 (and only 0x83!!) to 0x6f will cause 0x6c to switch + * from 0000 to ffff. */ +/* further I/O indices not saved/restored and not readable after writing, + * so probably not used */ -/*** I/O 2 area port indices ***/ + +/*** Gameport area port indices ***/ /* (only 0x06 of 0x08 bytes saved/restored by Windows driver) */ -#define AZF_IO_SIZE_IO2 0x08 -#define AZF_IO_SIZE_IO2_PM 0x06 +#define AZF_IO_SIZE_GAME 0x08 +#define AZF_IO_SIZE_GAME_PM 0x06 + +enum { + AZF_GAME_LEGACY_IO_PORT = 0x200 +} AZF_GAME_CONFIGS; + +#define IDX_GAME_LEGACY_COMPATIBLE 0x00 + /* in some operation mode, writing anything to this port + * triggers an interrupt: + * yup, that's in case IDX_GAME_01H has one of the + * axis measurement bits enabled + * (and of course one needs to have GAME_HWCFG_IRQ_ENABLE, too) */ + +#define IDX_GAME_AXES_CONFIG 0x01 + /* NOTE: layout of this register awfully similar (read: "identical??") + * to AD1815JS.pdf (p.29) */ + + /* enables axis 1 (X axis) measurement: */ + #define GAME_AXES_ENABLE_1 0x01 + /* enables axis 2 (Y axis) measurement: */ + #define GAME_AXES_ENABLE_2 0x02 + /* enables axis 3 (X axis) measurement: */ + #define GAME_AXES_ENABLE_3 0x04 + /* enables axis 4 (Y axis) measurement: */ + #define GAME_AXES_ENABLE_4 0x08 + /* selects the current axis to read the measured value of + * (at IDX_GAME_AXIS_VALUE): + * 00 = axis 1, 01 = axis 2, 10 = axis 3, 11 = axis 4: */ + #define GAME_AXES_READ_MASK 0x30 + /* enable to have the latch continuously accept ADC values + * (and continuously cause interrupts in case interrupts are enabled); + * AD1815JS.pdf says it's ~16ms interval there: */ + #define GAME_AXES_LATCH_ENABLE 0x40 + /* joystick data (measured axes) ready for reading: */ + #define GAME_AXES_SAMPLING_READY 0x80 + + /* NOTE: other card specs (SiS960 and others!) state that the + * game position latches should be frozen when reading and be freed + * (== reset?) after reading!!! + * Freezing most likely means disabling 0x40 (GAME_AXES_LATCH_ENABLE), + * but how to free the value? */ + /* An internet search for "gameport latch ADC" should provide some insight + * into how to program such a gameport system. */ + + /* writing 0xf0 to 01H once reset both counters to 0, in some special mode!? + * yup, in case 6AH 0x20 is not enabled + * (and 0x40 is sufficient, 0xf0 is not needed) */ + +#define IDX_GAME_AXIS_VALUE 0x02 + /* R: value of currently configured axis (word value!); + * W: trigger axis measurement */ + +#define IDX_GAME_HWCONFIG 0x04 + /* note: bits 4 to 7 are never set (== 0) when reading! + * --> reserved bits? */ + /* enables IRQ notification upon axes measurement ready: */ + #define GAME_HWCFG_IRQ_ENABLE 0x01 + /* these bits choose a different frequency for the + * internal ADC counter increment. + * hmm, seems to be a combo of bits: + * 00 --> standard frequency + * 10 --> 1/2 + * 01 --> 1/20 + * 11 --> 1/200: */ + #define GAME_HWCFG_ADC_COUNTER_FREQ_MASK 0x06 -#define IDX_IO2_LEGACY_ADDR 0x04 - #define LEGACY_SOMETHING 0x01 /* OPL3?? */ - #define LEGACY_JOY 0x08 + /* enable gameport legacy I/O address (0x200) + * I was unable to locate any configurability for a different address: */ + #define GAME_HWCFG_LEGACY_ADDRESS_ENABLE 0x08 +/*** MPU401 ***/ #define AZF_IO_SIZE_MPU 0x04 #define AZF_IO_SIZE_MPU_PM 0x04 -#define AZF_IO_SIZE_SYNTH 0x08 -#define AZF_IO_SIZE_SYNTH_PM 0x06 +/*** OPL3 synth ***/ +#define AZF_IO_SIZE_OPL3 0x08 +#define AZF_IO_SIZE_OPL3_PM 0x06 +/* hmm, given that a standard OPL3 has 4 registers only, + * there might be some enhanced functionality lurking at the end + * (especially since register 0x04 has a "non-empty" value 0xfe) */ /*** mixer I/O area port indices ***/ /* (only 0x22 of 0x40 bytes saved/restored by Windows driver) diff --git a/sound/pci/ca0106/ca0106_main.c b/sound/pci/ca0106/ca0106_main.c index ecbe79b67e4..2f8b28add27 100644 --- a/sound/pci/ca0106/ca0106_main.c +++ b/sound/pci/ca0106/ca0106_main.c @@ -249,6 +249,11 @@ static struct snd_ca0106_details ca0106_chip_details[] = { .name = "MSI K8N Diamond MB [SB0438]", .gpio_type = 2, .i2c_adc = 1 } , + /* Another MSI K8N Diamond MB, which has apprently a different SSID */ + { .serial = 0x10091102, + .name = "MSI K8N Diamond MB", + .gpio_type = 2, + .i2c_adc = 1 } , /* Shuttle XPC SD31P which has an onboard Creative Labs * Sound Blaster Live! 24-bit EAX * high-definition 7.1 audio processor". diff --git a/sound/pci/emu10k1/emu10k1_main.c b/sound/pci/emu10k1/emu10k1_main.c index 548c9cc81af..2f283ea6ad9 100644 --- a/sound/pci/emu10k1/emu10k1_main.c +++ b/sound/pci/emu10k1/emu10k1_main.c @@ -1528,6 +1528,7 @@ static struct snd_emu_chip_details emu_chip_details[] = { .ca0151_chip = 1, .spk71 = 1, .spdif_bug = 1, + .invert_shared_spdif = 1, /* digital/analog switch swapped */ .adc_1361t = 1, /* 24 bit capture instead of 16bit. Fixes ALSA bug#324 */ .ac97_chip = 1} , {.vendor = 0x1102, .device = 0x0004, .revision = 0x04, diff --git a/sound/pci/emu10k1/emumixer.c b/sound/pci/emu10k1/emumixer.c index fd221209abc..f34bbfb705f 100644 --- a/sound/pci/emu10k1/emumixer.c +++ b/sound/pci/emu10k1/emumixer.c @@ -1578,6 +1578,10 @@ static int snd_emu10k1_shared_spdif_get(struct snd_kcontrol *kcontrol, ucontrol->value.integer.value[0] = inl(emu->port + A_IOCFG) & A_IOCFG_GPOUT0 ? 1 : 0; else ucontrol->value.integer.value[0] = inl(emu->port + HCFG) & HCFG_GPOUT0 ? 1 : 0; + if (emu->card_capabilities->invert_shared_spdif) + ucontrol->value.integer.value[0] = + !ucontrol->value.integer.value[0]; + return 0; } @@ -1586,15 +1590,18 @@ static int snd_emu10k1_shared_spdif_put(struct snd_kcontrol *kcontrol, { unsigned long flags; struct snd_emu10k1 *emu = snd_kcontrol_chip(kcontrol); - unsigned int reg, val; + unsigned int reg, val, sw; int change = 0; + sw = ucontrol->value.integer.value[0]; + if (emu->card_capabilities->invert_shared_spdif) + sw = !sw; spin_lock_irqsave(&emu->reg_lock, flags); if ( emu->card_capabilities->i2c_adc) { /* Do nothing for Audigy 2 ZS Notebook */ } else if (emu->audigy) { reg = inl(emu->port + A_IOCFG); - val = ucontrol->value.integer.value[0] ? A_IOCFG_GPOUT0 : 0; + val = sw ? A_IOCFG_GPOUT0 : 0; change = (reg & A_IOCFG_GPOUT0) != val; if (change) { reg &= ~A_IOCFG_GPOUT0; @@ -1603,7 +1610,7 @@ static int snd_emu10k1_shared_spdif_put(struct snd_kcontrol *kcontrol, } } reg = inl(emu->port + HCFG); - val = ucontrol->value.integer.value[0] ? HCFG_GPOUT0 : 0; + val = sw ? HCFG_GPOUT0 : 0; change |= (reg & HCFG_GPOUT0) != val; if (change) { reg &= ~HCFG_GPOUT0; diff --git a/sound/pci/emu10k1/memory.c b/sound/pci/emu10k1/memory.c index 916c1dbcd53..7d379f5131f 100644 --- a/sound/pci/emu10k1/memory.c +++ b/sound/pci/emu10k1/memory.c @@ -437,43 +437,49 @@ static void get_single_page_range(struct snd_util_memhdr *hdr, *last_page_ret = last_page; } +/* release allocated pages */ +static void __synth_free_pages(struct snd_emu10k1 *emu, int first_page, + int last_page) +{ + int page; + + for (page = first_page; page <= last_page; page++) { + free_page((unsigned long)emu->page_ptr_table[page]); + emu->page_addr_table[page] = 0; + emu->page_ptr_table[page] = NULL; + } +} + /* * allocate kernel pages */ static int synth_alloc_pages(struct snd_emu10k1 *emu, struct snd_emu10k1_memblk *blk) { int page, first_page, last_page; - struct snd_dma_buffer dmab; emu10k1_memblk_init(blk); get_single_page_range(emu->memhdr, blk, &first_page, &last_page); /* allocate kernel pages */ for (page = first_page; page <= last_page; page++) { - if (snd_dma_alloc_pages(SNDRV_DMA_TYPE_DEV, snd_dma_pci_data(emu->pci), - PAGE_SIZE, &dmab) < 0) - goto __fail; - if (! is_valid_page(emu, dmab.addr)) { - snd_dma_free_pages(&dmab); - goto __fail; + /* first try to allocate from <4GB zone */ + struct page *p = alloc_page(GFP_KERNEL | GFP_DMA32 | + __GFP_NOWARN); + if (!p || (page_to_pfn(p) & ~(emu->dma_mask >> PAGE_SHIFT))) { + if (p) + __free_page(p); + /* try to allocate from <16MB zone */ + p = alloc_page(GFP_ATOMIC | GFP_DMA | + __GFP_NORETRY | /* no OOM-killer */ + __GFP_NOWARN); + } + if (!p) { + __synth_free_pages(emu, first_page, page - 1); + return -ENOMEM; } - emu->page_addr_table[page] = dmab.addr; - emu->page_ptr_table[page] = dmab.area; + emu->page_addr_table[page] = page_to_phys(p); + emu->page_ptr_table[page] = page_address(p); } return 0; - -__fail: - /* release allocated pages */ - last_page = page - 1; - for (page = first_page; page <= last_page; page++) { - dmab.area = emu->page_ptr_table[page]; - dmab.addr = emu->page_addr_table[page]; - dmab.bytes = PAGE_SIZE; - snd_dma_free_pages(&dmab); - emu->page_addr_table[page] = 0; - emu->page_ptr_table[page] = NULL; - } - - return -ENOMEM; } /* @@ -481,23 +487,10 @@ __fail: */ static int synth_free_pages(struct snd_emu10k1 *emu, struct snd_emu10k1_memblk *blk) { - int page, first_page, last_page; - struct snd_dma_buffer dmab; + int first_page, last_page; get_single_page_range(emu->memhdr, blk, &first_page, &last_page); - dmab.dev.type = SNDRV_DMA_TYPE_DEV; - dmab.dev.dev = snd_dma_pci_data(emu->pci); - for (page = first_page; page <= last_page; page++) { - if (emu->page_ptr_table[page] == NULL) - continue; - dmab.area = emu->page_ptr_table[page]; - dmab.addr = emu->page_addr_table[page]; - dmab.bytes = PAGE_SIZE; - snd_dma_free_pages(&dmab); - emu->page_addr_table[page] = 0; - emu->page_ptr_table[page] = NULL; - } - + __synth_free_pages(emu, first_page, last_page); return 0; } diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c index a6be6e3e871..d2e1093f8e9 100644 --- a/sound/pci/hda/hda_codec.c +++ b/sound/pci/hda/hda_codec.c @@ -2335,7 +2335,7 @@ int snd_hda_check_board_config(struct hda_codec *codec, if (!tbl) return -1; if (tbl->value >= 0 && tbl->value < num_configs) { -#ifdef CONFIG_SND_DEBUG_DETECT +#ifdef CONFIG_SND_DEBUG_VERBOSE char tmp[10]; const char *model = NULL; if (models) diff --git a/sound/pci/hda/hda_codec.h b/sound/pci/hda/hda_codec.h index dcd390b2bba..efc682888b3 100644 --- a/sound/pci/hda/hda_codec.h +++ b/sound/pci/hda/hda_codec.h @@ -78,7 +78,7 @@ enum { #define AC_VERB_GET_BEEP_CONTROL 0x0f0a #define AC_VERB_GET_EAPD_BTLENABLE 0x0f0c #define AC_VERB_GET_DIGI_CONVERT_1 0x0f0d -#define AC_VERB_GET_DIGI_CONVERT_2 0x0f0e +#define AC_VERB_GET_DIGI_CONVERT_2 0x0f0e /* unused */ #define AC_VERB_GET_VOLUME_KNOB_CONTROL 0x0f0f /* f10-f1a: GPIO */ #define AC_VERB_GET_GPIO_DATA 0x0f15 diff --git a/sound/pci/hda/hda_hwdep.c b/sound/pci/hda/hda_hwdep.c index 2177d9af533..6e18a422d99 100644 --- a/sound/pci/hda/hda_hwdep.c +++ b/sound/pci/hda/hda_hwdep.c @@ -88,7 +88,7 @@ static int hda_hwdep_ioctl_compat(struct snd_hwdep *hw, struct file *file, static int hda_hwdep_open(struct snd_hwdep *hw, struct file *file) { -#ifndef CONFIG_SND_DEBUG_DETECT +#ifndef CONFIG_SND_DEBUG_VERBOSE if (!capable(CAP_SYS_RAWIO)) return -EACCES; #endif diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index b3a618eb42c..16715a68ba5 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -55,6 +55,7 @@ static char *id[SNDRV_CARDS] = SNDRV_DEFAULT_STR; static int enable[SNDRV_CARDS] = SNDRV_DEFAULT_ENABLE_PNP; static char *model[SNDRV_CARDS]; static int position_fix[SNDRV_CARDS]; +static int bdl_pos_adj[SNDRV_CARDS] = {[0 ... (SNDRV_CARDS-1)] = -1}; static int probe_mask[SNDRV_CARDS] = {[0 ... (SNDRV_CARDS-1)] = -1}; static int single_cmd; static int enable_msi; @@ -69,7 +70,9 @@ module_param_array(model, charp, NULL, 0444); MODULE_PARM_DESC(model, "Use the given board model."); module_param_array(position_fix, int, NULL, 0444); MODULE_PARM_DESC(position_fix, "Fix DMA pointer " - "(0 = auto, 1 = none, 2 = POSBUF, 3 = FIFO size)."); + "(0 = auto, 1 = none, 2 = POSBUF)."); +module_param_array(bdl_pos_adj, int, NULL, 0644); +MODULE_PARM_DESC(bdl_pos_adj, "BDL position adjustment offset."); module_param_array(probe_mask, int, NULL, 0444); MODULE_PARM_DESC(probe_mask, "Bitmask to probe codecs (default = -1)."); module_param(single_cmd, bool, 0444); @@ -197,6 +200,10 @@ enum { SDI0, SDI1, SDI2, SDI3, SDO0, SDO1, SDO2, SDO3 }; #define ATIHDMI_NUM_CAPTURE 0 #define ATIHDMI_NUM_PLAYBACK 1 +/* TERA has 4 playback and 3 capture */ +#define TERA_NUM_CAPTURE 3 +#define TERA_NUM_PLAYBACK 4 + /* this number is statically defined for simplicity */ #define MAX_AZX_DEV 16 @@ -259,9 +266,8 @@ enum { SDI0, SDI1, SDI2, SDI3, SDO0, SDO1, SDO2, SDO3 }; /* position fix mode */ enum { POS_FIX_AUTO, - POS_FIX_NONE, + POS_FIX_LPIB, POS_FIX_POSBUF, - POS_FIX_FIFO, }; /* Defines for ATI HD Audio support in SB450 south bridge */ @@ -285,6 +291,7 @@ struct azx_dev { u32 *posbuf; /* position buffer pointer */ unsigned int bufsize; /* size of the play buffer in bytes */ + unsigned int period_bytes; /* size of the period in bytes */ unsigned int frags; /* number for period in the play buffer */ unsigned int fifo_size; /* FIFO size */ @@ -301,11 +308,11 @@ struct azx_dev { */ unsigned char stream_tag; /* assigned stream */ unsigned char index; /* stream index */ - /* for sanity check of position buffer */ - unsigned int period_intr; unsigned int opened :1; unsigned int running :1; + unsigned int irq_pending :1; + unsigned int irq_ignore :1; }; /* CORB/RIRB */ @@ -323,6 +330,7 @@ struct azx_rb { struct azx { struct snd_card *card; struct pci_dev *pci; + int dev_index; /* chip type specific */ int driver_type; @@ -366,9 +374,13 @@ struct azx { unsigned int single_cmd :1; unsigned int polling_mode :1; unsigned int msi :1; + unsigned int irq_pending_warned :1; /* for debugging */ unsigned int last_cmd; /* last issued command (to sync) */ + + /* for pending irqs */ + struct work_struct irq_pending_work; }; /* driver types */ @@ -381,6 +393,7 @@ enum { AZX_DRIVER_SIS, AZX_DRIVER_ULI, AZX_DRIVER_NVIDIA, + AZX_DRIVER_TERA, }; static char *driver_short_names[] __devinitdata = { @@ -392,6 +405,7 @@ static char *driver_short_names[] __devinitdata = { [AZX_DRIVER_SIS] = "HDA SIS966", [AZX_DRIVER_ULI] = "HDA ULI M5461", [AZX_DRIVER_NVIDIA] = "HDA NVidia", + [AZX_DRIVER_TERA] = "HDA Teradici", }; /* @@ -426,11 +440,6 @@ static char *driver_short_names[] __devinitdata = { /* for pcm support */ #define get_azx_dev(substream) (substream->runtime->private_data) -/* Get the upper 32bit of the given dma_addr_t - * Compiler should optimize and eliminate the code if dma_addr_t is 32bit - */ -#define upper_32bit(addr) (sizeof(addr) > 4 ? (u32)((addr) >> 32) : (u32)0) - static int azx_acquire_irq(struct azx *chip, int do_disconnect); /* @@ -461,7 +470,7 @@ static void azx_init_cmd_io(struct azx *chip) chip->corb.addr = chip->rb.addr; chip->corb.buf = (u32 *)chip->rb.area; azx_writel(chip, CORBLBASE, (u32)chip->corb.addr); - azx_writel(chip, CORBUBASE, upper_32bit(chip->corb.addr)); + azx_writel(chip, CORBUBASE, upper_32_bits(chip->corb.addr)); /* set the corb size to 256 entries (ULI requires explicitly) */ azx_writeb(chip, CORBSIZE, 0x02); @@ -476,7 +485,7 @@ static void azx_init_cmd_io(struct azx *chip) chip->rirb.addr = chip->rb.addr + 2048; chip->rirb.buf = (u32 *)(chip->rb.area + 2048); azx_writel(chip, RIRBLBASE, (u32)chip->rirb.addr); - azx_writel(chip, RIRBUBASE, upper_32bit(chip->rirb.addr)); + azx_writel(chip, RIRBUBASE, upper_32_bits(chip->rirb.addr)); /* set the rirb size to 256 entries (ULI requires explicitly) */ azx_writeb(chip, RIRBSIZE, 0x02); @@ -847,7 +856,7 @@ static void azx_init_chip(struct azx *chip) /* program the position buffer */ azx_writel(chip, DPLBASE, (u32)chip->posbuf.addr); - azx_writel(chip, DPUBASE, upper_32bit(chip->posbuf.addr)); + azx_writel(chip, DPUBASE, upper_32_bits(chip->posbuf.addr)); chip->initialized = 1; } @@ -908,6 +917,8 @@ static void azx_init_pci(struct azx *chip) } +static int azx_position_ok(struct azx *chip, struct azx_dev *azx_dev); + /* * interrupt handler */ @@ -930,11 +941,23 @@ static irqreturn_t azx_interrupt(int irq, void *dev_id) azx_dev = &chip->azx_dev[i]; if (status & azx_dev->sd_int_sta_mask) { azx_sd_writeb(azx_dev, SD_STS, SD_INT_MASK); - if (azx_dev->substream && azx_dev->running) { - azx_dev->period_intr++; + if (!azx_dev->substream || !azx_dev->running) + continue; + /* ignore the first dummy IRQ (due to pos_adj) */ + if (azx_dev->irq_ignore) { + azx_dev->irq_ignore = 0; + continue; + } + /* check whether this IRQ is really acceptable */ + if (azx_position_ok(chip, azx_dev)) { + azx_dev->irq_pending = 0; spin_unlock(&chip->reg_lock); snd_pcm_period_elapsed(azx_dev->substream); spin_lock(&chip->reg_lock); + } else { + /* bogus IRQ, process it later */ + azx_dev->irq_pending = 1; + schedule_work(&chip->irq_pending_work); } } } @@ -959,59 +982,107 @@ static irqreturn_t azx_interrupt(int irq, void *dev_id) /* + * set up a BDL entry + */ +static int setup_bdle(struct snd_pcm_substream *substream, + struct azx_dev *azx_dev, u32 **bdlp, + int ofs, int size, int with_ioc) +{ + struct snd_sg_buf *sgbuf = snd_pcm_substream_sgbuf(substream); + u32 *bdl = *bdlp; + + while (size > 0) { + dma_addr_t addr; + int chunk; + + if (azx_dev->frags >= AZX_MAX_BDL_ENTRIES) + return -EINVAL; + + addr = snd_pcm_sgbuf_get_addr(sgbuf, ofs); + /* program the address field of the BDL entry */ + bdl[0] = cpu_to_le32((u32)addr); + bdl[1] = cpu_to_le32(upper_32_bits(addr)); + /* program the size field of the BDL entry */ + chunk = PAGE_SIZE - (ofs % PAGE_SIZE); + if (size < chunk) + chunk = size; + bdl[2] = cpu_to_le32(chunk); + /* program the IOC to enable interrupt + * only when the whole fragment is processed + */ + size -= chunk; + bdl[3] = (size || !with_ioc) ? 0 : cpu_to_le32(0x01); + bdl += 4; + azx_dev->frags++; + ofs += chunk; + } + *bdlp = bdl; + return ofs; +} + +/* * set up BDL entries */ -static int azx_setup_periods(struct snd_pcm_substream *substream, +static int azx_setup_periods(struct azx *chip, + struct snd_pcm_substream *substream, struct azx_dev *azx_dev) { - struct snd_sg_buf *sgbuf = snd_pcm_substream_sgbuf(substream); u32 *bdl; int i, ofs, periods, period_bytes; + int pos_adj; /* reset BDL address */ azx_sd_writel(azx_dev, SD_BDLPL, 0); azx_sd_writel(azx_dev, SD_BDLPU, 0); period_bytes = snd_pcm_lib_period_bytes(substream); + azx_dev->period_bytes = period_bytes; periods = azx_dev->bufsize / period_bytes; /* program the initial BDL entries */ bdl = (u32 *)azx_dev->bdl.area; ofs = 0; azx_dev->frags = 0; - for (i = 0; i < periods; i++) { - int size, rest; - if (i >= AZX_MAX_BDL_ENTRIES) { - snd_printk(KERN_ERR "Too many BDL entries: " - "buffer=%d, period=%d\n", - azx_dev->bufsize, period_bytes); - /* reset */ - azx_sd_writel(azx_dev, SD_BDLPL, 0); - azx_sd_writel(azx_dev, SD_BDLPU, 0); - return -EINVAL; + azx_dev->irq_ignore = 0; + pos_adj = bdl_pos_adj[chip->dev_index]; + if (pos_adj > 0) { + struct snd_pcm_runtime *runtime = substream->runtime; + pos_adj = (pos_adj * runtime->rate + 47999) / 48000; + if (!pos_adj) + pos_adj = 1; + pos_adj = frames_to_bytes(runtime, pos_adj); + if (pos_adj >= period_bytes) { + snd_printk(KERN_WARNING "Too big adjustment %d\n", + bdl_pos_adj[chip->dev_index]); + pos_adj = 0; + } else { + ofs = setup_bdle(substream, azx_dev, + &bdl, ofs, pos_adj, 1); + if (ofs < 0) + goto error; + azx_dev->irq_ignore = 1; } - rest = period_bytes; - do { - dma_addr_t addr = snd_pcm_sgbuf_get_addr(sgbuf, ofs); - /* program the address field of the BDL entry */ - bdl[0] = cpu_to_le32((u32)addr); - bdl[1] = cpu_to_le32(upper_32bit(addr)); - /* program the size field of the BDL entry */ - size = PAGE_SIZE - (ofs % PAGE_SIZE); - if (rest < size) - size = rest; - bdl[2] = cpu_to_le32(size); - /* program the IOC to enable interrupt - * only when the whole fragment is processed - */ - rest -= size; - bdl[3] = rest ? 0 : cpu_to_le32(0x01); - bdl += 4; - azx_dev->frags++; - ofs += size; - } while (rest > 0); + } else + pos_adj = 0; + for (i = 0; i < periods; i++) { + if (i == periods - 1 && pos_adj) + ofs = setup_bdle(substream, azx_dev, &bdl, ofs, + period_bytes - pos_adj, 0); + else + ofs = setup_bdle(substream, azx_dev, &bdl, ofs, + period_bytes, 1); + if (ofs < 0) + goto error; } return 0; + + error: + snd_printk(KERN_ERR "Too many BDL entries: buffer=%d, period=%d\n", + azx_dev->bufsize, period_bytes); + /* reset */ + azx_sd_writel(azx_dev, SD_BDLPL, 0); + azx_sd_writel(azx_dev, SD_BDLPU, 0); + return -EINVAL; } /* @@ -1062,7 +1133,7 @@ static int azx_setup_controller(struct azx *chip, struct azx_dev *azx_dev) /* lower BDL address */ azx_sd_writel(azx_dev, SD_BDLPL, (u32)azx_dev->bdl.addr); /* upper BDL address */ - azx_sd_writel(azx_dev, SD_BDLPU, upper_32bit(azx_dev->bdl.addr)); + azx_sd_writel(azx_dev, SD_BDLPU, upper_32_bits(azx_dev->bdl.addr)); /* enable the position buffer */ if (chip->position_fix == POS_FIX_POSBUF || @@ -1085,7 +1156,7 @@ static int azx_setup_controller(struct azx *chip, struct azx_dev *azx_dev) */ static unsigned int azx_max_codecs[] __devinitdata = { - [AZX_DRIVER_ICH] = 3, + [AZX_DRIVER_ICH] = 4, /* Some ICH9 boards use SD3 */ [AZX_DRIVER_SCH] = 3, [AZX_DRIVER_ATI] = 4, [AZX_DRIVER_ATIHDMI] = 4, @@ -1093,6 +1164,7 @@ static unsigned int azx_max_codecs[] __devinitdata = { [AZX_DRIVER_SIS] = 3, /* FIXME: correct? */ [AZX_DRIVER_ULI] = 3, /* FIXME: correct? */ [AZX_DRIVER_NVIDIA] = 3, /* FIXME: correct? */ + [AZX_DRIVER_TERA] = 1, }; static int __devinit azx_codec_create(struct azx *chip, const char *model, @@ -1316,7 +1388,7 @@ static int azx_pcm_prepare(struct snd_pcm_substream *substream) snd_printdd("azx_pcm_prepare: bufsize=0x%x, format=0x%x\n", azx_dev->bufsize, azx_dev->format_val); - if (azx_setup_periods(substream, azx_dev) < 0) + if (azx_setup_periods(chip, substream, azx_dev) < 0) return -EINVAL; azx_setup_controller(chip, azx_dev); if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) @@ -1421,35 +1493,113 @@ static int azx_pcm_trigger(struct snd_pcm_substream *substream, int cmd) return 0; } -static snd_pcm_uframes_t azx_pcm_pointer(struct snd_pcm_substream *substream) +static unsigned int azx_get_position(struct azx *chip, + struct azx_dev *azx_dev) { - struct azx_pcm *apcm = snd_pcm_substream_chip(substream); - struct azx *chip = apcm->chip; - struct azx_dev *azx_dev = get_azx_dev(substream); unsigned int pos; if (chip->position_fix == POS_FIX_POSBUF || chip->position_fix == POS_FIX_AUTO) { /* use the position buffer */ pos = le32_to_cpu(*azx_dev->posbuf); - if (chip->position_fix == POS_FIX_AUTO && - azx_dev->period_intr == 1 && !pos) { - printk(KERN_WARNING - "hda-intel: Invalid position buffer, " - "using LPIB read method instead.\n"); - chip->position_fix = POS_FIX_NONE; - goto read_lpib; - } } else { - read_lpib: /* read LPIB */ pos = azx_sd_readl(azx_dev, SD_LPIB); - if (chip->position_fix == POS_FIX_FIFO) - pos += azx_dev->fifo_size; } if (pos >= azx_dev->bufsize) pos = 0; - return bytes_to_frames(substream->runtime, pos); + return pos; +} + +static snd_pcm_uframes_t azx_pcm_pointer(struct snd_pcm_substream *substream) +{ + struct azx_pcm *apcm = snd_pcm_substream_chip(substream); + struct azx *chip = apcm->chip; + struct azx_dev *azx_dev = get_azx_dev(substream); + return bytes_to_frames(substream->runtime, + azx_get_position(chip, azx_dev)); +} + +/* + * Check whether the current DMA position is acceptable for updating + * periods. Returns non-zero if it's OK. + * + * Many HD-audio controllers appear pretty inaccurate about + * the update-IRQ timing. The IRQ is issued before actually the + * data is processed. So, we need to process it afterwords in a + * workqueue. + */ +static int azx_position_ok(struct azx *chip, struct azx_dev *azx_dev) +{ + unsigned int pos; + + pos = azx_get_position(chip, azx_dev); + if (chip->position_fix == POS_FIX_AUTO) { + if (!pos) { + printk(KERN_WARNING + "hda-intel: Invalid position buffer, " + "using LPIB read method instead.\n"); + chip->position_fix = POS_FIX_LPIB; + pos = azx_get_position(chip, azx_dev); + } else + chip->position_fix = POS_FIX_POSBUF; + } + + if (pos % azx_dev->period_bytes > azx_dev->period_bytes / 2) + return 0; /* NG - it's below the period boundary */ + return 1; /* OK, it's fine */ +} + +/* + * The work for pending PCM period updates. + */ +static void azx_irq_pending_work(struct work_struct *work) +{ + struct azx *chip = container_of(work, struct azx, irq_pending_work); + int i, pending; + + if (!chip->irq_pending_warned) { + printk(KERN_WARNING + "hda-intel: IRQ timing workaround is activated " + "for card #%d. Suggest a bigger bdl_pos_adj.\n", + chip->card->number); + chip->irq_pending_warned = 1; + } + + for (;;) { + pending = 0; + spin_lock_irq(&chip->reg_lock); + for (i = 0; i < chip->num_streams; i++) { + struct azx_dev *azx_dev = &chip->azx_dev[i]; + if (!azx_dev->irq_pending || + !azx_dev->substream || + !azx_dev->running) + continue; + if (azx_position_ok(chip, azx_dev)) { + azx_dev->irq_pending = 0; + spin_unlock(&chip->reg_lock); + snd_pcm_period_elapsed(azx_dev->substream); + spin_lock(&chip->reg_lock); + } else + pending++; + } + spin_unlock_irq(&chip->reg_lock); + if (!pending) + return; + cond_resched(); + } +} + +/* clear irq_pending flags and assure no on-going workq */ +static void azx_clear_irq_pending(struct azx *chip) +{ + int i; + + spin_lock_irq(&chip->reg_lock); + for (i = 0; i < chip->num_streams; i++) + chip->azx_dev[i].irq_pending = 0; + spin_unlock_irq(&chip->reg_lock); + flush_scheduled_work(); } static struct snd_pcm_ops azx_pcm_ops = { @@ -1676,6 +1826,7 @@ static int azx_suspend(struct pci_dev *pci, pm_message_t state) int i; snd_power_change_state(card, SNDRV_CTL_POWER_D3hot); + azx_clear_irq_pending(chip); for (i = 0; i < AZX_MAX_PCMS; i++) snd_pcm_suspend_all(chip->pcm[i]); if (chip->initialized) @@ -1732,6 +1883,7 @@ static int azx_free(struct azx *chip) int i; if (chip->initialized) { + azx_clear_irq_pending(chip); for (i = 0; i < chip->num_streams; i++) azx_stream_stop(chip, &chip->azx_dev[i]); azx_stop_chip(chip); @@ -1770,9 +1922,9 @@ static int azx_dev_free(struct snd_device *device) * white/black-listing for position_fix */ static struct snd_pci_quirk position_fix_list[] __devinitdata = { - SND_PCI_QUIRK(0x1028, 0x01cc, "Dell D820", POS_FIX_NONE), - SND_PCI_QUIRK(0x1028, 0x01de, "Dell Precision 390", POS_FIX_NONE), - SND_PCI_QUIRK(0x1043, 0x813d, "ASUS P5AD2", POS_FIX_NONE), + SND_PCI_QUIRK(0x1028, 0x01cc, "Dell D820", POS_FIX_LPIB), + SND_PCI_QUIRK(0x1028, 0x01de, "Dell Precision 390", POS_FIX_LPIB), + SND_PCI_QUIRK(0x1043, 0x813d, "ASUS P5AD2", POS_FIX_LPIB), {} }; @@ -1857,12 +2009,25 @@ static int __devinit azx_create(struct snd_card *card, struct pci_dev *pci, chip->irq = -1; chip->driver_type = driver_type; chip->msi = enable_msi; + chip->dev_index = dev; + INIT_WORK(&chip->irq_pending_work, azx_irq_pending_work); chip->position_fix = check_position_fix(chip, position_fix[dev]); check_probe_mask(chip, dev); chip->single_cmd = single_cmd; + if (bdl_pos_adj[dev] < 0) { + switch (chip->driver_type) { + case AZX_DRIVER_ICH: + bdl_pos_adj[dev] = 1; + break; + default: + bdl_pos_adj[dev] = 32; + break; + } + } + #if BITS_PER_LONG != 64 /* Fix up base address on ULI M5461 */ if (chip->driver_type == AZX_DRIVER_ULI) { @@ -2089,6 +2254,7 @@ static struct pci_device_id azx_ids[] = { { PCI_DEVICE(0x8086, 0x27d8), .driver_data = AZX_DRIVER_ICH }, { PCI_DEVICE(0x8086, 0x269a), .driver_data = AZX_DRIVER_ICH }, { PCI_DEVICE(0x8086, 0x284b), .driver_data = AZX_DRIVER_ICH }, + { PCI_DEVICE(0x8086, 0x2911), .driver_data = AZX_DRIVER_ICH }, { PCI_DEVICE(0x8086, 0x293e), .driver_data = AZX_DRIVER_ICH }, { PCI_DEVICE(0x8086, 0x293f), .driver_data = AZX_DRIVER_ICH }, { PCI_DEVICE(0x8086, 0x3a3e), .driver_data = AZX_DRIVER_ICH }, @@ -2141,6 +2307,8 @@ static struct pci_device_id azx_ids[] = { { PCI_DEVICE(0x10de, 0x0bd5), .driver_data = AZX_DRIVER_NVIDIA }, { PCI_DEVICE(0x10de, 0x0bd6), .driver_data = AZX_DRIVER_NVIDIA }, { PCI_DEVICE(0x10de, 0x0bd7), .driver_data = AZX_DRIVER_NVIDIA }, + /* Teradici */ + { PCI_DEVICE(0x6549, 0x1200), .driver_data = AZX_DRIVER_TERA }, { 0, } }; MODULE_DEVICE_TABLE(pci, azx_ids); diff --git a/sound/pci/hda/hda_proc.c b/sound/pci/hda/hda_proc.c index 5633f77f8f3..1e5aff5c48d 100644 --- a/sound/pci/hda/hda_proc.c +++ b/sound/pci/hda/hda_proc.c @@ -366,8 +366,6 @@ static void print_digital_conv(struct snd_info_buffer *buffer, { unsigned int digi1 = snd_hda_codec_read(codec, nid, 0, AC_VERB_GET_DIGI_CONVERT_1, 0); - unsigned int digi2 = snd_hda_codec_read(codec, nid, 0, - AC_VERB_GET_DIGI_CONVERT_2, 0); snd_iprintf(buffer, " Digital:"); if (digi1 & AC_DIG1_ENABLE) snd_iprintf(buffer, " Enabled"); @@ -386,7 +384,8 @@ static void print_digital_conv(struct snd_info_buffer *buffer, if (digi1 & AC_DIG1_LEVEL) snd_iprintf(buffer, " GenLevel"); snd_iprintf(buffer, "\n"); - snd_iprintf(buffer, " Digital category: 0x%x\n", digi2 & AC_DIG2_CC); + snd_iprintf(buffer, " Digital category: 0x%x\n", + (digi1 >> 8) & AC_DIG2_CC); } static const char *get_pwr_state(u32 state) diff --git a/sound/pci/hda/patch_analog.c b/sound/pci/hda/patch_analog.c index a99e86d7427..e8003d99f0b 100644 --- a/sound/pci/hda/patch_analog.c +++ b/sound/pci/hda/patch_analog.c @@ -23,7 +23,6 @@ #include <linux/delay.h> #include <linux/slab.h> #include <linux/pci.h> -#include <linux/mutex.h> #include <sound/core.h> #include "hda_codec.h" @@ -64,7 +63,6 @@ struct ad198x_spec { /* PCM information */ struct hda_pcm pcm_rec[3]; /* used in alc_build_pcms() */ - struct mutex amp_mutex; /* PCM volume/mute control mutex */ unsigned int spdif_route; /* dynamic controls, init_verbs and input_mux */ @@ -1618,6 +1616,7 @@ static const char *ad1981_models[AD1981_MODELS] = { static struct snd_pci_quirk ad1981_cfg_tbl[] = { SND_PCI_QUIRK(0x1014, 0x0597, "Lenovo Z60", AD1981_THINKPAD), + SND_PCI_QUIRK(0x1014, 0x05b7, "Lenovo Z60m", AD1981_THINKPAD), /* All HP models */ SND_PCI_QUIRK(0x103c, 0, "HP nx", AD1981_HP), SND_PCI_QUIRK(0x1179, 0x0001, "Toshiba U205", AD1981_TOSHIBA), @@ -2623,7 +2622,7 @@ static int ad1988_auto_create_extra_out(struct hda_codec *codec, hda_nid_t pin, { struct ad198x_spec *spec = codec->spec; hda_nid_t nid; - int idx, err; + int i, idx, err; char name[32]; if (! pin) @@ -2631,16 +2630,26 @@ static int ad1988_auto_create_extra_out(struct hda_codec *codec, hda_nid_t pin, idx = ad1988_pin_idx(pin); nid = ad1988_idx_to_dac(codec, idx); - /* specify the DAC as the extra output */ - if (! spec->multiout.hp_nid) - spec->multiout.hp_nid = nid; - else - spec->multiout.extra_out_nid[0] = nid; - /* control HP volume/switch on the output mixer amp */ - sprintf(name, "%s Playback Volume", pfx); - if ((err = add_control(spec, AD_CTL_WIDGET_VOL, name, - HDA_COMPOSE_AMP_VAL(nid, 3, 0, HDA_OUTPUT))) < 0) - return err; + /* check whether the corresponding DAC was already taken */ + for (i = 0; i < spec->autocfg.line_outs; i++) { + hda_nid_t pin = spec->autocfg.line_out_pins[i]; + hda_nid_t dac = ad1988_idx_to_dac(codec, ad1988_pin_idx(pin)); + if (dac == nid) + break; + } + if (i >= spec->autocfg.line_outs) { + /* specify the DAC as the extra output */ + if (!spec->multiout.hp_nid) + spec->multiout.hp_nid = nid; + else + spec->multiout.extra_out_nid[0] = nid; + /* control HP volume/switch on the output mixer amp */ + sprintf(name, "%s Playback Volume", pfx); + err = add_control(spec, AD_CTL_WIDGET_VOL, name, + HDA_COMPOSE_AMP_VAL(nid, 3, 0, HDA_OUTPUT)); + if (err < 0) + return err; + } nid = ad1988_mixer_nids[idx]; sprintf(name, "%s Playback Switch", pfx); if ((err = add_control(spec, AD_CTL_BIND_MUTE, name, @@ -3177,7 +3186,6 @@ static int patch_ad1884(struct hda_codec *codec) if (spec == NULL) return -ENOMEM; - mutex_init(&spec->amp_mutex); codec->spec = spec; spec->multiout.max_channels = 2; @@ -3847,7 +3855,6 @@ static int patch_ad1884a(struct hda_codec *codec) if (spec == NULL) return -ENOMEM; - mutex_init(&spec->amp_mutex); codec->spec = spec; spec->multiout.max_channels = 2; @@ -4152,7 +4159,6 @@ static int patch_ad1882(struct hda_codec *codec) if (spec == NULL) return -ENOMEM; - mutex_init(&spec->amp_mutex); codec->spec = spec; spec->multiout.max_channels = 6; diff --git a/sound/pci/hda/patch_conexant.c b/sound/pci/hda/patch_conexant.c index 36fd8526003..7c1eb23f0ce 100644 --- a/sound/pci/hda/patch_conexant.c +++ b/sound/pci/hda/patch_conexant.c @@ -82,7 +82,6 @@ struct conexant_spec { /* PCM information */ struct hda_pcm pcm_rec[2]; /* used in build_pcms() */ - struct mutex amp_mutex; /* PCM volume/mute control mutex */ unsigned int spdif_route; /* dynamic controls, init_verbs and input_mux */ @@ -687,7 +686,7 @@ static struct snd_kcontrol_new cxt5045_mixers_hp530[] = { static struct hda_verb cxt5045_init_verbs[] = { /* Line in, Mic */ - {0x12, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN }, + {0x12, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN|AC_PINCTL_VREF_80 }, {0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN|AC_PINCTL_VREF_80 }, /* HP, Amp */ {0x10, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, @@ -907,10 +906,12 @@ static struct snd_pci_quirk cxt5045_cfg_tbl[] = { SND_PCI_QUIRK(0x103c, 0x30cf, "HP DV9533EG", CXT5045_LAPTOP_HPSENSE), SND_PCI_QUIRK(0x103c, 0x30d5, "HP 530", CXT5045_LAPTOP_HP530), SND_PCI_QUIRK(0x103c, 0x30d9, "HP Spartan", CXT5045_LAPTOP_HPSENSE), + SND_PCI_QUIRK(0x1179, 0xff31, "Toshiba P105", CXT5045_LAPTOP_MICSENSE), SND_PCI_QUIRK(0x152d, 0x0753, "Benq R55E", CXT5045_BENQ), SND_PCI_QUIRK(0x1734, 0x10ad, "Fujitsu Si1520", CXT5045_LAPTOP_MICSENSE), SND_PCI_QUIRK(0x1734, 0x10cb, "Fujitsu Si3515", CXT5045_LAPTOP_HPMICSENSE), - SND_PCI_QUIRK(0x1734, 0x110e, "Fujitsu V5505", CXT5045_LAPTOP_HPSENSE), + SND_PCI_QUIRK(0x1734, 0x110e, "Fujitsu V5505", + CXT5045_LAPTOP_HPMICSENSE), SND_PCI_QUIRK(0x1509, 0x1e40, "FIC", CXT5045_LAPTOP_HPMICSENSE), SND_PCI_QUIRK(0x1509, 0x2f05, "FIC", CXT5045_LAPTOP_HPMICSENSE), SND_PCI_QUIRK(0x1509, 0x2f06, "FIC", CXT5045_LAPTOP_HPMICSENSE), @@ -928,7 +929,6 @@ static int patch_cxt5045(struct hda_codec *codec) spec = kzalloc(sizeof(*spec), GFP_KERNEL); if (!spec) return -ENOMEM; - mutex_init(&spec->amp_mutex); codec->spec = spec; spec->multiout.max_channels = 2; @@ -963,6 +963,7 @@ static int patch_cxt5045(struct hda_codec *codec) codec->patch_ops.init = cxt5045_init; break; case CXT5045_LAPTOP_MICSENSE: + codec->patch_ops.unsol_event = cxt5045_hp_unsol_event; spec->input_mux = &cxt5045_capture_source; spec->num_init_verbs = 2; spec->init_verbs[1] = cxt5045_mic_sense_init_verbs; @@ -1007,15 +1008,19 @@ static int patch_cxt5045(struct hda_codec *codec) #endif } - /* - * Fix max PCM level to 0 dB - * (originall it has 0x2b steps with 0dB offset 0x14) - */ - snd_hda_override_amp_caps(codec, 0x17, HDA_INPUT, - (0x14 << AC_AMPCAP_OFFSET_SHIFT) | - (0x14 << AC_AMPCAP_NUM_STEPS_SHIFT) | - (0x05 << AC_AMPCAP_STEP_SIZE_SHIFT) | - (1 << AC_AMPCAP_MUTE_SHIFT)); + switch (codec->subsystem_id >> 16) { + case 0x103c: + /* HP laptop has a really bad sound over 0dB on NID 0x17. + * Fix max PCM level to 0 dB + * (originall it has 0x2b steps with 0dB offset 0x14) + */ + snd_hda_override_amp_caps(codec, 0x17, HDA_INPUT, + (0x14 << AC_AMPCAP_OFFSET_SHIFT) | + (0x14 << AC_AMPCAP_NUM_STEPS_SHIFT) | + (0x05 << AC_AMPCAP_STEP_SIZE_SHIFT) | + (1 << AC_AMPCAP_MUTE_SHIFT)); + break; + } return 0; } @@ -1477,7 +1482,6 @@ static int patch_cxt5047(struct hda_codec *codec) spec = kzalloc(sizeof(*spec), GFP_KERNEL); if (!spec) return -ENOMEM; - mutex_init(&spec->amp_mutex); codec->spec = spec; spec->multiout.max_channels = 2; @@ -1736,7 +1740,6 @@ static int patch_cxt5051(struct hda_codec *codec) spec = kzalloc(sizeof(*spec), GFP_KERNEL); if (!spec) return -ENOMEM; - mutex_init(&spec->amp_mutex); codec->spec = spec; codec->patch_ops = conexant_patch_ops; diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index b0a2a262ece..2807bc840d2 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -163,6 +163,10 @@ enum { ALC662_LENOVO_101E, ALC662_ASUS_EEEPC_P701, ALC662_ASUS_EEEPC_EP20, + ALC663_ASUS_M51VA, + ALC663_ASUS_G71V, + ALC663_ASUS_H13, + ALC663_ASUS_G50V, ALC662_AUTO, ALC662_MODEL_LAST, }; @@ -205,6 +209,7 @@ enum { ALC883_MITAC, ALC883_CLEVO_M720, ALC883_FUJITSU_PI2515, + ALC883_3ST_6ch_INTEL, ALC883_AUTO, ALC883_MODEL_LAST, }; @@ -280,6 +285,10 @@ struct alc_spec { #ifdef CONFIG_SND_HDA_POWER_SAVE struct hda_loopback_check loopback; #endif + + /* for PLL fix */ + hda_nid_t pll_nid; + unsigned int pll_coef_idx, pll_coef_bit; }; /* @@ -747,6 +756,38 @@ static struct hda_verb alc_gpio3_init_verbs[] = { { } }; +/* + * Fix hardware PLL issue + * On some codecs, the analog PLL gating control must be off while + * the default value is 1. + */ +static void alc_fix_pll(struct hda_codec *codec) +{ + struct alc_spec *spec = codec->spec; + unsigned int val; + + if (!spec->pll_nid) + return; + snd_hda_codec_write(codec, spec->pll_nid, 0, AC_VERB_SET_COEF_INDEX, + spec->pll_coef_idx); + val = snd_hda_codec_read(codec, spec->pll_nid, 0, + AC_VERB_GET_PROC_COEF, 0); + snd_hda_codec_write(codec, spec->pll_nid, 0, AC_VERB_SET_COEF_INDEX, + spec->pll_coef_idx); + snd_hda_codec_write(codec, spec->pll_nid, 0, AC_VERB_SET_PROC_COEF, + val & ~(1 << spec->pll_coef_bit)); +} + +static void alc_fix_pll_init(struct hda_codec *codec, hda_nid_t nid, + unsigned int coef_idx, unsigned int coef_bit) +{ + struct alc_spec *spec = codec->spec; + spec->pll_nid = nid; + spec->pll_coef_idx = coef_idx; + spec->pll_coef_bit = coef_bit; + alc_fix_pll(codec); +} + static void alc_sku_automute(struct hda_codec *codec) { struct alc_spec *spec = codec->spec; @@ -776,6 +817,24 @@ static void alc_sku_unsol_event(struct hda_codec *codec, unsigned int res) alc_sku_automute(codec); } +/* additional initialization for ALC888 variants */ +static void alc888_coef_init(struct hda_codec *codec) +{ + unsigned int tmp; + + snd_hda_codec_write(codec, 0x20, 0, AC_VERB_SET_COEF_INDEX, 0); + tmp = snd_hda_codec_read(codec, 0x20, 0, AC_VERB_GET_PROC_COEF, 0); + snd_hda_codec_write(codec, 0x20, 0, AC_VERB_SET_COEF_INDEX, 7); + if ((tmp & 0xf0) == 2) + /* alc888S-VC */ + snd_hda_codec_read(codec, 0x20, 0, + AC_VERB_SET_PROC_COEF, 0x830); + else + /* alc888-VB */ + snd_hda_codec_read(codec, 0x20, 0, + AC_VERB_SET_PROC_COEF, 0x3030); +} + /* 32-bit subsystem ID for BIOS loading in HD Audio codec. * 31 ~ 16 : Manufacture ID * 15 ~ 8 : SKU ID @@ -851,8 +910,10 @@ do_sku: case 0x10ec0267: case 0x10ec0268: case 0x10ec0269: + case 0x10ec0660: + case 0x10ec0662: + case 0x10ec0663: case 0x10ec0862: - case 0x10ec0662: case 0x10ec0889: snd_hda_codec_write(codec, 0x14, 0, AC_VERB_SET_EAPD_BTLENABLE, 2); @@ -877,7 +938,6 @@ do_sku: case 0x10ec0882: case 0x10ec0883: case 0x10ec0885: - case 0x10ec0888: case 0x10ec0889: snd_hda_codec_write(codec, 0x20, 0, AC_VERB_SET_COEF_INDEX, 7); @@ -889,6 +949,9 @@ do_sku: AC_VERB_SET_PROC_COEF, tmp | 0x2010); break; + case 0x10ec0888: + alc888_coef_init(codec); + break; case 0x10ec0267: case 0x10ec0268: snd_hda_codec_write(codec, 0x20, 0, @@ -2373,6 +2436,8 @@ static int alc_init(struct hda_codec *codec) struct alc_spec *spec = codec->spec; unsigned int i; + alc_fix_pll(codec); + for (i = 0; i < spec->num_init_verbs; i++) snd_hda_sequence_write(codec, spec->init_verbs[i]); @@ -3009,6 +3074,7 @@ static struct snd_pci_quirk alc880_cfg_tbl[] = { SND_PCI_QUIRK(0x1695, 0x400d, "EPoX", ALC880_5ST_DIG), SND_PCI_QUIRK(0x1695, 0x4012, "EPox EP-5LDA", ALC880_5ST_DIG), SND_PCI_QUIRK(0x1734, 0x107c, "FSC F1734", ALC880_F1734), + SND_PCI_QUIRK(0x1734, 0x1094, "FSC Amilo M1451G", ALC880_FUJITSU), SND_PCI_QUIRK(0x1734, 0x10ac, "FSC", ALC880_UNIWILL), SND_PCI_QUIRK(0x1734, 0x10b0, "Fujitsu", ALC880_FUJITSU), SND_PCI_QUIRK(0x1854, 0x0018, "LG LW20", ALC880_LG_LW), @@ -5101,7 +5167,7 @@ static struct snd_pci_quirk alc260_cfg_tbl[] = { SND_PCI_QUIRK(0x103c, 0x2808, "HP d5700", ALC260_HP_3013), SND_PCI_QUIRK(0x103c, 0x280a, "HP d5750", ALC260_HP_3013), SND_PCI_QUIRK(0x103c, 0x3010, "HP", ALC260_HP_3013), - SND_PCI_QUIRK(0x103c, 0x3011, "HP", ALC260_HP), + SND_PCI_QUIRK(0x103c, 0x3011, "HP", ALC260_HP_3013), SND_PCI_QUIRK(0x103c, 0x3012, "HP", ALC260_HP_3013), SND_PCI_QUIRK(0x103c, 0x3013, "HP", ALC260_HP_3013), SND_PCI_QUIRK(0x103c, 0x3014, "HP", ALC260_HP), @@ -6127,6 +6193,7 @@ static struct snd_pci_quirk alc882_cfg_tbl[] = { SND_PCI_QUIRK(0x1043, 0x817f, "Asus P5LD2", ALC882_6ST_DIG), SND_PCI_QUIRK(0x1043, 0x81d8, "Asus P5WD", ALC882_6ST_DIG), SND_PCI_QUIRK(0x105b, 0x6668, "Foxconn", ALC882_6ST_DIG), + SND_PCI_QUIRK(0x106b, 0x00a0, "Apple iMac 24''", ALC885_IMAC24), SND_PCI_QUIRK(0x1458, 0xa002, "Gigabyte P35 DS3R", ALC882_6ST_DIG), SND_PCI_QUIRK(0x1462, 0x28fb, "Targa T8", ALC882_TARGA), /* MSI-1049 T8 */ SND_PCI_QUIRK(0x1462, 0x6668, "MSI", ALC882_6ST_DIG), @@ -6353,7 +6420,9 @@ static void alc882_auto_init_analog_input(struct hda_codec *codec) continue; vref = PIN_IN; if (1 /*i <= AUTO_PIN_FRONT_MIC*/) { - if (snd_hda_param_read(codec, nid, AC_PAR_PIN_CAP) & + unsigned int pincap; + pincap = snd_hda_param_read(codec, nid, AC_PAR_PIN_CAP); + if ((pincap >> AC_PINCAP_VREF_SHIFT) & AC_PINCAP_VREF_80) vref = PIN_VREF80; } @@ -6450,8 +6519,9 @@ static int patch_alc882(struct hda_codec *codec) case 0x106b1000: /* iMac 24 */ board_config = ALC885_IMAC24; break; - case 0x106b00a1: /* Macbook */ + case 0x106b00a1: /* Macbook (might be wrong - PCI SSID?) */ case 0x106b2c00: /* Macbook Pro rev3 */ + case 0x106b3600: /* Macbook 3.1 */ board_config = ALC885_MBP3; break; default: @@ -6485,14 +6555,20 @@ static int patch_alc882(struct hda_codec *codec) if (board_config != ALC882_AUTO) setup_preset(spec, &alc882_presets[board_config]); - spec->stream_name_analog = "ALC882 Analog"; + if (codec->vendor_id == 0x10ec0885) { + spec->stream_name_analog = "ALC885 Analog"; + spec->stream_name_digital = "ALC885 Digital"; + } else { + spec->stream_name_analog = "ALC882 Analog"; + spec->stream_name_digital = "ALC882 Digital"; + } + spec->stream_analog_playback = &alc882_pcm_analog_playback; spec->stream_analog_capture = &alc882_pcm_analog_capture; /* FIXME: setup DAC5 */ /*spec->stream_analog_alt_playback = &alc880_pcm_analog_alt_playback;*/ spec->stream_analog_alt_capture = &alc880_pcm_analog_alt_capture; - spec->stream_name_digital = "ALC882 Digital"; spec->stream_digital_playback = &alc882_pcm_digital_playback; spec->stream_digital_capture = &alc882_pcm_digital_capture; @@ -6569,6 +6645,16 @@ static struct hda_input_mux alc883_capture_source = { }, }; +static struct hda_input_mux alc883_3stack_6ch_intel = { + .num_items = 4, + .items = { + { "Mic", 0x1 }, + { "Front Mic", 0x0 }, + { "Line", 0x2 }, + { "CD", 0x4 }, + }, +}; + static struct hda_input_mux alc883_lenovo_101e_capture_source = { .num_items = 2, .items = { @@ -6650,6 +6736,48 @@ static struct hda_channel_mode alc883_3ST_6ch_modes[3] = { }; /* + * 2ch mode + */ +static struct hda_verb alc883_3ST_ch2_intel_init[] = { + { 0x19, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80 }, + { 0x19, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE }, + { 0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN }, + { 0x1a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE }, + { } /* end */ +}; + +/* + * 4ch mode + */ +static struct hda_verb alc883_3ST_ch4_intel_init[] = { + { 0x19, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80 }, + { 0x19, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE }, + { 0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT }, + { 0x1a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE }, + { 0x1a, AC_VERB_SET_CONNECT_SEL, 0x01 }, + { } /* end */ +}; + +/* + * 6ch mode + */ +static struct hda_verb alc883_3ST_ch6_intel_init[] = { + { 0x19, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT }, + { 0x19, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE }, + { 0x19, AC_VERB_SET_CONNECT_SEL, 0x02 }, + { 0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT }, + { 0x1a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE }, + { 0x1a, AC_VERB_SET_CONNECT_SEL, 0x01 }, + { } /* end */ +}; + +static struct hda_channel_mode alc883_3ST_6ch_intel_modes[3] = { + { 2, alc883_3ST_ch2_intel_init }, + { 4, alc883_3ST_ch4_intel_init }, + { 6, alc883_3ST_ch6_intel_init }, +}; + +/* * 6ch mode */ static struct hda_verb alc883_sixstack_ch6_init[] = { @@ -6881,15 +7009,54 @@ static struct snd_kcontrol_new alc883_3ST_6ch_mixer[] = { { } /* end */ }; +static struct snd_kcontrol_new alc883_3ST_6ch_intel_mixer[] = { + HDA_CODEC_VOLUME("Front Playback Volume", 0x0c, 0x0, HDA_OUTPUT), + HDA_BIND_MUTE("Front Playback Switch", 0x0c, 2, HDA_INPUT), + HDA_CODEC_VOLUME("Surround Playback Volume", 0x0d, 0x0, HDA_OUTPUT), + HDA_BIND_MUTE("Surround Playback Switch", 0x0d, 2, HDA_INPUT), + HDA_CODEC_VOLUME_MONO("Center Playback Volume", 0x0e, 1, 0x0, + HDA_OUTPUT), + HDA_CODEC_VOLUME_MONO("LFE Playback Volume", 0x0e, 2, 0x0, HDA_OUTPUT), + HDA_BIND_MUTE_MONO("Center Playback Switch", 0x0e, 1, 2, HDA_INPUT), + HDA_BIND_MUTE_MONO("LFE Playback Switch", 0x0e, 2, 2, HDA_INPUT), + HDA_CODEC_MUTE("Headphone Playback Switch", 0x15, 0x0, HDA_OUTPUT), + HDA_CODEC_VOLUME("CD Playback Volume", 0x0b, 0x04, HDA_INPUT), + HDA_CODEC_MUTE("CD Playback Switch", 0x0b, 0x04, HDA_INPUT), + HDA_CODEC_VOLUME("Line Playback Volume", 0x0b, 0x02, HDA_INPUT), + HDA_CODEC_MUTE("Line Playback Switch", 0x0b, 0x02, HDA_INPUT), + HDA_CODEC_VOLUME("Mic Playback Volume", 0x0b, 0x1, HDA_INPUT), + HDA_CODEC_VOLUME("Mic Boost", 0x19, 0, HDA_INPUT), + HDA_CODEC_MUTE("Mic Playback Switch", 0x0b, 0x1, HDA_INPUT), + HDA_CODEC_VOLUME("Front Mic Playback Volume", 0x0b, 0x0, HDA_INPUT), + HDA_CODEC_VOLUME("Front Mic Boost", 0x18, 0, HDA_INPUT), + HDA_CODEC_MUTE("Front Mic Playback Switch", 0x0b, 0x0, HDA_INPUT), + HDA_CODEC_VOLUME("PC Speaker Playback Volume", 0x0b, 0x05, HDA_INPUT), + HDA_CODEC_MUTE("PC Speaker Playback Switch", 0x0b, 0x05, HDA_INPUT), + HDA_CODEC_VOLUME("Capture Volume", 0x08, 0x0, HDA_INPUT), + HDA_CODEC_MUTE("Capture Switch", 0x08, 0x0, HDA_INPUT), + HDA_CODEC_VOLUME_IDX("Capture Volume", 1, 0x09, 0x0, HDA_INPUT), + HDA_CODEC_MUTE_IDX("Capture Switch", 1, 0x09, 0x0, HDA_INPUT), + { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + /* .name = "Capture Source", */ + .name = "Input Source", + .count = 2, + .info = alc883_mux_enum_info, + .get = alc883_mux_enum_get, + .put = alc883_mux_enum_put, + }, + { } /* end */ +}; + static struct snd_kcontrol_new alc883_fivestack_mixer[] = { HDA_CODEC_VOLUME("Front Playback Volume", 0x0c, 0x0, HDA_OUTPUT), - HDA_CODEC_MUTE("Front Playback Switch", 0x14, 0x0, HDA_OUTPUT), + HDA_BIND_MUTE("Front Playback Switch", 0x0c, 2, HDA_INPUT), HDA_CODEC_VOLUME("Surround Playback Volume", 0x0d, 0x0, HDA_OUTPUT), - HDA_CODEC_MUTE("Surround Playback Switch", 0x15, 0x0, HDA_OUTPUT), + HDA_BIND_MUTE("Surround Playback Switch", 0x0d, 2, HDA_INPUT), HDA_CODEC_VOLUME_MONO("Center Playback Volume", 0x0e, 1, 0x0, HDA_OUTPUT), HDA_CODEC_VOLUME_MONO("LFE Playback Volume", 0x0e, 2, 0x0, HDA_OUTPUT), - HDA_CODEC_MUTE_MONO("Center Playback Switch", 0x16, 1, 0x0, HDA_OUTPUT), - HDA_CODEC_MUTE_MONO("LFE Playback Switch", 0x16, 2, 0x0, HDA_OUTPUT), + HDA_BIND_MUTE_MONO("Center Playback Switch", 0x0e, 1, 2, HDA_INPUT), + HDA_BIND_MUTE_MONO("LFE Playback Switch", 0x0e, 2, 2, HDA_INPUT), HDA_CODEC_MUTE("Headphone Playback Switch", 0x1b, 0x0, HDA_OUTPUT), HDA_CODEC_VOLUME("CD Playback Volume", 0x0b, 0x04, HDA_INPUT), HDA_CODEC_MUTE("CD Playback Switch", 0x0b, 0x04, HDA_INPUT), @@ -7729,6 +7896,7 @@ static const char *alc883_models[ALC883_MODEL_LAST] = { [ALC883_MITAC] = "mitac", [ALC883_CLEVO_M720] = "clevo-m720", [ALC883_FUJITSU_PI2515] = "fujitsu-pi2515", + [ALC883_3ST_6ch_INTEL] = "3stack-6ch-intel", [ALC883_AUTO] = "auto", }; @@ -7786,6 +7954,8 @@ static struct snd_pci_quirk alc883_cfg_tbl[] = { SND_PCI_QUIRK(0x17c0, 0x4071, "MEDION MD2", ALC883_MEDION_MD2), SND_PCI_QUIRK(0x17f2, 0x5000, "Albatron KI690-AM2", ALC883_6ST_DIG), SND_PCI_QUIRK(0x1991, 0x5625, "Haier W66", ALC883_HAIER_W66), + SND_PCI_QUIRK(0x8086, 0x0001, "DG33BUC", ALC883_3ST_6ch_INTEL), + SND_PCI_QUIRK(0x8086, 0x0002, "DG33FBC", ALC883_3ST_6ch_INTEL), SND_PCI_QUIRK(0x8086, 0xd601, "D102GGC", ALC883_3ST_6ch), {} }; @@ -7824,6 +7994,18 @@ static struct alc_config_preset alc883_presets[] = { .need_dac_fix = 1, .input_mux = &alc883_capture_source, }, + [ALC883_3ST_6ch_INTEL] = { + .mixers = { alc883_3ST_6ch_intel_mixer, alc883_chmode_mixer }, + .init_verbs = { alc883_init_verbs }, + .num_dacs = ARRAY_SIZE(alc883_dac_nids), + .dac_nids = alc883_dac_nids, + .dig_out_nid = ALC883_DIGOUT_NID, + .dig_in_nid = ALC883_DIGIN_NID, + .num_channel_mode = ARRAY_SIZE(alc883_3ST_6ch_intel_modes), + .channel_mode = alc883_3ST_6ch_intel_modes, + .need_dac_fix = 1, + .input_mux = &alc883_3stack_6ch_intel, + }, [ALC883_6ST_DIG] = { .mixers = { alc883_base_mixer, alc883_chmode_mixer }, .init_verbs = { alc883_init_verbs }, @@ -8145,6 +8327,8 @@ static int patch_alc883(struct hda_codec *codec) codec->spec = spec; + alc_fix_pll_init(codec, 0x20, 0x0a, 10); + board_config = snd_hda_check_board_config(codec, ALC883_MODEL_LAST, alc883_models, alc883_cfg_tbl); @@ -8171,12 +8355,25 @@ static int patch_alc883(struct hda_codec *codec) if (board_config != ALC883_AUTO) setup_preset(spec, &alc883_presets[board_config]); - spec->stream_name_analog = "ALC883 Analog"; + switch (codec->vendor_id) { + case 0x10ec0888: + spec->stream_name_analog = "ALC888 Analog"; + spec->stream_name_digital = "ALC888 Digital"; + break; + case 0x10ec0889: + spec->stream_name_analog = "ALC889 Analog"; + spec->stream_name_digital = "ALC889 Digital"; + break; + default: + spec->stream_name_analog = "ALC883 Analog"; + spec->stream_name_digital = "ALC883 Digital"; + break; + } + spec->stream_analog_playback = &alc883_pcm_analog_playback; spec->stream_analog_capture = &alc883_pcm_analog_capture; spec->stream_analog_alt_capture = &alc883_pcm_analog_alt_capture; - spec->stream_name_digital = "ALC883 Digital"; spec->stream_digital_playback = &alc883_pcm_digital_playback; spec->stream_digital_capture = &alc883_pcm_digital_capture; @@ -8189,6 +8386,9 @@ static int patch_alc883(struct hda_codec *codec) codec->patch_ops = alc_patch_ops; if (board_config == ALC883_AUTO) spec->init_hook = alc883_auto_init; + else if (codec->vendor_id == 0x10ec0888) + spec->init_hook = alc888_coef_init; + #ifdef CONFIG_SND_HDA_POWER_SAVE if (!spec->loopback.amplist) spec->loopback.amplist = alc883_loopbacks; @@ -9522,6 +9722,8 @@ static struct snd_pci_quirk alc262_cfg_tbl[] = { SND_PCI_QUIRK(0x104d, 0x820f, "Sony ASSAMD", ALC262_SONY_ASSAMD), SND_PCI_QUIRK(0x104d, 0x900e, "Sony ASSAMD", ALC262_SONY_ASSAMD), SND_PCI_QUIRK(0x104d, 0x9015, "Sony 0x9015", ALC262_SONY_ASSAMD), + SND_PCI_QUIRK(0x1179, 0x0001, "Toshiba dynabook SS RX1", + ALC262_SONY_ASSAMD), SND_PCI_QUIRK(0x10cf, 0x1397, "Fujitsu", ALC262_FUJITSU), SND_PCI_QUIRK(0x10cf, 0x142d, "Fujitsu Lifebook E8410", ALC262_FUJITSU), SND_PCI_QUIRK(0x144d, 0xc032, "Samsung Q1 Ultra", ALC262_ULTRA), @@ -9729,6 +9931,8 @@ static int patch_alc262(struct hda_codec *codec) } #endif + alc_fix_pll_init(codec, 0x20, 0x0a, 10); + board_config = snd_hda_check_board_config(codec, ALC262_MODEL_LAST, alc262_models, alc262_cfg_tbl); @@ -10674,12 +10878,18 @@ static int patch_alc268(struct hda_codec *codec) if (board_config != ALC268_AUTO) setup_preset(spec, &alc268_presets[board_config]); - spec->stream_name_analog = "ALC268 Analog"; + if (codec->vendor_id == 0x10ec0267) { + spec->stream_name_analog = "ALC267 Analog"; + spec->stream_name_digital = "ALC267 Digital"; + } else { + spec->stream_name_analog = "ALC268 Analog"; + spec->stream_name_digital = "ALC268 Digital"; + } + spec->stream_analog_playback = &alc268_pcm_analog_playback; spec->stream_analog_capture = &alc268_pcm_analog_capture; spec->stream_analog_alt_capture = &alc268_pcm_analog_alt_capture; - spec->stream_name_digital = "ALC268 Digital"; spec->stream_digital_playback = &alc268_pcm_digital_playback; if (!query_amp_caps(codec, 0x1d, HDA_INPUT)) @@ -11033,6 +11243,8 @@ static int patch_alc269(struct hda_codec *codec) codec->spec = spec; + alc_fix_pll_init(codec, 0x20, 0x04, 15); + board_config = snd_hda_check_board_config(codec, ALC269_MODEL_LAST, alc269_models, alc269_cfg_tbl); @@ -12631,6 +12843,12 @@ static struct hda_verb alc861vd_eapd_verbs[] = { { } }; +static struct hda_verb alc660vd_eapd_verbs[] = { + {0x14, AC_VERB_SET_EAPD_BTLENABLE, 2}, + {0x15, AC_VERB_SET_EAPD_BTLENABLE, 2}, + { } +}; + static struct hda_verb alc861vd_lenovo_unsol_verbs[] = { {0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, {0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)}, @@ -12786,6 +13004,7 @@ static struct snd_pci_quirk alc861vd_cfg_tbl[] = { SND_PCI_QUIRK(0x1565, 0x820d, "Biostar NF61S SE", ALC861VD_6ST_DIG), SND_PCI_QUIRK(0x17aa, 0x2066, "Lenovo", ALC861VD_LENOVO), SND_PCI_QUIRK(0x17aa, 0x3802, "Lenovo 3000 C200", ALC861VD_LENOVO), + SND_PCI_QUIRK(0x17aa, 0x384e, "Lenovo 3000 N200", ALC861VD_LENOVO), SND_PCI_QUIRK(0x1849, 0x0862, "ASRock K8NF6G-VSTA", ALC861VD_6ST_DIG), {} }; @@ -13168,11 +13387,19 @@ static int patch_alc861vd(struct hda_codec *codec) if (board_config != ALC861VD_AUTO) setup_preset(spec, &alc861vd_presets[board_config]); - spec->stream_name_analog = "ALC861VD Analog"; + if (codec->vendor_id == 0x10ec0660) { + spec->stream_name_analog = "ALC660-VD Analog"; + spec->stream_name_digital = "ALC660-VD Digital"; + /* always turn on EAPD */ + spec->init_verbs[spec->num_init_verbs++] = alc660vd_eapd_verbs; + } else { + spec->stream_name_analog = "ALC861VD Analog"; + spec->stream_name_digital = "ALC861VD Digital"; + } + spec->stream_analog_playback = &alc861vd_pcm_analog_playback; spec->stream_analog_capture = &alc861vd_pcm_analog_capture; - spec->stream_name_digital = "ALC861VD Digital"; spec->stream_digital_playback = &alc861vd_pcm_digital_playback; spec->stream_digital_capture = &alc861vd_pcm_digital_capture; @@ -13251,6 +13478,23 @@ static struct hda_input_mux alc662_eeepc_capture_source = { }, }; +static struct hda_input_mux alc663_capture_source = { + .num_items = 3, + .items = { + { "Mic", 0x0 }, + { "Front Mic", 0x1 }, + { "Line", 0x2 }, + }, +}; + +static struct hda_input_mux alc663_m51va_capture_source = { + .num_items = 2, + .items = { + { "Ext-Mic", 0x0 }, + { "D-Mic", 0x9 }, + }, +}; + #define alc662_mux_enum_info alc_mux_enum_info #define alc662_mux_enum_get alc_mux_enum_get #define alc662_mux_enum_put alc882_mux_enum_put @@ -13431,6 +13675,44 @@ static struct snd_kcontrol_new alc662_eeepc_ep20_mixer[] = { { } /* end */ }; +static struct snd_kcontrol_new alc663_m51va_mixer[] = { + HDA_CODEC_VOLUME("Speaker Playback Volume", 0x02, 0x0, HDA_OUTPUT), + HDA_CODEC_MUTE("Speaker Playback Switch", 0x14, 0x0, HDA_OUTPUT), + HDA_CODEC_MUTE("Headphone Playback Switch", 0x21, 0x0, HDA_OUTPUT), + HDA_CODEC_VOLUME("Mic Playback Volume", 0x0b, 0x0, HDA_INPUT), + HDA_CODEC_MUTE("Mic Playback Switch", 0x0b, 0x0, HDA_INPUT), + HDA_CODEC_MUTE("DMic Playback Switch", 0x23, 0x9, HDA_INPUT), + { } /* end */ +}; + +static struct snd_kcontrol_new alc663_g71v_mixer[] = { + HDA_CODEC_VOLUME("Speaker Playback Volume", 0x02, 0x0, HDA_OUTPUT), + HDA_CODEC_MUTE("Speaker Playback Switch", 0x14, 0x0, HDA_OUTPUT), + HDA_CODEC_VOLUME("Front Playback Volume", 0x03, 0x0, HDA_OUTPUT), + HDA_CODEC_MUTE("Front Playback Switch", 0x15, 0x0, HDA_OUTPUT), + HDA_CODEC_MUTE("Headphone Playback Switch", 0x21, 0x0, HDA_OUTPUT), + + HDA_CODEC_VOLUME("Mic Playback Volume", 0x0b, 0x0, HDA_INPUT), + HDA_CODEC_MUTE("Mic Playback Switch", 0x0b, 0x0, HDA_INPUT), + HDA_CODEC_VOLUME("i-Mic Playback Volume", 0x0b, 0x1, HDA_INPUT), + HDA_CODEC_MUTE("i-Mic Playback Switch", 0x0b, 0x1, HDA_INPUT), + { } /* end */ +}; + +static struct snd_kcontrol_new alc663_g50v_mixer[] = { + HDA_CODEC_VOLUME("Speaker Playback Volume", 0x02, 0x0, HDA_OUTPUT), + HDA_CODEC_MUTE("Speaker Playback Switch", 0x14, 0x0, HDA_OUTPUT), + HDA_CODEC_MUTE("Headphone Playback Switch", 0x21, 0x0, HDA_OUTPUT), + + HDA_CODEC_VOLUME("Mic Playback Volume", 0x0b, 0x0, HDA_INPUT), + HDA_CODEC_MUTE("Mic Playback Switch", 0x0b, 0x0, HDA_INPUT), + HDA_CODEC_VOLUME("i-Mic Playback Volume", 0x0b, 0x1, HDA_INPUT), + HDA_CODEC_MUTE("i-Mic Playback Switch", 0x0b, 0x1, HDA_INPUT), + HDA_CODEC_VOLUME("Line Playback Volume", 0x0b, 0x02, HDA_INPUT), + HDA_CODEC_MUTE("Line Playback Switch", 0x0b, 0x02, HDA_INPUT), + { } /* end */ +}; + static struct snd_kcontrol_new alc662_chmode_mixer[] = { { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, @@ -13501,6 +13783,11 @@ static struct hda_verb alc662_init_verbs[] = { {0x23, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)}, {0x23, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(2)}, {0x23, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(4)}, + + /* always trun on EAPD */ + {0x14, AC_VERB_SET_EAPD_BTLENABLE, 2}, + {0x15, AC_VERB_SET_EAPD_BTLENABLE, 2}, + { } }; @@ -13571,6 +13858,43 @@ static struct hda_verb alc662_auto_init_verbs[] = { { } }; +static struct hda_verb alc663_m51va_init_verbs[] = { + {0x21, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP}, + {0x21, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, + {0x21, AC_VERB_SET_CONNECT_SEL, 0x00}, /* Headphone */ + + {0x23, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(9)}, + + {0x18, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | ALC880_MIC_EVENT}, + {0x21, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | ALC880_HP_EVENT}, + {} +}; + +static struct hda_verb alc663_g71v_init_verbs[] = { + {0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP}, + /* {0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, */ + /* {0x15, AC_VERB_SET_CONNECT_SEL, 0x01}, */ /* Headphone */ + + {0x21, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP}, + {0x21, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, + {0x21, AC_VERB_SET_CONNECT_SEL, 0x00}, /* Headphone */ + + {0x15, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN|ALC880_FRONT_EVENT}, + {0x18, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN|ALC880_MIC_EVENT}, + {0x21, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN|ALC880_HP_EVENT}, + {} +}; + +static struct hda_verb alc663_g50v_init_verbs[] = { + {0x21, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP}, + {0x21, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, + {0x21, AC_VERB_SET_CONNECT_SEL, 0x00}, /* Headphone */ + + {0x18, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | ALC880_MIC_EVENT}, + {0x21, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | ALC880_HP_EVENT}, + {} +}; + /* capture mixer elements */ static struct snd_kcontrol_new alc662_capture_mixer[] = { HDA_CODEC_VOLUME("Capture Volume", 0x09, 0x0, HDA_INPUT), @@ -13692,6 +14016,125 @@ static void alc662_eeepc_ep20_inithook(struct hda_codec *codec) alc662_eeepc_ep20_automute(codec); } +static void alc663_m51va_speaker_automute(struct hda_codec *codec) +{ + unsigned int present; + unsigned char bits; + + present = snd_hda_codec_read(codec, 0x21, 0, + AC_VERB_GET_PIN_SENSE, 0) + & AC_PINSENSE_PRESENCE; + bits = present ? HDA_AMP_MUTE : 0; + snd_hda_codec_amp_stereo(codec, 0x14, HDA_OUTPUT, 0, + HDA_AMP_MUTE, bits); +} + +static void alc663_m51va_mic_automute(struct hda_codec *codec) +{ + unsigned int present; + + present = snd_hda_codec_read(codec, 0x18, 0, + AC_VERB_GET_PIN_SENSE, 0) + & AC_PINSENSE_PRESENCE; + snd_hda_codec_write_cache(codec, 0x22, 0, AC_VERB_SET_AMP_GAIN_MUTE, + 0x7000 | (0x00 << 8) | (present ? 0 : 0x80)); + snd_hda_codec_write_cache(codec, 0x23, 0, AC_VERB_SET_AMP_GAIN_MUTE, + 0x7000 | (0x00 << 8) | (present ? 0 : 0x80)); + snd_hda_codec_write_cache(codec, 0x22, 0, AC_VERB_SET_AMP_GAIN_MUTE, + 0x7000 | (0x09 << 8) | (present ? 0x80 : 0)); + snd_hda_codec_write_cache(codec, 0x23, 0, AC_VERB_SET_AMP_GAIN_MUTE, + 0x7000 | (0x09 << 8) | (present ? 0x80 : 0)); +} + +static void alc663_m51va_unsol_event(struct hda_codec *codec, + unsigned int res) +{ + switch (res >> 26) { + case ALC880_HP_EVENT: + alc663_m51va_speaker_automute(codec); + break; + case ALC880_MIC_EVENT: + alc663_m51va_mic_automute(codec); + break; + } +} + +static void alc663_m51va_inithook(struct hda_codec *codec) +{ + alc663_m51va_speaker_automute(codec); + alc663_m51va_mic_automute(codec); +} + +static void alc663_g71v_hp_automute(struct hda_codec *codec) +{ + unsigned int present; + unsigned char bits; + + present = snd_hda_codec_read(codec, 0x21, 0, + AC_VERB_GET_PIN_SENSE, 0) + & AC_PINSENSE_PRESENCE; + bits = present ? HDA_AMP_MUTE : 0; + snd_hda_codec_amp_stereo(codec, 0x15, HDA_OUTPUT, 0, + HDA_AMP_MUTE, bits); + snd_hda_codec_amp_stereo(codec, 0x14, HDA_OUTPUT, 0, + HDA_AMP_MUTE, bits); +} + +static void alc663_g71v_front_automute(struct hda_codec *codec) +{ + unsigned int present; + unsigned char bits; + + present = snd_hda_codec_read(codec, 0x15, 0, + AC_VERB_GET_PIN_SENSE, 0) + & AC_PINSENSE_PRESENCE; + bits = present ? HDA_AMP_MUTE : 0; + snd_hda_codec_amp_stereo(codec, 0x14, HDA_OUTPUT, 0, + HDA_AMP_MUTE, bits); +} + +static void alc663_g71v_unsol_event(struct hda_codec *codec, + unsigned int res) +{ + switch (res >> 26) { + case ALC880_HP_EVENT: + alc663_g71v_hp_automute(codec); + break; + case ALC880_FRONT_EVENT: + alc663_g71v_front_automute(codec); + break; + case ALC880_MIC_EVENT: + alc662_eeepc_mic_automute(codec); + break; + } +} + +static void alc663_g71v_inithook(struct hda_codec *codec) +{ + alc663_g71v_front_automute(codec); + alc663_g71v_hp_automute(codec); + alc662_eeepc_mic_automute(codec); +} + +static void alc663_g50v_unsol_event(struct hda_codec *codec, + unsigned int res) +{ + switch (res >> 26) { + case ALC880_HP_EVENT: + alc663_m51va_speaker_automute(codec); + break; + case ALC880_MIC_EVENT: + alc662_eeepc_mic_automute(codec); + break; + } +} + +static void alc663_g50v_inithook(struct hda_codec *codec) +{ + alc663_m51va_speaker_automute(codec); + alc662_eeepc_mic_automute(codec); +} + #ifdef CONFIG_SND_HDA_POWER_SAVE #define alc662_loopbacks alc880_loopbacks #endif @@ -13714,14 +14157,24 @@ static const char *alc662_models[ALC662_MODEL_LAST] = { [ALC662_LENOVO_101E] = "lenovo-101e", [ALC662_ASUS_EEEPC_P701] = "eeepc-p701", [ALC662_ASUS_EEEPC_EP20] = "eeepc-ep20", + [ALC663_ASUS_M51VA] = "m51va", + [ALC663_ASUS_G71V] = "g71v", + [ALC663_ASUS_H13] = "h13", + [ALC663_ASUS_G50V] = "g50v", [ALC662_AUTO] = "auto", }; static struct snd_pci_quirk alc662_cfg_tbl[] = { + SND_PCI_QUIRK(0x1043, 0x11c3, "ASUS G71V", ALC663_ASUS_G71V), + SND_PCI_QUIRK(0x1043, 0x1878, "ASUS M51VA", ALC663_ASUS_M51VA), + SND_PCI_QUIRK(0x1043, 0x19a3, "ASUS M51VA", ALC663_ASUS_G50V), SND_PCI_QUIRK(0x1043, 0x8290, "ASUS P5GC-MX", ALC662_3ST_6ch_DIG), SND_PCI_QUIRK(0x1043, 0x82a1, "ASUS Eeepc", ALC662_ASUS_EEEPC_P701), SND_PCI_QUIRK(0x1043, 0x82d1, "ASUS Eeepc EP20", ALC662_ASUS_EEEPC_EP20), SND_PCI_QUIRK(0x17aa, 0x101e, "Lenovo", ALC662_LENOVO_101E), + SND_PCI_QUIRK(0x1854, 0x2000, "ASUS H13-2000", ALC663_ASUS_H13), + SND_PCI_QUIRK(0x1854, 0x2001, "ASUS H13-2001", ALC663_ASUS_H13), + SND_PCI_QUIRK(0x1854, 0x2002, "ASUS H13-2002", ALC663_ASUS_H13), {} }; @@ -13809,7 +14262,53 @@ static struct alc_config_preset alc662_presets[] = { .unsol_event = alc662_eeepc_ep20_unsol_event, .init_hook = alc662_eeepc_ep20_inithook, }, - + [ALC663_ASUS_M51VA] = { + .mixers = { alc663_m51va_mixer, alc662_capture_mixer}, + .init_verbs = { alc662_init_verbs, alc663_m51va_init_verbs }, + .num_dacs = ARRAY_SIZE(alc662_dac_nids), + .dac_nids = alc662_dac_nids, + .dig_out_nid = ALC662_DIGOUT_NID, + .num_channel_mode = ARRAY_SIZE(alc662_3ST_2ch_modes), + .channel_mode = alc662_3ST_2ch_modes, + .input_mux = &alc663_m51va_capture_source, + .unsol_event = alc663_m51va_unsol_event, + .init_hook = alc663_m51va_inithook, + }, + [ALC663_ASUS_G71V] = { + .mixers = { alc663_g71v_mixer, alc662_capture_mixer}, + .init_verbs = { alc662_init_verbs, alc663_g71v_init_verbs }, + .num_dacs = ARRAY_SIZE(alc662_dac_nids), + .dac_nids = alc662_dac_nids, + .dig_out_nid = ALC662_DIGOUT_NID, + .num_channel_mode = ARRAY_SIZE(alc662_3ST_2ch_modes), + .channel_mode = alc662_3ST_2ch_modes, + .input_mux = &alc662_eeepc_capture_source, + .unsol_event = alc663_g71v_unsol_event, + .init_hook = alc663_g71v_inithook, + }, + [ALC663_ASUS_H13] = { + .mixers = { alc663_m51va_mixer, alc662_capture_mixer}, + .init_verbs = { alc662_init_verbs, alc663_m51va_init_verbs }, + .num_dacs = ARRAY_SIZE(alc662_dac_nids), + .dac_nids = alc662_dac_nids, + .num_channel_mode = ARRAY_SIZE(alc662_3ST_2ch_modes), + .channel_mode = alc662_3ST_2ch_modes, + .input_mux = &alc663_m51va_capture_source, + .unsol_event = alc663_m51va_unsol_event, + .init_hook = alc663_m51va_inithook, + }, + [ALC663_ASUS_G50V] = { + .mixers = { alc663_g50v_mixer, alc662_capture_mixer}, + .init_verbs = { alc662_init_verbs, alc663_g50v_init_verbs }, + .num_dacs = ARRAY_SIZE(alc662_dac_nids), + .dac_nids = alc662_dac_nids, + .dig_out_nid = ALC662_DIGOUT_NID, + .num_channel_mode = ARRAY_SIZE(alc662_3ST_6ch_modes), + .channel_mode = alc662_3ST_6ch_modes, + .input_mux = &alc663_capture_source, + .unsol_event = alc663_g50v_unsol_event, + .init_hook = alc663_g50v_inithook, + }, }; @@ -14082,6 +14581,8 @@ static int patch_alc662(struct hda_codec *codec) codec->spec = spec; + alc_fix_pll_init(codec, 0x20, 0x04, 15); + board_config = snd_hda_check_board_config(codec, ALC662_MODEL_LAST, alc662_models, alc662_cfg_tbl); @@ -14108,11 +14609,17 @@ static int patch_alc662(struct hda_codec *codec) if (board_config != ALC662_AUTO) setup_preset(spec, &alc662_presets[board_config]); - spec->stream_name_analog = "ALC662 Analog"; + if (codec->vendor_id == 0x10ec0663) { + spec->stream_name_analog = "ALC663 Analog"; + spec->stream_name_digital = "ALC663 Digital"; + } else { + spec->stream_name_analog = "ALC662 Analog"; + spec->stream_name_digital = "ALC662 Digital"; + } + spec->stream_analog_playback = &alc662_pcm_analog_playback; spec->stream_analog_capture = &alc662_pcm_analog_capture; - spec->stream_name_digital = "ALC662 Digital"; spec->stream_digital_playback = &alc662_pcm_digital_playback; spec->stream_digital_capture = &alc662_pcm_digital_capture; @@ -14151,6 +14658,7 @@ struct hda_codec_preset snd_hda_preset_realtek[] = { .patch = patch_alc883 }, { .id = 0x10ec0662, .rev = 0x100101, .name = "ALC662 rev1", .patch = patch_alc662 }, + { .id = 0x10ec0663, .name = "ALC663", .patch = patch_alc662 }, { .id = 0x10ec0880, .name = "ALC880", .patch = patch_alc880 }, { .id = 0x10ec0882, .name = "ALC882", .patch = patch_alc882 }, { .id = 0x10ec0883, .name = "ALC883", .patch = patch_alc883 }, diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c index a4f44a00bae..08cb77f5188 100644 --- a/sound/pci/hda/patch_sigmatel.c +++ b/sound/pci/hda/patch_sigmatel.c @@ -636,21 +636,28 @@ static struct hda_verb stac92hd71bxx_core_init[] = { { 0x0f, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, }; +#define HD_DISABLE_PORTF 3 static struct hda_verb stac92hd71bxx_analog_core_init[] = { + /* start of config #1 */ + + /* connect port 0f to audio mixer */ + { 0x0f, AC_VERB_SET_CONNECT_SEL, 0x2}, + { 0x0f, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, /* Speaker */ + /* unmute right and left channels for node 0x0f */ + { 0x0f, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, + /* start of config #2 */ + /* set master volume and direct control */ { 0x28, AC_VERB_SET_VOLUME_KNOB_CONTROL, 0xff}, /* connect headphone jack to dac1 */ { 0x0a, AC_VERB_SET_CONNECT_SEL, 0x01}, - /* connect ports 0d and 0f to audio mixer */ + /* connect port 0d to audio mixer */ { 0x0d, AC_VERB_SET_CONNECT_SEL, 0x2}, - { 0x0f, AC_VERB_SET_CONNECT_SEL, 0x2}, - { 0x0f, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, /* Speaker */ /* unmute dac0 input in audio mixer */ { 0x17, AC_VERB_SET_AMP_GAIN_MUTE, 0x701f}, - /* unmute right and left channels for nodes 0x0a, 0xd, 0x0f */ + /* unmute right and left channels for nodes 0x0a, 0xd */ { 0x0a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, { 0x0d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, - { 0x0f, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, {} }; @@ -818,6 +825,9 @@ static struct snd_kcontrol_new stac92hd71bxx_analog_mixer[] = { HDA_CODEC_MUTE_IDX("Capture Switch", 0x1, 0x1d, 0x0, HDA_OUTPUT), HDA_CODEC_VOLUME_IDX("Capture Mux Volume", 0x1, 0x1b, 0x0, HDA_OUTPUT), + HDA_CODEC_VOLUME("PC Beep Volume", 0x17, 0x2, HDA_INPUT), + HDA_CODEC_MUTE("PC Beep Switch", 0x17, 0x2, HDA_INPUT), + HDA_CODEC_MUTE("Analog Loopback 1", 0x17, 0x3, HDA_INPUT), HDA_CODEC_MUTE("Analog Loopback 2", 0x17, 0x4, HDA_INPUT), { } /* end */ @@ -1317,13 +1327,13 @@ static unsigned int ref92hd71bxx_pin_configs[10] = { 0x90a000f0, 0x01452050, }; -static unsigned int dell_m4_1_pin_configs[13] = { +static unsigned int dell_m4_1_pin_configs[10] = { 0x0421101f, 0x04a11221, 0x40f000f0, 0x90170110, 0x23a1902e, 0x23014250, 0x40f000f0, 0x90a000f0, 0x40f000f0, 0x4f0000f0, }; -static unsigned int dell_m4_2_pin_configs[13] = { +static unsigned int dell_m4_2_pin_configs[10] = { 0x0421101f, 0x04a11221, 0x90a70330, 0x90170110, 0x23a1902e, 0x23014250, 0x40f000f0, 0x40f000f0, 0x40f000f0, 0x044413b0, @@ -1754,12 +1764,8 @@ static struct snd_pci_quirk stac9205_cfg_tbl[] = { "unknown Dell", STAC_9205_DELL_M42), SND_PCI_QUIRK(PCI_VENDOR_ID_DELL, 0x01f8, "Dell Precision", STAC_9205_DELL_M43), - SND_PCI_QUIRK(PCI_VENDOR_ID_DELL, 0x021c, - "Dell Precision", STAC_9205_DELL_M43), SND_PCI_QUIRK(PCI_VENDOR_ID_DELL, 0x01f9, "Dell Precision", STAC_9205_DELL_M43), - SND_PCI_QUIRK(PCI_VENDOR_ID_DELL, 0x021b, - "Dell Precision", STAC_9205_DELL_M43), SND_PCI_QUIRK(PCI_VENDOR_ID_DELL, 0x01fa, "Dell Precision", STAC_9205_DELL_M43), SND_PCI_QUIRK(PCI_VENDOR_ID_DELL, 0x01fc, @@ -1770,18 +1776,14 @@ static struct snd_pci_quirk stac9205_cfg_tbl[] = { "Dell Precision", STAC_9205_DELL_M43), SND_PCI_QUIRK(PCI_VENDOR_ID_DELL, 0x01ff, "Dell Precision M4300", STAC_9205_DELL_M43), - SND_PCI_QUIRK(PCI_VENDOR_ID_DELL, 0x0206, - "Dell Precision", STAC_9205_DELL_M43), - SND_PCI_QUIRK(PCI_VENDOR_ID_DELL, 0x01f1, - "Dell Inspiron", STAC_9205_DELL_M44), - SND_PCI_QUIRK(PCI_VENDOR_ID_DELL, 0x01f2, - "Dell Inspiron", STAC_9205_DELL_M44), - SND_PCI_QUIRK(PCI_VENDOR_ID_DELL, 0x01fc, - "Dell Inspiron", STAC_9205_DELL_M44), - SND_PCI_QUIRK(PCI_VENDOR_ID_DELL, 0x01fd, - "Dell Inspiron", STAC_9205_DELL_M44), SND_PCI_QUIRK(PCI_VENDOR_ID_DELL, 0x0204, "unknown Dell", STAC_9205_DELL_M42), + SND_PCI_QUIRK(PCI_VENDOR_ID_DELL, 0x0206, + "Dell Precision", STAC_9205_DELL_M43), + SND_PCI_QUIRK(PCI_VENDOR_ID_DELL, 0x021b, + "Dell Precision", STAC_9205_DELL_M43), + SND_PCI_QUIRK(PCI_VENDOR_ID_DELL, 0x021c, + "Dell Precision", STAC_9205_DELL_M43), SND_PCI_QUIRK(PCI_VENDOR_ID_DELL, 0x021f, "Dell Inspiron", STAC_9205_DELL_M44), SND_PCI_QUIRK(PCI_VENDOR_ID_DELL, 0x0228, @@ -3103,13 +3105,16 @@ static int stac92xx_init(struct hda_codec *codec) 0, AC_VERB_GET_PIN_WIDGET_CONTROL, 0); int def_conf = snd_hda_codec_read(codec, spec->pwr_nids[i], 0, AC_VERB_GET_CONFIG_DEFAULT, 0); + def_conf = get_defcfg_connect(def_conf); /* outputs are only ports capable of power management * any attempts on powering down a input port cause the * referenced VREF to act quirky. */ if (pinctl & AC_PINCTL_IN_EN) continue; - if (get_defcfg_connect(def_conf) != AC_JACK_PORT_FIXED) + /* skip any ports that don't have jacks since presence + * detection is useless */ + if (def_conf && def_conf != AC_JACK_PORT_FIXED) continue; enable_pin_detect(codec, spec->pwr_nids[i], event | i); codec->patch_ops.unsol_event(codec, (event | i) << 26); @@ -3614,6 +3619,7 @@ static int patch_stac92hd71bxx(struct hda_codec *codec) codec->spec = spec; spec->num_pins = ARRAY_SIZE(stac92hd71bxx_pin_nids); + spec->num_pwrs = ARRAY_SIZE(stac92hd71bxx_pwr_nids); spec->pin_nids = stac92hd71bxx_pin_nids; spec->board_config = snd_hda_check_board_config(codec, STAC_92HD71BXX_MODELS, @@ -3642,6 +3648,19 @@ again: spec->mixer = stac92hd71bxx_mixer; spec->init = stac92hd71bxx_core_init; break; + case 0x111d7608: /* 5 Port with Analog Mixer */ + /* no output amps */ + spec->num_pwrs = 0; + spec->mixer = stac92hd71bxx_analog_mixer; + + /* disable VSW */ + spec->init = &stac92hd71bxx_analog_core_init[HD_DISABLE_PORTF]; + stac92xx_set_config_reg(codec, 0xf, 0x40f000f0); + break; + case 0x111d7603: /* 6 Port with Analog Mixer */ + /* no output amps */ + spec->num_pwrs = 0; + /* fallthru */ default: spec->mixer = stac92hd71bxx_analog_mixer; spec->init = stac92hd71bxx_analog_core_init; @@ -3653,22 +3672,19 @@ again: /* GPIO0 High = EAPD */ spec->gpio_mask = 0x01; spec->gpio_dir = 0x01; - spec->gpio_mask = 0x01; spec->gpio_data = 0x01; spec->mux_nids = stac92hd71bxx_mux_nids; spec->adc_nids = stac92hd71bxx_adc_nids; spec->dmic_nids = stac92hd71bxx_dmic_nids; spec->dmux_nids = stac92hd71bxx_dmux_nids; + spec->pwr_nids = stac92hd71bxx_pwr_nids; spec->num_muxes = ARRAY_SIZE(stac92hd71bxx_mux_nids); spec->num_adcs = ARRAY_SIZE(stac92hd71bxx_adc_nids); spec->num_dmics = STAC92HD71BXX_NUM_DMICS; spec->num_dmuxes = ARRAY_SIZE(stac92hd71bxx_dmux_nids); - spec->num_pwrs = ARRAY_SIZE(stac92hd71bxx_pwr_nids); - spec->pwr_nids = stac92hd71bxx_pwr_nids; - spec->multiout.num_dacs = 1; spec->multiout.hp_nid = 0x11; spec->multiout.dac_nids = stac92hd71bxx_dac_nids; @@ -4306,10 +4322,11 @@ struct hda_codec_preset snd_hda_preset_sigmatel[] = { { .id = 0x838476a5, .name = "STAC9255D", .patch = patch_stac9205 }, { .id = 0x838476a6, .name = "STAC9254", .patch = patch_stac9205 }, { .id = 0x838476a7, .name = "STAC9254D", .patch = patch_stac9205 }, + { .id = 0x111d7603, .name = "92HD75B3X5", .patch = patch_stac92hd71bxx}, + { .id = 0x111d7608, .name = "92HD75B2X5", .patch = patch_stac92hd71bxx}, { .id = 0x111d7674, .name = "92HD73D1X5", .patch = patch_stac92hd73xx }, { .id = 0x111d7675, .name = "92HD73C1X5", .patch = patch_stac92hd73xx }, { .id = 0x111d7676, .name = "92HD73E1X5", .patch = patch_stac92hd73xx }, - { .id = 0x111d7608, .name = "92HD71BXX", .patch = patch_stac92hd71bxx }, { .id = 0x111d76b0, .name = "92HD71B8X", .patch = patch_stac92hd71bxx }, { .id = 0x111d76b1, .name = "92HD71B8X", .patch = patch_stac92hd71bxx }, { .id = 0x111d76b2, .name = "92HD71B7X", .patch = patch_stac92hd71bxx }, diff --git a/sound/pci/ice1712/envy24ht.h b/sound/pci/ice1712/envy24ht.h index 43b9e3e858b..a0c5e009bb4 100644 --- a/sound/pci/ice1712/envy24ht.h +++ b/sound/pci/ice1712/envy24ht.h @@ -93,9 +93,13 @@ enum { #define VT1724_REG_MPU_TXFIFO 0x0a /*byte ro. number of bytes in TX fifo*/ #define VT1724_REG_MPU_RXFIFO 0x0b /*byte ro. number of bytes in RX fifo*/ -//are these 2 the wrong way around? they don't seem to be used yet anyway -#define VT1724_REG_MPU_CTRL 0x0c /* byte */ -#define VT1724_REG_MPU_DATA 0x0d /* byte */ +#define VT1724_REG_MPU_DATA 0x0c /* byte */ +#define VT1724_REG_MPU_CTRL 0x0d /* byte */ +#define VT1724_MPU_UART 0x01 +#define VT1724_MPU_TX_EMPTY 0x02 +#define VT1724_MPU_TX_FULL 0x04 +#define VT1724_MPU_RX_EMPTY 0x08 +#define VT1724_MPU_RX_FULL 0x10 #define VT1724_REG_MPU_FIFO_WM 0x0e /*byte set the high/low watermarks for RX/TX fifos*/ #define VT1724_MPU_RX_FIFO 0x20 //1=rx fifo watermark 0=tx fifo watermark diff --git a/sound/pci/ice1712/ice1712.h b/sound/pci/ice1712/ice1712.h index 3208901c740..762fbd7a750 100644 --- a/sound/pci/ice1712/ice1712.h +++ b/sound/pci/ice1712/ice1712.h @@ -333,6 +333,8 @@ struct snd_ice1712 { unsigned int has_spdif: 1; /* VT1720/4 - has SPDIF I/O */ unsigned int force_pdma4: 1; /* VT1720/4 - PDMA4 as non-spdif */ unsigned int force_rdma1: 1; /* VT1720/4 - RDMA1 as non-spdif */ + unsigned int midi_output: 1; /* VT1720/4: MIDI output triggered */ + unsigned int midi_input: 1; /* VT1720/4: MIDI input triggered */ unsigned int num_total_dacs; /* total DACs */ unsigned int num_total_adcs; /* total ADCs */ unsigned int cur_rate; /* current rate */ diff --git a/sound/pci/ice1712/ice1724.c b/sound/pci/ice1712/ice1724.c index 67350901772..e596d777d9d 100644 --- a/sound/pci/ice1712/ice1724.c +++ b/sound/pci/ice1712/ice1724.c @@ -32,7 +32,7 @@ #include <linux/mutex.h> #include <sound/core.h> #include <sound/info.h> -#include <sound/mpu401.h> +#include <sound/rawmidi.h> #include <sound/initval.h> #include <sound/asoundef.h> @@ -223,30 +223,153 @@ static unsigned int snd_vt1724_get_gpio_data(struct snd_ice1712 *ice) } /* - * MPU401 accessor + * MIDI */ -static unsigned char snd_vt1724_mpu401_read(struct snd_mpu401 *mpu, - unsigned long addr) + +static void vt1724_midi_clear_rx(struct snd_ice1712 *ice) +{ + unsigned int count; + + for (count = inb(ICEREG1724(ice, MPU_RXFIFO)); count > 0; --count) + inb(ICEREG1724(ice, MPU_DATA)); +} + +static inline struct snd_rawmidi_substream * +get_rawmidi_substream(struct snd_ice1712 *ice, unsigned int stream) { - /* fix status bits to the standard position */ - /* only RX_EMPTY and TX_FULL are checked */ - if (addr == MPU401C(mpu)) - return (inb(addr) & 0x0c) << 4; + return list_first_entry(&ice->rmidi[0]->streams[stream].substreams, + struct snd_rawmidi_substream, list); +} + +static void vt1724_midi_write(struct snd_ice1712 *ice) +{ + struct snd_rawmidi_substream *s; + int count, i; + u8 buffer[32]; + + s = get_rawmidi_substream(ice, SNDRV_RAWMIDI_STREAM_OUTPUT); + count = 31 - inb(ICEREG1724(ice, MPU_TXFIFO)); + if (count > 0) { + count = snd_rawmidi_transmit(s, buffer, count); + for (i = 0; i < count; ++i) + outb(buffer[i], ICEREG1724(ice, MPU_DATA)); + } +} + +static void vt1724_midi_read(struct snd_ice1712 *ice) +{ + struct snd_rawmidi_substream *s; + int count, i; + u8 buffer[32]; + + s = get_rawmidi_substream(ice, SNDRV_RAWMIDI_STREAM_INPUT); + count = inb(ICEREG1724(ice, MPU_RXFIFO)); + if (count > 0) { + count = min(count, 32); + for (i = 0; i < count; ++i) + buffer[i] = inb(ICEREG1724(ice, MPU_DATA)); + snd_rawmidi_receive(s, buffer, count); + } +} + +static void vt1724_enable_midi_irq(struct snd_rawmidi_substream *substream, + u8 flag, int enable) +{ + struct snd_ice1712 *ice = substream->rmidi->private_data; + u8 mask; + + spin_lock_irq(&ice->reg_lock); + mask = inb(ICEREG1724(ice, IRQMASK)); + if (enable) + mask &= ~flag; else - return inb(addr); + mask |= flag; + outb(mask, ICEREG1724(ice, IRQMASK)); + spin_unlock_irq(&ice->reg_lock); } -static void snd_vt1724_mpu401_write(struct snd_mpu401 *mpu, - unsigned char data, unsigned long addr) +static int vt1724_midi_output_open(struct snd_rawmidi_substream *s) { - if (addr == MPU401C(mpu)) { - if (data == MPU401_ENTER_UART) - outb(0x01, addr); - /* what else? */ - } else - outb(data, addr); + vt1724_enable_midi_irq(s, VT1724_IRQ_MPU_TX, 1); + return 0; +} + +static int vt1724_midi_output_close(struct snd_rawmidi_substream *s) +{ + vt1724_enable_midi_irq(s, VT1724_IRQ_MPU_TX, 0); + return 0; } +static void vt1724_midi_output_trigger(struct snd_rawmidi_substream *s, int up) +{ + struct snd_ice1712 *ice = s->rmidi->private_data; + unsigned long flags; + + spin_lock_irqsave(&ice->reg_lock, flags); + if (up) { + ice->midi_output = 1; + vt1724_midi_write(ice); + } else { + ice->midi_output = 0; + } + spin_unlock_irqrestore(&ice->reg_lock, flags); +} + +static void vt1724_midi_output_drain(struct snd_rawmidi_substream *s) +{ + struct snd_ice1712 *ice = s->rmidi->private_data; + unsigned long timeout; + + /* 32 bytes should be transmitted in less than about 12 ms */ + timeout = jiffies + msecs_to_jiffies(15); + do { + if (inb(ICEREG1724(ice, MPU_CTRL)) & VT1724_MPU_TX_EMPTY) + break; + schedule_timeout_uninterruptible(1); + } while (time_after(timeout, jiffies)); +} + +static struct snd_rawmidi_ops vt1724_midi_output_ops = { + .open = vt1724_midi_output_open, + .close = vt1724_midi_output_close, + .trigger = vt1724_midi_output_trigger, + .drain = vt1724_midi_output_drain, +}; + +static int vt1724_midi_input_open(struct snd_rawmidi_substream *s) +{ + vt1724_midi_clear_rx(s->rmidi->private_data); + vt1724_enable_midi_irq(s, VT1724_IRQ_MPU_RX, 1); + return 0; +} + +static int vt1724_midi_input_close(struct snd_rawmidi_substream *s) +{ + vt1724_enable_midi_irq(s, VT1724_IRQ_MPU_RX, 0); + return 0; +} + +static void vt1724_midi_input_trigger(struct snd_rawmidi_substream *s, int up) +{ + struct snd_ice1712 *ice = s->rmidi->private_data; + unsigned long flags; + + spin_lock_irqsave(&ice->reg_lock, flags); + if (up) { + ice->midi_input = 1; + vt1724_midi_read(ice); + } else { + ice->midi_input = 0; + } + spin_unlock_irqrestore(&ice->reg_lock, flags); +} + +static struct snd_rawmidi_ops vt1724_midi_input_ops = { + .open = vt1724_midi_input_open, + .close = vt1724_midi_input_close, + .trigger = vt1724_midi_input_trigger, +}; + /* * Interrupt handler @@ -278,13 +401,10 @@ static irqreturn_t snd_vt1724_interrupt(int irq, void *dev_id) #endif handled = 1; if (status & VT1724_IRQ_MPU_TX) { - if (ice->rmidi[0]) - snd_mpu401_uart_interrupt_tx(irq, - ice->rmidi[0]->private_data); - else /* disable TX to be sure */ - outb(inb(ICEREG1724(ice, IRQMASK)) | - VT1724_IRQ_MPU_TX, - ICEREG1724(ice, IRQMASK)); + spin_lock(&ice->reg_lock); + if (ice->midi_output) + vt1724_midi_write(ice); + spin_unlock(&ice->reg_lock); /* Due to mysterical reasons, MPU_TX is always * generated (and can't be cleared) when a PCM * playback is going. So let's ignore at the @@ -293,13 +413,12 @@ static irqreturn_t snd_vt1724_interrupt(int irq, void *dev_id) status_mask &= ~VT1724_IRQ_MPU_TX; } if (status & VT1724_IRQ_MPU_RX) { - if (ice->rmidi[0]) - snd_mpu401_uart_interrupt(irq, - ice->rmidi[0]->private_data); - else /* disable RX to be sure */ - outb(inb(ICEREG1724(ice, IRQMASK)) | - VT1724_IRQ_MPU_RX, - ICEREG1724(ice, IRQMASK)); + spin_lock(&ice->reg_lock); + if (ice->midi_input) + vt1724_midi_read(ice); + else + vt1724_midi_clear_rx(ice); + spin_unlock(&ice->reg_lock); } /* ack MPU irq */ outb(status, ICEREG1724(ice, IRQSTAT)); @@ -2425,28 +2544,30 @@ static int __devinit snd_vt1724_probe(struct pci_dev *pci, if (! c->no_mpu401) { if (ice->eeprom.data[ICE_EEP2_SYSCONF] & VT1724_CFG_MPU401) { - struct snd_mpu401 *mpu; - if ((err = snd_mpu401_uart_new(card, 0, MPU401_HW_ICE1712, - ICEREG1724(ice, MPU_CTRL), - (MPU401_INFO_INTEGRATED | - MPU401_INFO_NO_ACK | - MPU401_INFO_TX_IRQ), - ice->irq, 0, - &ice->rmidi[0])) < 0) { + struct snd_rawmidi *rmidi; + + err = snd_rawmidi_new(card, "MIDI", 0, 1, 1, &rmidi); + if (err < 0) { snd_card_free(card); return err; } - mpu = ice->rmidi[0]->private_data; - mpu->read = snd_vt1724_mpu401_read; - mpu->write = snd_vt1724_mpu401_write; - /* unmask MPU RX/TX irqs */ - outb(inb(ICEREG1724(ice, IRQMASK)) & - ~(VT1724_IRQ_MPU_RX | VT1724_IRQ_MPU_TX), - ICEREG1724(ice, IRQMASK)); + ice->rmidi[0] = rmidi; + rmidi->private_data = ice; + strcpy(rmidi->name, "ICE1724 MIDI"); + rmidi->info_flags = SNDRV_RAWMIDI_INFO_OUTPUT | + SNDRV_RAWMIDI_INFO_INPUT | + SNDRV_RAWMIDI_INFO_DUPLEX; + snd_rawmidi_set_ops(rmidi, SNDRV_RAWMIDI_STREAM_OUTPUT, + &vt1724_midi_output_ops); + snd_rawmidi_set_ops(rmidi, SNDRV_RAWMIDI_STREAM_INPUT, + &vt1724_midi_input_ops); + /* set watermarks */ outb(VT1724_MPU_RX_FIFO | 0x1, ICEREG1724(ice, MPU_FIFO_WM)); outb(0x1, ICEREG1724(ice, MPU_FIFO_WM)); + /* set UART mode */ + outb(VT1724_MPU_UART, ICEREG1724(ice, MPU_CTRL)); } } diff --git a/sound/pci/maestro3.c b/sound/pci/maestro3.c index a536c59fbea..f4788dee05c 100644 --- a/sound/pci/maestro3.c +++ b/sound/pci/maestro3.c @@ -2427,6 +2427,29 @@ snd_m3_amp_enable(struct snd_m3 *chip, int enable) outw(0xffff, io + GPIO_MASK); } +static void +snd_m3_hv_init(struct snd_m3 *chip) +{ + unsigned long io = chip->iobase; + u16 val = GPI_VOL_DOWN | GPI_VOL_UP; + + if (!chip->is_omnibook) + return; + + /* + * Volume buttons on some HP OmniBook laptops + * require some GPIO magic to work correctly. + */ + outw(0xffff, io + GPIO_MASK); + outw(0x0000, io + GPIO_DATA); + + outw(~val, io + GPIO_MASK); + outw(inw(io + GPIO_DIRECTION) & ~val, io + GPIO_DIRECTION); + outw(val, io + GPIO_MASK); + + outw(0xffff, io + GPIO_MASK); +} + static int snd_m3_chip_init(struct snd_m3 *chip) { @@ -2442,21 +2465,6 @@ snd_m3_chip_init(struct snd_m3 *chip) DISABLE_LEGACY); pci_write_config_word(pcidev, PCI_LEGACY_AUDIO_CTRL, w); - if (chip->is_omnibook) { - /* - * Volume buttons on some HP OmniBook laptops don't work - * correctly. This makes them work for the most part. - * - * Volume up and down buttons on the laptop side work. - * Fn+cursor_up (volme up) works. - * Fn+cursor_down (volume down) doesn't work. - * Fn+F7 (mute) works acts as volume up. - */ - outw(~(GPI_VOL_DOWN|GPI_VOL_UP), io + GPIO_MASK); - outw(inw(io + GPIO_DIRECTION) & ~(GPI_VOL_DOWN|GPI_VOL_UP), io + GPIO_DIRECTION); - outw((GPI_VOL_DOWN|GPI_VOL_UP), io + GPIO_DATA); - outw(0xffff, io + GPIO_MASK); - } pci_read_config_dword(pcidev, PCI_ALLEGRO_CONFIG, &n); n &= ~(HV_CTRL_ENABLE | REDUCED_DEBOUNCE | HV_BUTTON_FROM_GD); n |= chip->hv_config; @@ -2642,6 +2650,8 @@ static int m3_resume(struct pci_dev *pci) snd_m3_enable_ints(chip); snd_m3_amp_enable(chip, 1); + snd_m3_hv_init(chip); + snd_power_change_state(card, SNDRV_CTL_POWER_D0); return 0; } @@ -2781,6 +2791,8 @@ snd_m3_create(struct snd_card *card, struct pci_dev *pci, snd_m3_amp_enable(chip, 1); + snd_m3_hv_init(chip); + tasklet_init(&chip->hwvol_tq, snd_m3_update_hw_volume, (unsigned long)chip); if (request_irq(pci->irq, snd_m3_interrupt, IRQF_SHARED, diff --git a/sound/pci/nm256/nm256.c b/sound/pci/nm256/nm256.c index 7efb838d18a..06d13e71711 100644 --- a/sound/pci/nm256/nm256.c +++ b/sound/pci/nm256/nm256.c @@ -1302,8 +1302,8 @@ snd_nm256_mixer(struct nm256 *chip) .read = snd_nm256_ac97_read, }; - chip->ac97_regs = kcalloc(sizeof(short), - ARRAY_SIZE(nm256_ac97_init_val), GFP_KERNEL); + chip->ac97_regs = kcalloc(ARRAY_SIZE(nm256_ac97_init_val), + sizeof(short), GFP_KERNEL); if (! chip->ac97_regs) return -ENOMEM; diff --git a/sound/pci/oxygen/hifier.c b/sound/pci/oxygen/hifier.c index 090dd4354a2..7442460583d 100644 --- a/sound/pci/oxygen/hifier.c +++ b/sound/pci/oxygen/hifier.c @@ -28,7 +28,7 @@ MODULE_AUTHOR("Clemens Ladisch <clemens@ladisch.de>"); MODULE_DESCRIPTION("TempoTec HiFier driver"); -MODULE_LICENSE("GPL"); +MODULE_LICENSE("GPL v2"); static int index[SNDRV_CARDS] = SNDRV_DEFAULT_IDX; static char *id[SNDRV_CARDS] = SNDRV_DEFAULT_STR; @@ -62,16 +62,28 @@ static void ak4396_write(struct oxygen *chip, u8 reg, u8 value) AK4396_WRITE | (reg << 8) | value); } -static void hifier_init(struct oxygen *chip) +static void update_ak4396_volume(struct oxygen *chip) +{ + ak4396_write(chip, AK4396_LCH_ATT, chip->dac_volume[0]); + ak4396_write(chip, AK4396_RCH_ATT, chip->dac_volume[1]); +} + +static void hifier_registers_init(struct oxygen *chip) { struct hifier_data *data = chip->model_data; - data->ak4396_ctl2 = AK4396_SMUTE | AK4396_DEM_OFF | AK4396_DFS_NORMAL; ak4396_write(chip, AK4396_CONTROL_1, AK4396_DIF_24_MSB | AK4396_RSTN); ak4396_write(chip, AK4396_CONTROL_2, data->ak4396_ctl2); ak4396_write(chip, AK4396_CONTROL_3, AK4396_PCM); - ak4396_write(chip, AK4396_LCH_ATT, 0); - ak4396_write(chip, AK4396_RCH_ATT, 0); + update_ak4396_volume(chip); +} + +static void hifier_init(struct oxygen *chip) +{ + struct hifier_data *data = chip->model_data; + + data->ak4396_ctl2 = AK4396_SMUTE | AK4396_DEM_OFF | AK4396_DFS_NORMAL; + hifier_registers_init(chip); snd_component_add(chip->card, "AK4396"); snd_component_add(chip->card, "CS5340"); @@ -100,12 +112,6 @@ static void set_ak4396_params(struct oxygen *chip, ak4396_write(chip, AK4396_CONTROL_1, AK4396_DIF_24_MSB | AK4396_RSTN); } -static void update_ak4396_volume(struct oxygen *chip) -{ - ak4396_write(chip, AK4396_LCH_ATT, chip->dac_volume[0]); - ak4396_write(chip, AK4396_RCH_ATT, chip->dac_volume[1]); -} - static void update_ak4396_mute(struct oxygen *chip) { struct hifier_data *data = chip->model_data; @@ -140,6 +146,7 @@ static const struct oxygen_model model_hifier = { .init = hifier_init, .control_filter = hifier_control_filter, .cleanup = hifier_cleanup, + .resume = hifier_registers_init, .set_dac_params = set_ak4396_params, .set_adc_params = set_cs5340_params, .update_dac_volume = update_ak4396_volume, @@ -180,6 +187,10 @@ static struct pci_driver hifier_driver = { .id_table = hifier_ids, .probe = hifier_probe, .remove = __devexit_p(oxygen_pci_remove), +#ifdef CONFIG_PM + .suspend = oxygen_pci_suspend, + .resume = oxygen_pci_resume, +#endif }; static int __init alsa_card_hifier_init(void) diff --git a/sound/pci/oxygen/oxygen.c b/sound/pci/oxygen/oxygen.c index 63f185c1ed1..7c8ae31eb46 100644 --- a/sound/pci/oxygen/oxygen.c +++ b/sound/pci/oxygen/oxygen.c @@ -43,7 +43,7 @@ MODULE_AUTHOR("Clemens Ladisch <clemens@ladisch.de>"); MODULE_DESCRIPTION("C-Media CMI8788 driver"); -MODULE_LICENSE("GPL"); +MODULE_LICENSE("GPL v2"); MODULE_SUPPORTED_DEVICE("{{C-Media,CMI8788}}"); static int index[SNDRV_CARDS] = SNDRV_DEFAULT_IDX; @@ -80,6 +80,7 @@ MODULE_DEVICE_TABLE(pci, oxygen_ids); struct generic_data { u8 ak4396_ctl2; + u16 saved_wm8785_registers[2]; }; static void ak4396_write(struct oxygen *chip, unsigned int codec, @@ -99,20 +100,35 @@ static void ak4396_write(struct oxygen *chip, unsigned int codec, static void wm8785_write(struct oxygen *chip, u8 reg, unsigned int value) { + struct generic_data *data = chip->model_data; + oxygen_write_spi(chip, OXYGEN_SPI_TRIGGER | OXYGEN_SPI_DATA_LENGTH_2 | OXYGEN_SPI_CLOCK_160 | (3 << OXYGEN_SPI_CODEC_SHIFT) | OXYGEN_SPI_CEN_LATCH_CLOCK_LO, (reg << 9) | value); + if (reg < ARRAY_SIZE(data->saved_wm8785_registers)) + data->saved_wm8785_registers[reg] = value; } -static void ak4396_init(struct oxygen *chip) +static void update_ak4396_volume(struct oxygen *chip) +{ + unsigned int i; + + for (i = 0; i < 4; ++i) { + ak4396_write(chip, i, + AK4396_LCH_ATT, chip->dac_volume[i * 2]); + ak4396_write(chip, i, + AK4396_RCH_ATT, chip->dac_volume[i * 2 + 1]); + } +} + +static void ak4396_registers_init(struct oxygen *chip) { struct generic_data *data = chip->model_data; unsigned int i; - data->ak4396_ctl2 = AK4396_SMUTE | AK4396_DEM_OFF | AK4396_DFS_NORMAL; for (i = 0; i < 4; ++i) { ak4396_write(chip, i, AK4396_CONTROL_1, AK4396_DIF_24_MSB | AK4396_RSTN); @@ -120,9 +136,16 @@ static void ak4396_init(struct oxygen *chip) AK4396_CONTROL_2, data->ak4396_ctl2); ak4396_write(chip, i, AK4396_CONTROL_3, AK4396_PCM); - ak4396_write(chip, i, AK4396_LCH_ATT, 0); - ak4396_write(chip, i, AK4396_RCH_ATT, 0); } + update_ak4396_volume(chip); +} + +static void ak4396_init(struct oxygen *chip) +{ + struct generic_data *data = chip->model_data; + + data->ak4396_ctl2 = AK4396_SMUTE | AK4396_DEM_OFF | AK4396_DFS_NORMAL; + ak4396_registers_init(chip); snd_component_add(chip->card, "AK4396"); } @@ -133,12 +156,23 @@ static void ak5385_init(struct oxygen *chip) snd_component_add(chip->card, "AK5385"); } -static void wm8785_init(struct oxygen *chip) +static void wm8785_registers_init(struct oxygen *chip) { + struct generic_data *data = chip->model_data; + wm8785_write(chip, WM8785_R7, 0); - wm8785_write(chip, WM8785_R0, WM8785_MCR_SLAVE | - WM8785_OSR_SINGLE | WM8785_FORMAT_LJUST); - wm8785_write(chip, WM8785_R1, WM8785_WL_24); + wm8785_write(chip, WM8785_R0, data->saved_wm8785_registers[0]); + wm8785_write(chip, WM8785_R1, data->saved_wm8785_registers[1]); +} + +static void wm8785_init(struct oxygen *chip) +{ + struct generic_data *data = chip->model_data; + + data->saved_wm8785_registers[0] = WM8785_MCR_SLAVE | + WM8785_OSR_SINGLE | WM8785_FORMAT_LJUST; + data->saved_wm8785_registers[1] = WM8785_WL_24; + wm8785_registers_init(chip); snd_component_add(chip->card, "WM8785"); } @@ -158,6 +192,12 @@ static void generic_cleanup(struct oxygen *chip) { } +static void generic_resume(struct oxygen *chip) +{ + ak4396_registers_init(chip); + wm8785_registers_init(chip); +} + static void set_ak4396_params(struct oxygen *chip, struct snd_pcm_hw_params *params) { @@ -183,18 +223,6 @@ static void set_ak4396_params(struct oxygen *chip, } } -static void update_ak4396_volume(struct oxygen *chip) -{ - unsigned int i; - - for (i = 0; i < 4; ++i) { - ak4396_write(chip, i, - AK4396_LCH_ATT, chip->dac_volume[i * 2]); - ak4396_write(chip, i, - AK4396_RCH_ATT, chip->dac_volume[i * 2 + 1]); - } -} - static void update_ak4396_mute(struct oxygen *chip) { struct generic_data *data = chip->model_data; @@ -256,6 +284,7 @@ static const struct oxygen_model model_generic = { .owner = THIS_MODULE, .init = generic_init, .cleanup = generic_cleanup, + .resume = generic_resume, .set_dac_params = set_ak4396_params, .set_adc_params = set_wm8785_params, .update_dac_volume = update_ak4396_volume, @@ -283,6 +312,7 @@ static const struct oxygen_model model_meridian = { .owner = THIS_MODULE, .init = meridian_init, .cleanup = generic_cleanup, + .resume = ak4396_registers_init, .set_dac_params = set_ak4396_params, .set_adc_params = set_ak5385_params, .update_dac_volume = update_ak4396_volume, @@ -331,6 +361,10 @@ static struct pci_driver oxygen_driver = { .id_table = oxygen_ids, .probe = generic_oxygen_probe, .remove = __devexit_p(oxygen_pci_remove), +#ifdef CONFIG_PM + .suspend = oxygen_pci_suspend, + .resume = oxygen_pci_resume, +#endif }; static int __init alsa_card_oxygen_init(void) diff --git a/sound/pci/oxygen/oxygen.h b/sound/pci/oxygen/oxygen.h index a71c6e05926..74a64488007 100644 --- a/sound/pci/oxygen/oxygen.h +++ b/sound/pci/oxygen/oxygen.h @@ -16,6 +16,8 @@ #define PCM_AC97 5 #define PCM_COUNT 6 +#define OXYGEN_IO_SIZE 0x100 + /* model-specific configuration of outputs/inputs */ #define PLAYBACK_0_TO_I2S 0x001 #define PLAYBACK_1_TO_SPDIF 0x004 @@ -78,6 +80,12 @@ struct oxygen { struct work_struct spdif_input_bits_work; struct work_struct gpio_work; wait_queue_head_t ac97_waitqueue; + union { + u8 _8[OXYGEN_IO_SIZE]; + __le16 _16[OXYGEN_IO_SIZE / 2]; + __le32 _32[OXYGEN_IO_SIZE / 4]; + } saved_registers; + u16 saved_ac97_registers[2][0x40]; }; struct oxygen_model { @@ -89,6 +97,8 @@ struct oxygen_model { int (*control_filter)(struct snd_kcontrol_new *template); int (*mixer_init)(struct oxygen *chip); void (*cleanup)(struct oxygen *chip); + void (*suspend)(struct oxygen *chip); + void (*resume)(struct oxygen *chip); void (*pcm_hardware_filter)(unsigned int channel, struct snd_pcm_hardware *hardware); void (*set_dac_params)(struct oxygen *chip, @@ -117,6 +127,10 @@ struct oxygen_model { int oxygen_pci_probe(struct pci_dev *pci, int index, char *id, const struct oxygen_model *model); void oxygen_pci_remove(struct pci_dev *pci); +#ifdef CONFIG_PM +int oxygen_pci_suspend(struct pci_dev *pci, pm_message_t state); +int oxygen_pci_resume(struct pci_dev *pci); +#endif /* oxygen_mixer.c */ diff --git a/sound/pci/oxygen/oxygen_io.c b/sound/pci/oxygen/oxygen_io.c index 5569606ee87..83f135f80df 100644 --- a/sound/pci/oxygen/oxygen_io.c +++ b/sound/pci/oxygen/oxygen_io.c @@ -44,18 +44,21 @@ EXPORT_SYMBOL(oxygen_read32); void oxygen_write8(struct oxygen *chip, unsigned int reg, u8 value) { outb(value, chip->addr + reg); + chip->saved_registers._8[reg] = value; } EXPORT_SYMBOL(oxygen_write8); void oxygen_write16(struct oxygen *chip, unsigned int reg, u16 value) { outw(value, chip->addr + reg); + chip->saved_registers._16[reg / 2] = cpu_to_le16(value); } EXPORT_SYMBOL(oxygen_write16); void oxygen_write32(struct oxygen *chip, unsigned int reg, u32 value) { outl(value, chip->addr + reg); + chip->saved_registers._32[reg / 4] = cpu_to_le32(value); } EXPORT_SYMBOL(oxygen_write32); @@ -63,7 +66,10 @@ void oxygen_write8_masked(struct oxygen *chip, unsigned int reg, u8 value, u8 mask) { u8 tmp = inb(chip->addr + reg); - outb((tmp & ~mask) | (value & mask), chip->addr + reg); + tmp &= ~mask; + tmp |= value & mask; + outb(tmp, chip->addr + reg); + chip->saved_registers._8[reg] = tmp; } EXPORT_SYMBOL(oxygen_write8_masked); @@ -71,7 +77,10 @@ void oxygen_write16_masked(struct oxygen *chip, unsigned int reg, u16 value, u16 mask) { u16 tmp = inw(chip->addr + reg); - outw((tmp & ~mask) | (value & mask), chip->addr + reg); + tmp &= ~mask; + tmp |= value & mask; + outw(tmp, chip->addr + reg); + chip->saved_registers._16[reg / 2] = cpu_to_le16(tmp); } EXPORT_SYMBOL(oxygen_write16_masked); @@ -79,7 +88,10 @@ void oxygen_write32_masked(struct oxygen *chip, unsigned int reg, u32 value, u32 mask) { u32 tmp = inl(chip->addr + reg); - outl((tmp & ~mask) | (value & mask), chip->addr + reg); + tmp &= ~mask; + tmp |= value & mask; + outl(tmp, chip->addr + reg); + chip->saved_registers._32[reg / 4] = cpu_to_le32(tmp); } EXPORT_SYMBOL(oxygen_write32_masked); @@ -128,8 +140,10 @@ void oxygen_write_ac97(struct oxygen *chip, unsigned int codec, oxygen_write32(chip, OXYGEN_AC97_REGS, reg); /* require two "completed" writes, just to be sure */ if (oxygen_ac97_wait(chip, OXYGEN_AC97_INT_WRITE_DONE) >= 0 && - ++succeeded >= 2) + ++succeeded >= 2) { + chip->saved_ac97_registers[codec][index / 2] = data; return; + } } snd_printk(KERN_ERR "AC'97 write timeout\n"); } diff --git a/sound/pci/oxygen/oxygen_lib.c b/sound/pci/oxygen/oxygen_lib.c index 897697d4350..22f37851045 100644 --- a/sound/pci/oxygen/oxygen_lib.c +++ b/sound/pci/oxygen/oxygen_lib.c @@ -32,7 +32,7 @@ MODULE_AUTHOR("Clemens Ladisch <clemens@ladisch.de>"); MODULE_DESCRIPTION("C-Media CMI8788 helper library"); -MODULE_LICENSE("GPL"); +MODULE_LICENSE("GPL v2"); static irqreturn_t oxygen_interrupt(int dummy, void *dev_id) @@ -173,7 +173,7 @@ static void oxygen_proc_read(struct snd_info_entry *entry, int i, j; snd_iprintf(buffer, "CMI8788\n\n"); - for (i = 0; i < 0x100; i += 0x10) { + for (i = 0; i < OXYGEN_IO_SIZE; i += 0x10) { snd_iprintf(buffer, "%02x:", i); for (j = 0; j < 0x10; ++j) snd_iprintf(buffer, " %02x", oxygen_read8(chip, i + j)); @@ -314,6 +314,10 @@ static void oxygen_init(struct oxygen *chip) OXYGEN_SPDIF_LOCK_MASK | OXYGEN_SPDIF_RATE_MASK); oxygen_write32(chip, OXYGEN_SPDIF_OUTPUT_BITS, chip->spdif_bits); + oxygen_write16(chip, OXYGEN_2WIRE_BUS_STATUS, + OXYGEN_2WIRE_LENGTH_8 | + OXYGEN_2WIRE_INTERRUPT_MASK | + OXYGEN_2WIRE_SPEED_STANDARD); oxygen_clear_bits8(chip, OXYGEN_MPU401_CONTROL, OXYGEN_MPU401_LOOPBACK); oxygen_write8(chip, OXYGEN_GPI_INTERRUPT_MASK, 0); oxygen_write16(chip, OXYGEN_GPIO_INTERRUPT_MASK, 0); @@ -455,7 +459,7 @@ int oxygen_pci_probe(struct pci_dev *pci, int index, char *id, } if (!(pci_resource_flags(pci, 0) & IORESOURCE_IO) || - pci_resource_len(pci, 0) < 0x100) { + pci_resource_len(pci, 0) < OXYGEN_IO_SIZE) { snd_printk(KERN_ERR "invalid PCI I/O range\n"); err = -ENXIO; goto err_pci_regions; @@ -534,3 +538,99 @@ void oxygen_pci_remove(struct pci_dev *pci) pci_set_drvdata(pci, NULL); } EXPORT_SYMBOL(oxygen_pci_remove); + +#ifdef CONFIG_PM +int oxygen_pci_suspend(struct pci_dev *pci, pm_message_t state) +{ + struct snd_card *card = pci_get_drvdata(pci); + struct oxygen *chip = card->private_data; + unsigned int i, saved_interrupt_mask; + + snd_power_change_state(card, SNDRV_CTL_POWER_D3hot); + + for (i = 0; i < PCM_COUNT; ++i) + if (chip->streams[i]) + snd_pcm_suspend(chip->streams[i]); + + if (chip->model->suspend) + chip->model->suspend(chip); + + spin_lock_irq(&chip->reg_lock); + saved_interrupt_mask = chip->interrupt_mask; + chip->interrupt_mask = 0; + oxygen_write16(chip, OXYGEN_DMA_STATUS, 0); + oxygen_write16(chip, OXYGEN_INTERRUPT_MASK, 0); + spin_unlock_irq(&chip->reg_lock); + + synchronize_irq(chip->irq); + flush_scheduled_work(); + chip->interrupt_mask = saved_interrupt_mask; + + pci_disable_device(pci); + pci_save_state(pci); + pci_set_power_state(pci, pci_choose_state(pci, state)); + return 0; +} +EXPORT_SYMBOL(oxygen_pci_suspend); + +static const u32 registers_to_restore[OXYGEN_IO_SIZE / 32] = { + 0xffffffff, 0x00ff077f, 0x00011d08, 0x007f00ff, + 0x00300000, 0x00000fe4, 0x0ff7001f, 0x00000000 +}; +static const u32 ac97_registers_to_restore[2][0x40 / 32] = { + { 0x18284fa2, 0x03060000 }, + { 0x00007fa6, 0x00200000 } +}; + +static inline int is_bit_set(const u32 *bitmap, unsigned int bit) +{ + return bitmap[bit / 32] & (1 << (bit & 31)); +} + +static void oxygen_restore_ac97(struct oxygen *chip, unsigned int codec) +{ + unsigned int i; + + oxygen_write_ac97(chip, codec, AC97_RESET, 0); + msleep(1); + for (i = 1; i < 0x40; ++i) + if (is_bit_set(ac97_registers_to_restore[codec], i)) + oxygen_write_ac97(chip, codec, i * 2, + chip->saved_ac97_registers[codec][i]); +} + +int oxygen_pci_resume(struct pci_dev *pci) +{ + struct snd_card *card = pci_get_drvdata(pci); + struct oxygen *chip = card->private_data; + unsigned int i; + + pci_set_power_state(pci, PCI_D0); + pci_restore_state(pci); + if (pci_enable_device(pci) < 0) { + snd_printk(KERN_ERR "cannot reenable device"); + snd_card_disconnect(card); + return -EIO; + } + pci_set_master(pci); + + oxygen_write16(chip, OXYGEN_DMA_STATUS, 0); + oxygen_write16(chip, OXYGEN_INTERRUPT_MASK, 0); + for (i = 0; i < OXYGEN_IO_SIZE; ++i) + if (is_bit_set(registers_to_restore, i)) + oxygen_write8(chip, i, chip->saved_registers._8[i]); + if (chip->has_ac97_0) + oxygen_restore_ac97(chip, 0); + if (chip->has_ac97_1) + oxygen_restore_ac97(chip, 1); + + if (chip->model->resume) + chip->model->resume(chip); + + oxygen_write16(chip, OXYGEN_INTERRUPT_MASK, chip->interrupt_mask); + + snd_power_change_state(card, SNDRV_CTL_POWER_D0); + return 0; +} +EXPORT_SYMBOL(oxygen_pci_resume); +#endif /* CONFIG_PM */ diff --git a/sound/pci/oxygen/oxygen_pcm.c b/sound/pci/oxygen/oxygen_pcm.c index b17c405e069..c4ad65a3406 100644 --- a/sound/pci/oxygen/oxygen_pcm.c +++ b/sound/pci/oxygen/oxygen_pcm.c @@ -24,6 +24,16 @@ #include <sound/pcm_params.h> #include "oxygen.h" +/* most DMA channels have a 16-bit counter for 32-bit words */ +#define BUFFER_BYTES_MAX ((1 << 16) * 4) +/* the multichannel DMA channel has a 24-bit counter */ +#define BUFFER_BYTES_MAX_MULTICH ((1 << 24) * 4) + +#define PERIOD_BYTES_MIN 64 + +#define DEFAULT_BUFFER_BYTES (BUFFER_BYTES_MAX / 2) +#define DEFAULT_BUFFER_BYTES_MULTICH (1024 * 1024) + static const struct snd_pcm_hardware oxygen_stereo_hardware = { .info = SNDRV_PCM_INFO_MMAP | SNDRV_PCM_INFO_MMAP_VALID | @@ -44,11 +54,11 @@ static const struct snd_pcm_hardware oxygen_stereo_hardware = { .rate_max = 192000, .channels_min = 2, .channels_max = 2, - .buffer_bytes_max = 256 * 1024, - .period_bytes_min = 128, - .period_bytes_max = 128 * 1024, + .buffer_bytes_max = BUFFER_BYTES_MAX, + .period_bytes_min = PERIOD_BYTES_MIN, + .period_bytes_max = BUFFER_BYTES_MAX / 2, .periods_min = 2, - .periods_max = 2048, + .periods_max = BUFFER_BYTES_MAX / PERIOD_BYTES_MIN, }; static const struct snd_pcm_hardware oxygen_multichannel_hardware = { .info = SNDRV_PCM_INFO_MMAP | @@ -70,11 +80,11 @@ static const struct snd_pcm_hardware oxygen_multichannel_hardware = { .rate_max = 192000, .channels_min = 2, .channels_max = 8, - .buffer_bytes_max = 2048 * 1024, - .period_bytes_min = 128, - .period_bytes_max = 256 * 1024, + .buffer_bytes_max = BUFFER_BYTES_MAX_MULTICH, + .period_bytes_min = PERIOD_BYTES_MIN, + .period_bytes_max = BUFFER_BYTES_MAX_MULTICH / 2, .periods_min = 2, - .periods_max = 16384, + .periods_max = BUFFER_BYTES_MAX_MULTICH / PERIOD_BYTES_MIN, }; static const struct snd_pcm_hardware oxygen_ac97_hardware = { .info = SNDRV_PCM_INFO_MMAP | @@ -88,11 +98,11 @@ static const struct snd_pcm_hardware oxygen_ac97_hardware = { .rate_max = 48000, .channels_min = 2, .channels_max = 2, - .buffer_bytes_max = 256 * 1024, - .period_bytes_min = 128, - .period_bytes_max = 128 * 1024, + .buffer_bytes_max = BUFFER_BYTES_MAX, + .period_bytes_min = PERIOD_BYTES_MIN, + .period_bytes_max = BUFFER_BYTES_MAX / 2, .periods_min = 2, - .periods_max = 2048, + .periods_max = BUFFER_BYTES_MAX / PERIOD_BYTES_MIN, }; static const struct snd_pcm_hardware *const oxygen_hardware[PCM_COUNT] = { @@ -155,6 +165,12 @@ static int oxygen_open(struct snd_pcm_substream *substream, if (err < 0) return err; } + if (channel == PCM_MULTICH) { + err = snd_pcm_hw_constraint_minmax + (runtime, SNDRV_PCM_HW_PARAM_PERIOD_TIME, 0, 8192000); + if (err < 0) + return err; + } snd_pcm_set_sync(substream); chip->streams[channel] = substream; @@ -517,6 +533,7 @@ static int oxygen_trigger(struct snd_pcm_substream *substream, int cmd) switch (cmd) { case SNDRV_PCM_TRIGGER_STOP: case SNDRV_PCM_TRIGGER_START: + case SNDRV_PCM_TRIGGER_SUSPEND: pausing = 0; break; case SNDRV_PCM_TRIGGER_PAUSE_PUSH: @@ -663,12 +680,14 @@ int oxygen_pcm_init(struct oxygen *chip) snd_pcm_lib_preallocate_pages(pcm->streams[SNDRV_PCM_STREAM_PLAYBACK].substream, SNDRV_DMA_TYPE_DEV, snd_dma_pci_data(chip->pci), - 512 * 1024, 2048 * 1024); + DEFAULT_BUFFER_BYTES_MULTICH, + BUFFER_BYTES_MAX_MULTICH); if (ins) snd_pcm_lib_preallocate_pages(pcm->streams[SNDRV_PCM_STREAM_CAPTURE].substream, SNDRV_DMA_TYPE_DEV, snd_dma_pci_data(chip->pci), - 128 * 1024, 256 * 1024); + DEFAULT_BUFFER_BYTES, + BUFFER_BYTES_MAX); } outs = !!(chip->model->pcm_dev_cfg & PLAYBACK_1_TO_SPDIF); @@ -688,7 +707,8 @@ int oxygen_pcm_init(struct oxygen *chip) strcpy(pcm->name, "Digital"); snd_pcm_lib_preallocate_pages_for_all(pcm, SNDRV_DMA_TYPE_DEV, snd_dma_pci_data(chip->pci), - 128 * 1024, 256 * 1024); + DEFAULT_BUFFER_BYTES, + BUFFER_BYTES_MAX); } if (chip->has_ac97_1) { @@ -718,7 +738,8 @@ int oxygen_pcm_init(struct oxygen *chip) strcpy(pcm->name, outs ? "Front Panel" : "Analog 2"); snd_pcm_lib_preallocate_pages_for_all(pcm, SNDRV_DMA_TYPE_DEV, snd_dma_pci_data(chip->pci), - 128 * 1024, 256 * 1024); + DEFAULT_BUFFER_BYTES, + BUFFER_BYTES_MAX); } return 0; } diff --git a/sound/pci/oxygen/virtuoso.c b/sound/pci/oxygen/virtuoso.c index 7f84fa5deca..9a2c16bf94e 100644 --- a/sound/pci/oxygen/virtuoso.c +++ b/sound/pci/oxygen/virtuoso.c @@ -79,7 +79,7 @@ MODULE_AUTHOR("Clemens Ladisch <clemens@ladisch.de>"); MODULE_DESCRIPTION("Asus AVx00 driver"); -MODULE_LICENSE("GPL"); +MODULE_LICENSE("GPL v2"); MODULE_SUPPORTED_DEVICE("{{Asus,AV100},{Asus,AV200}}"); static int index[SNDRV_CARDS] = SNDRV_DEFAULT_IDX; @@ -132,6 +132,9 @@ struct xonar_data { u8 ext_power_int_reg; u8 ext_power_bit; u8 has_power; + u8 pcm1796_oversampling; + u8 cs4398_fm; + u8 cs4362a_fm; }; static void pcm1796_write(struct oxygen *chip, unsigned int codec, @@ -159,6 +162,14 @@ static void cs4362a_write(struct oxygen *chip, u8 reg, u8 value) oxygen_write_i2c(chip, I2C_DEVICE_CS4362A, reg, value); } +static void xonar_enable_output(struct oxygen *chip) +{ + struct xonar_data *data = chip->model_data; + + msleep(data->anti_pop_delay); + oxygen_set_bits16(chip, OXYGEN_GPIO_DATA, data->output_enable_bit); +} + static void xonar_common_init(struct oxygen *chip) { struct xonar_data *data = chip->model_data; @@ -170,32 +181,59 @@ static void xonar_common_init(struct oxygen *chip) data->has_power = !!(oxygen_read8(chip, data->ext_power_reg) & data->ext_power_bit); } - oxygen_set_bits16(chip, OXYGEN_GPIO_CONTROL, GPIO_CS53x1_M_MASK); + oxygen_set_bits16(chip, OXYGEN_GPIO_CONTROL, + GPIO_CS53x1_M_MASK | data->output_enable_bit); oxygen_write16_masked(chip, OXYGEN_GPIO_DATA, GPIO_CS53x1_M_SINGLE, GPIO_CS53x1_M_MASK); oxygen_ac97_set_bits(chip, 0, CM9780_JACK, CM9780_FMIC2MIC); - msleep(data->anti_pop_delay); - oxygen_set_bits16(chip, OXYGEN_GPIO_CONTROL, data->output_enable_bit); - oxygen_set_bits16(chip, OXYGEN_GPIO_DATA, data->output_enable_bit); + xonar_enable_output(chip); } -static void xonar_d2_init(struct oxygen *chip) +static void update_pcm1796_volume(struct oxygen *chip) { - struct xonar_data *data = chip->model_data; unsigned int i; - data->anti_pop_delay = 300; - data->output_enable_bit = GPIO_D2_OUTPUT_ENABLE; + for (i = 0; i < 4; ++i) { + pcm1796_write(chip, i, 16, chip->dac_volume[i * 2]); + pcm1796_write(chip, i, 17, chip->dac_volume[i * 2 + 1]); + } +} + +static void update_pcm1796_mute(struct oxygen *chip) +{ + unsigned int i; + u8 value; + + value = PCM1796_DMF_DISABLED | PCM1796_FMT_24_LJUST | PCM1796_ATLD; + if (chip->dac_mute) + value |= PCM1796_MUTE; + for (i = 0; i < 4; ++i) + pcm1796_write(chip, i, 18, value); +} + +static void pcm1796_init(struct oxygen *chip) +{ + struct xonar_data *data = chip->model_data; + unsigned int i; for (i = 0; i < 4; ++i) { - pcm1796_write(chip, i, 18, PCM1796_MUTE | PCM1796_DMF_DISABLED | - PCM1796_FMT_24_LJUST | PCM1796_ATLD); pcm1796_write(chip, i, 19, PCM1796_FLT_SHARP | PCM1796_ATS_1); - pcm1796_write(chip, i, 20, PCM1796_OS_64); + pcm1796_write(chip, i, 20, data->pcm1796_oversampling); pcm1796_write(chip, i, 21, 0); - pcm1796_write(chip, i, 16, 0x0f); /* set ATL/ATR after ATLD */ - pcm1796_write(chip, i, 17, 0x0f); } + update_pcm1796_mute(chip); /* set ATLD before ATL/ATR */ + update_pcm1796_volume(chip); +} + +static void xonar_d2_init(struct oxygen *chip) +{ + struct xonar_data *data = chip->model_data; + + data->anti_pop_delay = 300; + data->output_enable_bit = GPIO_D2_OUTPUT_ENABLE; + data->pcm1796_oversampling = PCM1796_OS_64; + + pcm1796_init(chip); oxygen_set_bits16(chip, OXYGEN_GPIO_CONTROL, GPIO_D2_ALT); oxygen_clear_bits16(chip, OXYGEN_GPIO_DATA, GPIO_D2_ALT); @@ -217,31 +255,47 @@ static void xonar_d2x_init(struct oxygen *chip) xonar_d2_init(chip); } -static void xonar_dx_init(struct oxygen *chip) +static void update_cs4362a_volumes(struct oxygen *chip) { - struct xonar_data *data = chip->model_data; + u8 mute; - data->anti_pop_delay = 800; - data->output_enable_bit = GPIO_DX_OUTPUT_ENABLE; - data->ext_power_reg = OXYGEN_GPI_DATA; - data->ext_power_int_reg = OXYGEN_GPI_INTERRUPT_MASK; - data->ext_power_bit = GPI_DX_EXT_POWER; + mute = chip->dac_mute ? CS4362A_MUTE : 0; + cs4362a_write(chip, 7, (127 - chip->dac_volume[2]) | mute); + cs4362a_write(chip, 8, (127 - chip->dac_volume[3]) | mute); + cs4362a_write(chip, 10, (127 - chip->dac_volume[4]) | mute); + cs4362a_write(chip, 11, (127 - chip->dac_volume[5]) | mute); + cs4362a_write(chip, 13, (127 - chip->dac_volume[6]) | mute); + cs4362a_write(chip, 14, (127 - chip->dac_volume[7]) | mute); +} - oxygen_write16(chip, OXYGEN_2WIRE_BUS_STATUS, - OXYGEN_2WIRE_LENGTH_8 | - OXYGEN_2WIRE_INTERRUPT_MASK | - OXYGEN_2WIRE_SPEED_FAST); +static void update_cs43xx_volume(struct oxygen *chip) +{ + cs4398_write(chip, 5, (127 - chip->dac_volume[0]) * 2); + cs4398_write(chip, 6, (127 - chip->dac_volume[1]) * 2); + update_cs4362a_volumes(chip); +} + +static void update_cs43xx_mute(struct oxygen *chip) +{ + u8 reg; + + reg = CS4398_MUTEP_LOW | CS4398_PAMUTE; + if (chip->dac_mute) + reg |= CS4398_MUTE_B | CS4398_MUTE_A; + cs4398_write(chip, 4, reg); + update_cs4362a_volumes(chip); +} + +static void cs43xx_init(struct oxygen *chip) +{ + struct xonar_data *data = chip->model_data; /* set CPEN (control port mode) and power down */ cs4398_write(chip, 8, CS4398_CPEN | CS4398_PDN); cs4362a_write(chip, 0x01, CS4362A_PDN | CS4362A_CPEN); /* configure */ - cs4398_write(chip, 2, CS4398_FM_SINGLE | - CS4398_DEM_NONE | CS4398_DIF_LJUST); + cs4398_write(chip, 2, data->cs4398_fm); cs4398_write(chip, 3, CS4398_ATAPI_B_R | CS4398_ATAPI_A_L); - cs4398_write(chip, 4, CS4398_MUTEP_LOW | CS4398_PAMUTE); - cs4398_write(chip, 5, 0xfe); - cs4398_write(chip, 6, 0xfe); cs4398_write(chip, 7, CS4398_RMP_DN | CS4398_RMP_UP | CS4398_ZERO_CROSS | CS4398_SOFT_RAMP); cs4362a_write(chip, 0x02, CS4362A_DIF_LJUST); @@ -249,21 +303,35 @@ static void xonar_dx_init(struct oxygen *chip) CS4362A_RMP_UP | CS4362A_ZERO_CROSS | CS4362A_SOFT_RAMP); cs4362a_write(chip, 0x04, CS4362A_RMP_DN | CS4362A_DEM_NONE); cs4362a_write(chip, 0x05, 0); - cs4362a_write(chip, 0x06, CS4362A_FM_SINGLE | - CS4362A_ATAPI_B_R | CS4362A_ATAPI_A_L); - cs4362a_write(chip, 0x07, 0x7f | CS4362A_MUTE); - cs4362a_write(chip, 0x08, 0x7f | CS4362A_MUTE); - cs4362a_write(chip, 0x09, CS4362A_FM_SINGLE | - CS4362A_ATAPI_B_R | CS4362A_ATAPI_A_L); - cs4362a_write(chip, 0x0a, 0x7f | CS4362A_MUTE); - cs4362a_write(chip, 0x0b, 0x7f | CS4362A_MUTE); - cs4362a_write(chip, 0x0c, CS4362A_FM_SINGLE | - CS4362A_ATAPI_B_R | CS4362A_ATAPI_A_L); - cs4362a_write(chip, 0x0d, 0x7f | CS4362A_MUTE); - cs4362a_write(chip, 0x0e, 0x7f | CS4362A_MUTE); + cs4362a_write(chip, 0x06, data->cs4362a_fm); + cs4362a_write(chip, 0x09, data->cs4362a_fm); + cs4362a_write(chip, 0x0c, data->cs4362a_fm); + update_cs43xx_volume(chip); + update_cs43xx_mute(chip); /* clear power down */ cs4398_write(chip, 8, CS4398_CPEN); cs4362a_write(chip, 0x01, CS4362A_CPEN); +} + +static void xonar_dx_init(struct oxygen *chip) +{ + struct xonar_data *data = chip->model_data; + + data->anti_pop_delay = 800; + data->output_enable_bit = GPIO_DX_OUTPUT_ENABLE; + data->ext_power_reg = OXYGEN_GPI_DATA; + data->ext_power_int_reg = OXYGEN_GPI_INTERRUPT_MASK; + data->ext_power_bit = GPI_DX_EXT_POWER; + data->cs4398_fm = CS4398_FM_SINGLE | CS4398_DEM_NONE | CS4398_DIF_LJUST; + data->cs4362a_fm = CS4362A_FM_SINGLE | + CS4362A_ATAPI_B_R | CS4362A_ATAPI_A_L; + + oxygen_write16(chip, OXYGEN_2WIRE_BUS_STATUS, + OXYGEN_2WIRE_LENGTH_8 | + OXYGEN_2WIRE_INTERRUPT_MASK | + OXYGEN_2WIRE_SPEED_FAST); + + cs43xx_init(chip); oxygen_set_bits16(chip, OXYGEN_GPIO_CONTROL, GPIO_DX_FRONT_PANEL | GPIO_DX_INPUT_ROUTE); @@ -291,37 +359,28 @@ static void xonar_dx_cleanup(struct oxygen *chip) oxygen_clear_bits8(chip, OXYGEN_FUNCTION, OXYGEN_FUNCTION_RESET_CODEC); } -static void set_pcm1796_params(struct oxygen *chip, - struct snd_pcm_hw_params *params) +static void xonar_d2_resume(struct oxygen *chip) { - unsigned int i; - u8 value; - - value = params_rate(params) >= 96000 ? PCM1796_OS_32 : PCM1796_OS_64; - for (i = 0; i < 4; ++i) - pcm1796_write(chip, i, 20, value); + pcm1796_init(chip); + xonar_enable_output(chip); } -static void update_pcm1796_volume(struct oxygen *chip) +static void xonar_dx_resume(struct oxygen *chip) { - unsigned int i; - - for (i = 0; i < 4; ++i) { - pcm1796_write(chip, i, 16, chip->dac_volume[i * 2]); - pcm1796_write(chip, i, 17, chip->dac_volume[i * 2 + 1]); - } + cs43xx_init(chip); + xonar_enable_output(chip); } -static void update_pcm1796_mute(struct oxygen *chip) +static void set_pcm1796_params(struct oxygen *chip, + struct snd_pcm_hw_params *params) { + struct xonar_data *data = chip->model_data; unsigned int i; - u8 value; - value = PCM1796_FMT_24_LJUST | PCM1796_ATLD; - if (chip->dac_mute) - value |= PCM1796_MUTE; + data->pcm1796_oversampling = + params_rate(params) >= 96000 ? PCM1796_OS_32 : PCM1796_OS_64; for (i = 0; i < 4; ++i) - pcm1796_write(chip, i, 18, value); + pcm1796_write(chip, i, 20, data->pcm1796_oversampling); } static void set_cs53x1_params(struct oxygen *chip, @@ -342,55 +401,24 @@ static void set_cs53x1_params(struct oxygen *chip, static void set_cs43xx_params(struct oxygen *chip, struct snd_pcm_hw_params *params) { - u8 fm_cs4398, fm_cs4362a; + struct xonar_data *data = chip->model_data; - fm_cs4398 = CS4398_DEM_NONE | CS4398_DIF_LJUST; - fm_cs4362a = CS4362A_ATAPI_B_R | CS4362A_ATAPI_A_L; + data->cs4398_fm = CS4398_DEM_NONE | CS4398_DIF_LJUST; + data->cs4362a_fm = CS4362A_ATAPI_B_R | CS4362A_ATAPI_A_L; if (params_rate(params) <= 50000) { - fm_cs4398 |= CS4398_FM_SINGLE; - fm_cs4362a |= CS4362A_FM_SINGLE; + data->cs4398_fm |= CS4398_FM_SINGLE; + data->cs4362a_fm |= CS4362A_FM_SINGLE; } else if (params_rate(params) <= 100000) { - fm_cs4398 |= CS4398_FM_DOUBLE; - fm_cs4362a |= CS4362A_FM_DOUBLE; + data->cs4398_fm |= CS4398_FM_DOUBLE; + data->cs4362a_fm |= CS4362A_FM_DOUBLE; } else { - fm_cs4398 |= CS4398_FM_QUAD; - fm_cs4362a |= CS4362A_FM_QUAD; + data->cs4398_fm |= CS4398_FM_QUAD; + data->cs4362a_fm |= CS4362A_FM_QUAD; } - cs4398_write(chip, 2, fm_cs4398); - cs4362a_write(chip, 0x06, fm_cs4362a); - cs4362a_write(chip, 0x09, fm_cs4362a); - cs4362a_write(chip, 0x0c, fm_cs4362a); -} - -static void update_cs4362a_volumes(struct oxygen *chip) -{ - u8 mute; - - mute = chip->dac_mute ? CS4362A_MUTE : 0; - cs4362a_write(chip, 7, (127 - chip->dac_volume[2]) | mute); - cs4362a_write(chip, 8, (127 - chip->dac_volume[3]) | mute); - cs4362a_write(chip, 10, (127 - chip->dac_volume[4]) | mute); - cs4362a_write(chip, 11, (127 - chip->dac_volume[5]) | mute); - cs4362a_write(chip, 13, (127 - chip->dac_volume[6]) | mute); - cs4362a_write(chip, 14, (127 - chip->dac_volume[7]) | mute); -} - -static void update_cs43xx_volume(struct oxygen *chip) -{ - cs4398_write(chip, 5, (127 - chip->dac_volume[0]) * 2); - cs4398_write(chip, 6, (127 - chip->dac_volume[1]) * 2); - update_cs4362a_volumes(chip); -} - -static void update_cs43xx_mute(struct oxygen *chip) -{ - u8 reg; - - reg = CS4398_MUTEP_LOW | CS4398_PAMUTE; - if (chip->dac_mute) - reg |= CS4398_MUTE_B | CS4398_MUTE_A; - cs4398_write(chip, 4, reg); - update_cs4362a_volumes(chip); + cs4398_write(chip, 2, data->cs4398_fm); + cs4362a_write(chip, 0x06, data->cs4362a_fm); + cs4362a_write(chip, 0x09, data->cs4362a_fm); + cs4362a_write(chip, 0x0c, data->cs4362a_fm); } static void xonar_gpio_changed(struct oxygen *chip) @@ -535,6 +563,8 @@ static const struct oxygen_model xonar_models[] = { .control_filter = xonar_d2_control_filter, .mixer_init = xonar_mixer_init, .cleanup = xonar_cleanup, + .suspend = xonar_cleanup, + .resume = xonar_d2_resume, .set_dac_params = set_pcm1796_params, .set_adc_params = set_cs53x1_params, .update_dac_volume = update_pcm1796_volume, @@ -563,6 +593,8 @@ static const struct oxygen_model xonar_models[] = { .control_filter = xonar_d2_control_filter, .mixer_init = xonar_mixer_init, .cleanup = xonar_cleanup, + .suspend = xonar_cleanup, + .resume = xonar_d2_resume, .set_dac_params = set_pcm1796_params, .set_adc_params = set_cs53x1_params, .update_dac_volume = update_pcm1796_volume, @@ -592,6 +624,8 @@ static const struct oxygen_model xonar_models[] = { .control_filter = xonar_dx_control_filter, .mixer_init = xonar_dx_mixer_init, .cleanup = xonar_dx_cleanup, + .suspend = xonar_dx_cleanup, + .resume = xonar_dx_resume, .set_dac_params = set_cs43xx_params, .set_adc_params = set_cs53x1_params, .update_dac_volume = update_cs43xx_volume, @@ -636,6 +670,10 @@ static struct pci_driver xonar_driver = { .id_table = xonar_ids, .probe = xonar_probe, .remove = __devexit_p(oxygen_pci_remove), +#ifdef CONFIG_PM + .suspend = oxygen_pci_suspend, + .resume = oxygen_pci_resume, +#endif }; static int __init alsa_card_xonar_init(void) diff --git a/sound/pci/pcxhr/pcxhr.c b/sound/pci/pcxhr/pcxhr.c index 7fdcdc8c6b6..2c7e2533679 100644 --- a/sound/pci/pcxhr/pcxhr.c +++ b/sound/pci/pcxhr/pcxhr.c @@ -516,7 +516,7 @@ static void pcxhr_trigger_tasklet(unsigned long arg) int capture_mask = 0; int playback_mask = 0; -#ifdef CONFIG_SND_DEBUG_DETECT +#ifdef CONFIG_SND_DEBUG_VERBOSE struct timeval my_tv1, my_tv2; do_gettimeofday(&my_tv1); #endif @@ -623,7 +623,7 @@ static void pcxhr_trigger_tasklet(unsigned long arg) mutex_unlock(&mgr->setup_mutex); -#ifdef CONFIG_SND_DEBUG_DETECT +#ifdef CONFIG_SND_DEBUG_VERBOSE do_gettimeofday(&my_tv2); snd_printdd("***TRIGGER TASKLET*** TIME = %ld (err = %x)\n", (long)(my_tv2.tv_usec - my_tv1.tv_usec), err); diff --git a/sound/pci/pcxhr/pcxhr_core.c b/sound/pci/pcxhr/pcxhr_core.c index 78aa81feaa4..abe5c59b72d 100644 --- a/sound/pci/pcxhr/pcxhr_core.c +++ b/sound/pci/pcxhr/pcxhr_core.c @@ -473,7 +473,7 @@ static struct pcxhr_cmd_info pcxhr_dsp_cmds[] = { [CMD_AUDIO_LEVEL_ADJUST] = { 0xc22000, 0, RMH_SSIZE_FIXED }, }; -#ifdef CONFIG_SND_DEBUG_DETECT +#ifdef CONFIG_SND_DEBUG_VERBOSE static char* cmd_names[] = { [CMD_VERSION] = "CMD_VERSION", [CMD_SUPPORTED] = "CMD_SUPPORTED", @@ -549,7 +549,7 @@ static int pcxhr_read_rmh_status(struct pcxhr_mgr *mgr, struct pcxhr_rmh *rmh) } } } -#ifdef CONFIG_SND_DEBUG_DETECT +#ifdef CONFIG_SND_DEBUG_VERBOSE if (rmh->cmd_idx < CMD_LAST_INDEX) snd_printdd(" stat[%d]=%x\n", i, data); #endif @@ -597,7 +597,7 @@ static int pcxhr_send_msg_nolock(struct pcxhr_mgr *mgr, struct pcxhr_rmh *rmh) data |= 0x008000; /* MASK_MORE_THAN_1_WORD_COMMAND */ else data &= 0xff7fff; /* MASK_1_WORD_COMMAND */ -#ifdef CONFIG_SND_DEBUG_DETECT +#ifdef CONFIG_SND_DEBUG_VERBOSE if (rmh->cmd_idx < CMD_LAST_INDEX) snd_printdd("MSG cmd[0]=%x (%s)\n", data, cmd_names[rmh->cmd_idx]); #endif @@ -624,7 +624,7 @@ static int pcxhr_send_msg_nolock(struct pcxhr_mgr *mgr, struct pcxhr_rmh *rmh) for (i=1; i < rmh->cmd_len; i++) { /* send other words */ data = rmh->cmd[i]; -#ifdef CONFIG_SND_DEBUG_DETECT +#ifdef CONFIG_SND_DEBUG_VERBOSE if (rmh->cmd_idx < CMD_LAST_INDEX) snd_printdd(" cmd[%d]=%x\n", i, data); #endif @@ -847,7 +847,7 @@ int pcxhr_set_pipe_state(struct pcxhr_mgr *mgr, int playback_mask, int capture_m int state, i, err; int audio_mask; -#ifdef CONFIG_SND_DEBUG_DETECT +#ifdef CONFIG_SND_DEBUG_VERBOSE struct timeval my_tv1, my_tv2; do_gettimeofday(&my_tv1); #endif @@ -894,7 +894,7 @@ int pcxhr_set_pipe_state(struct pcxhr_mgr *mgr, int playback_mask, int capture_m if (err) return err; } -#ifdef CONFIG_SND_DEBUG_DETECT +#ifdef CONFIG_SND_DEBUG_VERBOSE do_gettimeofday(&my_tv2); snd_printdd("***SET PIPE STATE*** TIME = %ld (err = %x)\n", (long)(my_tv2.tv_usec - my_tv1.tv_usec), err); @@ -951,7 +951,7 @@ static int pcxhr_handle_async_err(struct pcxhr_mgr *mgr, u32 err, enum pcxhr_async_err_src err_src, int pipe, int is_capture) { -#ifdef CONFIG_SND_DEBUG_DETECT +#ifdef CONFIG_SND_DEBUG_VERBOSE static char* err_src_name[] = { [PCXHR_ERR_PIPE] = "Pipe", [PCXHR_ERR_STREAM] = "Stream", @@ -1169,7 +1169,7 @@ irqreturn_t pcxhr_interrupt(int irq, void *dev_id) mgr->dsp_time_last, dsp_time_new); mgr->dsp_time_err++; } -#ifdef CONFIG_SND_DEBUG_DETECT +#ifdef CONFIG_SND_DEBUG_VERBOSE if (dsp_time_diff == 0) snd_printdd("ERROR DSP TIME NO DIFF time(%d)\n", dsp_time_new); else if (dsp_time_diff >= (2*PCXHR_GRANULARITY)) @@ -1208,7 +1208,7 @@ irqreturn_t pcxhr_interrupt(int irq, void *dev_id) mgr->src_it_dsp = reg; tasklet_hi_schedule(&mgr->msg_taskq); } -#ifdef CONFIG_SND_DEBUG_DETECT +#ifdef CONFIG_SND_DEBUG_VERBOSE if (reg & PCXHR_FATAL_DSP_ERR) snd_printdd("FATAL DSP ERROR : %x\n", reg); #endif diff --git a/sound/pci/trident/trident_main.c b/sound/pci/trident/trident_main.c index bbcee2c09ae..a69b4206c69 100644 --- a/sound/pci/trident/trident_main.c +++ b/sound/pci/trident/trident_main.c @@ -1590,7 +1590,10 @@ static int snd_trident_trigger(struct snd_pcm_substream *substream, if (spdif_flag) { if (trident->device != TRIDENT_DEVICE_ID_SI7018) { outl(trident->spdif_pcm_bits, TRID_REG(trident, NX_SPCSTATUS)); - outb(trident->spdif_pcm_ctrl, TRID_REG(trident, NX_SPCTRL_SPCSO + 3)); + val = trident->spdif_pcm_ctrl; + if (!go) + val &= ~(0x28); + outb(val, TRID_REG(trident, NX_SPCTRL_SPCSO + 3)); } else { outl(trident->spdif_pcm_bits, TRID_REG(trident, SI_SPDIF_CS)); val = inl(TRID_REG(trident, SI_SERIAL_INTF_CTRL)) | SPDIF_EN; diff --git a/sound/pci/trident/trident_memory.c b/sound/pci/trident/trident_memory.c index df9b487fa17..3fd7f1b29b0 100644 --- a/sound/pci/trident/trident_memory.c +++ b/sound/pci/trident/trident_memory.c @@ -310,181 +310,3 @@ int snd_trident_free_pages(struct snd_trident *trident, mutex_unlock(&hdr->block_mutex); return 0; } - - -/*---------------------------------------------------------------- - * memory allocation using multiple pages (for synth) - *---------------------------------------------------------------- - * Unlike the DMA allocation above, non-contiguous pages are - * assigned to TLB. - *----------------------------------------------------------------*/ - -/* - */ -static int synth_alloc_pages(struct snd_trident *hw, struct snd_util_memblk *blk); -static int synth_free_pages(struct snd_trident *hw, struct snd_util_memblk *blk); - -/* - * allocate a synth sample area - */ -struct snd_util_memblk * -snd_trident_synth_alloc(struct snd_trident *hw, unsigned int size) -{ - struct snd_util_memblk *blk; - struct snd_util_memhdr *hdr = hw->tlb.memhdr; - - mutex_lock(&hdr->block_mutex); - blk = __snd_util_mem_alloc(hdr, size); - if (blk == NULL) { - mutex_unlock(&hdr->block_mutex); - return NULL; - } - if (synth_alloc_pages(hw, blk)) { - __snd_util_mem_free(hdr, blk); - mutex_unlock(&hdr->block_mutex); - return NULL; - } - mutex_unlock(&hdr->block_mutex); - return blk; -} - -EXPORT_SYMBOL(snd_trident_synth_alloc); - -/* - * free a synth sample area - */ -int -snd_trident_synth_free(struct snd_trident *hw, struct snd_util_memblk *blk) -{ - struct snd_util_memhdr *hdr = hw->tlb.memhdr; - - mutex_lock(&hdr->block_mutex); - synth_free_pages(hw, blk); - __snd_util_mem_free(hdr, blk); - mutex_unlock(&hdr->block_mutex); - return 0; -} - -EXPORT_SYMBOL(snd_trident_synth_free); - -/* - * reset TLB entry and free kernel page - */ -static void clear_tlb(struct snd_trident *trident, int page) -{ - void *ptr = page_to_ptr(trident, page); - dma_addr_t addr = page_to_addr(trident, page); - set_silent_tlb(trident, page); - if (ptr) { - struct snd_dma_buffer dmab; - dmab.dev.type = SNDRV_DMA_TYPE_DEV; - dmab.dev.dev = snd_dma_pci_data(trident->pci); - dmab.area = ptr; - dmab.addr = addr; - dmab.bytes = ALIGN_PAGE_SIZE; - snd_dma_free_pages(&dmab); - } -} - -/* check new allocation range */ -static void get_single_page_range(struct snd_util_memhdr *hdr, - struct snd_util_memblk *blk, - int *first_page_ret, int *last_page_ret) -{ - struct list_head *p; - struct snd_util_memblk *q; - int first_page, last_page; - first_page = firstpg(blk); - if ((p = blk->list.prev) != &hdr->block) { - q = list_entry(p, struct snd_util_memblk, list); - if (lastpg(q) == first_page) - first_page++; /* first page was already allocated */ - } - last_page = lastpg(blk); - if ((p = blk->list.next) != &hdr->block) { - q = list_entry(p, struct snd_util_memblk, list); - if (firstpg(q) == last_page) - last_page--; /* last page was already allocated */ - } - *first_page_ret = first_page; - *last_page_ret = last_page; -} - -/* - * allocate kernel pages and assign them to TLB - */ -static int synth_alloc_pages(struct snd_trident *hw, struct snd_util_memblk *blk) -{ - int page, first_page, last_page; - struct snd_dma_buffer dmab; - - firstpg(blk) = get_aligned_page(blk->offset); - lastpg(blk) = get_aligned_page(blk->offset + blk->size - 1); - get_single_page_range(hw->tlb.memhdr, blk, &first_page, &last_page); - - /* allocate a kernel page for each Trident page - - * fortunately Trident page size and kernel PAGE_SIZE is identical! - */ - for (page = first_page; page <= last_page; page++) { - if (snd_dma_alloc_pages(SNDRV_DMA_TYPE_DEV, snd_dma_pci_data(hw->pci), - ALIGN_PAGE_SIZE, &dmab) < 0) - goto __fail; - if (! is_valid_page(dmab.addr)) { - snd_dma_free_pages(&dmab); - goto __fail; - } - set_tlb_bus(hw, page, (unsigned long)dmab.area, dmab.addr); - } - return 0; - -__fail: - /* release allocated pages */ - last_page = page - 1; - for (page = first_page; page <= last_page; page++) - clear_tlb(hw, page); - - return -ENOMEM; -} - -/* - * free pages - */ -static int synth_free_pages(struct snd_trident *trident, struct snd_util_memblk *blk) -{ - int page, first_page, last_page; - - get_single_page_range(trident->tlb.memhdr, blk, &first_page, &last_page); - for (page = first_page; page <= last_page; page++) - clear_tlb(trident, page); - - return 0; -} - -/* - * copy_from_user(blk + offset, data, size) - */ -int snd_trident_synth_copy_from_user(struct snd_trident *trident, - struct snd_util_memblk *blk, - int offset, const char __user *data, int size) -{ - int page, nextofs, end_offset, temp, temp1; - - offset += blk->offset; - end_offset = offset + size; - page = get_aligned_page(offset) + 1; - do { - nextofs = aligned_page_offset(page); - temp = nextofs - offset; - temp1 = end_offset - offset; - if (temp1 < temp) - temp = temp1; - if (copy_from_user(offset_ptr(trident, offset), data, temp)) - return -EFAULT; - offset = nextofs; - data += temp; - page++; - } while (offset < end_offset); - return 0; -} - -EXPORT_SYMBOL(snd_trident_synth_copy_from_user); diff --git a/sound/pci/via82xx.c b/sound/pci/via82xx.c index b585cc3e4c4..6781be9e307 100644 --- a/sound/pci/via82xx.c +++ b/sound/pci/via82xx.c @@ -1757,6 +1757,12 @@ static struct ac97_quirk ac97_quirks[] = { .type = AC97_TUNE_HP_ONLY }, { + .subvendor = 0x1019, + .subdevice = 0x1841, + .name = "ECS K7VTA3", + .type = AC97_TUNE_HP_ONLY + }, + { .subvendor = 0x1849, .subdevice = 0x3059, .name = "ASRock K7VM2", diff --git a/sound/pci/ymfpci/ymfpci_main.c b/sound/pci/ymfpci/ymfpci_main.c index 29b3056c510..7129df5f315 100644 --- a/sound/pci/ymfpci/ymfpci_main.c +++ b/sound/pci/ymfpci/ymfpci_main.c @@ -2205,6 +2205,7 @@ static int __devinit snd_ymfpci_memalloc(struct snd_ymfpci *chip) for (reg = 0x80; reg < 0xc0; reg += 4) snd_ymfpci_writel(chip, reg, 0); snd_ymfpci_writel(chip, YDSXGR_NATIVEDACOUTVOL, 0x3fff3fff); + snd_ymfpci_writel(chip, YDSXGR_BUF441OUTVOL, 0x3fff3fff); snd_ymfpci_writel(chip, YDSXGR_ZVOUTVOL, 0x3fff3fff); snd_ymfpci_writel(chip, YDSXGR_SPDIFOUTVOL, 0x3fff3fff); snd_ymfpci_writel(chip, YDSXGR_NATIVEADCINVOL, 0x3fff3fff); @@ -2324,6 +2325,7 @@ int snd_ymfpci_suspend(struct pci_dev *pci, pm_message_t state) chip->saved_regs[i] = snd_ymfpci_readl(chip, saved_regs_index[i]); chip->saved_ydsxgr_mode = snd_ymfpci_readl(chip, YDSXGR_MODE); snd_ymfpci_writel(chip, YDSXGR_NATIVEDACOUTVOL, 0); + snd_ymfpci_writel(chip, YDSXGR_BUF441OUTVOL, 0); snd_ymfpci_disable_dsp(chip); pci_disable_device(pci); pci_save_state(pci); diff --git a/sound/pcmcia/Kconfig b/sound/pcmcia/Kconfig index c9fa1a2bc58..7fbb190adf6 100644 --- a/sound/pcmcia/Kconfig +++ b/sound/pcmcia/Kconfig @@ -1,11 +1,16 @@ # ALSA PCMCIA drivers -menu "PCMCIA devices" - depends on SND!=n && PCMCIA +menuconfig SND_PCMCIA + bool "PCMCIA sound devices" + depends on PCMCIA + default y + help + Support for sound devices connected via the PCMCIA bus. + +if SND_PCMCIA && PCMCIA config SND_VXPOCKET tristate "Digigram VXpocket" - depends on SND && PCMCIA select SND_VX_LIB help Say Y here to include support for Digigram VXpocket and @@ -16,7 +21,6 @@ config SND_VXPOCKET config SND_PDAUDIOCF tristate "Sound Core PDAudioCF" - depends on SND && PCMCIA select SND_PCM help Say Y here to include support for Sound Core PDAudioCF @@ -25,4 +29,5 @@ config SND_PDAUDIOCF To compile this driver as a module, choose M here: the module will be called snd-pdaudiocf. -endmenu +endif # SND_PCMCIA + diff --git a/sound/pcmcia/vx/vxp_ops.c b/sound/pcmcia/vx/vxp_ops.c index 157b0b539f3..99bf2a65a6f 100644 --- a/sound/pcmcia/vx/vxp_ops.c +++ b/sound/pcmcia/vx/vxp_ops.c @@ -151,7 +151,7 @@ static int vxp_load_xilinx_binary(struct vx_core *_chip, const struct firmware * unsigned int i; int c; int regCSUER, regRUER; - unsigned char *image; + const unsigned char *image; unsigned char data; /* Switch to programmation mode */ diff --git a/sound/ppc/Kconfig b/sound/ppc/Kconfig index cacb0b13688..777de2b1717 100644 --- a/sound/ppc/Kconfig +++ b/sound/ppc/Kconfig @@ -1,17 +1,17 @@ # ALSA PowerMac drivers -menu "ALSA PowerMac devices" - depends on SND!=n && PPC - -comment "ALSA PowerMac requires I2C" - depends on SND && I2C=n +menuconfig SND_PPC + bool "PowerPC sound devices" + depends on PPC64 || PPC32 + default y + help + Support for sound devices specific to PowerPC architectures. -comment "ALSA PowerMac requires INPUT" - depends on SND && INPUT=n +if SND_PPC config SND_POWERMAC tristate "PowerMac (AWACS, DACA, Burgundy, Tumbler, Keywest)" - depends on SND && I2C && INPUT && PPC_PMAC + depends on I2C && INPUT && PPC_PMAC select SND_PCM help Say Y here to include support for the integrated sound device. @@ -32,14 +32,9 @@ config SND_POWERMAC_AUTO_DRC Note that you can turn on/off DRC manually even without this option. -endmenu - -menu "ALSA PowerPC devices" - depends on SND!=n && ( PPC64 || PPC32 ) - config SND_PS3 tristate "PS3 Audio support" - depends on SND && PS3_PS3AV + depends on PS3_PS3AV select SND_PCM default m help @@ -52,4 +47,5 @@ config SND_PS3_DEFAULT_START_DELAY int "Startup delay time in ms" depends on SND_PS3 default "2000" -endmenu + +endif # SND_PPC diff --git a/sound/ppc/daca.c b/sound/ppc/daca.c index ca9452901a5..8a5b2903193 100644 --- a/sound/ppc/daca.c +++ b/sound/ppc/daca.c @@ -249,9 +249,7 @@ int __init snd_pmac_daca_init(struct snd_pmac *chip) int i, err; struct pmac_daca *mix; -#ifdef CONFIG_KMOD request_module("i2c-powermac"); -#endif /* CONFIG_KMOD */ mix = kzalloc(sizeof(*mix), GFP_KERNEL); if (! mix) diff --git a/sound/ppc/tumbler.c b/sound/ppc/tumbler.c index 3f8d7164cef..009df8dd37a 100644 --- a/sound/ppc/tumbler.c +++ b/sound/ppc/tumbler.c @@ -1350,9 +1350,7 @@ int __init snd_pmac_tumbler_init(struct snd_pmac *chip) struct device_node *tas_node, *np; char *chipname; -#ifdef CONFIG_KMOD request_module("i2c-powermac"); -#endif /* CONFIG_KMOD */ mix = kzalloc(sizeof(*mix), GFP_KERNEL); if (! mix) diff --git a/sound/sh/Kconfig b/sound/sh/Kconfig index b7e08ef22a9..cfc14398580 100644 --- a/sound/sh/Kconfig +++ b/sound/sh/Kconfig @@ -1,14 +1,22 @@ # ALSA SH drivers -menu "SUPERH devices" - depends on SND!=n && SUPERH +menuconfig SND_SUPERH + bool "SUPERH sound devices" + depends on SUPERH + default y + help + Support for sound devices specific to SUPERH architectures. + Drivers that are implemented on ASoC can be found in + "ALSA for SoC audio support" section. + +if SND_SUPERH config SND_AICA tristate "Dreamcast Yamaha AICA sound" - depends on SH_DREAMCAST && SND + depends on SH_DREAMCAST select SND_PCM help ALSA Sound driver for the SEGA Dreamcast console. -endmenu +endif # SND_SUPERH diff --git a/sound/soc/Kconfig b/sound/soc/Kconfig index 18f28ac4bfe..f743530add8 100644 --- a/sound/soc/Kconfig +++ b/sound/soc/Kconfig @@ -2,15 +2,8 @@ # SoC audio configuration # -menu "System on Chip audio support" - depends on SND!=n - -config SND_SOC_AC97_BUS - bool - -config SND_SOC +menuconfig SND_SOC tristate "ALSA for SoC audio support" - depends on SND select SND_PCM ---help--- @@ -23,8 +16,15 @@ config SND_SOC This ASoC audio support can also be built as a module. If so, the module will be called snd-soc-core. +if SND_SOC + +config SND_SOC_AC97_BUS + bool + # All the supported Soc's +source "sound/soc/at32/Kconfig" source "sound/soc/at91/Kconfig" +source "sound/soc/au1x/Kconfig" source "sound/soc/pxa/Kconfig" source "sound/soc/s3c24xx/Kconfig" source "sound/soc/sh/Kconfig" @@ -35,4 +35,5 @@ source "sound/soc/omap/Kconfig" # Supported codecs source "sound/soc/codecs/Kconfig" -endmenu +endif # SND_SOC + diff --git a/sound/soc/Makefile b/sound/soc/Makefile index 782db212710..933a66d3080 100644 --- a/sound/soc/Makefile +++ b/sound/soc/Makefile @@ -1,4 +1,5 @@ snd-soc-core-objs := soc-core.o soc-dapm.o obj-$(CONFIG_SND_SOC) += snd-soc-core.o -obj-$(CONFIG_SND_SOC) += codecs/ at91/ pxa/ s3c24xx/ sh/ fsl/ davinci/ omap/ +obj-$(CONFIG_SND_SOC) += codecs/ at32/ at91/ pxa/ s3c24xx/ sh/ fsl/ davinci/ +obj-$(CONFIG_SND_SOC) += omap/ au1x/ diff --git a/sound/soc/at32/Kconfig b/sound/soc/at32/Kconfig new file mode 100644 index 00000000000..b0765e86c08 --- /dev/null +++ b/sound/soc/at32/Kconfig @@ -0,0 +1,34 @@ +config SND_AT32_SOC + tristate "SoC Audio for the Atmel AT32 System-on-a-Chip" + depends on AVR32 && SND_SOC + help + Say Y or M if you want to add support for codecs attached to + the AT32 SSC interface. You will also need to + to select the audio interfaces to support below. + + +config SND_AT32_SOC_SSC + tristate + + + +config SND_AT32_SOC_PLAYPAQ + tristate "SoC Audio support for PlayPaq with WM8510" + depends on SND_AT32_SOC && BOARD_PLAYPAQ + select SND_AT32_SOC_SSC + select SND_SOC_WM8510 + help + Say Y or M here if you want to add support for SoC audio + on the LRS PlayPaq. + + + +config SND_AT32_SOC_PLAYPAQ_SLAVE + bool "Run CODEC on PlayPaq in slave mode" + depends on SND_AT32_SOC_PLAYPAQ + default n + help + Say Y if you want to run with the AT32 SSC generating the BCLK + and FRAME signals on the PlayPaq. Unless you want to play + with the AT32 as the SSC master, you probably want to say N here, + as this will give you better sound quality. diff --git a/sound/soc/at32/Makefile b/sound/soc/at32/Makefile new file mode 100644 index 00000000000..c03e55ecece --- /dev/null +++ b/sound/soc/at32/Makefile @@ -0,0 +1,11 @@ +# AT32 Platform Support +snd-soc-at32-objs := at32-pcm.o +snd-soc-at32-ssc-objs := at32-ssc.o + +obj-$(CONFIG_SND_AT32_SOC) += snd-soc-at32.o +obj-$(CONFIG_SND_AT32_SOC_SSC) += snd-soc-at32-ssc.o + +# AT32 Machine Support +snd-soc-playpaq-objs := playpaq_wm8510.o + +obj-$(CONFIG_SND_AT32_SOC_PLAYPAQ) += snd-soc-playpaq.o diff --git a/sound/soc/at32/at32-pcm.c b/sound/soc/at32/at32-pcm.c new file mode 100644 index 00000000000..435f1daf177 --- /dev/null +++ b/sound/soc/at32/at32-pcm.c @@ -0,0 +1,491 @@ +/* sound/soc/at32/at32-pcm.c + * ASoC PCM interface for Atmel AT32 SoC + * + * Copyright (C) 2008 Long Range Systems + * Geoffrey Wossum <gwossum@acm.org> + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. + * + * Note that this is basically a port of the sound/soc/at91-pcm.c to + * the AVR32 kernel. Thanks to Frank Mandarino for that code. + */ + +#include <linux/module.h> +#include <linux/init.h> +#include <linux/platform_device.h> +#include <linux/slab.h> +#include <linux/dma-mapping.h> +#include <linux/atmel_pdc.h> + +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> + +#include "at32-pcm.h" + + + +/*--------------------------------------------------------------------------*\ + * Hardware definition +\*--------------------------------------------------------------------------*/ +/* TODO: These values were taken from the AT91 platform driver, check + * them against real values for AT32 + */ +static const struct snd_pcm_hardware at32_pcm_hardware = { + .info = (SNDRV_PCM_INFO_MMAP | + SNDRV_PCM_INFO_MMAP_VALID | + SNDRV_PCM_INFO_INTERLEAVED | + SNDRV_PCM_INFO_BLOCK_TRANSFER | + SNDRV_PCM_INFO_PAUSE), + + .formats = SNDRV_PCM_FMTBIT_S16, + .period_bytes_min = 32, + .period_bytes_max = 8192, /* 512 frames * 16 bytes / frame */ + .periods_min = 2, + .periods_max = 1024, + .buffer_bytes_max = 32 * 1024, +}; + + + +/*--------------------------------------------------------------------------*\ + * Data types +\*--------------------------------------------------------------------------*/ +struct at32_runtime_data { + struct at32_pcm_dma_params *params; + dma_addr_t dma_buffer; /* physical address of DMA buffer */ + dma_addr_t dma_buffer_end; /* first address beyond DMA buffer */ + size_t period_size; + + dma_addr_t period_ptr; /* physical address of next period */ + int periods; /* period index of period_ptr */ + + /* Save PDC registers (for power management) */ + u32 pdc_xpr_save; + u32 pdc_xcr_save; + u32 pdc_xnpr_save; + u32 pdc_xncr_save; +}; + + + +/*--------------------------------------------------------------------------*\ + * Helper functions +\*--------------------------------------------------------------------------*/ +static int at32_pcm_preallocate_dma_buffer(struct snd_pcm *pcm, int stream) +{ + struct snd_pcm_substream *substream = pcm->streams[stream].substream; + struct snd_dma_buffer *dmabuf = &substream->dma_buffer; + size_t size = at32_pcm_hardware.buffer_bytes_max; + + dmabuf->dev.type = SNDRV_DMA_TYPE_DEV; + dmabuf->dev.dev = pcm->card->dev; + dmabuf->private_data = NULL; + dmabuf->area = dma_alloc_coherent(pcm->card->dev, size, + &dmabuf->addr, GFP_KERNEL); + pr_debug("at32_pcm: preallocate_dma_buffer: " + "area=%p, addr=%p, size=%ld\n", + (void *)dmabuf->area, (void *)dmabuf->addr, size); + + if (!dmabuf->area) + return -ENOMEM; + + dmabuf->bytes = size; + return 0; +} + + + +/*--------------------------------------------------------------------------*\ + * ISR +\*--------------------------------------------------------------------------*/ +static void at32_pcm_dma_irq(u32 ssc_sr, struct snd_pcm_substream *substream) +{ + struct snd_pcm_runtime *rtd = substream->runtime; + struct at32_runtime_data *prtd = rtd->private_data; + struct at32_pcm_dma_params *params = prtd->params; + static int count; + + count++; + if (ssc_sr & params->mask->ssc_endbuf) { + pr_warning("at32-pcm: buffer %s on %s (SSC_SR=%#x, count=%d)\n", + substream->stream == SNDRV_PCM_STREAM_PLAYBACK ? + "underrun" : "overrun", params->name, ssc_sr, count); + + /* re-start the PDC */ + ssc_writex(params->ssc->regs, ATMEL_PDC_PTCR, + params->mask->pdc_disable); + prtd->period_ptr += prtd->period_size; + if (prtd->period_ptr >= prtd->dma_buffer_end) + prtd->period_ptr = prtd->dma_buffer; + + + ssc_writex(params->ssc->regs, params->pdc->xpr, + prtd->period_ptr); + ssc_writex(params->ssc->regs, params->pdc->xcr, + prtd->period_size / params->pdc_xfer_size); + ssc_writex(params->ssc->regs, ATMEL_PDC_PTCR, + params->mask->pdc_enable); + } + + + if (ssc_sr & params->mask->ssc_endx) { + /* Load the PDC next pointer and counter registers */ + prtd->period_ptr += prtd->period_size; + if (prtd->period_ptr >= prtd->dma_buffer_end) + prtd->period_ptr = prtd->dma_buffer; + ssc_writex(params->ssc->regs, params->pdc->xnpr, + prtd->period_ptr); + ssc_writex(params->ssc->regs, params->pdc->xncr, + prtd->period_size / params->pdc_xfer_size); + } + + + snd_pcm_period_elapsed(substream); +} + + + +/*--------------------------------------------------------------------------*\ + * PCM operations +\*--------------------------------------------------------------------------*/ +static int at32_pcm_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params) +{ + struct snd_pcm_runtime *runtime = substream->runtime; + struct at32_runtime_data *prtd = runtime->private_data; + struct snd_soc_pcm_runtime *rtd = substream->private_data; + + /* this may get called several times by oss emulation + * with different params + */ + snd_pcm_set_runtime_buffer(substream, &substream->dma_buffer); + runtime->dma_bytes = params_buffer_bytes(params); + + prtd->params = rtd->dai->cpu_dai->dma_data; + prtd->params->dma_intr_handler = at32_pcm_dma_irq; + + prtd->dma_buffer = runtime->dma_addr; + prtd->dma_buffer_end = runtime->dma_addr + runtime->dma_bytes; + prtd->period_size = params_period_bytes(params); + + pr_debug("hw_params: DMA for %s initialized " + "(dma_bytes=%ld, period_size=%ld)\n", + prtd->params->name, runtime->dma_bytes, prtd->period_size); + + return 0; +} + + + +static int at32_pcm_hw_free(struct snd_pcm_substream *substream) +{ + struct at32_runtime_data *prtd = substream->runtime->private_data; + struct at32_pcm_dma_params *params = prtd->params; + + if (params != NULL) { + ssc_writex(params->ssc->regs, SSC_PDC_PTCR, + params->mask->pdc_disable); + prtd->params->dma_intr_handler = NULL; + } + + return 0; +} + + + +static int at32_pcm_prepare(struct snd_pcm_substream *substream) +{ + struct at32_runtime_data *prtd = substream->runtime->private_data; + struct at32_pcm_dma_params *params = prtd->params; + + ssc_writex(params->ssc->regs, SSC_IDR, + params->mask->ssc_endx | params->mask->ssc_endbuf); + ssc_writex(params->ssc->regs, ATMEL_PDC_PTCR, + params->mask->pdc_disable); + + return 0; +} + + +static int at32_pcm_trigger(struct snd_pcm_substream *substream, int cmd) +{ + struct snd_pcm_runtime *rtd = substream->runtime; + struct at32_runtime_data *prtd = rtd->private_data; + struct at32_pcm_dma_params *params = prtd->params; + int ret = 0; + + pr_debug("at32_pcm_trigger: buffer_size = %ld, " + "dma_area = %p, dma_bytes = %ld\n", + rtd->buffer_size, rtd->dma_area, rtd->dma_bytes); + + switch (cmd) { + case SNDRV_PCM_TRIGGER_START: + prtd->period_ptr = prtd->dma_buffer; + + ssc_writex(params->ssc->regs, params->pdc->xpr, + prtd->period_ptr); + ssc_writex(params->ssc->regs, params->pdc->xcr, + prtd->period_size / params->pdc_xfer_size); + + prtd->period_ptr += prtd->period_size; + ssc_writex(params->ssc->regs, params->pdc->xnpr, + prtd->period_ptr); + ssc_writex(params->ssc->regs, params->pdc->xncr, + prtd->period_size / params->pdc_xfer_size); + + pr_debug("trigger: period_ptr=%lx, xpr=%x, " + "xcr=%d, xnpr=%x, xncr=%d\n", + (unsigned long)prtd->period_ptr, + ssc_readx(params->ssc->regs, params->pdc->xpr), + ssc_readx(params->ssc->regs, params->pdc->xcr), + ssc_readx(params->ssc->regs, params->pdc->xnpr), + ssc_readx(params->ssc->regs, params->pdc->xncr)); + + ssc_writex(params->ssc->regs, SSC_IER, + params->mask->ssc_endx | params->mask->ssc_endbuf); + ssc_writex(params->ssc->regs, SSC_PDC_PTCR, + params->mask->pdc_enable); + + pr_debug("sr=%x, imr=%x\n", + ssc_readx(params->ssc->regs, SSC_SR), + ssc_readx(params->ssc->regs, SSC_IER)); + break; /* SNDRV_PCM_TRIGGER_START */ + + + + case SNDRV_PCM_TRIGGER_STOP: + case SNDRV_PCM_TRIGGER_SUSPEND: + case SNDRV_PCM_TRIGGER_PAUSE_PUSH: + ssc_writex(params->ssc->regs, ATMEL_PDC_PTCR, + params->mask->pdc_disable); + break; + + + case SNDRV_PCM_TRIGGER_RESUME: + case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: + ssc_writex(params->ssc->regs, ATMEL_PDC_PTCR, + params->mask->pdc_enable); + break; + + default: + ret = -EINVAL; + } + + return ret; +} + + + +static snd_pcm_uframes_t at32_pcm_pointer(struct snd_pcm_substream *substream) +{ + struct snd_pcm_runtime *runtime = substream->runtime; + struct at32_runtime_data *prtd = runtime->private_data; + struct at32_pcm_dma_params *params = prtd->params; + dma_addr_t ptr; + snd_pcm_uframes_t x; + + ptr = (dma_addr_t) ssc_readx(params->ssc->regs, params->pdc->xpr); + x = bytes_to_frames(runtime, ptr - prtd->dma_buffer); + + if (x == runtime->buffer_size) + x = 0; + + return x; +} + + + +static int at32_pcm_open(struct snd_pcm_substream *substream) +{ + struct snd_pcm_runtime *runtime = substream->runtime; + struct at32_runtime_data *prtd; + int ret = 0; + + snd_soc_set_runtime_hwparams(substream, &at32_pcm_hardware); + + /* ensure that buffer size is a multiple of period size */ + ret = snd_pcm_hw_constraint_integer(runtime, + SNDRV_PCM_HW_PARAM_PERIODS); + if (ret < 0) + goto out; + + prtd = kzalloc(sizeof(*prtd), GFP_KERNEL); + if (prtd == NULL) { + ret = -ENOMEM; + goto out; + } + runtime->private_data = prtd; + + +out: + return ret; +} + + + +static int at32_pcm_close(struct snd_pcm_substream *substream) +{ + struct at32_runtime_data *prtd = substream->runtime->private_data; + + kfree(prtd); + return 0; +} + + +static int at32_pcm_mmap(struct snd_pcm_substream *substream, + struct vm_area_struct *vma) +{ + return remap_pfn_range(vma, vma->vm_start, + substream->dma_buffer.addr >> PAGE_SHIFT, + vma->vm_end - vma->vm_start, vma->vm_page_prot); +} + + + +static struct snd_pcm_ops at32_pcm_ops = { + .open = at32_pcm_open, + .close = at32_pcm_close, + .ioctl = snd_pcm_lib_ioctl, + .hw_params = at32_pcm_hw_params, + .hw_free = at32_pcm_hw_free, + .prepare = at32_pcm_prepare, + .trigger = at32_pcm_trigger, + .pointer = at32_pcm_pointer, + .mmap = at32_pcm_mmap, +}; + + + +/*--------------------------------------------------------------------------*\ + * ASoC platform driver +\*--------------------------------------------------------------------------*/ +static u64 at32_pcm_dmamask = 0xffffffff; + +static int at32_pcm_new(struct snd_card *card, + struct snd_soc_dai *dai, + struct snd_pcm *pcm) +{ + int ret = 0; + + if (!card->dev->dma_mask) + card->dev->dma_mask = &at32_pcm_dmamask; + if (!card->dev->coherent_dma_mask) + card->dev->coherent_dma_mask = 0xffffffff; + + if (dai->playback.channels_min) { + ret = at32_pcm_preallocate_dma_buffer( + pcm, SNDRV_PCM_STREAM_PLAYBACK); + if (ret) + goto out; + } + + if (dai->capture.channels_min) { + pr_debug("at32-pcm: Allocating PCM capture DMA buffer\n"); + ret = at32_pcm_preallocate_dma_buffer( + pcm, SNDRV_PCM_STREAM_CAPTURE); + if (ret) + goto out; + } + + +out: + return ret; +} + + + +static void at32_pcm_free_dma_buffers(struct snd_pcm *pcm) +{ + struct snd_pcm_substream *substream; + struct snd_dma_buffer *buf; + int stream; + + for (stream = 0; stream < 2; stream++) { + substream = pcm->streams[stream].substream; + if (substream == NULL) + continue; + + buf = &substream->dma_buffer; + if (!buf->area) + continue; + dma_free_coherent(pcm->card->dev, buf->bytes, + buf->area, buf->addr); + buf->area = NULL; + } +} + + + +#ifdef CONFIG_PM +static int at32_pcm_suspend(struct platform_device *pdev, + struct snd_soc_dai *dai) +{ + struct snd_pcm_runtime *runtime = dai->runtime; + struct at32_runtime_data *prtd; + struct at32_pcm_dma_params *params; + + if (runtime == NULL) + return 0; + prtd = runtime->private_data; + params = prtd->params; + + /* Disable the PDC and save the PDC registers */ + ssc_writex(params->ssc->regs, PDC_PTCR, params->mask->pdc_disable); + + prtd->pdc_xpr_save = ssc_readx(params->ssc->regs, params->pdc->xpr); + prtd->pdc_xcr_save = ssc_readx(params->ssc->regs, params->pdc->xcr); + prtd->pdc_xnpr_save = ssc_readx(params->ssc->regs, params->pdc->xnpr); + prtd->pdc_xncr_save = ssc_readx(params->ssc->regs, params->pdc->xncr); + + return 0; +} + + + +static int at32_pcm_resume(struct platform_device *pdev, + struct snd_soc_dai *dai) +{ + struct snd_pcm_runtime *runtime = dai->runtime; + struct at32_runtime_data *prtd; + struct at32_pcm_dma_params *params; + + if (runtime == NULL) + return 0; + prtd = runtime->private_data; + params = prtd->params; + + /* Restore the PDC registers and enable the PDC */ + ssc_writex(params->ssc->regs, params->pdc->xpr, prtd->pdc_xpr_save); + ssc_writex(params->ssc->regs, params->pdc->xcr, prtd->pdc_xcr_save); + ssc_writex(params->ssc->regs, params->pdc->xnpr, prtd->pdc_xnpr_save); + ssc_writex(params->ssc->regs, params->pdc->xncr, prtd->pdc_xncr_save); + + ssc_writex(params->ssc->regs, PDC_PTCR, params->mask->pdc_enable); + return 0; +} +#else /* CONFIG_PM */ +# define at32_pcm_suspend NULL +# define at32_pcm_resume NULL +#endif /* CONFIG_PM */ + + + +struct snd_soc_platform at32_soc_platform = { + .name = "at32-audio", + .pcm_ops = &at32_pcm_ops, + .pcm_new = at32_pcm_new, + .pcm_free = at32_pcm_free_dma_buffers, + .suspend = at32_pcm_suspend, + .resume = at32_pcm_resume, +}; +EXPORT_SYMBOL_GPL(at32_soc_platform); + + + +MODULE_AUTHOR("Geoffrey Wossum <gwossum@acm.org>"); +MODULE_DESCRIPTION("Atmel AT32 PCM module"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/at32/at32-pcm.h b/sound/soc/at32/at32-pcm.h new file mode 100644 index 00000000000..2a52430417d --- /dev/null +++ b/sound/soc/at32/at32-pcm.h @@ -0,0 +1,79 @@ +/* sound/soc/at32/at32-pcm.h + * ASoC PCM interface for Atmel AT32 SoC + * + * Copyright (C) 2008 Long Range Systems + * Geoffrey Wossum <gwossum@acm.org> + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. + */ + +#ifndef __SOUND_SOC_AT32_AT32_PCM_H +#define __SOUND_SOC_AT32_AT32_PCM_H __FILE__ + +#include <linux/atmel-ssc.h> + + +/* + * Registers and status bits that are required by the PCM driver + * TODO: Is ptcr really used? + */ +struct at32_pdc_regs { + u32 xpr; /* PDC RX/TX pointer */ + u32 xcr; /* PDC RX/TX counter */ + u32 xnpr; /* PDC next RX/TX pointer */ + u32 xncr; /* PDC next RX/TX counter */ + u32 ptcr; /* PDC transfer control */ +}; + + + +/* + * SSC mask info + */ +struct at32_ssc_mask { + u32 ssc_enable; /* SSC RX/TX enable */ + u32 ssc_disable; /* SSC RX/TX disable */ + u32 ssc_endx; /* SSC ENDTX or ENDRX */ + u32 ssc_endbuf; /* SSC TXBUFF or RXBUFF */ + u32 pdc_enable; /* PDC RX/TX enable */ + u32 pdc_disable; /* PDC RX/TX disable */ +}; + + + +/* + * This structure, shared between the PCM driver and the interface, + * contains all information required by the PCM driver to perform the + * PDC DMA operation. All fields except dma_intr_handler() are initialized + * by the interface. The dms_intr_handler() pointer is set by the PCM + * driver and called by the interface SSC interrupt handler if it is + * non-NULL. + */ +struct at32_pcm_dma_params { + char *name; /* stream identifier */ + int pdc_xfer_size; /* PDC counter increment in bytes */ + struct ssc_device *ssc; /* SSC device for stream */ + struct at32_pdc_regs *pdc; /* PDC register info */ + struct at32_ssc_mask *mask; /* SSC mask info */ + struct snd_pcm_substream *substream; + void (*dma_intr_handler) (u32, struct snd_pcm_substream *); +}; + + + +/* + * The AT32 ASoC platform driver + */ +extern struct snd_soc_platform at32_soc_platform; + + + +/* + * SSC register access (since ssc_writel() / ssc_readl() require literal name) + */ +#define ssc_readx(base, reg) (__raw_readl((base) + (reg))) +#define ssc_writex(base, reg, value) __raw_writel((value), (base) + (reg)) + +#endif /* __SOUND_SOC_AT32_AT32_PCM_H */ diff --git a/sound/soc/at32/at32-ssc.c b/sound/soc/at32/at32-ssc.c new file mode 100644 index 00000000000..4ef6492c902 --- /dev/null +++ b/sound/soc/at32/at32-ssc.c @@ -0,0 +1,849 @@ +/* sound/soc/at32/at32-ssc.c + * ASoC platform driver for AT32 using SSC as DAI + * + * Copyright (C) 2008 Long Range Systems + * Geoffrey Wossum <gwossum@acm.org> + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. + * + * Note that this is basically a port of the sound/soc/at91-ssc.c to + * the AVR32 kernel. Thanks to Frank Mandarino for that code. + */ + +/* #define DEBUG */ + +#include <linux/init.h> +#include <linux/module.h> +#include <linux/interrupt.h> +#include <linux/device.h> +#include <linux/delay.h> +#include <linux/clk.h> +#include <linux/io.h> +#include <linux/atmel_pdc.h> +#include <linux/atmel-ssc.h> + +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/initval.h> +#include <sound/soc.h> + +#include "at32-pcm.h" +#include "at32-ssc.h" + + + +/*-------------------------------------------------------------------------*\ + * Constants +\*-------------------------------------------------------------------------*/ +#define NUM_SSC_DEVICES 3 + +/* + * SSC direction masks + */ +#define SSC_DIR_MASK_UNUSED 0 +#define SSC_DIR_MASK_PLAYBACK 1 +#define SSC_DIR_MASK_CAPTURE 2 + +/* + * SSC register values that Atmel left out of <linux/atmel-ssc.h>. These + * are expected to be used with SSC_BF + */ +/* START bit field values */ +#define SSC_START_CONTINUOUS 0 +#define SSC_START_TX_RX 1 +#define SSC_START_LOW_RF 2 +#define SSC_START_HIGH_RF 3 +#define SSC_START_FALLING_RF 4 +#define SSC_START_RISING_RF 5 +#define SSC_START_LEVEL_RF 6 +#define SSC_START_EDGE_RF 7 +#define SSS_START_COMPARE_0 8 + +/* CKI bit field values */ +#define SSC_CKI_FALLING 0 +#define SSC_CKI_RISING 1 + +/* CKO bit field values */ +#define SSC_CKO_NONE 0 +#define SSC_CKO_CONTINUOUS 1 +#define SSC_CKO_TRANSFER 2 + +/* CKS bit field values */ +#define SSC_CKS_DIV 0 +#define SSC_CKS_CLOCK 1 +#define SSC_CKS_PIN 2 + +/* FSEDGE bit field values */ +#define SSC_FSEDGE_POSITIVE 0 +#define SSC_FSEDGE_NEGATIVE 1 + +/* FSOS bit field values */ +#define SSC_FSOS_NONE 0 +#define SSC_FSOS_NEGATIVE 1 +#define SSC_FSOS_POSITIVE 2 +#define SSC_FSOS_LOW 3 +#define SSC_FSOS_HIGH 4 +#define SSC_FSOS_TOGGLE 5 + +#define START_DELAY 1 + + + +/*-------------------------------------------------------------------------*\ + * Module data +\*-------------------------------------------------------------------------*/ +/* + * SSC PDC registered required by the PCM DMA engine + */ +static struct at32_pdc_regs pdc_tx_reg = { + .xpr = SSC_PDC_TPR, + .xcr = SSC_PDC_TCR, + .xnpr = SSC_PDC_TNPR, + .xncr = SSC_PDC_TNCR, +}; + + + +static struct at32_pdc_regs pdc_rx_reg = { + .xpr = SSC_PDC_RPR, + .xcr = SSC_PDC_RCR, + .xnpr = SSC_PDC_RNPR, + .xncr = SSC_PDC_RNCR, +}; + + + +/* + * SSC and PDC status bits for transmit and receive + */ +static struct at32_ssc_mask ssc_tx_mask = { + .ssc_enable = SSC_BIT(CR_TXEN), + .ssc_disable = SSC_BIT(CR_TXDIS), + .ssc_endx = SSC_BIT(SR_ENDTX), + .ssc_endbuf = SSC_BIT(SR_TXBUFE), + .pdc_enable = SSC_BIT(PDC_PTCR_TXTEN), + .pdc_disable = SSC_BIT(PDC_PTCR_TXTDIS), +}; + + + +static struct at32_ssc_mask ssc_rx_mask = { + .ssc_enable = SSC_BIT(CR_RXEN), + .ssc_disable = SSC_BIT(CR_RXDIS), + .ssc_endx = SSC_BIT(SR_ENDRX), + .ssc_endbuf = SSC_BIT(SR_RXBUFF), + .pdc_enable = SSC_BIT(PDC_PTCR_RXTEN), + .pdc_disable = SSC_BIT(PDC_PTCR_RXTDIS), +}; + + + +/* + * DMA parameters for each SSC + */ +static struct at32_pcm_dma_params ssc_dma_params[NUM_SSC_DEVICES][2] = { + { + { + .name = "SSC0 PCM out", + .pdc = &pdc_tx_reg, + .mask = &ssc_tx_mask, + }, + { + .name = "SSC0 PCM in", + .pdc = &pdc_rx_reg, + .mask = &ssc_rx_mask, + }, + }, + { + { + .name = "SSC1 PCM out", + .pdc = &pdc_tx_reg, + .mask = &ssc_tx_mask, + }, + { + .name = "SSC1 PCM in", + .pdc = &pdc_rx_reg, + .mask = &ssc_rx_mask, + }, + }, + { + { + .name = "SSC2 PCM out", + .pdc = &pdc_tx_reg, + .mask = &ssc_tx_mask, + }, + { + .name = "SSC2 PCM in", + .pdc = &pdc_rx_reg, + .mask = &ssc_rx_mask, + }, + }, +}; + + + +static struct at32_ssc_info ssc_info[NUM_SSC_DEVICES] = { + { + .name = "ssc0", + .lock = __SPIN_LOCK_UNLOCKED(ssc_info[0].lock), + .dir_mask = SSC_DIR_MASK_UNUSED, + .initialized = 0, + }, + { + .name = "ssc1", + .lock = __SPIN_LOCK_UNLOCKED(ssc_info[1].lock), + .dir_mask = SSC_DIR_MASK_UNUSED, + .initialized = 0, + }, + { + .name = "ssc2", + .lock = __SPIN_LOCK_UNLOCKED(ssc_info[2].lock), + .dir_mask = SSC_DIR_MASK_UNUSED, + .initialized = 0, + }, +}; + + + + +/*-------------------------------------------------------------------------*\ + * ISR +\*-------------------------------------------------------------------------*/ +/* + * SSC interrupt handler. Passes PDC interrupts to the DMA interrupt + * handler in the PCM driver. + */ +static irqreturn_t at32_ssc_interrupt(int irq, void *dev_id) +{ + struct at32_ssc_info *ssc_p = dev_id; + struct at32_pcm_dma_params *dma_params; + u32 ssc_sr; + u32 ssc_substream_mask; + int i; + + ssc_sr = (ssc_readl(ssc_p->ssc->regs, SR) & + ssc_readl(ssc_p->ssc->regs, IMR)); + + /* + * Loop through substreams attached to this SSC. If a DMA-related + * interrupt occured on that substream, call the DMA interrupt + * handler function, if one has been registered in the dma_param + * structure by the PCM driver. + */ + for (i = 0; i < ARRAY_SIZE(ssc_p->dma_params); i++) { + dma_params = ssc_p->dma_params[i]; + + if ((dma_params != NULL) && + (dma_params->dma_intr_handler != NULL)) { + ssc_substream_mask = (dma_params->mask->ssc_endx | + dma_params->mask->ssc_endbuf); + if (ssc_sr & ssc_substream_mask) { + dma_params->dma_intr_handler(ssc_sr, + dma_params-> + substream); + } + } + } + + + return IRQ_HANDLED; +} + +/*-------------------------------------------------------------------------*\ + * DAI functions +\*-------------------------------------------------------------------------*/ +/* + * Startup. Only that one substream allowed in each direction. + */ +static int at32_ssc_startup(struct snd_pcm_substream *substream) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct at32_ssc_info *ssc_p = &ssc_info[rtd->dai->cpu_dai->id]; + int dir_mask; + + dir_mask = ((substream->stream == SNDRV_PCM_STREAM_PLAYBACK) ? + SSC_DIR_MASK_PLAYBACK : SSC_DIR_MASK_CAPTURE); + + spin_lock_irq(&ssc_p->lock); + if (ssc_p->dir_mask & dir_mask) { + spin_unlock_irq(&ssc_p->lock); + return -EBUSY; + } + ssc_p->dir_mask |= dir_mask; + spin_unlock_irq(&ssc_p->lock); + + return 0; +} + + + +/* + * Shutdown. Clear DMA parameters and shutdown the SSC if there + * are no other substreams open. + */ +static void at32_ssc_shutdown(struct snd_pcm_substream *substream) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct at32_ssc_info *ssc_p = &ssc_info[rtd->dai->cpu_dai->id]; + struct at32_pcm_dma_params *dma_params; + int dir_mask; + + dma_params = ssc_p->dma_params[substream->stream]; + + if (dma_params != NULL) { + ssc_writel(dma_params->ssc->regs, CR, + dma_params->mask->ssc_disable); + pr_debug("%s disabled SSC_SR=0x%08x\n", + (substream->stream ? "receiver" : "transmit"), + ssc_readl(ssc_p->ssc->regs, SR)); + + dma_params->ssc = NULL; + dma_params->substream = NULL; + ssc_p->dma_params[substream->stream] = NULL; + } + + + dir_mask = 1 << substream->stream; + spin_lock_irq(&ssc_p->lock); + ssc_p->dir_mask &= ~dir_mask; + if (!ssc_p->dir_mask) { + /* Shutdown the SSC clock */ + pr_debug("at32-ssc: Stopping user %d clock\n", + ssc_p->ssc->user); + clk_disable(ssc_p->ssc->clk); + + if (ssc_p->initialized) { + free_irq(ssc_p->ssc->irq, ssc_p); + ssc_p->initialized = 0; + } + + /* Reset the SSC */ + ssc_writel(ssc_p->ssc->regs, CR, SSC_BIT(CR_SWRST)); + + /* clear the SSC dividers */ + ssc_p->cmr_div = 0; + ssc_p->tcmr_period = 0; + ssc_p->rcmr_period = 0; + } + spin_unlock_irq(&ssc_p->lock); +} + + + +/* + * Set the SSC system clock rate + */ +static int at32_ssc_set_dai_sysclk(struct snd_soc_dai *cpu_dai, + int clk_id, unsigned int freq, int dir) +{ + /* TODO: What the heck do I do here? */ + return 0; +} + + + +/* + * Record DAI format for use by hw_params() + */ +static int at32_ssc_set_dai_fmt(struct snd_soc_dai *cpu_dai, + unsigned int fmt) +{ + struct at32_ssc_info *ssc_p = &ssc_info[cpu_dai->id]; + + ssc_p->daifmt = fmt; + return 0; +} + + + +/* + * Record SSC clock dividers for use in hw_params() + */ +static int at32_ssc_set_dai_clkdiv(struct snd_soc_dai *cpu_dai, + int div_id, int div) +{ + struct at32_ssc_info *ssc_p = &ssc_info[cpu_dai->id]; + + switch (div_id) { + case AT32_SSC_CMR_DIV: + /* + * The same master clock divider is used for both + * transmit and receive, so if a value has already + * been set, it must match this value + */ + if (ssc_p->cmr_div == 0) + ssc_p->cmr_div = div; + else if (div != ssc_p->cmr_div) + return -EBUSY; + break; + + case AT32_SSC_TCMR_PERIOD: + ssc_p->tcmr_period = div; + break; + + case AT32_SSC_RCMR_PERIOD: + ssc_p->rcmr_period = div; + break; + + default: + return -EINVAL; + } + + return 0; +} + + + +/* + * Configure the SSC + */ +static int at32_ssc_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + int id = rtd->dai->cpu_dai->id; + struct at32_ssc_info *ssc_p = &ssc_info[id]; + struct at32_pcm_dma_params *dma_params; + int channels, bits; + u32 tfmr, rfmr, tcmr, rcmr; + int start_event; + int ret; + + + /* + * Currently, there is only one set of dma_params for each direction. + * If more are added, this code will have to be changed to select + * the proper set + */ + dma_params = &ssc_dma_params[id][substream->stream]; + dma_params->ssc = ssc_p->ssc; + dma_params->substream = substream; + + ssc_p->dma_params[substream->stream] = dma_params; + + + /* + * The cpu_dai->dma_data field is only used to communicate the + * appropriate DMA parameters to the PCM driver's hw_params() + * function. It should not be used for other purposes as it + * is common to all substreams. + */ + rtd->dai->cpu_dai->dma_data = dma_params; + + channels = params_channels(params); + + + /* + * Determine sample size in bits and the PDC increment + */ + switch (params_format(params)) { + case SNDRV_PCM_FORMAT_S8: + bits = 8; + dma_params->pdc_xfer_size = 1; + break; + + case SNDRV_PCM_FORMAT_S16: + bits = 16; + dma_params->pdc_xfer_size = 2; + break; + + case SNDRV_PCM_FORMAT_S24: + bits = 24; + dma_params->pdc_xfer_size = 4; + break; + + case SNDRV_PCM_FORMAT_S32: + bits = 32; + dma_params->pdc_xfer_size = 4; + break; + + default: + pr_warning("at32-ssc: Unsupported PCM format %d", + params_format(params)); + return -EINVAL; + } + pr_debug("at32-ssc: bits = %d, pdc_xfer_size = %d, channels = %d\n", + bits, dma_params->pdc_xfer_size, channels); + + + /* + * The SSC only supports up to 16-bit samples in I2S format, due + * to the size of the Frame Mode Register FSLEN field. + */ + if ((ssc_p->daifmt & SND_SOC_DAIFMT_FORMAT_MASK) == SND_SOC_DAIFMT_I2S) + if (bits > 16) { + pr_warning("at32-ssc: " + "sample size %d is too large for I2S\n", + bits); + return -EINVAL; + } + + + /* + * Compute the SSC register settings + */ + switch (ssc_p->daifmt & (SND_SOC_DAIFMT_FORMAT_MASK | + SND_SOC_DAIFMT_MASTER_MASK)) { + case SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_CBS_CFS: + /* + * I2S format, SSC provides BCLK and LRS clocks. + * + * The SSC transmit and receive clocks are generated from the + * MCK divider, and the BCLK signal is output on the SSC TK line + */ + pr_debug("at32-ssc: SSC mode is I2S BCLK / FRAME master\n"); + rcmr = (SSC_BF(RCMR_PERIOD, ssc_p->rcmr_period) | + SSC_BF(RCMR_STTDLY, START_DELAY) | + SSC_BF(RCMR_START, SSC_START_FALLING_RF) | + SSC_BF(RCMR_CKI, SSC_CKI_RISING) | + SSC_BF(RCMR_CKO, SSC_CKO_NONE) | + SSC_BF(RCMR_CKS, SSC_CKS_DIV)); + + rfmr = (SSC_BF(RFMR_FSEDGE, SSC_FSEDGE_POSITIVE) | + SSC_BF(RFMR_FSOS, SSC_FSOS_NEGATIVE) | + SSC_BF(RFMR_FSLEN, bits - 1) | + SSC_BF(RFMR_DATNB, channels - 1) | + SSC_BIT(RFMR_MSBF) | SSC_BF(RFMR_DATLEN, bits - 1)); + + tcmr = (SSC_BF(TCMR_PERIOD, ssc_p->tcmr_period) | + SSC_BF(TCMR_STTDLY, START_DELAY) | + SSC_BF(TCMR_START, SSC_START_FALLING_RF) | + SSC_BF(TCMR_CKI, SSC_CKI_FALLING) | + SSC_BF(TCMR_CKO, SSC_CKO_CONTINUOUS) | + SSC_BF(TCMR_CKS, SSC_CKS_DIV)); + + tfmr = (SSC_BF(TFMR_FSEDGE, SSC_FSEDGE_POSITIVE) | + SSC_BF(TFMR_FSOS, SSC_FSOS_NEGATIVE) | + SSC_BF(TFMR_FSLEN, bits - 1) | + SSC_BF(TFMR_DATNB, channels - 1) | SSC_BIT(TFMR_MSBF) | + SSC_BF(TFMR_DATLEN, bits - 1)); + break; + + + case SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_CBM_CFM: + /* + * I2S format, CODEC supplies BCLK and LRC clock. + * + * The SSC transmit clock is obtained from the BCLK signal + * on the TK line, and the SSC receive clock is generated from + * the transmit clock. + * + * For single channel data, one sample is transferred on the + * falling edge of the LRC clock. For two channel data, one + * sample is transferred on both edges of the LRC clock. + */ + pr_debug("at32-ssc: SSC mode is I2S BCLK / FRAME slave\n"); + start_event = ((channels == 1) ? + SSC_START_FALLING_RF : SSC_START_EDGE_RF); + + rcmr = (SSC_BF(RCMR_STTDLY, START_DELAY) | + SSC_BF(RCMR_START, start_event) | + SSC_BF(RCMR_CKI, SSC_CKI_RISING) | + SSC_BF(RCMR_CKO, SSC_CKO_NONE) | + SSC_BF(RCMR_CKS, SSC_CKS_CLOCK)); + + rfmr = (SSC_BF(RFMR_FSEDGE, SSC_FSEDGE_POSITIVE) | + SSC_BF(RFMR_FSOS, SSC_FSOS_NONE) | + SSC_BIT(RFMR_MSBF) | SSC_BF(RFMR_DATLEN, bits - 1)); + + tcmr = (SSC_BF(TCMR_STTDLY, START_DELAY) | + SSC_BF(TCMR_START, start_event) | + SSC_BF(TCMR_CKI, SSC_CKI_FALLING) | + SSC_BF(TCMR_CKO, SSC_CKO_NONE) | + SSC_BF(TCMR_CKS, SSC_CKS_PIN)); + + tfmr = (SSC_BF(TFMR_FSEDGE, SSC_FSEDGE_POSITIVE) | + SSC_BF(TFMR_FSOS, SSC_FSOS_NONE) | + SSC_BIT(TFMR_MSBF) | SSC_BF(TFMR_DATLEN, bits - 1)); + break; + + + case SND_SOC_DAIFMT_DSP_A | SND_SOC_DAIFMT_CBS_CFS: + /* + * DSP/PCM Mode A format, SSC provides BCLK and LRC clocks. + * + * The SSC transmit and receive clocks are generated from the + * MCK divider, and the BCLK signal is output on the SSC TK line + */ + pr_debug("at32-ssc: SSC mode is DSP A BCLK / FRAME master\n"); + rcmr = (SSC_BF(RCMR_PERIOD, ssc_p->rcmr_period) | + SSC_BF(RCMR_STTDLY, 1) | + SSC_BF(RCMR_START, SSC_START_RISING_RF) | + SSC_BF(RCMR_CKI, SSC_CKI_RISING) | + SSC_BF(RCMR_CKO, SSC_CKO_NONE) | + SSC_BF(RCMR_CKS, SSC_CKS_DIV)); + + rfmr = (SSC_BF(RFMR_FSEDGE, SSC_FSEDGE_POSITIVE) | + SSC_BF(RFMR_FSOS, SSC_FSOS_POSITIVE) | + SSC_BF(RFMR_DATNB, channels - 1) | + SSC_BIT(RFMR_MSBF) | SSC_BF(RFMR_DATLEN, bits - 1)); + + tcmr = (SSC_BF(TCMR_PERIOD, ssc_p->tcmr_period) | + SSC_BF(TCMR_STTDLY, 1) | + SSC_BF(TCMR_START, SSC_START_RISING_RF) | + SSC_BF(TCMR_CKI, SSC_CKI_RISING) | + SSC_BF(TCMR_CKO, SSC_CKO_CONTINUOUS) | + SSC_BF(TCMR_CKS, SSC_CKS_DIV)); + + tfmr = (SSC_BF(TFMR_FSEDGE, SSC_FSEDGE_POSITIVE) | + SSC_BF(TFMR_FSOS, SSC_FSOS_POSITIVE) | + SSC_BF(TFMR_DATNB, channels - 1) | + SSC_BIT(TFMR_MSBF) | SSC_BF(TFMR_DATLEN, bits - 1)); + break; + + + case SND_SOC_DAIFMT_DSP_A | SND_SOC_DAIFMT_CBM_CFM: + default: + pr_warning("at32-ssc: unsupported DAI format 0x%x\n", + ssc_p->daifmt); + return -EINVAL; + break; + } + pr_debug("at32-ssc: RCMR=%08x RFMR=%08x TCMR=%08x TFMR=%08x\n", + rcmr, rfmr, tcmr, tfmr); + + + if (!ssc_p->initialized) { + /* enable peripheral clock */ + pr_debug("at32-ssc: Starting clock\n"); + clk_enable(ssc_p->ssc->clk); + + /* Reset the SSC and its PDC registers */ + ssc_writel(ssc_p->ssc->regs, CR, SSC_BIT(CR_SWRST)); + + ssc_writel(ssc_p->ssc->regs, PDC_RPR, 0); + ssc_writel(ssc_p->ssc->regs, PDC_RCR, 0); + ssc_writel(ssc_p->ssc->regs, PDC_RNPR, 0); + ssc_writel(ssc_p->ssc->regs, PDC_RNCR, 0); + + ssc_writel(ssc_p->ssc->regs, PDC_TPR, 0); + ssc_writel(ssc_p->ssc->regs, PDC_TCR, 0); + ssc_writel(ssc_p->ssc->regs, PDC_TNPR, 0); + ssc_writel(ssc_p->ssc->regs, PDC_TNCR, 0); + + ret = request_irq(ssc_p->ssc->irq, at32_ssc_interrupt, 0, + ssc_p->name, ssc_p); + if (ret < 0) { + pr_warning("at32-ssc: request irq failed (%d)\n", ret); + pr_debug("at32-ssc: Stopping clock\n"); + clk_disable(ssc_p->ssc->clk); + return ret; + } + + ssc_p->initialized = 1; + } + + /* Set SSC clock mode register */ + ssc_writel(ssc_p->ssc->regs, CMR, ssc_p->cmr_div); + + /* set receive clock mode and format */ + ssc_writel(ssc_p->ssc->regs, RCMR, rcmr); + ssc_writel(ssc_p->ssc->regs, RFMR, rfmr); + + /* set transmit clock mode and format */ + ssc_writel(ssc_p->ssc->regs, TCMR, tcmr); + ssc_writel(ssc_p->ssc->regs, TFMR, tfmr); + + pr_debug("at32-ssc: SSC initialized\n"); + return 0; +} + + + +static int at32_ssc_prepare(struct snd_pcm_substream *substream) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct at32_ssc_info *ssc_p = &ssc_info[rtd->dai->cpu_dai->id]; + struct at32_pcm_dma_params *dma_params; + + dma_params = ssc_p->dma_params[substream->stream]; + + ssc_writel(dma_params->ssc->regs, CR, dma_params->mask->ssc_enable); + + return 0; +} + + + +#ifdef CONFIG_PM +static int at32_ssc_suspend(struct platform_device *pdev, + struct snd_soc_dai *cpu_dai) +{ + struct at32_ssc_info *ssc_p; + + if (!cpu_dai->active) + return 0; + + ssc_p = &ssc_info[cpu_dai->id]; + + /* Save the status register before disabling transmit and receive */ + ssc_p->ssc_state.ssc_sr = ssc_readl(ssc_p->ssc->regs, SR); + ssc_writel(ssc_p->ssc->regs, CR, SSC_BIT(CR_TXDIS) | SSC_BIT(CR_RXDIS)); + + /* Save the current interrupt mask, then disable unmasked interrupts */ + ssc_p->ssc_state.ssc_imr = ssc_readl(ssc_p->ssc->regs, IMR); + ssc_writel(ssc_p->ssc->regs, IDR, ssc_p->ssc_state.ssc_imr); + + ssc_p->ssc_state.ssc_cmr = ssc_readl(ssc_p->ssc->regs, CMR); + ssc_p->ssc_state.ssc_rcmr = ssc_readl(ssc_p->ssc->regs, RCMR); + ssc_p->ssc_state.ssc_rfmr = ssc_readl(ssc_p->ssc->regs, RFMR); + ssc_p->ssc_state.ssc_tcmr = ssc_readl(ssc_p->ssc->regs, TCMR); + ssc_p->ssc_state.ssc_tfmr = ssc_readl(ssc_p->ssc->regs, TFMR); + + return 0; +} + + + +static int at32_ssc_resume(struct platform_device *pdev, + struct snd_soc_dai *cpu_dai) +{ + struct at32_ssc_info *ssc_p; + u32 cr; + + if (!cpu_dai->active) + return 0; + + ssc_p = &ssc_info[cpu_dai->id]; + + /* restore SSC register settings */ + ssc_writel(ssc_p->ssc->regs, TFMR, ssc_p->ssc_state.ssc_tfmr); + ssc_writel(ssc_p->ssc->regs, TCMR, ssc_p->ssc_state.ssc_tcmr); + ssc_writel(ssc_p->ssc->regs, RFMR, ssc_p->ssc_state.ssc_rfmr); + ssc_writel(ssc_p->ssc->regs, RCMR, ssc_p->ssc_state.ssc_rcmr); + ssc_writel(ssc_p->ssc->regs, CMR, ssc_p->ssc_state.ssc_cmr); + + /* re-enable interrupts */ + ssc_writel(ssc_p->ssc->regs, IER, ssc_p->ssc_state.ssc_imr); + + /* Re-enable recieve and transmit as appropriate */ + cr = 0; + cr |= + (ssc_p->ssc_state.ssc_sr & SSC_BIT(SR_RXEN)) ? SSC_BIT(CR_RXEN) : 0; + cr |= + (ssc_p->ssc_state.ssc_sr & SSC_BIT(SR_TXEN)) ? SSC_BIT(CR_TXEN) : 0; + ssc_writel(ssc_p->ssc->regs, CR, cr); + + return 0; +} +#else /* CONFIG_PM */ +# define at32_ssc_suspend NULL +# define at32_ssc_resume NULL +#endif /* CONFIG_PM */ + + +#define AT32_SSC_RATES \ + (SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_11025 | SNDRV_PCM_RATE_16000 | \ + SNDRV_PCM_RATE_22050 | SNDRV_PCM_RATE_32000 | SNDRV_PCM_RATE_44100 | \ + SNDRV_PCM_RATE_48000 | SNDRV_PCM_RATE_88200 | SNDRV_PCM_RATE_96000) + + +#define AT32_SSC_FORMATS \ + (SNDRV_PCM_FMTBIT_S8 | SNDRV_PCM_FMTBIT_S16 | \ + SNDRV_PCM_FMTBIT_S24 | SNDRV_PCM_FMTBIT_S32) + + +struct snd_soc_dai at32_ssc_dai[NUM_SSC_DEVICES] = { + { + .name = "at32-ssc0", + .id = 0, + .type = SND_SOC_DAI_PCM, + .suspend = at32_ssc_suspend, + .resume = at32_ssc_resume, + .playback = { + .channels_min = 1, + .channels_max = 2, + .rates = AT32_SSC_RATES, + .formats = AT32_SSC_FORMATS, + }, + .capture = { + .channels_min = 1, + .channels_max = 2, + .rates = AT32_SSC_RATES, + .formats = AT32_SSC_FORMATS, + }, + .ops = { + .startup = at32_ssc_startup, + .shutdown = at32_ssc_shutdown, + .prepare = at32_ssc_prepare, + .hw_params = at32_ssc_hw_params, + }, + .dai_ops = { + .set_sysclk = at32_ssc_set_dai_sysclk, + .set_fmt = at32_ssc_set_dai_fmt, + .set_clkdiv = at32_ssc_set_dai_clkdiv, + }, + .private_data = &ssc_info[0], + }, + { + .name = "at32-ssc1", + .id = 1, + .type = SND_SOC_DAI_PCM, + .suspend = at32_ssc_suspend, + .resume = at32_ssc_resume, + .playback = { + .channels_min = 1, + .channels_max = 2, + .rates = AT32_SSC_RATES, + .formats = AT32_SSC_FORMATS, + }, + .capture = { + .channels_min = 1, + .channels_max = 2, + .rates = AT32_SSC_RATES, + .formats = AT32_SSC_FORMATS, + }, + .ops = { + .startup = at32_ssc_startup, + .shutdown = at32_ssc_shutdown, + .prepare = at32_ssc_prepare, + .hw_params = at32_ssc_hw_params, + }, + .dai_ops = { + .set_sysclk = at32_ssc_set_dai_sysclk, + .set_fmt = at32_ssc_set_dai_fmt, + .set_clkdiv = at32_ssc_set_dai_clkdiv, + }, + .private_data = &ssc_info[1], + }, + { + .name = "at32-ssc2", + .id = 2, + .type = SND_SOC_DAI_PCM, + .suspend = at32_ssc_suspend, + .resume = at32_ssc_resume, + .playback = { + .channels_min = 1, + .channels_max = 2, + .rates = AT32_SSC_RATES, + .formats = AT32_SSC_FORMATS, + }, + .capture = { + .channels_min = 1, + .channels_max = 2, + .rates = AT32_SSC_RATES, + .formats = AT32_SSC_FORMATS, + }, + .ops = { + .startup = at32_ssc_startup, + .shutdown = at32_ssc_shutdown, + .prepare = at32_ssc_prepare, + .hw_params = at32_ssc_hw_params, + }, + .dai_ops = { + .set_sysclk = at32_ssc_set_dai_sysclk, + .set_fmt = at32_ssc_set_dai_fmt, + .set_clkdiv = at32_ssc_set_dai_clkdiv, + }, + .private_data = &ssc_info[2], + }, +}; +EXPORT_SYMBOL_GPL(at32_ssc_dai); + + +MODULE_AUTHOR("Geoffrey Wossum <gwossum@acm.org>"); +MODULE_DESCRIPTION("AT32 SSC ASoC Interface"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/at32/at32-ssc.h b/sound/soc/at32/at32-ssc.h new file mode 100644 index 00000000000..3c052dbbe46 --- /dev/null +++ b/sound/soc/at32/at32-ssc.h @@ -0,0 +1,59 @@ +/* sound/soc/at32/at32-ssc.h + * ASoC SSC interface for Atmel AT32 SoC + * + * Copyright (C) 2008 Long Range Systems + * Geoffrey Wossum <gwossum@acm.org> + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. + */ + +#ifndef __SOUND_SOC_AT32_AT32_SSC_H +#define __SOUND_SOC_AT32_AT32_SSC_H __FILE__ + +#include <linux/types.h> +#include <linux/atmel-ssc.h> + +#include "at32-pcm.h" + + + +struct at32_ssc_state { + u32 ssc_cmr; + u32 ssc_rcmr; + u32 ssc_rfmr; + u32 ssc_tcmr; + u32 ssc_tfmr; + u32 ssc_sr; + u32 ssc_imr; +}; + + + +struct at32_ssc_info { + char *name; + struct ssc_device *ssc; + spinlock_t lock; /* lock for dir_mask */ + unsigned short dir_mask; /* 0=unused, 1=playback, 2=capture */ + unsigned short initialized; /* true if SSC has been initialized */ + unsigned short daifmt; + unsigned short cmr_div; + unsigned short tcmr_period; + unsigned short rcmr_period; + struct at32_pcm_dma_params *dma_params[2]; + struct at32_ssc_state ssc_state; +}; + + +/* SSC divider ids */ +#define AT32_SSC_CMR_DIV 0 /* MCK divider for BCLK */ +#define AT32_SSC_TCMR_PERIOD 1 /* BCLK divider for transmit FS */ +#define AT32_SSC_RCMR_PERIOD 2 /* BCLK divider for receive FS */ + + +extern struct snd_soc_dai at32_ssc_dai[]; + + + +#endif /* __SOUND_SOC_AT32_AT32_SSC_H */ diff --git a/sound/soc/at32/playpaq_wm8510.c b/sound/soc/at32/playpaq_wm8510.c new file mode 100644 index 00000000000..fee5f8e5895 --- /dev/null +++ b/sound/soc/at32/playpaq_wm8510.c @@ -0,0 +1,522 @@ +/* sound/soc/at32/playpaq_wm8510.c + * ASoC machine driver for PlayPaq using WM8510 codec + * + * Copyright (C) 2008 Long Range Systems + * Geoffrey Wossum <gwossum@acm.org> + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. + * + * This code is largely inspired by sound/soc/at91/eti_b1_wm8731.c + * + * NOTE: If you don't have the AT32 enhanced portmux configured (which + * isn't currently in the mainline or Atmel patched kernel), you will + * need to set the MCLK pin (PA30) to peripheral A in your board initialization + * code. Something like: + * at32_select_periph(GPIO_PIN_PA(30), GPIO_PERIPH_A, 0); + * + */ + +/* #define DEBUG */ + +#include <linux/module.h> +#include <linux/moduleparam.h> +#include <linux/version.h> +#include <linux/kernel.h> +#include <linux/errno.h> +#include <linux/clk.h> +#include <linux/timer.h> +#include <linux/interrupt.h> +#include <linux/platform_device.h> + +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> +#include <sound/soc-dapm.h> + +#include <asm/arch/at32ap700x.h> +#include <asm/arch/portmux.h> + +#include "../codecs/wm8510.h" +#include "at32-pcm.h" +#include "at32-ssc.h" + + +/*-------------------------------------------------------------------------*\ + * constants +\*-------------------------------------------------------------------------*/ +#define MCLK_PIN GPIO_PIN_PA(30) +#define MCLK_PERIPH GPIO_PERIPH_A + + +/*-------------------------------------------------------------------------*\ + * data types +\*-------------------------------------------------------------------------*/ +/* SSC clocking data */ +struct ssc_clock_data { + /* CMR div */ + unsigned int cmr_div; + + /* Frame period (as needed by xCMR.PERIOD) */ + unsigned int period; + + /* The SSC clock rate these settings where calculated for */ + unsigned long ssc_rate; +}; + + +/*-------------------------------------------------------------------------*\ + * module data +\*-------------------------------------------------------------------------*/ +static struct clk *_gclk0; +static struct clk *_pll0; + +#define CODEC_CLK (_gclk0) + + +/*-------------------------------------------------------------------------*\ + * Sound SOC operations +\*-------------------------------------------------------------------------*/ +#if defined CONFIG_SND_AT32_SOC_PLAYPAQ_SLAVE +static struct ssc_clock_data playpaq_wm8510_calc_ssc_clock( + struct snd_pcm_hw_params *params, + struct snd_soc_dai *cpu_dai) +{ + struct at32_ssc_info *ssc_p = cpu_dai->private_data; + struct ssc_device *ssc = ssc_p->ssc; + struct ssc_clock_data cd; + unsigned int rate, width_bits, channels; + unsigned int bitrate, ssc_div; + unsigned actual_rate; + + + /* + * Figure out required bitrate + */ + rate = params_rate(params); + channels = params_channels(params); + width_bits = snd_pcm_format_physical_width(params_format(params)); + bitrate = rate * width_bits * channels; + + + /* + * Figure out required SSC divider and period for required bitrate + */ + cd.ssc_rate = clk_get_rate(ssc->clk); + ssc_div = cd.ssc_rate / bitrate; + cd.cmr_div = ssc_div / 2; + if (ssc_div & 1) { + /* round cmr_div up */ + cd.cmr_div++; + } + cd.period = width_bits - 1; + + + /* + * Find actual rate, compare to requested rate + */ + actual_rate = (cd.ssc_rate / (cd.cmr_div * 2)) / (2 * (cd.period + 1)); + pr_debug("playpaq_wm8510: Request rate = %d, actual rate = %d\n", + rate, actual_rate); + + + return cd; +} +#endif /* CONFIG_SND_AT32_SOC_PLAYPAQ_SLAVE */ + + + +static int playpaq_wm8510_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_dai *codec_dai = rtd->dai->codec_dai; + struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai; + struct at32_ssc_info *ssc_p = cpu_dai->private_data; + struct ssc_device *ssc = ssc_p->ssc; + unsigned int pll_out = 0, bclk = 0, mclk_div = 0; + int ret; + + + /* Due to difficulties with getting the correct clocks from the AT32's + * PLL0, we're going to let the CODEC be in charge of all the clocks + */ +#if !defined CONFIG_SND_AT32_SOC_PLAYPAQ_SLAVE + const unsigned int fmt = (SND_SOC_DAIFMT_I2S | + SND_SOC_DAIFMT_NB_NF | + SND_SOC_DAIFMT_CBM_CFM); +#else + struct ssc_clock_data cd; + const unsigned int fmt = (SND_SOC_DAIFMT_I2S | + SND_SOC_DAIFMT_NB_NF | + SND_SOC_DAIFMT_CBS_CFS); +#endif + + if (ssc == NULL) { + pr_warning("playpaq_wm8510_hw_params: ssc is NULL!\n"); + return -EINVAL; + } + + + /* + * Figure out PLL and BCLK dividers for WM8510 + */ + switch (params_rate(params)) { + case 48000: + pll_out = 12288000; + mclk_div = WM8510_MCLKDIV_1; + bclk = WM8510_BCLKDIV_8; + break; + + case 44100: + pll_out = 11289600; + mclk_div = WM8510_MCLKDIV_1; + bclk = WM8510_BCLKDIV_8; + break; + + case 22050: + pll_out = 11289600; + mclk_div = WM8510_MCLKDIV_2; + bclk = WM8510_BCLKDIV_8; + break; + + case 16000: + pll_out = 12288000; + mclk_div = WM8510_MCLKDIV_3; + bclk = WM8510_BCLKDIV_8; + break; + + case 11025: + pll_out = 11289600; + mclk_div = WM8510_MCLKDIV_4; + bclk = WM8510_BCLKDIV_8; + break; + + case 8000: + pll_out = 12288000; + mclk_div = WM8510_MCLKDIV_6; + bclk = WM8510_BCLKDIV_8; + break; + + default: + pr_warning("playpaq_wm8510: Unsupported sample rate %d\n", + params_rate(params)); + return -EINVAL; + } + + + /* + * set CPU and CODEC DAI configuration + */ + ret = snd_soc_dai_set_fmt(codec_dai, fmt); + if (ret < 0) { + pr_warning("playpaq_wm8510: " + "Failed to set CODEC DAI format (%d)\n", + ret); + return ret; + } + ret = snd_soc_dai_set_fmt(cpu_dai, fmt); + if (ret < 0) { + pr_warning("playpaq_wm8510: " + "Failed to set CPU DAI format (%d)\n", + ret); + return ret; + } + + + /* + * Set CPU clock configuration + */ +#if defined CONFIG_SND_AT32_SOC_PLAYPAQ_SLAVE + cd = playpaq_wm8510_calc_ssc_clock(params, cpu_dai); + pr_debug("playpaq_wm8510: cmr_div = %d, period = %d\n", + cd.cmr_div, cd.period); + ret = snd_soc_dai_set_clkdiv(cpu_dai, AT32_SSC_CMR_DIV, cd.cmr_div); + if (ret < 0) { + pr_warning("playpaq_wm8510: Failed to set CPU CMR_DIV (%d)\n", + ret); + return ret; + } + ret = snd_soc_dai_set_clkdiv(cpu_dai, AT32_SSC_TCMR_PERIOD, + cd.period); + if (ret < 0) { + pr_warning("playpaq_wm8510: " + "Failed to set CPU transmit period (%d)\n", + ret); + return ret; + } +#endif /* CONFIG_SND_AT32_SOC_PLAYPAQ_SLAVE */ + + + /* + * Set CODEC clock configuration + */ + pr_debug("playpaq_wm8510: " + "pll_in = %ld, pll_out = %u, bclk = %x, mclk = %x\n", + clk_get_rate(CODEC_CLK), pll_out, bclk, mclk_div); + + +#if !defined CONFIG_SND_AT32_SOC_PLAYPAQ_SLAVE + ret = snd_soc_dai_set_clkdiv(codec_dai, WM8510_BCLKDIV, bclk); + if (ret < 0) { + pr_warning + ("playpaq_wm8510: Failed to set CODEC DAI BCLKDIV (%d)\n", + ret); + return ret; + } +#endif /* CONFIG_SND_AT32_SOC_PLAYPAQ_SLAVE */ + + + ret = snd_soc_dai_set_pll(codec_dai, 0, + clk_get_rate(CODEC_CLK), pll_out); + if (ret < 0) { + pr_warning("playpaq_wm8510: Failed to set CODEC DAI PLL (%d)\n", + ret); + return ret; + } + + + ret = snd_soc_dai_set_clkdiv(codec_dai, WM8510_MCLKDIV, mclk_div); + if (ret < 0) { + pr_warning("playpaq_wm8510: Failed to set CODEC MCLKDIV (%d)\n", + ret); + return ret; + } + + + return 0; +} + + + +static struct snd_soc_ops playpaq_wm8510_ops = { + .hw_params = playpaq_wm8510_hw_params, +}; + + + +static const struct snd_soc_dapm_widget playpaq_dapm_widgets[] = { + SND_SOC_DAPM_MIC("Int Mic", NULL), + SND_SOC_DAPM_SPK("Ext Spk", NULL), +}; + + + +static const char *intercon[][3] = { + /* speaker connected to SPKOUT */ + {"Ext Spk", NULL, "SPKOUTP"}, + {"Ext Spk", NULL, "SPKOUTN"}, + + {"Mic Bias", NULL, "Int Mic"}, + {"MICN", NULL, "Mic Bias"}, + {"MICP", NULL, "Mic Bias"}, + + /* Terminator */ + {NULL, NULL, NULL}, +}; + + + +static int playpaq_wm8510_init(struct snd_soc_codec *codec) +{ + int i; + + /* + * Add DAPM widgets + */ + for (i = 0; i < ARRAY_SIZE(playpaq_dapm_widgets); i++) + snd_soc_dapm_new_control(codec, &playpaq_dapm_widgets[i]); + + + + /* + * Setup audio path interconnects + */ + for (i = 0; intercon[i][0] != NULL; i++) { + snd_soc_dapm_connect_input(codec, + intercon[i][0], + intercon[i][1], intercon[i][2]); + } + + + /* always connected pins */ + snd_soc_dapm_enable_pin(codec, "Int Mic"); + snd_soc_dapm_enable_pin(codec, "Ext Spk"); + snd_soc_dapm_sync(codec); + + + + /* Make CSB show PLL rate */ + snd_soc_dai_set_clkdiv(codec->dai, WM8510_OPCLKDIV, + WM8510_OPCLKDIV_1 | 4); + + return 0; +} + + + +static struct snd_soc_dai_link playpaq_wm8510_dai = { + .name = "WM8510", + .stream_name = "WM8510 PCM", + .cpu_dai = &at32_ssc_dai[0], + .codec_dai = &wm8510_dai, + .init = playpaq_wm8510_init, + .ops = &playpaq_wm8510_ops, +}; + + + +static struct snd_soc_machine snd_soc_machine_playpaq = { + .name = "LRS_PlayPaq_WM8510", + .dai_link = &playpaq_wm8510_dai, + .num_links = 1, +}; + + + +static struct wm8510_setup_data playpaq_wm8510_setup = { + .i2c_address = 0x1a, +}; + + + +static struct snd_soc_device playpaq_wm8510_snd_devdata = { + .machine = &snd_soc_machine_playpaq, + .platform = &at32_soc_platform, + .codec_dev = &soc_codec_dev_wm8510, + .codec_data = &playpaq_wm8510_setup, +}; + +static struct platform_device *playpaq_snd_device; + + +static int __init playpaq_asoc_init(void) +{ + int ret = 0; + struct at32_ssc_info *ssc_p = playpaq_wm8510_dai.cpu_dai->private_data; + struct ssc_device *ssc = NULL; + + + /* + * Request SSC device + */ + ssc = ssc_request(0); + if (IS_ERR(ssc)) { + ret = PTR_ERR(ssc); + ssc = NULL; + goto err_ssc; + } + ssc_p->ssc = ssc; + + + /* + * Configure MCLK for WM8510 + */ + _gclk0 = clk_get(NULL, "gclk0"); + if (IS_ERR(_gclk0)) { + _gclk0 = NULL; + goto err_gclk0; + } + _pll0 = clk_get(NULL, "pll0"); + if (IS_ERR(_pll0)) { + _pll0 = NULL; + goto err_pll0; + } + if (clk_set_parent(_gclk0, _pll0)) { + pr_warning("snd-soc-playpaq: " + "Failed to set PLL0 as parent for DAC clock\n"); + goto err_set_clk; + } + clk_set_rate(CODEC_CLK, 12000000); + clk_enable(CODEC_CLK); + +#if defined CONFIG_AT32_ENHANCED_PORTMUX + at32_select_periph(MCLK_PIN, MCLK_PERIPH, 0); +#endif + + + /* + * Create and register platform device + */ + playpaq_snd_device = platform_device_alloc("soc-audio", 0); + if (playpaq_snd_device == NULL) { + ret = -ENOMEM; + goto err_device_alloc; + } + + platform_set_drvdata(playpaq_snd_device, &playpaq_wm8510_snd_devdata); + playpaq_wm8510_snd_devdata.dev = &playpaq_snd_device->dev; + + ret = platform_device_add(playpaq_snd_device); + if (ret) { + pr_warning("playpaq_wm8510: platform_device_add failed (%d)\n", + ret); + goto err_device_add; + } + + return 0; + + +err_device_add: + if (playpaq_snd_device != NULL) { + platform_device_put(playpaq_snd_device); + playpaq_snd_device = NULL; + } +err_device_alloc: +err_set_clk: + if (_pll0 != NULL) { + clk_put(_pll0); + _pll0 = NULL; + } +err_pll0: + if (_gclk0 != NULL) { + clk_put(_gclk0); + _gclk0 = NULL; + } +err_gclk0: + if (ssc != NULL) { + ssc_free(ssc); + ssc = NULL; + } +err_ssc: + return ret; +} + + +static void __exit playpaq_asoc_exit(void) +{ + struct at32_ssc_info *ssc_p = playpaq_wm8510_dai.cpu_dai->private_data; + struct ssc_device *ssc; + + if (ssc_p != NULL) { + ssc = ssc_p->ssc; + if (ssc != NULL) + ssc_free(ssc); + ssc_p->ssc = NULL; + } + + if (_gclk0 != NULL) { + clk_put(_gclk0); + _gclk0 = NULL; + } + if (_pll0 != NULL) { + clk_put(_pll0); + _pll0 = NULL; + } + +#if defined CONFIG_AT32_ENHANCED_PORTMUX + at32_free_pin(MCLK_PIN); +#endif + + platform_device_unregister(playpaq_snd_device); + playpaq_snd_device = NULL; +} + +module_init(playpaq_asoc_init); +module_exit(playpaq_asoc_exit); + +MODULE_AUTHOR("Geoffrey Wossum <gwossum@acm.org>"); +MODULE_DESCRIPTION("ASoC machine driver for LRS PlayPaq"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/at91/Kconfig b/sound/soc/at91/Kconfig index 5cb93fd3a40..905186502e0 100644 --- a/sound/soc/at91/Kconfig +++ b/sound/soc/at91/Kconfig @@ -1,6 +1,6 @@ config SND_AT91_SOC tristate "SoC Audio for the Atmel AT91 System-on-Chip" - depends on ARCH_AT91 && SND_SOC + depends on ARCH_AT91 help Say Y or M if you want to add support for codecs attached to the AT91 SSC interface. You will also need diff --git a/sound/soc/at91/at91-pcm.c b/sound/soc/at91/at91-pcm.c index ccac6bd2889..d47492b2b6e 100644 --- a/sound/soc/at91/at91-pcm.c +++ b/sound/soc/at91/at91-pcm.c @@ -318,7 +318,7 @@ static int at91_pcm_preallocate_dma_buffer(struct snd_pcm *pcm, static u64 at91_pcm_dmamask = 0xffffffff; static int at91_pcm_new(struct snd_card *card, - struct snd_soc_codec_dai *dai, struct snd_pcm *pcm) + struct snd_soc_dai *dai, struct snd_pcm *pcm) { int ret = 0; @@ -367,7 +367,7 @@ static void at91_pcm_free_dma_buffers(struct snd_pcm *pcm) #ifdef CONFIG_PM static int at91_pcm_suspend(struct platform_device *pdev, - struct snd_soc_cpu_dai *dai) + struct snd_soc_dai *dai) { struct snd_pcm_runtime *runtime = dai->runtime; struct at91_runtime_data *prtd; @@ -392,7 +392,7 @@ static int at91_pcm_suspend(struct platform_device *pdev, } static int at91_pcm_resume(struct platform_device *pdev, - struct snd_soc_cpu_dai *dai) + struct snd_soc_dai *dai) { struct snd_pcm_runtime *runtime = dai->runtime; struct at91_runtime_data *prtd; diff --git a/sound/soc/at91/at91-ssc.c b/sound/soc/at91/at91-ssc.c index bc35d00a38f..c3625b665c5 100644 --- a/sound/soc/at91/at91-ssc.c +++ b/sound/soc/at91/at91-ssc.c @@ -281,7 +281,7 @@ static void at91_ssc_shutdown(struct snd_pcm_substream *substream) /* * Record the SSC system clock rate. */ -static int at91_ssc_set_dai_sysclk(struct snd_soc_cpu_dai *cpu_dai, +static int at91_ssc_set_dai_sysclk(struct snd_soc_dai *cpu_dai, int clk_id, unsigned int freq, int dir) { /* @@ -303,7 +303,7 @@ static int at91_ssc_set_dai_sysclk(struct snd_soc_cpu_dai *cpu_dai, /* * Record the DAI format for use in hw_params(). */ -static int at91_ssc_set_dai_fmt(struct snd_soc_cpu_dai *cpu_dai, +static int at91_ssc_set_dai_fmt(struct snd_soc_dai *cpu_dai, unsigned int fmt) { struct at91_ssc_info *ssc_p = &ssc_info[cpu_dai->id]; @@ -315,7 +315,7 @@ static int at91_ssc_set_dai_fmt(struct snd_soc_cpu_dai *cpu_dai, /* * Record SSC clock dividers for use in hw_params(). */ -static int at91_ssc_set_dai_clkdiv(struct snd_soc_cpu_dai *cpu_dai, +static int at91_ssc_set_dai_clkdiv(struct snd_soc_dai *cpu_dai, int div_id, int div) { struct at91_ssc_info *ssc_p = &ssc_info[cpu_dai->id]; @@ -634,7 +634,7 @@ static int at91_ssc_prepare(struct snd_pcm_substream *substream) #ifdef CONFIG_PM static int at91_ssc_suspend(struct platform_device *pdev, - struct snd_soc_cpu_dai *cpu_dai) + struct snd_soc_dai *cpu_dai) { struct at91_ssc_info *ssc_p; @@ -662,7 +662,7 @@ static int at91_ssc_suspend(struct platform_device *pdev, } static int at91_ssc_resume(struct platform_device *pdev, - struct snd_soc_cpu_dai *cpu_dai) + struct snd_soc_dai *cpu_dai) { struct at91_ssc_info *ssc_p; @@ -700,7 +700,7 @@ static int at91_ssc_resume(struct platform_device *pdev, #define AT91_SSC_FORMATS (SNDRV_PCM_FMTBIT_S8 | SNDRV_PCM_FMTBIT_S16_LE |\ SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_S32_LE) -struct snd_soc_cpu_dai at91_ssc_dai[NUM_SSC_DEVICES] = { +struct snd_soc_dai at91_ssc_dai[NUM_SSC_DEVICES] = { { .name = "at91-ssc0", .id = 0, .type = SND_SOC_DAI_PCM, diff --git a/sound/soc/at91/at91-ssc.h b/sound/soc/at91/at91-ssc.h index b188f973df9..6b7bf382d06 100644 --- a/sound/soc/at91/at91-ssc.h +++ b/sound/soc/at91/at91-ssc.h @@ -21,7 +21,7 @@ #define AT91SSC_TCMR_PERIOD 1 /* BCLK divider for transmit FS */ #define AT91SSC_RCMR_PERIOD 2 /* BCLK divider for receive FS */ -extern struct snd_soc_cpu_dai at91_ssc_dai[]; +extern struct snd_soc_dai at91_ssc_dai[]; #endif /* _AT91_SSC_H */ diff --git a/sound/soc/at91/eti_b1_wm8731.c b/sound/soc/at91/eti_b1_wm8731.c index 1347dcf3f80..d532de95424 100644 --- a/sound/soc/at91/eti_b1_wm8731.c +++ b/sound/soc/at91/eti_b1_wm8731.c @@ -53,18 +53,18 @@ static struct clk *pllb_clk; static int eti_b1_startup(struct snd_pcm_substream *substream) { struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_codec_dai *codec_dai = rtd->dai->codec_dai; - struct snd_soc_cpu_dai *cpu_dai = rtd->dai->cpu_dai; + struct snd_soc_dai *codec_dai = rtd->dai->codec_dai; + struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai; int ret; /* cpu clock is the AT91 master clock sent to the SSC */ - ret = cpu_dai->dai_ops.set_sysclk(cpu_dai, AT91_SYSCLK_MCK, + ret = snd_soc_dai_set_sysclk(cpu_dai, AT91_SYSCLK_MCK, 60000000, SND_SOC_CLOCK_IN); if (ret < 0) return ret; /* codec system clock is supplied by PCK1, set to 12MHz */ - ret = codec_dai->dai_ops.set_sysclk(codec_dai, WM8731_SYSCLK, + ret = snd_soc_dai_set_sysclk(codec_dai, WM8731_SYSCLK, 12000000, SND_SOC_CLOCK_IN); if (ret < 0) return ret; @@ -87,8 +87,8 @@ static int eti_b1_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params) { struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_codec_dai *codec_dai = rtd->dai->codec_dai; - struct snd_soc_cpu_dai *cpu_dai = rtd->dai->cpu_dai; + struct snd_soc_dai *codec_dai = rtd->dai->codec_dai; + struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai; int ret; #ifdef CONFIG_SND_AT91_SOC_ETI_SLAVE @@ -96,13 +96,13 @@ static int eti_b1_hw_params(struct snd_pcm_substream *substream, int cmr_div, period; /* set codec DAI configuration */ - ret = codec_dai->dai_ops.set_fmt(codec_dai, SND_SOC_DAIFMT_I2S | + ret = snd_soc_dai_set_fmt(codec_dai, SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBS_CFS); if (ret < 0) return ret; /* set cpu DAI configuration */ - ret = cpu_dai->dai_ops.set_fmt(cpu_dai, SND_SOC_DAIFMT_I2S | + ret = snd_soc_dai_set_fmt(cpu_dai, SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBS_CFS); if (ret < 0) return ret; @@ -141,17 +141,17 @@ static int eti_b1_hw_params(struct snd_pcm_substream *substream, } /* set the MCK divider for BCLK */ - ret = cpu_dai->dai_ops.set_clkdiv(cpu_dai, AT91SSC_CMR_DIV, cmr_div); + ret = snd_soc_dai_set_clkdiv(cpu_dai, AT91SSC_CMR_DIV, cmr_div); if (ret < 0) return ret; if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { /* set the BCLK divider for DACLRC */ - ret = cpu_dai->dai_ops.set_clkdiv(cpu_dai, + ret = snd_soc_dai_set_clkdiv(cpu_dai, AT91SSC_TCMR_PERIOD, period); } else { /* set the BCLK divider for ADCLRC */ - ret = cpu_dai->dai_ops.set_clkdiv(cpu_dai, + ret = snd_soc_dai_set_clkdiv(cpu_dai, AT91SSC_RCMR_PERIOD, period); } if (ret < 0) @@ -163,13 +163,13 @@ static int eti_b1_hw_params(struct snd_pcm_substream *substream, */ /* set codec DAI configuration */ - ret = codec_dai->dai_ops.set_fmt(codec_dai, SND_SOC_DAIFMT_I2S | + ret = snd_soc_dai_set_fmt(codec_dai, SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBM_CFM); if (ret < 0) return ret; /* set cpu DAI configuration */ - ret = cpu_dai->dai_ops.set_fmt(cpu_dai, SND_SOC_DAIFMT_I2S | + ret = snd_soc_dai_set_fmt(cpu_dai, SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBM_CFM); if (ret < 0) return ret; @@ -191,7 +191,7 @@ static const struct snd_soc_dapm_widget eti_b1_dapm_widgets[] = { SND_SOC_DAPM_SPK("Ext Spk", NULL), }; -static const char *intercon[][3] = { +static const struct snd_soc_dapm_route intercon[] = { /* speaker connected to LHPOUT */ {"Ext Spk", NULL, "LHPOUT"}, @@ -199,9 +199,6 @@ static const char *intercon[][3] = { /* mic is connected to Mic Jack, with WM8731 Mic Bias */ {"MICIN", NULL, "Mic Bias"}, {"Mic Bias", NULL, "Int Mic"}, - - /* terminator */ - {NULL, NULL, NULL}, }; /* @@ -209,30 +206,24 @@ static const char *intercon[][3] = { */ static int eti_b1_wm8731_init(struct snd_soc_codec *codec) { - int i; - DBG("eti_b1_wm8731_init() called\n"); /* Add specific widgets */ - for(i = 0; i < ARRAY_SIZE(eti_b1_dapm_widgets); i++) { - snd_soc_dapm_new_control(codec, &eti_b1_dapm_widgets[i]); - } + snd_soc_dapm_new_controls(codec, eti_b1_dapm_widgets, + ARRAY_SIZE(eti_b1_dapm_widgets)); /* Set up specific audio path interconnects */ - for(i = 0; intercon[i][0] != NULL; i++) { - snd_soc_dapm_connect_input(codec, intercon[i][0], - intercon[i][1], intercon[i][2]); - } + snd_soc_dapm_add_route(codec, intercon, ARRAY_SIZE(intercon)); /* not connected */ - snd_soc_dapm_set_endpoint(codec, "RLINEIN", 0); - snd_soc_dapm_set_endpoint(codec, "LLINEIN", 0); + snd_soc_dapm_disable_pin(codec, "RLINEIN"); + snd_soc_dapm_disable_pin(codec, "LLINEIN"); /* always connected */ - snd_soc_dapm_set_endpoint(codec, "Int Mic", 1); - snd_soc_dapm_set_endpoint(codec, "Ext Spk", 1); + snd_soc_dapm_enable_pin(codec, "Int Mic"); + snd_soc_dapm_enable_pin(codec, "Ext Spk"); - snd_soc_dapm_sync_endpoints(codec); + snd_soc_dapm_sync(codec); return 0; } diff --git a/sound/soc/au1x/Kconfig b/sound/soc/au1x/Kconfig new file mode 100644 index 00000000000..410a893aa66 --- /dev/null +++ b/sound/soc/au1x/Kconfig @@ -0,0 +1,32 @@ +## +## Au1200/Au1550 PSC + DBDMA +## +config SND_SOC_AU1XPSC + tristate "SoC Audio for Au1200/Au1250/Au1550" + depends on SOC_AU1200 || SOC_AU1550 + help + This option enables support for the Programmable Serial + Controllers in AC97 and I2S mode, and the Descriptor-Based DMA + Controller (DBDMA) as found on the Au1200/Au1250/Au1550 SoC. + +config SND_SOC_AU1XPSC_I2S + tristate + +config SND_SOC_AU1XPSC_AC97 + tristate + select AC97_BUS + select SND_AC97_CODEC + select SND_SOC_AC97_BUS + + +## +## Boards +## +config SND_SOC_SAMPLE_PSC_AC97 + tristate "Sample Au12x0/Au1550 PSC AC97 sound machine" + depends on SND_SOC_AU1XPSC + select SND_SOC_AU1XPSC_AC97 + select SND_SOC_AC97_CODEC + help + This is a sample AC97 sound machine for use in Au12x0/Au1550 + based systems which have audio on PSC1 (e.g. Db1200 demoboard). diff --git a/sound/soc/au1x/Makefile b/sound/soc/au1x/Makefile new file mode 100644 index 00000000000..6c6950b8003 --- /dev/null +++ b/sound/soc/au1x/Makefile @@ -0,0 +1,13 @@ +# Au1200/Au1550 PSC audio +snd-soc-au1xpsc-dbdma-objs := dbdma2.o +snd-soc-au1xpsc-i2s-objs := psc-i2s.o +snd-soc-au1xpsc-ac97-objs := psc-ac97.o + +obj-$(CONFIG_SND_SOC_AU1XPSC) += snd-soc-au1xpsc-dbdma.o +obj-$(CONFIG_SND_SOC_AU1XPSC_I2S) += snd-soc-au1xpsc-i2s.o +obj-$(CONFIG_SND_SOC_AU1XPSC_AC97) += snd-soc-au1xpsc-ac97.o + +# Boards +snd-soc-sample-ac97-objs := sample-ac97.o + +obj-$(CONFIG_SND_SOC_SAMPLE_PSC_AC97) += snd-soc-sample-ac97.o diff --git a/sound/soc/au1x/dbdma2.c b/sound/soc/au1x/dbdma2.c new file mode 100644 index 00000000000..1466d932880 --- /dev/null +++ b/sound/soc/au1x/dbdma2.c @@ -0,0 +1,421 @@ +/* + * Au12x0/Au1550 PSC ALSA ASoC audio support. + * + * (c) 2007-2008 MSC Vertriebsges.m.b.H., + * Manuel Lauss <mano@roarinelk.homelinux.net> + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. + * + * DMA glue for Au1x-PSC audio. + * + * NOTE: all of these drivers can only work with a SINGLE instance + * of a PSC. Multiple independent audio devices are impossible + * with ASoC v1. + */ + + +#include <linux/module.h> +#include <linux/init.h> +#include <linux/platform_device.h> +#include <linux/slab.h> +#include <linux/dma-mapping.h> + +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> + +#include <asm/mach-au1x00/au1000.h> +#include <asm/mach-au1x00/au1xxx_dbdma.h> +#include <asm/mach-au1x00/au1xxx_psc.h> + +#include "psc.h" + +/*#define PCM_DEBUG*/ + +#define MSG(x...) printk(KERN_INFO "au1xpsc_pcm: " x) +#ifdef PCM_DEBUG +#define DBG MSG +#else +#define DBG(x...) do {} while (0) +#endif + +struct au1xpsc_audio_dmadata { + /* DDMA control data */ + unsigned int ddma_id; /* DDMA direction ID for this PSC */ + u32 ddma_chan; /* DDMA context */ + + /* PCM context (for irq handlers) */ + struct snd_pcm_substream *substream; + unsigned long curr_period; /* current segment DDMA is working on */ + unsigned long q_period; /* queue period(s) */ + unsigned long dma_area; /* address of queued DMA area */ + unsigned long dma_area_s; /* start address of DMA area */ + unsigned long pos; /* current byte position being played */ + unsigned long periods; /* number of SG segments in total */ + unsigned long period_bytes; /* size in bytes of one SG segment */ + + /* runtime data */ + int msbits; +}; + +/* instance data. There can be only one, MacLeod!!!! */ +static struct au1xpsc_audio_dmadata *au1xpsc_audio_pcmdma[2]; + +/* + * These settings are somewhat okay, at least on my machine audio plays + * almost skip-free. Especially the 64kB buffer seems to help a LOT. + */ +#define AU1XPSC_PERIOD_MIN_BYTES 1024 +#define AU1XPSC_BUFFER_MIN_BYTES 65536 + +#define AU1XPSC_PCM_FMTS \ + (SNDRV_PCM_FMTBIT_S8 | SNDRV_PCM_FMTBIT_U8 | \ + SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S16_BE | \ + SNDRV_PCM_FMTBIT_U16_LE | SNDRV_PCM_FMTBIT_U16_BE | \ + SNDRV_PCM_FMTBIT_S32_LE | SNDRV_PCM_FMTBIT_S32_BE | \ + SNDRV_PCM_FMTBIT_U32_LE | SNDRV_PCM_FMTBIT_U32_BE | \ + 0) + +/* PCM hardware DMA capabilities - platform specific */ +static const struct snd_pcm_hardware au1xpsc_pcm_hardware = { + .info = SNDRV_PCM_INFO_MMAP | SNDRV_PCM_INFO_MMAP_VALID | + SNDRV_PCM_INFO_INTERLEAVED, + .formats = AU1XPSC_PCM_FMTS, + .period_bytes_min = AU1XPSC_PERIOD_MIN_BYTES, + .period_bytes_max = 4096 * 1024 - 1, + .periods_min = 2, + .periods_max = 4096, /* 2 to as-much-as-you-like */ + .buffer_bytes_max = 4096 * 1024 - 1, + .fifo_size = 16, /* fifo entries of AC97/I2S PSC */ +}; + +static void au1x_pcm_queue_tx(struct au1xpsc_audio_dmadata *cd) +{ + au1xxx_dbdma_put_source_flags(cd->ddma_chan, + (void *)phys_to_virt(cd->dma_area), + cd->period_bytes, DDMA_FLAGS_IE); + + /* update next-to-queue period */ + ++cd->q_period; + cd->dma_area += cd->period_bytes; + if (cd->q_period >= cd->periods) { + cd->q_period = 0; + cd->dma_area = cd->dma_area_s; + } +} + +static void au1x_pcm_queue_rx(struct au1xpsc_audio_dmadata *cd) +{ + au1xxx_dbdma_put_dest_flags(cd->ddma_chan, + (void *)phys_to_virt(cd->dma_area), + cd->period_bytes, DDMA_FLAGS_IE); + + /* update next-to-queue period */ + ++cd->q_period; + cd->dma_area += cd->period_bytes; + if (cd->q_period >= cd->periods) { + cd->q_period = 0; + cd->dma_area = cd->dma_area_s; + } +} + +static void au1x_pcm_dmatx_cb(int irq, void *dev_id) +{ + struct au1xpsc_audio_dmadata *cd = dev_id; + + cd->pos += cd->period_bytes; + if (++cd->curr_period >= cd->periods) { + cd->pos = 0; + cd->curr_period = 0; + } + snd_pcm_period_elapsed(cd->substream); + au1x_pcm_queue_tx(cd); +} + +static void au1x_pcm_dmarx_cb(int irq, void *dev_id) +{ + struct au1xpsc_audio_dmadata *cd = dev_id; + + cd->pos += cd->period_bytes; + if (++cd->curr_period >= cd->periods) { + cd->pos = 0; + cd->curr_period = 0; + } + snd_pcm_period_elapsed(cd->substream); + au1x_pcm_queue_rx(cd); +} + +static void au1x_pcm_dbdma_free(struct au1xpsc_audio_dmadata *pcd) +{ + if (pcd->ddma_chan) { + au1xxx_dbdma_stop(pcd->ddma_chan); + au1xxx_dbdma_reset(pcd->ddma_chan); + au1xxx_dbdma_chan_free(pcd->ddma_chan); + pcd->ddma_chan = 0; + pcd->msbits = 0; + } +} + +/* in case of missing DMA ring or changed TX-source / RX-dest bit widths, + * allocate (or reallocate) a 2-descriptor DMA ring with bit depth according + * to ALSA-supplied sample depth. This is due to limitations in the dbdma api + * (cannot adjust source/dest widths of already allocated descriptor ring). + */ +static int au1x_pcm_dbdma_realloc(struct au1xpsc_audio_dmadata *pcd, + int stype, int msbits) +{ + /* DMA only in 8/16/32 bit widths */ + if (msbits == 24) + msbits = 32; + + /* check current config: correct bits and descriptors allocated? */ + if ((pcd->ddma_chan) && (msbits == pcd->msbits)) + goto out; /* all ok! */ + + au1x_pcm_dbdma_free(pcd); + + if (stype == PCM_RX) + pcd->ddma_chan = au1xxx_dbdma_chan_alloc(pcd->ddma_id, + DSCR_CMD0_ALWAYS, + au1x_pcm_dmarx_cb, (void *)pcd); + else + pcd->ddma_chan = au1xxx_dbdma_chan_alloc(DSCR_CMD0_ALWAYS, + pcd->ddma_id, + au1x_pcm_dmatx_cb, (void *)pcd); + + if (!pcd->ddma_chan) + return -ENOMEM;; + + au1xxx_dbdma_set_devwidth(pcd->ddma_chan, msbits); + au1xxx_dbdma_ring_alloc(pcd->ddma_chan, 2); + + pcd->msbits = msbits; + + au1xxx_dbdma_stop(pcd->ddma_chan); + au1xxx_dbdma_reset(pcd->ddma_chan); + +out: + return 0; +} + +static int au1xpsc_pcm_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params) +{ + struct snd_pcm_runtime *runtime = substream->runtime; + struct au1xpsc_audio_dmadata *pcd; + int stype, ret; + + ret = snd_pcm_lib_malloc_pages(substream, params_buffer_bytes(params)); + if (ret < 0) + goto out; + + stype = SUBSTREAM_TYPE(substream); + pcd = au1xpsc_audio_pcmdma[stype]; + + DBG("runtime->dma_area = 0x%08lx dma_addr_t = 0x%08lx dma_size = %d " + "runtime->min_align %d\n", + (unsigned long)runtime->dma_area, + (unsigned long)runtime->dma_addr, runtime->dma_bytes, + runtime->min_align); + + DBG("bits %d frags %d frag_bytes %d is_rx %d\n", params->msbits, + params_periods(params), params_period_bytes(params), stype); + + ret = au1x_pcm_dbdma_realloc(pcd, stype, params->msbits); + if (ret) { + MSG("DDMA channel (re)alloc failed!\n"); + goto out; + } + + pcd->substream = substream; + pcd->period_bytes = params_period_bytes(params); + pcd->periods = params_periods(params); + pcd->dma_area_s = pcd->dma_area = (unsigned long)runtime->dma_addr; + pcd->q_period = 0; + pcd->curr_period = 0; + pcd->pos = 0; + + ret = 0; +out: + return ret; +} + +static int au1xpsc_pcm_hw_free(struct snd_pcm_substream *substream) +{ + snd_pcm_lib_free_pages(substream); + return 0; +} + +static int au1xpsc_pcm_prepare(struct snd_pcm_substream *substream) +{ + struct au1xpsc_audio_dmadata *pcd = + au1xpsc_audio_pcmdma[SUBSTREAM_TYPE(substream)]; + + au1xxx_dbdma_reset(pcd->ddma_chan); + + if (SUBSTREAM_TYPE(substream) == PCM_RX) { + au1x_pcm_queue_rx(pcd); + au1x_pcm_queue_rx(pcd); + } else { + au1x_pcm_queue_tx(pcd); + au1x_pcm_queue_tx(pcd); + } + + return 0; +} + +static int au1xpsc_pcm_trigger(struct snd_pcm_substream *substream, int cmd) +{ + u32 c = au1xpsc_audio_pcmdma[SUBSTREAM_TYPE(substream)]->ddma_chan; + + switch (cmd) { + case SNDRV_PCM_TRIGGER_START: + case SNDRV_PCM_TRIGGER_RESUME: + au1xxx_dbdma_start(c); + break; + case SNDRV_PCM_TRIGGER_STOP: + case SNDRV_PCM_TRIGGER_SUSPEND: + au1xxx_dbdma_stop(c); + break; + default: + return -EINVAL; + } + return 0; +} + +static snd_pcm_uframes_t +au1xpsc_pcm_pointer(struct snd_pcm_substream *substream) +{ + return bytes_to_frames(substream->runtime, + au1xpsc_audio_pcmdma[SUBSTREAM_TYPE(substream)]->pos); +} + +static int au1xpsc_pcm_open(struct snd_pcm_substream *substream) +{ + snd_soc_set_runtime_hwparams(substream, &au1xpsc_pcm_hardware); + return 0; +} + +static int au1xpsc_pcm_close(struct snd_pcm_substream *substream) +{ + au1x_pcm_dbdma_free(au1xpsc_audio_pcmdma[SUBSTREAM_TYPE(substream)]); + return 0; +} + +struct snd_pcm_ops au1xpsc_pcm_ops = { + .open = au1xpsc_pcm_open, + .close = au1xpsc_pcm_close, + .ioctl = snd_pcm_lib_ioctl, + .hw_params = au1xpsc_pcm_hw_params, + .hw_free = au1xpsc_pcm_hw_free, + .prepare = au1xpsc_pcm_prepare, + .trigger = au1xpsc_pcm_trigger, + .pointer = au1xpsc_pcm_pointer, +}; + +static void au1xpsc_pcm_free_dma_buffers(struct snd_pcm *pcm) +{ + snd_pcm_lib_preallocate_free_for_all(pcm); +} + +static int au1xpsc_pcm_new(struct snd_card *card, + struct snd_soc_dai *dai, + struct snd_pcm *pcm) +{ + snd_pcm_lib_preallocate_pages_for_all(pcm, SNDRV_DMA_TYPE_DEV, + card->dev, AU1XPSC_BUFFER_MIN_BYTES, (4096 * 1024) - 1); + + return 0; +} + +static int au1xpsc_pcm_probe(struct platform_device *pdev) +{ + struct resource *r; + int ret; + + if (au1xpsc_audio_pcmdma[PCM_TX] || au1xpsc_audio_pcmdma[PCM_RX]) + return -EBUSY; + + /* TX DMA */ + au1xpsc_audio_pcmdma[PCM_TX] + = kzalloc(sizeof(struct au1xpsc_audio_dmadata), GFP_KERNEL); + if (!au1xpsc_audio_pcmdma[PCM_TX]) + return -ENOMEM; + + r = platform_get_resource(pdev, IORESOURCE_DMA, 0); + if (!r) { + ret = -ENODEV; + goto out1; + } + (au1xpsc_audio_pcmdma[PCM_TX])->ddma_id = r->start; + + /* RX DMA */ + au1xpsc_audio_pcmdma[PCM_RX] + = kzalloc(sizeof(struct au1xpsc_audio_dmadata), GFP_KERNEL); + if (!au1xpsc_audio_pcmdma[PCM_RX]) + return -ENOMEM; + + r = platform_get_resource(pdev, IORESOURCE_DMA, 1); + if (!r) { + ret = -ENODEV; + goto out2; + } + (au1xpsc_audio_pcmdma[PCM_RX])->ddma_id = r->start; + + return 0; + +out2: + kfree(au1xpsc_audio_pcmdma[PCM_RX]); + au1xpsc_audio_pcmdma[PCM_RX] = NULL; +out1: + kfree(au1xpsc_audio_pcmdma[PCM_TX]); + au1xpsc_audio_pcmdma[PCM_TX] = NULL; + return ret; +} + +static int au1xpsc_pcm_remove(struct platform_device *pdev) +{ + int i; + + for (i = 0; i < 2; i++) { + if (au1xpsc_audio_pcmdma[i]) { + au1x_pcm_dbdma_free(au1xpsc_audio_pcmdma[i]); + kfree(au1xpsc_audio_pcmdma[i]); + au1xpsc_audio_pcmdma[i] = NULL; + } + } + + return 0; +} + +/* au1xpsc audio platform */ +struct snd_soc_platform au1xpsc_soc_platform = { + .name = "au1xpsc-pcm-dbdma", + .probe = au1xpsc_pcm_probe, + .remove = au1xpsc_pcm_remove, + .pcm_ops = &au1xpsc_pcm_ops, + .pcm_new = au1xpsc_pcm_new, + .pcm_free = au1xpsc_pcm_free_dma_buffers, +}; +EXPORT_SYMBOL_GPL(au1xpsc_soc_platform); + +static int __init au1xpsc_audio_dbdma_init(void) +{ + au1xpsc_audio_pcmdma[PCM_TX] = NULL; + au1xpsc_audio_pcmdma[PCM_RX] = NULL; + return 0; +} + +static void __exit au1xpsc_audio_dbdma_exit(void) +{ +} + +module_init(au1xpsc_audio_dbdma_init); +module_exit(au1xpsc_audio_dbdma_exit); + +MODULE_LICENSE("GPL"); +MODULE_DESCRIPTION("Au12x0/Au1550 PSC Audio DMA driver"); +MODULE_AUTHOR("Manuel Lauss <mano@roarinelk.homelinux.net>"); diff --git a/sound/soc/au1x/psc-ac97.c b/sound/soc/au1x/psc-ac97.c new file mode 100644 index 00000000000..57facbad682 --- /dev/null +++ b/sound/soc/au1x/psc-ac97.c @@ -0,0 +1,387 @@ +/* + * Au12x0/Au1550 PSC ALSA ASoC audio support. + * + * (c) 2007-2008 MSC Vertriebsges.m.b.H., + * Manuel Lauss <mano@roarinelk.homelinux.net> + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. + * + * Au1xxx-PSC AC97 glue. + * + * NOTE: all of these drivers can only work with a SINGLE instance + * of a PSC. Multiple independent audio devices are impossible + * with ASoC v1. + */ + +#include <linux/init.h> +#include <linux/module.h> +#include <linux/device.h> +#include <linux/delay.h> +#include <linux/suspend.h> +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/initval.h> +#include <sound/soc.h> +#include <asm/mach-au1x00/au1000.h> +#include <asm/mach-au1x00/au1xxx_psc.h> + +#include "psc.h" + +#define AC97_DIR \ + (SND_SOC_DAIDIR_PLAYBACK | SND_SOC_DAIDIR_CAPTURE) + +#define AC97_RATES \ + SNDRV_PCM_RATE_8000_48000 + +#define AC97_FMTS \ + (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3BE) + +#define AC97PCR_START(stype) \ + ((stype) == PCM_TX ? PSC_AC97PCR_TS : PSC_AC97PCR_RS) +#define AC97PCR_STOP(stype) \ + ((stype) == PCM_TX ? PSC_AC97PCR_TP : PSC_AC97PCR_RP) +#define AC97PCR_CLRFIFO(stype) \ + ((stype) == PCM_TX ? PSC_AC97PCR_TC : PSC_AC97PCR_RC) + +/* instance data. There can be only one, MacLeod!!!! */ +static struct au1xpsc_audio_data *au1xpsc_ac97_workdata; + +/* AC97 controller reads codec register */ +static unsigned short au1xpsc_ac97_read(struct snd_ac97 *ac97, + unsigned short reg) +{ + /* FIXME */ + struct au1xpsc_audio_data *pscdata = au1xpsc_ac97_workdata; + unsigned short data, tmo; + + au_writel(PSC_AC97CDC_RD | PSC_AC97CDC_INDX(reg), AC97_CDC(pscdata)); + au_sync(); + + tmo = 1000; + while ((!(au_readl(AC97_EVNT(pscdata)) & PSC_AC97EVNT_CD)) && --tmo) + udelay(2); + + if (!tmo) + data = 0xffff; + else + data = au_readl(AC97_CDC(pscdata)) & 0xffff; + + au_writel(PSC_AC97EVNT_CD, AC97_EVNT(pscdata)); + au_sync(); + + return data; +} + +/* AC97 controller writes to codec register */ +static void au1xpsc_ac97_write(struct snd_ac97 *ac97, unsigned short reg, + unsigned short val) +{ + /* FIXME */ + struct au1xpsc_audio_data *pscdata = au1xpsc_ac97_workdata; + unsigned int tmo; + + au_writel(PSC_AC97CDC_INDX(reg) | (val & 0xffff), AC97_CDC(pscdata)); + au_sync(); + tmo = 1000; + while ((!(au_readl(AC97_EVNT(pscdata)) & PSC_AC97EVNT_CD)) && --tmo) + au_sync(); + + au_writel(PSC_AC97EVNT_CD, AC97_EVNT(pscdata)); + au_sync(); +} + +/* AC97 controller asserts a warm reset */ +static void au1xpsc_ac97_warm_reset(struct snd_ac97 *ac97) +{ + /* FIXME */ + struct au1xpsc_audio_data *pscdata = au1xpsc_ac97_workdata; + + au_writel(PSC_AC97RST_SNC, AC97_RST(pscdata)); + au_sync(); + msleep(10); + au_writel(0, AC97_RST(pscdata)); + au_sync(); +} + +static void au1xpsc_ac97_cold_reset(struct snd_ac97 *ac97) +{ + /* FIXME */ + struct au1xpsc_audio_data *pscdata = au1xpsc_ac97_workdata; + int i; + + /* disable PSC during cold reset */ + au_writel(0, AC97_CFG(au1xpsc_ac97_workdata)); + au_sync(); + au_writel(PSC_CTRL_DISABLE, PSC_CTRL(pscdata)); + au_sync(); + + /* issue cold reset */ + au_writel(PSC_AC97RST_RST, AC97_RST(pscdata)); + au_sync(); + msleep(500); + au_writel(0, AC97_RST(pscdata)); + au_sync(); + + /* enable PSC */ + au_writel(PSC_CTRL_ENABLE, PSC_CTRL(pscdata)); + au_sync(); + + /* wait for PSC to indicate it's ready */ + i = 100000; + while (!((au_readl(AC97_STAT(pscdata)) & PSC_AC97STAT_SR)) && (--i)) + au_sync(); + + if (i == 0) { + printk(KERN_ERR "au1xpsc-ac97: PSC not ready!\n"); + return; + } + + /* enable the ac97 function */ + au_writel(pscdata->cfg | PSC_AC97CFG_DE_ENABLE, AC97_CFG(pscdata)); + au_sync(); + + /* wait for AC97 core to become ready */ + i = 100000; + while (!((au_readl(AC97_STAT(pscdata)) & PSC_AC97STAT_DR)) && (--i)) + au_sync(); + if (i == 0) + printk(KERN_ERR "au1xpsc-ac97: AC97 ctrl not ready\n"); +} + +/* AC97 controller operations */ +struct snd_ac97_bus_ops soc_ac97_ops = { + .read = au1xpsc_ac97_read, + .write = au1xpsc_ac97_write, + .reset = au1xpsc_ac97_cold_reset, + .warm_reset = au1xpsc_ac97_warm_reset, +}; +EXPORT_SYMBOL_GPL(soc_ac97_ops); + +static int au1xpsc_ac97_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params) +{ + /* FIXME */ + struct au1xpsc_audio_data *pscdata = au1xpsc_ac97_workdata; + unsigned long r, stat; + int chans, stype = SUBSTREAM_TYPE(substream); + + chans = params_channels(params); + + r = au_readl(AC97_CFG(pscdata)); + stat = au_readl(AC97_STAT(pscdata)); + + /* already active? */ + if (stat & (PSC_AC97STAT_TB | PSC_AC97STAT_RB)) { + /* reject parameters not currently set up */ + if ((PSC_AC97CFG_GET_LEN(r) != params->msbits) || + (pscdata->rate != params_rate(params))) + return -EINVAL; + } else { + /* disable AC97 device controller first */ + au_writel(r & ~PSC_AC97CFG_DE_ENABLE, AC97_CFG(pscdata)); + au_sync(); + + /* set sample bitdepth: REG[24:21]=(BITS-2)/2 */ + r &= ~PSC_AC97CFG_LEN_MASK; + r |= PSC_AC97CFG_SET_LEN(params->msbits); + + /* channels: enable slots for front L/R channel */ + if (stype == PCM_TX) { + r &= ~PSC_AC97CFG_TXSLOT_MASK; + r |= PSC_AC97CFG_TXSLOT_ENA(3); + r |= PSC_AC97CFG_TXSLOT_ENA(4); + } else { + r &= ~PSC_AC97CFG_RXSLOT_MASK; + r |= PSC_AC97CFG_RXSLOT_ENA(3); + r |= PSC_AC97CFG_RXSLOT_ENA(4); + } + + /* finally enable the AC97 controller again */ + au_writel(r | PSC_AC97CFG_DE_ENABLE, AC97_CFG(pscdata)); + au_sync(); + + pscdata->cfg = r; + pscdata->rate = params_rate(params); + } + + return 0; +} + +static int au1xpsc_ac97_trigger(struct snd_pcm_substream *substream, + int cmd) +{ + /* FIXME */ + struct au1xpsc_audio_data *pscdata = au1xpsc_ac97_workdata; + int ret, stype = SUBSTREAM_TYPE(substream); + + ret = 0; + + switch (cmd) { + case SNDRV_PCM_TRIGGER_START: + case SNDRV_PCM_TRIGGER_RESUME: + au_writel(AC97PCR_START(stype), AC97_PCR(pscdata)); + au_sync(); + break; + case SNDRV_PCM_TRIGGER_STOP: + case SNDRV_PCM_TRIGGER_SUSPEND: + au_writel(AC97PCR_STOP(stype), AC97_PCR(pscdata)); + au_sync(); + break; + default: + ret = -EINVAL; + } + return ret; +} + +static int au1xpsc_ac97_probe(struct platform_device *pdev, + struct snd_soc_dai *dai) +{ + int ret; + struct resource *r; + unsigned long sel; + + if (au1xpsc_ac97_workdata) + return -EBUSY; + + au1xpsc_ac97_workdata = + kzalloc(sizeof(struct au1xpsc_audio_data), GFP_KERNEL); + if (!au1xpsc_ac97_workdata) + return -ENOMEM; + + r = platform_get_resource(pdev, IORESOURCE_MEM, 0); + if (!r) { + ret = -ENODEV; + goto out0; + } + + ret = -EBUSY; + au1xpsc_ac97_workdata->ioarea = + request_mem_region(r->start, r->end - r->start + 1, + "au1xpsc_ac97"); + if (!au1xpsc_ac97_workdata->ioarea) + goto out0; + + au1xpsc_ac97_workdata->mmio = ioremap(r->start, 0xffff); + if (!au1xpsc_ac97_workdata->mmio) + goto out1; + + /* configuration: max dma trigger threshold, enable ac97 */ + au1xpsc_ac97_workdata->cfg = PSC_AC97CFG_RT_FIFO8 | + PSC_AC97CFG_TT_FIFO8 | + PSC_AC97CFG_DE_ENABLE; + + /* preserve PSC clock source set up by platform (dev.platform_data + * is already occupied by soc layer) + */ + sel = au_readl(PSC_SEL(au1xpsc_ac97_workdata)) & PSC_SEL_CLK_MASK; + au_writel(PSC_CTRL_DISABLE, PSC_CTRL(au1xpsc_ac97_workdata)); + au_sync(); + au_writel(0, PSC_SEL(au1xpsc_ac97_workdata)); + au_sync(); + au_writel(PSC_SEL_PS_AC97MODE | sel, PSC_SEL(au1xpsc_ac97_workdata)); + au_sync(); + /* next up: cold reset. Dont check for PSC-ready now since + * there may not be any codec clock yet. + */ + + return 0; + +out1: + release_resource(au1xpsc_ac97_workdata->ioarea); + kfree(au1xpsc_ac97_workdata->ioarea); +out0: + kfree(au1xpsc_ac97_workdata); + au1xpsc_ac97_workdata = NULL; + return ret; +} + +static void au1xpsc_ac97_remove(struct platform_device *pdev, + struct snd_soc_dai *dai) +{ + /* disable PSC completely */ + au_writel(0, AC97_CFG(au1xpsc_ac97_workdata)); + au_sync(); + au_writel(PSC_CTRL_DISABLE, PSC_CTRL(au1xpsc_ac97_workdata)); + au_sync(); + + iounmap(au1xpsc_ac97_workdata->mmio); + release_resource(au1xpsc_ac97_workdata->ioarea); + kfree(au1xpsc_ac97_workdata->ioarea); + kfree(au1xpsc_ac97_workdata); + au1xpsc_ac97_workdata = NULL; +} + +static int au1xpsc_ac97_suspend(struct platform_device *pdev, + struct snd_soc_dai *dai) +{ + /* save interesting registers and disable PSC */ + au1xpsc_ac97_workdata->pm[0] = + au_readl(PSC_SEL(au1xpsc_ac97_workdata)); + + au_writel(0, AC97_CFG(au1xpsc_ac97_workdata)); + au_sync(); + au_writel(PSC_CTRL_DISABLE, PSC_CTRL(au1xpsc_ac97_workdata)); + au_sync(); + + return 0; +} + +static int au1xpsc_ac97_resume(struct platform_device *pdev, + struct snd_soc_dai *dai) +{ + /* restore PSC clock config */ + au_writel(au1xpsc_ac97_workdata->pm[0] | PSC_SEL_PS_AC97MODE, + PSC_SEL(au1xpsc_ac97_workdata)); + au_sync(); + + /* after this point the ac97 core will cold-reset the codec. + * During cold-reset the PSC is reinitialized and the last + * configuration set up in hw_params() is restored. + */ + return 0; +} + +struct snd_soc_dai au1xpsc_ac97_dai = { + .name = "au1xpsc_ac97", + .type = SND_SOC_DAI_AC97, + .probe = au1xpsc_ac97_probe, + .remove = au1xpsc_ac97_remove, + .suspend = au1xpsc_ac97_suspend, + .resume = au1xpsc_ac97_resume, + .playback = { + .rates = AC97_RATES, + .formats = AC97_FMTS, + .channels_min = 2, + .channels_max = 2, + }, + .capture = { + .rates = AC97_RATES, + .formats = AC97_FMTS, + .channels_min = 2, + .channels_max = 2, + }, + .ops = { + .trigger = au1xpsc_ac97_trigger, + .hw_params = au1xpsc_ac97_hw_params, + }, +}; +EXPORT_SYMBOL_GPL(au1xpsc_ac97_dai); + +static int __init au1xpsc_ac97_init(void) +{ + au1xpsc_ac97_workdata = NULL; + return 0; +} + +static void __exit au1xpsc_ac97_exit(void) +{ +} + +module_init(au1xpsc_ac97_init); +module_exit(au1xpsc_ac97_exit); + +MODULE_LICENSE("GPL"); +MODULE_DESCRIPTION("Au12x0/Au1550 PSC AC97 ALSA ASoC audio driver"); +MODULE_AUTHOR("Manuel Lauss <mano@roarinelk.homelinux.net>"); diff --git a/sound/soc/au1x/psc-i2s.c b/sound/soc/au1x/psc-i2s.c new file mode 100644 index 00000000000..ba4b5c199f2 --- /dev/null +++ b/sound/soc/au1x/psc-i2s.c @@ -0,0 +1,414 @@ +/* + * Au12x0/Au1550 PSC ALSA ASoC audio support. + * + * (c) 2007-2008 MSC Vertriebsges.m.b.H., + * Manuel Lauss <mano@roarinelk.homelinux.net> + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. + * + * Au1xxx-PSC I2S glue. + * + * NOTE: all of these drivers can only work with a SINGLE instance + * of a PSC. Multiple independent audio devices are impossible + * with ASoC v1. + * NOTE: so far only PSC slave mode (bit- and frameclock) is supported. + */ + +#include <linux/init.h> +#include <linux/module.h> +#include <linux/suspend.h> +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/initval.h> +#include <sound/soc.h> +#include <asm/mach-au1x00/au1000.h> +#include <asm/mach-au1x00/au1xxx_psc.h> + +#include "psc.h" + +/* supported I2S DAI hardware formats */ +#define AU1XPSC_I2S_DAIFMT \ + (SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_LEFT_J | \ + SND_SOC_DAIFMT_NB_NF) + +/* supported I2S direction */ +#define AU1XPSC_I2S_DIR \ + (SND_SOC_DAIDIR_PLAYBACK | SND_SOC_DAIDIR_CAPTURE) + +#define AU1XPSC_I2S_RATES \ + SNDRV_PCM_RATE_8000_192000 + +#define AU1XPSC_I2S_FMTS \ + (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S24_LE) + +#define I2SSTAT_BUSY(stype) \ + ((stype) == PCM_TX ? PSC_I2SSTAT_TB : PSC_I2SSTAT_RB) +#define I2SPCR_START(stype) \ + ((stype) == PCM_TX ? PSC_I2SPCR_TS : PSC_I2SPCR_RS) +#define I2SPCR_STOP(stype) \ + ((stype) == PCM_TX ? PSC_I2SPCR_TP : PSC_I2SPCR_RP) +#define I2SPCR_CLRFIFO(stype) \ + ((stype) == PCM_TX ? PSC_I2SPCR_TC : PSC_I2SPCR_RC) + + +/* instance data. There can be only one, MacLeod!!!! */ +static struct au1xpsc_audio_data *au1xpsc_i2s_workdata; + +static int au1xpsc_i2s_set_fmt(struct snd_soc_dai *cpu_dai, + unsigned int fmt) +{ + struct au1xpsc_audio_data *pscdata = au1xpsc_i2s_workdata; + unsigned long ct; + int ret; + + ret = -EINVAL; + + ct = pscdata->cfg; + + ct &= ~(PSC_I2SCFG_XM | PSC_I2SCFG_MLJ); /* left-justified */ + switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { + case SND_SOC_DAIFMT_I2S: + ct |= PSC_I2SCFG_XM; /* enable I2S mode */ + break; + case SND_SOC_DAIFMT_MSB: + break; + case SND_SOC_DAIFMT_LSB: + ct |= PSC_I2SCFG_MLJ; /* LSB (right-) justified */ + break; + default: + goto out; + } + + ct &= ~(PSC_I2SCFG_BI | PSC_I2SCFG_WI); /* IB-IF */ + switch (fmt & SND_SOC_DAIFMT_INV_MASK) { + case SND_SOC_DAIFMT_NB_NF: + ct |= PSC_I2SCFG_BI | PSC_I2SCFG_WI; + break; + case SND_SOC_DAIFMT_NB_IF: + ct |= PSC_I2SCFG_BI; + break; + case SND_SOC_DAIFMT_IB_NF: + ct |= PSC_I2SCFG_WI; + break; + case SND_SOC_DAIFMT_IB_IF: + break; + default: + goto out; + } + + switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { + case SND_SOC_DAIFMT_CBM_CFM: /* CODEC master */ + ct |= PSC_I2SCFG_MS; /* PSC I2S slave mode */ + break; + case SND_SOC_DAIFMT_CBS_CFS: /* CODEC slave */ + ct &= ~PSC_I2SCFG_MS; /* PSC I2S Master mode */ + break; + default: + goto out; + } + + pscdata->cfg = ct; + ret = 0; +out: + return ret; +} + +static int au1xpsc_i2s_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params) +{ + struct au1xpsc_audio_data *pscdata = au1xpsc_i2s_workdata; + + int cfgbits; + unsigned long stat; + + /* check if the PSC is already streaming data */ + stat = au_readl(I2S_STAT(pscdata)); + if (stat & (PSC_I2SSTAT_TB | PSC_I2SSTAT_RB)) { + /* reject parameters not currently set up in hardware */ + cfgbits = au_readl(I2S_CFG(pscdata)); + if ((PSC_I2SCFG_GET_LEN(cfgbits) != params->msbits) || + (params_rate(params) != pscdata->rate)) + return -EINVAL; + } else { + /* set sample bitdepth */ + pscdata->cfg &= ~(0x1f << 4); + pscdata->cfg |= PSC_I2SCFG_SET_LEN(params->msbits); + /* remember current rate for other stream */ + pscdata->rate = params_rate(params); + } + return 0; +} + +/* Configure PSC late: on my devel systems the codec is I2S master and + * supplies the i2sbitclock __AND__ i2sMclk (!) to the PSC unit. ASoC + * uses aggressive PM and switches the codec off when it is not in use + * which also means the PSC unit doesn't get any clocks and is therefore + * dead. That's why this chunk here gets called from the trigger callback + * because I can be reasonably certain the codec is driving the clocks. + */ +static int au1xpsc_i2s_configure(struct au1xpsc_audio_data *pscdata) +{ + unsigned long tmo; + + /* bring PSC out of sleep, and configure I2S unit */ + au_writel(PSC_CTRL_ENABLE, PSC_CTRL(pscdata)); + au_sync(); + + tmo = 1000000; + while (!(au_readl(I2S_STAT(pscdata)) & PSC_I2SSTAT_SR) && tmo) + tmo--; + + if (!tmo) + goto psc_err; + + au_writel(0, I2S_CFG(pscdata)); + au_sync(); + au_writel(pscdata->cfg | PSC_I2SCFG_DE_ENABLE, I2S_CFG(pscdata)); + au_sync(); + + /* wait for I2S controller to become ready */ + tmo = 1000000; + while (!(au_readl(I2S_STAT(pscdata)) & PSC_I2SSTAT_DR) && tmo) + tmo--; + + if (tmo) + return 0; + +psc_err: + au_writel(0, I2S_CFG(pscdata)); + au_writel(PSC_CTRL_SUSPEND, PSC_CTRL(pscdata)); + au_sync(); + return -ETIMEDOUT; +} + +static int au1xpsc_i2s_start(struct au1xpsc_audio_data *pscdata, int stype) +{ + unsigned long tmo, stat; + int ret; + + ret = 0; + + /* if both TX and RX are idle, configure the PSC */ + stat = au_readl(I2S_STAT(pscdata)); + if (!(stat & (PSC_I2SSTAT_TB | PSC_I2SSTAT_RB))) { + ret = au1xpsc_i2s_configure(pscdata); + if (ret) + goto out; + } + + au_writel(I2SPCR_CLRFIFO(stype), I2S_PCR(pscdata)); + au_sync(); + au_writel(I2SPCR_START(stype), I2S_PCR(pscdata)); + au_sync(); + + /* wait for start confirmation */ + tmo = 1000000; + while (!(au_readl(I2S_STAT(pscdata)) & I2SSTAT_BUSY(stype)) && tmo) + tmo--; + + if (!tmo) { + au_writel(I2SPCR_STOP(stype), I2S_PCR(pscdata)); + au_sync(); + ret = -ETIMEDOUT; + } +out: + return ret; +} + +static int au1xpsc_i2s_stop(struct au1xpsc_audio_data *pscdata, int stype) +{ + unsigned long tmo, stat; + + au_writel(I2SPCR_STOP(stype), I2S_PCR(pscdata)); + au_sync(); + + /* wait for stop confirmation */ + tmo = 1000000; + while ((au_readl(I2S_STAT(pscdata)) & I2SSTAT_BUSY(stype)) && tmo) + tmo--; + + /* if both TX and RX are idle, disable PSC */ + stat = au_readl(I2S_STAT(pscdata)); + if (!(stat & (PSC_I2SSTAT_RB | PSC_I2SSTAT_RB))) { + au_writel(0, I2S_CFG(pscdata)); + au_sync(); + au_writel(PSC_CTRL_SUSPEND, PSC_CTRL(pscdata)); + au_sync(); + } + return 0; +} + +static int au1xpsc_i2s_trigger(struct snd_pcm_substream *substream, int cmd) +{ + struct au1xpsc_audio_data *pscdata = au1xpsc_i2s_workdata; + int ret, stype = SUBSTREAM_TYPE(substream); + + switch (cmd) { + case SNDRV_PCM_TRIGGER_START: + case SNDRV_PCM_TRIGGER_RESUME: + ret = au1xpsc_i2s_start(pscdata, stype); + break; + case SNDRV_PCM_TRIGGER_STOP: + case SNDRV_PCM_TRIGGER_SUSPEND: + ret = au1xpsc_i2s_stop(pscdata, stype); + break; + default: + ret = -EINVAL; + } + return ret; +} + +static int au1xpsc_i2s_probe(struct platform_device *pdev, + struct snd_soc_dai *dai) +{ + struct resource *r; + unsigned long sel; + int ret; + + if (au1xpsc_i2s_workdata) + return -EBUSY; + + au1xpsc_i2s_workdata = + kzalloc(sizeof(struct au1xpsc_audio_data), GFP_KERNEL); + if (!au1xpsc_i2s_workdata) + return -ENOMEM; + + r = platform_get_resource(pdev, IORESOURCE_MEM, 0); + if (!r) { + ret = -ENODEV; + goto out0; + } + + ret = -EBUSY; + au1xpsc_i2s_workdata->ioarea = + request_mem_region(r->start, r->end - r->start + 1, + "au1xpsc_i2s"); + if (!au1xpsc_i2s_workdata->ioarea) + goto out0; + + au1xpsc_i2s_workdata->mmio = ioremap(r->start, 0xffff); + if (!au1xpsc_i2s_workdata->mmio) + goto out1; + + /* preserve PSC clock source set up by platform (dev.platform_data + * is already occupied by soc layer) + */ + sel = au_readl(PSC_SEL(au1xpsc_i2s_workdata)) & PSC_SEL_CLK_MASK; + au_writel(PSC_CTRL_DISABLE, PSC_CTRL(au1xpsc_i2s_workdata)); + au_sync(); + au_writel(PSC_SEL_PS_I2SMODE | sel, PSC_SEL(au1xpsc_i2s_workdata)); + au_writel(0, I2S_CFG(au1xpsc_i2s_workdata)); + au_sync(); + + /* preconfigure: set max rx/tx fifo depths */ + au1xpsc_i2s_workdata->cfg |= + PSC_I2SCFG_RT_FIFO8 | PSC_I2SCFG_TT_FIFO8; + + /* don't wait for I2S core to become ready now; clocks may not + * be running yet; depending on clock input for PSC a wait might + * time out. + */ + + return 0; + +out1: + release_resource(au1xpsc_i2s_workdata->ioarea); + kfree(au1xpsc_i2s_workdata->ioarea); +out0: + kfree(au1xpsc_i2s_workdata); + au1xpsc_i2s_workdata = NULL; + return ret; +} + +static void au1xpsc_i2s_remove(struct platform_device *pdev, + struct snd_soc_dai *dai) +{ + au_writel(0, I2S_CFG(au1xpsc_i2s_workdata)); + au_sync(); + au_writel(PSC_CTRL_DISABLE, PSC_CTRL(au1xpsc_i2s_workdata)); + au_sync(); + + iounmap(au1xpsc_i2s_workdata->mmio); + release_resource(au1xpsc_i2s_workdata->ioarea); + kfree(au1xpsc_i2s_workdata->ioarea); + kfree(au1xpsc_i2s_workdata); + au1xpsc_i2s_workdata = NULL; +} + +static int au1xpsc_i2s_suspend(struct platform_device *pdev, + struct snd_soc_dai *cpu_dai) +{ + /* save interesting register and disable PSC */ + au1xpsc_i2s_workdata->pm[0] = + au_readl(PSC_SEL(au1xpsc_i2s_workdata)); + + au_writel(0, I2S_CFG(au1xpsc_i2s_workdata)); + au_sync(); + au_writel(PSC_CTRL_DISABLE, PSC_CTRL(au1xpsc_i2s_workdata)); + au_sync(); + + return 0; +} + +static int au1xpsc_i2s_resume(struct platform_device *pdev, + struct snd_soc_dai *cpu_dai) +{ + /* select I2S mode and PSC clock */ + au_writel(PSC_CTRL_DISABLE, PSC_CTRL(au1xpsc_i2s_workdata)); + au_sync(); + au_writel(0, PSC_SEL(au1xpsc_i2s_workdata)); + au_sync(); + au_writel(au1xpsc_i2s_workdata->pm[0], + PSC_SEL(au1xpsc_i2s_workdata)); + au_sync(); + + return 0; +} + +struct snd_soc_dai au1xpsc_i2s_dai = { + .name = "au1xpsc_i2s", + .type = SND_SOC_DAI_I2S, + .probe = au1xpsc_i2s_probe, + .remove = au1xpsc_i2s_remove, + .suspend = au1xpsc_i2s_suspend, + .resume = au1xpsc_i2s_resume, + .playback = { + .rates = AU1XPSC_I2S_RATES, + .formats = AU1XPSC_I2S_FMTS, + .channels_min = 2, + .channels_max = 8, /* 2 without external help */ + }, + .capture = { + .rates = AU1XPSC_I2S_RATES, + .formats = AU1XPSC_I2S_FMTS, + .channels_min = 2, + .channels_max = 8, /* 2 without external help */ + }, + .ops = { + .trigger = au1xpsc_i2s_trigger, + .hw_params = au1xpsc_i2s_hw_params, + }, + .dai_ops = { + .set_fmt = au1xpsc_i2s_set_fmt, + }, +}; +EXPORT_SYMBOL(au1xpsc_i2s_dai); + +static int __init au1xpsc_i2s_init(void) +{ + au1xpsc_i2s_workdata = NULL; + return 0; +} + +static void __exit au1xpsc_i2s_exit(void) +{ +} + +module_init(au1xpsc_i2s_init); +module_exit(au1xpsc_i2s_exit); + +MODULE_LICENSE("GPL"); +MODULE_DESCRIPTION("Au12x0/Au1550 PSC I2S ALSA ASoC audio driver"); +MODULE_AUTHOR("Manuel Lauss <mano@roarinelk.homelinux.net>"); diff --git a/sound/soc/au1x/psc.h b/sound/soc/au1x/psc.h new file mode 100644 index 00000000000..8fdb1a04a07 --- /dev/null +++ b/sound/soc/au1x/psc.h @@ -0,0 +1,53 @@ +/* + * Au12x0/Au1550 PSC ALSA ASoC audio support. + * + * (c) 2007-2008 MSC Vertriebsges.m.b.H., + * Manuel Lauss <mano@roarinelk.homelinux.net> + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. + * + * NOTE: all of these drivers can only work with a SINGLE instance + * of a PSC. Multiple independent audio devices are impossible + * with ASoC v1. + */ + +#ifndef _AU1X_PCM_H +#define _AU1X_PCM_H + +extern struct snd_soc_dai au1xpsc_ac97_dai; +extern struct snd_soc_dai au1xpsc_i2s_dai; +extern struct snd_soc_platform au1xpsc_soc_platform; +extern struct snd_ac97_bus_ops soc_ac97_ops; + +struct au1xpsc_audio_data { + void __iomem *mmio; + + unsigned long cfg; + unsigned long rate; + + unsigned long pm[2]; + struct resource *ioarea; +}; + +#define PCM_TX 0 +#define PCM_RX 1 + +#define SUBSTREAM_TYPE(substream) \ + ((substream)->stream == SNDRV_PCM_STREAM_PLAYBACK ? PCM_TX : PCM_RX) + +/* easy access macros */ +#define PSC_CTRL(x) ((unsigned long)((x)->mmio) + PSC_CTRL_OFFSET) +#define PSC_SEL(x) ((unsigned long)((x)->mmio) + PSC_SEL_OFFSET) +#define I2S_STAT(x) ((unsigned long)((x)->mmio) + PSC_I2SSTAT_OFFSET) +#define I2S_CFG(x) ((unsigned long)((x)->mmio) + PSC_I2SCFG_OFFSET) +#define I2S_PCR(x) ((unsigned long)((x)->mmio) + PSC_I2SPCR_OFFSET) +#define AC97_CFG(x) ((unsigned long)((x)->mmio) + PSC_AC97CFG_OFFSET) +#define AC97_CDC(x) ((unsigned long)((x)->mmio) + PSC_AC97CDC_OFFSET) +#define AC97_EVNT(x) ((unsigned long)((x)->mmio) + PSC_AC97EVNT_OFFSET) +#define AC97_PCR(x) ((unsigned long)((x)->mmio) + PSC_AC97PCR_OFFSET) +#define AC97_RST(x) ((unsigned long)((x)->mmio) + PSC_AC97RST_OFFSET) +#define AC97_STAT(x) ((unsigned long)((x)->mmio) + PSC_AC97STAT_OFFSET) + +#endif diff --git a/sound/soc/au1x/sample-ac97.c b/sound/soc/au1x/sample-ac97.c new file mode 100644 index 00000000000..f75ae7f62c3 --- /dev/null +++ b/sound/soc/au1x/sample-ac97.c @@ -0,0 +1,144 @@ +/* + * Sample Au12x0/Au1550 PSC AC97 sound machine. + * + * Copyright (c) 2007-2008 Manuel Lauss <mano@roarinelk.homelinux.net> + * + * This program is free software; you can redistribute it and/or modify + * it under the terms outlined in the file COPYING at the root of this + * source archive. + * + * This is a very generic AC97 sound machine driver for boards which + * have (AC97) audio at PSC1 (e.g. DB1200 demoboards). + */ + +#include <linux/module.h> +#include <linux/moduleparam.h> +#include <linux/timer.h> +#include <linux/interrupt.h> +#include <linux/platform_device.h> +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/soc.h> +#include <sound/soc-dapm.h> +#include <asm/mach-au1x00/au1000.h> +#include <asm/mach-au1x00/au1xxx_psc.h> +#include <asm/mach-au1x00/au1xxx_dbdma.h> + +#include "../codecs/ac97.h" +#include "psc.h" + +static int au1xpsc_sample_ac97_init(struct snd_soc_codec *codec) +{ + snd_soc_dapm_sync(codec); + return 0; +} + +static struct snd_soc_dai_link au1xpsc_sample_ac97_dai = { + .name = "AC97", + .stream_name = "AC97 HiFi", + .cpu_dai = &au1xpsc_ac97_dai, /* see psc-ac97.c */ + .codec_dai = &ac97_dai, /* see codecs/ac97.c */ + .init = au1xpsc_sample_ac97_init, + .ops = NULL, +}; + +static struct snd_soc_machine au1xpsc_sample_ac97_machine = { + .name = "Au1xxx PSC AC97 Audio", + .dai_link = &au1xpsc_sample_ac97_dai, + .num_links = 1, +}; + +static struct snd_soc_device au1xpsc_sample_ac97_devdata = { + .machine = &au1xpsc_sample_ac97_machine, + .platform = &au1xpsc_soc_platform, /* see dbdma2.c */ + .codec_dev = &soc_codec_dev_ac97, +}; + +static struct resource au1xpsc_psc1_res[] = { + [0] = { + .start = CPHYSADDR(PSC1_BASE_ADDR), + .end = CPHYSADDR(PSC1_BASE_ADDR) + 0x000fffff, + .flags = IORESOURCE_MEM, + }, + [1] = { +#ifdef CONFIG_SOC_AU1200 + .start = AU1200_PSC1_INT, + .end = AU1200_PSC1_INT, +#elif defined(CONFIG_SOC_AU1550) + .start = AU1550_PSC1_INT, + .end = AU1550_PSC1_INT, +#endif + .flags = IORESOURCE_IRQ, + }, + [2] = { + .start = DSCR_CMD0_PSC1_TX, + .end = DSCR_CMD0_PSC1_TX, + .flags = IORESOURCE_DMA, + }, + [3] = { + .start = DSCR_CMD0_PSC1_RX, + .end = DSCR_CMD0_PSC1_RX, + .flags = IORESOURCE_DMA, + }, +}; + +static struct platform_device *au1xpsc_sample_ac97_dev; + +static int __init au1xpsc_sample_ac97_load(void) +{ + int ret; + +#ifdef CONFIG_SOC_AU1200 + unsigned long io; + + /* modify sys_pinfunc for AC97 on PSC1 */ + io = au_readl(SYS_PINFUNC); + io |= SYS_PINFUNC_P1C; + io &= ~(SYS_PINFUNC_P1A | SYS_PINFUNC_P1B); + au_writel(io, SYS_PINFUNC); + au_sync(); +#endif + + ret = -ENOMEM; + + /* setup PSC clock source for AC97 part: external clock provided + * by codec. The psc-ac97.c driver depends on this setting! + */ + au_writel(PSC_SEL_CLK_SERCLK, PSC1_BASE_ADDR + PSC_SEL_OFFSET); + au_sync(); + + au1xpsc_sample_ac97_dev = platform_device_alloc("soc-audio", -1); + if (!au1xpsc_sample_ac97_dev) + goto out; + + au1xpsc_sample_ac97_dev->resource = + kmemdup(au1xpsc_psc1_res, sizeof(struct resource) * + ARRAY_SIZE(au1xpsc_psc1_res), GFP_KERNEL); + au1xpsc_sample_ac97_dev->num_resources = ARRAY_SIZE(au1xpsc_psc1_res); + au1xpsc_sample_ac97_dev->id = 1; + + platform_set_drvdata(au1xpsc_sample_ac97_dev, + &au1xpsc_sample_ac97_devdata); + au1xpsc_sample_ac97_devdata.dev = &au1xpsc_sample_ac97_dev->dev; + ret = platform_device_add(au1xpsc_sample_ac97_dev); + + if (ret) { + platform_device_put(au1xpsc_sample_ac97_dev); + au1xpsc_sample_ac97_dev = NULL; + } + +out: + return ret; +} + +static void __exit au1xpsc_sample_ac97_exit(void) +{ + platform_device_unregister(au1xpsc_sample_ac97_dev); +} + +module_init(au1xpsc_sample_ac97_load); +module_exit(au1xpsc_sample_ac97_exit); + +MODULE_LICENSE("GPL"); +MODULE_DESCRIPTION("Au1xxx PSC sample AC97 machine"); +MODULE_AUTHOR("Manuel Lauss <mano@roarinelk.homelinux.net>"); diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index 3903ab7dfa4..1db04a28a53 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -1,31 +1,37 @@ config SND_SOC_AC97_CODEC tristate - depends on SND_SOC + select SND_AC97_CODEC + +config SND_SOC_AK4535 + tristate + +config SND_SOC_UDA1380 + tristate + +config SND_SOC_WM8510 + tristate config SND_SOC_WM8731 tristate - depends on SND_SOC config SND_SOC_WM8750 tristate - depends on SND_SOC config SND_SOC_WM8753 tristate - depends on SND_SOC + +config SND_SOC_WM8990 + tristate config SND_SOC_WM9712 tristate - depends on SND_SOC config SND_SOC_WM9713 tristate - depends on SND_SOC # Cirrus Logic CS4270 Codec config SND_SOC_CS4270 tristate - depends on SND_SOC # Cirrus Logic CS4270 Codec Hardware Mute Support # Select if you have external muting circuitry attached to your CS4270. @@ -43,4 +49,4 @@ config SND_SOC_CS4270_VD33_ERRATA config SND_SOC_TLV320AIC3X tristate - depends on SND_SOC && I2C + depends on I2C diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index 4e1314c9d3e..d7b97abcf72 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -1,16 +1,24 @@ snd-soc-ac97-objs := ac97.o +snd-soc-ak4535-objs := ak4535.o +snd-soc-uda1380-objs := uda1380.o +snd-soc-wm8510-objs := wm8510.o snd-soc-wm8731-objs := wm8731.o snd-soc-wm8750-objs := wm8750.o snd-soc-wm8753-objs := wm8753.o +snd-soc-wm8990-objs := wm8990.o snd-soc-wm9712-objs := wm9712.o snd-soc-wm9713-objs := wm9713.o snd-soc-cs4270-objs := cs4270.o snd-soc-tlv320aic3x-objs := tlv320aic3x.o obj-$(CONFIG_SND_SOC_AC97_CODEC) += snd-soc-ac97.o +obj-$(CONFIG_SND_SOC_AK4535) += snd-soc-ak4535.o +obj-$(CONFIG_SND_SOC_UDA1380) += snd-soc-uda1380.o +obj-$(CONFIG_SND_SOC_WM8510) += snd-soc-wm8510.o obj-$(CONFIG_SND_SOC_WM8731) += snd-soc-wm8731.o obj-$(CONFIG_SND_SOC_WM8750) += snd-soc-wm8750.o obj-$(CONFIG_SND_SOC_WM8753) += snd-soc-wm8753.o +obj-$(CONFIG_SND_SOC_WM8990) += snd-soc-wm8990.o obj-$(CONFIG_SND_SOC_WM9712) += snd-soc-wm9712.o obj-$(CONFIG_SND_SOC_WM9713) += snd-soc-wm9713.o obj-$(CONFIG_SND_SOC_CS4270) += snd-soc-cs4270.o diff --git a/sound/soc/codecs/ac97.c b/sound/soc/codecs/ac97.c index 2a1ffe39690..61fd96ca7bc 100644 --- a/sound/soc/codecs/ac97.c +++ b/sound/soc/codecs/ac97.c @@ -10,9 +10,6 @@ * Free Software Foundation; either version 2 of the License, or (at your * option) any later version. * - * Revision history - * 17th Oct 2005 Initial version. - * * Generic AC97 support. */ @@ -24,6 +21,7 @@ #include <sound/ac97_codec.h> #include <sound/initval.h> #include <sound/soc.h> +#include "ac97.h" #define AC97_VERSION "0.6" @@ -43,7 +41,7 @@ static int ac97_prepare(struct snd_pcm_substream *substream) SNDRV_PCM_RATE_22050 | SNDRV_PCM_RATE_44100 |\ SNDRV_PCM_RATE_48000) -struct snd_soc_codec_dai ac97_dai = { +struct snd_soc_dai ac97_dai = { .name = "AC97 HiFi", .type = SND_SOC_DAI_AC97, .playback = { @@ -146,9 +144,34 @@ static int ac97_soc_remove(struct platform_device *pdev) return 0; } +#ifdef CONFIG_PM +static int ac97_soc_suspend(struct platform_device *pdev, pm_message_t msg) +{ + struct snd_soc_device *socdev = platform_get_drvdata(pdev); + + snd_ac97_suspend(socdev->codec->ac97); + + return 0; +} + +static int ac97_soc_resume(struct platform_device *pdev) +{ + struct snd_soc_device *socdev = platform_get_drvdata(pdev); + + snd_ac97_resume(socdev->codec->ac97); + + return 0; +} +#else +#define ac97_soc_suspend NULL +#define ac97_soc_resume NULL +#endif + struct snd_soc_codec_device soc_codec_dev_ac97 = { .probe = ac97_soc_probe, .remove = ac97_soc_remove, + .suspend = ac97_soc_suspend, + .resume = ac97_soc_resume, }; EXPORT_SYMBOL_GPL(soc_codec_dev_ac97); diff --git a/sound/soc/codecs/ac97.h b/sound/soc/codecs/ac97.h index 2bf6d69fd06..281aa42e2bb 100644 --- a/sound/soc/codecs/ac97.h +++ b/sound/soc/codecs/ac97.h @@ -14,6 +14,6 @@ #define __LINUX_SND_SOC_AC97_H extern struct snd_soc_codec_device soc_codec_dev_ac97; -extern struct snd_soc_codec_dai ac97_dai; +extern struct snd_soc_dai ac97_dai; #endif diff --git a/sound/soc/codecs/ak4535.c b/sound/soc/codecs/ak4535.c new file mode 100644 index 00000000000..b26003c4f3e --- /dev/null +++ b/sound/soc/codecs/ak4535.c @@ -0,0 +1,696 @@ +/* + * ak4535.c -- AK4535 ALSA Soc Audio driver + * + * Copyright 2005 Openedhand Ltd. + * + * Author: Richard Purdie <richard@openedhand.com> + * + * Based on wm8753.c by Liam Girdwood + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. + */ + +#include <linux/module.h> +#include <linux/moduleparam.h> +#include <linux/init.h> +#include <linux/delay.h> +#include <linux/pm.h> +#include <linux/i2c.h> +#include <linux/platform_device.h> +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> +#include <sound/soc-dapm.h> +#include <sound/initval.h> + +#include "ak4535.h" + +#define AUDIO_NAME "ak4535" +#define AK4535_VERSION "0.3" + +struct snd_soc_codec_device soc_codec_dev_ak4535; + +/* codec private data */ +struct ak4535_priv { + unsigned int sysclk; +}; + +/* + * ak4535 register cache + */ +static const u16 ak4535_reg[AK4535_CACHEREGNUM] = { + 0x0000, 0x0080, 0x0000, 0x0003, + 0x0002, 0x0000, 0x0011, 0x0001, + 0x0000, 0x0040, 0x0036, 0x0010, + 0x0000, 0x0000, 0x0057, 0x0000, +}; + +/* + * read ak4535 register cache + */ +static inline unsigned int ak4535_read_reg_cache(struct snd_soc_codec *codec, + unsigned int reg) +{ + u16 *cache = codec->reg_cache; + if (reg >= AK4535_CACHEREGNUM) + return -1; + return cache[reg]; +} + +static inline unsigned int ak4535_read(struct snd_soc_codec *codec, + unsigned int reg) +{ + u8 data; + data = reg; + + if (codec->hw_write(codec->control_data, &data, 1) != 1) + return -EIO; + + if (codec->hw_read(codec->control_data, &data, 1) != 1) + return -EIO; + + return data; +}; + +/* + * write ak4535 register cache + */ +static inline void ak4535_write_reg_cache(struct snd_soc_codec *codec, + u16 reg, unsigned int value) +{ + u16 *cache = codec->reg_cache; + if (reg >= AK4535_CACHEREGNUM) + return; + cache[reg] = value; +} + +/* + * write to the AK4535 register space + */ +static int ak4535_write(struct snd_soc_codec *codec, unsigned int reg, + unsigned int value) +{ + u8 data[2]; + + /* data is + * D15..D8 AK4535 register offset + * D7...D0 register data + */ + data[0] = reg & 0xff; + data[1] = value & 0xff; + + ak4535_write_reg_cache(codec, reg, value); + if (codec->hw_write(codec->control_data, data, 2) == 2) + return 0; + else + return -EIO; +} + +static int ak4535_sync(struct snd_soc_codec *codec) +{ + u16 *cache = codec->reg_cache; + int i, r = 0; + + for (i = 0; i < AK4535_CACHEREGNUM; i++) + r |= ak4535_write(codec, i, cache[i]); + + return r; +}; + +static const char *ak4535_mono_gain[] = {"+6dB", "-17dB"}; +static const char *ak4535_mono_out[] = {"(L + R)/2", "Hi-Z"}; +static const char *ak4535_hp_out[] = {"Stereo", "Mono"}; +static const char *ak4535_deemp[] = {"44.1kHz", "Off", "48kHz", "32kHz"}; +static const char *ak4535_mic_select[] = {"Internal", "External"}; + +static const struct soc_enum ak4535_enum[] = { + SOC_ENUM_SINGLE(AK4535_SIG1, 7, 2, ak4535_mono_gain), + SOC_ENUM_SINGLE(AK4535_SIG1, 6, 2, ak4535_mono_out), + SOC_ENUM_SINGLE(AK4535_MODE2, 2, 2, ak4535_hp_out), + SOC_ENUM_SINGLE(AK4535_DAC, 0, 4, ak4535_deemp), + SOC_ENUM_SINGLE(AK4535_MIC, 1, 2, ak4535_mic_select), +}; + +static const struct snd_kcontrol_new ak4535_snd_controls[] = { + SOC_SINGLE("ALC2 Switch", AK4535_SIG1, 1, 1, 0), + SOC_ENUM("Mono 1 Output", ak4535_enum[1]), + SOC_ENUM("Mono 1 Gain", ak4535_enum[0]), + SOC_ENUM("Headphone Output", ak4535_enum[2]), + SOC_ENUM("Playback Deemphasis", ak4535_enum[3]), + SOC_SINGLE("Bass Volume", AK4535_DAC, 2, 3, 0), + SOC_SINGLE("Mic Boost (+20dB) Switch", AK4535_MIC, 0, 1, 0), + SOC_ENUM("Mic Select", ak4535_enum[4]), + SOC_SINGLE("ALC Operation Time", AK4535_TIMER, 0, 3, 0), + SOC_SINGLE("ALC Recovery Time", AK4535_TIMER, 2, 3, 0), + SOC_SINGLE("ALC ZC Time", AK4535_TIMER, 4, 3, 0), + SOC_SINGLE("ALC 1 Switch", AK4535_ALC1, 5, 1, 0), + SOC_SINGLE("ALC 2 Switch", AK4535_ALC1, 6, 1, 0), + SOC_SINGLE("ALC Volume", AK4535_ALC2, 0, 127, 0), + SOC_SINGLE("Capture Volume", AK4535_PGA, 0, 127, 0), + SOC_SINGLE("Left Playback Volume", AK4535_LATT, 0, 127, 1), + SOC_SINGLE("Right Playback Volume", AK4535_RATT, 0, 127, 1), + SOC_SINGLE("AUX Bypass Volume", AK4535_VOL, 0, 15, 0), + SOC_SINGLE("Mic Sidetone Volume", AK4535_VOL, 4, 7, 0), +}; + +/* add non dapm controls */ +static int ak4535_add_controls(struct snd_soc_codec *codec) +{ + int err, i; + + for (i = 0; i < ARRAY_SIZE(ak4535_snd_controls); i++) { + err = snd_ctl_add(codec->card, + snd_soc_cnew(&ak4535_snd_controls[i], codec, NULL)); + if (err < 0) + return err; + } + + return 0; +} + +/* Mono 1 Mixer */ +static const struct snd_kcontrol_new ak4535_mono1_mixer_controls[] = { + SOC_DAPM_SINGLE("Mic Sidetone Switch", AK4535_SIG1, 4, 1, 0), + SOC_DAPM_SINGLE("Mono Playback Switch", AK4535_SIG1, 5, 1, 0), +}; + +/* Stereo Mixer */ +static const struct snd_kcontrol_new ak4535_stereo_mixer_controls[] = { + SOC_DAPM_SINGLE("Mic Sidetone Switch", AK4535_SIG2, 4, 1, 0), + SOC_DAPM_SINGLE("Playback Switch", AK4535_SIG2, 7, 1, 0), + SOC_DAPM_SINGLE("Aux Bypass Switch", AK4535_SIG2, 5, 1, 0), +}; + +/* Input Mixer */ +static const struct snd_kcontrol_new ak4535_input_mixer_controls[] = { + SOC_DAPM_SINGLE("Mic Capture Switch", AK4535_MIC, 2, 1, 0), + SOC_DAPM_SINGLE("Aux Capture Switch", AK4535_MIC, 5, 1, 0), +}; + +/* Input mux */ +static const struct snd_kcontrol_new ak4535_input_mux_control = + SOC_DAPM_ENUM("Input Select", ak4535_enum[4]); + +/* HP L switch */ +static const struct snd_kcontrol_new ak4535_hpl_control = + SOC_DAPM_SINGLE("Switch", AK4535_SIG2, 1, 1, 1); + +/* HP R switch */ +static const struct snd_kcontrol_new ak4535_hpr_control = + SOC_DAPM_SINGLE("Switch", AK4535_SIG2, 0, 1, 1); + +/* mono 2 switch */ +static const struct snd_kcontrol_new ak4535_mono2_control = + SOC_DAPM_SINGLE("Switch", AK4535_SIG1, 0, 1, 0); + +/* Line out switch */ +static const struct snd_kcontrol_new ak4535_line_control = + SOC_DAPM_SINGLE("Switch", AK4535_SIG2, 6, 1, 0); + +/* ak4535 dapm widgets */ +static const struct snd_soc_dapm_widget ak4535_dapm_widgets[] = { + SND_SOC_DAPM_MIXER("Stereo Mixer", SND_SOC_NOPM, 0, 0, + &ak4535_stereo_mixer_controls[0], + ARRAY_SIZE(ak4535_stereo_mixer_controls)), + SND_SOC_DAPM_MIXER("Mono1 Mixer", SND_SOC_NOPM, 0, 0, + &ak4535_mono1_mixer_controls[0], + ARRAY_SIZE(ak4535_mono1_mixer_controls)), + SND_SOC_DAPM_MIXER("Input Mixer", SND_SOC_NOPM, 0, 0, + &ak4535_input_mixer_controls[0], + ARRAY_SIZE(ak4535_input_mixer_controls)), + SND_SOC_DAPM_MUX("Input Mux", SND_SOC_NOPM, 0, 0, + &ak4535_input_mux_control), + SND_SOC_DAPM_DAC("DAC", "Playback", AK4535_PM2, 0, 0), + SND_SOC_DAPM_SWITCH("Mono 2 Enable", SND_SOC_NOPM, 0, 0, + &ak4535_mono2_control), + /* speaker powersave bit */ + SND_SOC_DAPM_PGA("Speaker Enable", AK4535_MODE2, 0, 0, NULL, 0), + SND_SOC_DAPM_SWITCH("Line Out Enable", SND_SOC_NOPM, 0, 0, + &ak4535_line_control), + SND_SOC_DAPM_SWITCH("Left HP Enable", SND_SOC_NOPM, 0, 0, + &ak4535_hpl_control), + SND_SOC_DAPM_SWITCH("Right HP Enable", SND_SOC_NOPM, 0, 0, + &ak4535_hpr_control), + SND_SOC_DAPM_OUTPUT("LOUT"), + SND_SOC_DAPM_OUTPUT("HPL"), + SND_SOC_DAPM_OUTPUT("ROUT"), + SND_SOC_DAPM_OUTPUT("HPR"), + SND_SOC_DAPM_OUTPUT("SPP"), + SND_SOC_DAPM_OUTPUT("SPN"), + SND_SOC_DAPM_OUTPUT("MOUT1"), + SND_SOC_DAPM_OUTPUT("MOUT2"), + SND_SOC_DAPM_OUTPUT("MICOUT"), + SND_SOC_DAPM_ADC("ADC", "Capture", AK4535_PM1, 0, 0), + SND_SOC_DAPM_PGA("Spk Amp", AK4535_PM2, 3, 0, NULL, 0), + SND_SOC_DAPM_PGA("HP R Amp", AK4535_PM2, 1, 0, NULL, 0), + SND_SOC_DAPM_PGA("HP L Amp", AK4535_PM2, 2, 0, NULL, 0), + SND_SOC_DAPM_PGA("Mic", AK4535_PM1, 1, 0, NULL, 0), + SND_SOC_DAPM_PGA("Line Out", AK4535_PM1, 4, 0, NULL, 0), + SND_SOC_DAPM_PGA("Mono Out", AK4535_PM1, 3, 0, NULL, 0), + SND_SOC_DAPM_PGA("AUX In", AK4535_PM1, 2, 0, NULL, 0), + + SND_SOC_DAPM_MICBIAS("Mic Int Bias", AK4535_MIC, 3, 0), + SND_SOC_DAPM_MICBIAS("Mic Ext Bias", AK4535_MIC, 4, 0), + SND_SOC_DAPM_INPUT("MICIN"), + SND_SOC_DAPM_INPUT("MICEXT"), + SND_SOC_DAPM_INPUT("AUX"), + SND_SOC_DAPM_INPUT("MIN"), + SND_SOC_DAPM_INPUT("AIN"), +}; + +static const struct snd_soc_dapm_route audio_map[] = { + /*stereo mixer */ + {"Stereo Mixer", "Playback Switch", "DAC"}, + {"Stereo Mixer", "Mic Sidetone Switch", "Mic"}, + {"Stereo Mixer", "Aux Bypass Switch", "AUX In"}, + + /* mono1 mixer */ + {"Mono1 Mixer", "Mic Sidetone Switch", "Mic"}, + {"Mono1 Mixer", "Mono Playback Switch", "DAC"}, + + /* Mic */ + {"Mic", NULL, "AIN"}, + {"Input Mux", "Internal", "Mic Int Bias"}, + {"Input Mux", "External", "Mic Ext Bias"}, + {"Mic Int Bias", NULL, "MICIN"}, + {"Mic Ext Bias", NULL, "MICEXT"}, + {"MICOUT", NULL, "Input Mux"}, + + /* line out */ + {"LOUT", NULL, "Line Out Enable"}, + {"ROUT", NULL, "Line Out Enable"}, + {"Line Out Enable", "Switch", "Line Out"}, + {"Line Out", NULL, "Stereo Mixer"}, + + /* mono1 out */ + {"MOUT1", NULL, "Mono Out"}, + {"Mono Out", NULL, "Mono1 Mixer"}, + + /* left HP */ + {"HPL", NULL, "Left HP Enable"}, + {"Left HP Enable", "Switch", "HP L Amp"}, + {"HP L Amp", NULL, "Stereo Mixer"}, + + /* right HP */ + {"HPR", NULL, "Right HP Enable"}, + {"Right HP Enable", "Switch", "HP R Amp"}, + {"HP R Amp", NULL, "Stereo Mixer"}, + + /* speaker */ + {"SPP", NULL, "Speaker Enable"}, + {"SPN", NULL, "Speaker Enable"}, + {"Speaker Enable", "Switch", "Spk Amp"}, + {"Spk Amp", NULL, "MIN"}, + + /* mono 2 */ + {"MOUT2", NULL, "Mono 2 Enable"}, + {"Mono 2 Enable", "Switch", "Stereo Mixer"}, + + /* Aux In */ + {"Aux In", NULL, "AUX"}, + + /* ADC */ + {"ADC", NULL, "Input Mixer"}, + {"Input Mixer", "Mic Capture Switch", "Mic"}, + {"Input Mixer", "Aux Capture Switch", "Aux In"}, +}; + +static int ak4535_add_widgets(struct snd_soc_codec *codec) +{ + snd_soc_dapm_new_controls(codec, ak4535_dapm_widgets, + ARRAY_SIZE(ak4535_dapm_widgets)); + + snd_soc_dapm_add_routes(codec, audio_map, ARRAY_SIZE(audio_map)); + + snd_soc_dapm_new_widgets(codec); + return 0; +} + +static int ak4535_set_dai_sysclk(struct snd_soc_dai *codec_dai, + int clk_id, unsigned int freq, int dir) +{ + struct snd_soc_codec *codec = codec_dai->codec; + struct ak4535_priv *ak4535 = codec->private_data; + + ak4535->sysclk = freq; + return 0; +} + +static int ak4535_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_device *socdev = rtd->socdev; + struct snd_soc_codec *codec = socdev->codec; + struct ak4535_priv *ak4535 = codec->private_data; + u8 mode2 = ak4535_read_reg_cache(codec, AK4535_MODE2) & ~(0x3 << 5); + int rate = params_rate(params), fs = 256; + + if (rate) + fs = ak4535->sysclk / rate; + + /* set fs */ + switch (fs) { + case 1024: + mode2 |= (0x2 << 5); + break; + case 512: + mode2 |= (0x1 << 5); + break; + case 256: + break; + } + + /* set rate */ + ak4535_write(codec, AK4535_MODE2, mode2); + return 0; +} + +static int ak4535_set_dai_fmt(struct snd_soc_dai *codec_dai, + unsigned int fmt) +{ + struct snd_soc_codec *codec = codec_dai->codec; + u8 mode1 = 0; + + /* interface format */ + switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { + case SND_SOC_DAIFMT_I2S: + mode1 = 0x0002; + break; + case SND_SOC_DAIFMT_LEFT_J: + mode1 = 0x0001; + break; + default: + return -EINVAL; + } + + /* use 32 fs for BCLK to save power */ + mode1 |= 0x4; + + ak4535_write(codec, AK4535_MODE1, mode1); + return 0; +} + +static int ak4535_mute(struct snd_soc_dai *dai, int mute) +{ + struct snd_soc_codec *codec = dai->codec; + u16 mute_reg = ak4535_read_reg_cache(codec, AK4535_DAC) & 0xffdf; + if (!mute) + ak4535_write(codec, AK4535_DAC, mute_reg); + else + ak4535_write(codec, AK4535_DAC, mute_reg | 0x20); + return 0; +} + +static int ak4535_set_bias_level(struct snd_soc_codec *codec, + enum snd_soc_bias_level level) +{ + u16 i; + + switch (level) { + case SND_SOC_BIAS_ON: + ak4535_mute(codec->dai, 0); + break; + case SND_SOC_BIAS_PREPARE: + ak4535_mute(codec->dai, 1); + break; + case SND_SOC_BIAS_STANDBY: + i = ak4535_read_reg_cache(codec, AK4535_PM1); + ak4535_write(codec, AK4535_PM1, i | 0x80); + i = ak4535_read_reg_cache(codec, AK4535_PM2); + ak4535_write(codec, AK4535_PM2, i & (~0x80)); + break; + case SND_SOC_BIAS_OFF: + i = ak4535_read_reg_cache(codec, AK4535_PM1); + ak4535_write(codec, AK4535_PM1, i & (~0x80)); + break; + } + codec->bias_level = level; + return 0; +} + +#define AK4535_RATES (SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_11025 |\ + SNDRV_PCM_RATE_16000 | SNDRV_PCM_RATE_22050 |\ + SNDRV_PCM_RATE_44100 | SNDRV_PCM_RATE_48000) + +struct snd_soc_dai ak4535_dai = { + .name = "AK4535", + .playback = { + .stream_name = "Playback", + .channels_min = 1, + .channels_max = 2, + .rates = AK4535_RATES, + .formats = SNDRV_PCM_FMTBIT_S16_LE,}, + .capture = { + .stream_name = "Capture", + .channels_min = 1, + .channels_max = 2, + .rates = AK4535_RATES, + .formats = SNDRV_PCM_FMTBIT_S16_LE,}, + .ops = { + .hw_params = ak4535_hw_params, + }, + .dai_ops = { + .set_fmt = ak4535_set_dai_fmt, + .digital_mute = ak4535_mute, + .set_sysclk = ak4535_set_dai_sysclk, + }, +}; +EXPORT_SYMBOL_GPL(ak4535_dai); + +static int ak4535_suspend(struct platform_device *pdev, pm_message_t state) +{ + struct snd_soc_device *socdev = platform_get_drvdata(pdev); + struct snd_soc_codec *codec = socdev->codec; + + ak4535_set_bias_level(codec, SND_SOC_BIAS_OFF); + return 0; +} + +static int ak4535_resume(struct platform_device *pdev) +{ + struct snd_soc_device *socdev = platform_get_drvdata(pdev); + struct snd_soc_codec *codec = socdev->codec; + ak4535_sync(codec); + ak4535_set_bias_level(codec, SND_SOC_BIAS_STANDBY); + ak4535_set_bias_level(codec, codec->suspend_bias_level); + return 0; +} + +/* + * initialise the AK4535 driver + * register the mixer and dsp interfaces with the kernel + */ +static int ak4535_init(struct snd_soc_device *socdev) +{ + struct snd_soc_codec *codec = socdev->codec; + int ret = 0; + + codec->name = "AK4535"; + codec->owner = THIS_MODULE; + codec->read = ak4535_read_reg_cache; + codec->write = ak4535_write; + codec->set_bias_level = ak4535_set_bias_level; + codec->dai = &ak4535_dai; + codec->num_dai = 1; + codec->reg_cache_size = ARRAY_SIZE(ak4535_reg); + codec->reg_cache = kmemdup(ak4535_reg, sizeof(ak4535_reg), GFP_KERNEL); + + if (codec->reg_cache == NULL) + return -ENOMEM; + + /* register pcms */ + ret = snd_soc_new_pcms(socdev, SNDRV_DEFAULT_IDX1, SNDRV_DEFAULT_STR1); + if (ret < 0) { + printk(KERN_ERR "ak4535: failed to create pcms\n"); + goto pcm_err; + } + + /* power on device */ + ak4535_set_bias_level(codec, SND_SOC_BIAS_STANDBY); + + ak4535_add_controls(codec); + ak4535_add_widgets(codec); + ret = snd_soc_register_card(socdev); + if (ret < 0) { + printk(KERN_ERR "ak4535: failed to register card\n"); + goto card_err; + } + + return ret; + +card_err: + snd_soc_free_pcms(socdev); + snd_soc_dapm_free(socdev); +pcm_err: + kfree(codec->reg_cache); + + return ret; +} + +static struct snd_soc_device *ak4535_socdev; + +#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE) + +#define I2C_DRIVERID_AK4535 0xfefe /* liam - need a proper id */ + +static unsigned short normal_i2c[] = { 0, I2C_CLIENT_END }; + +/* Magic definition of all other variables and things */ +I2C_CLIENT_INSMOD; + +static struct i2c_driver ak4535_i2c_driver; +static struct i2c_client client_template; + +/* If the i2c layer weren't so broken, we could pass this kind of data + around */ +static int ak4535_codec_probe(struct i2c_adapter *adap, int addr, int kind) +{ + struct snd_soc_device *socdev = ak4535_socdev; + struct ak4535_setup_data *setup = socdev->codec_data; + struct snd_soc_codec *codec = socdev->codec; + struct i2c_client *i2c; + int ret; + + if (addr != setup->i2c_address) + return -ENODEV; + + client_template.adapter = adap; + client_template.addr = addr; + + i2c = kmemdup(&client_template, sizeof(client_template), GFP_KERNEL); + if (i2c == NULL) { + kfree(codec); + return -ENOMEM; + } + i2c_set_clientdata(i2c, codec); + codec->control_data = i2c; + + ret = i2c_attach_client(i2c); + if (ret < 0) { + printk(KERN_ERR "failed to attach codec at addr %x\n", addr); + goto err; + } + + ret = ak4535_init(socdev); + if (ret < 0) { + printk(KERN_ERR "failed to initialise AK4535\n"); + goto err; + } + return ret; + +err: + kfree(codec); + kfree(i2c); + return ret; +} + +static int ak4535_i2c_detach(struct i2c_client *client) +{ + struct snd_soc_codec *codec = i2c_get_clientdata(client); + i2c_detach_client(client); + kfree(codec->reg_cache); + kfree(client); + return 0; +} + +static int ak4535_i2c_attach(struct i2c_adapter *adap) +{ + return i2c_probe(adap, &addr_data, ak4535_codec_probe); +} + +/* corgi i2c codec control layer */ +static struct i2c_driver ak4535_i2c_driver = { + .driver = { + .name = "AK4535 I2C Codec", + .owner = THIS_MODULE, + }, + .id = I2C_DRIVERID_AK4535, + .attach_adapter = ak4535_i2c_attach, + .detach_client = ak4535_i2c_detach, + .command = NULL, +}; + +static struct i2c_client client_template = { + .name = "AK4535", + .driver = &ak4535_i2c_driver, +}; +#endif + +static int ak4535_probe(struct platform_device *pdev) +{ + struct snd_soc_device *socdev = platform_get_drvdata(pdev); + struct ak4535_setup_data *setup; + struct snd_soc_codec *codec; + struct ak4535_priv *ak4535; + int ret = 0; + + printk(KERN_INFO "AK4535 Audio Codec %s", AK4535_VERSION); + + setup = socdev->codec_data; + codec = kzalloc(sizeof(struct snd_soc_codec), GFP_KERNEL); + if (codec == NULL) + return -ENOMEM; + + ak4535 = kzalloc(sizeof(struct ak4535_priv), GFP_KERNEL); + if (ak4535 == NULL) { + kfree(codec); + return -ENOMEM; + } + + codec->private_data = ak4535; + socdev->codec = codec; + mutex_init(&codec->mutex); + INIT_LIST_HEAD(&codec->dapm_widgets); + INIT_LIST_HEAD(&codec->dapm_paths); + + ak4535_socdev = socdev; +#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE) + if (setup->i2c_address) { + normal_i2c[0] = setup->i2c_address; + codec->hw_write = (hw_write_t)i2c_master_send; + codec->hw_read = (hw_read_t)i2c_master_recv; + ret = i2c_add_driver(&ak4535_i2c_driver); + if (ret != 0) + printk(KERN_ERR "can't add i2c driver"); + } +#else + /* Add other interfaces here */ +#endif + return ret; +} + +/* power down chip */ +static int ak4535_remove(struct platform_device *pdev) +{ + struct snd_soc_device *socdev = platform_get_drvdata(pdev); + struct snd_soc_codec *codec = socdev->codec; + + if (codec->control_data) + ak4535_set_bias_level(codec, SND_SOC_BIAS_OFF); + + snd_soc_free_pcms(socdev); + snd_soc_dapm_free(socdev); +#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE) + i2c_del_driver(&ak4535_i2c_driver); +#endif + kfree(codec->private_data); + kfree(codec); + + return 0; +} + +struct snd_soc_codec_device soc_codec_dev_ak4535 = { + .probe = ak4535_probe, + .remove = ak4535_remove, + .suspend = ak4535_suspend, + .resume = ak4535_resume, +}; +EXPORT_SYMBOL_GPL(soc_codec_dev_ak4535); + +MODULE_DESCRIPTION("Soc AK4535 driver"); +MODULE_AUTHOR("Richard Purdie"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/codecs/ak4535.h b/sound/soc/codecs/ak4535.h new file mode 100644 index 00000000000..e9fe30e2c05 --- /dev/null +++ b/sound/soc/codecs/ak4535.h @@ -0,0 +1,46 @@ +/* + * ak4535.h -- AK4535 Soc Audio driver + * + * Copyright 2005 Openedhand Ltd. + * + * Author: Richard Purdie <richard@openedhand.com> + * + * Based on wm8753.h + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. + */ + +#ifndef _AK4535_H +#define _AK4535_H + +/* AK4535 register space */ + +#define AK4535_PM1 0x0 +#define AK4535_PM2 0x1 +#define AK4535_SIG1 0x2 +#define AK4535_SIG2 0x3 +#define AK4535_MODE1 0x4 +#define AK4535_MODE2 0x5 +#define AK4535_DAC 0x6 +#define AK4535_MIC 0x7 +#define AK4535_TIMER 0x8 +#define AK4535_ALC1 0x9 +#define AK4535_ALC2 0xa +#define AK4535_PGA 0xb +#define AK4535_LATT 0xc +#define AK4535_RATT 0xd +#define AK4535_VOL 0xe +#define AK4535_STATUS 0xf + +#define AK4535_CACHEREGNUM 0x10 + +struct ak4535_setup_data { + unsigned short i2c_address; +}; + +extern struct snd_soc_dai ak4535_dai; +extern struct snd_soc_codec_device soc_codec_dev_ak4535; + +#endif diff --git a/sound/soc/codecs/cs4270.c b/sound/soc/codecs/cs4270.c index e73fcfd9f5c..9deb8c74fdf 100644 --- a/sound/soc/codecs/cs4270.c +++ b/sound/soc/codecs/cs4270.c @@ -201,7 +201,7 @@ static struct { * driver what the input settings can be. This would need to be implemented * for stand-alone mode to work. */ -static int cs4270_set_dai_sysclk(struct snd_soc_codec_dai *codec_dai, +static int cs4270_set_dai_sysclk(struct snd_soc_dai *codec_dai, int clk_id, unsigned int freq, int dir) { struct snd_soc_codec *codec = codec_dai->codec; @@ -251,7 +251,7 @@ static int cs4270_set_dai_sysclk(struct snd_soc_codec_dai *codec_dai, * data for playback only, but ASoC currently does not support different * formats for playback vs. record. */ -static int cs4270_set_dai_fmt(struct snd_soc_codec_dai *codec_dai, +static int cs4270_set_dai_fmt(struct snd_soc_dai *codec_dai, unsigned int format) { struct snd_soc_codec *codec = codec_dai->codec; @@ -471,7 +471,7 @@ static int cs4270_hw_params(struct snd_pcm_substream *substream, * board does not have the MUTEA or MUTEB pins connected to such circuitry, * then this function will do nothing. */ -static int cs4270_mute(struct snd_soc_codec_dai *dai, int mute) +static int cs4270_mute(struct snd_soc_dai *dai, int mute) { struct snd_soc_codec *codec = dai->codec; int reg6; @@ -667,7 +667,7 @@ error: #endif /* USE_I2C*/ -struct snd_soc_codec_dai cs4270_dai = { +struct snd_soc_dai cs4270_dai = { .name = "CS4270", .playback = { .stream_name = "Playback", diff --git a/sound/soc/codecs/cs4270.h b/sound/soc/codecs/cs4270.h index 0ced49b7804..adc6cd9667d 100644 --- a/sound/soc/codecs/cs4270.h +++ b/sound/soc/codecs/cs4270.h @@ -16,7 +16,7 @@ * The ASoC codec DAI structure for the CS4270. Assign this structure to * the .codec_dai field of your machine driver's snd_soc_dai_link structure. */ -extern struct snd_soc_codec_dai cs4270_dai; +extern struct snd_soc_dai cs4270_dai; /* * The ASoC codec device structure for the CS4270. Assign this structure diff --git a/sound/soc/codecs/tlv320aic3x.c b/sound/soc/codecs/tlv320aic3x.c index 09b1661b8a3..b1dce5f459d 100644 --- a/sound/soc/codecs/tlv320aic3x.c +++ b/sound/soc/codecs/tlv320aic3x.c @@ -29,7 +29,7 @@ * --------------------------------------- * * Hence the machine layer should disable unsupported inputs/outputs by - * snd_soc_dapm_set_endpoint(codec, "MONO_LOUT", 0), etc. + * snd_soc_dapm_disable_pin(codec, "MONO_LOUT"), etc. */ #include <linux/module.h> @@ -49,7 +49,7 @@ #include "tlv320aic3x.h" #define AUDIO_NAME "aic3x" -#define AIC3X_VERSION "0.1" +#define AIC3X_VERSION "0.2" /* codec private data */ struct aic3x_priv { @@ -138,6 +138,20 @@ static int aic3x_write(struct snd_soc_codec *codec, unsigned int reg, return -EIO; } +/* + * read from the aic3x register space + */ +static int aic3x_read(struct snd_soc_codec *codec, unsigned int reg, + u8 *value) +{ + *value = reg & 0xff; + if (codec->hw_read(codec->control_data, value, 1) != 1) + return -EIO; + + aic3x_write_reg_cache(codec, reg, *value); + return 0; +} + #define SOC_DAPM_SINGLE_AIC3X(xname, reg, shift, mask, invert) \ { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, \ .info = snd_soc_info_volsw, \ @@ -192,7 +206,7 @@ static int snd_soc_dapm_put_volsw_aic3x(struct snd_kcontrol *kcontrol, } if (found) - snd_soc_dapm_sync_endpoints(widget->codec); + snd_soc_dapm_sync(widget->codec); } ret = snd_soc_update_bits(widget->codec, reg, val_mask, val); @@ -209,6 +223,8 @@ static const char *aic3x_right_hpcom_mux[] = { "differential of HPROUT", "constant VCM", "single-ended", "differential of HPLCOM", "external feedback" }; static const char *aic3x_linein_mode_mux[] = { "single-ended", "differential" }; +static const char *aic3x_adc_hpf[] = + { "Disabled", "0.0045xFs", "0.0125xFs", "0.025xFs" }; #define LDAC_ENUM 0 #define RDAC_ENUM 1 @@ -218,6 +234,7 @@ static const char *aic3x_linein_mode_mux[] = { "single-ended", "differential" }; #define LINE1R_ENUM 5 #define LINE2L_ENUM 6 #define LINE2R_ENUM 7 +#define ADC_HPF_ENUM 8 static const struct soc_enum aic3x_enum[] = { SOC_ENUM_SINGLE(DAC_LINE_MUX, 6, 3, aic3x_left_dac_mux), @@ -228,6 +245,7 @@ static const struct soc_enum aic3x_enum[] = { SOC_ENUM_SINGLE(LINE1R_2_RADC_CTRL, 7, 2, aic3x_linein_mode_mux), SOC_ENUM_SINGLE(LINE2L_2_LADC_CTRL, 7, 2, aic3x_linein_mode_mux), SOC_ENUM_SINGLE(LINE2R_2_RADC_CTRL, 7, 2, aic3x_linein_mode_mux), + SOC_ENUM_DOUBLE(AIC3X_CODEC_DFILT_CTRL, 6, 4, 4, aic3x_adc_hpf), }; static const struct snd_kcontrol_new aic3x_snd_controls[] = { @@ -278,6 +296,8 @@ static const struct snd_kcontrol_new aic3x_snd_controls[] = { /* Input */ SOC_DOUBLE_R("PGA Capture Volume", LADC_VOL, RADC_VOL, 0, 0x7f, 0), SOC_DOUBLE_R("PGA Capture Switch", LADC_VOL, RADC_VOL, 7, 0x01, 1), + + SOC_ENUM("ADC HPF Cut-off", aic3x_enum[ADC_HPF_ENUM]), }; /* add non dapm controls */ @@ -441,11 +461,34 @@ static const struct snd_soc_dapm_widget aic3x_dapm_widgets[] = { SND_SOC_DAPM_MUX("Right Line2R Mux", SND_SOC_NOPM, 0, 0, &aic3x_right_line2_mux_controls), + /* + * Not a real mic bias widget but similar function. This is for dynamic + * control of GPIO1 digital mic modulator clock output function when + * using digital mic. + */ + SND_SOC_DAPM_REG(snd_soc_dapm_micbias, "GPIO1 dmic modclk", + AIC3X_GPIO1_REG, 4, 0xf, + AIC3X_GPIO1_FUNC_DIGITAL_MIC_MODCLK, + AIC3X_GPIO1_FUNC_DISABLED), + + /* + * Also similar function like mic bias. Selects digital mic with + * configurable oversampling rate instead of ADC converter. + */ + SND_SOC_DAPM_REG(snd_soc_dapm_micbias, "DMic Rate 128", + AIC3X_ASD_INTF_CTRLA, 0, 3, 1, 0), + SND_SOC_DAPM_REG(snd_soc_dapm_micbias, "DMic Rate 64", + AIC3X_ASD_INTF_CTRLA, 0, 3, 2, 0), + SND_SOC_DAPM_REG(snd_soc_dapm_micbias, "DMic Rate 32", + AIC3X_ASD_INTF_CTRLA, 0, 3, 3, 0), + /* Mic Bias */ - SND_SOC_DAPM_MICBIAS("Mic Bias 2V", MICBIAS_CTRL, 6, 0), - SND_SOC_DAPM_MICBIAS("Mic Bias 2.5V", MICBIAS_CTRL, 7, 0), - SND_SOC_DAPM_MICBIAS("Mic Bias AVDD", MICBIAS_CTRL, 6, 0), - SND_SOC_DAPM_MICBIAS("Mic Bias AVDD", MICBIAS_CTRL, 7, 0), + SND_SOC_DAPM_REG(snd_soc_dapm_micbias, "Mic Bias 2V", + MICBIAS_CTRL, 6, 3, 1, 0), + SND_SOC_DAPM_REG(snd_soc_dapm_micbias, "Mic Bias 2.5V", + MICBIAS_CTRL, 6, 3, 2, 0), + SND_SOC_DAPM_REG(snd_soc_dapm_micbias, "Mic Bias AVDD", + MICBIAS_CTRL, 6, 3, 3, 0), /* Left PGA to Left Output bypass */ SND_SOC_DAPM_MIXER("Left PGA Bypass Mixer", SND_SOC_NOPM, 0, 0, @@ -483,7 +526,7 @@ static const struct snd_soc_dapm_widget aic3x_dapm_widgets[] = { SND_SOC_DAPM_INPUT("LINE2R"), }; -static const char *intercon[][3] = { +static const struct snd_soc_dapm_route intercon[] = { /* Left Output */ {"Left DAC Mux", "DAC_L1", "Left DAC"}, {"Left DAC Mux", "DAC_L2", "Left DAC"}, @@ -554,6 +597,7 @@ static const char *intercon[][3] = { {"Left PGA Mixer", "Mic3L Switch", "MIC3L"}, {"Left ADC", NULL, "Left PGA Mixer"}, + {"Left ADC", NULL, "GPIO1 dmic modclk"}, /* Right Input */ {"Right Line1R Mux", "single-ended", "LINE1R"}, @@ -567,6 +611,7 @@ static const char *intercon[][3] = { {"Right PGA Mixer", "Mic3R Switch", "MIC3R"}, {"Right ADC", NULL, "Right PGA Mixer"}, + {"Right ADC", NULL, "GPIO1 dmic modclk"}, /* Left PGA Bypass */ {"Left PGA Bypass Mixer", "Line Switch", "Left PGA Mixer"}, @@ -628,101 +673,27 @@ static const char *intercon[][3] = { {"Mono Out", NULL, "Right Line2 Bypass Mixer"}, {"Right HP Out", NULL, "Right Line2 Bypass Mixer"}, - /* terminator */ - {NULL, NULL, NULL}, + /* + * Logical path between digital mic enable and GPIO1 modulator clock + * output function + */ + {"GPIO1 dmic modclk", NULL, "DMic Rate 128"}, + {"GPIO1 dmic modclk", NULL, "DMic Rate 64"}, + {"GPIO1 dmic modclk", NULL, "DMic Rate 32"}, }; static int aic3x_add_widgets(struct snd_soc_codec *codec) { - int i; - - for (i = 0; i < ARRAY_SIZE(aic3x_dapm_widgets); i++) - snd_soc_dapm_new_control(codec, &aic3x_dapm_widgets[i]); + snd_soc_dapm_new_controls(codec, aic3x_dapm_widgets, + ARRAY_SIZE(aic3x_dapm_widgets)); /* set up audio path interconnects */ - for (i = 0; intercon[i][0] != NULL; i++) - snd_soc_dapm_connect_input(codec, intercon[i][0], - intercon[i][1], intercon[i][2]); + snd_soc_dapm_add_routes(codec, intercon, ARRAY_SIZE(intercon)); snd_soc_dapm_new_widgets(codec); return 0; } -struct aic3x_rate_divs { - u32 mclk; - u32 rate; - u32 fsref_reg; - u8 sr_reg:4; - u8 pllj_reg; - u16 plld_reg; -}; - -/* AIC3X codec mclk clock divider coefficients */ -static const struct aic3x_rate_divs aic3x_divs[] = { - /* 8k */ - {12000000, 8000, 48000, 0xa, 16, 3840}, - {19200000, 8000, 48000, 0xa, 10, 2400}, - {22579200, 8000, 48000, 0xa, 8, 7075}, - {33868800, 8000, 48000, 0xa, 5, 8049}, - /* 11.025k */ - {12000000, 11025, 44100, 0x6, 15, 528}, - {19200000, 11025, 44100, 0x6, 9, 4080}, - {22579200, 11025, 44100, 0x6, 8, 0}, - {33868800, 11025, 44100, 0x6, 5, 3333}, - /* 16k */ - {12000000, 16000, 48000, 0x4, 16, 3840}, - {19200000, 16000, 48000, 0x4, 10, 2400}, - {22579200, 16000, 48000, 0x4, 8, 7075}, - {33868800, 16000, 48000, 0x4, 5, 8049}, - /* 22.05k */ - {12000000, 22050, 44100, 0x2, 15, 528}, - {19200000, 22050, 44100, 0x2, 9, 4080}, - {22579200, 22050, 44100, 0x2, 8, 0}, - {33868800, 22050, 44100, 0x2, 5, 3333}, - /* 32k */ - {12000000, 32000, 48000, 0x1, 16, 3840}, - {19200000, 32000, 48000, 0x1, 10, 2400}, - {22579200, 32000, 48000, 0x1, 8, 7075}, - {33868800, 32000, 48000, 0x1, 5, 8049}, - /* 44.1k */ - {12000000, 44100, 44100, 0x0, 15, 528}, - {19200000, 44100, 44100, 0x0, 9, 4080}, - {22579200, 44100, 44100, 0x0, 8, 0}, - {33868800, 44100, 44100, 0x0, 5, 3333}, - /* 48k */ - {12000000, 48000, 48000, 0x0, 16, 3840}, - {19200000, 48000, 48000, 0x0, 10, 2400}, - {22579200, 48000, 48000, 0x0, 8, 7075}, - {33868800, 48000, 48000, 0x0, 5, 8049}, - /* 64k */ - {12000000, 64000, 96000, 0x1, 16, 3840}, - {19200000, 64000, 96000, 0x1, 10, 2400}, - {22579200, 64000, 96000, 0x1, 8, 7075}, - {33868800, 64000, 96000, 0x1, 5, 8049}, - /* 88.2k */ - {12000000, 88200, 88200, 0x0, 15, 528}, - {19200000, 88200, 88200, 0x0, 9, 4080}, - {22579200, 88200, 88200, 0x0, 8, 0}, - {33868800, 88200, 88200, 0x0, 5, 3333}, - /* 96k */ - {12000000, 96000, 96000, 0x0, 16, 3840}, - {19200000, 96000, 96000, 0x0, 10, 2400}, - {22579200, 96000, 96000, 0x0, 8, 7075}, - {33868800, 96000, 96000, 0x0, 5, 8049}, -}; - -static inline int aic3x_get_divs(int mclk, int rate) -{ - int i; - - for (i = 0; i < ARRAY_SIZE(aic3x_divs); i++) { - if (aic3x_divs[i].rate == rate && aic3x_divs[i].mclk == mclk) - return i; - } - - return 0; -} - static int aic3x_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params) { @@ -730,49 +701,107 @@ static int aic3x_hw_params(struct snd_pcm_substream *substream, struct snd_soc_device *socdev = rtd->socdev; struct snd_soc_codec *codec = socdev->codec; struct aic3x_priv *aic3x = codec->private_data; - int i; - u8 data, pll_p, pll_r, pll_j; - u16 pll_d; - - i = aic3x_get_divs(aic3x->sysclk, params_rate(params)); + int codec_clk = 0, bypass_pll = 0, fsref, last_clk = 0; + u8 data, r, p, pll_q, pll_p = 1, pll_r = 1, pll_j = 1; + u16 pll_d = 1; - /* Route Left DAC to left channel input and - * right DAC to right channel input */ - data = (LDAC2LCH | RDAC2RCH); - switch (aic3x_divs[i].fsref_reg) { - case 44100: - data |= FSREF_44100; + /* select data word length */ + data = + aic3x_read_reg_cache(codec, AIC3X_ASD_INTF_CTRLB) & (~(0x3 << 4)); + switch (params_format(params)) { + case SNDRV_PCM_FORMAT_S16_LE: break; - case 48000: - data |= FSREF_48000; + case SNDRV_PCM_FORMAT_S20_3LE: + data |= (0x01 << 4); break; - case 88200: - data |= FSREF_44100 | DUAL_RATE_MODE; + case SNDRV_PCM_FORMAT_S24_LE: + data |= (0x02 << 4); break; - case 96000: - data |= FSREF_48000 | DUAL_RATE_MODE; + case SNDRV_PCM_FORMAT_S32_LE: + data |= (0x03 << 4); break; } + aic3x_write(codec, AIC3X_ASD_INTF_CTRLB, data); + + /* Fsref can be 44100 or 48000 */ + fsref = (params_rate(params) % 11025 == 0) ? 44100 : 48000; + + /* Try to find a value for Q which allows us to bypass the PLL and + * generate CODEC_CLK directly. */ + for (pll_q = 2; pll_q < 18; pll_q++) + if (aic3x->sysclk / (128 * pll_q) == fsref) { + bypass_pll = 1; + break; + } + + if (bypass_pll) { + pll_q &= 0xf; + aic3x_write(codec, AIC3X_PLL_PROGA_REG, pll_q << PLLQ_SHIFT); + aic3x_write(codec, AIC3X_GPIOB_REG, CODEC_CLKIN_CLKDIV); + } else + aic3x_write(codec, AIC3X_GPIOB_REG, CODEC_CLKIN_PLLDIV); + + /* Route Left DAC to left channel input and + * right DAC to right channel input */ + data = (LDAC2LCH | RDAC2RCH); + data |= (fsref == 44100) ? FSREF_44100 : FSREF_48000; + if (params_rate(params) >= 64000) + data |= DUAL_RATE_MODE; aic3x_write(codec, AIC3X_CODEC_DATAPATH_REG, data); /* codec sample rate select */ - data = aic3x_divs[i].sr_reg; + data = (fsref * 20) / params_rate(params); + if (params_rate(params) < 64000) + data /= 2; + data /= 5; + data -= 2; data |= (data << 4); aic3x_write(codec, AIC3X_SAMPLE_RATE_SEL_REG, data); - /* Use PLL for generation Fsref by equation: - * Fsref = (MCLK * K * R)/(2048 * P); - * Fix P = 2 and R = 1 and calculate K, if - * K = J.D, i.e. J - an interger portion of K and D is the fractional - * one with 4 digits of precision; - * Example: - * For MCLK = 22.5792 MHz and Fsref = 48kHz: - * Select P = 2, R= 1, K = 8.7074, which results in J = 8, D = 7074 + if (bypass_pll) + return 0; + + /* Use PLL + * find an apropriate setup for j, d, r and p by iterating over + * p and r - j and d are calculated for each fraction. + * Up to 128 values are probed, the closest one wins the game. + * The sysclk is divided by 1000 to prevent integer overflows. */ - pll_p = 2; - pll_r = 1; - pll_j = aic3x_divs[i].pllj_reg; - pll_d = aic3x_divs[i].plld_reg; + codec_clk = (2048 * fsref) / (aic3x->sysclk / 1000); + + for (r = 1; r <= 16; r++) + for (p = 1; p <= 8; p++) { + int clk, tmp = (codec_clk * pll_r * 10) / pll_p; + u8 j = tmp / 10000; + u16 d = tmp % 10000; + + if (j > 63) + continue; + + if (d != 0 && aic3x->sysclk < 10000000) + continue; + + /* This is actually 1000 * ((j + (d/10000)) * r) / p + * The term had to be converted to get rid of the + * division by 10000 */ + clk = ((10000 * j * r) + (d * r)) / (10 * p); + + /* check whether this values get closer than the best + * ones we had before */ + if (abs(codec_clk - clk) < abs(codec_clk - last_clk)) { + pll_j = j; pll_d = d; pll_r = r; pll_p = p; + last_clk = clk; + } + + /* Early exit for exact matches */ + if (clk == codec_clk) + break; + } + + if (last_clk == 0) { + printk(KERN_ERR "%s(): unable to setup PLL\n", __func__); + return -EINVAL; + } data = aic3x_read_reg_cache(codec, AIC3X_PLL_PROGA_REG); aic3x_write(codec, AIC3X_PLL_PROGA_REG, data | (pll_p << PLLP_SHIFT)); @@ -782,28 +811,10 @@ static int aic3x_hw_params(struct snd_pcm_substream *substream, aic3x_write(codec, AIC3X_PLL_PROGD_REG, (pll_d & 0x3F) << PLLD_LSB_SHIFT); - /* select data word length */ - data = - aic3x_read_reg_cache(codec, AIC3X_ASD_INTF_CTRLB) & (~(0x3 << 4)); - switch (params_format(params)) { - case SNDRV_PCM_FORMAT_S16_LE: - break; - case SNDRV_PCM_FORMAT_S20_3LE: - data |= (0x01 << 4); - break; - case SNDRV_PCM_FORMAT_S24_LE: - data |= (0x02 << 4); - break; - case SNDRV_PCM_FORMAT_S32_LE: - data |= (0x03 << 4); - break; - } - aic3x_write(codec, AIC3X_ASD_INTF_CTRLB, data); - return 0; } -static int aic3x_mute(struct snd_soc_codec_dai *dai, int mute) +static int aic3x_mute(struct snd_soc_dai *dai, int mute) { struct snd_soc_codec *codec = dai->codec; u8 ldac_reg = aic3x_read_reg_cache(codec, LDAC_VOL) & ~MUTE_ON; @@ -820,31 +831,25 @@ static int aic3x_mute(struct snd_soc_codec_dai *dai, int mute) return 0; } -static int aic3x_set_dai_sysclk(struct snd_soc_codec_dai *codec_dai, +static int aic3x_set_dai_sysclk(struct snd_soc_dai *codec_dai, int clk_id, unsigned int freq, int dir) { struct snd_soc_codec *codec = codec_dai->codec; struct aic3x_priv *aic3x = codec->private_data; - switch (freq) { - case 12000000: - case 19200000: - case 22579200: - case 33868800: - aic3x->sysclk = freq; - return 0; - } - - return -EINVAL; + aic3x->sysclk = freq; + return 0; } -static int aic3x_set_dai_fmt(struct snd_soc_codec_dai *codec_dai, +static int aic3x_set_dai_fmt(struct snd_soc_dai *codec_dai, unsigned int fmt) { struct snd_soc_codec *codec = codec_dai->codec; struct aic3x_priv *aic3x = codec->private_data; - u8 iface_areg = 0; - u8 iface_breg = 0; + u8 iface_areg, iface_breg; + + iface_areg = aic3x_read_reg_cache(codec, AIC3X_ASD_INTF_CTRLA) & 0x3f; + iface_breg = aic3x_read_reg_cache(codec, AIC3X_ASD_INTF_CTRLB) & 0x3f; /* set master/slave audio interface */ switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { @@ -883,13 +888,14 @@ static int aic3x_set_dai_fmt(struct snd_soc_codec_dai *codec_dai, return 0; } -static int aic3x_dapm_event(struct snd_soc_codec *codec, int event) +static int aic3x_set_bias_level(struct snd_soc_codec *codec, + enum snd_soc_bias_level level) { struct aic3x_priv *aic3x = codec->private_data; u8 reg; - switch (event) { - case SNDRV_CTL_POWER_D0: + switch (level) { + case SND_SOC_BIAS_ON: /* all power is driven by DAPM system */ if (aic3x->master) { /* enable pll */ @@ -898,10 +904,9 @@ static int aic3x_dapm_event(struct snd_soc_codec *codec, int event) reg | PLL_ENABLE); } break; - case SNDRV_CTL_POWER_D1: - case SNDRV_CTL_POWER_D2: + case SND_SOC_BIAS_PREPARE: break; - case SNDRV_CTL_POWER_D3hot: + case SND_SOC_BIAS_STANDBY: /* * all power is driven by DAPM system, * so output power is safe if bypass was set @@ -913,7 +918,7 @@ static int aic3x_dapm_event(struct snd_soc_codec *codec, int event) reg & ~PLL_ENABLE); } break; - case SNDRV_CTL_POWER_D3cold: + case SND_SOC_BIAS_OFF: /* force all power off */ reg = aic3x_read_reg_cache(codec, LINE1L_2_LADC_CTRL); aic3x_write(codec, LINE1L_2_LADC_CTRL, reg & ~LADC_PWR_ON); @@ -949,16 +954,43 @@ static int aic3x_dapm_event(struct snd_soc_codec *codec, int event) } break; } - codec->dapm_state = event; + codec->bias_level = level; return 0; } +void aic3x_set_gpio(struct snd_soc_codec *codec, int gpio, int state) +{ + u8 reg = gpio ? AIC3X_GPIO2_REG : AIC3X_GPIO1_REG; + u8 bit = gpio ? 3: 0; + u8 val = aic3x_read_reg_cache(codec, reg) & ~(1 << bit); + aic3x_write(codec, reg, val | (!!state << bit)); +} +EXPORT_SYMBOL_GPL(aic3x_set_gpio); + +int aic3x_get_gpio(struct snd_soc_codec *codec, int gpio) +{ + u8 reg = gpio ? AIC3X_GPIO2_REG : AIC3X_GPIO1_REG; + u8 val, bit = gpio ? 2: 1; + + aic3x_read(codec, reg, &val); + return (val >> bit) & 1; +} +EXPORT_SYMBOL_GPL(aic3x_get_gpio); + +int aic3x_headset_detected(struct snd_soc_codec *codec) +{ + u8 val; + aic3x_read(codec, AIC3X_RT_IRQ_FLAGS_REG, &val); + return (val >> 2) & 1; +} +EXPORT_SYMBOL_GPL(aic3x_headset_detected); + #define AIC3X_RATES SNDRV_PCM_RATE_8000_96000 #define AIC3X_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE | \ SNDRV_PCM_FMTBIT_S24_3LE | SNDRV_PCM_FMTBIT_S32_LE) -struct snd_soc_codec_dai aic3x_dai = { +struct snd_soc_dai aic3x_dai = { .name = "aic3x", .playback = { .stream_name = "Playback", @@ -988,7 +1020,7 @@ static int aic3x_suspend(struct platform_device *pdev, pm_message_t state) struct snd_soc_device *socdev = platform_get_drvdata(pdev); struct snd_soc_codec *codec = socdev->codec; - aic3x_dapm_event(codec, SNDRV_CTL_POWER_D3cold); + aic3x_set_bias_level(codec, SND_SOC_BIAS_OFF); return 0; } @@ -1008,7 +1040,7 @@ static int aic3x_resume(struct platform_device *pdev) codec->hw_write(codec->control_data, data, 2); } - aic3x_dapm_event(codec, codec->suspend_dapm_state); + aic3x_set_bias_level(codec, codec->suspend_bias_level); return 0; } @@ -1020,16 +1052,17 @@ static int aic3x_resume(struct platform_device *pdev) static int aic3x_init(struct snd_soc_device *socdev) { struct snd_soc_codec *codec = socdev->codec; + struct aic3x_setup_data *setup = socdev->codec_data; int reg, ret = 0; codec->name = "aic3x"; codec->owner = THIS_MODULE; codec->read = aic3x_read_reg_cache; codec->write = aic3x_write; - codec->dapm_event = aic3x_dapm_event; + codec->set_bias_level = aic3x_set_bias_level; codec->dai = &aic3x_dai; codec->num_dai = 1; - codec->reg_cache_size = sizeof(aic3x_reg); + codec->reg_cache_size = ARRAY_SIZE(aic3x_reg); codec->reg_cache = kmemdup(aic3x_reg, sizeof(aic3x_reg), GFP_KERNEL); if (codec->reg_cache == NULL) return -ENOMEM; @@ -1108,7 +1141,11 @@ static int aic3x_init(struct snd_soc_device *socdev) aic3x_write(codec, LINE2R_2_MONOLOPM_VOL, DEFAULT_VOL); /* off, with power on */ - aic3x_dapm_event(codec, SNDRV_CTL_POWER_D3hot); + aic3x_set_bias_level(codec, SND_SOC_BIAS_STANDBY); + + /* setup GPIO functions */ + aic3x_write(codec, AIC3X_GPIO1_REG, (setup->gpio_func[0] & 0xf) << 4); + aic3x_write(codec, AIC3X_GPIO2_REG, (setup->gpio_func[1] & 0xf) << 4); aic3x_add_controls(codec); aic3x_add_widgets(codec); @@ -1217,6 +1254,12 @@ static struct i2c_client client_template = { .name = "AIC3X", .driver = &aic3x_i2c_driver, }; + +static int aic3x_i2c_read(struct i2c_client *client, u8 *value, int len) +{ + value[0] = i2c_smbus_read_byte_data(client, value[0]); + return (len == 1); +} #endif static int aic3x_probe(struct platform_device *pdev) @@ -1251,6 +1294,7 @@ static int aic3x_probe(struct platform_device *pdev) if (setup->i2c_address) { normal_i2c[0] = setup->i2c_address; codec->hw_write = (hw_write_t) i2c_master_send; + codec->hw_read = (hw_read_t) aic3x_i2c_read; ret = i2c_add_driver(&aic3x_i2c_driver); if (ret != 0) printk(KERN_ERR "can't add i2c driver"); @@ -1268,7 +1312,7 @@ static int aic3x_remove(struct platform_device *pdev) /* power down chip */ if (codec->control_data) - aic3x_dapm_event(codec, SNDRV_CTL_POWER_D3); + aic3x_set_bias_level(codec, SND_SOC_BIAS_OFF); snd_soc_free_pcms(socdev); snd_soc_dapm_free(socdev); diff --git a/sound/soc/codecs/tlv320aic3x.h b/sound/soc/codecs/tlv320aic3x.h index d0cdeeb629d..d76c079b86e 100644 --- a/sound/soc/codecs/tlv320aic3x.h +++ b/sound/soc/codecs/tlv320aic3x.h @@ -37,6 +37,8 @@ #define AIC3X_ASD_INTF_CTRLB 9 /* Audio overflow status and PLL R value programming register */ #define AIC3X_OVRF_STATUS_AND_PLLR_REG 11 +/* Audio codec digital filter control register */ +#define AIC3X_CODEC_DFILT_CTRL 12 /* ADC PGA Gain control registers */ #define LADC_VOL 15 @@ -108,6 +110,13 @@ #define DACR1_2_RLOPM_VOL 92 #define LLOPM_CTRL 86 #define RLOPM_CTRL 93 +/* GPIO/IRQ registers */ +#define AIC3X_STICKY_IRQ_FLAGS_REG 96 +#define AIC3X_RT_IRQ_FLAGS_REG 97 +#define AIC3X_GPIO1_REG 98 +#define AIC3X_GPIO2_REG 99 +#define AIC3X_GPIOA_REG 100 +#define AIC3X_GPIOB_REG 101 /* Clock generation control register */ #define AIC3X_CLKGEN_CTRL_REG 102 @@ -128,12 +137,15 @@ /* PLL registers bitfields */ #define PLLP_SHIFT 0 +#define PLLQ_SHIFT 3 #define PLLR_SHIFT 0 #define PLLJ_SHIFT 2 #define PLLD_MSB_SHIFT 0 #define PLLD_LSB_SHIFT 2 /* Clock generation register bits */ +#define CODEC_CLKIN_PLLDIV 0 +#define CODEC_CLKIN_CLKDIV 1 #define PLL_CLKIN_SHIFT 4 #define MCLK_SOURCE 0x0 #define PLL_CLKDIV_SHIFT 0 @@ -171,11 +183,52 @@ /* Default input volume */ #define DEFAULT_GAIN 0x20 +/* GPIO API */ +enum { + AIC3X_GPIO1_FUNC_DISABLED = 0, + AIC3X_GPIO1_FUNC_AUDIO_WORDCLK_ADC = 1, + AIC3X_GPIO1_FUNC_CLOCK_MUX = 2, + AIC3X_GPIO1_FUNC_CLOCK_MUX_DIV2 = 3, + AIC3X_GPIO1_FUNC_CLOCK_MUX_DIV4 = 4, + AIC3X_GPIO1_FUNC_CLOCK_MUX_DIV8 = 5, + AIC3X_GPIO1_FUNC_SHORT_CIRCUIT_IRQ = 6, + AIC3X_GPIO1_FUNC_AGC_NOISE_IRQ = 7, + AIC3X_GPIO1_FUNC_INPUT = 8, + AIC3X_GPIO1_FUNC_OUTPUT = 9, + AIC3X_GPIO1_FUNC_DIGITAL_MIC_MODCLK = 10, + AIC3X_GPIO1_FUNC_AUDIO_WORDCLK = 11, + AIC3X_GPIO1_FUNC_BUTTON_IRQ = 12, + AIC3X_GPIO1_FUNC_HEADSET_DETECT_IRQ = 13, + AIC3X_GPIO1_FUNC_HEADSET_DETECT_OR_BUTTON_IRQ = 14, + AIC3X_GPIO1_FUNC_ALL_IRQ = 16 +}; + +enum { + AIC3X_GPIO2_FUNC_DISABLED = 0, + AIC3X_GPIO2_FUNC_HEADSET_DETECT_IRQ = 2, + AIC3X_GPIO2_FUNC_INPUT = 3, + AIC3X_GPIO2_FUNC_OUTPUT = 4, + AIC3X_GPIO2_FUNC_DIGITAL_MIC_INPUT = 5, + AIC3X_GPIO2_FUNC_AUDIO_BITCLK = 8, + AIC3X_GPIO2_FUNC_HEADSET_DETECT_OR_BUTTON_IRQ = 9, + AIC3X_GPIO2_FUNC_ALL_IRQ = 10, + AIC3X_GPIO2_FUNC_SHORT_CIRCUIT_OR_AGC_IRQ = 11, + AIC3X_GPIO2_FUNC_HEADSET_OR_BUTTON_PRESS_OR_SHORT_CIRCUIT_IRQ = 12, + AIC3X_GPIO2_FUNC_SHORT_CIRCUIT_IRQ = 13, + AIC3X_GPIO2_FUNC_AGC_NOISE_IRQ = 14, + AIC3X_GPIO2_FUNC_BUTTON_PRESS_IRQ = 15 +}; + +void aic3x_set_gpio(struct snd_soc_codec *codec, int gpio, int state); +int aic3x_get_gpio(struct snd_soc_codec *codec, int gpio); +int aic3x_headset_detected(struct snd_soc_codec *codec); + struct aic3x_setup_data { unsigned short i2c_address; + unsigned int gpio_func[2]; }; -extern struct snd_soc_codec_dai aic3x_dai; +extern struct snd_soc_dai aic3x_dai; extern struct snd_soc_codec_device soc_codec_dev_aic3x; #endif /* _AIC3X_H */ diff --git a/sound/soc/codecs/uda1380.c b/sound/soc/codecs/uda1380.c new file mode 100644 index 00000000000..a52d6d9e007 --- /dev/null +++ b/sound/soc/codecs/uda1380.c @@ -0,0 +1,852 @@ +/* + * uda1380.c - Philips UDA1380 ALSA SoC audio driver + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. + * + * Copyright (c) 2007 Philipp Zabel <philipp.zabel@gmail.com> + * Improved support for DAPM and audio routing/mixing capabilities, + * added TLV support. + * + * Modified by Richard Purdie <richard@openedhand.com> to fit into SoC + * codec model. + * + * Copyright (c) 2005 Giorgio Padrin <giorgio@mandarinlogiq.org> + * Copyright 2005 Openedhand Ltd. + */ + +#include <linux/module.h> +#include <linux/init.h> +#include <linux/types.h> +#include <linux/string.h> +#include <linux/slab.h> +#include <linux/errno.h> +#include <linux/ioctl.h> +#include <linux/delay.h> +#include <linux/i2c.h> +#include <sound/core.h> +#include <sound/control.h> +#include <sound/initval.h> +#include <sound/info.h> +#include <sound/soc.h> +#include <sound/soc-dapm.h> +#include <sound/tlv.h> + +#include "uda1380.h" + +#define UDA1380_VERSION "0.6" +#define AUDIO_NAME "uda1380" + +/* + * uda1380 register cache + */ +static const u16 uda1380_reg[UDA1380_CACHEREGNUM] = { + 0x0502, 0x0000, 0x0000, 0x3f3f, + 0x0202, 0x0000, 0x0000, 0x0000, + 0x0000, 0x0000, 0x0000, 0x0000, + 0x0000, 0x0000, 0x0000, 0x0000, + 0x0000, 0xff00, 0x0000, 0x4800, + 0x0000, 0x0000, 0x0000, 0x0000, + 0x0000, 0x0000, 0x0000, 0x0000, + 0x0000, 0x0000, 0x0000, 0x0000, + 0x0000, 0x8000, 0x0002, 0x0000, +}; + +/* + * read uda1380 register cache + */ +static inline unsigned int uda1380_read_reg_cache(struct snd_soc_codec *codec, + unsigned int reg) +{ + u16 *cache = codec->reg_cache; + if (reg == UDA1380_RESET) + return 0; + if (reg >= UDA1380_CACHEREGNUM) + return -1; + return cache[reg]; +} + +/* + * write uda1380 register cache + */ +static inline void uda1380_write_reg_cache(struct snd_soc_codec *codec, + u16 reg, unsigned int value) +{ + u16 *cache = codec->reg_cache; + if (reg >= UDA1380_CACHEREGNUM) + return; + cache[reg] = value; +} + +/* + * write to the UDA1380 register space + */ +static int uda1380_write(struct snd_soc_codec *codec, unsigned int reg, + unsigned int value) +{ + u8 data[3]; + + /* data is + * data[0] is register offset + * data[1] is MS byte + * data[2] is LS byte + */ + data[0] = reg; + data[1] = (value & 0xff00) >> 8; + data[2] = value & 0x00ff; + + uda1380_write_reg_cache(codec, reg, value); + + /* the interpolator & decimator regs must only be written when the + * codec DAI is active. + */ + if (!codec->active && (reg >= UDA1380_MVOL)) + return 0; + pr_debug("uda1380: hw write %x val %x\n", reg, value); + if (codec->hw_write(codec->control_data, data, 3) == 3) { + unsigned int val; + i2c_master_send(codec->control_data, data, 1); + i2c_master_recv(codec->control_data, data, 2); + val = (data[0]<<8) | data[1]; + if (val != value) { + pr_debug("uda1380: READ BACK VAL %x\n", + (data[0]<<8) | data[1]); + return -EIO; + } + return 0; + } else + return -EIO; +} + +#define uda1380_reset(c) uda1380_write(c, UDA1380_RESET, 0) + +/* declarations of ALSA reg_elem_REAL controls */ +static const char *uda1380_deemp[] = { + "None", + "32kHz", + "44.1kHz", + "48kHz", + "96kHz", +}; +static const char *uda1380_input_sel[] = { + "Line", + "Mic + Line R", + "Line L", + "Mic", +}; +static const char *uda1380_output_sel[] = { + "DAC", + "Analog Mixer", +}; +static const char *uda1380_spf_mode[] = { + "Flat", + "Minimum1", + "Minimum2", + "Maximum" +}; +static const char *uda1380_capture_sel[] = { + "ADC", + "Digital Mixer" +}; +static const char *uda1380_sel_ns[] = { + "3rd-order", + "5th-order" +}; +static const char *uda1380_mix_control[] = { + "off", + "PCM only", + "before sound processing", + "after sound processing" +}; +static const char *uda1380_sdet_setting[] = { + "3200", + "4800", + "9600", + "19200" +}; +static const char *uda1380_os_setting[] = { + "single-speed", + "double-speed (no mixing)", + "quad-speed (no mixing)" +}; + +static const struct soc_enum uda1380_deemp_enum[] = { + SOC_ENUM_SINGLE(UDA1380_DEEMP, 8, 5, uda1380_deemp), + SOC_ENUM_SINGLE(UDA1380_DEEMP, 0, 5, uda1380_deemp), +}; +static const struct soc_enum uda1380_input_sel_enum = + SOC_ENUM_SINGLE(UDA1380_ADC, 2, 4, uda1380_input_sel); /* SEL_MIC, SEL_LNA */ +static const struct soc_enum uda1380_output_sel_enum = + SOC_ENUM_SINGLE(UDA1380_PM, 7, 2, uda1380_output_sel); /* R02_EN_AVC */ +static const struct soc_enum uda1380_spf_enum = + SOC_ENUM_SINGLE(UDA1380_MODE, 14, 4, uda1380_spf_mode); /* M */ +static const struct soc_enum uda1380_capture_sel_enum = + SOC_ENUM_SINGLE(UDA1380_IFACE, 6, 2, uda1380_capture_sel); /* SEL_SOURCE */ +static const struct soc_enum uda1380_sel_ns_enum = + SOC_ENUM_SINGLE(UDA1380_MIXER, 14, 2, uda1380_sel_ns); /* SEL_NS */ +static const struct soc_enum uda1380_mix_enum = + SOC_ENUM_SINGLE(UDA1380_MIXER, 12, 4, uda1380_mix_control); /* MIX, MIX_POS */ +static const struct soc_enum uda1380_sdet_enum = + SOC_ENUM_SINGLE(UDA1380_MIXER, 4, 4, uda1380_sdet_setting); /* SD_VALUE */ +static const struct soc_enum uda1380_os_enum = + SOC_ENUM_SINGLE(UDA1380_MIXER, 0, 3, uda1380_os_setting); /* OS */ + +/* + * from -48 dB in 1.5 dB steps (mute instead of -49.5 dB) + */ +static DECLARE_TLV_DB_SCALE(amix_tlv, -4950, 150, 1); + +/* + * from -78 dB in 1 dB steps (3 dB steps, really. LSB are ignored), + * from -66 dB in 0.5 dB steps (2 dB steps, really) and + * from -52 dB in 0.25 dB steps + */ +static const unsigned int mvol_tlv[] = { + TLV_DB_RANGE_HEAD(3), + 0, 15, TLV_DB_SCALE_ITEM(-8200, 100, 1), + 16, 43, TLV_DB_SCALE_ITEM(-6600, 50, 0), + 44, 252, TLV_DB_SCALE_ITEM(-5200, 25, 0), +}; + +/* + * from -72 dB in 1.5 dB steps (6 dB steps really), + * from -66 dB in 0.75 dB steps (3 dB steps really), + * from -60 dB in 0.5 dB steps (2 dB steps really) and + * from -46 dB in 0.25 dB steps + */ +static const unsigned int vc_tlv[] = { + TLV_DB_RANGE_HEAD(4), + 0, 7, TLV_DB_SCALE_ITEM(-7800, 150, 1), + 8, 15, TLV_DB_SCALE_ITEM(-6600, 75, 0), + 16, 43, TLV_DB_SCALE_ITEM(-6000, 50, 0), + 44, 228, TLV_DB_SCALE_ITEM(-4600, 25, 0), +}; + +/* from 0 to 6 dB in 2 dB steps if SPF mode != flat */ +static DECLARE_TLV_DB_SCALE(tr_tlv, 0, 200, 0); + +/* from 0 to 24 dB in 2 dB steps, if SPF mode == maximum, otherwise cuts + * off at 18 dB max) */ +static DECLARE_TLV_DB_SCALE(bb_tlv, 0, 200, 0); + +/* from -63 to 24 dB in 0.5 dB steps (-128...48) */ +static DECLARE_TLV_DB_SCALE(dec_tlv, -6400, 50, 1); + +/* from 0 to 24 dB in 3 dB steps */ +static DECLARE_TLV_DB_SCALE(pga_tlv, 0, 300, 0); + +/* from 0 to 30 dB in 2 dB steps */ +static DECLARE_TLV_DB_SCALE(vga_tlv, 0, 200, 0); + +static const struct snd_kcontrol_new uda1380_snd_controls[] = { + SOC_DOUBLE_TLV("Analog Mixer Volume", UDA1380_AMIX, 0, 8, 44, 1, amix_tlv), /* AVCR, AVCL */ + SOC_DOUBLE_TLV("Master Playback Volume", UDA1380_MVOL, 0, 8, 252, 1, mvol_tlv), /* MVCL, MVCR */ + SOC_SINGLE_TLV("ADC Playback Volume", UDA1380_MIXVOL, 8, 228, 1, vc_tlv), /* VC2 */ + SOC_SINGLE_TLV("PCM Playback Volume", UDA1380_MIXVOL, 0, 228, 1, vc_tlv), /* VC1 */ + SOC_ENUM("Sound Processing Filter", uda1380_spf_enum), /* M */ + SOC_DOUBLE_TLV("Tone Control - Treble", UDA1380_MODE, 4, 12, 3, 0, tr_tlv), /* TRL, TRR */ + SOC_DOUBLE_TLV("Tone Control - Bass", UDA1380_MODE, 0, 8, 15, 0, bb_tlv), /* BBL, BBR */ +/**/ SOC_SINGLE("Master Playback Switch", UDA1380_DEEMP, 14, 1, 1), /* MTM */ + SOC_SINGLE("ADC Playback Switch", UDA1380_DEEMP, 11, 1, 1), /* MT2 from decimation filter */ + SOC_ENUM("ADC Playback De-emphasis", uda1380_deemp_enum[0]), /* DE2 */ + SOC_SINGLE("PCM Playback Switch", UDA1380_DEEMP, 3, 1, 1), /* MT1, from digital data input */ + SOC_ENUM("PCM Playback De-emphasis", uda1380_deemp_enum[1]), /* DE1 */ + SOC_SINGLE("DAC Polarity inverting Switch", UDA1380_MIXER, 15, 1, 0), /* DA_POL_INV */ + SOC_ENUM("Noise Shaper", uda1380_sel_ns_enum), /* SEL_NS */ + SOC_ENUM("Digital Mixer Signal Control", uda1380_mix_enum), /* MIX_POS, MIX */ + SOC_SINGLE("Silence Switch", UDA1380_MIXER, 7, 1, 0), /* SILENCE, force DAC output to silence */ + SOC_SINGLE("Silence Detector Switch", UDA1380_MIXER, 6, 1, 0), /* SDET_ON */ + SOC_ENUM("Silence Detector Setting", uda1380_sdet_enum), /* SD_VALUE */ + SOC_ENUM("Oversampling Input", uda1380_os_enum), /* OS */ + SOC_DOUBLE_S8_TLV("ADC Capture Volume", UDA1380_DEC, -128, 48, dec_tlv), /* ML_DEC, MR_DEC */ +/**/ SOC_SINGLE("ADC Capture Switch", UDA1380_PGA, 15, 1, 1), /* MT_ADC */ + SOC_DOUBLE_TLV("Line Capture Volume", UDA1380_PGA, 0, 8, 8, 0, pga_tlv), /* PGA_GAINCTRLL, PGA_GAINCTRLR */ + SOC_SINGLE("ADC Polarity inverting Switch", UDA1380_ADC, 12, 1, 0), /* ADCPOL_INV */ + SOC_SINGLE_TLV("Mic Capture Volume", UDA1380_ADC, 8, 15, 0, vga_tlv), /* VGA_CTRL */ + SOC_SINGLE("DC Filter Bypass Switch", UDA1380_ADC, 1, 1, 0), /* SKIP_DCFIL (before decimator) */ + SOC_SINGLE("DC Filter Enable Switch", UDA1380_ADC, 0, 1, 0), /* EN_DCFIL (at output of decimator) */ + SOC_SINGLE("AGC Timing", UDA1380_AGC, 8, 7, 0), /* TODO: enum, see table 62 */ + SOC_SINGLE("AGC Target level", UDA1380_AGC, 2, 3, 1), /* AGC_LEVEL */ + /* -5.5, -8, -11.5, -14 dBFS */ + SOC_SINGLE("AGC Switch", UDA1380_AGC, 0, 1, 0), +}; + +/* add non dapm controls */ +static int uda1380_add_controls(struct snd_soc_codec *codec) +{ + int err, i; + + for (i = 0; i < ARRAY_SIZE(uda1380_snd_controls); i++) { + err = snd_ctl_add(codec->card, + snd_soc_cnew(&uda1380_snd_controls[i], codec, NULL)); + if (err < 0) + return err; + } + + return 0; +} + +/* Input mux */ +static const struct snd_kcontrol_new uda1380_input_mux_control = + SOC_DAPM_ENUM("Route", uda1380_input_sel_enum); + +/* Output mux */ +static const struct snd_kcontrol_new uda1380_output_mux_control = + SOC_DAPM_ENUM("Route", uda1380_output_sel_enum); + +/* Capture mux */ +static const struct snd_kcontrol_new uda1380_capture_mux_control = + SOC_DAPM_ENUM("Route", uda1380_capture_sel_enum); + + +static const struct snd_soc_dapm_widget uda1380_dapm_widgets[] = { + SND_SOC_DAPM_MUX("Input Mux", SND_SOC_NOPM, 0, 0, + &uda1380_input_mux_control), + SND_SOC_DAPM_MUX("Output Mux", SND_SOC_NOPM, 0, 0, + &uda1380_output_mux_control), + SND_SOC_DAPM_MUX("Capture Mux", SND_SOC_NOPM, 0, 0, + &uda1380_capture_mux_control), + SND_SOC_DAPM_PGA("Left PGA", UDA1380_PM, 3, 0, NULL, 0), + SND_SOC_DAPM_PGA("Right PGA", UDA1380_PM, 1, 0, NULL, 0), + SND_SOC_DAPM_PGA("Mic LNA", UDA1380_PM, 4, 0, NULL, 0), + SND_SOC_DAPM_ADC("Left ADC", "Left Capture", UDA1380_PM, 2, 0), + SND_SOC_DAPM_ADC("Right ADC", "Right Capture", UDA1380_PM, 0, 0), + SND_SOC_DAPM_INPUT("VINM"), + SND_SOC_DAPM_INPUT("VINL"), + SND_SOC_DAPM_INPUT("VINR"), + SND_SOC_DAPM_MIXER("Analog Mixer", UDA1380_PM, 6, 0, NULL, 0), + SND_SOC_DAPM_OUTPUT("VOUTLHP"), + SND_SOC_DAPM_OUTPUT("VOUTRHP"), + SND_SOC_DAPM_OUTPUT("VOUTL"), + SND_SOC_DAPM_OUTPUT("VOUTR"), + SND_SOC_DAPM_DAC("DAC", "Playback", UDA1380_PM, 10, 0), + SND_SOC_DAPM_PGA("HeadPhone Driver", UDA1380_PM, 13, 0, NULL, 0), +}; + +static const struct snd_soc_dapm_route audio_map[] = { + + /* output mux */ + {"HeadPhone Driver", NULL, "Output Mux"}, + {"VOUTR", NULL, "Output Mux"}, + {"VOUTL", NULL, "Output Mux"}, + + {"Analog Mixer", NULL, "VINR"}, + {"Analog Mixer", NULL, "VINL"}, + {"Analog Mixer", NULL, "DAC"}, + + {"Output Mux", "DAC", "DAC"}, + {"Output Mux", "Analog Mixer", "Analog Mixer"}, + + /* {"DAC", "Digital Mixer", "I2S" } */ + + /* headphone driver */ + {"VOUTLHP", NULL, "HeadPhone Driver"}, + {"VOUTRHP", NULL, "HeadPhone Driver"}, + + /* input mux */ + {"Left ADC", NULL, "Input Mux"}, + {"Input Mux", "Mic", "Mic LNA"}, + {"Input Mux", "Mic + Line R", "Mic LNA"}, + {"Input Mux", "Line L", "Left PGA"}, + {"Input Mux", "Line", "Left PGA"}, + + /* right input */ + {"Right ADC", "Mic + Line R", "Right PGA"}, + {"Right ADC", "Line", "Right PGA"}, + + /* inputs */ + {"Mic LNA", NULL, "VINM"}, + {"Left PGA", NULL, "VINL"}, + {"Right PGA", NULL, "VINR"}, +}; + +static int uda1380_add_widgets(struct snd_soc_codec *codec) +{ + snd_soc_dapm_new_controls(codec, uda1380_dapm_widgets, + ARRAY_SIZE(uda1380_dapm_widgets)); + + snd_soc_dapm_add_routes(codec, audio_map, ARRAY_SIZE(audio_map)); + + snd_soc_dapm_new_widgets(codec); + return 0; +} + +static int uda1380_set_dai_fmt(struct snd_soc_dai *codec_dai, + unsigned int fmt) +{ + struct snd_soc_codec *codec = codec_dai->codec; + int iface; + + /* set up DAI based upon fmt */ + iface = uda1380_read_reg_cache(codec, UDA1380_IFACE); + iface &= ~(R01_SFORI_MASK | R01_SIM | R01_SFORO_MASK); + + /* FIXME: how to select I2S for DATAO and MSB for DATAI correctly? */ + switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { + case SND_SOC_DAIFMT_I2S: + iface |= R01_SFORI_I2S | R01_SFORO_I2S; + break; + case SND_SOC_DAIFMT_LSB: + iface |= R01_SFORI_LSB16 | R01_SFORO_I2S; + break; + case SND_SOC_DAIFMT_MSB: + iface |= R01_SFORI_MSB | R01_SFORO_I2S; + } + + if ((fmt & SND_SOC_DAIFMT_MASTER_MASK) == SND_SOC_DAIFMT_CBM_CFM) + iface |= R01_SIM; + + uda1380_write(codec, UDA1380_IFACE, iface); + + return 0; +} + +/* + * Flush reg cache + * We can only write the interpolator and decimator registers + * when the DAI is being clocked by the CPU DAI. It's up to the + * machine and cpu DAI driver to do this before we are called. + */ +static int uda1380_pcm_prepare(struct snd_pcm_substream *substream) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_device *socdev = rtd->socdev; + struct snd_soc_codec *codec = socdev->codec; + int reg, reg_start, reg_end, clk; + + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { + reg_start = UDA1380_MVOL; + reg_end = UDA1380_MIXER; + } else { + reg_start = UDA1380_DEC; + reg_end = UDA1380_AGC; + } + + /* FIXME disable DAC_CLK */ + clk = uda1380_read_reg_cache(codec, UDA1380_CLK); + uda1380_write(codec, UDA1380_CLK, clk & ~R00_DAC_CLK); + + for (reg = reg_start; reg <= reg_end; reg++) { + pr_debug("uda1380: flush reg %x val %x:", reg, + uda1380_read_reg_cache(codec, reg)); + uda1380_write(codec, reg, uda1380_read_reg_cache(codec, reg)); + } + + /* FIXME enable DAC_CLK */ + uda1380_write(codec, UDA1380_CLK, clk | R00_DAC_CLK); + + return 0; +} + +static int uda1380_pcm_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_device *socdev = rtd->socdev; + struct snd_soc_codec *codec = socdev->codec; + u16 clk = uda1380_read_reg_cache(codec, UDA1380_CLK); + + /* set WSPLL power and divider if running from this clock */ + if (clk & R00_DAC_CLK) { + int rate = params_rate(params); + u16 pm = uda1380_read_reg_cache(codec, UDA1380_PM); + clk &= ~0x3; /* clear SEL_LOOP_DIV */ + switch (rate) { + case 6250 ... 12500: + clk |= 0x0; + break; + case 12501 ... 25000: + clk |= 0x1; + break; + case 25001 ... 50000: + clk |= 0x2; + break; + case 50001 ... 100000: + clk |= 0x3; + break; + } + uda1380_write(codec, UDA1380_PM, R02_PON_PLL | pm); + } + + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + clk |= R00_EN_DAC | R00_EN_INT; + else + clk |= R00_EN_ADC | R00_EN_DEC; + + uda1380_write(codec, UDA1380_CLK, clk); + return 0; +} + +static void uda1380_pcm_shutdown(struct snd_pcm_substream *substream) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_device *socdev = rtd->socdev; + struct snd_soc_codec *codec = socdev->codec; + u16 clk = uda1380_read_reg_cache(codec, UDA1380_CLK); + + /* shut down WSPLL power if running from this clock */ + if (clk & R00_DAC_CLK) { + u16 pm = uda1380_read_reg_cache(codec, UDA1380_PM); + uda1380_write(codec, UDA1380_PM, ~R02_PON_PLL & pm); + } + + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + clk &= ~(R00_EN_DAC | R00_EN_INT); + else + clk &= ~(R00_EN_ADC | R00_EN_DEC); + + uda1380_write(codec, UDA1380_CLK, clk); +} + +static int uda1380_mute(struct snd_soc_dai *codec_dai, int mute) +{ + struct snd_soc_codec *codec = codec_dai->codec; + u16 mute_reg = uda1380_read_reg_cache(codec, UDA1380_DEEMP) & ~R13_MTM; + + /* FIXME: mute(codec,0) is called when the magician clock is already + * set to WSPLL, but for some unknown reason writing to interpolator + * registers works only when clocked by SYSCLK */ + u16 clk = uda1380_read_reg_cache(codec, UDA1380_CLK); + uda1380_write(codec, UDA1380_CLK, ~R00_DAC_CLK & clk); + if (mute) + uda1380_write(codec, UDA1380_DEEMP, mute_reg | R13_MTM); + else + uda1380_write(codec, UDA1380_DEEMP, mute_reg); + uda1380_write(codec, UDA1380_CLK, clk); + return 0; +} + +static int uda1380_set_bias_level(struct snd_soc_codec *codec, + enum snd_soc_bias_level level) +{ + int pm = uda1380_read_reg_cache(codec, UDA1380_PM); + + switch (level) { + case SND_SOC_BIAS_ON: + case SND_SOC_BIAS_PREPARE: + uda1380_write(codec, UDA1380_PM, R02_PON_BIAS | pm); + break; + case SND_SOC_BIAS_STANDBY: + uda1380_write(codec, UDA1380_PM, R02_PON_BIAS); + break; + case SND_SOC_BIAS_OFF: + uda1380_write(codec, UDA1380_PM, 0x0); + break; + } + codec->bias_level = level; + return 0; +} + +#define UDA1380_RATES (SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_11025 |\ + SNDRV_PCM_RATE_16000 | SNDRV_PCM_RATE_22050 |\ + SNDRV_PCM_RATE_44100 | SNDRV_PCM_RATE_48000) + +struct snd_soc_dai uda1380_dai[] = { +{ + .name = "UDA1380", + .playback = { + .stream_name = "Playback", + .channels_min = 1, + .channels_max = 2, + .rates = UDA1380_RATES, + .formats = SNDRV_PCM_FMTBIT_S16_LE,}, + .capture = { + .stream_name = "Capture", + .channels_min = 1, + .channels_max = 2, + .rates = UDA1380_RATES, + .formats = SNDRV_PCM_FMTBIT_S16_LE,}, + .ops = { + .hw_params = uda1380_pcm_hw_params, + .shutdown = uda1380_pcm_shutdown, + .prepare = uda1380_pcm_prepare, + }, + .dai_ops = { + .digital_mute = uda1380_mute, + .set_fmt = uda1380_set_dai_fmt, + }, +}, +{ /* playback only - dual interface */ + .name = "UDA1380", + .playback = { + .stream_name = "Playback", + .channels_min = 1, + .channels_max = 2, + .rates = UDA1380_RATES, + .formats = SNDRV_PCM_FMTBIT_S16_LE, + }, + .ops = { + .hw_params = uda1380_pcm_hw_params, + .shutdown = uda1380_pcm_shutdown, + .prepare = uda1380_pcm_prepare, + }, + .dai_ops = { + .digital_mute = uda1380_mute, + .set_fmt = uda1380_set_dai_fmt, + }, +}, +{ /* capture only - dual interface*/ + .name = "UDA1380", + .capture = { + .stream_name = "Capture", + .channels_min = 1, + .channels_max = 2, + .rates = UDA1380_RATES, + .formats = SNDRV_PCM_FMTBIT_S16_LE, + }, + .ops = { + .hw_params = uda1380_pcm_hw_params, + .shutdown = uda1380_pcm_shutdown, + .prepare = uda1380_pcm_prepare, + }, + .dai_ops = { + .set_fmt = uda1380_set_dai_fmt, + }, +}, +}; +EXPORT_SYMBOL_GPL(uda1380_dai); + +static int uda1380_suspend(struct platform_device *pdev, pm_message_t state) +{ + struct snd_soc_device *socdev = platform_get_drvdata(pdev); + struct snd_soc_codec *codec = socdev->codec; + + uda1380_set_bias_level(codec, SND_SOC_BIAS_OFF); + return 0; +} + +static int uda1380_resume(struct platform_device *pdev) +{ + struct snd_soc_device *socdev = platform_get_drvdata(pdev); + struct snd_soc_codec *codec = socdev->codec; + int i; + u8 data[2]; + u16 *cache = codec->reg_cache; + + /* Sync reg_cache with the hardware */ + for (i = 0; i < ARRAY_SIZE(uda1380_reg); i++) { + data[0] = (i << 1) | ((cache[i] >> 8) & 0x0001); + data[1] = cache[i] & 0x00ff; + codec->hw_write(codec->control_data, data, 2); + } + uda1380_set_bias_level(codec, SND_SOC_BIAS_STANDBY); + uda1380_set_bias_level(codec, codec->suspend_bias_level); + return 0; +} + +/* + * initialise the UDA1380 driver + * register mixer and dsp interfaces with the kernel + */ +static int uda1380_init(struct snd_soc_device *socdev, int dac_clk) +{ + struct snd_soc_codec *codec = socdev->codec; + int ret = 0; + + codec->name = "UDA1380"; + codec->owner = THIS_MODULE; + codec->read = uda1380_read_reg_cache; + codec->write = uda1380_write; + codec->set_bias_level = uda1380_set_bias_level; + codec->dai = uda1380_dai; + codec->num_dai = ARRAY_SIZE(uda1380_dai); + codec->reg_cache = kmemdup(uda1380_reg, sizeof(uda1380_reg), + GFP_KERNEL); + if (codec->reg_cache == NULL) + return -ENOMEM; + codec->reg_cache_size = ARRAY_SIZE(uda1380_reg); + codec->reg_cache_step = 1; + uda1380_reset(codec); + + /* register pcms */ + ret = snd_soc_new_pcms(socdev, SNDRV_DEFAULT_IDX1, SNDRV_DEFAULT_STR1); + if (ret < 0) { + pr_err("uda1380: failed to create pcms\n"); + goto pcm_err; + } + + /* power on device */ + uda1380_set_bias_level(codec, SND_SOC_BIAS_STANDBY); + /* set clock input */ + switch (dac_clk) { + case UDA1380_DAC_CLK_SYSCLK: + uda1380_write(codec, UDA1380_CLK, 0); + break; + case UDA1380_DAC_CLK_WSPLL: + uda1380_write(codec, UDA1380_CLK, R00_DAC_CLK); + break; + } + + /* uda1380 init */ + uda1380_add_controls(codec); + uda1380_add_widgets(codec); + ret = snd_soc_register_card(socdev); + if (ret < 0) { + pr_err("uda1380: failed to register card\n"); + goto card_err; + } + + return ret; + +card_err: + snd_soc_free_pcms(socdev); + snd_soc_dapm_free(socdev); +pcm_err: + kfree(codec->reg_cache); + return ret; +} + +static struct snd_soc_device *uda1380_socdev; + +#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE) + +#define I2C_DRIVERID_UDA1380 0xfefe /* liam - need a proper id */ + +static unsigned short normal_i2c[] = { 0, I2C_CLIENT_END }; + +/* Magic definition of all other variables and things */ +I2C_CLIENT_INSMOD; + +static struct i2c_driver uda1380_i2c_driver; +static struct i2c_client client_template; + +/* If the i2c layer weren't so broken, we could pass this kind of data + around */ + +static int uda1380_codec_probe(struct i2c_adapter *adap, int addr, int kind) +{ + struct snd_soc_device *socdev = uda1380_socdev; + struct uda1380_setup_data *setup = socdev->codec_data; + struct snd_soc_codec *codec = socdev->codec; + struct i2c_client *i2c; + int ret; + + if (addr != setup->i2c_address) + return -ENODEV; + + client_template.adapter = adap; + client_template.addr = addr; + + i2c = kmemdup(&client_template, sizeof(client_template), GFP_KERNEL); + if (i2c == NULL) { + kfree(codec); + return -ENOMEM; + } + i2c_set_clientdata(i2c, codec); + codec->control_data = i2c; + + ret = i2c_attach_client(i2c); + if (ret < 0) { + pr_err("uda1380: failed to attach codec at addr %x\n", addr); + goto err; + } + + ret = uda1380_init(socdev, setup->dac_clk); + if (ret < 0) { + pr_err("uda1380: failed to initialise UDA1380\n"); + goto err; + } + return ret; + +err: + kfree(codec); + kfree(i2c); + return ret; +} + +static int uda1380_i2c_detach(struct i2c_client *client) +{ + struct snd_soc_codec *codec = i2c_get_clientdata(client); + i2c_detach_client(client); + kfree(codec->reg_cache); + kfree(client); + return 0; +} + +static int uda1380_i2c_attach(struct i2c_adapter *adap) +{ + return i2c_probe(adap, &addr_data, uda1380_codec_probe); +} + +static struct i2c_driver uda1380_i2c_driver = { + .driver = { + .name = "UDA1380 I2C Codec", + .owner = THIS_MODULE, + }, + .id = I2C_DRIVERID_UDA1380, + .attach_adapter = uda1380_i2c_attach, + .detach_client = uda1380_i2c_detach, + .command = NULL, +}; + +static struct i2c_client client_template = { + .name = "UDA1380", + .driver = &uda1380_i2c_driver, +}; +#endif + +static int uda1380_probe(struct platform_device *pdev) +{ + struct snd_soc_device *socdev = platform_get_drvdata(pdev); + struct uda1380_setup_data *setup; + struct snd_soc_codec *codec; + int ret = 0; + + pr_info("UDA1380 Audio Codec %s", UDA1380_VERSION); + + setup = socdev->codec_data; + codec = kzalloc(sizeof(struct snd_soc_codec), GFP_KERNEL); + if (codec == NULL) + return -ENOMEM; + + socdev->codec = codec; + mutex_init(&codec->mutex); + INIT_LIST_HEAD(&codec->dapm_widgets); + INIT_LIST_HEAD(&codec->dapm_paths); + + uda1380_socdev = socdev; +#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE) + if (setup->i2c_address) { + normal_i2c[0] = setup->i2c_address; + codec->hw_write = (hw_write_t)i2c_master_send; + ret = i2c_add_driver(&uda1380_i2c_driver); + if (ret != 0) + printk(KERN_ERR "can't add i2c driver"); + } +#else + /* Add other interfaces here */ +#endif + return ret; +} + +/* power down chip */ +static int uda1380_remove(struct platform_device *pdev) +{ + struct snd_soc_device *socdev = platform_get_drvdata(pdev); + struct snd_soc_codec *codec = socdev->codec; + + if (codec->control_data) + uda1380_set_bias_level(codec, SND_SOC_BIAS_OFF); + + snd_soc_free_pcms(socdev); + snd_soc_dapm_free(socdev); +#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE) + i2c_del_driver(&uda1380_i2c_driver); +#endif + kfree(codec); + + return 0; +} + +struct snd_soc_codec_device soc_codec_dev_uda1380 = { + .probe = uda1380_probe, + .remove = uda1380_remove, + .suspend = uda1380_suspend, + .resume = uda1380_resume, +}; +EXPORT_SYMBOL_GPL(soc_codec_dev_uda1380); + +MODULE_AUTHOR("Giorgio Padrin"); +MODULE_DESCRIPTION("Audio support for codec Philips UDA1380"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/codecs/uda1380.h b/sound/soc/codecs/uda1380.h new file mode 100644 index 00000000000..50c603e2c9f --- /dev/null +++ b/sound/soc/codecs/uda1380.h @@ -0,0 +1,89 @@ +/* + * Audio support for Philips UDA1380 + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. + * + * Copyright (c) 2005 Giorgio Padrin <giorgio@mandarinlogiq.org> + */ + +#ifndef _UDA1380_H +#define _UDA1380_H + +#define UDA1380_CLK 0x00 +#define UDA1380_IFACE 0x01 +#define UDA1380_PM 0x02 +#define UDA1380_AMIX 0x03 +#define UDA1380_HP 0x04 +#define UDA1380_MVOL 0x10 +#define UDA1380_MIXVOL 0x11 +#define UDA1380_MODE 0x12 +#define UDA1380_DEEMP 0x13 +#define UDA1380_MIXER 0x14 +#define UDA1380_INTSTAT 0x18 +#define UDA1380_DEC 0x20 +#define UDA1380_PGA 0x21 +#define UDA1380_ADC 0x22 +#define UDA1380_AGC 0x23 +#define UDA1380_DECSTAT 0x28 +#define UDA1380_RESET 0x7f + +#define UDA1380_CACHEREGNUM 0x24 + +/* Register flags */ +#define R00_EN_ADC 0x0800 +#define R00_EN_DEC 0x0400 +#define R00_EN_DAC 0x0200 +#define R00_EN_INT 0x0100 +#define R00_DAC_CLK 0x0010 +#define R01_SFORI_I2S 0x0000 +#define R01_SFORI_LSB16 0x0100 +#define R01_SFORI_LSB18 0x0200 +#define R01_SFORI_LSB20 0x0300 +#define R01_SFORI_MSB 0x0500 +#define R01_SFORI_MASK 0x0700 +#define R01_SFORO_I2S 0x0000 +#define R01_SFORO_LSB16 0x0001 +#define R01_SFORO_LSB18 0x0002 +#define R01_SFORO_LSB20 0x0003 +#define R01_SFORO_LSB24 0x0004 +#define R01_SFORO_MSB 0x0005 +#define R01_SFORO_MASK 0x0007 +#define R01_SEL_SOURCE 0x0040 +#define R01_SIM 0x0010 +#define R02_PON_PLL 0x8000 +#define R02_PON_HP 0x2000 +#define R02_PON_DAC 0x0400 +#define R02_PON_BIAS 0x0100 +#define R02_EN_AVC 0x0080 +#define R02_PON_AVC 0x0040 +#define R02_PON_LNA 0x0010 +#define R02_PON_PGAL 0x0008 +#define R02_PON_ADCL 0x0004 +#define R02_PON_PGAR 0x0002 +#define R02_PON_ADCR 0x0001 +#define R13_MTM 0x4000 +#define R14_SILENCE 0x0080 +#define R14_SDET_ON 0x0040 +#define R21_MT_ADC 0x8000 +#define R22_SEL_LNA 0x0008 +#define R22_SEL_MIC 0x0004 +#define R22_SKIP_DCFIL 0x0002 +#define R23_AGC_EN 0x0001 + +struct uda1380_setup_data { + unsigned short i2c_address; + int dac_clk; +#define UDA1380_DAC_CLK_SYSCLK 0 +#define UDA1380_DAC_CLK_WSPLL 1 +}; + +#define UDA1380_DAI_DUPLEX 0 /* playback and capture on single DAI */ +#define UDA1380_DAI_PLAYBACK 1 /* playback DAI */ +#define UDA1380_DAI_CAPTURE 2 /* capture DAI */ + +extern struct snd_soc_dai uda1380_dai[3]; +extern struct snd_soc_codec_device soc_codec_dev_uda1380; + +#endif /* _UDA1380_H */ diff --git a/sound/soc/codecs/wm8510.c b/sound/soc/codecs/wm8510.c new file mode 100644 index 00000000000..67325fd9544 --- /dev/null +++ b/sound/soc/codecs/wm8510.c @@ -0,0 +1,817 @@ +/* + * wm8510.c -- WM8510 ALSA Soc Audio driver + * + * Copyright 2006 Wolfson Microelectronics PLC. + * + * Author: Liam Girdwood <liam.girdwood@wolfsonmicro.com> + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. + */ + +#include <linux/module.h> +#include <linux/moduleparam.h> +#include <linux/kernel.h> +#include <linux/init.h> +#include <linux/delay.h> +#include <linux/pm.h> +#include <linux/i2c.h> +#include <linux/platform_device.h> +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> +#include <sound/soc-dapm.h> +#include <sound/initval.h> + +#include "wm8510.h" + +#define AUDIO_NAME "wm8510" +#define WM8510_VERSION "0.6" + +struct snd_soc_codec_device soc_codec_dev_wm8510; + +/* + * wm8510 register cache + * We can't read the WM8510 register space when we are + * using 2 wire for device control, so we cache them instead. + */ +static const u16 wm8510_reg[WM8510_CACHEREGNUM] = { + 0x0000, 0x0000, 0x0000, 0x0000, + 0x0050, 0x0000, 0x0140, 0x0000, + 0x0000, 0x0000, 0x0000, 0x00ff, + 0x0000, 0x0000, 0x0100, 0x00ff, + 0x0000, 0x0000, 0x012c, 0x002c, + 0x002c, 0x002c, 0x002c, 0x0000, + 0x0032, 0x0000, 0x0000, 0x0000, + 0x0000, 0x0000, 0x0000, 0x0000, + 0x0038, 0x000b, 0x0032, 0x0000, + 0x0008, 0x000c, 0x0093, 0x00e9, + 0x0000, 0x0000, 0x0000, 0x0000, + 0x0003, 0x0010, 0x0000, 0x0000, + 0x0000, 0x0002, 0x0001, 0x0000, + 0x0000, 0x0000, 0x0039, 0x0000, + 0x0001, +}; + +/* + * read wm8510 register cache + */ +static inline unsigned int wm8510_read_reg_cache(struct snd_soc_codec *codec, + unsigned int reg) +{ + u16 *cache = codec->reg_cache; + if (reg == WM8510_RESET) + return 0; + if (reg >= WM8510_CACHEREGNUM) + return -1; + return cache[reg]; +} + +/* + * write wm8510 register cache + */ +static inline void wm8510_write_reg_cache(struct snd_soc_codec *codec, + u16 reg, unsigned int value) +{ + u16 *cache = codec->reg_cache; + if (reg >= WM8510_CACHEREGNUM) + return; + cache[reg] = value; +} + +/* + * write to the WM8510 register space + */ +static int wm8510_write(struct snd_soc_codec *codec, unsigned int reg, + unsigned int value) +{ + u8 data[2]; + + /* data is + * D15..D9 WM8510 register offset + * D8...D0 register data + */ + data[0] = (reg << 1) | ((value >> 8) & 0x0001); + data[1] = value & 0x00ff; + + wm8510_write_reg_cache(codec, reg, value); + if (codec->hw_write(codec->control_data, data, 2) == 2) + return 0; + else + return -EIO; +} + +#define wm8510_reset(c) wm8510_write(c, WM8510_RESET, 0) + +static const char *wm8510_companding[] = { "Off", "NC", "u-law", "A-law" }; +static const char *wm8510_deemp[] = { "None", "32kHz", "44.1kHz", "48kHz" }; +static const char *wm8510_alc[] = { "ALC", "Limiter" }; + +static const struct soc_enum wm8510_enum[] = { + SOC_ENUM_SINGLE(WM8510_COMP, 1, 4, wm8510_companding), /* adc */ + SOC_ENUM_SINGLE(WM8510_COMP, 3, 4, wm8510_companding), /* dac */ + SOC_ENUM_SINGLE(WM8510_DAC, 4, 4, wm8510_deemp), + SOC_ENUM_SINGLE(WM8510_ALC3, 8, 2, wm8510_alc), +}; + +static const struct snd_kcontrol_new wm8510_snd_controls[] = { + +SOC_SINGLE("Digital Loopback Switch", WM8510_COMP, 0, 1, 0), + +SOC_ENUM("DAC Companding", wm8510_enum[1]), +SOC_ENUM("ADC Companding", wm8510_enum[0]), + +SOC_ENUM("Playback De-emphasis", wm8510_enum[2]), +SOC_SINGLE("DAC Inversion Switch", WM8510_DAC, 0, 1, 0), + +SOC_SINGLE("Master Playback Volume", WM8510_DACVOL, 0, 127, 0), + +SOC_SINGLE("High Pass Filter Switch", WM8510_ADC, 8, 1, 0), +SOC_SINGLE("High Pass Cut Off", WM8510_ADC, 4, 7, 0), +SOC_SINGLE("ADC Inversion Switch", WM8510_COMP, 0, 1, 0), + +SOC_SINGLE("Capture Volume", WM8510_ADCVOL, 0, 127, 0), + +SOC_SINGLE("DAC Playback Limiter Switch", WM8510_DACLIM1, 8, 1, 0), +SOC_SINGLE("DAC Playback Limiter Decay", WM8510_DACLIM1, 4, 15, 0), +SOC_SINGLE("DAC Playback Limiter Attack", WM8510_DACLIM1, 0, 15, 0), + +SOC_SINGLE("DAC Playback Limiter Threshold", WM8510_DACLIM2, 4, 7, 0), +SOC_SINGLE("DAC Playback Limiter Boost", WM8510_DACLIM2, 0, 15, 0), + +SOC_SINGLE("ALC Enable Switch", WM8510_ALC1, 8, 1, 0), +SOC_SINGLE("ALC Capture Max Gain", WM8510_ALC1, 3, 7, 0), +SOC_SINGLE("ALC Capture Min Gain", WM8510_ALC1, 0, 7, 0), + +SOC_SINGLE("ALC Capture ZC Switch", WM8510_ALC2, 8, 1, 0), +SOC_SINGLE("ALC Capture Hold", WM8510_ALC2, 4, 7, 0), +SOC_SINGLE("ALC Capture Target", WM8510_ALC2, 0, 15, 0), + +SOC_ENUM("ALC Capture Mode", wm8510_enum[3]), +SOC_SINGLE("ALC Capture Decay", WM8510_ALC3, 4, 15, 0), +SOC_SINGLE("ALC Capture Attack", WM8510_ALC3, 0, 15, 0), + +SOC_SINGLE("ALC Capture Noise Gate Switch", WM8510_NGATE, 3, 1, 0), +SOC_SINGLE("ALC Capture Noise Gate Threshold", WM8510_NGATE, 0, 7, 0), + +SOC_SINGLE("Capture PGA ZC Switch", WM8510_INPPGA, 7, 1, 0), +SOC_SINGLE("Capture PGA Volume", WM8510_INPPGA, 0, 63, 0), + +SOC_SINGLE("Speaker Playback ZC Switch", WM8510_SPKVOL, 7, 1, 0), +SOC_SINGLE("Speaker Playback Switch", WM8510_SPKVOL, 6, 1, 1), +SOC_SINGLE("Speaker Playback Volume", WM8510_SPKVOL, 0, 63, 0), +SOC_SINGLE("Speaker Boost", WM8510_OUTPUT, 2, 1, 0), + +SOC_SINGLE("Capture Boost(+20dB)", WM8510_ADCBOOST, 8, 1, 0), +SOC_SINGLE("Mono Playback Switch", WM8510_MONOMIX, 6, 1, 1), +}; + +/* add non dapm controls */ +static int wm8510_add_controls(struct snd_soc_codec *codec) +{ + int err, i; + + for (i = 0; i < ARRAY_SIZE(wm8510_snd_controls); i++) { + err = snd_ctl_add(codec->card, + snd_soc_cnew(&wm8510_snd_controls[i], codec, + NULL)); + if (err < 0) + return err; + } + + return 0; +} + +/* Speaker Output Mixer */ +static const struct snd_kcontrol_new wm8510_speaker_mixer_controls[] = { +SOC_DAPM_SINGLE("Line Bypass Switch", WM8510_SPKMIX, 1, 1, 0), +SOC_DAPM_SINGLE("Aux Playback Switch", WM8510_SPKMIX, 5, 1, 0), +SOC_DAPM_SINGLE("PCM Playback Switch", WM8510_SPKMIX, 0, 1, 0), +}; + +/* Mono Output Mixer */ +static const struct snd_kcontrol_new wm8510_mono_mixer_controls[] = { +SOC_DAPM_SINGLE("Line Bypass Switch", WM8510_MONOMIX, 1, 1, 0), +SOC_DAPM_SINGLE("Aux Playback Switch", WM8510_MONOMIX, 2, 1, 0), +SOC_DAPM_SINGLE("PCM Playback Switch", WM8510_MONOMIX, 0, 1, 0), +}; + +static const struct snd_kcontrol_new wm8510_boost_controls[] = { +SOC_DAPM_SINGLE("Mic PGA Switch", WM8510_INPPGA, 6, 1, 0), +SOC_DAPM_SINGLE("Aux Volume", WM8510_ADCBOOST, 0, 7, 0), +SOC_DAPM_SINGLE("Mic Volume", WM8510_ADCBOOST, 4, 7, 0), +}; + +static const struct snd_kcontrol_new wm8510_micpga_controls[] = { +SOC_DAPM_SINGLE("MICP Switch", WM8510_INPUT, 0, 1, 0), +SOC_DAPM_SINGLE("MICN Switch", WM8510_INPUT, 1, 1, 0), +SOC_DAPM_SINGLE("AUX Switch", WM8510_INPUT, 2, 1, 0), +}; + +static const struct snd_soc_dapm_widget wm8510_dapm_widgets[] = { +SND_SOC_DAPM_MIXER("Speaker Mixer", WM8510_POWER3, 2, 0, + &wm8510_speaker_mixer_controls[0], + ARRAY_SIZE(wm8510_speaker_mixer_controls)), +SND_SOC_DAPM_MIXER("Mono Mixer", WM8510_POWER3, 3, 0, + &wm8510_mono_mixer_controls[0], + ARRAY_SIZE(wm8510_mono_mixer_controls)), +SND_SOC_DAPM_DAC("DAC", "HiFi Playback", WM8510_POWER3, 0, 0), +SND_SOC_DAPM_ADC("ADC", "HiFi Capture", WM8510_POWER2, 0, 0), +SND_SOC_DAPM_PGA("Aux Input", WM8510_POWER1, 6, 0, NULL, 0), +SND_SOC_DAPM_PGA("SpkN Out", WM8510_POWER3, 5, 0, NULL, 0), +SND_SOC_DAPM_PGA("SpkP Out", WM8510_POWER3, 6, 0, NULL, 0), +SND_SOC_DAPM_PGA("Mono Out", WM8510_POWER3, 7, 0, NULL, 0), + +SND_SOC_DAPM_PGA("Mic PGA", WM8510_POWER2, 2, 0, + &wm8510_micpga_controls[0], + ARRAY_SIZE(wm8510_micpga_controls)), +SND_SOC_DAPM_MIXER("Boost Mixer", WM8510_POWER2, 4, 0, + &wm8510_boost_controls[0], + ARRAY_SIZE(wm8510_boost_controls)), + +SND_SOC_DAPM_MICBIAS("Mic Bias", WM8510_POWER1, 4, 0), + +SND_SOC_DAPM_INPUT("MICN"), +SND_SOC_DAPM_INPUT("MICP"), +SND_SOC_DAPM_INPUT("AUX"), +SND_SOC_DAPM_OUTPUT("MONOOUT"), +SND_SOC_DAPM_OUTPUT("SPKOUTP"), +SND_SOC_DAPM_OUTPUT("SPKOUTN"), +}; + +static const struct snd_soc_dapm_route audio_map[] = { + /* Mono output mixer */ + {"Mono Mixer", "PCM Playback Switch", "DAC"}, + {"Mono Mixer", "Aux Playback Switch", "Aux Input"}, + {"Mono Mixer", "Line Bypass Switch", "Boost Mixer"}, + + /* Speaker output mixer */ + {"Speaker Mixer", "PCM Playback Switch", "DAC"}, + {"Speaker Mixer", "Aux Playback Switch", "Aux Input"}, + {"Speaker Mixer", "Line Bypass Switch", "Boost Mixer"}, + + /* Outputs */ + {"Mono Out", NULL, "Mono Mixer"}, + {"MONOOUT", NULL, "Mono Out"}, + {"SpkN Out", NULL, "Speaker Mixer"}, + {"SpkP Out", NULL, "Speaker Mixer"}, + {"SPKOUTN", NULL, "SpkN Out"}, + {"SPKOUTP", NULL, "SpkP Out"}, + + /* Microphone PGA */ + {"Mic PGA", "MICN Switch", "MICN"}, + {"Mic PGA", "MICP Switch", "MICP"}, + { "Mic PGA", "AUX Switch", "Aux Input" }, + + /* Boost Mixer */ + {"Boost Mixer", "Mic PGA Switch", "Mic PGA"}, + {"Boost Mixer", "Mic Volume", "MICP"}, + {"Boost Mixer", "Aux Volume", "Aux Input"}, + + {"ADC", NULL, "Boost Mixer"}, +}; + +static int wm8510_add_widgets(struct snd_soc_codec *codec) +{ + snd_soc_dapm_new_controls(codec, wm8510_dapm_widgets, + ARRAY_SIZE(wm8510_dapm_widgets)); + + snd_soc_dapm_add_routes(codec, audio_map, ARRAY_SIZE(audio_map)); + + snd_soc_dapm_new_widgets(codec); + return 0; +} + +struct pll_ { + unsigned int pre_div:4; /* prescale - 1 */ + unsigned int n:4; + unsigned int k; +}; + +static struct pll_ pll_div; + +/* The size in bits of the pll divide multiplied by 10 + * to allow rounding later */ +#define FIXED_PLL_SIZE ((1 << 24) * 10) + +static void pll_factors(unsigned int target, unsigned int source) +{ + unsigned long long Kpart; + unsigned int K, Ndiv, Nmod; + + Ndiv = target / source; + if (Ndiv < 6) { + source >>= 1; + pll_div.pre_div = 1; + Ndiv = target / source; + } else + pll_div.pre_div = 0; + + if ((Ndiv < 6) || (Ndiv > 12)) + printk(KERN_WARNING + "WM8510 N value %d outwith recommended range!d\n", + Ndiv); + + pll_div.n = Ndiv; + Nmod = target % source; + Kpart = FIXED_PLL_SIZE * (long long)Nmod; + + do_div(Kpart, source); + + K = Kpart & 0xFFFFFFFF; + + /* Check if we need to round */ + if ((K % 10) >= 5) + K += 5; + + /* Move down to proper range now rounding is done */ + K /= 10; + + pll_div.k = K; +} + +static int wm8510_set_dai_pll(struct snd_soc_dai *codec_dai, + int pll_id, unsigned int freq_in, unsigned int freq_out) +{ + struct snd_soc_codec *codec = codec_dai->codec; + u16 reg; + + if (freq_in == 0 || freq_out == 0) { + /* Clock CODEC directly from MCLK */ + reg = wm8510_read_reg_cache(codec, WM8510_CLOCK); + wm8510_write(codec, WM8510_CLOCK, reg & 0x0ff); + + /* Turn off PLL */ + reg = wm8510_read_reg_cache(codec, WM8510_POWER1); + wm8510_write(codec, WM8510_POWER1, reg & 0x1df); + return 0; + } + + pll_factors(freq_out*8, freq_in); + + wm8510_write(codec, WM8510_PLLN, (pll_div.pre_div << 4) | pll_div.n); + wm8510_write(codec, WM8510_PLLK1, pll_div.k >> 18); + wm8510_write(codec, WM8510_PLLK2, (pll_div.k >> 9) & 0x1ff); + wm8510_write(codec, WM8510_PLLK3, pll_div.k & 0x1ff); + reg = wm8510_read_reg_cache(codec, WM8510_POWER1); + wm8510_write(codec, WM8510_POWER1, reg | 0x020); + + /* Run CODEC from PLL instead of MCLK */ + reg = wm8510_read_reg_cache(codec, WM8510_CLOCK); + wm8510_write(codec, WM8510_CLOCK, reg | 0x100); + + return 0; +} + +/* + * Configure WM8510 clock dividers. + */ +static int wm8510_set_dai_clkdiv(struct snd_soc_dai *codec_dai, + int div_id, int div) +{ + struct snd_soc_codec *codec = codec_dai->codec; + u16 reg; + + switch (div_id) { + case WM8510_OPCLKDIV: + reg = wm8510_read_reg_cache(codec, WM8510_GPIO) & 0x1cf; + wm8510_write(codec, WM8510_GPIO, reg | div); + break; + case WM8510_MCLKDIV: + reg = wm8510_read_reg_cache(codec, WM8510_CLOCK) & 0x1f; + wm8510_write(codec, WM8510_CLOCK, reg | div); + break; + case WM8510_ADCCLK: + reg = wm8510_read_reg_cache(codec, WM8510_ADC) & 0x1f7; + wm8510_write(codec, WM8510_ADC, reg | div); + break; + case WM8510_DACCLK: + reg = wm8510_read_reg_cache(codec, WM8510_DAC) & 0x1f7; + wm8510_write(codec, WM8510_DAC, reg | div); + break; + case WM8510_BCLKDIV: + reg = wm8510_read_reg_cache(codec, WM8510_CLOCK) & 0x1e3; + wm8510_write(codec, WM8510_CLOCK, reg | div); + break; + default: + return -EINVAL; + } + + return 0; +} + +static int wm8510_set_dai_fmt(struct snd_soc_dai *codec_dai, + unsigned int fmt) +{ + struct snd_soc_codec *codec = codec_dai->codec; + u16 iface = 0; + u16 clk = wm8510_read_reg_cache(codec, WM8510_CLOCK) & 0x1fe; + + /* set master/slave audio interface */ + switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { + case SND_SOC_DAIFMT_CBM_CFM: + clk |= 0x0001; + break; + case SND_SOC_DAIFMT_CBS_CFS: + break; + default: + return -EINVAL; + } + + /* interface format */ + switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { + case SND_SOC_DAIFMT_I2S: + iface |= 0x0010; + break; + case SND_SOC_DAIFMT_RIGHT_J: + break; + case SND_SOC_DAIFMT_LEFT_J: + iface |= 0x0008; + break; + case SND_SOC_DAIFMT_DSP_A: + iface |= 0x00018; + break; + default: + return -EINVAL; + } + + /* clock inversion */ + switch (fmt & SND_SOC_DAIFMT_INV_MASK) { + case SND_SOC_DAIFMT_NB_NF: + break; + case SND_SOC_DAIFMT_IB_IF: + iface |= 0x0180; + break; + case SND_SOC_DAIFMT_IB_NF: + iface |= 0x0100; + break; + case SND_SOC_DAIFMT_NB_IF: + iface |= 0x0080; + break; + default: + return -EINVAL; + } + + wm8510_write(codec, WM8510_IFACE, iface); + wm8510_write(codec, WM8510_CLOCK, clk); + return 0; +} + +static int wm8510_pcm_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_device *socdev = rtd->socdev; + struct snd_soc_codec *codec = socdev->codec; + u16 iface = wm8510_read_reg_cache(codec, WM8510_IFACE) & 0x19f; + u16 adn = wm8510_read_reg_cache(codec, WM8510_ADD) & 0x1f1; + + /* bit size */ + switch (params_format(params)) { + case SNDRV_PCM_FORMAT_S16_LE: + break; + case SNDRV_PCM_FORMAT_S20_3LE: + iface |= 0x0020; + break; + case SNDRV_PCM_FORMAT_S24_LE: + iface |= 0x0040; + break; + case SNDRV_PCM_FORMAT_S32_LE: + iface |= 0x0060; + break; + } + + /* filter coefficient */ + switch (params_rate(params)) { + case SNDRV_PCM_RATE_8000: + adn |= 0x5 << 1; + break; + case SNDRV_PCM_RATE_11025: + adn |= 0x4 << 1; + break; + case SNDRV_PCM_RATE_16000: + adn |= 0x3 << 1; + break; + case SNDRV_PCM_RATE_22050: + adn |= 0x2 << 1; + break; + case SNDRV_PCM_RATE_32000: + adn |= 0x1 << 1; + break; + case SNDRV_PCM_RATE_44100: + case SNDRV_PCM_RATE_48000: + break; + } + + wm8510_write(codec, WM8510_IFACE, iface); + wm8510_write(codec, WM8510_ADD, adn); + return 0; +} + +static int wm8510_mute(struct snd_soc_dai *dai, int mute) +{ + struct snd_soc_codec *codec = dai->codec; + u16 mute_reg = wm8510_read_reg_cache(codec, WM8510_DAC) & 0xffbf; + + if (mute) + wm8510_write(codec, WM8510_DAC, mute_reg | 0x40); + else + wm8510_write(codec, WM8510_DAC, mute_reg); + return 0; +} + +/* liam need to make this lower power with dapm */ +static int wm8510_set_bias_level(struct snd_soc_codec *codec, + enum snd_soc_bias_level level) +{ + + switch (level) { + case SND_SOC_BIAS_ON: + wm8510_write(codec, WM8510_POWER1, 0x1ff); + wm8510_write(codec, WM8510_POWER2, 0x1ff); + wm8510_write(codec, WM8510_POWER3, 0x1ff); + break; + case SND_SOC_BIAS_PREPARE: + case SND_SOC_BIAS_STANDBY: + break; + case SND_SOC_BIAS_OFF: + /* everything off, dac mute, inactive */ + wm8510_write(codec, WM8510_POWER1, 0x0); + wm8510_write(codec, WM8510_POWER2, 0x0); + wm8510_write(codec, WM8510_POWER3, 0x0); + break; + } + codec->bias_level = level; + return 0; +} + +#define WM8510_RATES (SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_11025 |\ + SNDRV_PCM_RATE_16000 | SNDRV_PCM_RATE_22050 |\ + SNDRV_PCM_RATE_44100 | SNDRV_PCM_RATE_48000) + +#define WM8510_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE |\ + SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_S32_LE) + +struct snd_soc_dai wm8510_dai = { + .name = "WM8510 HiFi", + .playback = { + .stream_name = "Playback", + .channels_min = 2, + .channels_max = 2, + .rates = WM8510_RATES, + .formats = WM8510_FORMATS,}, + .capture = { + .stream_name = "Capture", + .channels_min = 2, + .channels_max = 2, + .rates = WM8510_RATES, + .formats = WM8510_FORMATS,}, + .ops = { + .hw_params = wm8510_pcm_hw_params, + }, + .dai_ops = { + .digital_mute = wm8510_mute, + .set_fmt = wm8510_set_dai_fmt, + .set_clkdiv = wm8510_set_dai_clkdiv, + .set_pll = wm8510_set_dai_pll, + }, +}; +EXPORT_SYMBOL_GPL(wm8510_dai); + +static int wm8510_suspend(struct platform_device *pdev, pm_message_t state) +{ + struct snd_soc_device *socdev = platform_get_drvdata(pdev); + struct snd_soc_codec *codec = socdev->codec; + + wm8510_set_bias_level(codec, SND_SOC_BIAS_OFF); + return 0; +} + +static int wm8510_resume(struct platform_device *pdev) +{ + struct snd_soc_device *socdev = platform_get_drvdata(pdev); + struct snd_soc_codec *codec = socdev->codec; + int i; + u8 data[2]; + u16 *cache = codec->reg_cache; + + /* Sync reg_cache with the hardware */ + for (i = 0; i < ARRAY_SIZE(wm8510_reg); i++) { + data[0] = (i << 1) | ((cache[i] >> 8) & 0x0001); + data[1] = cache[i] & 0x00ff; + codec->hw_write(codec->control_data, data, 2); + } + wm8510_set_bias_level(codec, SND_SOC_BIAS_STANDBY); + wm8510_set_bias_level(codec, codec->suspend_bias_level); + return 0; +} + +/* + * initialise the WM8510 driver + * register the mixer and dsp interfaces with the kernel + */ +static int wm8510_init(struct snd_soc_device *socdev) +{ + struct snd_soc_codec *codec = socdev->codec; + int ret = 0; + + codec->name = "WM8510"; + codec->owner = THIS_MODULE; + codec->read = wm8510_read_reg_cache; + codec->write = wm8510_write; + codec->set_bias_level = wm8510_set_bias_level; + codec->dai = &wm8510_dai; + codec->num_dai = 1; + codec->reg_cache_size = ARRAY_SIZE(wm8510_reg); + codec->reg_cache = kmemdup(wm8510_reg, sizeof(wm8510_reg), GFP_KERNEL); + + if (codec->reg_cache == NULL) + return -ENOMEM; + + wm8510_reset(codec); + + /* register pcms */ + ret = snd_soc_new_pcms(socdev, SNDRV_DEFAULT_IDX1, SNDRV_DEFAULT_STR1); + if (ret < 0) { + printk(KERN_ERR "wm8510: failed to create pcms\n"); + goto pcm_err; + } + + /* power on device */ + wm8510_set_bias_level(codec, SND_SOC_BIAS_STANDBY); + wm8510_add_controls(codec); + wm8510_add_widgets(codec); + ret = snd_soc_register_card(socdev); + if (ret < 0) { + printk(KERN_ERR "wm8510: failed to register card\n"); + goto card_err; + } + return ret; + +card_err: + snd_soc_free_pcms(socdev); + snd_soc_dapm_free(socdev); +pcm_err: + kfree(codec->reg_cache); + return ret; +} + +static struct snd_soc_device *wm8510_socdev; + +#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE) + +/* + * WM8510 2 wire address is 0x1a + */ +#define I2C_DRIVERID_WM8510 0xfefe /* liam - need a proper id */ + +static unsigned short normal_i2c[] = { 0, I2C_CLIENT_END }; + +/* Magic definition of all other variables and things */ +I2C_CLIENT_INSMOD; + +static struct i2c_driver wm8510_i2c_driver; +static struct i2c_client client_template; + +/* If the i2c layer weren't so broken, we could pass this kind of data + around */ + +static int wm8510_codec_probe(struct i2c_adapter *adap, int addr, int kind) +{ + struct snd_soc_device *socdev = wm8510_socdev; + struct wm8510_setup_data *setup = socdev->codec_data; + struct snd_soc_codec *codec = socdev->codec; + struct i2c_client *i2c; + int ret; + + if (addr != setup->i2c_address) + return -ENODEV; + + client_template.adapter = adap; + client_template.addr = addr; + + i2c = kmemdup(&client_template, sizeof(client_template), GFP_KERNEL); + if (i2c == NULL) { + kfree(codec); + return -ENOMEM; + } + i2c_set_clientdata(i2c, codec); + codec->control_data = i2c; + + ret = i2c_attach_client(i2c); + if (ret < 0) { + pr_err("failed to attach codec at addr %x\n", addr); + goto err; + } + + ret = wm8510_init(socdev); + if (ret < 0) { + pr_err("failed to initialise WM8510\n"); + goto err; + } + return ret; + +err: + kfree(codec); + kfree(i2c); + return ret; +} + +static int wm8510_i2c_detach(struct i2c_client *client) +{ + struct snd_soc_codec *codec = i2c_get_clientdata(client); + i2c_detach_client(client); + kfree(codec->reg_cache); + kfree(client); + return 0; +} + +static int wm8510_i2c_attach(struct i2c_adapter *adap) +{ + return i2c_probe(adap, &addr_data, wm8510_codec_probe); +} + +/* corgi i2c codec control layer */ +static struct i2c_driver wm8510_i2c_driver = { + .driver = { + .name = "WM8510 I2C Codec", + .owner = THIS_MODULE, + }, + .id = I2C_DRIVERID_WM8510, + .attach_adapter = wm8510_i2c_attach, + .detach_client = wm8510_i2c_detach, + .command = NULL, +}; + +static struct i2c_client client_template = { + .name = "WM8510", + .driver = &wm8510_i2c_driver, +}; +#endif + +static int wm8510_probe(struct platform_device *pdev) +{ + struct snd_soc_device *socdev = platform_get_drvdata(pdev); + struct wm8510_setup_data *setup; + struct snd_soc_codec *codec; + int ret = 0; + + pr_info("WM8510 Audio Codec %s", WM8510_VERSION); + + setup = socdev->codec_data; + codec = kzalloc(sizeof(struct snd_soc_codec), GFP_KERNEL); + if (codec == NULL) + return -ENOMEM; + + socdev->codec = codec; + mutex_init(&codec->mutex); + INIT_LIST_HEAD(&codec->dapm_widgets); + INIT_LIST_HEAD(&codec->dapm_paths); + + wm8510_socdev = socdev; +#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE) + if (setup->i2c_address) { + normal_i2c[0] = setup->i2c_address; + codec->hw_write = (hw_write_t)i2c_master_send; + ret = i2c_add_driver(&wm8510_i2c_driver); + if (ret != 0) + printk(KERN_ERR "can't add i2c driver"); + } +#else + /* Add other interfaces here */ +#endif + return ret; +} + +/* power down chip */ +static int wm8510_remove(struct platform_device *pdev) +{ + struct snd_soc_device *socdev = platform_get_drvdata(pdev); + struct snd_soc_codec *codec = socdev->codec; + + if (codec->control_data) + wm8510_set_bias_level(codec, SND_SOC_BIAS_OFF); + + snd_soc_free_pcms(socdev); + snd_soc_dapm_free(socdev); +#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE) + i2c_del_driver(&wm8510_i2c_driver); +#endif + kfree(codec); + + return 0; +} + +struct snd_soc_codec_device soc_codec_dev_wm8510 = { + .probe = wm8510_probe, + .remove = wm8510_remove, + .suspend = wm8510_suspend, + .resume = wm8510_resume, +}; +EXPORT_SYMBOL_GPL(soc_codec_dev_wm8510); + +MODULE_DESCRIPTION("ASoC WM8510 driver"); +MODULE_AUTHOR("Liam Girdwood"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/codecs/wm8510.h b/sound/soc/codecs/wm8510.h new file mode 100644 index 00000000000..f5d2e42eb3f --- /dev/null +++ b/sound/soc/codecs/wm8510.h @@ -0,0 +1,103 @@ +/* + * wm8510.h -- WM8510 Soc Audio driver + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. + */ + +#ifndef _WM8510_H +#define _WM8510_H + +/* WM8510 register space */ + +#define WM8510_RESET 0x0 +#define WM8510_POWER1 0x1 +#define WM8510_POWER2 0x2 +#define WM8510_POWER3 0x3 +#define WM8510_IFACE 0x4 +#define WM8510_COMP 0x5 +#define WM8510_CLOCK 0x6 +#define WM8510_ADD 0x7 +#define WM8510_GPIO 0x8 +#define WM8510_DAC 0xa +#define WM8510_DACVOL 0xb +#define WM8510_ADC 0xe +#define WM8510_ADCVOL 0xf +#define WM8510_EQ1 0x12 +#define WM8510_EQ2 0x13 +#define WM8510_EQ3 0x14 +#define WM8510_EQ4 0x15 +#define WM8510_EQ5 0x16 +#define WM8510_DACLIM1 0x18 +#define WM8510_DACLIM2 0x19 +#define WM8510_NOTCH1 0x1b +#define WM8510_NOTCH2 0x1c +#define WM8510_NOTCH3 0x1d +#define WM8510_NOTCH4 0x1e +#define WM8510_ALC1 0x20 +#define WM8510_ALC2 0x21 +#define WM8510_ALC3 0x22 +#define WM8510_NGATE 0x23 +#define WM8510_PLLN 0x24 +#define WM8510_PLLK1 0x25 +#define WM8510_PLLK2 0x26 +#define WM8510_PLLK3 0x27 +#define WM8510_ATTEN 0x28 +#define WM8510_INPUT 0x2c +#define WM8510_INPPGA 0x2d +#define WM8510_ADCBOOST 0x2f +#define WM8510_OUTPUT 0x31 +#define WM8510_SPKMIX 0x32 +#define WM8510_SPKVOL 0x36 +#define WM8510_MONOMIX 0x38 + +#define WM8510_CACHEREGNUM 57 + +/* Clock divider Id's */ +#define WM8510_OPCLKDIV 0 +#define WM8510_MCLKDIV 1 +#define WM8510_ADCCLK 2 +#define WM8510_DACCLK 3 +#define WM8510_BCLKDIV 4 + +/* DAC clock dividers */ +#define WM8510_DACCLK_F2 (1 << 3) +#define WM8510_DACCLK_F4 (0 << 3) + +/* ADC clock dividers */ +#define WM8510_ADCCLK_F2 (1 << 3) +#define WM8510_ADCCLK_F4 (0 << 3) + +/* PLL Out dividers */ +#define WM8510_OPCLKDIV_1 (0 << 4) +#define WM8510_OPCLKDIV_2 (1 << 4) +#define WM8510_OPCLKDIV_3 (2 << 4) +#define WM8510_OPCLKDIV_4 (3 << 4) + +/* BCLK clock dividers */ +#define WM8510_BCLKDIV_1 (0 << 2) +#define WM8510_BCLKDIV_2 (1 << 2) +#define WM8510_BCLKDIV_4 (2 << 2) +#define WM8510_BCLKDIV_8 (3 << 2) +#define WM8510_BCLKDIV_16 (4 << 2) +#define WM8510_BCLKDIV_32 (5 << 2) + +/* MCLK clock dividers */ +#define WM8510_MCLKDIV_1 (0 << 5) +#define WM8510_MCLKDIV_1_5 (1 << 5) +#define WM8510_MCLKDIV_2 (2 << 5) +#define WM8510_MCLKDIV_3 (3 << 5) +#define WM8510_MCLKDIV_4 (4 << 5) +#define WM8510_MCLKDIV_6 (5 << 5) +#define WM8510_MCLKDIV_8 (6 << 5) +#define WM8510_MCLKDIV_12 (7 << 5) + +struct wm8510_setup_data { + unsigned short i2c_address; +}; + +extern struct snd_soc_dai wm8510_dai; +extern struct snd_soc_codec_device soc_codec_dev_wm8510; + +#endif diff --git a/sound/soc/codecs/wm8731.c b/sound/soc/codecs/wm8731.c index 0cf9265fca8..369d39c3f74 100644 --- a/sound/soc/codecs/wm8731.c +++ b/sound/soc/codecs/wm8731.c @@ -31,25 +31,6 @@ #define AUDIO_NAME "wm8731" #define WM8731_VERSION "0.13" -/* - * Debug - */ - -#define WM8731_DEBUG 0 - -#ifdef WM8731_DEBUG -#define dbg(format, arg...) \ - printk(KERN_DEBUG AUDIO_NAME ": " format "\n" , ## arg) -#else -#define dbg(format, arg...) do {} while (0) -#endif -#define err(format, arg...) \ - printk(KERN_ERR AUDIO_NAME ": " format "\n" , ## arg) -#define info(format, arg...) \ - printk(KERN_INFO AUDIO_NAME ": " format "\n" , ## arg) -#define warn(format, arg...) \ - printk(KERN_WARNING AUDIO_NAME ": " format "\n" , ## arg) - struct snd_soc_codec_device soc_codec_dev_wm8731; /* codec private data */ @@ -193,7 +174,7 @@ SND_SOC_DAPM_INPUT("RLINEIN"), SND_SOC_DAPM_INPUT("LLINEIN"), }; -static const char *intercon[][3] = { +static const struct snd_soc_dapm_route intercon[] = { /* output mixer */ {"Output Mixer", "Line Bypass Switch", "Line Input"}, {"Output Mixer", "HiFi Playback Switch", "DAC"}, @@ -214,22 +195,14 @@ static const char *intercon[][3] = { {"Line Input", NULL, "LLINEIN"}, {"Line Input", NULL, "RLINEIN"}, {"Mic Bias", NULL, "MICIN"}, - - /* terminator */ - {NULL, NULL, NULL}, }; static int wm8731_add_widgets(struct snd_soc_codec *codec) { - int i; - - for (i = 0; i < ARRAY_SIZE(wm8731_dapm_widgets); i++) - snd_soc_dapm_new_control(codec, &wm8731_dapm_widgets[i]); + snd_soc_dapm_new_controls(codec, wm8731_dapm_widgets, + ARRAY_SIZE(wm8731_dapm_widgets)); - /* set up audio path interconnects */ - for (i = 0; intercon[i][0] != NULL; i++) - snd_soc_dapm_connect_input(codec, intercon[i][0], - intercon[i][1], intercon[i][2]); + snd_soc_dapm_add_routes(codec, intercon, ARRAY_SIZE(intercon)); snd_soc_dapm_new_widgets(codec); return 0; @@ -345,7 +318,7 @@ static void wm8731_shutdown(struct snd_pcm_substream *substream) } } -static int wm8731_mute(struct snd_soc_codec_dai *dai, int mute) +static int wm8731_mute(struct snd_soc_dai *dai, int mute) { struct snd_soc_codec *codec = dai->codec; u16 mute_reg = wm8731_read_reg_cache(codec, WM8731_APDIGI) & 0xfff7; @@ -357,7 +330,7 @@ static int wm8731_mute(struct snd_soc_codec_dai *dai, int mute) return 0; } -static int wm8731_set_dai_sysclk(struct snd_soc_codec_dai *codec_dai, +static int wm8731_set_dai_sysclk(struct snd_soc_dai *codec_dai, int clk_id, unsigned int freq, int dir) { struct snd_soc_codec *codec = codec_dai->codec; @@ -376,7 +349,7 @@ static int wm8731_set_dai_sysclk(struct snd_soc_codec_dai *codec_dai, } -static int wm8731_set_dai_fmt(struct snd_soc_codec_dai *codec_dai, +static int wm8731_set_dai_fmt(struct snd_soc_dai *codec_dai, unsigned int fmt) { struct snd_soc_codec *codec = codec_dai->codec; @@ -435,29 +408,29 @@ static int wm8731_set_dai_fmt(struct snd_soc_codec_dai *codec_dai, return 0; } -static int wm8731_dapm_event(struct snd_soc_codec *codec, int event) +static int wm8731_set_bias_level(struct snd_soc_codec *codec, + enum snd_soc_bias_level level) { u16 reg = wm8731_read_reg_cache(codec, WM8731_PWR) & 0xff7f; - switch (event) { - case SNDRV_CTL_POWER_D0: /* full On */ + switch (level) { + case SND_SOC_BIAS_ON: /* vref/mid, osc on, dac unmute */ wm8731_write(codec, WM8731_PWR, reg); break; - case SNDRV_CTL_POWER_D1: /* partial On */ - case SNDRV_CTL_POWER_D2: /* partial On */ + case SND_SOC_BIAS_PREPARE: break; - case SNDRV_CTL_POWER_D3hot: /* Off, with power */ + case SND_SOC_BIAS_STANDBY: /* everything off except vref/vmid, */ wm8731_write(codec, WM8731_PWR, reg | 0x0040); break; - case SNDRV_CTL_POWER_D3cold: /* Off, without power */ + case SND_SOC_BIAS_OFF: /* everything off, dac mute, inactive */ wm8731_write(codec, WM8731_ACTIVE, 0x0); wm8731_write(codec, WM8731_PWR, 0xffff); break; } - codec->dapm_state = event; + codec->bias_level = level; return 0; } @@ -470,7 +443,7 @@ static int wm8731_dapm_event(struct snd_soc_codec *codec, int event) #define WM8731_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE |\ SNDRV_PCM_FMTBIT_S24_LE) -struct snd_soc_codec_dai wm8731_dai = { +struct snd_soc_dai wm8731_dai = { .name = "WM8731", .playback = { .stream_name = "Playback", @@ -503,7 +476,7 @@ static int wm8731_suspend(struct platform_device *pdev, pm_message_t state) struct snd_soc_codec *codec = socdev->codec; wm8731_write(codec, WM8731_ACTIVE, 0x0); - wm8731_dapm_event(codec, SNDRV_CTL_POWER_D3cold); + wm8731_set_bias_level(codec, SND_SOC_BIAS_OFF); return 0; } @@ -521,8 +494,8 @@ static int wm8731_resume(struct platform_device *pdev) data[1] = cache[i] & 0x00ff; codec->hw_write(codec->control_data, data, 2); } - wm8731_dapm_event(codec, SNDRV_CTL_POWER_D3hot); - wm8731_dapm_event(codec, codec->suspend_dapm_state); + wm8731_set_bias_level(codec, SND_SOC_BIAS_STANDBY); + wm8731_set_bias_level(codec, codec->suspend_bias_level); return 0; } @@ -539,10 +512,10 @@ static int wm8731_init(struct snd_soc_device *socdev) codec->owner = THIS_MODULE; codec->read = wm8731_read_reg_cache; codec->write = wm8731_write; - codec->dapm_event = wm8731_dapm_event; + codec->set_bias_level = wm8731_set_bias_level; codec->dai = &wm8731_dai; codec->num_dai = 1; - codec->reg_cache_size = sizeof(wm8731_reg); + codec->reg_cache_size = ARRAY_SIZE(wm8731_reg); codec->reg_cache = kmemdup(wm8731_reg, sizeof(wm8731_reg), GFP_KERNEL); if (codec->reg_cache == NULL) return -ENOMEM; @@ -557,7 +530,7 @@ static int wm8731_init(struct snd_soc_device *socdev) } /* power on device */ - wm8731_dapm_event(codec, SNDRV_CTL_POWER_D3hot); + wm8731_set_bias_level(codec, SND_SOC_BIAS_STANDBY); /* set the update bits */ reg = wm8731_read_reg_cache(codec, WM8731_LOUT1V); @@ -632,13 +605,13 @@ static int wm8731_codec_probe(struct i2c_adapter *adap, int addr, int kind) ret = i2c_attach_client(i2c); if (ret < 0) { - err("failed to attach codec at addr %x\n", addr); + pr_err("failed to attach codec at addr %x\n", addr); goto err; } ret = wm8731_init(socdev); if (ret < 0) { - err("failed to initialise WM8731\n"); + pr_err("failed to initialise WM8731\n"); goto err; } return ret; @@ -689,7 +662,7 @@ static int wm8731_probe(struct platform_device *pdev) struct wm8731_priv *wm8731; int ret = 0; - info("WM8731 Audio Codec %s", WM8731_VERSION); + pr_info("WM8731 Audio Codec %s", WM8731_VERSION); setup = socdev->codec_data; codec = kzalloc(sizeof(struct snd_soc_codec), GFP_KERNEL); @@ -730,7 +703,7 @@ static int wm8731_remove(struct platform_device *pdev) struct snd_soc_codec *codec = socdev->codec; if (codec->control_data) - wm8731_dapm_event(codec, SNDRV_CTL_POWER_D3cold); + wm8731_set_bias_level(codec, SND_SOC_BIAS_OFF); snd_soc_free_pcms(socdev); snd_soc_dapm_free(socdev); diff --git a/sound/soc/codecs/wm8731.h b/sound/soc/codecs/wm8731.h index 5bcab6a7afb..99f2e3c60e3 100644 --- a/sound/soc/codecs/wm8731.h +++ b/sound/soc/codecs/wm8731.h @@ -38,7 +38,7 @@ struct wm8731_setup_data { unsigned short i2c_address; }; -extern struct snd_soc_codec_dai wm8731_dai; +extern struct snd_soc_dai wm8731_dai; extern struct snd_soc_codec_device soc_codec_dev_wm8731; #endif diff --git a/sound/soc/codecs/wm8750.c b/sound/soc/codecs/wm8750.c index 16cd5d4d5ad..e23cb09f0d1 100644 --- a/sound/soc/codecs/wm8750.c +++ b/sound/soc/codecs/wm8750.c @@ -31,25 +31,6 @@ #define AUDIO_NAME "WM8750" #define WM8750_VERSION "0.12" -/* - * Debug - */ - -#define WM8750_DEBUG 0 - -#ifdef WM8750_DEBUG -#define dbg(format, arg...) \ - printk(KERN_DEBUG AUDIO_NAME ": " format "\n" , ## arg) -#else -#define dbg(format, arg...) do {} while (0) -#endif -#define err(format, arg...) \ - printk(KERN_ERR AUDIO_NAME ": " format "\n" , ## arg) -#define info(format, arg...) \ - printk(KERN_INFO AUDIO_NAME ": " format "\n" , ## arg) -#define warn(format, arg...) \ - printk(KERN_WARNING AUDIO_NAME ": " format "\n" , ## arg) - /* codec private data */ struct wm8750_priv { unsigned int sysclk; @@ -378,7 +359,7 @@ static const struct snd_soc_dapm_widget wm8750_dapm_widgets[] = { SND_SOC_DAPM_INPUT("RINPUT3"), }; -static const char *audio_map[][3] = { +static const struct snd_soc_dapm_route audio_map[] = { /* left mixer */ {"Left Mixer", "Playback Switch", "Left DAC"}, {"Left Mixer", "Left Bypass Switch", "Left Line Mux"}, @@ -470,22 +451,14 @@ static const char *audio_map[][3] = { /* ADC */ {"Left ADC", NULL, "Left ADC Mux"}, {"Right ADC", NULL, "Right ADC Mux"}, - - /* terminator */ - {NULL, NULL, NULL}, }; static int wm8750_add_widgets(struct snd_soc_codec *codec) { - int i; - - for (i = 0; i < ARRAY_SIZE(wm8750_dapm_widgets); i++) - snd_soc_dapm_new_control(codec, &wm8750_dapm_widgets[i]); + snd_soc_dapm_new_controls(codec, wm8750_dapm_widgets, + ARRAY_SIZE(wm8750_dapm_widgets)); - /* set up audio path audio_mapnects */ - for (i = 0; audio_map[i][0] != NULL; i++) - snd_soc_dapm_connect_input(codec, audio_map[i][0], - audio_map[i][1], audio_map[i][2]); + snd_soc_dapm_add_routes(codec, audio_map, ARRAY_SIZE(audio_map)); snd_soc_dapm_new_widgets(codec); return 0; @@ -563,7 +536,7 @@ static inline int get_coeff(int mclk, int rate) return -EINVAL; } -static int wm8750_set_dai_sysclk(struct snd_soc_codec_dai *codec_dai, +static int wm8750_set_dai_sysclk(struct snd_soc_dai *codec_dai, int clk_id, unsigned int freq, int dir) { struct snd_soc_codec *codec = codec_dai->codec; @@ -581,7 +554,7 @@ static int wm8750_set_dai_sysclk(struct snd_soc_codec_dai *codec_dai, return -EINVAL; } -static int wm8750_set_dai_fmt(struct snd_soc_codec_dai *codec_dai, +static int wm8750_set_dai_fmt(struct snd_soc_dai *codec_dai, unsigned int fmt) { struct snd_soc_codec *codec = codec_dai->codec; @@ -674,7 +647,7 @@ static int wm8750_pcm_hw_params(struct snd_pcm_substream *substream, return 0; } -static int wm8750_mute(struct snd_soc_codec_dai *dai, int mute) +static int wm8750_mute(struct snd_soc_dai *dai, int mute) { struct snd_soc_codec *codec = dai->codec; u16 mute_reg = wm8750_read_reg_cache(codec, WM8750_ADCDAC) & 0xfff7; @@ -686,29 +659,29 @@ static int wm8750_mute(struct snd_soc_codec_dai *dai, int mute) return 0; } -static int wm8750_dapm_event(struct snd_soc_codec *codec, int event) +static int wm8750_set_bias_level(struct snd_soc_codec *codec, + enum snd_soc_bias_level level) { u16 pwr_reg = wm8750_read_reg_cache(codec, WM8750_PWR1) & 0xfe3e; - switch (event) { - case SNDRV_CTL_POWER_D0: /* full On */ + switch (level) { + case SND_SOC_BIAS_ON: /* set vmid to 50k and unmute dac */ wm8750_write(codec, WM8750_PWR1, pwr_reg | 0x00c0); break; - case SNDRV_CTL_POWER_D1: /* partial On */ - case SNDRV_CTL_POWER_D2: /* partial On */ + case SND_SOC_BIAS_PREPARE: /* set vmid to 5k for quick power up */ wm8750_write(codec, WM8750_PWR1, pwr_reg | 0x01c1); break; - case SNDRV_CTL_POWER_D3hot: /* Off, with power */ + case SND_SOC_BIAS_STANDBY: /* mute dac and set vmid to 500k, enable VREF */ wm8750_write(codec, WM8750_PWR1, pwr_reg | 0x0141); break; - case SNDRV_CTL_POWER_D3cold: /* Off, without power */ + case SND_SOC_BIAS_OFF: wm8750_write(codec, WM8750_PWR1, 0x0001); break; } - codec->dapm_state = event; + codec->bias_level = level; return 0; } @@ -719,7 +692,7 @@ static int wm8750_dapm_event(struct snd_soc_codec *codec, int event) #define WM8750_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE |\ SNDRV_PCM_FMTBIT_S24_LE) -struct snd_soc_codec_dai wm8750_dai = { +struct snd_soc_dai wm8750_dai = { .name = "WM8750", .playback = { .stream_name = "Playback", @@ -748,7 +721,7 @@ static void wm8750_work(struct work_struct *work) { struct snd_soc_codec *codec = container_of(work, struct snd_soc_codec, delayed_work.work); - wm8750_dapm_event(codec, codec->dapm_state); + wm8750_set_bias_level(codec, codec->bias_level); } static int wm8750_suspend(struct platform_device *pdev, pm_message_t state) @@ -756,7 +729,7 @@ static int wm8750_suspend(struct platform_device *pdev, pm_message_t state) struct snd_soc_device *socdev = platform_get_drvdata(pdev); struct snd_soc_codec *codec = socdev->codec; - wm8750_dapm_event(codec, SNDRV_CTL_POWER_D3cold); + wm8750_set_bias_level(codec, SND_SOC_BIAS_OFF); return 0; } @@ -777,12 +750,12 @@ static int wm8750_resume(struct platform_device *pdev) codec->hw_write(codec->control_data, data, 2); } - wm8750_dapm_event(codec, SNDRV_CTL_POWER_D3hot); + wm8750_set_bias_level(codec, SND_SOC_BIAS_STANDBY); /* charge wm8750 caps */ - if (codec->suspend_dapm_state == SNDRV_CTL_POWER_D0) { - wm8750_dapm_event(codec, SNDRV_CTL_POWER_D2); - codec->dapm_state = SNDRV_CTL_POWER_D0; + if (codec->suspend_bias_level == SND_SOC_BIAS_ON) { + wm8750_set_bias_level(codec, SND_SOC_BIAS_PREPARE); + codec->bias_level = SND_SOC_BIAS_ON; schedule_delayed_work(&codec->delayed_work, msecs_to_jiffies(1000)); } @@ -803,10 +776,10 @@ static int wm8750_init(struct snd_soc_device *socdev) codec->owner = THIS_MODULE; codec->read = wm8750_read_reg_cache; codec->write = wm8750_write; - codec->dapm_event = wm8750_dapm_event; + codec->set_bias_level = wm8750_set_bias_level; codec->dai = &wm8750_dai; codec->num_dai = 1; - codec->reg_cache_size = sizeof(wm8750_reg); + codec->reg_cache_size = ARRAY_SIZE(wm8750_reg); codec->reg_cache = kmemdup(wm8750_reg, sizeof(wm8750_reg), GFP_KERNEL); if (codec->reg_cache == NULL) return -ENOMEM; @@ -821,8 +794,8 @@ static int wm8750_init(struct snd_soc_device *socdev) } /* charge output caps */ - wm8750_dapm_event(codec, SNDRV_CTL_POWER_D2); - codec->dapm_state = SNDRV_CTL_POWER_D3hot; + wm8750_set_bias_level(codec, SND_SOC_BIAS_PREPARE); + codec->bias_level = SND_SOC_BIAS_STANDBY; schedule_delayed_work(&codec->delayed_work, msecs_to_jiffies(1000)); /* set the update bits */ @@ -904,13 +877,13 @@ static int wm8750_codec_probe(struct i2c_adapter *adap, int addr, int kind) ret = i2c_attach_client(i2c); if (ret < 0) { - err("failed to attach codec at addr %x\n", addr); + pr_err("failed to attach codec at addr %x\n", addr); goto err; } ret = wm8750_init(socdev); if (ret < 0) { - err("failed to initialise WM8750\n"); + pr_err("failed to initialise WM8750\n"); goto err; } return ret; @@ -961,7 +934,7 @@ static int wm8750_probe(struct platform_device *pdev) struct wm8750_priv *wm8750; int ret = 0; - info("WM8750 Audio Codec %s", WM8750_VERSION); + pr_info("WM8750 Audio Codec %s", WM8750_VERSION); codec = kzalloc(sizeof(struct snd_soc_codec), GFP_KERNEL); if (codec == NULL) return -ENOMEM; @@ -1021,7 +994,7 @@ static int wm8750_remove(struct platform_device *pdev) struct snd_soc_codec *codec = socdev->codec; if (codec->control_data) - wm8750_dapm_event(codec, SNDRV_CTL_POWER_D3cold); + wm8750_set_bias_level(codec, SND_SOC_BIAS_OFF); run_delayed_work(&codec->delayed_work); snd_soc_free_pcms(socdev); snd_soc_dapm_free(socdev); diff --git a/sound/soc/codecs/wm8750.h b/sound/soc/codecs/wm8750.h index a97a54a6348..8ef30e628b2 100644 --- a/sound/soc/codecs/wm8750.h +++ b/sound/soc/codecs/wm8750.h @@ -61,7 +61,7 @@ struct wm8750_setup_data { unsigned short i2c_address; }; -extern struct snd_soc_codec_dai wm8750_dai; +extern struct snd_soc_dai wm8750_dai; extern struct snd_soc_codec_device soc_codec_dev_wm8750; #endif diff --git a/sound/soc/codecs/wm8753.c b/sound/soc/codecs/wm8753.c index fb41826c4c4..8604809f0c3 100644 --- a/sound/soc/codecs/wm8753.c +++ b/sound/soc/codecs/wm8753.c @@ -55,25 +55,6 @@ #define AUDIO_NAME "wm8753" #define WM8753_VERSION "0.16" -/* - * Debug - */ - -#define WM8753_DEBUG 0 - -#ifdef WM8753_DEBUG -#define dbg(format, arg...) \ - printk(KERN_DEBUG AUDIO_NAME ": " format "\n" , ## arg) -#else -#define dbg(format, arg...) do {} while (0) -#endif -#define err(format, arg...) \ - printk(KERN_ERR AUDIO_NAME ": " format "\n" , ## arg) -#define info(format, arg...) \ - printk(KERN_INFO AUDIO_NAME ": " format "\n" , ## arg) -#define warn(format, arg...) \ - printk(KERN_WARNING AUDIO_NAME ": " format "\n" , ## arg) - static int caps_charge = 2000; module_param(caps_charge, int, 0); MODULE_PARM_DESC(caps_charge, "WM8753 cap charge time (msecs)"); @@ -260,28 +241,50 @@ static int wm8753_set_dai(struct snd_kcontrol *kcontrol, return 1; } -static const DECLARE_TLV_DB_LINEAR(rec_mix_tlv, -1500, 600); +static const DECLARE_TLV_DB_SCALE(rec_mix_tlv, -1500, 300, 0); +static const DECLARE_TLV_DB_SCALE(mic_preamp_tlv, 1200, 600, 0); +static const DECLARE_TLV_DB_SCALE(adc_tlv, -9750, 50, 1); +static const DECLARE_TLV_DB_SCALE(dac_tlv, -12750, 50, 1); +static const unsigned int out_tlv[] = { + TLV_DB_RANGE_HEAD(2), + /* 0000000 - 0101111 = "Analogue mute" */ + 0, 48, TLV_DB_SCALE_ITEM(-25500, 0, 0), + 48, 127, TLV_DB_SCALE_ITEM(-7300, 100, 0), +}; +static const DECLARE_TLV_DB_SCALE(mix_tlv, -1500, 300, 0); +static const DECLARE_TLV_DB_SCALE(voice_mix_tlv, -1200, 300, 0); +static const DECLARE_TLV_DB_SCALE(pga_tlv, -1725, 75, 0); static const struct snd_kcontrol_new wm8753_snd_controls[] = { -SOC_DOUBLE_R("PCM Volume", WM8753_LDAC, WM8753_RDAC, 0, 255, 0), - -SOC_DOUBLE_R("ADC Capture Volume", WM8753_LADC, WM8753_RADC, 0, 255, 0), - -SOC_DOUBLE_R("Headphone Playback Volume", WM8753_LOUT1V, WM8753_ROUT1V, 0, 127, 0), -SOC_DOUBLE_R("Speaker Playback Volume", WM8753_LOUT2V, WM8753_ROUT2V, 0, 127, 0), - -SOC_SINGLE("Mono Playback Volume", WM8753_MOUTV, 0, 127, 0), - -SOC_DOUBLE_R("Bypass Playback Volume", WM8753_LOUTM1, WM8753_ROUTM1, 4, 7, 1), -SOC_DOUBLE_R("Sidetone Playback Volume", WM8753_LOUTM2, WM8753_ROUTM2, 4, 7, 1), -SOC_DOUBLE_R("Voice Playback Volume", WM8753_LOUTM2, WM8753_ROUTM2, 0, 7, 1), - -SOC_DOUBLE_R("Headphone Playback ZC Switch", WM8753_LOUT1V, WM8753_ROUT1V, 7, 1, 0), -SOC_DOUBLE_R("Speaker Playback ZC Switch", WM8753_LOUT2V, WM8753_ROUT2V, 7, 1, 0), - -SOC_SINGLE("Mono Bypass Playback Volume", WM8753_MOUTM1, 4, 7, 1), -SOC_SINGLE("Mono Sidetone Playback Volume", WM8753_MOUTM2, 4, 7, 1), -SOC_SINGLE("Mono Voice Playback Volume", WM8753_MOUTM2, 0, 7, 1), +SOC_DOUBLE_R_TLV("PCM Volume", WM8753_LDAC, WM8753_RDAC, 0, 255, 0, dac_tlv), + +SOC_DOUBLE_R_TLV("ADC Capture Volume", WM8753_LADC, WM8753_RADC, 0, 255, 0, + adc_tlv), + +SOC_DOUBLE_R_TLV("Headphone Playback Volume", WM8753_LOUT1V, WM8753_ROUT1V, + 0, 127, 0, out_tlv), +SOC_DOUBLE_R_TLV("Speaker Playback Volume", WM8753_LOUT2V, WM8753_ROUT2V, 0, + 127, 0, out_tlv), + +SOC_SINGLE_TLV("Mono Playback Volume", WM8753_MOUTV, 0, 127, 0, out_tlv), + +SOC_DOUBLE_R_TLV("Bypass Playback Volume", WM8753_LOUTM1, WM8753_ROUTM1, 4, 7, + 1, mix_tlv), +SOC_DOUBLE_R_TLV("Sidetone Playback Volume", WM8753_LOUTM2, WM8753_ROUTM2, 4, + 7, 1, mix_tlv), +SOC_DOUBLE_R_TLV("Voice Playback Volume", WM8753_LOUTM2, WM8753_ROUTM2, 0, 7, + 1, voice_mix_tlv), + +SOC_DOUBLE_R("Headphone Playback ZC Switch", WM8753_LOUT1V, WM8753_ROUT1V, 7, + 1, 0), +SOC_DOUBLE_R("Speaker Playback ZC Switch", WM8753_LOUT2V, WM8753_ROUT2V, 7, + 1, 0), + +SOC_SINGLE_TLV("Mono Bypass Playback Volume", WM8753_MOUTM1, 4, 7, 1, mix_tlv), +SOC_SINGLE_TLV("Mono Sidetone Playback Volume", WM8753_MOUTM2, 4, 7, 1, + mix_tlv), +SOC_SINGLE_TLV("Mono Voice Playback Volume", WM8753_MOUTM2, 0, 7, 1, + voice_mix_tlv), SOC_SINGLE("Mono Playback ZC Switch", WM8753_MOUTV, 7, 1, 0), SOC_ENUM("Bass Boost", wm8753_enum[0]), @@ -291,10 +294,13 @@ SOC_SINGLE("Bass Volume", WM8753_BASS, 0, 15, 1), SOC_SINGLE("Treble Volume", WM8753_TREBLE, 0, 15, 1), SOC_ENUM("Treble Cut-off", wm8753_enum[2]), -SOC_DOUBLE_TLV("Sidetone Capture Volume", WM8753_RECMIX1, 0, 4, 7, 1, rec_mix_tlv), -SOC_SINGLE_TLV("Voice Sidetone Capture Volume", WM8753_RECMIX2, 0, 7, 1, rec_mix_tlv), +SOC_DOUBLE_TLV("Sidetone Capture Volume", WM8753_RECMIX1, 0, 4, 7, 1, + rec_mix_tlv), +SOC_SINGLE_TLV("Voice Sidetone Capture Volume", WM8753_RECMIX2, 0, 7, 1, + rec_mix_tlv), -SOC_DOUBLE_R("Capture Volume", WM8753_LINVOL, WM8753_RINVOL, 0, 63, 0), +SOC_DOUBLE_R_TLV("Capture Volume", WM8753_LINVOL, WM8753_RINVOL, 0, 63, 0, + pga_tlv), SOC_DOUBLE_R("Capture ZC Switch", WM8753_LINVOL, WM8753_RINVOL, 6, 1, 0), SOC_DOUBLE_R("Capture Switch", WM8753_LINVOL, WM8753_RINVOL, 7, 1, 1), @@ -326,8 +332,8 @@ SOC_ENUM("De-emphasis", wm8753_enum[8]), SOC_ENUM("Playback Mono Mix", wm8753_enum[9]), SOC_ENUM("Playback Phase", wm8753_enum[10]), -SOC_SINGLE("Mic2 Capture Volume", WM8753_INCTL1, 7, 3, 0), -SOC_SINGLE("Mic1 Capture Volume", WM8753_INCTL1, 5, 3, 0), +SOC_SINGLE_TLV("Mic2 Capture Volume", WM8753_INCTL1, 7, 3, 0, mic_preamp_tlv), +SOC_SINGLE_TLV("Mic1 Capture Volume", WM8753_INCTL1, 5, 3, 0, mic_preamp_tlv), SOC_ENUM_EXT("DAI Mode", wm8753_enum[26], wm8753_get_dai, wm8753_set_dai), @@ -523,7 +529,7 @@ SND_SOC_DAPM_INPUT("MIC2"), SND_SOC_DAPM_VMID("VREF"), }; -static const char *audio_map[][3] = { +static const struct snd_soc_dapm_route audio_map[] = { /* left mixer */ {"Left Mixer", "Left Playback Switch", "Left DAC"}, {"Left Mixer", "Voice Playback Switch", "Voice DAC"}, @@ -674,23 +680,14 @@ static const char *audio_map[][3] = { /* ACOP */ {"ACOP", NULL, "ALC Mixer"}, - - /* terminator */ - {NULL, NULL, NULL}, }; static int wm8753_add_widgets(struct snd_soc_codec *codec) { - int i; + snd_soc_dapm_new_controls(codec, wm8753_dapm_widgets, + ARRAY_SIZE(wm8753_dapm_widgets)); - for (i = 0; i < ARRAY_SIZE(wm8753_dapm_widgets); i++) - snd_soc_dapm_new_control(codec, &wm8753_dapm_widgets[i]); - - /* set up the WM8753 audio map */ - for (i = 0; audio_map[i][0] != NULL; i++) { - snd_soc_dapm_connect_input(codec, audio_map[i][0], - audio_map[i][1], audio_map[i][2]); - } + snd_soc_dapm_add_routes(codec, audio_map, ARRAY_SIZE(audio_map)); snd_soc_dapm_new_widgets(codec); return 0; @@ -743,7 +740,7 @@ static void pll_factors(struct _pll_div *pll_div, unsigned int target, pll_div->k = K; } -static int wm8753_set_dai_pll(struct snd_soc_codec_dai *codec_dai, +static int wm8753_set_dai_pll(struct snd_soc_dai *codec_dai, int pll_id, unsigned int freq_in, unsigned int freq_out) { u16 reg, enable; @@ -866,7 +863,7 @@ static int get_coeff(int mclk, int rate) /* * Clock after PLL and dividers */ -static int wm8753_set_dai_sysclk(struct snd_soc_codec_dai *codec_dai, +static int wm8753_set_dai_sysclk(struct snd_soc_dai *codec_dai, int clk_id, unsigned int freq, int dir) { struct snd_soc_codec *codec = codec_dai->codec; @@ -893,7 +890,7 @@ static int wm8753_set_dai_sysclk(struct snd_soc_codec_dai *codec_dai, /* * Set's ADC and Voice DAC format. */ -static int wm8753_vdac_adc_set_dai_fmt(struct snd_soc_codec_dai *codec_dai, +static int wm8753_vdac_adc_set_dai_fmt(struct snd_soc_dai *codec_dai, unsigned int fmt) { struct snd_soc_codec *codec = codec_dai->codec; @@ -963,7 +960,7 @@ static int wm8753_pcm_hw_params(struct snd_pcm_substream *substream, /* * Set's PCM dai fmt and BCLK. */ -static int wm8753_pcm_set_dai_fmt(struct snd_soc_codec_dai *codec_dai, +static int wm8753_pcm_set_dai_fmt(struct snd_soc_dai *codec_dai, unsigned int fmt) { struct snd_soc_codec *codec = codec_dai->codec; @@ -1029,7 +1026,7 @@ static int wm8753_pcm_set_dai_fmt(struct snd_soc_codec_dai *codec_dai, return 0; } -static int wm8753_set_dai_clkdiv(struct snd_soc_codec_dai *codec_dai, +static int wm8753_set_dai_clkdiv(struct snd_soc_dai *codec_dai, int div_id, int div) { struct snd_soc_codec *codec = codec_dai->codec; @@ -1057,7 +1054,7 @@ static int wm8753_set_dai_clkdiv(struct snd_soc_codec_dai *codec_dai, /* * Set's HiFi DAC format. */ -static int wm8753_hdac_set_dai_fmt(struct snd_soc_codec_dai *codec_dai, +static int wm8753_hdac_set_dai_fmt(struct snd_soc_dai *codec_dai, unsigned int fmt) { struct snd_soc_codec *codec = codec_dai->codec; @@ -1090,7 +1087,7 @@ static int wm8753_hdac_set_dai_fmt(struct snd_soc_codec_dai *codec_dai, /* * Set's I2S DAI format. */ -static int wm8753_i2s_set_dai_fmt(struct snd_soc_codec_dai *codec_dai, +static int wm8753_i2s_set_dai_fmt(struct snd_soc_dai *codec_dai, unsigned int fmt) { struct snd_soc_codec *codec = codec_dai->codec; @@ -1198,7 +1195,7 @@ static int wm8753_i2s_hw_params(struct snd_pcm_substream *substream, return 0; } -static int wm8753_mode1v_set_dai_fmt(struct snd_soc_codec_dai *codec_dai, +static int wm8753_mode1v_set_dai_fmt(struct snd_soc_dai *codec_dai, unsigned int fmt) { struct snd_soc_codec *codec = codec_dai->codec; @@ -1213,7 +1210,7 @@ static int wm8753_mode1v_set_dai_fmt(struct snd_soc_codec_dai *codec_dai, return wm8753_pcm_set_dai_fmt(codec_dai, fmt); } -static int wm8753_mode1h_set_dai_fmt(struct snd_soc_codec_dai *codec_dai, +static int wm8753_mode1h_set_dai_fmt(struct snd_soc_dai *codec_dai, unsigned int fmt) { if (wm8753_hdac_set_dai_fmt(codec_dai, fmt) < 0) @@ -1221,7 +1218,7 @@ static int wm8753_mode1h_set_dai_fmt(struct snd_soc_codec_dai *codec_dai, return wm8753_i2s_set_dai_fmt(codec_dai, fmt); } -static int wm8753_mode2_set_dai_fmt(struct snd_soc_codec_dai *codec_dai, +static int wm8753_mode2_set_dai_fmt(struct snd_soc_dai *codec_dai, unsigned int fmt) { struct snd_soc_codec *codec = codec_dai->codec; @@ -1236,7 +1233,7 @@ static int wm8753_mode2_set_dai_fmt(struct snd_soc_codec_dai *codec_dai, return wm8753_i2s_set_dai_fmt(codec_dai, fmt); } -static int wm8753_mode3_4_set_dai_fmt(struct snd_soc_codec_dai *codec_dai, +static int wm8753_mode3_4_set_dai_fmt(struct snd_soc_dai *codec_dai, unsigned int fmt) { struct snd_soc_codec *codec = codec_dai->codec; @@ -1253,7 +1250,7 @@ static int wm8753_mode3_4_set_dai_fmt(struct snd_soc_codec_dai *codec_dai, return wm8753_i2s_set_dai_fmt(codec_dai, fmt); } -static int wm8753_mute(struct snd_soc_codec_dai *dai, int mute) +static int wm8753_mute(struct snd_soc_dai *dai, int mute) { struct snd_soc_codec *codec = dai->codec; u16 mute_reg = wm8753_read_reg_cache(codec, WM8753_DAC) & 0xfff7; @@ -1274,29 +1271,29 @@ static int wm8753_mute(struct snd_soc_codec_dai *dai, int mute) return 0; } -static int wm8753_dapm_event(struct snd_soc_codec *codec, int event) +static int wm8753_set_bias_level(struct snd_soc_codec *codec, + enum snd_soc_bias_level level) { u16 pwr_reg = wm8753_read_reg_cache(codec, WM8753_PWR1) & 0xfe3e; - switch (event) { - case SNDRV_CTL_POWER_D0: /* full On */ + switch (level) { + case SND_SOC_BIAS_ON: /* set vmid to 50k and unmute dac */ wm8753_write(codec, WM8753_PWR1, pwr_reg | 0x00c0); break; - case SNDRV_CTL_POWER_D1: /* partial On */ - case SNDRV_CTL_POWER_D2: /* partial On */ + case SND_SOC_BIAS_PREPARE: /* set vmid to 5k for quick power up */ wm8753_write(codec, WM8753_PWR1, pwr_reg | 0x01c1); break; - case SNDRV_CTL_POWER_D3hot: /* Off, with power */ + case SND_SOC_BIAS_STANDBY: /* mute dac and set vmid to 500k, enable VREF */ wm8753_write(codec, WM8753_PWR1, pwr_reg | 0x0141); break; - case SNDRV_CTL_POWER_D3cold: /* Off, without power */ + case SND_SOC_BIAS_OFF: wm8753_write(codec, WM8753_PWR1, 0x0001); break; } - codec->dapm_state = event; + codec->bias_level = level; return 0; } @@ -1319,7 +1316,7 @@ static int wm8753_dapm_event(struct snd_soc_codec *codec, int event) * 3. Voice disabled - HIFI over HIFI * 4. Voice disabled - HIFI over HIFI, uses voice DAI LRC for capture */ -static const struct snd_soc_codec_dai wm8753_all_dai[] = { +static const struct snd_soc_dai wm8753_all_dai[] = { /* DAI HiFi mode 1 */ { .name = "WM8753 HiFi", .id = 1, @@ -1459,7 +1456,7 @@ static const struct snd_soc_codec_dai wm8753_all_dai[] = { }, }; -struct snd_soc_codec_dai wm8753_dai[2]; +struct snd_soc_dai wm8753_dai[2]; EXPORT_SYMBOL_GPL(wm8753_dai); static void wm8753_set_dai_mode(struct snd_soc_codec *codec, unsigned int mode) @@ -1500,7 +1497,7 @@ static void wm8753_work(struct work_struct *work) { struct snd_soc_codec *codec = container_of(work, struct snd_soc_codec, delayed_work.work); - wm8753_dapm_event(codec, codec->dapm_state); + wm8753_set_bias_level(codec, codec->bias_level); } static int wm8753_suspend(struct platform_device *pdev, pm_message_t state) @@ -1512,7 +1509,7 @@ static int wm8753_suspend(struct platform_device *pdev, pm_message_t state) if (!codec->card) return 0; - wm8753_dapm_event(codec, SNDRV_CTL_POWER_D3cold); + wm8753_set_bias_level(codec, SND_SOC_BIAS_OFF); return 0; } @@ -1537,12 +1534,12 @@ static int wm8753_resume(struct platform_device *pdev) codec->hw_write(codec->control_data, data, 2); } - wm8753_dapm_event(codec, SNDRV_CTL_POWER_D3hot); + wm8753_set_bias_level(codec, SND_SOC_BIAS_STANDBY); /* charge wm8753 caps */ - if (codec->suspend_dapm_state == SNDRV_CTL_POWER_D0) { - wm8753_dapm_event(codec, SNDRV_CTL_POWER_D2); - codec->dapm_state = SNDRV_CTL_POWER_D0; + if (codec->suspend_bias_level == SND_SOC_BIAS_ON) { + wm8753_set_bias_level(codec, SND_SOC_BIAS_PREPARE); + codec->bias_level = SND_SOC_BIAS_ON; schedule_delayed_work(&codec->delayed_work, msecs_to_jiffies(caps_charge)); } @@ -1563,10 +1560,10 @@ static int wm8753_init(struct snd_soc_device *socdev) codec->owner = THIS_MODULE; codec->read = wm8753_read_reg_cache; codec->write = wm8753_write; - codec->dapm_event = wm8753_dapm_event; + codec->set_bias_level = wm8753_set_bias_level; codec->dai = wm8753_dai; codec->num_dai = 2; - codec->reg_cache_size = sizeof(wm8753_reg); + codec->reg_cache_size = ARRAY_SIZE(wm8753_reg); codec->reg_cache = kmemdup(wm8753_reg, sizeof(wm8753_reg), GFP_KERNEL); if (codec->reg_cache == NULL) @@ -1584,8 +1581,8 @@ static int wm8753_init(struct snd_soc_device *socdev) } /* charge output caps */ - wm8753_dapm_event(codec, SNDRV_CTL_POWER_D2); - codec->dapm_state = SNDRV_CTL_POWER_D3hot; + wm8753_set_bias_level(codec, SND_SOC_BIAS_PREPARE); + codec->bias_level = SND_SOC_BIAS_STANDBY; schedule_delayed_work(&codec->delayed_work, msecs_to_jiffies(caps_charge)); @@ -1673,13 +1670,13 @@ static int wm8753_codec_probe(struct i2c_adapter *adap, int addr, int kind) ret = i2c_attach_client(i2c); if (ret < 0) { - err("failed to attach codec at addr %x\n", addr); + pr_err("failed to attach codec at addr %x\n", addr); goto err; } ret = wm8753_init(socdev); if (ret < 0) { - err("failed to initialise WM8753\n"); + pr_err("failed to initialise WM8753\n"); goto err; } @@ -1731,7 +1728,7 @@ static int wm8753_probe(struct platform_device *pdev) struct wm8753_priv *wm8753; int ret = 0; - info("WM8753 Audio Codec %s", WM8753_VERSION); + pr_info("WM8753 Audio Codec %s", WM8753_VERSION); setup = socdev->codec_data; codec = kzalloc(sizeof(struct snd_soc_codec), GFP_KERNEL); @@ -1792,7 +1789,7 @@ static int wm8753_remove(struct platform_device *pdev) struct snd_soc_codec *codec = socdev->codec; if (codec->control_data) - wm8753_dapm_event(codec, SNDRV_CTL_POWER_D3cold); + wm8753_set_bias_level(codec, SND_SOC_BIAS_OFF); run_delayed_work(&codec->delayed_work); snd_soc_free_pcms(socdev); snd_soc_dapm_free(socdev); diff --git a/sound/soc/codecs/wm8753.h b/sound/soc/codecs/wm8753.h index 95e2a1f5316..44f5f1ff0cc 100644 --- a/sound/soc/codecs/wm8753.h +++ b/sound/soc/codecs/wm8753.h @@ -120,7 +120,7 @@ struct wm8753_setup_data { #define WM8753_DAI_HIFI 0 #define WM8753_DAI_VOICE 1 -extern struct snd_soc_codec_dai wm8753_dai[2]; +extern struct snd_soc_dai wm8753_dai[2]; extern struct snd_soc_codec_device soc_codec_dev_wm8753; #endif diff --git a/sound/soc/codecs/wm8990.c b/sound/soc/codecs/wm8990.c new file mode 100644 index 00000000000..3ecce5168e9 --- /dev/null +++ b/sound/soc/codecs/wm8990.c @@ -0,0 +1,1626 @@ +/* + * wm8990.c -- WM8990 ALSA Soc Audio driver + * + * Copyright 2008 Wolfson Microelectronics PLC. + * Author: Liam Girdwood + * lg@opensource.wolfsonmicro.com or linux@wolfsonmicro.com + * + * This program is free software; you can redistribute it and/or modify it + * under the terms of the GNU General Public License as published by the + * Free Software Foundation; either version 2 of the License, or (at your + * option) any later version. + */ + +#include <linux/module.h> +#include <linux/moduleparam.h> +#include <linux/kernel.h> +#include <linux/init.h> +#include <linux/delay.h> +#include <linux/pm.h> +#include <linux/i2c.h> +#include <linux/platform_device.h> +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> +#include <sound/soc-dapm.h> +#include <sound/initval.h> +#include <sound/tlv.h> +#include <asm/div64.h> + +#include "wm8990.h" + +#define AUDIO_NAME "wm8990" +#define WM8990_VERSION "0.2" + +/* codec private data */ +struct wm8990_priv { + unsigned int sysclk; + unsigned int pcmclk; +}; + +/* + * wm8990 register cache. Note that register 0 is not included in the + * cache. + */ +static const u16 wm8990_reg[] = { + 0x8990, /* R0 - Reset */ + 0x0000, /* R1 - Power Management (1) */ + 0x6000, /* R2 - Power Management (2) */ + 0x0000, /* R3 - Power Management (3) */ + 0x4050, /* R4 - Audio Interface (1) */ + 0x4000, /* R5 - Audio Interface (2) */ + 0x01C8, /* R6 - Clocking (1) */ + 0x0000, /* R7 - Clocking (2) */ + 0x0040, /* R8 - Audio Interface (3) */ + 0x0040, /* R9 - Audio Interface (4) */ + 0x0004, /* R10 - DAC CTRL */ + 0x00C0, /* R11 - Left DAC Digital Volume */ + 0x00C0, /* R12 - Right DAC Digital Volume */ + 0x0000, /* R13 - Digital Side Tone */ + 0x0100, /* R14 - ADC CTRL */ + 0x00C0, /* R15 - Left ADC Digital Volume */ + 0x00C0, /* R16 - Right ADC Digital Volume */ + 0x0000, /* R17 */ + 0x0000, /* R18 - GPIO CTRL 1 */ + 0x1000, /* R19 - GPIO1 & GPIO2 */ + 0x1010, /* R20 - GPIO3 & GPIO4 */ + 0x1010, /* R21 - GPIO5 & GPIO6 */ + 0x8000, /* R22 - GPIOCTRL 2 */ + 0x0800, /* R23 - GPIO_POL */ + 0x008B, /* R24 - Left Line Input 1&2 Volume */ + 0x008B, /* R25 - Left Line Input 3&4 Volume */ + 0x008B, /* R26 - Right Line Input 1&2 Volume */ + 0x008B, /* R27 - Right Line Input 3&4 Volume */ + 0x0000, /* R28 - Left Output Volume */ + 0x0000, /* R29 - Right Output Volume */ + 0x0066, /* R30 - Line Outputs Volume */ + 0x0022, /* R31 - Out3/4 Volume */ + 0x0079, /* R32 - Left OPGA Volume */ + 0x0079, /* R33 - Right OPGA Volume */ + 0x0003, /* R34 - Speaker Volume */ + 0x0003, /* R35 - ClassD1 */ + 0x0000, /* R36 */ + 0x0100, /* R37 - ClassD3 */ + 0x0000, /* R38 */ + 0x0000, /* R39 - Input Mixer1 */ + 0x0000, /* R40 - Input Mixer2 */ + 0x0000, /* R41 - Input Mixer3 */ + 0x0000, /* R42 - Input Mixer4 */ + 0x0000, /* R43 - Input Mixer5 */ + 0x0000, /* R44 - Input Mixer6 */ + 0x0000, /* R45 - Output Mixer1 */ + 0x0000, /* R46 - Output Mixer2 */ + 0x0000, /* R47 - Output Mixer3 */ + 0x0000, /* R48 - Output Mixer4 */ + 0x0000, /* R49 - Output Mixer5 */ + 0x0000, /* R50 - Output Mixer6 */ + 0x0180, /* R51 - Out3/4 Mixer */ + 0x0000, /* R52 - Line Mixer1 */ + 0x0000, /* R53 - Line Mixer2 */ + 0x0000, /* R54 - Speaker Mixer */ + 0x0000, /* R55 - Additional Control */ + 0x0000, /* R56 - AntiPOP1 */ + 0x0000, /* R57 - AntiPOP2 */ + 0x0000, /* R58 - MICBIAS */ + 0x0000, /* R59 */ + 0x0008, /* R60 - PLL1 */ + 0x0031, /* R61 - PLL2 */ + 0x0026, /* R62 - PLL3 */ +}; + +/* + * read wm8990 register cache + */ +static inline unsigned int wm8990_read_reg_cache(struct snd_soc_codec *codec, + unsigned int reg) +{ + u16 *cache = codec->reg_cache; + BUG_ON(reg > (ARRAY_SIZE(wm8990_reg)) - 1); + return cache[reg]; +} + +/* + * write wm8990 register cache + */ +static inline void wm8990_write_reg_cache(struct snd_soc_codec *codec, + unsigned int reg, unsigned int value) +{ + u16 *cache = codec->reg_cache; + BUG_ON(reg > (ARRAY_SIZE(wm8990_reg)) - 1); + + /* Reset register is uncached */ + if (reg == 0) + return; + + cache[reg] = value; +} + +/* + * write to the wm8990 register space + */ +static int wm8990_write(struct snd_soc_codec *codec, unsigned int reg, + unsigned int value) +{ + u8 data[3]; + + data[0] = reg & 0xFF; + data[1] = (value >> 8) & 0xFF; + data[2] = value & 0xFF; + + wm8990_write_reg_cache(codec, reg, value); + + if (codec->hw_write(codec->control_data, data, 3) == 2) + return 0; + else + return -EIO; +} + +#define wm8990_reset(c) wm8990_write(c, WM8990_RESET, 0) + +static const DECLARE_TLV_DB_LINEAR(rec_mix_tlv, -1500, 600); + +static const DECLARE_TLV_DB_LINEAR(in_pga_tlv, -1650, 3000); + +static const DECLARE_TLV_DB_LINEAR(out_mix_tlv, 0, -2100); + +static const DECLARE_TLV_DB_LINEAR(out_pga_tlv, -7300, 600); + +static const DECLARE_TLV_DB_LINEAR(out_omix_tlv, -600, 0); + +static const DECLARE_TLV_DB_LINEAR(out_dac_tlv, -7163, 0); + +static const DECLARE_TLV_DB_LINEAR(in_adc_tlv, -7163, 1763); + +static const DECLARE_TLV_DB_LINEAR(out_sidetone_tlv, -3600, 0); + +static int wm899x_outpga_put_volsw_vu(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); + int reg = kcontrol->private_value & 0xff; + int ret; + u16 val; + + ret = snd_soc_put_volsw(kcontrol, ucontrol); + if (ret < 0) + return ret; + + /* now hit the volume update bits (always bit 8) */ + val = wm8990_read_reg_cache(codec, reg); + return wm8990_write(codec, reg, val | 0x0100); +} + +#define SOC_WM899X_OUTPGA_SINGLE_R_TLV(xname, reg, shift, max, invert,\ + tlv_array) {\ + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = (xname), \ + .access = SNDRV_CTL_ELEM_ACCESS_TLV_READ |\ + SNDRV_CTL_ELEM_ACCESS_READWRITE,\ + .tlv.p = (tlv_array), \ + .info = snd_soc_info_volsw, \ + .get = snd_soc_get_volsw, .put = wm899x_outpga_put_volsw_vu, \ + .private_value = SOC_SINGLE_VALUE(reg, shift, max, invert) } + + +static const char *wm8990_digital_sidetone[] = + {"None", "Left ADC", "Right ADC", "Reserved"}; + +static const struct soc_enum wm8990_left_digital_sidetone_enum = +SOC_ENUM_SINGLE(WM8990_DIGITAL_SIDE_TONE, + WM8990_ADC_TO_DACL_SHIFT, + WM8990_ADC_TO_DACL_MASK, + wm8990_digital_sidetone); + +static const struct soc_enum wm8990_right_digital_sidetone_enum = +SOC_ENUM_SINGLE(WM8990_DIGITAL_SIDE_TONE, + WM8990_ADC_TO_DACR_SHIFT, + WM8990_ADC_TO_DACR_MASK, + wm8990_digital_sidetone); + +static const char *wm8990_adcmode[] = + {"Hi-fi mode", "Voice mode 1", "Voice mode 2", "Voice mode 3"}; + +static const struct soc_enum wm8990_right_adcmode_enum = +SOC_ENUM_SINGLE(WM8990_ADC_CTRL, + WM8990_ADC_HPF_CUT_SHIFT, + WM8990_ADC_HPF_CUT_MASK, + wm8990_adcmode); + +static const struct snd_kcontrol_new wm8990_snd_controls[] = { +/* INMIXL */ +SOC_SINGLE("LIN12 PGA Boost", WM8990_INPUT_MIXER3, WM8990_L12MNBST_BIT, 1, 0), +SOC_SINGLE("LIN34 PGA Boost", WM8990_INPUT_MIXER3, WM8990_L34MNBST_BIT, 1, 0), +/* INMIXR */ +SOC_SINGLE("RIN12 PGA Boost", WM8990_INPUT_MIXER3, WM8990_R12MNBST_BIT, 1, 0), +SOC_SINGLE("RIN34 PGA Boost", WM8990_INPUT_MIXER3, WM8990_R34MNBST_BIT, 1, 0), + +/* LOMIX */ +SOC_SINGLE_TLV("LOMIX LIN3 Bypass Volume", WM8990_OUTPUT_MIXER3, + WM8990_LLI3LOVOL_SHIFT, WM8990_LLI3LOVOL_MASK, 1, out_mix_tlv), +SOC_SINGLE_TLV("LOMIX RIN12 PGA Bypass Volume", WM8990_OUTPUT_MIXER3, + WM8990_LR12LOVOL_SHIFT, WM8990_LR12LOVOL_MASK, 1, out_mix_tlv), +SOC_SINGLE_TLV("LOMIX LIN12 PGA Bypass Volume", WM8990_OUTPUT_MIXER3, + WM8990_LL12LOVOL_SHIFT, WM8990_LL12LOVOL_MASK, 1, out_mix_tlv), +SOC_SINGLE_TLV("LOMIX RIN3 Bypass Volume", WM8990_OUTPUT_MIXER5, + WM8990_LRI3LOVOL_SHIFT, WM8990_LRI3LOVOL_MASK, 1, out_mix_tlv), +SOC_SINGLE_TLV("LOMIX AINRMUX Bypass Volume", WM8990_OUTPUT_MIXER5, + WM8990_LRBLOVOL_SHIFT, WM8990_LRBLOVOL_MASK, 1, out_mix_tlv), +SOC_SINGLE_TLV("LOMIX AINLMUX Bypass Volume", WM8990_OUTPUT_MIXER5, + WM8990_LRBLOVOL_SHIFT, WM8990_LRBLOVOL_MASK, 1, out_mix_tlv), + +/* ROMIX */ +SOC_SINGLE_TLV("ROMIX RIN3 Bypass Volume", WM8990_OUTPUT_MIXER4, + WM8990_RRI3ROVOL_SHIFT, WM8990_RRI3ROVOL_MASK, 1, out_mix_tlv), +SOC_SINGLE_TLV("ROMIX LIN12 PGA Bypass Volume", WM8990_OUTPUT_MIXER4, + WM8990_RL12ROVOL_SHIFT, WM8990_RL12ROVOL_MASK, 1, out_mix_tlv), +SOC_SINGLE_TLV("ROMIX RIN12 PGA Bypass Volume", WM8990_OUTPUT_MIXER4, + WM8990_RR12ROVOL_SHIFT, WM8990_RR12ROVOL_MASK, 1, out_mix_tlv), +SOC_SINGLE_TLV("ROMIX LIN3 Bypass Volume", WM8990_OUTPUT_MIXER6, + WM8990_RLI3ROVOL_SHIFT, WM8990_RLI3ROVOL_MASK, 1, out_mix_tlv), +SOC_SINGLE_TLV("ROMIX AINLMUX Bypass Volume", WM8990_OUTPUT_MIXER6, + WM8990_RLBROVOL_SHIFT, WM8990_RLBROVOL_MASK, 1, out_mix_tlv), +SOC_SINGLE_TLV("ROMIX AINRMUX Bypass Volume", WM8990_OUTPUT_MIXER6, + WM8990_RRBROVOL_SHIFT, WM8990_RRBROVOL_MASK, 1, out_mix_tlv), + +/* LOUT */ +SOC_WM899X_OUTPGA_SINGLE_R_TLV("LOUT Volume", WM8990_LEFT_OUTPUT_VOLUME, + WM8990_LOUTVOL_SHIFT, WM8990_LOUTVOL_MASK, 0, out_pga_tlv), +SOC_SINGLE("LOUT ZC", WM8990_LEFT_OUTPUT_VOLUME, WM8990_LOZC_BIT, 1, 0), + +/* ROUT */ +SOC_WM899X_OUTPGA_SINGLE_R_TLV("ROUT Volume", WM8990_RIGHT_OUTPUT_VOLUME, + WM8990_ROUTVOL_SHIFT, WM8990_ROUTVOL_MASK, 0, out_pga_tlv), +SOC_SINGLE("ROUT ZC", WM8990_RIGHT_OUTPUT_VOLUME, WM8990_ROZC_BIT, 1, 0), + +/* LOPGA */ +SOC_WM899X_OUTPGA_SINGLE_R_TLV("LOPGA Volume", WM8990_LEFT_OPGA_VOLUME, + WM8990_LOPGAVOL_SHIFT, WM8990_LOPGAVOL_MASK, 0, out_pga_tlv), +SOC_SINGLE("LOPGA ZC Switch", WM8990_LEFT_OPGA_VOLUME, + WM8990_LOPGAZC_BIT, 1, 0), + +/* ROPGA */ +SOC_WM899X_OUTPGA_SINGLE_R_TLV("ROPGA Volume", WM8990_RIGHT_OPGA_VOLUME, + WM8990_ROPGAVOL_SHIFT, WM8990_ROPGAVOL_MASK, 0, out_pga_tlv), +SOC_SINGLE("ROPGA ZC Switch", WM8990_RIGHT_OPGA_VOLUME, + WM8990_ROPGAZC_BIT, 1, 0), + +SOC_SINGLE("LON Mute Switch", WM8990_LINE_OUTPUTS_VOLUME, + WM8990_LONMUTE_BIT, 1, 0), +SOC_SINGLE("LOP Mute Switch", WM8990_LINE_OUTPUTS_VOLUME, + WM8990_LOPMUTE_BIT, 1, 0), +SOC_SINGLE("LOP Attenuation Switch", WM8990_LINE_OUTPUTS_VOLUME, + WM8990_LOATTN_BIT, 1, 0), +SOC_SINGLE("RON Mute Switch", WM8990_LINE_OUTPUTS_VOLUME, + WM8990_RONMUTE_BIT, 1, 0), +SOC_SINGLE("ROP Mute Switch", WM8990_LINE_OUTPUTS_VOLUME, + WM8990_ROPMUTE_BIT, 1, 0), +SOC_SINGLE("ROP Attenuation Switch", WM8990_LINE_OUTPUTS_VOLUME, + WM8990_ROATTN_BIT, 1, 0), + +SOC_SINGLE("OUT3 Mute Switch", WM8990_OUT3_4_VOLUME, + WM8990_OUT3MUTE_BIT, 1, 0), +SOC_SINGLE("OUT3 Attenuation Switch", WM8990_OUT3_4_VOLUME, + WM8990_OUT3ATTN_BIT, 1, 0), + +SOC_SINGLE("OUT4 Mute Switch", WM8990_OUT3_4_VOLUME, + WM8990_OUT4MUTE_BIT, 1, 0), +SOC_SINGLE("OUT4 Attenuation Switch", WM8990_OUT3_4_VOLUME, + WM8990_OUT4ATTN_BIT, 1, 0), + +SOC_SINGLE("Speaker Mode Switch", WM8990_CLASSD1, + WM8990_CDMODE_BIT, 1, 0), + +SOC_SINGLE("Speaker Output Attenuation Volume", WM8990_SPEAKER_VOLUME, + WM8990_SPKVOL_SHIFT, WM8990_SPKVOL_MASK, 0), +SOC_SINGLE("Speaker DC Boost Volume", WM8990_CLASSD3, + WM8990_DCGAIN_SHIFT, WM8990_DCGAIN_MASK, 0), +SOC_SINGLE("Speaker AC Boost Volume", WM8990_CLASSD3, + WM8990_ACGAIN_SHIFT, WM8990_ACGAIN_MASK, 0), + +SOC_WM899X_OUTPGA_SINGLE_R_TLV("Left DAC Digital Volume", + WM8990_LEFT_DAC_DIGITAL_VOLUME, + WM8990_DACL_VOL_SHIFT, + WM8990_DACL_VOL_MASK, + 0, + out_dac_tlv), + +SOC_WM899X_OUTPGA_SINGLE_R_TLV("Right DAC Digital Volume", + WM8990_RIGHT_DAC_DIGITAL_VOLUME, + WM8990_DACR_VOL_SHIFT, + WM8990_DACR_VOL_MASK, + 0, + out_dac_tlv), + +SOC_ENUM("Left Digital Sidetone", wm8990_left_digital_sidetone_enum), +SOC_ENUM("Right Digital Sidetone", wm8990_right_digital_sidetone_enum), + +SOC_SINGLE_TLV("Left Digital Sidetone Volume", WM8990_DIGITAL_SIDE_TONE, + WM8990_ADCL_DAC_SVOL_SHIFT, WM8990_ADCL_DAC_SVOL_MASK, 0, + out_sidetone_tlv), +SOC_SINGLE_TLV("Right Digital Sidetone Volume", WM8990_DIGITAL_SIDE_TONE, + WM8990_ADCR_DAC_SVOL_SHIFT, WM8990_ADCR_DAC_SVOL_MASK, 0, + out_sidetone_tlv), + +SOC_SINGLE("ADC Digital High Pass Filter Switch", WM8990_ADC_CTRL, + WM8990_ADC_HPF_ENA_BIT, 1, 0), + +SOC_ENUM("ADC HPF Mode", wm8990_right_adcmode_enum), + +SOC_WM899X_OUTPGA_SINGLE_R_TLV("Left ADC Digital Volume", + WM8990_LEFT_ADC_DIGITAL_VOLUME, + WM8990_ADCL_VOL_SHIFT, + WM8990_ADCL_VOL_MASK, + 0, + in_adc_tlv), + +SOC_WM899X_OUTPGA_SINGLE_R_TLV("Right ADC Digital Volume", + WM8990_RIGHT_ADC_DIGITAL_VOLUME, + WM8990_ADCR_VOL_SHIFT, + WM8990_ADCR_VOL_MASK, + 0, + in_adc_tlv), + +SOC_WM899X_OUTPGA_SINGLE_R_TLV("LIN12 Volume", + WM8990_LEFT_LINE_INPUT_1_2_VOLUME, + WM8990_LIN12VOL_SHIFT, + WM8990_LIN12VOL_MASK, + 0, + in_pga_tlv), + +SOC_SINGLE("LIN12 ZC Switch", WM8990_LEFT_LINE_INPUT_1_2_VOLUME, + WM8990_LI12ZC_BIT, 1, 0), + +SOC_SINGLE("LIN12 Mute Switch", WM8990_LEFT_LINE_INPUT_1_2_VOLUME, + WM8990_LI12MUTE_BIT, 1, 0), + +SOC_WM899X_OUTPGA_SINGLE_R_TLV("LIN34 Volume", + WM8990_LEFT_LINE_INPUT_3_4_VOLUME, + WM8990_LIN34VOL_SHIFT, + WM8990_LIN34VOL_MASK, + 0, + in_pga_tlv), + +SOC_SINGLE("LIN34 ZC Switch", WM8990_LEFT_LINE_INPUT_3_4_VOLUME, + WM8990_LI34ZC_BIT, 1, 0), + +SOC_SINGLE("LIN34 Mute Switch", WM8990_LEFT_LINE_INPUT_3_4_VOLUME, + WM8990_LI34MUTE_BIT, 1, 0), + +SOC_WM899X_OUTPGA_SINGLE_R_TLV("RIN12 Volume", + WM8990_RIGHT_LINE_INPUT_1_2_VOLUME, + WM8990_RIN12VOL_SHIFT, + WM8990_RIN12VOL_MASK, + 0, + in_pga_tlv), + +SOC_SINGLE("RIN12 ZC Switch", WM8990_RIGHT_LINE_INPUT_1_2_VOLUME, + WM8990_RI12ZC_BIT, 1, 0), + +SOC_SINGLE("RIN12 Mute Switch", WM8990_RIGHT_LINE_INPUT_1_2_VOLUME, + WM8990_RI12MUTE_BIT, 1, 0), + +SOC_WM899X_OUTPGA_SINGLE_R_TLV("RIN34 Volume", + WM8990_RIGHT_LINE_INPUT_3_4_VOLUME, + WM8990_RIN34VOL_SHIFT, + WM8990_RIN34VOL_MASK, + 0, + in_pga_tlv), + +SOC_SINGLE("RIN34 ZC Switch", WM8990_RIGHT_LINE_INPUT_3_4_VOLUME, + WM8990_RI34ZC_BIT, 1, 0), + +SOC_SINGLE("RIN34 Mute Switch", WM8990_RIGHT_LINE_INPUT_3_4_VOLUME, + WM8990_RI34MUTE_BIT, 1, 0), + +}; + +/* add non dapm controls */ +static int wm8990_add_controls(struct snd_soc_codec *codec) +{ + int err, i; + + for (i = 0; i < ARRAY_SIZE(wm8990_snd_controls); i++) { + err = snd_ctl_add(codec->card, + snd_soc_cnew(&wm8990_snd_controls[i], codec, + NULL)); + if (err < 0) + return err; + } + return 0; +} + +/* + * _DAPM_ Controls + */ + +static int inmixer_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + u16 reg, fakepower; + + reg = wm8990_read_reg_cache(w->codec, WM8990_POWER_MANAGEMENT_2); + fakepower = wm8990_read_reg_cache(w->codec, WM8990_INTDRIVBITS); + + if (fakepower & ((1 << WM8990_INMIXL_PWR_BIT) | + (1 << WM8990_AINLMUX_PWR_BIT))) { + reg |= WM8990_AINL_ENA; + } else { + reg &= ~WM8990_AINL_ENA; + } + + if (fakepower & ((1 << WM8990_INMIXR_PWR_BIT) | + (1 << WM8990_AINRMUX_PWR_BIT))) { + reg |= WM8990_AINR_ENA; + } else { + reg &= ~WM8990_AINL_ENA; + } + wm8990_write(w->codec, WM8990_POWER_MANAGEMENT_2, reg); + + return 0; +} + +static int outmixer_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + u32 reg_shift = kcontrol->private_value & 0xfff; + int ret = 0; + u16 reg; + + switch (reg_shift) { + case WM8990_SPEAKER_MIXER | (WM8990_LDSPK_BIT << 8) : + reg = wm8990_read_reg_cache(w->codec, WM8990_OUTPUT_MIXER1); + if (reg & WM8990_LDLO) { + printk(KERN_WARNING + "Cannot set as Output Mixer 1 LDLO Set\n"); + ret = -1; + } + break; + case WM8990_SPEAKER_MIXER | (WM8990_RDSPK_BIT << 8): + reg = wm8990_read_reg_cache(w->codec, WM8990_OUTPUT_MIXER2); + if (reg & WM8990_RDRO) { + printk(KERN_WARNING + "Cannot set as Output Mixer 2 RDRO Set\n"); + ret = -1; + } + break; + case WM8990_OUTPUT_MIXER1 | (WM8990_LDLO_BIT << 8): + reg = wm8990_read_reg_cache(w->codec, WM8990_SPEAKER_MIXER); + if (reg & WM8990_LDSPK) { + printk(KERN_WARNING + "Cannot set as Speaker Mixer LDSPK Set\n"); + ret = -1; + } + break; + case WM8990_OUTPUT_MIXER2 | (WM8990_RDRO_BIT << 8): + reg = wm8990_read_reg_cache(w->codec, WM8990_SPEAKER_MIXER); + if (reg & WM8990_RDSPK) { + printk(KERN_WARNING + "Cannot set as Speaker Mixer RDSPK Set\n"); + ret = -1; + } + break; + } + + return ret; +} + +/* INMIX dB values */ +static const unsigned int in_mix_tlv[] = { + TLV_DB_RANGE_HEAD(1), + 0, 7, TLV_DB_LINEAR_ITEM(-1200, 600), +}; + +/* Left In PGA Connections */ +static const struct snd_kcontrol_new wm8990_dapm_lin12_pga_controls[] = { +SOC_DAPM_SINGLE("LIN1 Switch", WM8990_INPUT_MIXER2, WM8990_LMN1_BIT, 1, 0), +SOC_DAPM_SINGLE("LIN2 Switch", WM8990_INPUT_MIXER2, WM8990_LMP2_BIT, 1, 0), +}; + +static const struct snd_kcontrol_new wm8990_dapm_lin34_pga_controls[] = { +SOC_DAPM_SINGLE("LIN3 Switch", WM8990_INPUT_MIXER2, WM8990_LMN3_BIT, 1, 0), +SOC_DAPM_SINGLE("LIN4 Switch", WM8990_INPUT_MIXER2, WM8990_LMP4_BIT, 1, 0), +}; + +/* Right In PGA Connections */ +static const struct snd_kcontrol_new wm8990_dapm_rin12_pga_controls[] = { +SOC_DAPM_SINGLE("RIN1 Switch", WM8990_INPUT_MIXER2, WM8990_RMN1_BIT, 1, 0), +SOC_DAPM_SINGLE("RIN2 Switch", WM8990_INPUT_MIXER2, WM8990_RMP2_BIT, 1, 0), +}; + +static const struct snd_kcontrol_new wm8990_dapm_rin34_pga_controls[] = { +SOC_DAPM_SINGLE("RIN3 Switch", WM8990_INPUT_MIXER2, WM8990_RMN3_BIT, 1, 0), +SOC_DAPM_SINGLE("RIN4 Switch", WM8990_INPUT_MIXER2, WM8990_RMP4_BIT, 1, 0), +}; + +/* INMIXL */ +static const struct snd_kcontrol_new wm8990_dapm_inmixl_controls[] = { +SOC_DAPM_SINGLE_TLV("Record Left Volume", WM8990_INPUT_MIXER3, + WM8990_LDBVOL_SHIFT, WM8990_LDBVOL_MASK, 0, in_mix_tlv), +SOC_DAPM_SINGLE_TLV("LIN2 Volume", WM8990_INPUT_MIXER5, WM8990_LI2BVOL_SHIFT, + 7, 0, in_mix_tlv), +SOC_DAPM_SINGLE("LINPGA12 Switch", WM8990_INPUT_MIXER3, WM8990_L12MNB_BIT, + 1, 0), +SOC_DAPM_SINGLE("LINPGA34 Switch", WM8990_INPUT_MIXER3, WM8990_L34MNB_BIT, + 1, 0), +}; + +/* INMIXR */ +static const struct snd_kcontrol_new wm8990_dapm_inmixr_controls[] = { +SOC_DAPM_SINGLE_TLV("Record Right Volume", WM8990_INPUT_MIXER4, + WM8990_RDBVOL_SHIFT, WM8990_RDBVOL_MASK, 0, in_mix_tlv), +SOC_DAPM_SINGLE_TLV("RIN2 Volume", WM8990_INPUT_MIXER6, WM8990_RI2BVOL_SHIFT, + 7, 0, in_mix_tlv), +SOC_DAPM_SINGLE("RINPGA12 Switch", WM8990_INPUT_MIXER3, WM8990_L12MNB_BIT, + 1, 0), +SOC_DAPM_SINGLE("RINPGA34 Switch", WM8990_INPUT_MIXER3, WM8990_L34MNB_BIT, + 1, 0), +}; + +/* AINLMUX */ +static const char *wm8990_ainlmux[] = + {"INMIXL Mix", "RXVOICE Mix", "DIFFINL Mix"}; + +static const struct soc_enum wm8990_ainlmux_enum = +SOC_ENUM_SINGLE(WM8990_INPUT_MIXER1, WM8990_AINLMODE_SHIFT, + ARRAY_SIZE(wm8990_ainlmux), wm8990_ainlmux); + +static const struct snd_kcontrol_new wm8990_dapm_ainlmux_controls = +SOC_DAPM_ENUM("Route", wm8990_ainlmux_enum); + +/* DIFFINL */ + +/* AINRMUX */ +static const char *wm8990_ainrmux[] = + {"INMIXR Mix", "RXVOICE Mix", "DIFFINR Mix"}; + +static const struct soc_enum wm8990_ainrmux_enum = +SOC_ENUM_SINGLE(WM8990_INPUT_MIXER1, WM8990_AINRMODE_SHIFT, + ARRAY_SIZE(wm8990_ainrmux), wm8990_ainrmux); + +static const struct snd_kcontrol_new wm8990_dapm_ainrmux_controls = +SOC_DAPM_ENUM("Route", wm8990_ainrmux_enum); + +/* RXVOICE */ +static const struct snd_kcontrol_new wm8990_dapm_rxvoice_controls[] = { +SOC_DAPM_SINGLE_TLV("LIN4/RXN", WM8990_INPUT_MIXER5, WM8990_LR4BVOL_SHIFT, + WM8990_LR4BVOL_MASK, 0, in_mix_tlv), +SOC_DAPM_SINGLE_TLV("RIN4/RXP", WM8990_INPUT_MIXER6, WM8990_RL4BVOL_SHIFT, + WM8990_RL4BVOL_MASK, 0, in_mix_tlv), +}; + +/* LOMIX */ +static const struct snd_kcontrol_new wm8990_dapm_lomix_controls[] = { +SOC_DAPM_SINGLE("LOMIX Right ADC Bypass Switch", WM8990_OUTPUT_MIXER1, + WM8990_LRBLO_BIT, 1, 0), +SOC_DAPM_SINGLE("LOMIX Left ADC Bypass Switch", WM8990_OUTPUT_MIXER1, + WM8990_LLBLO_BIT, 1, 0), +SOC_DAPM_SINGLE("LOMIX RIN3 Bypass Switch", WM8990_OUTPUT_MIXER1, + WM8990_LRI3LO_BIT, 1, 0), +SOC_DAPM_SINGLE("LOMIX LIN3 Bypass Switch", WM8990_OUTPUT_MIXER1, + WM8990_LLI3LO_BIT, 1, 0), +SOC_DAPM_SINGLE("LOMIX RIN12 PGA Bypass Switch", WM8990_OUTPUT_MIXER1, + WM8990_LR12LO_BIT, 1, 0), +SOC_DAPM_SINGLE("LOMIX LIN12 PGA Bypass Switch", WM8990_OUTPUT_MIXER1, + WM8990_LL12LO_BIT, 1, 0), +SOC_DAPM_SINGLE("LOMIX Left DAC Switch", WM8990_OUTPUT_MIXER1, + WM8990_LDLO_BIT, 1, 0), +}; + +/* ROMIX */ +static const struct snd_kcontrol_new wm8990_dapm_romix_controls[] = { +SOC_DAPM_SINGLE("ROMIX Left ADC Bypass Switch", WM8990_OUTPUT_MIXER2, + WM8990_RLBRO_BIT, 1, 0), +SOC_DAPM_SINGLE("ROMIX Right ADC Bypass Switch", WM8990_OUTPUT_MIXER2, + WM8990_RRBRO_BIT, 1, 0), +SOC_DAPM_SINGLE("ROMIX LIN3 Bypass Switch", WM8990_OUTPUT_MIXER2, + WM8990_RLI3RO_BIT, 1, 0), +SOC_DAPM_SINGLE("ROMIX RIN3 Bypass Switch", WM8990_OUTPUT_MIXER2, + WM8990_RRI3RO_BIT, 1, 0), +SOC_DAPM_SINGLE("ROMIX LIN12 PGA Bypass Switch", WM8990_OUTPUT_MIXER2, + WM8990_RL12RO_BIT, 1, 0), +SOC_DAPM_SINGLE("ROMIX RIN12 PGA Bypass Switch", WM8990_OUTPUT_MIXER2, + WM8990_RR12RO_BIT, 1, 0), +SOC_DAPM_SINGLE("ROMIX Right DAC Switch", WM8990_OUTPUT_MIXER2, + WM8990_RDRO_BIT, 1, 0), +}; + +/* LONMIX */ +static const struct snd_kcontrol_new wm8990_dapm_lonmix_controls[] = { +SOC_DAPM_SINGLE("LONMIX Left Mixer PGA Switch", WM8990_LINE_MIXER1, + WM8990_LLOPGALON_BIT, 1, 0), +SOC_DAPM_SINGLE("LONMIX Right Mixer PGA Switch", WM8990_LINE_MIXER1, + WM8990_LROPGALON_BIT, 1, 0), +SOC_DAPM_SINGLE("LONMIX Inverted LOP Switch", WM8990_LINE_MIXER1, + WM8990_LOPLON_BIT, 1, 0), +}; + +/* LOPMIX */ +static const struct snd_kcontrol_new wm8990_dapm_lopmix_controls[] = { +SOC_DAPM_SINGLE("LOPMIX Right Mic Bypass Switch", WM8990_LINE_MIXER1, + WM8990_LR12LOP_BIT, 1, 0), +SOC_DAPM_SINGLE("LOPMIX Left Mic Bypass Switch", WM8990_LINE_MIXER1, + WM8990_LL12LOP_BIT, 1, 0), +SOC_DAPM_SINGLE("LOPMIX Left Mixer PGA Switch", WM8990_LINE_MIXER1, + WM8990_LLOPGALOP_BIT, 1, 0), +}; + +/* RONMIX */ +static const struct snd_kcontrol_new wm8990_dapm_ronmix_controls[] = { +SOC_DAPM_SINGLE("RONMIX Right Mixer PGA Switch", WM8990_LINE_MIXER2, + WM8990_RROPGARON_BIT, 1, 0), +SOC_DAPM_SINGLE("RONMIX Left Mixer PGA Switch", WM8990_LINE_MIXER2, + WM8990_RLOPGARON_BIT, 1, 0), +SOC_DAPM_SINGLE("RONMIX Inverted ROP Switch", WM8990_LINE_MIXER2, + WM8990_ROPRON_BIT, 1, 0), +}; + +/* ROPMIX */ +static const struct snd_kcontrol_new wm8990_dapm_ropmix_controls[] = { +SOC_DAPM_SINGLE("ROPMIX Left Mic Bypass Switch", WM8990_LINE_MIXER2, + WM8990_RL12ROP_BIT, 1, 0), +SOC_DAPM_SINGLE("ROPMIX Right Mic Bypass Switch", WM8990_LINE_MIXER2, + WM8990_RR12ROP_BIT, 1, 0), +SOC_DAPM_SINGLE("ROPMIX Right Mixer PGA Switch", WM8990_LINE_MIXER2, + WM8990_RROPGAROP_BIT, 1, 0), +}; + +/* OUT3MIX */ +static const struct snd_kcontrol_new wm8990_dapm_out3mix_controls[] = { +SOC_DAPM_SINGLE("OUT3MIX LIN4/RXP Bypass Switch", WM8990_OUT3_4_MIXER, + WM8990_LI4O3_BIT, 1, 0), +SOC_DAPM_SINGLE("OUT3MIX Left Out PGA Switch", WM8990_OUT3_4_MIXER, + WM8990_LPGAO3_BIT, 1, 0), +}; + +/* OUT4MIX */ +static const struct snd_kcontrol_new wm8990_dapm_out4mix_controls[] = { +SOC_DAPM_SINGLE("OUT4MIX Right Out PGA Switch", WM8990_OUT3_4_MIXER, + WM8990_RPGAO4_BIT, 1, 0), +SOC_DAPM_SINGLE("OUT4MIX RIN4/RXP Bypass Switch", WM8990_OUT3_4_MIXER, + WM8990_RI4O4_BIT, 1, 0), +}; + +/* SPKMIX */ +static const struct snd_kcontrol_new wm8990_dapm_spkmix_controls[] = { +SOC_DAPM_SINGLE("SPKMIX LIN2 Bypass Switch", WM8990_SPEAKER_MIXER, + WM8990_LI2SPK_BIT, 1, 0), +SOC_DAPM_SINGLE("SPKMIX LADC Bypass Switch", WM8990_SPEAKER_MIXER, + WM8990_LB2SPK_BIT, 1, 0), +SOC_DAPM_SINGLE("SPKMIX Left Mixer PGA Switch", WM8990_SPEAKER_MIXER, + WM8990_LOPGASPK_BIT, 1, 0), +SOC_DAPM_SINGLE("SPKMIX Left DAC Switch", WM8990_SPEAKER_MIXER, + WM8990_LDSPK_BIT, 1, 0), +SOC_DAPM_SINGLE("SPKMIX Right DAC Switch", WM8990_SPEAKER_MIXER, + WM8990_RDSPK_BIT, 1, 0), +SOC_DAPM_SINGLE("SPKMIX Right Mixer PGA Switch", WM8990_SPEAKER_MIXER, + WM8990_ROPGASPK_BIT, 1, 0), +SOC_DAPM_SINGLE("SPKMIX RADC Bypass Switch", WM8990_SPEAKER_MIXER, + WM8990_RL12ROP_BIT, 1, 0), +SOC_DAPM_SINGLE("SPKMIX RIN2 Bypass Switch", WM8990_SPEAKER_MIXER, + WM8990_RI2SPK_BIT, 1, 0), +}; + +static const struct snd_soc_dapm_widget wm8990_dapm_widgets[] = { +/* Input Side */ +/* Input Lines */ +SND_SOC_DAPM_INPUT("LIN1"), +SND_SOC_DAPM_INPUT("LIN2"), +SND_SOC_DAPM_INPUT("LIN3"), +SND_SOC_DAPM_INPUT("LIN4/RXN"), +SND_SOC_DAPM_INPUT("RIN3"), +SND_SOC_DAPM_INPUT("RIN4/RXP"), +SND_SOC_DAPM_INPUT("RIN1"), +SND_SOC_DAPM_INPUT("RIN2"), +SND_SOC_DAPM_INPUT("Internal ADC Source"), + +/* DACs */ +SND_SOC_DAPM_ADC("Left ADC", "Left Capture", WM8990_POWER_MANAGEMENT_2, + WM8990_ADCL_ENA_BIT, 0), +SND_SOC_DAPM_ADC("Right ADC", "Right Capture", WM8990_POWER_MANAGEMENT_2, + WM8990_ADCR_ENA_BIT, 0), + +/* Input PGAs */ +SND_SOC_DAPM_MIXER("LIN12 PGA", WM8990_POWER_MANAGEMENT_2, WM8990_LIN12_ENA_BIT, + 0, &wm8990_dapm_lin12_pga_controls[0], + ARRAY_SIZE(wm8990_dapm_lin12_pga_controls)), +SND_SOC_DAPM_MIXER("LIN34 PGA", WM8990_POWER_MANAGEMENT_2, WM8990_LIN34_ENA_BIT, + 0, &wm8990_dapm_lin34_pga_controls[0], + ARRAY_SIZE(wm8990_dapm_lin34_pga_controls)), +SND_SOC_DAPM_MIXER("RIN12 PGA", WM8990_POWER_MANAGEMENT_2, WM8990_RIN12_ENA_BIT, + 0, &wm8990_dapm_rin12_pga_controls[0], + ARRAY_SIZE(wm8990_dapm_rin12_pga_controls)), +SND_SOC_DAPM_MIXER("RIN34 PGA", WM8990_POWER_MANAGEMENT_2, WM8990_RIN34_ENA_BIT, + 0, &wm8990_dapm_rin34_pga_controls[0], + ARRAY_SIZE(wm8990_dapm_rin34_pga_controls)), + +/* INMIXL */ +SND_SOC_DAPM_MIXER_E("INMIXL", WM8990_INTDRIVBITS, WM8990_INMIXL_PWR_BIT, 0, + &wm8990_dapm_inmixl_controls[0], + ARRAY_SIZE(wm8990_dapm_inmixl_controls), + inmixer_event, SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_POST_PMD), + +/* AINLMUX */ +SND_SOC_DAPM_MUX_E("AILNMUX", WM8990_INTDRIVBITS, WM8990_AINLMUX_PWR_BIT, 0, + &wm8990_dapm_ainlmux_controls, inmixer_event, + SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_POST_PMD), + +/* INMIXR */ +SND_SOC_DAPM_MIXER_E("INMIXR", WM8990_INTDRIVBITS, WM8990_INMIXR_PWR_BIT, 0, + &wm8990_dapm_inmixr_controls[0], + ARRAY_SIZE(wm8990_dapm_inmixr_controls), + inmixer_event, SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_POST_PMD), + +/* AINRMUX */ +SND_SOC_DAPM_MUX_E("AIRNMUX", WM8990_INTDRIVBITS, WM8990_AINRMUX_PWR_BIT, 0, + &wm8990_dapm_ainrmux_controls, inmixer_event, + SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_POST_PMD), + +/* Output Side */ +/* DACs */ +SND_SOC_DAPM_DAC("Left DAC", "Left Playback", WM8990_POWER_MANAGEMENT_3, + WM8990_DACL_ENA_BIT, 0), +SND_SOC_DAPM_DAC("Right DAC", "Right Playback", WM8990_POWER_MANAGEMENT_3, + WM8990_DACR_ENA_BIT, 0), + +/* LOMIX */ +SND_SOC_DAPM_MIXER_E("LOMIX", WM8990_POWER_MANAGEMENT_3, WM8990_LOMIX_ENA_BIT, + 0, &wm8990_dapm_lomix_controls[0], + ARRAY_SIZE(wm8990_dapm_lomix_controls), + outmixer_event, SND_SOC_DAPM_PRE_REG), + +/* LONMIX */ +SND_SOC_DAPM_MIXER("LONMIX", WM8990_POWER_MANAGEMENT_3, WM8990_LON_ENA_BIT, 0, + &wm8990_dapm_lonmix_controls[0], + ARRAY_SIZE(wm8990_dapm_lonmix_controls)), + +/* LOPMIX */ +SND_SOC_DAPM_MIXER("LOPMIX", WM8990_POWER_MANAGEMENT_3, WM8990_LOP_ENA_BIT, 0, + &wm8990_dapm_lopmix_controls[0], + ARRAY_SIZE(wm8990_dapm_lopmix_controls)), + +/* OUT3MIX */ +SND_SOC_DAPM_MIXER("OUT3MIX", WM8990_POWER_MANAGEMENT_1, WM8990_OUT3_ENA_BIT, 0, + &wm8990_dapm_out3mix_controls[0], + ARRAY_SIZE(wm8990_dapm_out3mix_controls)), + +/* SPKMIX */ +SND_SOC_DAPM_MIXER_E("SPKMIX", WM8990_POWER_MANAGEMENT_1, WM8990_SPK_ENA_BIT, 0, + &wm8990_dapm_spkmix_controls[0], + ARRAY_SIZE(wm8990_dapm_spkmix_controls), outmixer_event, + SND_SOC_DAPM_PRE_REG), + +/* OUT4MIX */ +SND_SOC_DAPM_MIXER("OUT4MIX", WM8990_POWER_MANAGEMENT_1, WM8990_OUT4_ENA_BIT, 0, + &wm8990_dapm_out4mix_controls[0], + ARRAY_SIZE(wm8990_dapm_out4mix_controls)), + +/* ROPMIX */ +SND_SOC_DAPM_MIXER("ROPMIX", WM8990_POWER_MANAGEMENT_3, WM8990_ROP_ENA_BIT, 0, + &wm8990_dapm_ropmix_controls[0], + ARRAY_SIZE(wm8990_dapm_ropmix_controls)), + +/* RONMIX */ +SND_SOC_DAPM_MIXER("RONMIX", WM8990_POWER_MANAGEMENT_3, WM8990_RON_ENA_BIT, 0, + &wm8990_dapm_ronmix_controls[0], + ARRAY_SIZE(wm8990_dapm_ronmix_controls)), + +/* ROMIX */ +SND_SOC_DAPM_MIXER_E("ROMIX", WM8990_POWER_MANAGEMENT_3, WM8990_ROMIX_ENA_BIT, + 0, &wm8990_dapm_romix_controls[0], + ARRAY_SIZE(wm8990_dapm_romix_controls), + outmixer_event, SND_SOC_DAPM_PRE_REG), + +/* LOUT PGA */ +SND_SOC_DAPM_PGA("LOUT PGA", WM8990_POWER_MANAGEMENT_1, WM8990_LOUT_ENA_BIT, 0, + NULL, 0), + +/* ROUT PGA */ +SND_SOC_DAPM_PGA("ROUT PGA", WM8990_POWER_MANAGEMENT_1, WM8990_ROUT_ENA_BIT, 0, + NULL, 0), + +/* LOPGA */ +SND_SOC_DAPM_PGA("LOPGA", WM8990_POWER_MANAGEMENT_3, WM8990_LOPGA_ENA_BIT, 0, + NULL, 0), + +/* ROPGA */ +SND_SOC_DAPM_PGA("ROPGA", WM8990_POWER_MANAGEMENT_3, WM8990_ROPGA_ENA_BIT, 0, + NULL, 0), + +/* MICBIAS */ +SND_SOC_DAPM_MICBIAS("MICBIAS", WM8990_POWER_MANAGEMENT_1, + WM8990_MICBIAS_ENA_BIT, 0), + +SND_SOC_DAPM_OUTPUT("LON"), +SND_SOC_DAPM_OUTPUT("LOP"), +SND_SOC_DAPM_OUTPUT("OUT3"), +SND_SOC_DAPM_OUTPUT("LOUT"), +SND_SOC_DAPM_OUTPUT("SPKN"), +SND_SOC_DAPM_OUTPUT("SPKP"), +SND_SOC_DAPM_OUTPUT("ROUT"), +SND_SOC_DAPM_OUTPUT("OUT4"), +SND_SOC_DAPM_OUTPUT("ROP"), +SND_SOC_DAPM_OUTPUT("RON"), + +SND_SOC_DAPM_OUTPUT("Internal DAC Sink"), +}; + +static const struct snd_soc_dapm_route audio_map[] = { + /* Make DACs turn on when playing even if not mixed into any outputs */ + {"Internal DAC Sink", NULL, "Left DAC"}, + {"Internal DAC Sink", NULL, "Right DAC"}, + + /* Make ADCs turn on when recording even if not mixed from any inputs */ + {"Left ADC", NULL, "Internal ADC Source"}, + {"Right ADC", NULL, "Internal ADC Source"}, + + /* Input Side */ + /* LIN12 PGA */ + {"LIN12 PGA", "LIN1 Switch", "LIN1"}, + {"LIN12 PGA", "LIN2 Switch", "LIN2"}, + /* LIN34 PGA */ + {"LIN34 PGA", "LIN3 Switch", "LIN3"}, + {"LIN34 PGA", "LIN4 Switch", "LIN4"}, + /* INMIXL */ + {"INMIXL", "Record Left Volume", "LOMIX"}, + {"INMIXL", "LIN2 Volume", "LIN2"}, + {"INMIXL", "LINPGA12 Switch", "LIN12 PGA"}, + {"INMIXL", "LINPGA34 Switch", "LIN34 PGA"}, + /* AILNMUX */ + {"AILNMUX", "INMIXL Mix", "INMIXL"}, + {"AILNMUX", "DIFFINL Mix", "LIN12PGA"}, + {"AILNMUX", "DIFFINL Mix", "LIN34PGA"}, + {"AILNMUX", "RXVOICE Mix", "LIN4/RXN"}, + {"AILNMUX", "RXVOICE Mix", "RIN4/RXP"}, + /* ADC */ + {"Left ADC", NULL, "AILNMUX"}, + + /* RIN12 PGA */ + {"RIN12 PGA", "RIN1 Switch", "RIN1"}, + {"RIN12 PGA", "RIN2 Switch", "RIN2"}, + /* RIN34 PGA */ + {"RIN34 PGA", "RIN3 Switch", "RIN3"}, + {"RIN34 PGA", "RIN4 Switch", "RIN4"}, + /* INMIXL */ + {"INMIXR", "Record Right Volume", "ROMIX"}, + {"INMIXR", "RIN2 Volume", "RIN2"}, + {"INMIXR", "RINPGA12 Switch", "RIN12 PGA"}, + {"INMIXR", "RINPGA34 Switch", "RIN34 PGA"}, + /* AIRNMUX */ + {"AIRNMUX", "INMIXR Mix", "INMIXR"}, + {"AIRNMUX", "DIFFINR Mix", "RIN12PGA"}, + {"AIRNMUX", "DIFFINR Mix", "RIN34PGA"}, + {"AIRNMUX", "RXVOICE Mix", "RIN4/RXN"}, + {"AIRNMUX", "RXVOICE Mix", "RIN4/RXP"}, + /* ADC */ + {"Right ADC", NULL, "AIRNMUX"}, + + /* LOMIX */ + {"LOMIX", "LOMIX RIN3 Bypass Switch", "RIN3"}, + {"LOMIX", "LOMIX LIN3 Bypass Switch", "LIN3"}, + {"LOMIX", "LOMIX LIN12 PGA Bypass Switch", "LIN12 PGA"}, + {"LOMIX", "LOMIX RIN12 PGA Bypass Switch", "RIN12 PGA"}, + {"LOMIX", "LOMIX Right ADC Bypass Switch", "AINRMUX"}, + {"LOMIX", "LOMIX Left ADC Bypass Switch", "AINLMUX"}, + {"LOMIX", "LOMIX Left DAC Switch", "Left DAC"}, + + /* ROMIX */ + {"ROMIX", "ROMIX RIN3 Bypass Switch", "RIN3"}, + {"ROMIX", "ROMIX LIN3 Bypass Switch", "LIN3"}, + {"ROMIX", "ROMIX LIN12 PGA Bypass Switch", "LIN12 PGA"}, + {"ROMIX", "ROMIX RIN12 PGA Bypass Switch", "RIN12 PGA"}, + {"ROMIX", "ROMIX Right ADC Bypass Switch", "AINRMUX"}, + {"ROMIX", "ROMIX Left ADC Bypass Switch", "AINLMUX"}, + {"ROMIX", "ROMIX Right DAC Switch", "Right DAC"}, + + /* SPKMIX */ + {"SPKMIX", "SPKMIX LIN2 Bypass Switch", "LIN2"}, + {"SPKMIX", "SPKMIX RIN2 Bypass Switch", "RIN2"}, + {"SPKMIX", "SPKMIX LADC Bypass Switch", "AINLMUX"}, + {"SPKMIX", "SPKMIX RADC Bypass Switch", "AINRMUX"}, + {"SPKMIX", "SPKMIX Left Mixer PGA Switch", "LOPGA"}, + {"SPKMIX", "SPKMIX Right Mixer PGA Switch", "ROPGA"}, + {"SPKMIX", "SPKMIX Right DAC Switch", "Right DAC"}, + {"SPKMIX", "SPKMIX Left DAC Switch", "Right DAC"}, + + /* LONMIX */ + {"LONMIX", "LONMIX Left Mixer PGA Switch", "LOPGA"}, + {"LONMIX", "LONMIX Right Mixer PGA Switch", "ROPGA"}, + {"LONMIX", "LONMIX Inverted LOP Switch", "LOPMIX"}, + + /* LOPMIX */ + {"LOPMIX", "LOPMIX Right Mic Bypass Switch", "RIN12 PGA"}, + {"LOPMIX", "LOPMIX Left Mic Bypass Switch", "LIN12 PGA"}, + {"LOPMIX", "LOPMIX Left Mixer PGA Switch", "LOPGA"}, + + /* OUT3MIX */ + {"OUT3MIX", "OUT3MIX LIN4/RXP Bypass Switch", "LIN4/RXP"}, + {"OUT3MIX", "OUT3MIX Left Out PGA Switch", "LOPGA"}, + + /* OUT4MIX */ + {"OUT4MIX", "OUT4MIX Right Out PGA Switch", "ROPGA"}, + {"OUT4MIX", "OUT4MIX RIN4/RXP Bypass Switch", "RIN4/RXP"}, + + /* RONMIX */ + {"RONMIX", "RONMIX Right Mixer PGA Switch", "ROPGA"}, + {"RONMIX", "RONMIX Left Mixer PGA Switch", "LOPGA"}, + {"RONMIX", "RONMIX Inverted ROP Switch", "ROPMIX"}, + + /* ROPMIX */ + {"ROPMIX", "ROPMIX Left Mic Bypass Switch", "LIN12 PGA"}, + {"ROPMIX", "ROPMIX Right Mic Bypass Switch", "RIN12 PGA"}, + {"ROPMIX", "ROPMIX Right Mixer PGA Switch", "ROPGA"}, + + /* Out Mixer PGAs */ + {"LOPGA", NULL, "LOMIX"}, + {"ROPGA", NULL, "ROMIX"}, + + {"LOUT PGA", NULL, "LOMIX"}, + {"ROUT PGA", NULL, "ROMIX"}, + + /* Output Pins */ + {"LON", NULL, "LONMIX"}, + {"LOP", NULL, "LOPMIX"}, + {"OUT", NULL, "OUT3MIX"}, + {"LOUT", NULL, "LOUT PGA"}, + {"SPKN", NULL, "SPKMIX"}, + {"ROUT", NULL, "ROUT PGA"}, + {"OUT4", NULL, "OUT4MIX"}, + {"ROP", NULL, "ROPMIX"}, + {"RON", NULL, "RONMIX"}, +}; + +static int wm8990_add_widgets(struct snd_soc_codec *codec) +{ + snd_soc_dapm_new_controls(codec, wm8990_dapm_widgets, + ARRAY_SIZE(wm8990_dapm_widgets)); + + /* set up the WM8990 audio map */ + snd_soc_dapm_add_routes(codec, audio_map, ARRAY_SIZE(audio_map)); + + snd_soc_dapm_new_widgets(codec); + return 0; +} + +/* PLL divisors */ +struct _pll_div { + u32 div2; + u32 n; + u32 k; +}; + +/* The size in bits of the pll divide multiplied by 10 + * to allow rounding later */ +#define FIXED_PLL_SIZE ((1 << 16) * 10) + +static void pll_factors(struct _pll_div *pll_div, unsigned int target, + unsigned int source) +{ + u64 Kpart; + unsigned int K, Ndiv, Nmod; + + + Ndiv = target / source; + if (Ndiv < 6) { + source >>= 1; + pll_div->div2 = 1; + Ndiv = target / source; + } else + pll_div->div2 = 0; + + if ((Ndiv < 6) || (Ndiv > 12)) + printk(KERN_WARNING + "WM8990 N value outwith recommended range! N = %d\n", Ndiv); + + pll_div->n = Ndiv; + Nmod = target % source; + Kpart = FIXED_PLL_SIZE * (long long)Nmod; + + do_div(Kpart, source); + + K = Kpart & 0xFFFFFFFF; + + /* Check if we need to round */ + if ((K % 10) >= 5) + K += 5; + + /* Move down to proper range now rounding is done */ + K /= 10; + + pll_div->k = K; +} + +static int wm8990_set_dai_pll(struct snd_soc_dai *codec_dai, + int pll_id, unsigned int freq_in, unsigned int freq_out) +{ + u16 reg; + struct snd_soc_codec *codec = codec_dai->codec; + struct _pll_div pll_div; + + if (freq_in && freq_out) { + pll_factors(&pll_div, freq_out * 4, freq_in); + + /* Turn on PLL */ + reg = wm8990_read_reg_cache(codec, WM8990_POWER_MANAGEMENT_2); + reg |= WM8990_PLL_ENA; + wm8990_write(codec, WM8990_POWER_MANAGEMENT_2, reg); + + /* sysclk comes from PLL */ + reg = wm8990_read_reg_cache(codec, WM8990_CLOCKING_2); + wm8990_write(codec, WM8990_CLOCKING_2, reg | WM8990_SYSCLK_SRC); + + /* set up N , fractional mode and pre-divisor if neccessary */ + wm8990_write(codec, WM8990_PLL1, pll_div.n | WM8990_SDM | + (pll_div.div2?WM8990_PRESCALE:0)); + wm8990_write(codec, WM8990_PLL2, (u8)(pll_div.k>>8)); + wm8990_write(codec, WM8990_PLL3, (u8)(pll_div.k & 0xFF)); + } else { + /* Turn on PLL */ + reg = wm8990_read_reg_cache(codec, WM8990_POWER_MANAGEMENT_2); + reg &= ~WM8990_PLL_ENA; + wm8990_write(codec, WM8990_POWER_MANAGEMENT_2, reg); + } + return 0; +} + +/* + * Clock after PLL and dividers + */ +static int wm8990_set_dai_sysclk(struct snd_soc_dai *codec_dai, + int clk_id, unsigned int freq, int dir) +{ + struct snd_soc_codec *codec = codec_dai->codec; + struct wm8990_priv *wm8990 = codec->private_data; + + wm8990->sysclk = freq; + return 0; +} + +/* + * Set's ADC and Voice DAC format. + */ +static int wm8990_set_dai_fmt(struct snd_soc_dai *codec_dai, + unsigned int fmt) +{ + struct snd_soc_codec *codec = codec_dai->codec; + u16 audio1, audio3; + + audio1 = wm8990_read_reg_cache(codec, WM8990_AUDIO_INTERFACE_1); + audio3 = wm8990_read_reg_cache(codec, WM8990_AUDIO_INTERFACE_3); + + /* set master/slave audio interface */ + switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { + case SND_SOC_DAIFMT_CBS_CFS: + audio3 &= ~WM8990_AIF_MSTR1; + break; + case SND_SOC_DAIFMT_CBM_CFM: + audio3 |= WM8990_AIF_MSTR1; + break; + default: + return -EINVAL; + } + + audio1 &= ~WM8990_AIF_FMT_MASK; + + /* interface format */ + switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { + case SND_SOC_DAIFMT_I2S: + audio1 |= WM8990_AIF_TMF_I2S; + audio1 &= ~WM8990_AIF_LRCLK_INV; + break; + case SND_SOC_DAIFMT_RIGHT_J: + audio1 |= WM8990_AIF_TMF_RIGHTJ; + audio1 &= ~WM8990_AIF_LRCLK_INV; + break; + case SND_SOC_DAIFMT_LEFT_J: + audio1 |= WM8990_AIF_TMF_LEFTJ; + audio1 &= ~WM8990_AIF_LRCLK_INV; + break; + case SND_SOC_DAIFMT_DSP_A: + audio1 |= WM8990_AIF_TMF_DSP; + audio1 &= ~WM8990_AIF_LRCLK_INV; + break; + case SND_SOC_DAIFMT_DSP_B: + audio1 |= WM8990_AIF_TMF_DSP | WM8990_AIF_LRCLK_INV; + break; + default: + return -EINVAL; + } + + wm8990_write(codec, WM8990_AUDIO_INTERFACE_1, audio1); + wm8990_write(codec, WM8990_AUDIO_INTERFACE_3, audio3); + return 0; +} + +static int wm8990_set_dai_clkdiv(struct snd_soc_dai *codec_dai, + int div_id, int div) +{ + struct snd_soc_codec *codec = codec_dai->codec; + u16 reg; + + switch (div_id) { + case WM8990_MCLK_DIV: + reg = wm8990_read_reg_cache(codec, WM8990_CLOCKING_2) & + ~WM8990_MCLK_DIV_MASK; + wm8990_write(codec, WM8990_CLOCKING_2, reg | div); + break; + case WM8990_DACCLK_DIV: + reg = wm8990_read_reg_cache(codec, WM8990_CLOCKING_2) & + ~WM8990_DAC_CLKDIV_MASK; + wm8990_write(codec, WM8990_CLOCKING_2, reg | div); + break; + case WM8990_ADCCLK_DIV: + reg = wm8990_read_reg_cache(codec, WM8990_CLOCKING_2) & + ~WM8990_ADC_CLKDIV_MASK; + wm8990_write(codec, WM8990_CLOCKING_2, reg | div); + break; + case WM8990_BCLK_DIV: + reg = wm8990_read_reg_cache(codec, WM8990_CLOCKING_1) & + ~WM8990_BCLK_DIV_MASK; + wm8990_write(codec, WM8990_CLOCKING_1, reg | div); + break; + default: + return -EINVAL; + } + + return 0; +} + +/* + * Set PCM DAI bit size and sample rate. + */ +static int wm8990_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_device *socdev = rtd->socdev; + struct snd_soc_codec *codec = socdev->codec; + u16 audio1 = wm8990_read_reg_cache(codec, WM8990_AUDIO_INTERFACE_1); + + audio1 &= ~WM8990_AIF_WL_MASK; + /* bit size */ + switch (params_format(params)) { + case SNDRV_PCM_FORMAT_S16_LE: + break; + case SNDRV_PCM_FORMAT_S20_3LE: + audio1 |= WM8990_AIF_WL_20BITS; + break; + case SNDRV_PCM_FORMAT_S24_LE: + audio1 |= WM8990_AIF_WL_24BITS; + break; + case SNDRV_PCM_FORMAT_S32_LE: + audio1 |= WM8990_AIF_WL_32BITS; + break; + } + + wm8990_write(codec, WM8990_AUDIO_INTERFACE_1, audio1); + return 0; +} + +static int wm8990_mute(struct snd_soc_dai *dai, int mute) +{ + struct snd_soc_codec *codec = dai->codec; + u16 val; + + val = wm8990_read_reg_cache(codec, WM8990_DAC_CTRL) & ~WM8990_DAC_MUTE; + + if (mute) + wm8990_write(codec, WM8990_DAC_CTRL, val | WM8990_DAC_MUTE); + else + wm8990_write(codec, WM8990_DAC_CTRL, val); + + return 0; +} + +static int wm8990_set_bias_level(struct snd_soc_codec *codec, + enum snd_soc_bias_level level) +{ + u16 val; + + switch (level) { + case SND_SOC_BIAS_ON: + break; + case SND_SOC_BIAS_PREPARE: + break; + case SND_SOC_BIAS_STANDBY: + if (codec->bias_level == SND_SOC_BIAS_OFF) { + /* Enable all output discharge bits */ + wm8990_write(codec, WM8990_ANTIPOP1, WM8990_DIS_LLINE | + WM8990_DIS_RLINE | WM8990_DIS_OUT3 | + WM8990_DIS_OUT4 | WM8990_DIS_LOUT | + WM8990_DIS_ROUT); + + /* Enable POBCTRL, SOFT_ST, VMIDTOG and BUFDCOPEN */ + wm8990_write(codec, WM8990_ANTIPOP2, WM8990_SOFTST | + WM8990_BUFDCOPEN | WM8990_POBCTRL | + WM8990_VMIDTOG); + + /* Delay to allow output caps to discharge */ + msleep(msecs_to_jiffies(300)); + + /* Disable VMIDTOG */ + wm8990_write(codec, WM8990_ANTIPOP2, WM8990_SOFTST | + WM8990_BUFDCOPEN | WM8990_POBCTRL); + + /* disable all output discharge bits */ + wm8990_write(codec, WM8990_ANTIPOP1, 0); + + /* Enable outputs */ + wm8990_write(codec, WM8990_POWER_MANAGEMENT_1, 0x1b00); + + msleep(msecs_to_jiffies(50)); + + /* Enable VMID at 2x50k */ + wm8990_write(codec, WM8990_POWER_MANAGEMENT_1, 0x1f02); + + msleep(msecs_to_jiffies(100)); + + /* Enable VREF */ + wm8990_write(codec, WM8990_POWER_MANAGEMENT_1, 0x1f03); + + msleep(msecs_to_jiffies(600)); + + /* Enable BUFIOEN */ + wm8990_write(codec, WM8990_ANTIPOP2, WM8990_SOFTST | + WM8990_BUFDCOPEN | WM8990_POBCTRL | + WM8990_BUFIOEN); + + /* Disable outputs */ + wm8990_write(codec, WM8990_POWER_MANAGEMENT_1, 0x3); + + /* disable POBCTRL, SOFT_ST and BUFDCOPEN */ + wm8990_write(codec, WM8990_ANTIPOP2, WM8990_BUFIOEN); + } else { + /* ON -> standby */ + + } + break; + + case SND_SOC_BIAS_OFF: + /* Enable POBCTRL and SOFT_ST */ + wm8990_write(codec, WM8990_ANTIPOP2, WM8990_SOFTST | + WM8990_POBCTRL | WM8990_BUFIOEN); + + /* Enable POBCTRL, SOFT_ST and BUFDCOPEN */ + wm8990_write(codec, WM8990_ANTIPOP2, WM8990_SOFTST | + WM8990_BUFDCOPEN | WM8990_POBCTRL | + WM8990_BUFIOEN); + + /* mute DAC */ + val = wm8990_read_reg_cache(codec, WM8990_DAC_CTRL); + wm8990_write(codec, WM8990_DAC_CTRL, val | WM8990_DAC_MUTE); + + /* Enable any disabled outputs */ + wm8990_write(codec, WM8990_POWER_MANAGEMENT_1, 0x1f03); + + /* Disable VMID */ + wm8990_write(codec, WM8990_POWER_MANAGEMENT_1, 0x1f01); + + msleep(msecs_to_jiffies(300)); + + /* Enable all output discharge bits */ + wm8990_write(codec, WM8990_ANTIPOP1, WM8990_DIS_LLINE | + WM8990_DIS_RLINE | WM8990_DIS_OUT3 | + WM8990_DIS_OUT4 | WM8990_DIS_LOUT | + WM8990_DIS_ROUT); + + /* Disable VREF */ + wm8990_write(codec, WM8990_POWER_MANAGEMENT_1, 0x0); + + /* disable POBCTRL, SOFT_ST and BUFDCOPEN */ + wm8990_write(codec, WM8990_ANTIPOP2, 0x0); + break; + } + + codec->bias_level = level; + return 0; +} + +#define WM8990_RATES (SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_11025 |\ + SNDRV_PCM_RATE_16000 | SNDRV_PCM_RATE_22050 | SNDRV_PCM_RATE_44100 | \ + SNDRV_PCM_RATE_48000) + +#define WM8990_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE |\ + SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_S32_LE) + +/* + * The WM8990 supports 2 different and mutually exclusive DAI + * configurations. + * + * 1. ADC/DAC on Primary Interface + * 2. ADC on Primary Interface/DAC on secondary + */ +struct snd_soc_dai wm8990_dai = { +/* ADC/DAC on primary */ + .name = "WM8990 ADC/DAC Primary", + .id = 1, + .playback = { + .stream_name = "Playback", + .channels_min = 1, + .channels_max = 2, + .rates = WM8990_RATES, + .formats = WM8990_FORMATS,}, + .capture = { + .stream_name = "Capture", + .channels_min = 1, + .channels_max = 2, + .rates = WM8990_RATES, + .formats = WM8990_FORMATS,}, + .ops = { + .hw_params = wm8990_hw_params,}, + .dai_ops = { + .digital_mute = wm8990_mute, + .set_fmt = wm8990_set_dai_fmt, + .set_clkdiv = wm8990_set_dai_clkdiv, + .set_pll = wm8990_set_dai_pll, + .set_sysclk = wm8990_set_dai_sysclk, + }, +}; +EXPORT_SYMBOL_GPL(wm8990_dai); + +static int wm8990_suspend(struct platform_device *pdev, pm_message_t state) +{ + struct snd_soc_device *socdev = platform_get_drvdata(pdev); + struct snd_soc_codec *codec = socdev->codec; + + /* we only need to suspend if we are a valid card */ + if (!codec->card) + return 0; + + wm8990_set_bias_level(codec, SND_SOC_BIAS_OFF); + return 0; +} + +static int wm8990_resume(struct platform_device *pdev) +{ + struct snd_soc_device *socdev = platform_get_drvdata(pdev); + struct snd_soc_codec *codec = socdev->codec; + int i; + u8 data[2]; + u16 *cache = codec->reg_cache; + + /* we only need to resume if we are a valid card */ + if (!codec->card) + return 0; + + /* Sync reg_cache with the hardware */ + for (i = 0; i < ARRAY_SIZE(wm8990_reg); i++) { + if (i + 1 == WM8990_RESET) + continue; + data[0] = ((i + 1) << 1) | ((cache[i] >> 8) & 0x0001); + data[1] = cache[i] & 0x00ff; + codec->hw_write(codec->control_data, data, 2); + } + + wm8990_set_bias_level(codec, SND_SOC_BIAS_STANDBY); + return 0; +} + +/* + * initialise the WM8990 driver + * register the mixer and dsp interfaces with the kernel + */ +static int wm8990_init(struct snd_soc_device *socdev) +{ + struct snd_soc_codec *codec = socdev->codec; + u16 reg; + int ret = 0; + + codec->name = "WM8990"; + codec->owner = THIS_MODULE; + codec->read = wm8990_read_reg_cache; + codec->write = wm8990_write; + codec->set_bias_level = wm8990_set_bias_level; + codec->dai = &wm8990_dai; + codec->num_dai = 2; + codec->reg_cache_size = ARRAY_SIZE(wm8990_reg); + codec->reg_cache = kmemdup(wm8990_reg, sizeof(wm8990_reg), GFP_KERNEL); + + if (codec->reg_cache == NULL) + return -ENOMEM; + + wm8990_reset(codec); + + /* register pcms */ + ret = snd_soc_new_pcms(socdev, SNDRV_DEFAULT_IDX1, SNDRV_DEFAULT_STR1); + if (ret < 0) { + printk(KERN_ERR "wm8990: failed to create pcms\n"); + goto pcm_err; + } + + /* charge output caps */ + codec->bias_level = SND_SOC_BIAS_OFF; + wm8990_set_bias_level(codec, SND_SOC_BIAS_STANDBY); + + reg = wm8990_read_reg_cache(codec, WM8990_AUDIO_INTERFACE_4); + wm8990_write(codec, WM8990_AUDIO_INTERFACE_4, reg | WM8990_ALRCGPIO1); + + reg = wm8990_read_reg_cache(codec, WM8990_GPIO1_GPIO2) & + ~WM8990_GPIO1_SEL_MASK; + wm8990_write(codec, WM8990_GPIO1_GPIO2, reg | 1); + + reg = wm8990_read_reg_cache(codec, WM8990_POWER_MANAGEMENT_2); + wm8990_write(codec, WM8990_POWER_MANAGEMENT_2, reg | WM8990_OPCLK_ENA); + + wm8990_write(codec, WM8990_LEFT_OUTPUT_VOLUME, 0x50 | (1<<8)); + wm8990_write(codec, WM8990_RIGHT_OUTPUT_VOLUME, 0x50 | (1<<8)); + + wm8990_add_controls(codec); + wm8990_add_widgets(codec); + ret = snd_soc_register_card(socdev); + if (ret < 0) { + printk(KERN_ERR "wm8990: failed to register card\n"); + goto card_err; + } + return ret; + +card_err: + snd_soc_free_pcms(socdev); + snd_soc_dapm_free(socdev); +pcm_err: + kfree(codec->reg_cache); + return ret; +} + +/* If the i2c layer weren't so broken, we could pass this kind of data + around */ +static struct snd_soc_device *wm8990_socdev; + +#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE) + +/* + * WM891 2 wire address is determined by GPIO5 + * state during powerup. + * low = 0x34 + * high = 0x36 + */ +static unsigned short normal_i2c[] = { 0, I2C_CLIENT_END }; + +/* Magic definition of all other variables and things */ +I2C_CLIENT_INSMOD; + +static struct i2c_driver wm8990_i2c_driver; +static struct i2c_client client_template; + +static int wm8990_codec_probe(struct i2c_adapter *adap, int addr, int kind) +{ + struct snd_soc_device *socdev = wm8990_socdev; + struct wm8990_setup_data *setup = socdev->codec_data; + struct snd_soc_codec *codec = socdev->codec; + struct i2c_client *i2c; + int ret; + + if (addr != setup->i2c_address) + return -ENODEV; + + client_template.adapter = adap; + client_template.addr = addr; + + i2c = kmemdup(&client_template, sizeof(client_template), GFP_KERNEL); + if (i2c == NULL) { + kfree(codec); + return -ENOMEM; + } + i2c_set_clientdata(i2c, codec); + codec->control_data = i2c; + + ret = i2c_attach_client(i2c); + if (ret < 0) { + pr_err("failed to attach codec at addr %x\n", addr); + goto err; + } + + ret = wm8990_init(socdev); + if (ret < 0) { + pr_err("failed to initialise WM8990\n"); + goto err; + } + return ret; + +err: + kfree(codec); + kfree(i2c); + return ret; +} + +static int wm8990_i2c_detach(struct i2c_client *client) +{ + struct snd_soc_codec *codec = i2c_get_clientdata(client); + i2c_detach_client(client); + kfree(codec->reg_cache); + kfree(client); + return 0; +} + +static int wm8990_i2c_attach(struct i2c_adapter *adap) +{ + return i2c_probe(adap, &addr_data, wm8990_codec_probe); +} + +static struct i2c_driver wm8990_i2c_driver = { + .driver = { + .name = "WM8990 I2C Codec", + .owner = THIS_MODULE, + }, + .attach_adapter = wm8990_i2c_attach, + .detach_client = wm8990_i2c_detach, + .command = NULL, +}; + +static struct i2c_client client_template = { + .name = "WM8990", + .driver = &wm8990_i2c_driver, +}; +#endif + +static int wm8990_probe(struct platform_device *pdev) +{ + struct snd_soc_device *socdev = platform_get_drvdata(pdev); + struct wm8990_setup_data *setup; + struct snd_soc_codec *codec; + struct wm8990_priv *wm8990; + int ret = 0; + + pr_info("WM8990 Audio Codec %s\n", WM8990_VERSION); + + setup = socdev->codec_data; + codec = kzalloc(sizeof(struct snd_soc_codec), GFP_KERNEL); + if (codec == NULL) + return -ENOMEM; + + wm8990 = kzalloc(sizeof(struct wm8990_priv), GFP_KERNEL); + if (wm8990 == NULL) { + kfree(codec); + return -ENOMEM; + } + + codec->private_data = wm8990; + socdev->codec = codec; + mutex_init(&codec->mutex); + INIT_LIST_HEAD(&codec->dapm_widgets); + INIT_LIST_HEAD(&codec->dapm_paths); + wm8990_socdev = socdev; + +#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE) + if (setup->i2c_address) { + normal_i2c[0] = setup->i2c_address; + codec->hw_write = (hw_write_t)i2c_master_send; + ret = i2c_add_driver(&wm8990_i2c_driver); + if (ret != 0) + printk(KERN_ERR "can't add i2c driver"); + } +#else + /* Add other interfaces here */ +#endif + return ret; +} + +/* power down chip */ +static int wm8990_remove(struct platform_device *pdev) +{ + struct snd_soc_device *socdev = platform_get_drvdata(pdev); + struct snd_soc_codec *codec = socdev->codec; + + if (codec->control_data) + wm8990_set_bias_level(codec, SND_SOC_BIAS_OFF); + snd_soc_free_pcms(socdev); + snd_soc_dapm_free(socdev); +#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE) + i2c_del_driver(&wm8990_i2c_driver); +#endif + kfree(codec->private_data); + kfree(codec); + + return 0; +} + +struct snd_soc_codec_device soc_codec_dev_wm8990 = { + .probe = wm8990_probe, + .remove = wm8990_remove, + .suspend = wm8990_suspend, + .resume = wm8990_resume, +}; +EXPORT_SYMBOL_GPL(soc_codec_dev_wm8990); + +MODULE_DESCRIPTION("ASoC WM8990 driver"); +MODULE_AUTHOR("Liam Girdwood"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/codecs/wm8990.h b/sound/soc/codecs/wm8990.h new file mode 100644 index 00000000000..6bea5748528 --- /dev/null +++ b/sound/soc/codecs/wm8990.h @@ -0,0 +1,832 @@ +/* + * wm8990.h -- audio driver for WM8990 + * + * Copyright 2007 Wolfson Microelectronics PLC. + * Author: Graeme Gregory + * graeme.gregory@wolfsonmicro.com or linux@wolfsonmicro.com + * + * This program is free software; you can redistribute it and/or modify it + * under the terms of the GNU General Public License as published by the + * Free Software Foundation; either version 2 of the License, or (at your + * option) any later version. + * + */ + +#ifndef __WM8990REGISTERDEFS_H__ +#define __WM8990REGISTERDEFS_H__ + +/* + * Register values. + */ +#define WM8990_RESET 0x00 +#define WM8990_POWER_MANAGEMENT_1 0x01 +#define WM8990_POWER_MANAGEMENT_2 0x02 +#define WM8990_POWER_MANAGEMENT_3 0x03 +#define WM8990_AUDIO_INTERFACE_1 0x04 +#define WM8990_AUDIO_INTERFACE_2 0x05 +#define WM8990_CLOCKING_1 0x06 +#define WM8990_CLOCKING_2 0x07 +#define WM8990_AUDIO_INTERFACE_3 0x08 +#define WM8990_AUDIO_INTERFACE_4 0x09 +#define WM8990_DAC_CTRL 0x0A +#define WM8990_LEFT_DAC_DIGITAL_VOLUME 0x0B +#define WM8990_RIGHT_DAC_DIGITAL_VOLUME 0x0C +#define WM8990_DIGITAL_SIDE_TONE 0x0D +#define WM8990_ADC_CTRL 0x0E +#define WM8990_LEFT_ADC_DIGITAL_VOLUME 0x0F +#define WM8990_RIGHT_ADC_DIGITAL_VOLUME 0x10 +#define WM8990_GPIO_CTRL_1 0x12 +#define WM8990_GPIO1_GPIO2 0x13 +#define WM8990_GPIO3_GPIO4 0x14 +#define WM8990_GPIO5_GPIO6 0x15 +#define WM8990_GPIOCTRL_2 0x16 +#define WM8990_GPIO_POL 0x17 +#define WM8990_LEFT_LINE_INPUT_1_2_VOLUME 0x18 +#define WM8990_LEFT_LINE_INPUT_3_4_VOLUME 0x19 +#define WM8990_RIGHT_LINE_INPUT_1_2_VOLUME 0x1A +#define WM8990_RIGHT_LINE_INPUT_3_4_VOLUME 0x1B +#define WM8990_LEFT_OUTPUT_VOLUME 0x1C +#define WM8990_RIGHT_OUTPUT_VOLUME 0x1D +#define WM8990_LINE_OUTPUTS_VOLUME 0x1E +#define WM8990_OUT3_4_VOLUME 0x1F +#define WM8990_LEFT_OPGA_VOLUME 0x20 +#define WM8990_RIGHT_OPGA_VOLUME 0x21 +#define WM8990_SPEAKER_VOLUME 0x22 +#define WM8990_CLASSD1 0x23 +#define WM8990_CLASSD3 0x25 +#define WM8990_INPUT_MIXER1 0x27 +#define WM8990_INPUT_MIXER2 0x28 +#define WM8990_INPUT_MIXER3 0x29 +#define WM8990_INPUT_MIXER4 0x2A +#define WM8990_INPUT_MIXER5 0x2B +#define WM8990_INPUT_MIXER6 0x2C +#define WM8990_OUTPUT_MIXER1 0x2D +#define WM8990_OUTPUT_MIXER2 0x2E +#define WM8990_OUTPUT_MIXER3 0x2F +#define WM8990_OUTPUT_MIXER4 0x30 +#define WM8990_OUTPUT_MIXER5 0x31 +#define WM8990_OUTPUT_MIXER6 0x32 +#define WM8990_OUT3_4_MIXER 0x33 +#define WM8990_LINE_MIXER1 0x34 +#define WM8990_LINE_MIXER2 0x35 +#define WM8990_SPEAKER_MIXER 0x36 +#define WM8990_ADDITIONAL_CONTROL 0x37 +#define WM8990_ANTIPOP1 0x38 +#define WM8990_ANTIPOP2 0x39 +#define WM8990_MICBIAS 0x3A +#define WM8990_PLL1 0x3C +#define WM8990_PLL2 0x3D +#define WM8990_PLL3 0x3E +#define WM8990_INTDRIVBITS 0x3F + +#define WM8990_REGISTER_COUNT 60 +#define WM8990_MAX_REGISTER 0x3F + +/* + * Field Definitions. + */ + +/* + * R0 (0x00) - Reset + */ +#define WM8990_SW_RESET_CHIP_ID_MASK 0xFFFF /* SW_RESET_CHIP_ID */ + +/* + * R1 (0x01) - Power Management (1) + */ +#define WM8990_SPK_ENA 0x1000 /* SPK_ENA */ +#define WM8990_SPK_ENA_BIT 12 +#define WM8990_OUT3_ENA 0x0800 /* OUT3_ENA */ +#define WM8990_OUT3_ENA_BIT 11 +#define WM8990_OUT4_ENA 0x0400 /* OUT4_ENA */ +#define WM8990_OUT4_ENA_BIT 10 +#define WM8990_LOUT_ENA 0x0200 /* LOUT_ENA */ +#define WM8990_LOUT_ENA_BIT 9 +#define WM8990_ROUT_ENA 0x0100 /* ROUT_ENA */ +#define WM8990_ROUT_ENA_BIT 8 +#define WM8990_MICBIAS_ENA 0x0010 /* MICBIAS_ENA */ +#define WM8990_MICBIAS_ENA_BIT 4 +#define WM8990_VMID_MODE_MASK 0x0006 /* VMID_MODE - [2:1] */ +#define WM8990_VREF_ENA 0x0001 /* VREF_ENA */ +#define WM8990_VREF_ENA_BIT 0 + +/* + * R2 (0x02) - Power Management (2) + */ +#define WM8990_PLL_ENA 0x8000 /* PLL_ENA */ +#define WM8990_PLL_ENA_BIT 15 +#define WM8990_TSHUT_ENA 0x4000 /* TSHUT_ENA */ +#define WM8990_TSHUT_ENA_BIT 14 +#define WM8990_TSHUT_OPDIS 0x2000 /* TSHUT_OPDIS */ +#define WM8990_TSHUT_OPDIS_BIT 13 +#define WM8990_OPCLK_ENA 0x0800 /* OPCLK_ENA */ +#define WM8990_OPCLK_ENA_BIT 11 +#define WM8990_AINL_ENA 0x0200 /* AINL_ENA */ +#define WM8990_AINL_ENA_BIT 9 +#define WM8990_AINR_ENA 0x0100 /* AINR_ENA */ +#define WM8990_AINR_ENA_BIT 8 +#define WM8990_LIN34_ENA 0x0080 /* LIN34_ENA */ +#define WM8990_LIN34_ENA_BIT 7 +#define WM8990_LIN12_ENA 0x0040 /* LIN12_ENA */ +#define WM8990_LIN12_ENA_BIT 6 +#define WM8990_RIN34_ENA 0x0020 /* RIN34_ENA */ +#define WM8990_RIN34_ENA_BIT 5 +#define WM8990_RIN12_ENA 0x0010 /* RIN12_ENA */ +#define WM8990_RIN12_ENA_BIT 4 +#define WM8990_ADCL_ENA 0x0002 /* ADCL_ENA */ +#define WM8990_ADCL_ENA_BIT 1 +#define WM8990_ADCR_ENA 0x0001 /* ADCR_ENA */ +#define WM8990_ADCR_ENA_BIT 0 + +/* + * R3 (0x03) - Power Management (3) + */ +#define WM8990_LON_ENA 0x2000 /* LON_ENA */ +#define WM8990_LON_ENA_BIT 13 +#define WM8990_LOP_ENA 0x1000 /* LOP_ENA */ +#define WM8990_LOP_ENA_BIT 12 +#define WM8990_RON_ENA 0x0800 /* RON_ENA */ +#define WM8990_RON_ENA_BIT 11 +#define WM8990_ROP_ENA 0x0400 /* ROP_ENA */ +#define WM8990_ROP_ENA_BIT 10 +#define WM8990_LOPGA_ENA 0x0080 /* LOPGA_ENA */ +#define WM8990_LOPGA_ENA_BIT 7 +#define WM8990_ROPGA_ENA 0x0040 /* ROPGA_ENA */ +#define WM8990_ROPGA_ENA_BIT 6 +#define WM8990_LOMIX_ENA 0x0020 /* LOMIX_ENA */ +#define WM8990_LOMIX_ENA_BIT 5 +#define WM8990_ROMIX_ENA 0x0010 /* ROMIX_ENA */ +#define WM8990_ROMIX_ENA_BIT 4 +#define WM8990_DACL_ENA 0x0002 /* DACL_ENA */ +#define WM8990_DACL_ENA_BIT 1 +#define WM8990_DACR_ENA 0x0001 /* DACR_ENA */ +#define WM8990_DACR_ENA_BIT 0 + +/* + * R4 (0x04) - Audio Interface (1) + */ +#define WM8990_AIFADCL_SRC 0x8000 /* AIFADCL_SRC */ +#define WM8990_AIFADCR_SRC 0x4000 /* AIFADCR_SRC */ +#define WM8990_AIFADC_TDM 0x2000 /* AIFADC_TDM */ +#define WM8990_AIFADC_TDM_CHAN 0x1000 /* AIFADC_TDM_CHAN */ +#define WM8990_AIF_BCLK_INV 0x0100 /* AIF_BCLK_INV */ +#define WM8990_AIF_LRCLK_INV 0x0080 /* AIF_LRCLK_INV */ +#define WM8990_AIF_WL_MASK 0x0060 /* AIF_WL - [6:5] */ +#define WM8990_AIF_WL_16BITS (0 << 5) +#define WM8990_AIF_WL_20BITS (1 << 5) +#define WM8990_AIF_WL_24BITS (2 << 5) +#define WM8990_AIF_WL_32BITS (3 << 5) +#define WM8990_AIF_FMT_MASK 0x0018 /* AIF_FMT - [4:3] */ +#define WM8990_AIF_TMF_RIGHTJ (0 << 3) +#define WM8990_AIF_TMF_LEFTJ (1 << 3) +#define WM8990_AIF_TMF_I2S (2 << 3) +#define WM8990_AIF_TMF_DSP (3 << 3) + +/* + * R5 (0x05) - Audio Interface (2) + */ +#define WM8990_DACL_SRC 0x8000 /* DACL_SRC */ +#define WM8990_DACR_SRC 0x4000 /* DACR_SRC */ +#define WM8990_AIFDAC_TDM 0x2000 /* AIFDAC_TDM */ +#define WM8990_AIFDAC_TDM_CHAN 0x1000 /* AIFDAC_TDM_CHAN */ +#define WM8990_DAC_BOOST_MASK 0x0C00 /* DAC_BOOST */ +#define WM8990_DAC_COMP 0x0010 /* DAC_COMP */ +#define WM8990_DAC_COMPMODE 0x0008 /* DAC_COMPMODE */ +#define WM8990_ADC_COMP 0x0004 /* ADC_COMP */ +#define WM8990_ADC_COMPMODE 0x0002 /* ADC_COMPMODE */ +#define WM8990_LOOPBACK 0x0001 /* LOOPBACK */ + +/* + * R6 (0x06) - Clocking (1) + */ +#define WM8990_TOCLK_RATE 0x8000 /* TOCLK_RATE */ +#define WM8990_TOCLK_ENA 0x4000 /* TOCLK_ENA */ +#define WM8990_OPCLKDIV_MASK 0x1E00 /* OPCLKDIV - [12:9] */ +#define WM8990_DCLKDIV_MASK 0x01C0 /* DCLKDIV - [8:6] */ +#define WM8990_BCLK_DIV_MASK 0x001E /* BCLK_DIV - [4:1] */ +#define WM8990_BCLK_DIV_1 (0x0 << 1) +#define WM8990_BCLK_DIV_1_5 (0x1 << 1) +#define WM8990_BCLK_DIV_2 (0x2 << 1) +#define WM8990_BCLK_DIV_3 (0x3 << 1) +#define WM8990_BCLK_DIV_4 (0x4 << 1) +#define WM8990_BCLK_DIV_5_5 (0x5 << 1) +#define WM8990_BCLK_DIV_6 (0x6 << 1) +#define WM8990_BCLK_DIV_8 (0x7 << 1) +#define WM8990_BCLK_DIV_11 (0x8 << 1) +#define WM8990_BCLK_DIV_12 (0x9 << 1) +#define WM8990_BCLK_DIV_16 (0xA << 1) +#define WM8990_BCLK_DIV_22 (0xB << 1) +#define WM8990_BCLK_DIV_24 (0xC << 1) +#define WM8990_BCLK_DIV_32 (0xD << 1) +#define WM8990_BCLK_DIV_44 (0xE << 1) +#define WM8990_BCLK_DIV_48 (0xF << 1) + +/* + * R7 (0x07) - Clocking (2) + */ +#define WM8990_MCLK_SRC 0x8000 /* MCLK_SRC */ +#define WM8990_SYSCLK_SRC 0x4000 /* SYSCLK_SRC */ +#define WM8990_CLK_FORCE 0x2000 /* CLK_FORCE */ +#define WM8990_MCLK_DIV_MASK 0x1800 /* MCLK_DIV - [12:11] */ +#define WM8990_MCLK_DIV_1 (0 << 11) +#define WM8990_MCLK_DIV_2 (2 << 11) +#define WM8990_MCLK_INV 0x0400 /* MCLK_INV */ +#define WM8990_ADC_CLKDIV_MASK 0x00E0 /* ADC_CLKDIV */ +#define WM8990_ADC_CLKDIV_1 (0 << 5) +#define WM8990_ADC_CLKDIV_1_5 (1 << 5) +#define WM8990_ADC_CLKDIV_2 (2 << 5) +#define WM8990_ADC_CLKDIV_3 (3 << 5) +#define WM8990_ADC_CLKDIV_4 (4 << 5) +#define WM8990_ADC_CLKDIV_5_5 (5 << 5) +#define WM8990_ADC_CLKDIV_6 (6 << 5) +#define WM8990_DAC_CLKDIV_MASK 0x001C /* DAC_CLKDIV - [4:2] */ +#define WM8990_DAC_CLKDIV_1 (0 << 2) +#define WM8990_DAC_CLKDIV_1_5 (1 << 2) +#define WM8990_DAC_CLKDIV_2 (2 << 2) +#define WM8990_DAC_CLKDIV_3 (3 << 2) +#define WM8990_DAC_CLKDIV_4 (4 << 2) +#define WM8990_DAC_CLKDIV_5_5 (5 << 2) +#define WM8990_DAC_CLKDIV_6 (6 << 2) + +/* + * R8 (0x08) - Audio Interface (3) + */ +#define WM8990_AIF_MSTR1 0x8000 /* AIF_MSTR1 */ +#define WM8990_AIF_MSTR2 0x4000 /* AIF_MSTR2 */ +#define WM8990_AIF_SEL 0x2000 /* AIF_SEL */ +#define WM8990_ADCLRC_DIR 0x0800 /* ADCLRC_DIR */ +#define WM8990_ADCLRC_RATE_MASK 0x07FF /* ADCLRC_RATE */ + +/* + * R9 (0x09) - Audio Interface (4) + */ +#define WM8990_ALRCGPIO1 0x8000 /* ALRCGPIO1 */ +#define WM8990_ALRCBGPIO6 0x4000 /* ALRCBGPIO6 */ +#define WM8990_AIF_TRIS 0x2000 /* AIF_TRIS */ +#define WM8990_DACLRC_DIR 0x0800 /* DACLRC_DIR */ +#define WM8990_DACLRC_RATE_MASK 0x07FF /* DACLRC_RATE */ + +/* + * R10 (0x0A) - DAC CTRL + */ +#define WM8990_AIF_LRCLKRATE 0x0400 /* AIF_LRCLKRATE */ +#define WM8990_DAC_MONO 0x0200 /* DAC_MONO */ +#define WM8990_DAC_SB_FILT 0x0100 /* DAC_SB_FILT */ +#define WM8990_DAC_MUTERATE 0x0080 /* DAC_MUTERATE */ +#define WM8990_DAC_MUTEMODE 0x0040 /* DAC_MUTEMODE */ +#define WM8990_DEEMP_MASK 0x0030 /* DEEMP - [5:4] */ +#define WM8990_DAC_MUTE 0x0004 /* DAC_MUTE */ +#define WM8990_DACL_DATINV 0x0002 /* DACL_DATINV */ +#define WM8990_DACR_DATINV 0x0001 /* DACR_DATINV */ + +/* + * R11 (0x0B) - Left DAC Digital Volume + */ +#define WM8990_DAC_VU 0x0100 /* DAC_VU */ +#define WM8990_DACL_VOL_MASK 0x00FF /* DACL_VOL - [7:0] */ +#define WM8990_DACL_VOL_SHIFT 0 +/* + * R12 (0x0C) - Right DAC Digital Volume + */ +#define WM8990_DAC_VU 0x0100 /* DAC_VU */ +#define WM8990_DACR_VOL_MASK 0x00FF /* DACR_VOL - [7:0] */ +#define WM8990_DACR_VOL_SHIFT 0 +/* + * R13 (0x0D) - Digital Side Tone + */ +#define WM8990_ADCL_DAC_SVOL_MASK 0x0F /* ADCL_DAC_SVOL */ +#define WM8990_ADCL_DAC_SVOL_SHIFT 9 +#define WM8990_ADCR_DAC_SVOL_MASK 0x0F /* ADCR_DAC_SVOL */ +#define WM8990_ADCR_DAC_SVOL_SHIFT 5 +#define WM8990_ADC_TO_DACL_MASK 0x03 /* ADC_TO_DACL - [3:2] */ +#define WM8990_ADC_TO_DACL_SHIFT 2 +#define WM8990_ADC_TO_DACR_MASK 0x03 /* ADC_TO_DACR - [1:0] */ +#define WM8990_ADC_TO_DACR_SHIFT 0 + +/* + * R14 (0x0E) - ADC CTRL + */ +#define WM8990_ADC_HPF_ENA 0x0100 /* ADC_HPF_ENA */ +#define WM8990_ADC_HPF_ENA_BIT 8 +#define WM8990_ADC_HPF_CUT_MASK 0x03 /* ADC_HPF_CUT - [6:5] */ +#define WM8990_ADC_HPF_CUT_SHIFT 5 +#define WM8990_ADCL_DATINV 0x0002 /* ADCL_DATINV */ +#define WM8990_ADCL_DATINV_BIT 1 +#define WM8990_ADCR_DATINV 0x0001 /* ADCR_DATINV */ +#define WM8990_ADCR_DATINV_BIT 0 + +/* + * R15 (0x0F) - Left ADC Digital Volume + */ +#define WM8990_ADC_VU 0x0100 /* ADC_VU */ +#define WM8990_ADCL_VOL_MASK 0x00FF /* ADCL_VOL - [7:0] */ +#define WM8990_ADCL_VOL_SHIFT 0 + +/* + * R16 (0x10) - Right ADC Digital Volume + */ +#define WM8990_ADC_VU 0x0100 /* ADC_VU */ +#define WM8990_ADCR_VOL_MASK 0x00FF /* ADCR_VOL - [7:0] */ +#define WM8990_ADCR_VOL_SHIFT 0 + +/* + * R18 (0x12) - GPIO CTRL 1 + */ +#define WM8990_IRQ 0x1000 /* IRQ */ +#define WM8990_TEMPOK 0x0800 /* TEMPOK */ +#define WM8990_MICSHRT 0x0400 /* MICSHRT */ +#define WM8990_MICDET 0x0200 /* MICDET */ +#define WM8990_PLL_LCK 0x0100 /* PLL_LCK */ +#define WM8990_GPI8_STATUS 0x0080 /* GPI8_STATUS */ +#define WM8990_GPI7_STATUS 0x0040 /* GPI7_STATUS */ +#define WM8990_GPIO6_STATUS 0x0020 /* GPIO6_STATUS */ +#define WM8990_GPIO5_STATUS 0x0010 /* GPIO5_STATUS */ +#define WM8990_GPIO4_STATUS 0x0008 /* GPIO4_STATUS */ +#define WM8990_GPIO3_STATUS 0x0004 /* GPIO3_STATUS */ +#define WM8990_GPIO2_STATUS 0x0002 /* GPIO2_STATUS */ +#define WM8990_GPIO1_STATUS 0x0001 /* GPIO1_STATUS */ + +/* + * R19 (0x13) - GPIO1 & GPIO2 + */ +#define WM8990_GPIO2_DEB_ENA 0x8000 /* GPIO2_DEB_ENA */ +#define WM8990_GPIO2_IRQ_ENA 0x4000 /* GPIO2_IRQ_ENA */ +#define WM8990_GPIO2_PU 0x2000 /* GPIO2_PU */ +#define WM8990_GPIO2_PD 0x1000 /* GPIO2_PD */ +#define WM8990_GPIO2_SEL_MASK 0x0F00 /* GPIO2_SEL - [11:8] */ +#define WM8990_GPIO1_DEB_ENA 0x0080 /* GPIO1_DEB_ENA */ +#define WM8990_GPIO1_IRQ_ENA 0x0040 /* GPIO1_IRQ_ENA */ +#define WM8990_GPIO1_PU 0x0020 /* GPIO1_PU */ +#define WM8990_GPIO1_PD 0x0010 /* GPIO1_PD */ +#define WM8990_GPIO1_SEL_MASK 0x000F /* GPIO1_SEL - [3:0] */ + +/* + * R20 (0x14) - GPIO3 & GPIO4 + */ +#define WM8990_GPIO4_DEB_ENA 0x8000 /* GPIO4_DEB_ENA */ +#define WM8990_GPIO4_IRQ_ENA 0x4000 /* GPIO4_IRQ_ENA */ +#define WM8990_GPIO4_PU 0x2000 /* GPIO4_PU */ +#define WM8990_GPIO4_PD 0x1000 /* GPIO4_PD */ +#define WM8990_GPIO4_SEL_MASK 0x0F00 /* GPIO4_SEL - [11:8] */ +#define WM8990_GPIO3_DEB_ENA 0x0080 /* GPIO3_DEB_ENA */ +#define WM8990_GPIO3_IRQ_ENA 0x0040 /* GPIO3_IRQ_ENA */ +#define WM8990_GPIO3_PU 0x0020 /* GPIO3_PU */ +#define WM8990_GPIO3_PD 0x0010 /* GPIO3_PD */ +#define WM8990_GPIO3_SEL_MASK 0x000F /* GPIO3_SEL - [3:0] */ + +/* + * R21 (0x15) - GPIO5 & GPIO6 + */ +#define WM8990_GPIO6_DEB_ENA 0x8000 /* GPIO6_DEB_ENA */ +#define WM8990_GPIO6_IRQ_ENA 0x4000 /* GPIO6_IRQ_ENA */ +#define WM8990_GPIO6_PU 0x2000 /* GPIO6_PU */ +#define WM8990_GPIO6_PD 0x1000 /* GPIO6_PD */ +#define WM8990_GPIO6_SEL_MASK 0x0F00 /* GPIO6_SEL - [11:8] */ +#define WM8990_GPIO5_DEB_ENA 0x0080 /* GPIO5_DEB_ENA */ +#define WM8990_GPIO5_IRQ_ENA 0x0040 /* GPIO5_IRQ_ENA */ +#define WM8990_GPIO5_PU 0x0020 /* GPIO5_PU */ +#define WM8990_GPIO5_PD 0x0010 /* GPIO5_PD */ +#define WM8990_GPIO5_SEL_MASK 0x000F /* GPIO5_SEL - [3:0] */ + +/* + * R22 (0x16) - GPIOCTRL 2 + */ +#define WM8990_RD_3W_ENA 0x8000 /* RD_3W_ENA */ +#define WM8990_MODE_3W4W 0x4000 /* MODE_3W4W */ +#define WM8990_TEMPOK_IRQ_ENA 0x0800 /* TEMPOK_IRQ_ENA */ +#define WM8990_MICSHRT_IRQ_ENA 0x0400 /* MICSHRT_IRQ_ENA */ +#define WM8990_MICDET_IRQ_ENA 0x0200 /* MICDET_IRQ_ENA */ +#define WM8990_PLL_LCK_IRQ_ENA 0x0100 /* PLL_LCK_IRQ_ENA */ +#define WM8990_GPI8_DEB_ENA 0x0080 /* GPI8_DEB_ENA */ +#define WM8990_GPI8_IRQ_ENA 0x0040 /* GPI8_IRQ_ENA */ +#define WM8990_GPI8_ENA 0x0010 /* GPI8_ENA */ +#define WM8990_GPI7_DEB_ENA 0x0008 /* GPI7_DEB_ENA */ +#define WM8990_GPI7_IRQ_ENA 0x0004 /* GPI7_IRQ_ENA */ +#define WM8990_GPI7_ENA 0x0001 /* GPI7_ENA */ + +/* + * R23 (0x17) - GPIO_POL + */ +#define WM8990_IRQ_INV 0x1000 /* IRQ_INV */ +#define WM8990_TEMPOK_POL 0x0800 /* TEMPOK_POL */ +#define WM8990_MICSHRT_POL 0x0400 /* MICSHRT_POL */ +#define WM8990_MICDET_POL 0x0200 /* MICDET_POL */ +#define WM8990_PLL_LCK_POL 0x0100 /* PLL_LCK_POL */ +#define WM8990_GPI8_POL 0x0080 /* GPI8_POL */ +#define WM8990_GPI7_POL 0x0040 /* GPI7_POL */ +#define WM8990_GPIO6_POL 0x0020 /* GPIO6_POL */ +#define WM8990_GPIO5_POL 0x0010 /* GPIO5_POL */ +#define WM8990_GPIO4_POL 0x0008 /* GPIO4_POL */ +#define WM8990_GPIO3_POL 0x0004 /* GPIO3_POL */ +#define WM8990_GPIO2_POL 0x0002 /* GPIO2_POL */ +#define WM8990_GPIO1_POL 0x0001 /* GPIO1_POL */ + +/* + * R24 (0x18) - Left Line Input 1&2 Volume + */ +#define WM8990_IPVU 0x0100 /* IPVU */ +#define WM8990_LI12MUTE 0x0080 /* LI12MUTE */ +#define WM8990_LI12MUTE_BIT 7 +#define WM8990_LI12ZC 0x0040 /* LI12ZC */ +#define WM8990_LI12ZC_BIT 6 +#define WM8990_LIN12VOL_MASK 0x001F /* LIN12VOL - [4:0] */ +#define WM8990_LIN12VOL_SHIFT 0 +/* + * R25 (0x19) - Left Line Input 3&4 Volume + */ +#define WM8990_IPVU 0x0100 /* IPVU */ +#define WM8990_LI34MUTE 0x0080 /* LI34MUTE */ +#define WM8990_LI34MUTE_BIT 7 +#define WM8990_LI34ZC 0x0040 /* LI34ZC */ +#define WM8990_LI34ZC_BIT 6 +#define WM8990_LIN34VOL_MASK 0x001F /* LIN34VOL - [4:0] */ +#define WM8990_LIN34VOL_SHIFT 0 + +/* + * R26 (0x1A) - Right Line Input 1&2 Volume + */ +#define WM8990_IPVU 0x0100 /* IPVU */ +#define WM8990_RI12MUTE 0x0080 /* RI12MUTE */ +#define WM8990_RI12MUTE_BIT 7 +#define WM8990_RI12ZC 0x0040 /* RI12ZC */ +#define WM8990_RI12ZC_BIT 6 +#define WM8990_RIN12VOL_MASK 0x001F /* RIN12VOL - [4:0] */ +#define WM8990_RIN12VOL_SHIFT 0 + +/* + * R27 (0x1B) - Right Line Input 3&4 Volume + */ +#define WM8990_IPVU 0x0100 /* IPVU */ +#define WM8990_RI34MUTE 0x0080 /* RI34MUTE */ +#define WM8990_RI34MUTE_BIT 7 +#define WM8990_RI34ZC 0x0040 /* RI34ZC */ +#define WM8990_RI34ZC_BIT 6 +#define WM8990_RIN34VOL_MASK 0x001F /* RIN34VOL - [4:0] */ +#define WM8990_RIN34VOL_SHIFT 0 + +/* + * R28 (0x1C) - Left Output Volume + */ +#define WM8990_OPVU 0x0100 /* OPVU */ +#define WM8990_LOZC 0x0080 /* LOZC */ +#define WM8990_LOZC_BIT 7 +#define WM8990_LOUTVOL_MASK 0x007F /* LOUTVOL - [6:0] */ +#define WM8990_LOUTVOL_SHIFT 0 +/* + * R29 (0x1D) - Right Output Volume + */ +#define WM8990_OPVU 0x0100 /* OPVU */ +#define WM8990_ROZC 0x0080 /* ROZC */ +#define WM8990_ROZC_BIT 7 +#define WM8990_ROUTVOL_MASK 0x007F /* ROUTVOL - [6:0] */ +#define WM8990_ROUTVOL_SHIFT 0 +/* + * R30 (0x1E) - Line Outputs Volume + */ +#define WM8990_LONMUTE 0x0040 /* LONMUTE */ +#define WM8990_LONMUTE_BIT 6 +#define WM8990_LOPMUTE 0x0020 /* LOPMUTE */ +#define WM8990_LOPMUTE_BIT 5 +#define WM8990_LOATTN 0x0010 /* LOATTN */ +#define WM8990_LOATTN_BIT 4 +#define WM8990_RONMUTE 0x0004 /* RONMUTE */ +#define WM8990_RONMUTE_BIT 2 +#define WM8990_ROPMUTE 0x0002 /* ROPMUTE */ +#define WM8990_ROPMUTE_BIT 1 +#define WM8990_ROATTN 0x0001 /* ROATTN */ +#define WM8990_ROATTN_BIT 0 + +/* + * R31 (0x1F) - Out3/4 Volume + */ +#define WM8990_OUT3MUTE 0x0020 /* OUT3MUTE */ +#define WM8990_OUT3MUTE_BIT 5 +#define WM8990_OUT3ATTN 0x0010 /* OUT3ATTN */ +#define WM8990_OUT3ATTN_BIT 4 +#define WM8990_OUT4MUTE 0x0002 /* OUT4MUTE */ +#define WM8990_OUT4MUTE_BIT 1 +#define WM8990_OUT4ATTN 0x0001 /* OUT4ATTN */ +#define WM8990_OUT4ATTN_BIT 0 + +/* + * R32 (0x20) - Left OPGA Volume + */ +#define WM8990_OPVU 0x0100 /* OPVU */ +#define WM8990_LOPGAZC 0x0080 /* LOPGAZC */ +#define WM8990_LOPGAZC_BIT 7 +#define WM8990_LOPGAVOL_MASK 0x007F /* LOPGAVOL - [6:0] */ +#define WM8990_LOPGAVOL_SHIFT 0 + +/* + * R33 (0x21) - Right OPGA Volume + */ +#define WM8990_OPVU 0x0100 /* OPVU */ +#define WM8990_ROPGAZC 0x0080 /* ROPGAZC */ +#define WM8990_ROPGAZC_BIT 7 +#define WM8990_ROPGAVOL_MASK 0x007F /* ROPGAVOL - [6:0] */ +#define WM8990_ROPGAVOL_SHIFT 0 +/* + * R34 (0x22) - Speaker Volume + */ +#define WM8990_SPKVOL_MASK 0x0003 /* SPKVOL - [1:0] */ +#define WM8990_SPKVOL_SHIFT 0 + +/* + * R35 (0x23) - ClassD1 + */ +#define WM8990_CDMODE 0x0100 /* CDMODE */ +#define WM8990_CDMODE_BIT 8 + +/* + * R37 (0x25) - ClassD3 + */ +#define WM8990_DCGAIN_MASK 0x0007 /* DCGAIN - [5:3] */ +#define WM8990_DCGAIN_SHIFT 3 +#define WM8990_ACGAIN_MASK 0x0007 /* ACGAIN - [2:0] */ +#define WM8990_ACGAIN_SHIFT 0 +/* + * R39 (0x27) - Input Mixer1 + */ +#define WM8990_AINLMODE_MASK 0x000C /* AINLMODE - [3:2] */ +#define WM8990_AINLMODE_SHIFT 2 +#define WM8990_AINRMODE_MASK 0x0003 /* AINRMODE - [1:0] */ +#define WM8990_AINRMODE_SHIFT 0 + +/* + * R40 (0x28) - Input Mixer2 + */ +#define WM8990_LMP4 0x0080 /* LMP4 */ +#define WM8990_LMP4_BIT 7 /* LMP4 */ +#define WM8990_LMN3 0x0040 /* LMN3 */ +#define WM8990_LMN3_BIT 6 /* LMN3 */ +#define WM8990_LMP2 0x0020 /* LMP2 */ +#define WM8990_LMP2_BIT 5 /* LMP2 */ +#define WM8990_LMN1 0x0010 /* LMN1 */ +#define WM8990_LMN1_BIT 4 /* LMN1 */ +#define WM8990_RMP4 0x0008 /* RMP4 */ +#define WM8990_RMP4_BIT 3 /* RMP4 */ +#define WM8990_RMN3 0x0004 /* RMN3 */ +#define WM8990_RMN3_BIT 2 /* RMN3 */ +#define WM8990_RMP2 0x0002 /* RMP2 */ +#define WM8990_RMP2_BIT 1 /* RMP2 */ +#define WM8990_RMN1 0x0001 /* RMN1 */ +#define WM8990_RMN1_BIT 0 /* RMN1 */ + +/* + * R41 (0x29) - Input Mixer3 + */ +#define WM8990_L34MNB 0x0100 /* L34MNB */ +#define WM8990_L34MNB_BIT 8 +#define WM8990_L34MNBST 0x0080 /* L34MNBST */ +#define WM8990_L34MNBST_BIT 7 +#define WM8990_L12MNB 0x0020 /* L12MNB */ +#define WM8990_L12MNB_BIT 5 +#define WM8990_L12MNBST 0x0010 /* L12MNBST */ +#define WM8990_L12MNBST_BIT 4 +#define WM8990_LDBVOL_MASK 0x0007 /* LDBVOL - [2:0] */ +#define WM8990_LDBVOL_SHIFT 0 + +/* + * R42 (0x2A) - Input Mixer4 + */ +#define WM8990_R34MNB 0x0100 /* R34MNB */ +#define WM8990_R34MNB_BIT 8 +#define WM8990_R34MNBST 0x0080 /* R34MNBST */ +#define WM8990_R34MNBST_BIT 7 +#define WM8990_R12MNB 0x0020 /* R12MNB */ +#define WM8990_R12MNB_BIT 5 +#define WM8990_R12MNBST 0x0010 /* R12MNBST */ +#define WM8990_R12MNBST_BIT 4 +#define WM8990_RDBVOL_MASK 0x0007 /* RDBVOL - [2:0] */ +#define WM8990_RDBVOL_SHIFT 0 + +/* + * R43 (0x2B) - Input Mixer5 + */ +#define WM8990_LI2BVOL_MASK 0x07 /* LI2BVOL - [8:6] */ +#define WM8990_LI2BVOL_SHIFT 6 +#define WM8990_LR4BVOL_MASK 0x07 /* LR4BVOL - [5:3] */ +#define WM8990_LR4BVOL_SHIFT 3 +#define WM8990_LL4BVOL_MASK 0x07 /* LL4BVOL - [2:0] */ +#define WM8990_LL4BVOL_SHIFT 0 + +/* + * R44 (0x2C) - Input Mixer6 + */ +#define WM8990_RI2BVOL_MASK 0x07 /* RI2BVOL - [8:6] */ +#define WM8990_RI2BVOL_SHIFT 6 +#define WM8990_RL4BVOL_MASK 0x07 /* RL4BVOL - [5:3] */ +#define WM8990_RL4BVOL_SHIFT 3 +#define WM8990_RR4BVOL_MASK 0x07 /* RR4BVOL - [2:0] */ +#define WM8990_RR4BVOL_SHIFT 0 + +/* + * R45 (0x2D) - Output Mixer1 + */ +#define WM8990_LRBLO 0x0080 /* LRBLO */ +#define WM8990_LRBLO_BIT 7 +#define WM8990_LLBLO 0x0040 /* LLBLO */ +#define WM8990_LLBLO_BIT 6 +#define WM8990_LRI3LO 0x0020 /* LRI3LO */ +#define WM8990_LRI3LO_BIT 5 +#define WM8990_LLI3LO 0x0010 /* LLI3LO */ +#define WM8990_LLI3LO_BIT 4 +#define WM8990_LR12LO 0x0008 /* LR12LO */ +#define WM8990_LR12LO_BIT 3 +#define WM8990_LL12LO 0x0004 /* LL12LO */ +#define WM8990_LL12LO_BIT 2 +#define WM8990_LDLO 0x0001 /* LDLO */ +#define WM8990_LDLO_BIT 0 + +/* + * R46 (0x2E) - Output Mixer2 + */ +#define WM8990_RLBRO 0x0080 /* RLBRO */ +#define WM8990_RLBRO_BIT 7 +#define WM8990_RRBRO 0x0040 /* RRBRO */ +#define WM8990_RRBRO_BIT 6 +#define WM8990_RLI3RO 0x0020 /* RLI3RO */ +#define WM8990_RLI3RO_BIT 5 +#define WM8990_RRI3RO 0x0010 /* RRI3RO */ +#define WM8990_RRI3RO_BIT 4 +#define WM8990_RL12RO 0x0008 /* RL12RO */ +#define WM8990_RL12RO_BIT 3 +#define WM8990_RR12RO 0x0004 /* RR12RO */ +#define WM8990_RR12RO_BIT 2 +#define WM8990_RDRO 0x0001 /* RDRO */ +#define WM8990_RDRO_BIT 0 + +/* + * R47 (0x2F) - Output Mixer3 + */ +#define WM8990_LLI3LOVOL_MASK 0x07 /* LLI3LOVOL - [8:6] */ +#define WM8990_LLI3LOVOL_SHIFT 6 +#define WM8990_LR12LOVOL_MASK 0x07 /* LR12LOVOL - [5:3] */ +#define WM8990_LR12LOVOL_SHIFT 3 +#define WM8990_LL12LOVOL_MASK 0x07 /* LL12LOVOL - [2:0] */ +#define WM8990_LL12LOVOL_SHIFT 0 + +/* + * R48 (0x30) - Output Mixer4 + */ +#define WM8990_RRI3ROVOL_MASK 0x07 /* RRI3ROVOL - [8:6] */ +#define WM8990_RRI3ROVOL_SHIFT 6 +#define WM8990_RL12ROVOL_MASK 0x07 /* RL12ROVOL - [5:3] */ +#define WM8990_RL12ROVOL_SHIFT 3 +#define WM8990_RR12ROVOL_MASK 0x07 /* RR12ROVOL - [2:0] */ +#define WM8990_RR12ROVOL_SHIFT 0 + +/* + * R49 (0x31) - Output Mixer5 + */ +#define WM8990_LRI3LOVOL_MASK 0x07 /* LRI3LOVOL - [8:6] */ +#define WM8990_LRI3LOVOL_SHIFT 6 +#define WM8990_LRBLOVOL_MASK 0x07 /* LRBLOVOL - [5:3] */ +#define WM8990_LRBLOVOL_SHIFT 3 +#define WM8990_LLBLOVOL_MASK 0x07 /* LLBLOVOL - [2:0] */ +#define WM8990_LLBLOVOL_SHIFT 0 + +/* + * R50 (0x32) - Output Mixer6 + */ +#define WM8990_RLI3ROVOL_MASK 0x07 /* RLI3ROVOL - [8:6] */ +#define WM8990_RLI3ROVOL_SHIFT 6 +#define WM8990_RLBROVOL_MASK 0x07 /* RLBROVOL - [5:3] */ +#define WM8990_RLBROVOL_SHIFT 3 +#define WM8990_RRBROVOL_MASK 0x07 /* RRBROVOL - [2:0] */ +#define WM8990_RRBROVOL_SHIFT 0 + +/* + * R51 (0x33) - Out3/4 Mixer + */ +#define WM8990_VSEL_MASK 0x0180 /* VSEL - [8:7] */ +#define WM8990_LI4O3 0x0020 /* LI4O3 */ +#define WM8990_LI4O3_BIT 5 +#define WM8990_LPGAO3 0x0010 /* LPGAO3 */ +#define WM8990_LPGAO3_BIT 4 +#define WM8990_RI4O4 0x0002 /* RI4O4 */ +#define WM8990_RI4O4_BIT 1 +#define WM8990_RPGAO4 0x0001 /* RPGAO4 */ +#define WM8990_RPGAO4_BIT 0 +/* + * R52 (0x34) - Line Mixer1 + */ +#define WM8990_LLOPGALON 0x0040 /* LLOPGALON */ +#define WM8990_LLOPGALON_BIT 6 +#define WM8990_LROPGALON 0x0020 /* LROPGALON */ +#define WM8990_LROPGALON_BIT 5 +#define WM8990_LOPLON 0x0010 /* LOPLON */ +#define WM8990_LOPLON_BIT 4 +#define WM8990_LR12LOP 0x0004 /* LR12LOP */ +#define WM8990_LR12LOP_BIT 2 +#define WM8990_LL12LOP 0x0002 /* LL12LOP */ +#define WM8990_LL12LOP_BIT 1 +#define WM8990_LLOPGALOP 0x0001 /* LLOPGALOP */ +#define WM8990_LLOPGALOP_BIT 0 +/* + * R53 (0x35) - Line Mixer2 + */ +#define WM8990_RROPGARON 0x0040 /* RROPGARON */ +#define WM8990_RROPGARON_BIT 6 +#define WM8990_RLOPGARON 0x0020 /* RLOPGARON */ +#define WM8990_RLOPGARON_BIT 5 +#define WM8990_ROPRON 0x0010 /* ROPRON */ +#define WM8990_ROPRON_BIT 4 +#define WM8990_RL12ROP 0x0004 /* RL12ROP */ +#define WM8990_RL12ROP_BIT 2 +#define WM8990_RR12ROP 0x0002 /* RR12ROP */ +#define WM8990_RR12ROP_BIT 1 +#define WM8990_RROPGAROP 0x0001 /* RROPGAROP */ +#define WM8990_RROPGAROP_BIT 0 + +/* + * R54 (0x36) - Speaker Mixer + */ +#define WM8990_LB2SPK 0x0080 /* LB2SPK */ +#define WM8990_LB2SPK_BIT 7 +#define WM8990_RB2SPK 0x0040 /* RB2SPK */ +#define WM8990_RB2SPK_BIT 6 +#define WM8990_LI2SPK 0x0020 /* LI2SPK */ +#define WM8990_LI2SPK_BIT 5 +#define WM8990_RI2SPK 0x0010 /* RI2SPK */ +#define WM8990_RI2SPK_BIT 4 +#define WM8990_LOPGASPK 0x0008 /* LOPGASPK */ +#define WM8990_LOPGASPK_BIT 3 +#define WM8990_ROPGASPK 0x0004 /* ROPGASPK */ +#define WM8990_ROPGASPK_BIT 2 +#define WM8990_LDSPK 0x0002 /* LDSPK */ +#define WM8990_LDSPK_BIT 1 +#define WM8990_RDSPK 0x0001 /* RDSPK */ +#define WM8990_RDSPK_BIT 0 + +/* + * R55 (0x37) - Additional Control + */ +#define WM8990_VROI 0x0001 /* VROI */ + +/* + * R56 (0x38) - AntiPOP1 + */ +#define WM8990_DIS_LLINE 0x0020 /* DIS_LLINE */ +#define WM8990_DIS_RLINE 0x0010 /* DIS_RLINE */ +#define WM8990_DIS_OUT3 0x0008 /* DIS_OUT3 */ +#define WM8990_DIS_OUT4 0x0004 /* DIS_OUT4 */ +#define WM8990_DIS_LOUT 0x0002 /* DIS_LOUT */ +#define WM8990_DIS_ROUT 0x0001 /* DIS_ROUT */ + +/* + * R57 (0x39) - AntiPOP2 + */ +#define WM8990_SOFTST 0x0040 /* SOFTST */ +#define WM8990_BUFIOEN 0x0008 /* BUFIOEN */ +#define WM8990_BUFDCOPEN 0x0004 /* BUFDCOPEN */ +#define WM8990_POBCTRL 0x0002 /* POBCTRL */ +#define WM8990_VMIDTOG 0x0001 /* VMIDTOG */ + +/* + * R58 (0x3A) - MICBIAS + */ +#define WM8990_MCDSCTH_MASK 0x00C0 /* MCDSCTH - [7:6] */ +#define WM8990_MCDTHR_MASK 0x0038 /* MCDTHR - [5:3] */ +#define WM8990_MCD 0x0004 /* MCD */ +#define WM8990_MBSEL 0x0001 /* MBSEL */ + +/* + * R60 (0x3C) - PLL1 + */ +#define WM8990_SDM 0x0080 /* SDM */ +#define WM8990_PRESCALE 0x0040 /* PRESCALE */ +#define WM8990_PLLN_MASK 0x000F /* PLLN - [3:0] */ + +/* + * R61 (0x3D) - PLL2 + */ +#define WM8990_PLLK1_MASK 0x00FF /* PLLK1 - [7:0] */ + +/* + * R62 (0x3E) - PLL3 + */ +#define WM8990_PLLK2_MASK 0x00FF /* PLLK2 - [7:0] */ + +/* + * R63 (0x3F) - Internal Driver Bits + */ +#define WM8990_INMIXL_PWR_BIT 0 +#define WM8990_AINLMUX_PWR_BIT 1 +#define WM8990_INMIXR_PWR_BIT 2 +#define WM8990_AINRMUX_PWR_BIT 3 + +struct wm8990_setup_data { + unsigned short i2c_address; +}; + +#define WM8990_MCLK_DIV 0 +#define WM8990_DACCLK_DIV 1 +#define WM8990_ADCCLK_DIV 2 +#define WM8990_BCLK_DIV 3 + +extern struct snd_soc_dai wm8990_dai; +extern struct snd_soc_codec_device soc_codec_dev_wm8990; + +#endif /* __WM8990REGISTERDEFS_H__ */ +/*------------------------------ END OF FILE ---------------------------------*/ diff --git a/sound/soc/codecs/wm9712.c b/sound/soc/codecs/wm9712.c index 76c1e2d33e7..9fc8edd8222 100644 --- a/sound/soc/codecs/wm9712.c +++ b/sound/soc/codecs/wm9712.c @@ -9,9 +9,6 @@ * under the terms of the GNU General Public License as published by the * Free Software Foundation; either version 2 of the License, or (at your * option) any later version. - * - * Revision history - * 4th Feb 2006 Initial version. */ #include <linux/init.h> @@ -25,6 +22,7 @@ #include <sound/initval.h> #include <sound/soc.h> #include <sound/soc-dapm.h> +#include "wm9712.h" #define WM9712_VERSION "0.4" @@ -351,7 +349,7 @@ SND_SOC_DAPM_INPUT("MIC1"), SND_SOC_DAPM_INPUT("MIC2"), }; -static const char *audio_map[][3] = { +static const struct snd_soc_dapm_route audio_map[] = { /* virtual mixer - mixes left & right channels for spk and mono */ {"AC97 Mixer", NULL, "Left DAC"}, {"AC97 Mixer", NULL, "Right DAC"}, @@ -446,21 +444,14 @@ static const char *audio_map[][3] = { {"Speaker PGA", NULL, "Speaker Mux"}, {"LOUT2", NULL, "Speaker PGA"}, {"ROUT2", NULL, "Speaker PGA"}, - - {NULL, NULL, NULL}, }; static int wm9712_add_widgets(struct snd_soc_codec *codec) { - int i; - - for (i = 0; i < ARRAY_SIZE(wm9712_dapm_widgets); i++) - snd_soc_dapm_new_control(codec, &wm9712_dapm_widgets[i]); + snd_soc_dapm_new_controls(codec, wm9712_dapm_widgets, + ARRAY_SIZE(wm9712_dapm_widgets)); - /* set up audio path connects */ - for (i = 0; audio_map[i][0] != NULL; i++) - snd_soc_dapm_connect_input(codec, audio_map[i][0], - audio_map[i][1], audio_map[i][2]); + snd_soc_dapm_add_routes(codec, audio_map, ARRAY_SIZE(audio_map)); snd_soc_dapm_new_widgets(codec); return 0; @@ -541,7 +532,7 @@ static int ac97_aux_prepare(struct snd_pcm_substream *substream) SNDRV_PCM_RATE_22050 | SNDRV_PCM_RATE_44100 |\ SNDRV_PCM_RATE_48000) -struct snd_soc_codec_dai wm9712_dai[] = { +struct snd_soc_dai wm9712_dai[] = { { .name = "AC97 HiFi", .type = SND_SOC_DAI_AC97_BUS, @@ -574,23 +565,23 @@ struct snd_soc_codec_dai wm9712_dai[] = { }; EXPORT_SYMBOL_GPL(wm9712_dai); -static int wm9712_dapm_event(struct snd_soc_codec *codec, int event) +static int wm9712_set_bias_level(struct snd_soc_codec *codec, + enum snd_soc_bias_level level) { - switch (event) { - case SNDRV_CTL_POWER_D0: /* full On */ - case SNDRV_CTL_POWER_D1: /* partial On */ - case SNDRV_CTL_POWER_D2: /* partial On */ + switch (level) { + case SND_SOC_BIAS_ON: + case SND_SOC_BIAS_PREPARE: break; - case SNDRV_CTL_POWER_D3hot: /* Off, with power */ + case SND_SOC_BIAS_STANDBY: ac97_write(codec, AC97_POWERDOWN, 0x0000); break; - case SNDRV_CTL_POWER_D3cold: /* Off, without power */ + case SND_SOC_BIAS_OFF: /* disable everything including AC link */ ac97_write(codec, AC97_EXTENDED_MSTATUS, 0xffff); ac97_write(codec, AC97_POWERDOWN, 0xffff); break; } - codec->dapm_state = event; + codec->bias_level = level; return 0; } @@ -598,12 +589,12 @@ static int wm9712_reset(struct snd_soc_codec *codec, int try_warm) { if (try_warm && soc_ac97_ops.warm_reset) { soc_ac97_ops.warm_reset(codec->ac97); - if (!(ac97_read(codec, 0) & 0x8000)) + if (ac97_read(codec, 0) == wm9712_reg[0]) return 1; } soc_ac97_ops.reset(codec->ac97); - if (ac97_read(codec, 0) & 0x8000) + if (ac97_read(codec, 0) != wm9712_reg[0]) goto err; return 0; @@ -618,7 +609,7 @@ static int wm9712_soc_suspend(struct platform_device *pdev, struct snd_soc_device *socdev = platform_get_drvdata(pdev); struct snd_soc_codec *codec = socdev->codec; - wm9712_dapm_event(codec, SNDRV_CTL_POWER_D3cold); + wm9712_set_bias_level(codec, SND_SOC_BIAS_OFF); return 0; } @@ -635,7 +626,7 @@ static int wm9712_soc_resume(struct platform_device *pdev) return ret; } - wm9712_dapm_event(codec, SNDRV_CTL_POWER_D3hot); + wm9712_set_bias_level(codec, SND_SOC_BIAS_STANDBY); if (ret == 0) { /* Sync reg_cache with the hardware after cold reset */ @@ -647,8 +638,8 @@ static int wm9712_soc_resume(struct platform_device *pdev) } } - if (codec->suspend_dapm_state == SNDRV_CTL_POWER_D0) - wm9712_dapm_event(codec, SNDRV_CTL_POWER_D0); + if (codec->suspend_bias_level == SND_SOC_BIAS_ON) + wm9712_set_bias_level(codec, SND_SOC_BIAS_ON); return ret; } @@ -682,7 +673,7 @@ static int wm9712_soc_probe(struct platform_device *pdev) codec->num_dai = ARRAY_SIZE(wm9712_dai); codec->write = ac97_write; codec->read = ac97_read; - codec->dapm_event = wm9712_dapm_event; + codec->set_bias_level = wm9712_set_bias_level; INIT_LIST_HEAD(&codec->dapm_widgets); INIT_LIST_HEAD(&codec->dapm_paths); @@ -706,7 +697,7 @@ static int wm9712_soc_probe(struct platform_device *pdev) /* set alc mux to none */ ac97_write(codec, AC97_VIDEO, ac97_read(codec, AC97_VIDEO) | 0x3000); - wm9712_dapm_event(codec, SNDRV_CTL_POWER_D3hot); + wm9712_set_bias_level(codec, SND_SOC_BIAS_STANDBY); wm9712_add_controls(codec); wm9712_add_widgets(codec); ret = snd_soc_register_card(socdev); diff --git a/sound/soc/codecs/wm9712.h b/sound/soc/codecs/wm9712.h index 719105d61e6..d29e8a18ca6 100644 --- a/sound/soc/codecs/wm9712.h +++ b/sound/soc/codecs/wm9712.h @@ -8,7 +8,7 @@ #define WM9712_DAI_AC97_HIFI 0 #define WM9712_DAI_AC97_AUX 1 -extern struct snd_soc_codec_dai wm9712_dai[2]; +extern struct snd_soc_dai wm9712_dai[2]; extern struct snd_soc_codec_device soc_codec_dev_wm9712; #endif diff --git a/sound/soc/codecs/wm9713.c b/sound/soc/codecs/wm9713.c index 1f241161445..38d1fe0971f 100644 --- a/sound/soc/codecs/wm9713.c +++ b/sound/soc/codecs/wm9713.c @@ -10,9 +10,6 @@ * Free Software Foundation; either version 2 of the License, or (at your * option) any later version. * - * Revision history - * 4th Feb 2006 Initial version. - * * Features:- * * o Support for AC97 Codec, Voice DAC and Aux DAC @@ -456,7 +453,7 @@ SND_SOC_DAPM_INPUT("MIC2B"), SND_SOC_DAPM_VMID("VMID"), }; -static const char *audio_map[][3] = { +static const struct snd_soc_dapm_route audio_map[] = { /* left HP mixer */ {"Left HP Mixer", "PC Beep Playback Switch", "PCBEEP"}, {"Left HP Mixer", "Voice Playback Switch", "Voice DAC"}, @@ -607,21 +604,14 @@ static const char *audio_map[][3] = { {"Capture Mono Mux", "Stereo", "Capture Mixer"}, {"Capture Mono Mux", "Left", "Left Capture Source"}, {"Capture Mono Mux", "Right", "Right Capture Source"}, - - {NULL, NULL, NULL}, }; static int wm9713_add_widgets(struct snd_soc_codec *codec) { - int i; - - for (i = 0; i < ARRAY_SIZE(wm9713_dapm_widgets); i++) - snd_soc_dapm_new_control(codec, &wm9713_dapm_widgets[i]); + snd_soc_dapm_new_controls(codec, wm9713_dapm_widgets, + ARRAY_SIZE(wm9713_dapm_widgets)); - /* set up audio path audio_mapnects */ - for (i = 0; audio_map[i][0] != NULL; i++) - snd_soc_dapm_connect_input(codec, audio_map[i][0], - audio_map[i][1], audio_map[i][2]); + snd_soc_dapm_add_routes(codec, audio_map, ARRAY_SIZE(audio_map)); snd_soc_dapm_new_widgets(codec); return 0; @@ -799,7 +789,7 @@ static int wm9713_set_pll(struct snd_soc_codec *codec, return 0; } -static int wm9713_set_dai_pll(struct snd_soc_codec_dai *codec_dai, +static int wm9713_set_dai_pll(struct snd_soc_dai *codec_dai, int pll_id, unsigned int freq_in, unsigned int freq_out) { struct snd_soc_codec *codec = codec_dai->codec; @@ -810,7 +800,7 @@ static int wm9713_set_dai_pll(struct snd_soc_codec_dai *codec_dai, * Tristate the PCM DAI lines, tristate can be disabled by calling * wm9713_set_dai_fmt() */ -static int wm9713_set_dai_tristate(struct snd_soc_codec_dai *codec_dai, +static int wm9713_set_dai_tristate(struct snd_soc_dai *codec_dai, int tristate) { struct snd_soc_codec *codec = codec_dai->codec; @@ -826,7 +816,7 @@ static int wm9713_set_dai_tristate(struct snd_soc_codec_dai *codec_dai, * Configure WM9713 clock dividers. * Voice DAC needs 256 FS */ -static int wm9713_set_dai_clkdiv(struct snd_soc_codec_dai *codec_dai, +static int wm9713_set_dai_clkdiv(struct snd_soc_dai *codec_dai, int div_id, int div) { struct snd_soc_codec *codec = codec_dai->codec; @@ -868,7 +858,7 @@ static int wm9713_set_dai_clkdiv(struct snd_soc_codec_dai *codec_dai, return 0; } -static int wm9713_set_dai_fmt(struct snd_soc_codec_dai *codec_dai, +static int wm9713_set_dai_fmt(struct snd_soc_dai *codec_dai, unsigned int fmt) { struct snd_soc_codec *codec = codec_dai->codec; @@ -886,7 +876,7 @@ static int wm9713_set_dai_fmt(struct snd_soc_codec_dai *codec_dai, gpio |= 0x0018; break; case SND_SOC_DAIFMT_CBS_CFS: - reg |= 0x0200; + reg |= 0x2000; gpio |= 0x001a; break; case SND_SOC_DAIFMT_CBS_CFM: @@ -1011,15 +1001,24 @@ static int ac97_aux_prepare(struct snd_pcm_substream *substream) return ac97_write(codec, AC97_PCM_SURR_DAC_RATE, runtime->rate); } -#define WM9713_RATES (SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_11025 |\ - SNDRV_PCM_RATE_22050 | SNDRV_PCM_RATE_44100 |\ - SNDRV_PCM_RATE_48000) +#define WM9713_RATES (SNDRV_PCM_RATE_8000 | \ + SNDRV_PCM_RATE_11025 | \ + SNDRV_PCM_RATE_22050 | \ + SNDRV_PCM_RATE_44100 | \ + SNDRV_PCM_RATE_48000) + +#define WM9713_PCM_RATES (SNDRV_PCM_RATE_8000 | \ + SNDRV_PCM_RATE_11025 | \ + SNDRV_PCM_RATE_16000 | \ + SNDRV_PCM_RATE_22050 | \ + SNDRV_PCM_RATE_44100 | \ + SNDRV_PCM_RATE_48000) #define WM9713_PCM_FORMATS \ (SNDRV_PCM_FORMAT_S16_LE | SNDRV_PCM_FORMAT_S20_3LE | \ SNDRV_PCM_FORMAT_S24_LE) -struct snd_soc_codec_dai wm9713_dai[] = { +struct snd_soc_dai wm9713_dai[] = { { .name = "AC97 HiFi", .type = SND_SOC_DAI_AC97_BUS, @@ -1061,13 +1060,13 @@ struct snd_soc_codec_dai wm9713_dai[] = { .stream_name = "Voice Playback", .channels_min = 1, .channels_max = 1, - .rates = WM9713_RATES, + .rates = WM9713_PCM_RATES, .formats = WM9713_PCM_FORMATS,}, .capture = { .stream_name = "Voice Capture", .channels_min = 1, .channels_max = 2, - .rates = WM9713_RATES, + .rates = WM9713_PCM_RATES, .formats = WM9713_PCM_FORMATS,}, .ops = { .hw_params = wm9713_pcm_hw_params, @@ -1086,44 +1085,44 @@ int wm9713_reset(struct snd_soc_codec *codec, int try_warm) { if (try_warm && soc_ac97_ops.warm_reset) { soc_ac97_ops.warm_reset(codec->ac97); - if (!(ac97_read(codec, 0) & 0x8000)) + if (ac97_read(codec, 0) == wm9713_reg[0]) return 1; } soc_ac97_ops.reset(codec->ac97); - if (ac97_read(codec, 0) & 0x8000) + if (ac97_read(codec, 0) != wm9713_reg[0]) return -EIO; return 0; } EXPORT_SYMBOL_GPL(wm9713_reset); -static int wm9713_dapm_event(struct snd_soc_codec *codec, int event) +static int wm9713_set_bias_level(struct snd_soc_codec *codec, + enum snd_soc_bias_level level) { u16 reg; - switch (event) { - case SNDRV_CTL_POWER_D0: /* full On */ + switch (level) { + case SND_SOC_BIAS_ON: /* enable thermal shutdown */ reg = ac97_read(codec, AC97_EXTENDED_MID) & 0x1bff; ac97_write(codec, AC97_EXTENDED_MID, reg); break; - case SNDRV_CTL_POWER_D1: /* partial On */ - case SNDRV_CTL_POWER_D2: /* partial On */ + case SND_SOC_BIAS_PREPARE: break; - case SNDRV_CTL_POWER_D3hot: /* Off, with power */ + case SND_SOC_BIAS_STANDBY: /* enable master bias and vmid */ reg = ac97_read(codec, AC97_EXTENDED_MID) & 0x3bff; ac97_write(codec, AC97_EXTENDED_MID, reg); ac97_write(codec, AC97_POWERDOWN, 0x0000); break; - case SNDRV_CTL_POWER_D3cold: /* Off, without power */ + case SND_SOC_BIAS_OFF: /* disable everything including AC link */ ac97_write(codec, AC97_EXTENDED_MID, 0xffff); ac97_write(codec, AC97_EXTENDED_MSTATUS, 0xffff); ac97_write(codec, AC97_POWERDOWN, 0xffff); break; } - codec->dapm_state = event; + codec->bias_level = level; return 0; } @@ -1160,7 +1159,7 @@ static int wm9713_soc_resume(struct platform_device *pdev) return ret; } - wm9713_dapm_event(codec, SNDRV_CTL_POWER_D3hot); + wm9713_set_bias_level(codec, SND_SOC_BIAS_STANDBY); /* do we need to re-start the PLL ? */ if (wm9713->pll_out) @@ -1176,8 +1175,8 @@ static int wm9713_soc_resume(struct platform_device *pdev) } } - if (codec->suspend_dapm_state == SNDRV_CTL_POWER_D0) - wm9713_dapm_event(codec, SNDRV_CTL_POWER_D0); + if (codec->suspend_bias_level == SND_SOC_BIAS_ON) + wm9713_set_bias_level(codec, SND_SOC_BIAS_ON); return ret; } @@ -1216,7 +1215,7 @@ static int wm9713_soc_probe(struct platform_device *pdev) codec->num_dai = ARRAY_SIZE(wm9713_dai); codec->write = ac97_write; codec->read = ac97_read; - codec->dapm_event = wm9713_dapm_event; + codec->set_bias_level = wm9713_set_bias_level; INIT_LIST_HEAD(&codec->dapm_widgets); INIT_LIST_HEAD(&codec->dapm_paths); @@ -1238,7 +1237,7 @@ static int wm9713_soc_probe(struct platform_device *pdev) goto reset_err; } - wm9713_dapm_event(codec, SNDRV_CTL_POWER_D3hot); + wm9713_set_bias_level(codec, SND_SOC_BIAS_STANDBY); /* unmute the adc - move to kcontrol */ reg = ac97_read(codec, AC97_CD) & 0x7fff; diff --git a/sound/soc/codecs/wm9713.h b/sound/soc/codecs/wm9713.h index d357b6c8134..63b8d81756e 100644 --- a/sound/soc/codecs/wm9713.h +++ b/sound/soc/codecs/wm9713.h @@ -46,7 +46,7 @@ #define WM9713_DAI_PCM_VOICE 2 extern struct snd_soc_codec_device soc_codec_dev_wm9713; -extern struct snd_soc_codec_dai wm9713_dai[3]; +extern struct snd_soc_dai wm9713_dai[3]; int wm9713_reset(struct snd_soc_codec *codec, int try_warm); diff --git a/sound/soc/davinci/Kconfig b/sound/soc/davinci/Kconfig index 20680c551aa..8f7e3383490 100644 --- a/sound/soc/davinci/Kconfig +++ b/sound/soc/davinci/Kconfig @@ -1,6 +1,6 @@ config SND_DAVINCI_SOC tristate "SoC Audio for the TI DAVINCI chip" - depends on ARCH_DAVINCI && SND_SOC + depends on ARCH_DAVINCI help Say Y or M if you want to add support for codecs attached to the DAVINCI AC97 or I2S interface. You will also need diff --git a/sound/soc/davinci/davinci-evm.c b/sound/soc/davinci/davinci-evm.c index fcd16524033..5e2c306399e 100644 --- a/sound/soc/davinci/davinci-evm.c +++ b/sound/soc/davinci/davinci-evm.c @@ -33,24 +33,24 @@ static int evm_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params) { struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_codec_dai *codec_dai = rtd->dai->codec_dai; - struct snd_soc_cpu_dai *cpu_dai = rtd->dai->cpu_dai; + struct snd_soc_dai *codec_dai = rtd->dai->codec_dai; + struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai; int ret = 0; /* set codec DAI configuration */ - ret = codec_dai->dai_ops.set_fmt(codec_dai, SND_SOC_DAIFMT_I2S | + ret = snd_soc_dai_set_fmt(codec_dai, SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_CBM_CFM); if (ret < 0) return ret; /* set cpu DAI configuration */ - ret = cpu_dai->dai_ops.set_fmt(cpu_dai, SND_SOC_DAIFMT_CBM_CFM | + ret = snd_soc_dai_set_fmt(cpu_dai, SND_SOC_DAIFMT_CBM_CFM | SND_SOC_DAIFMT_IB_NF); if (ret < 0) return ret; /* set the codec system clock */ - ret = codec_dai->dai_ops.set_sysclk(codec_dai, 0, EVM_CODEC_CLOCK, + ret = snd_soc_dai_set_sysclk(codec_dai, 0, EVM_CODEC_CLOCK, SND_SOC_CLOCK_OUT); if (ret < 0) return ret; @@ -71,7 +71,7 @@ static const struct snd_soc_dapm_widget aic3x_dapm_widgets[] = { }; /* davinci-evm machine audio_mapnections to the codec pins */ -static const char *audio_map[][3] = { +static const struct snd_soc_dapm_route audio_map[] = { /* Headphone connected to HPLOUT, HPROUT */ {"Headphone Jack", NULL, "HPLOUT"}, {"Headphone Jack", NULL, "HPROUT"}, @@ -90,36 +90,30 @@ static const char *audio_map[][3] = { {"LINE2L", NULL, "Line In"}, {"LINE1R", NULL, "Line In"}, {"LINE2R", NULL, "Line In"}, - - {NULL, NULL, NULL}, }; /* Logic for a aic3x as connected on a davinci-evm */ static int evm_aic3x_init(struct snd_soc_codec *codec) { - int i; - /* Add davinci-evm specific widgets */ - for (i = 0; i < ARRAY_SIZE(aic3x_dapm_widgets); i++) - snd_soc_dapm_new_control(codec, &aic3x_dapm_widgets[i]); + snd_soc_dapm_new_controls(codec, aic3x_dapm_widgets, + ARRAY_SIZE(aic3x_dapm_widgets)); /* Set up davinci-evm specific audio path audio_map */ - for (i = 0; audio_map[i][0] != NULL; i++) - snd_soc_dapm_connect_input(codec, audio_map[i][0], - audio_map[i][1], audio_map[i][2]); + snd_soc_dapm_add_routes(codec, audio_map, ARRAY_SIZE(audio_map)); /* not connected */ - snd_soc_dapm_set_endpoint(codec, "MONO_LOUT", 0); - snd_soc_dapm_set_endpoint(codec, "HPLCOM", 0); - snd_soc_dapm_set_endpoint(codec, "HPRCOM", 0); + snd_soc_dapm_disable_pin(codec, "MONO_LOUT"); + snd_soc_dapm_disable_pin(codec, "HPLCOM"); + snd_soc_dapm_disable_pin(codec, "HPRCOM"); /* always connected */ - snd_soc_dapm_set_endpoint(codec, "Headphone Jack", 1); - snd_soc_dapm_set_endpoint(codec, "Line Out", 1); - snd_soc_dapm_set_endpoint(codec, "Mic Jack", 1); - snd_soc_dapm_set_endpoint(codec, "Line In", 1); + snd_soc_dapm_enable_pin(codec, "Headphone Jack"); + snd_soc_dapm_enable_pin(codec, "Line Out"); + snd_soc_dapm_enable_pin(codec, "Mic Jack"); + snd_soc_dapm_enable_pin(codec, "Line In"); - snd_soc_dapm_sync_endpoints(codec); + snd_soc_dapm_sync(codec); return 0; } diff --git a/sound/soc/davinci/davinci-i2s.c b/sound/soc/davinci/davinci-i2s.c index c421774b33e..5ebf1ff71c4 100644 --- a/sound/soc/davinci/davinci-i2s.c +++ b/sound/soc/davinci/davinci-i2s.c @@ -147,7 +147,7 @@ static void davinci_mcbsp_stop(struct snd_pcm_substream *substream) static int davinci_i2s_startup(struct snd_pcm_substream *substream) { struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_cpu_dai *cpu_dai = rtd->dai->cpu_dai; + struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai; struct davinci_mcbsp_dev *dev = rtd->dai->cpu_dai->private_data; cpu_dai->dma_data = dev->dma_params[substream->stream]; @@ -155,7 +155,7 @@ static int davinci_i2s_startup(struct snd_pcm_substream *substream) return 0; } -static int davinci_i2s_set_dai_fmt(struct snd_soc_cpu_dai *cpu_dai, +static int davinci_i2s_set_dai_fmt(struct snd_soc_dai *cpu_dai, unsigned int fmt) { struct davinci_mcbsp_dev *dev = cpu_dai->private_data; @@ -295,11 +295,12 @@ static int davinci_i2s_trigger(struct snd_pcm_substream *substream, int cmd) return ret; } -static int davinci_i2s_probe(struct platform_device *pdev) +static int davinci_i2s_probe(struct platform_device *pdev, + struct snd_soc_dai *dai) { struct snd_soc_device *socdev = platform_get_drvdata(pdev); struct snd_soc_machine *machine = socdev->machine; - struct snd_soc_cpu_dai *cpu_dai = machine->dai_link[pdev->id].cpu_dai; + struct snd_soc_dai *cpu_dai = machine->dai_link[pdev->id].cpu_dai; struct davinci_mcbsp_dev *dev; struct resource *mem, *ioarea; struct evm_snd_platform_data *pdata; @@ -356,11 +357,12 @@ err_release_region: return ret; } -static void davinci_i2s_remove(struct platform_device *pdev) +static void davinci_i2s_remove(struct platform_device *pdev, + struct snd_soc_dai *dai) { struct snd_soc_device *socdev = platform_get_drvdata(pdev); struct snd_soc_machine *machine = socdev->machine; - struct snd_soc_cpu_dai *cpu_dai = machine->dai_link[pdev->id].cpu_dai; + struct snd_soc_dai *cpu_dai = machine->dai_link[pdev->id].cpu_dai; struct davinci_mcbsp_dev *dev = cpu_dai->private_data; struct resource *mem; @@ -376,7 +378,7 @@ static void davinci_i2s_remove(struct platform_device *pdev) #define DAVINCI_I2S_RATES SNDRV_PCM_RATE_8000_96000 -struct snd_soc_cpu_dai davinci_i2s_dai = { +struct snd_soc_dai davinci_i2s_dai = { .name = "davinci-i2s", .id = 0, .type = SND_SOC_DAI_I2S, diff --git a/sound/soc/davinci/davinci-i2s.h b/sound/soc/davinci/davinci-i2s.h index 9592d17db32..c5b091807ee 100644 --- a/sound/soc/davinci/davinci-i2s.h +++ b/sound/soc/davinci/davinci-i2s.h @@ -12,6 +12,6 @@ #ifndef _DAVINCI_I2S_H #define _DAVINCI_I2S_H -extern struct snd_soc_cpu_dai davinci_i2s_dai; +extern struct snd_soc_dai davinci_i2s_dai; #endif diff --git a/sound/soc/davinci/davinci-pcm.c b/sound/soc/davinci/davinci-pcm.c index 6a76927c997..6a5e56a782b 100644 --- a/sound/soc/davinci/davinci-pcm.c +++ b/sound/soc/davinci/davinci-pcm.c @@ -350,7 +350,7 @@ static void davinci_pcm_free(struct snd_pcm *pcm) static u64 davinci_pcm_dmamask = 0xffffffff; static int davinci_pcm_new(struct snd_card *card, - struct snd_soc_codec_dai *dai, struct snd_pcm *pcm) + struct snd_soc_dai *dai, struct snd_pcm *pcm) { int ret; diff --git a/sound/soc/fsl/Kconfig b/sound/soc/fsl/Kconfig index 257101f44e9..3368ace6097 100644 --- a/sound/soc/fsl/Kconfig +++ b/sound/soc/fsl/Kconfig @@ -1,8 +1,6 @@ -menu "ALSA SoC audio for Freescale SOCs" - config SND_SOC_MPC8610 bool "ALSA SoC support for the MPC8610 SOC" - depends on SND_SOC && MPC8610_HPCD + depends on MPC8610_HPCD default y if MPC8610 help Say Y if you want to add support for codecs attached to the SSI @@ -16,5 +14,3 @@ config SND_SOC_MPC8610_HPCD default y if MPC8610_HPCD help Say Y if you want to enable audio on the Freescale MPC8610 HPCD. - -endmenu diff --git a/sound/soc/fsl/fsl_dma.c b/sound/soc/fsl/fsl_dma.c index 78de7168d2b..da2bc590286 100644 --- a/sound/soc/fsl/fsl_dma.c +++ b/sound/soc/fsl/fsl_dma.c @@ -282,7 +282,7 @@ static irqreturn_t fsl_dma_isr(int irq, void *dev_id) * once for each .dai_link in the machine driver's snd_soc_machine * structure. */ -static int fsl_dma_new(struct snd_card *card, struct snd_soc_codec_dai *dai, +static int fsl_dma_new(struct snd_card *card, struct snd_soc_dai *dai, struct snd_pcm *pcm) { static u64 fsl_dma_dmamask = DMA_BIT_MASK(32); diff --git a/sound/soc/fsl/fsl_dma.h b/sound/soc/fsl/fsl_dma.h index 430a6ce8b0d..385d4a42603 100644 --- a/sound/soc/fsl/fsl_dma.h +++ b/sound/soc/fsl/fsl_dma.h @@ -126,7 +126,7 @@ struct fsl_dma_link_descriptor { u8 res[4]; /* Reserved */ } __attribute__ ((aligned(32), packed)); -/* DMA information needed to create a snd_soc_cpu_dai object +/* DMA information needed to create a snd_soc_dai object * * ssi_stx_phys: bus address of SSI STX register to use * ssi_srx_phys: bus address of SSI SRX register to use diff --git a/sound/soc/fsl/fsl_ssi.c b/sound/soc/fsl/fsl_ssi.c index f588545698f..71bff33f552 100644 --- a/sound/soc/fsl/fsl_ssi.c +++ b/sound/soc/fsl/fsl_ssi.c @@ -82,7 +82,7 @@ struct fsl_ssi_private { struct device *dev; unsigned int playback; unsigned int capture; - struct snd_soc_cpu_dai cpu_dai; + struct snd_soc_dai cpu_dai; struct device_attribute dev_attr; struct { @@ -479,7 +479,7 @@ static void fsl_ssi_shutdown(struct snd_pcm_substream *substream) * @freq: the frequency of the given clock ID, currently ignored * @dir: SND_SOC_CLOCK_IN (clock slave) or SND_SOC_CLOCK_OUT (clock master) */ -static int fsl_ssi_set_sysclk(struct snd_soc_cpu_dai *cpu_dai, +static int fsl_ssi_set_sysclk(struct snd_soc_dai *cpu_dai, int clk_id, unsigned int freq, int dir) { @@ -497,7 +497,7 @@ static int fsl_ssi_set_sysclk(struct snd_soc_cpu_dai *cpu_dai, * * @format: one of SND_SOC_DAIFMT_xxx */ -static int fsl_ssi_set_fmt(struct snd_soc_cpu_dai *cpu_dai, unsigned int format) +static int fsl_ssi_set_fmt(struct snd_soc_dai *cpu_dai, unsigned int format) { return (format == SND_SOC_DAIFMT_I2S) ? 0 : -EINVAL; } @@ -505,7 +505,7 @@ static int fsl_ssi_set_fmt(struct snd_soc_cpu_dai *cpu_dai, unsigned int format) /** * fsl_ssi_dai_template: template CPU DAI for the SSI */ -static struct snd_soc_cpu_dai fsl_ssi_dai_template = { +static struct snd_soc_dai fsl_ssi_dai_template = { .playback = { /* The SSI does not support monaural audio. */ .channels_min = 2, @@ -569,15 +569,15 @@ static ssize_t fsl_sysfs_ssi_show(struct device *dev, } /** - * fsl_ssi_create_dai: create a snd_soc_cpu_dai structure + * fsl_ssi_create_dai: create a snd_soc_dai structure * - * This function is called by the machine driver to create a snd_soc_cpu_dai + * This function is called by the machine driver to create a snd_soc_dai * structure. The function creates an ssi_private object, which contains - * the snd_soc_cpu_dai. It also creates the sysfs statistics device. + * the snd_soc_dai. It also creates the sysfs statistics device. */ -struct snd_soc_cpu_dai *fsl_ssi_create_dai(struct fsl_ssi_info *ssi_info) +struct snd_soc_dai *fsl_ssi_create_dai(struct fsl_ssi_info *ssi_info) { - struct snd_soc_cpu_dai *fsl_ssi_dai; + struct snd_soc_dai *fsl_ssi_dai; struct fsl_ssi_private *ssi_private; int ret = 0; struct device_attribute *dev_attr; @@ -588,7 +588,7 @@ struct snd_soc_cpu_dai *fsl_ssi_create_dai(struct fsl_ssi_info *ssi_info) return NULL; } memcpy(&ssi_private->cpu_dai, &fsl_ssi_dai_template, - sizeof(struct snd_soc_cpu_dai)); + sizeof(struct snd_soc_dai)); fsl_ssi_dai = &ssi_private->cpu_dai; dev_attr = &ssi_private->dev_attr; @@ -623,11 +623,11 @@ struct snd_soc_cpu_dai *fsl_ssi_create_dai(struct fsl_ssi_info *ssi_info) EXPORT_SYMBOL_GPL(fsl_ssi_create_dai); /** - * fsl_ssi_destroy_dai: destroy the snd_soc_cpu_dai object + * fsl_ssi_destroy_dai: destroy the snd_soc_dai object * * This function undoes the operations of fsl_ssi_create_dai() */ -void fsl_ssi_destroy_dai(struct snd_soc_cpu_dai *fsl_ssi_dai) +void fsl_ssi_destroy_dai(struct snd_soc_dai *fsl_ssi_dai) { struct fsl_ssi_private *ssi_private = container_of(fsl_ssi_dai, struct fsl_ssi_private, cpu_dai); diff --git a/sound/soc/fsl/fsl_ssi.h b/sound/soc/fsl/fsl_ssi.h index c5ce88e1565..83b44d700e3 100644 --- a/sound/soc/fsl/fsl_ssi.h +++ b/sound/soc/fsl/fsl_ssi.h @@ -217,8 +217,8 @@ struct fsl_ssi_info { struct device *dev; }; -struct snd_soc_cpu_dai *fsl_ssi_create_dai(struct fsl_ssi_info *ssi_info); -void fsl_ssi_destroy_dai(struct snd_soc_cpu_dai *fsl_ssi_dai); +struct snd_soc_dai *fsl_ssi_create_dai(struct fsl_ssi_info *ssi_info); +void fsl_ssi_destroy_dai(struct snd_soc_dai *fsl_ssi_dai); #endif diff --git a/sound/soc/fsl/mpc8610_hpcd.c b/sound/soc/fsl/mpc8610_hpcd.c index a00aac7a71f..4bdc9d8fc90 100644 --- a/sound/soc/fsl/mpc8610_hpcd.c +++ b/sound/soc/fsl/mpc8610_hpcd.c @@ -58,9 +58,9 @@ static int mpc8610_hpcd_machine_probe(struct platform_device *sound_device) sound_device->dev.platform_data; /* Program the signal routing between the SSI and the DMA */ - guts_set_dmacr(machine_data->guts, machine_data->dma_id + 1, + guts_set_dmacr(machine_data->guts, machine_data->dma_id, machine_data->dma_channel_id[0], CCSR_GUTS_DMACR_DEV_SSI); - guts_set_dmacr(machine_data->guts, machine_data->dma_id + 1, + guts_set_dmacr(machine_data->guts, machine_data->dma_id, machine_data->dma_channel_id[1], CCSR_GUTS_DMACR_DEV_SSI); guts_set_pmuxcr_dma(machine_data->guts, machine_data->dma_id, @@ -96,62 +96,52 @@ static int mpc8610_hpcd_machine_probe(struct platform_device *sound_device) static int mpc8610_hpcd_startup(struct snd_pcm_substream *substream) { struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_codec_dai *codec_dai = rtd->dai->codec_dai; - struct snd_soc_cpu_dai *cpu_dai = rtd->dai->cpu_dai; + struct snd_soc_dai *codec_dai = rtd->dai->codec_dai; + struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai; struct mpc8610_hpcd_data *machine_data = rtd->socdev->dev->platform_data; int ret = 0; /* Tell the CPU driver what the serial protocol is. */ - if (cpu_dai->dai_ops.set_fmt) { - ret = cpu_dai->dai_ops.set_fmt(cpu_dai, - machine_data->dai_format); - if (ret < 0) { - dev_err(substream->pcm->card->dev, - "could not set CPU driver audio format\n"); - return ret; - } + ret = snd_soc_dai_set_fmt(cpu_dai, machine_data->dai_format); + if (ret < 0) { + dev_err(substream->pcm->card->dev, + "could not set CPU driver audio format\n"); + return ret; } /* Tell the codec driver what the serial protocol is. */ - if (codec_dai->dai_ops.set_fmt) { - ret = codec_dai->dai_ops.set_fmt(codec_dai, - machine_data->dai_format); - if (ret < 0) { - dev_err(substream->pcm->card->dev, - "could not set codec driver audio format\n"); - return ret; - } + ret = snd_soc_dai_set_fmt(codec_dai, machine_data->dai_format); + if (ret < 0) { + dev_err(substream->pcm->card->dev, + "could not set codec driver audio format\n"); + return ret; } /* * Tell the CPU driver what the clock frequency is, and whether it's a * slave or master. */ - if (cpu_dai->dai_ops.set_sysclk) { - ret = cpu_dai->dai_ops.set_sysclk(cpu_dai, 0, - machine_data->clk_frequency, - machine_data->cpu_clk_direction); - if (ret < 0) { - dev_err(substream->pcm->card->dev, - "could not set CPU driver clock parameters\n"); - return ret; - } + ret = snd_soc_dai_set_sysclk(cpu_dai, 0, + machine_data->clk_frequency, + machine_data->cpu_clk_direction); + if (ret < 0) { + dev_err(substream->pcm->card->dev, + "could not set CPU driver clock parameters\n"); + return ret; } /* * Tell the codec driver what the MCLK frequency is, and whether it's * a slave or master. */ - if (codec_dai->dai_ops.set_sysclk) { - ret = codec_dai->dai_ops.set_sysclk(codec_dai, 0, - machine_data->clk_frequency, - machine_data->codec_clk_direction); - if (ret < 0) { - dev_err(substream->pcm->card->dev, - "could not set codec driver clock params\n"); - return ret; - } + ret = snd_soc_dai_set_sysclk(codec_dai, 0, + machine_data->clk_frequency, + machine_data->codec_clk_direction); + if (ret < 0) { + dev_err(substream->pcm->card->dev, + "could not set codec driver clock params\n"); + return ret; } return 0; @@ -170,9 +160,9 @@ int mpc8610_hpcd_machine_remove(struct platform_device *sound_device) /* Restore the signal routing */ - guts_set_dmacr(machine_data->guts, machine_data->dma_id + 1, + guts_set_dmacr(machine_data->guts, machine_data->dma_id, machine_data->dma_channel_id[0], 0); - guts_set_dmacr(machine_data->guts, machine_data->dma_id + 1, + guts_set_dmacr(machine_data->guts, machine_data->dma_id, machine_data->dma_channel_id[1], 0); switch (machine_data->ssi_id) { @@ -182,7 +172,7 @@ int mpc8610_hpcd_machine_remove(struct platform_device *sound_device) break; case 1: clrsetbits_be32(&machine_data->guts->pmuxcr, - CCSR_GUTS_PMUXCR_SSI2_MASK, CCSR_GUTS_PMUXCR_SSI1_LA); + CCSR_GUTS_PMUXCR_SSI2_MASK, CCSR_GUTS_PMUXCR_SSI2_LA); break; } diff --git a/sound/soc/omap/Kconfig b/sound/soc/omap/Kconfig index 0230d83e8e5..aea27e70043 100644 --- a/sound/soc/omap/Kconfig +++ b/sound/soc/omap/Kconfig @@ -1,5 +1,3 @@ -menu "SoC Audio for the Texas Instruments OMAP" - config SND_OMAP_SOC tristate "SoC Audio for the Texas Instruments OMAP chips" depends on ARCH_OMAP && SND_SOC @@ -15,5 +13,3 @@ config SND_OMAP_SOC_N810 select SND_SOC_TLV320AIC3X help Say Y if you want to add support for SoC audio on Nokia N810. - -endmenu diff --git a/sound/soc/omap/n810.c b/sound/soc/omap/n810.c index 6533563a601..02cec96859b 100644 --- a/sound/soc/omap/n810.c +++ b/sound/soc/omap/n810.c @@ -30,15 +30,15 @@ #include <asm/mach-types.h> #include <asm/arch/hardware.h> -#include <asm/arch/gpio.h> +#include <linux/gpio.h> #include <asm/arch/mcbsp.h> #include "omap-mcbsp.h" #include "omap-pcm.h" #include "../codecs/tlv320aic3x.h" -#define RX44_HEADSET_AMP_GPIO 10 -#define RX44_SPEAKER_AMP_GPIO 101 +#define N810_HEADSET_AMP_GPIO 10 +#define N810_SPEAKER_AMP_GPIO 101 static struct clk *sys_clkout2; static struct clk *sys_clkout2_src; @@ -46,13 +46,26 @@ static struct clk *func96m_clk; static int n810_spk_func; static int n810_jack_func; +static int n810_dmic_func; static void n810_ext_control(struct snd_soc_codec *codec) { - snd_soc_dapm_set_endpoint(codec, "Ext Spk", n810_spk_func); - snd_soc_dapm_set_endpoint(codec, "Headphone Jack", n810_jack_func); + if (n810_spk_func) + snd_soc_dapm_enable_pin(codec, "Ext Spk"); + else + snd_soc_dapm_disable_pin(codec, "Ext Spk"); + + if (n810_jack_func) + snd_soc_dapm_enable_pin(codec, "Headphone Jack"); + else + snd_soc_dapm_disable_pin(codec, "Headphone Jack"); - snd_soc_dapm_sync_endpoints(codec); + if (n810_dmic_func) + snd_soc_dapm_enable_pin(codec, "DMic"); + else + snd_soc_dapm_disable_pin(codec, "DMic"); + + snd_soc_dapm_sync(codec); } static int n810_startup(struct snd_pcm_substream *substream) @@ -73,12 +86,12 @@ static int n810_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params) { struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_codec_dai *codec_dai = rtd->dai->codec_dai; - struct snd_soc_cpu_dai *cpu_dai = rtd->dai->cpu_dai; + struct snd_soc_dai *codec_dai = rtd->dai->codec_dai; + struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai; int err; /* Set codec DAI configuration */ - err = codec_dai->dai_ops.set_fmt(codec_dai, + err = snd_soc_dai_set_fmt(codec_dai, SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBM_CFM); @@ -86,7 +99,7 @@ static int n810_hw_params(struct snd_pcm_substream *substream, return err; /* Set cpu DAI configuration */ - err = cpu_dai->dai_ops.set_fmt(cpu_dai, + err = snd_soc_dai_set_fmt(cpu_dai, SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBM_CFM); @@ -94,7 +107,7 @@ static int n810_hw_params(struct snd_pcm_substream *substream, return err; /* Set the codec system clock for DAC and ADC */ - err = codec_dai->dai_ops.set_sysclk(codec_dai, 0, 12000000, + err = snd_soc_dai_set_sysclk(codec_dai, 0, 12000000, SND_SOC_CLOCK_IN); return err; @@ -150,13 +163,35 @@ static int n810_set_jack(struct snd_kcontrol *kcontrol, return 1; } +static int n810_get_input(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + ucontrol->value.integer.value[0] = n810_dmic_func; + + return 0; +} + +static int n810_set_input(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); + + if (n810_dmic_func == ucontrol->value.integer.value[0]) + return 0; + + n810_dmic_func = ucontrol->value.integer.value[0]; + n810_ext_control(codec); + + return 1; +} + static int n810_spk_event(struct snd_soc_dapm_widget *w, struct snd_kcontrol *k, int event) { if (SND_SOC_DAPM_EVENT_ON(event)) - omap_set_gpio_dataout(RX44_SPEAKER_AMP_GPIO, 1); + gpio_set_value(N810_SPEAKER_AMP_GPIO, 1); else - omap_set_gpio_dataout(RX44_SPEAKER_AMP_GPIO, 0); + gpio_set_value(N810_SPEAKER_AMP_GPIO, 0); return 0; } @@ -165,9 +200,9 @@ static int n810_jack_event(struct snd_soc_dapm_widget *w, struct snd_kcontrol *k, int event) { if (SND_SOC_DAPM_EVENT_ON(event)) - omap_set_gpio_dataout(RX44_HEADSET_AMP_GPIO, 1); + gpio_set_value(N810_HEADSET_AMP_GPIO, 1); else - omap_set_gpio_dataout(RX44_HEADSET_AMP_GPIO, 0); + gpio_set_value(N810_HEADSET_AMP_GPIO, 0); return 0; } @@ -175,21 +210,27 @@ static int n810_jack_event(struct snd_soc_dapm_widget *w, static const struct snd_soc_dapm_widget aic33_dapm_widgets[] = { SND_SOC_DAPM_SPK("Ext Spk", n810_spk_event), SND_SOC_DAPM_HP("Headphone Jack", n810_jack_event), + SND_SOC_DAPM_MIC("DMic", NULL), }; -static const char *audio_map[][3] = { +static const struct snd_soc_dapm_route audio_map[] = { {"Headphone Jack", NULL, "HPLOUT"}, {"Headphone Jack", NULL, "HPROUT"}, {"Ext Spk", NULL, "LLOUT"}, {"Ext Spk", NULL, "RLOUT"}, + + {"DMic Rate 64", NULL, "Mic Bias 2V"}, + {"Mic Bias 2V", NULL, "DMic"}, }; static const char *spk_function[] = {"Off", "On"}; static const char *jack_function[] = {"Off", "Headphone"}; +static const char *input_function[] = {"ADC", "Digital Mic"}; static const struct soc_enum n810_enum[] = { SOC_ENUM_SINGLE_EXT(ARRAY_SIZE(spk_function), spk_function), SOC_ENUM_SINGLE_EXT(ARRAY_SIZE(jack_function), jack_function), + SOC_ENUM_SINGLE_EXT(ARRAY_SIZE(input_function), input_function), }; static const struct snd_kcontrol_new aic33_n810_controls[] = { @@ -197,6 +238,8 @@ static const struct snd_kcontrol_new aic33_n810_controls[] = { n810_get_spk, n810_set_spk), SOC_ENUM_EXT("Jack Function", n810_enum[1], n810_get_jack, n810_set_jack), + SOC_ENUM_EXT("Input Select", n810_enum[2], + n810_get_input, n810_set_input), }; static int n810_aic33_init(struct snd_soc_codec *codec) @@ -204,9 +247,9 @@ static int n810_aic33_init(struct snd_soc_codec *codec) int i, err; /* Not connected */ - snd_soc_dapm_set_endpoint(codec, "MONO_LOUT", 0); - snd_soc_dapm_set_endpoint(codec, "HPLCOM", 0); - snd_soc_dapm_set_endpoint(codec, "HPRCOM", 0); + snd_soc_dapm_disable_pin(codec, "MONO_LOUT"); + snd_soc_dapm_disable_pin(codec, "HPLCOM"); + snd_soc_dapm_disable_pin(codec, "HPRCOM"); /* Add N810 specific controls */ for (i = 0; i < ARRAY_SIZE(aic33_n810_controls); i++) { @@ -217,15 +260,13 @@ static int n810_aic33_init(struct snd_soc_codec *codec) } /* Add N810 specific widgets */ - for (i = 0; i < ARRAY_SIZE(aic33_dapm_widgets); i++) - snd_soc_dapm_new_control(codec, &aic33_dapm_widgets[i]); + snd_soc_dapm_new_controls(codec, aic33_dapm_widgets, + ARRAY_SIZE(aic33_dapm_widgets)); /* Set up N810 specific audio path audio_map */ - for (i = 0; i < ARRAY_SIZE(audio_map); i++) - snd_soc_dapm_connect_input(codec, audio_map[i][0], - audio_map[i][1], audio_map[i][2]); + snd_soc_dapm_add_routes(codec, audio_map, ARRAY_SIZE(audio_map)); - snd_soc_dapm_sync_endpoints(codec); + snd_soc_dapm_sync(codec); return 0; } @@ -250,6 +291,8 @@ static struct snd_soc_machine snd_soc_machine_n810 = { /* Audio private data */ static struct aic3x_setup_data n810_aic33_setup = { .i2c_address = 0x18, + .gpio_func[0] = AIC3X_GPIO1_FUNC_DISABLED, + .gpio_func[1] = AIC3X_GPIO2_FUNC_DIGITAL_MIC_INPUT, }; /* Audio subsystem */ @@ -267,7 +310,7 @@ static int __init n810_soc_init(void) int err; struct device *dev; - if (!machine_is_nokia_n810()) + if (!(machine_is_nokia_n810() || machine_is_nokia_n810_wimax())) return -ENODEV; n810_snd_device = platform_device_alloc("soc-audio", -1); @@ -305,12 +348,12 @@ static int __init n810_soc_init(void) clk_set_parent(sys_clkout2_src, func96m_clk); clk_set_rate(sys_clkout2, 12000000); - if (omap_request_gpio(RX44_HEADSET_AMP_GPIO) < 0) + if (gpio_request(N810_HEADSET_AMP_GPIO, "hs_amp") < 0) BUG(); - if (omap_request_gpio(RX44_SPEAKER_AMP_GPIO) < 0) + if (gpio_request(N810_SPEAKER_AMP_GPIO, "spk_amp") < 0) BUG(); - omap_set_gpio_direction(RX44_HEADSET_AMP_GPIO, 0); - omap_set_gpio_direction(RX44_SPEAKER_AMP_GPIO, 0); + gpio_direction_output(N810_HEADSET_AMP_GPIO, 0); + gpio_direction_output(N810_SPEAKER_AMP_GPIO, 0); return 0; err2: @@ -325,6 +368,9 @@ err1: static void __exit n810_soc_exit(void) { + gpio_free(N810_SPEAKER_AMP_GPIO); + gpio_free(N810_HEADSET_AMP_GPIO); + platform_device_unregister(n810_snd_device); } diff --git a/sound/soc/omap/omap-mcbsp.c b/sound/soc/omap/omap-mcbsp.c index 40d87e6d0de..00b0c9d73cd 100644 --- a/sound/soc/omap/omap-mcbsp.c +++ b/sound/soc/omap/omap-mcbsp.c @@ -103,7 +103,7 @@ static const unsigned long omap2420_mcbsp_port[][2] = {}; static int omap_mcbsp_dai_startup(struct snd_pcm_substream *substream) { struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_cpu_dai *cpu_dai = rtd->dai->cpu_dai; + struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai; struct omap_mcbsp_data *mcbsp_data = to_mcbsp(cpu_dai->private_data); int err = 0; @@ -116,7 +116,7 @@ static int omap_mcbsp_dai_startup(struct snd_pcm_substream *substream) static void omap_mcbsp_dai_shutdown(struct snd_pcm_substream *substream) { struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_cpu_dai *cpu_dai = rtd->dai->cpu_dai; + struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai; struct omap_mcbsp_data *mcbsp_data = to_mcbsp(cpu_dai->private_data); if (!cpu_dai->active) { @@ -128,7 +128,7 @@ static void omap_mcbsp_dai_shutdown(struct snd_pcm_substream *substream) static int omap_mcbsp_dai_trigger(struct snd_pcm_substream *substream, int cmd) { struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_cpu_dai *cpu_dai = rtd->dai->cpu_dai; + struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai; struct omap_mcbsp_data *mcbsp_data = to_mcbsp(cpu_dai->private_data); int err = 0; @@ -157,7 +157,7 @@ static int omap_mcbsp_dai_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params) { struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_cpu_dai *cpu_dai = rtd->dai->cpu_dai; + struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai; struct omap_mcbsp_data *mcbsp_data = to_mcbsp(cpu_dai->private_data); struct omap_mcbsp_reg_cfg *regs = &mcbsp_data->regs; int dma, bus_id = mcbsp_data->bus_id, id = cpu_dai->id; @@ -223,7 +223,7 @@ static int omap_mcbsp_dai_hw_params(struct snd_pcm_substream *substream, * This must be called before _set_clkdiv and _set_sysclk since McBSP register * cache is initialized here */ -static int omap_mcbsp_dai_set_dai_fmt(struct snd_soc_cpu_dai *cpu_dai, +static int omap_mcbsp_dai_set_dai_fmt(struct snd_soc_dai *cpu_dai, unsigned int fmt) { struct omap_mcbsp_data *mcbsp_data = to_mcbsp(cpu_dai->private_data); @@ -292,7 +292,7 @@ static int omap_mcbsp_dai_set_dai_fmt(struct snd_soc_cpu_dai *cpu_dai, return 0; } -static int omap_mcbsp_dai_set_clkdiv(struct snd_soc_cpu_dai *cpu_dai, +static int omap_mcbsp_dai_set_clkdiv(struct snd_soc_dai *cpu_dai, int div_id, int div) { struct omap_mcbsp_data *mcbsp_data = to_mcbsp(cpu_dai->private_data); @@ -347,7 +347,7 @@ static int omap_mcbsp_dai_set_clks_src(struct omap_mcbsp_data *mcbsp_data, return 0; } -static int omap_mcbsp_dai_set_dai_sysclk(struct snd_soc_cpu_dai *cpu_dai, +static int omap_mcbsp_dai_set_dai_sysclk(struct snd_soc_dai *cpu_dai, int clk_id, unsigned int freq, int dir) { @@ -376,7 +376,7 @@ static int omap_mcbsp_dai_set_dai_sysclk(struct snd_soc_cpu_dai *cpu_dai, return err; } -struct snd_soc_cpu_dai omap_mcbsp_dai[NUM_LINKS] = { +struct snd_soc_dai omap_mcbsp_dai[NUM_LINKS] = { { .name = "omap-mcbsp-dai", .id = 0, diff --git a/sound/soc/omap/omap-mcbsp.h b/sound/soc/omap/omap-mcbsp.h index 9965fd4b042..ed8afb55067 100644 --- a/sound/soc/omap/omap-mcbsp.h +++ b/sound/soc/omap/omap-mcbsp.h @@ -44,6 +44,6 @@ enum omap_mcbsp_div { */ #define NUM_LINKS 1 -extern struct snd_soc_cpu_dai omap_mcbsp_dai[NUM_LINKS]; +extern struct snd_soc_dai omap_mcbsp_dai[NUM_LINKS]; #endif diff --git a/sound/soc/omap/omap-pcm.c b/sound/soc/omap/omap-pcm.c index 62370202c64..e092f3d836d 100644 --- a/sound/soc/omap/omap-pcm.c +++ b/sound/soc/omap/omap-pcm.c @@ -316,7 +316,7 @@ static void omap_pcm_free_dma_buffers(struct snd_pcm *pcm) } } -int omap_pcm_new(struct snd_card *card, struct snd_soc_codec_dai *dai, +int omap_pcm_new(struct snd_card *card, struct snd_soc_dai *dai, struct snd_pcm *pcm) { int ret = 0; diff --git a/sound/soc/pxa/Kconfig b/sound/soc/pxa/Kconfig index 484f883459e..12f6ac99b04 100644 --- a/sound/soc/pxa/Kconfig +++ b/sound/soc/pxa/Kconfig @@ -1,6 +1,6 @@ config SND_PXA2XX_SOC tristate "SoC Audio for the Intel PXA2xx chip" - depends on ARCH_PXA && SND_SOC + depends on ARCH_PXA help Say Y or M if you want to add support for codecs attached to the PXA2xx AC97, I2S or SSP interface. You will also need @@ -62,3 +62,12 @@ config SND_PXA2XX_SOC_E800 help Say Y if you want to add support for SoC audio on the Toshiba e800 PDA + +config SND_PXA2XX_SOC_EM_X270 + tristate "SoC Audio support for CompuLab EM-x270" + depends on SND_PXA2XX_SOC && MACH_EM_X270 + select SND_PXA2XX_SOC_AC97 + select SND_SOC_WM9712 + help + Say Y if you want to add support for SoC audio on + CompuLab EM-x270. diff --git a/sound/soc/pxa/Makefile b/sound/soc/pxa/Makefile index 04e5646f75b..5bc8edf9dca 100644 --- a/sound/soc/pxa/Makefile +++ b/sound/soc/pxa/Makefile @@ -13,10 +13,11 @@ snd-soc-poodle-objs := poodle.o snd-soc-tosa-objs := tosa.o snd-soc-e800-objs := e800_wm9712.o snd-soc-spitz-objs := spitz.o +snd-soc-em-x270-objs := em-x270.o obj-$(CONFIG_SND_PXA2XX_SOC_CORGI) += snd-soc-corgi.o obj-$(CONFIG_SND_PXA2XX_SOC_POODLE) += snd-soc-poodle.o obj-$(CONFIG_SND_PXA2XX_SOC_TOSA) += snd-soc-tosa.o obj-$(CONFIG_SND_PXA2XX_SOC_E800) += snd-soc-e800.o obj-$(CONFIG_SND_PXA2XX_SOC_SPITZ) += snd-soc-spitz.o - +obj-$(CONFIG_SND_PXA2XX_SOC_EM_X270) += snd-soc-em-x270.o diff --git a/sound/soc/pxa/corgi.c b/sound/soc/pxa/corgi.c index 7f32a116757..c0294464a23 100644 --- a/sound/soc/pxa/corgi.c +++ b/sound/soc/pxa/corgi.c @@ -11,10 +11,6 @@ * under the terms of the GNU General Public License as published by the * Free Software Foundation; either version 2 of the License, or (at your * option) any later version. - * - * Revision history - * 30th Nov 2005 Initial version. - * */ #include <linux/module.h> @@ -54,47 +50,51 @@ static int corgi_spk_func; static void corgi_ext_control(struct snd_soc_codec *codec) { - int spk = 0, mic = 0, line = 0, hp = 0, hs = 0; - /* set up jack connection */ switch (corgi_jack_func) { case CORGI_HP: - hp = 1; /* set = unmute headphone */ set_scoop_gpio(&corgiscoop_device.dev, CORGI_SCP_MUTE_L); set_scoop_gpio(&corgiscoop_device.dev, CORGI_SCP_MUTE_R); + snd_soc_dapm_disable_pin(codec, "Mic Jack"); + snd_soc_dapm_disable_pin(codec, "Line Jack"); + snd_soc_dapm_enable_pin(codec, "Headphone Jack"); + snd_soc_dapm_disable_pin(codec, "Headset Jack"); break; case CORGI_MIC: - mic = 1; /* reset = mute headphone */ reset_scoop_gpio(&corgiscoop_device.dev, CORGI_SCP_MUTE_L); reset_scoop_gpio(&corgiscoop_device.dev, CORGI_SCP_MUTE_R); + snd_soc_dapm_enable_pin(codec, "Mic Jack"); + snd_soc_dapm_disable_pin(codec, "Line Jack"); + snd_soc_dapm_disable_pin(codec, "Headphone Jack"); + snd_soc_dapm_disable_pin(codec, "Headset Jack"); break; case CORGI_LINE: - line = 1; reset_scoop_gpio(&corgiscoop_device.dev, CORGI_SCP_MUTE_L); reset_scoop_gpio(&corgiscoop_device.dev, CORGI_SCP_MUTE_R); + snd_soc_dapm_disable_pin(codec, "Mic Jack"); + snd_soc_dapm_enable_pin(codec, "Line Jack"); + snd_soc_dapm_disable_pin(codec, "Headphone Jack"); + snd_soc_dapm_disable_pin(codec, "Headset Jack"); break; case CORGI_HEADSET: - hs = 1; - mic = 1; reset_scoop_gpio(&corgiscoop_device.dev, CORGI_SCP_MUTE_L); set_scoop_gpio(&corgiscoop_device.dev, CORGI_SCP_MUTE_R); + snd_soc_dapm_enable_pin(codec, "Mic Jack"); + snd_soc_dapm_disable_pin(codec, "Line Jack"); + snd_soc_dapm_disable_pin(codec, "Headphone Jack"); + snd_soc_dapm_enable_pin(codec, "Headset Jack"); break; } if (corgi_spk_func == CORGI_SPK_ON) - spk = 1; - - /* set the enpoints to their new connetion states */ - snd_soc_dapm_set_endpoint(codec, "Ext Spk", spk); - snd_soc_dapm_set_endpoint(codec, "Mic Jack", mic); - snd_soc_dapm_set_endpoint(codec, "Line Jack", line); - snd_soc_dapm_set_endpoint(codec, "Headphone Jack", hp); - snd_soc_dapm_set_endpoint(codec, "Headset Jack", hs); + snd_soc_dapm_enable_pin(codec, "Ext Spk"); + else + snd_soc_dapm_disable_pin(codec, "Ext Spk"); /* signal a DAPM event */ - snd_soc_dapm_sync_endpoints(codec); + snd_soc_dapm_sync(codec); } static int corgi_startup(struct snd_pcm_substream *substream) @@ -123,8 +123,8 @@ static int corgi_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params) { struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_codec_dai *codec_dai = rtd->dai->codec_dai; - struct snd_soc_cpu_dai *cpu_dai = rtd->dai->cpu_dai; + struct snd_soc_dai *codec_dai = rtd->dai->codec_dai; + struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai; unsigned int clk = 0; int ret = 0; @@ -143,25 +143,25 @@ static int corgi_hw_params(struct snd_pcm_substream *substream, } /* set codec DAI configuration */ - ret = codec_dai->dai_ops.set_fmt(codec_dai, SND_SOC_DAIFMT_I2S | + ret = snd_soc_dai_set_fmt(codec_dai, SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBS_CFS); if (ret < 0) return ret; /* set cpu DAI configuration */ - ret = cpu_dai->dai_ops.set_fmt(cpu_dai, SND_SOC_DAIFMT_I2S | + ret = snd_soc_dai_set_fmt(cpu_dai, SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBS_CFS); if (ret < 0) return ret; /* set the codec system clock for DAC and ADC */ - ret = codec_dai->dai_ops.set_sysclk(codec_dai, WM8731_SYSCLK, clk, + ret = snd_soc_dai_set_sysclk(codec_dai, WM8731_SYSCLK, clk, SND_SOC_CLOCK_IN); if (ret < 0) return ret; /* set the I2S system clock as input (unused) */ - ret = cpu_dai->dai_ops.set_sysclk(cpu_dai, PXA2XX_I2S_SYSCLK, 0, + ret = snd_soc_dai_set_sysclk(cpu_dai, PXA2XX_I2S_SYSCLK, 0, SND_SOC_CLOCK_IN); if (ret < 0) return ret; @@ -247,7 +247,7 @@ SND_SOC_DAPM_HP("Headset Jack", NULL), }; /* Corgi machine audio map (connections to the codec pins) */ -static const char *audio_map[][3] = { +static const struct snd_soc_dapm_route audio_map[] = { /* headset Jack - in = micin, out = LHPOUT*/ {"Headset Jack", NULL, "LHPOUT"}, @@ -265,8 +265,6 @@ static const char *audio_map[][3] = { /* Same as the above but no mic bias for line signals */ {"MICIN", NULL, "Line Jack"}, - - {NULL, NULL, NULL}, }; static const char *jack_function[] = {"Headphone", "Mic", "Line", "Headset", @@ -291,8 +289,8 @@ static int corgi_wm8731_init(struct snd_soc_codec *codec) { int i, err; - snd_soc_dapm_set_endpoint(codec, "LLINEIN", 0); - snd_soc_dapm_set_endpoint(codec, "RLINEIN", 0); + snd_soc_dapm_disable_pin(codec, "LLINEIN"); + snd_soc_dapm_disable_pin(codec, "RLINEIN"); /* Add corgi specific controls */ for (i = 0; i < ARRAY_SIZE(wm8731_corgi_controls); i++) { @@ -303,15 +301,13 @@ static int corgi_wm8731_init(struct snd_soc_codec *codec) } /* Add corgi specific widgets */ - for (i = 0; i < ARRAY_SIZE(wm8731_dapm_widgets); i++) - snd_soc_dapm_new_control(codec, &wm8731_dapm_widgets[i]); + snd_soc_dapm_new_controls(codec, wm8731_dapm_widgets, + ARRAY_SIZE(wm8731_dapm_widgets)); /* Set up corgi specific audio path audio_map */ - for (i = 0; audio_map[i][0] != NULL; i++) - snd_soc_dapm_connect_input(codec, audio_map[i][0], - audio_map[i][1], audio_map[i][2]); + snd_soc_dapm_add_routes(codec, audio_map, ARRAY_SIZE(audio_map)); - snd_soc_dapm_sync_endpoints(codec); + snd_soc_dapm_sync(codec); return 0; } diff --git a/sound/soc/pxa/em-x270.c b/sound/soc/pxa/em-x270.c new file mode 100644 index 00000000000..02dcac39cdf --- /dev/null +++ b/sound/soc/pxa/em-x270.c @@ -0,0 +1,102 @@ +/* + * em-x270.c -- SoC audio for EM-X270 + * + * Copyright 2007 CompuLab, Ltd. + * + * Author: Mike Rapoport <mike@compulab.co.il> + * + * Copied from tosa.c: + * Copyright 2005 Wolfson Microelectronics PLC. + * Copyright 2005 Openedhand Ltd. + * + * Authors: Liam Girdwood <liam.girdwood@wolfsonmicro.com> + * Richard Purdie <richard@openedhand.com> + * + * This program is free software; you can redistribute it and/or modify it + * under the terms of the GNU General Public License as published by the + * Free Software Foundation; either version 2 of the License, or (at your + * option) any later version. + * + */ + +#include <linux/module.h> +#include <linux/moduleparam.h> +#include <linux/device.h> + +#include <sound/driver.h> +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/soc.h> +#include <sound/soc-dapm.h> + +#include <asm/mach-types.h> +#include <asm/arch/pxa-regs.h> +#include <asm/arch/hardware.h> +#include <asm/arch/audio.h> + +#include "../codecs/wm9712.h" +#include "pxa2xx-pcm.h" +#include "pxa2xx-ac97.h" + +static struct snd_soc_dai_link em_x270_dai[] = { + { + .name = "AC97", + .stream_name = "AC97 HiFi", + .cpu_dai = &pxa_ac97_dai[PXA2XX_DAI_AC97_HIFI], + .codec_dai = &wm9712_dai[WM9712_DAI_AC97_HIFI], + }, + { + .name = "AC97 Aux", + .stream_name = "AC97 Aux", + .cpu_dai = &pxa_ac97_dai[PXA2XX_DAI_AC97_AUX], + .codec_dai = &wm9712_dai[WM9712_DAI_AC97_AUX], + }, +}; + +static struct snd_soc_machine em_x270 = { + .name = "EM-X270", + .dai_link = em_x270_dai, + .num_links = ARRAY_SIZE(em_x270_dai), +}; + +static struct snd_soc_device em_x270_snd_devdata = { + .machine = &em_x270, + .platform = &pxa2xx_soc_platform, + .codec_dev = &soc_codec_dev_wm9712, +}; + +static struct platform_device *em_x270_snd_device; + +static int __init em_x270_init(void) +{ + int ret; + + if (!machine_is_em_x270()) + return -ENODEV; + + em_x270_snd_device = platform_device_alloc("soc-audio", -1); + if (!em_x270_snd_device) + return -ENOMEM; + + platform_set_drvdata(em_x270_snd_device, &em_x270_snd_devdata); + em_x270_snd_devdata.dev = &em_x270_snd_device->dev; + ret = platform_device_add(em_x270_snd_device); + + if (ret) + platform_device_put(em_x270_snd_device); + + return ret; +} + +static void __exit em_x270_exit(void) +{ + platform_device_unregister(em_x270_snd_device); +} + +module_init(em_x270_init); +module_exit(em_x270_exit); + +/* Module information */ +MODULE_AUTHOR("Mike Rapoport"); +MODULE_DESCRIPTION("ALSA SoC EM-X270"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/pxa/poodle.c b/sound/soc/pxa/poodle.c index 7e830b21894..65a4e9a8c39 100644 --- a/sound/soc/pxa/poodle.c +++ b/sound/soc/pxa/poodle.c @@ -48,8 +48,6 @@ static int poodle_spk_func; static void poodle_ext_control(struct snd_soc_codec *codec) { - int spk = 0; - /* set up jack connection */ if (poodle_jack_func == POODLE_HP) { /* set = unmute headphone */ @@ -57,23 +55,23 @@ static void poodle_ext_control(struct snd_soc_codec *codec) POODLE_LOCOMO_GPIO_MUTE_L, 1); locomo_gpio_write(&poodle_locomo_device.dev, POODLE_LOCOMO_GPIO_MUTE_R, 1); - snd_soc_dapm_set_endpoint(codec, "Headphone Jack", 1); + snd_soc_dapm_enable_pin(codec, "Headphone Jack"); } else { locomo_gpio_write(&poodle_locomo_device.dev, POODLE_LOCOMO_GPIO_MUTE_L, 0); locomo_gpio_write(&poodle_locomo_device.dev, POODLE_LOCOMO_GPIO_MUTE_R, 0); - snd_soc_dapm_set_endpoint(codec, "Headphone Jack", 0); + snd_soc_dapm_disable_pin(codec, "Headphone Jack"); } - if (poodle_spk_func == POODLE_SPK_ON) - spk = 1; - /* set the enpoints to their new connetion states */ - snd_soc_dapm_set_endpoint(codec, "Ext Spk", spk); + if (poodle_spk_func == POODLE_SPK_ON) + snd_soc_dapm_enable_pin(codec, "Ext Spk"); + else + snd_soc_dapm_disable_pin(codec, "Ext Spk"); /* signal a DAPM event */ - snd_soc_dapm_sync_endpoints(codec); + snd_soc_dapm_sync(codec); } static int poodle_startup(struct snd_pcm_substream *substream) @@ -104,8 +102,8 @@ static int poodle_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params) { struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_codec_dai *codec_dai = rtd->dai->codec_dai; - struct snd_soc_cpu_dai *cpu_dai = rtd->dai->cpu_dai; + struct snd_soc_dai *codec_dai = rtd->dai->codec_dai; + struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai; unsigned int clk = 0; int ret = 0; @@ -124,25 +122,25 @@ static int poodle_hw_params(struct snd_pcm_substream *substream, } /* set codec DAI configuration */ - ret = codec_dai->dai_ops.set_fmt(codec_dai, SND_SOC_DAIFMT_I2S | + ret = snd_soc_dai_set_fmt(codec_dai, SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBS_CFS); if (ret < 0) return ret; /* set cpu DAI configuration */ - ret = cpu_dai->dai_ops.set_fmt(cpu_dai, SND_SOC_DAIFMT_I2S | + ret = snd_soc_dai_set_fmt(cpu_dai, SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBS_CFS); if (ret < 0) return ret; /* set the codec system clock for DAC and ADC */ - ret = codec_dai->dai_ops.set_sysclk(codec_dai, WM8731_SYSCLK, clk, + ret = snd_soc_dai_set_sysclk(codec_dai, WM8731_SYSCLK, clk, SND_SOC_CLOCK_IN); if (ret < 0) return ret; /* set the I2S system clock as input (unused) */ - ret = cpu_dai->dai_ops.set_sysclk(cpu_dai, PXA2XX_I2S_SYSCLK, 0, + ret = snd_soc_dai_set_sysclk(cpu_dai, PXA2XX_I2S_SYSCLK, 0, SND_SOC_CLOCK_IN); if (ret < 0) return ret; @@ -215,8 +213,8 @@ SND_SOC_DAPM_HP("Headphone Jack", NULL), SND_SOC_DAPM_SPK("Ext Spk", poodle_amp_event), }; -/* Corgi machine audio_mapnections to the codec pins */ -static const char *audio_map[][3] = { +/* Corgi machine connections to the codec pins */ +static const struct snd_soc_dapm_route audio_map[] = { /* headphone connected to LHPOUT1, RHPOUT1 */ {"Headphone Jack", NULL, "LHPOUT"}, @@ -225,8 +223,6 @@ static const char *audio_map[][3] = { /* speaker connected to LOUT, ROUT */ {"Ext Spk", NULL, "ROUT"}, {"Ext Spk", NULL, "LOUT"}, - - {NULL, NULL, NULL}, }; static const char *jack_function[] = {"Off", "Headphone"}; @@ -250,9 +246,9 @@ static int poodle_wm8731_init(struct snd_soc_codec *codec) { int i, err; - snd_soc_dapm_set_endpoint(codec, "LLINEIN", 0); - snd_soc_dapm_set_endpoint(codec, "RLINEIN", 0); - snd_soc_dapm_set_endpoint(codec, "MICIN", 1); + snd_soc_dapm_disable_pin(codec, "LLINEIN"); + snd_soc_dapm_disable_pin(codec, "RLINEIN"); + snd_soc_dapm_enable_pin(codec, "MICIN"); /* Add poodle specific controls */ for (i = 0; i < ARRAY_SIZE(wm8731_poodle_controls); i++) { @@ -263,15 +259,13 @@ static int poodle_wm8731_init(struct snd_soc_codec *codec) } /* Add poodle specific widgets */ - for (i = 0; i < ARRAY_SIZE(wm8731_dapm_widgets); i++) - snd_soc_dapm_new_control(codec, &wm8731_dapm_widgets[i]); + snd_soc_dapm_new_controls(codec, wm8731_dapm_widgets, + ARRAY_SIZE(wm8731_dapm_widgets)); /* Set up poodle specific audio path audio_map */ - for (i = 0; audio_map[i][0] != NULL; i++) - snd_soc_dapm_connect_input(codec, audio_map[i][0], - audio_map[i][1], audio_map[i][2]); + snd_soc_dapm_add_routes(codec, audio_map, ARRAY_SIZE(audio_map)); - snd_soc_dapm_sync_endpoints(codec); + snd_soc_dapm_sync(codec); return 0; } diff --git a/sound/soc/pxa/pxa2xx-ac97.c b/sound/soc/pxa/pxa2xx-ac97.c index 97ec2d90547..059af815ea0 100644 --- a/sound/soc/pxa/pxa2xx-ac97.c +++ b/sound/soc/pxa/pxa2xx-ac97.c @@ -283,7 +283,7 @@ static struct pxa2xx_pcm_dma_params pxa2xx_ac97_pcm_mic_mono_in = { #ifdef CONFIG_PM static int pxa2xx_ac97_suspend(struct platform_device *pdev, - struct snd_soc_cpu_dai *dai) + struct snd_soc_dai *dai) { GCR |= GCR_ACLINK_OFF; clk_disable(ac97_clk); @@ -291,7 +291,7 @@ static int pxa2xx_ac97_suspend(struct platform_device *pdev, } static int pxa2xx_ac97_resume(struct platform_device *pdev, - struct snd_soc_cpu_dai *dai) + struct snd_soc_dai *dai) { pxa_gpio_mode(GPIO31_SYNC_AC97_MD); pxa_gpio_mode(GPIO30_SDATA_OUT_AC97_MD); @@ -310,7 +310,8 @@ static int pxa2xx_ac97_resume(struct platform_device *pdev, #define pxa2xx_ac97_resume NULL #endif -static int pxa2xx_ac97_probe(struct platform_device *pdev) +static int pxa2xx_ac97_probe(struct platform_device *pdev, + struct snd_soc_dai *dai) { int ret; @@ -355,7 +356,8 @@ static int pxa2xx_ac97_probe(struct platform_device *pdev) return ret; } -static void pxa2xx_ac97_remove(struct platform_device *pdev) +static void pxa2xx_ac97_remove(struct platform_device *pdev, + struct snd_soc_dai *dai) { GCR |= GCR_ACLINK_OFF; free_irq(IRQ_AC97, NULL); @@ -372,7 +374,7 @@ static int pxa2xx_ac97_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params) { struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_cpu_dai *cpu_dai = rtd->dai->cpu_dai; + struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai; if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) cpu_dai->dma_data = &pxa2xx_ac97_pcm_stereo_out; @@ -386,7 +388,7 @@ static int pxa2xx_ac97_hw_aux_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params) { struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_cpu_dai *cpu_dai = rtd->dai->cpu_dai; + struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai; if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) cpu_dai->dma_data = &pxa2xx_ac97_pcm_aux_mono_out; @@ -400,7 +402,7 @@ static int pxa2xx_ac97_hw_mic_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params) { struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_cpu_dai *cpu_dai = rtd->dai->cpu_dai; + struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai; if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) return -ENODEV; @@ -418,7 +420,7 @@ static int pxa2xx_ac97_hw_mic_params(struct snd_pcm_substream *substream, * There is only 1 physical AC97 interface for pxa2xx, but it * has extra fifo's that can be used for aux DACs and ADCs. */ -struct snd_soc_cpu_dai pxa_ac97_dai[] = { +struct snd_soc_dai pxa_ac97_dai[] = { { .name = "pxa2xx-ac97", .id = 0, diff --git a/sound/soc/pxa/pxa2xx-ac97.h b/sound/soc/pxa/pxa2xx-ac97.h index b8ccfee095c..e390de8edcd 100644 --- a/sound/soc/pxa/pxa2xx-ac97.h +++ b/sound/soc/pxa/pxa2xx-ac97.h @@ -14,7 +14,7 @@ #define PXA2XX_DAI_AC97_AUX 1 #define PXA2XX_DAI_AC97_MIC 2 -extern struct snd_soc_cpu_dai pxa_ac97_dai[3]; +extern struct snd_soc_dai pxa_ac97_dai[3]; /* platform data */ extern struct snd_ac97_bus_ops pxa2xx_ac97_ops; diff --git a/sound/soc/pxa/pxa2xx-i2s.c b/sound/soc/pxa/pxa2xx-i2s.c index 42507103097..9c06553b926 100644 --- a/sound/soc/pxa/pxa2xx-i2s.c +++ b/sound/soc/pxa/pxa2xx-i2s.c @@ -9,9 +9,6 @@ * under the terms of the GNU General Public License as published by the * Free Software Foundation; either version 2 of the License, or (at your * option) any later version. - * - * Revision history - * 12th Aug 2005 Initial version. */ #include <linux/init.h> @@ -80,7 +77,7 @@ static struct pxa2xx_gpio gpio_bus[] = { static int pxa2xx_i2s_startup(struct snd_pcm_substream *substream) { struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_cpu_dai *cpu_dai = rtd->dai->cpu_dai; + struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai; if (!cpu_dai->active) { SACR0 |= SACR0_RST; @@ -101,7 +98,7 @@ static int pxa_i2s_wait(void) return 0; } -static int pxa2xx_i2s_set_dai_fmt(struct snd_soc_cpu_dai *cpu_dai, +static int pxa2xx_i2s_set_dai_fmt(struct snd_soc_dai *cpu_dai, unsigned int fmt) { /* interface format */ @@ -127,7 +124,7 @@ static int pxa2xx_i2s_set_dai_fmt(struct snd_soc_cpu_dai *cpu_dai, return 0; } -static int pxa2xx_i2s_set_dai_sysclk(struct snd_soc_cpu_dai *cpu_dai, +static int pxa2xx_i2s_set_dai_sysclk(struct snd_soc_dai *cpu_dai, int clk_id, unsigned int freq, int dir) { if (clk_id != PXA2XX_I2S_SYSCLK) @@ -143,7 +140,7 @@ static int pxa2xx_i2s_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params) { struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_cpu_dai *cpu_dai = rtd->dai->cpu_dai; + struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai; pxa_gpio_mode(gpio_bus[pxa_i2s.master].rx); pxa_gpio_mode(gpio_bus[pxa_i2s.master].tx); @@ -240,7 +237,7 @@ static void pxa2xx_i2s_shutdown(struct snd_pcm_substream *substream) #ifdef CONFIG_PM static int pxa2xx_i2s_suspend(struct platform_device *dev, - struct snd_soc_cpu_dai *dai) + struct snd_soc_dai *dai) { if (!dai->active) return 0; @@ -258,7 +255,7 @@ static int pxa2xx_i2s_suspend(struct platform_device *dev, } static int pxa2xx_i2s_resume(struct platform_device *pdev, - struct snd_soc_cpu_dai *dai) + struct snd_soc_dai *dai) { if (!dai->active) return 0; @@ -283,7 +280,7 @@ static int pxa2xx_i2s_resume(struct platform_device *pdev, SNDRV_PCM_RATE_16000 | SNDRV_PCM_RATE_22050 | SNDRV_PCM_RATE_44100 | \ SNDRV_PCM_RATE_48000 | SNDRV_PCM_RATE_96000) -struct snd_soc_cpu_dai pxa_i2s_dai = { +struct snd_soc_dai pxa_i2s_dai = { .name = "pxa2xx-i2s", .id = 0, .type = SND_SOC_DAI_I2S, diff --git a/sound/soc/pxa/pxa2xx-i2s.h b/sound/soc/pxa/pxa2xx-i2s.h index 4435bd9f884..e2def441153 100644 --- a/sound/soc/pxa/pxa2xx-i2s.h +++ b/sound/soc/pxa/pxa2xx-i2s.h @@ -15,6 +15,6 @@ /* I2S clock */ #define PXA2XX_I2S_SYSCLK 0 -extern struct snd_soc_cpu_dai pxa_i2s_dai; +extern struct snd_soc_dai pxa_i2s_dai; #endif diff --git a/sound/soc/pxa/pxa2xx-pcm.c b/sound/soc/pxa/pxa2xx-pcm.c index 01ad7bf716b..2df03ee5819 100644 --- a/sound/soc/pxa/pxa2xx-pcm.c +++ b/sound/soc/pxa/pxa2xx-pcm.c @@ -330,7 +330,7 @@ static void pxa2xx_pcm_free_dma_buffers(struct snd_pcm *pcm) static u64 pxa2xx_pcm_dmamask = DMA_32BIT_MASK; -int pxa2xx_pcm_new(struct snd_card *card, struct snd_soc_codec_dai *dai, +int pxa2xx_pcm_new(struct snd_card *card, struct snd_soc_dai *dai, struct snd_pcm *pcm) { int ret = 0; diff --git a/sound/soc/pxa/spitz.c b/sound/soc/pxa/spitz.c index d8b8372db00..64385797da5 100644 --- a/sound/soc/pxa/spitz.c +++ b/sound/soc/pxa/spitz.c @@ -12,9 +12,6 @@ * Free Software Foundation; either version 2 of the License, or (at your * option) any later version. * - * Revision history - * 30th Nov 2005 Initial version. - * */ #include <linux/module.h> @@ -54,60 +51,60 @@ static int spitz_spk_func; static void spitz_ext_control(struct snd_soc_codec *codec) { if (spitz_spk_func == SPITZ_SPK_ON) - snd_soc_dapm_set_endpoint(codec, "Ext Spk", 1); + snd_soc_dapm_enable_pin(codec, "Ext Spk"); else - snd_soc_dapm_set_endpoint(codec, "Ext Spk", 0); + snd_soc_dapm_disable_pin(codec, "Ext Spk"); /* set up jack connection */ switch (spitz_jack_func) { case SPITZ_HP: /* enable and unmute hp jack, disable mic bias */ - snd_soc_dapm_set_endpoint(codec, "Headset Jack", 0); - snd_soc_dapm_set_endpoint(codec, "Mic Jack", 0); - snd_soc_dapm_set_endpoint(codec, "Line Jack", 0); - snd_soc_dapm_set_endpoint(codec, "Headphone Jack", 1); + snd_soc_dapm_disable_pin(codec, "Headset Jack"); + snd_soc_dapm_disable_pin(codec, "Mic Jack"); + snd_soc_dapm_disable_pin(codec, "Line Jack"); + snd_soc_dapm_enable_pin(codec, "Headphone Jack"); set_scoop_gpio(&spitzscoop_device.dev, SPITZ_SCP_MUTE_L); set_scoop_gpio(&spitzscoop_device.dev, SPITZ_SCP_MUTE_R); break; case SPITZ_MIC: /* enable mic jack and bias, mute hp */ - snd_soc_dapm_set_endpoint(codec, "Headphone Jack", 0); - snd_soc_dapm_set_endpoint(codec, "Headset Jack", 0); - snd_soc_dapm_set_endpoint(codec, "Line Jack", 0); - snd_soc_dapm_set_endpoint(codec, "Mic Jack", 1); + snd_soc_dapm_disable_pin(codec, "Headphone Jack"); + snd_soc_dapm_disable_pin(codec, "Headset Jack"); + snd_soc_dapm_disable_pin(codec, "Line Jack"); + snd_soc_dapm_enable_pin(codec, "Mic Jack"); reset_scoop_gpio(&spitzscoop_device.dev, SPITZ_SCP_MUTE_L); reset_scoop_gpio(&spitzscoop_device.dev, SPITZ_SCP_MUTE_R); break; case SPITZ_LINE: /* enable line jack, disable mic bias and mute hp */ - snd_soc_dapm_set_endpoint(codec, "Headphone Jack", 0); - snd_soc_dapm_set_endpoint(codec, "Headset Jack", 0); - snd_soc_dapm_set_endpoint(codec, "Mic Jack", 0); - snd_soc_dapm_set_endpoint(codec, "Line Jack", 1); + snd_soc_dapm_disable_pin(codec, "Headphone Jack"); + snd_soc_dapm_disable_pin(codec, "Headset Jack"); + snd_soc_dapm_disable_pin(codec, "Mic Jack"); + snd_soc_dapm_enable_pin(codec, "Line Jack"); reset_scoop_gpio(&spitzscoop_device.dev, SPITZ_SCP_MUTE_L); reset_scoop_gpio(&spitzscoop_device.dev, SPITZ_SCP_MUTE_R); break; case SPITZ_HEADSET: /* enable and unmute headset jack enable mic bias, mute L hp */ - snd_soc_dapm_set_endpoint(codec, "Headphone Jack", 0); - snd_soc_dapm_set_endpoint(codec, "Mic Jack", 1); - snd_soc_dapm_set_endpoint(codec, "Line Jack", 0); - snd_soc_dapm_set_endpoint(codec, "Headset Jack", 1); + snd_soc_dapm_disable_pin(codec, "Headphone Jack"); + snd_soc_dapm_enable_pin(codec, "Mic Jack"); + snd_soc_dapm_disable_pin(codec, "Line Jack"); + snd_soc_dapm_enable_pin(codec, "Headset Jack"); reset_scoop_gpio(&spitzscoop_device.dev, SPITZ_SCP_MUTE_L); set_scoop_gpio(&spitzscoop_device.dev, SPITZ_SCP_MUTE_R); break; case SPITZ_HP_OFF: /* jack removed, everything off */ - snd_soc_dapm_set_endpoint(codec, "Headphone Jack", 0); - snd_soc_dapm_set_endpoint(codec, "Headset Jack", 0); - snd_soc_dapm_set_endpoint(codec, "Mic Jack", 0); - snd_soc_dapm_set_endpoint(codec, "Line Jack", 0); + snd_soc_dapm_disable_pin(codec, "Headphone Jack"); + snd_soc_dapm_disable_pin(codec, "Headset Jack"); + snd_soc_dapm_disable_pin(codec, "Mic Jack"); + snd_soc_dapm_disable_pin(codec, "Line Jack"); reset_scoop_gpio(&spitzscoop_device.dev, SPITZ_SCP_MUTE_L); reset_scoop_gpio(&spitzscoop_device.dev, SPITZ_SCP_MUTE_R); break; } - snd_soc_dapm_sync_endpoints(codec); + snd_soc_dapm_sync(codec); } static int spitz_startup(struct snd_pcm_substream *substream) @@ -124,8 +121,8 @@ static int spitz_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params) { struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_codec_dai *codec_dai = rtd->dai->codec_dai; - struct snd_soc_cpu_dai *cpu_dai = rtd->dai->cpu_dai; + struct snd_soc_dai *codec_dai = rtd->dai->codec_dai; + struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai; unsigned int clk = 0; int ret = 0; @@ -144,25 +141,25 @@ static int spitz_hw_params(struct snd_pcm_substream *substream, } /* set codec DAI configuration */ - ret = codec_dai->dai_ops.set_fmt(codec_dai, SND_SOC_DAIFMT_I2S | + ret = snd_soc_dai_set_fmt(codec_dai, SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBS_CFS); if (ret < 0) return ret; /* set cpu DAI configuration */ - ret = cpu_dai->dai_ops.set_fmt(cpu_dai, SND_SOC_DAIFMT_I2S | + ret = snd_soc_dai_set_fmt(cpu_dai, SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBS_CFS); if (ret < 0) return ret; /* set the codec system clock for DAC and ADC */ - ret = codec_dai->dai_ops.set_sysclk(codec_dai, WM8750_SYSCLK, clk, + ret = snd_soc_dai_set_sysclk(codec_dai, WM8750_SYSCLK, clk, SND_SOC_CLOCK_IN); if (ret < 0) return ret; /* set the I2S system clock as input (unused) */ - ret = cpu_dai->dai_ops.set_sysclk(cpu_dai, PXA2XX_I2S_SYSCLK, 0, + ret = snd_soc_dai_set_sysclk(cpu_dai, PXA2XX_I2S_SYSCLK, 0, SND_SOC_CLOCK_IN); if (ret < 0) return ret; @@ -250,7 +247,7 @@ static const struct snd_soc_dapm_widget wm8750_dapm_widgets[] = { }; /* Spitz machine audio_map */ -static const char *audio_map[][3] = { +static const struct snd_soc_dapm_route audio_map[] = { /* headphone connected to LOUT1, ROUT1 */ {"Headphone Jack", NULL, "LOUT1"}, @@ -269,8 +266,6 @@ static const char *audio_map[][3] = { /* line is connected to input 1 - no bias */ {"LINPUT1", NULL, "Line Jack"}, - - {NULL, NULL, NULL}, }; static const char *jack_function[] = {"Headphone", "Mic", "Line", "Headset", @@ -296,13 +291,13 @@ static int spitz_wm8750_init(struct snd_soc_codec *codec) int i, err; /* NC codec pins */ - snd_soc_dapm_set_endpoint(codec, "RINPUT1", 0); - snd_soc_dapm_set_endpoint(codec, "LINPUT2", 0); - snd_soc_dapm_set_endpoint(codec, "RINPUT2", 0); - snd_soc_dapm_set_endpoint(codec, "LINPUT3", 0); - snd_soc_dapm_set_endpoint(codec, "RINPUT3", 0); - snd_soc_dapm_set_endpoint(codec, "OUT3", 0); - snd_soc_dapm_set_endpoint(codec, "MONO", 0); + snd_soc_dapm_disable_pin(codec, "RINPUT1"); + snd_soc_dapm_disable_pin(codec, "LINPUT2"); + snd_soc_dapm_disable_pin(codec, "RINPUT2"); + snd_soc_dapm_disable_pin(codec, "LINPUT3"); + snd_soc_dapm_disable_pin(codec, "RINPUT3"); + snd_soc_dapm_disable_pin(codec, "OUT3"); + snd_soc_dapm_disable_pin(codec, "MONO"); /* Add spitz specific controls */ for (i = 0; i < ARRAY_SIZE(wm8750_spitz_controls); i++) { @@ -313,15 +308,13 @@ static int spitz_wm8750_init(struct snd_soc_codec *codec) } /* Add spitz specific widgets */ - for (i = 0; i < ARRAY_SIZE(wm8750_dapm_widgets); i++) - snd_soc_dapm_new_control(codec, &wm8750_dapm_widgets[i]); + snd_soc_dapm_new_controls(codec, wm8750_dapm_widgets, + ARRAY_SIZE(wm8750_dapm_widgets)); - /* Set up spitz specific audio path audio_map */ - for (i = 0; audio_map[i][0] != NULL; i++) - snd_soc_dapm_connect_input(codec, audio_map[i][0], - audio_map[i][1], audio_map[i][2]); + /* Set up spitz specific audio paths */ + snd_soc_dapm_add_routes(codec, audio_map, ARRAY_SIZE(audio_map)); - snd_soc_dapm_sync_endpoints(codec); + snd_soc_dapm_sync(codec); return 0; } diff --git a/sound/soc/pxa/tosa.c b/sound/soc/pxa/tosa.c index 7346d7e5d06..b6edb61a3a3 100644 --- a/sound/soc/pxa/tosa.c +++ b/sound/soc/pxa/tosa.c @@ -12,9 +12,6 @@ * Free Software Foundation; either version 2 of the License, or (at your * option) any later version. * - * Revision history - * 30th Nov 2005 Initial version. - * * GPIO's * 1 - Jack Insertion * 5 - Hookswitch (headset answer/hang up switch) @@ -55,29 +52,31 @@ static int tosa_spk_func; static void tosa_ext_control(struct snd_soc_codec *codec) { - int spk = 0, mic_int = 0, hp = 0, hs = 0; - /* set up jack connection */ switch (tosa_jack_func) { case TOSA_HP: - hp = 1; + snd_soc_dapm_disable_pin(codec, "Mic (Internal)"); + snd_soc_dapm_enable_pin(codec, "Headphone Jack"); + snd_soc_dapm_disable_pin(codec, "Headset Jack"); break; case TOSA_MIC_INT: - mic_int = 1; + snd_soc_dapm_enable_pin(codec, "Mic (Internal)"); + snd_soc_dapm_disable_pin(codec, "Headphone Jack"); + snd_soc_dapm_disable_pin(codec, "Headset Jack"); break; case TOSA_HEADSET: - hs = 1; + snd_soc_dapm_disable_pin(codec, "Mic (Internal)"); + snd_soc_dapm_disable_pin(codec, "Headphone Jack"); + snd_soc_dapm_enable_pin(codec, "Headset Jack"); break; } if (tosa_spk_func == TOSA_SPK_ON) - spk = 1; + snd_soc_dapm_enable_pin(codec, "Speaker"); + else + snd_soc_dapm_disable_pin(codec, "Speaker"); - snd_soc_dapm_set_endpoint(codec, "Speaker", spk); - snd_soc_dapm_set_endpoint(codec, "Mic (Internal)", mic_int); - snd_soc_dapm_set_endpoint(codec, "Headphone Jack", hp); - snd_soc_dapm_set_endpoint(codec, "Headset Jack", hs); - snd_soc_dapm_sync_endpoints(codec); + snd_soc_dapm_sync(codec); } static int tosa_startup(struct snd_pcm_substream *substream) @@ -154,7 +153,7 @@ SND_SOC_DAPM_SPK("Speaker", NULL), }; /* tosa audio map */ -static const char *audio_map[][3] = { +static const struct snd_soc_dapm_route audio_map[] = { /* headphone connected to HPOUTL, HPOUTR */ {"Headphone Jack", NULL, "HPOUTL"}, @@ -173,8 +172,6 @@ static const char *audio_map[][3] = { {"Headset Jack", NULL, "HPOUTR"}, {"LINEINR", NULL, "Mic Bias"}, {"Mic Bias", NULL, "Headset Jack"}, - - {NULL, NULL, NULL}, }; static const char *jack_function[] = {"Headphone", "Mic", "Line", "Headset", @@ -196,8 +193,8 @@ static int tosa_ac97_init(struct snd_soc_codec *codec) { int i, err; - snd_soc_dapm_set_endpoint(codec, "OUT3", 0); - snd_soc_dapm_set_endpoint(codec, "MONOOUT", 0); + snd_soc_dapm_disable_pin(codec, "OUT3"); + snd_soc_dapm_disable_pin(codec, "MONOOUT"); /* add tosa specific controls */ for (i = 0; i < ARRAY_SIZE(tosa_controls); i++) { @@ -208,17 +205,13 @@ static int tosa_ac97_init(struct snd_soc_codec *codec) } /* add tosa specific widgets */ - for (i = 0; i < ARRAY_SIZE(tosa_dapm_widgets); i++) { - snd_soc_dapm_new_control(codec, &tosa_dapm_widgets[i]); - } + snd_soc_dapm_new_controls(codec, tosa_dapm_widgets, + ARRAY_SIZE(tosa_dapm_widgets)); /* set up tosa specific audio path audio_map */ - for (i = 0; audio_map[i][0] != NULL; i++) { - snd_soc_dapm_connect_input(codec, audio_map[i][0], - audio_map[i][1], audio_map[i][2]); - } + snd_soc_dapm_add_routes(codec, audio_map, ARRAY_SIZE(audio_map)); - snd_soc_dapm_sync_endpoints(codec); + snd_soc_dapm_sync(codec); return 0; } diff --git a/sound/soc/s3c24xx/Kconfig b/sound/soc/s3c24xx/Kconfig index 1f6dbfc4caa..b9f2353effe 100644 --- a/sound/soc/s3c24xx/Kconfig +++ b/sound/soc/s3c24xx/Kconfig @@ -1,7 +1,6 @@ config SND_S3C24XX_SOC tristate "SoC Audio for the Samsung S3C24XX chips" - depends on ARCH_S3C2410 && SND_SOC - select SND_PCM + depends on ARCH_S3C2410 help Say Y or M if you want to add support for codecs attached to the S3C24XX AC97, I2S or SSP interface. You will also need @@ -16,7 +15,6 @@ config SND_S3C2412_SOC_I2S config SND_S3C2443_SOC_AC97 tristate select AC97_BUS - select SND_AC97_CODEC select SND_SOC_AC97_BUS config SND_S3C24XX_SOC_NEO1973_WM8753 diff --git a/sound/soc/s3c24xx/neo1973_wm8753.c b/sound/soc/s3c24xx/neo1973_wm8753.c index 0e9d1c5f248..4d7a9aa15f1 100644 --- a/sound/soc/s3c24xx/neo1973_wm8753.c +++ b/sound/soc/s3c24xx/neo1973_wm8753.c @@ -10,10 +10,6 @@ * Free Software Foundation; either version 2 of the License, or (at your * option) any later version. * - * Revision history - * 20th Jan 2007 Initial version. - * 05th Feb 2007 Rename all to Neo1973 - * */ #include <linux/module.h> @@ -26,6 +22,7 @@ #include <sound/pcm.h> #include <sound/soc.h> #include <sound/soc-dapm.h> +#include <sound/tlv.h> #include <asm/mach-types.h> #include <asm/hardware/scoop.h> @@ -43,6 +40,14 @@ #include "s3c24xx-pcm.h" #include "s3c24xx-i2s.h" +/* Debugging stuff */ +#define S3C24XX_SOC_NEO1973_WM8753_DEBUG 0 +#if S3C24XX_SOC_NEO1973_WM8753_DEBUG +#define DBG(x...) printk(KERN_DEBUG "s3c24xx-soc-neo1973-wm8753: " x) +#else +#define DBG(x...) +#endif + /* define the scenarios */ #define NEO_AUDIO_OFF 0 #define NEO_GSM_CALL_AUDIO_HANDSET 1 @@ -61,12 +66,14 @@ static int neo1973_hifi_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params) { struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_codec_dai *codec_dai = rtd->dai->codec_dai; - struct snd_soc_cpu_dai *cpu_dai = rtd->dai->cpu_dai; + struct snd_soc_dai *codec_dai = rtd->dai->codec_dai; + struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai; unsigned int pll_out = 0, bclk = 0; int ret = 0; unsigned long iis_clkrate; + DBG("Entered %s\n", __func__); + iis_clkrate = s3c24xx_i2s_get_clockrate(); switch (params_rate(params)) { @@ -101,44 +108,44 @@ static int neo1973_hifi_hw_params(struct snd_pcm_substream *substream, } /* set codec DAI configuration */ - ret = codec_dai->dai_ops.set_fmt(codec_dai, + ret = snd_soc_dai_set_fmt(codec_dai, SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBM_CFM); if (ret < 0) return ret; /* set cpu DAI configuration */ - ret = cpu_dai->dai_ops.set_fmt(cpu_dai, + ret = snd_soc_dai_set_fmt(cpu_dai, SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBM_CFM); if (ret < 0) return ret; /* set the codec system clock for DAC and ADC */ - ret = codec_dai->dai_ops.set_sysclk(codec_dai, WM8753_MCLK, pll_out, + ret = snd_soc_dai_set_sysclk(codec_dai, WM8753_MCLK, pll_out, SND_SOC_CLOCK_IN); if (ret < 0) return ret; /* set MCLK division for sample rate */ - ret = cpu_dai->dai_ops.set_clkdiv(cpu_dai, S3C24XX_DIV_MCLK, + ret = snd_soc_dai_set_clkdiv(cpu_dai, S3C24XX_DIV_MCLK, S3C2410_IISMOD_32FS); if (ret < 0) return ret; /* set codec BCLK division for sample rate */ - ret = codec_dai->dai_ops.set_clkdiv(codec_dai, WM8753_BCLKDIV, bclk); + ret = snd_soc_dai_set_clkdiv(codec_dai, WM8753_BCLKDIV, bclk); if (ret < 0) return ret; /* set prescaler division for sample rate */ - ret = cpu_dai->dai_ops.set_clkdiv(cpu_dai, S3C24XX_DIV_PRESCALER, + ret = snd_soc_dai_set_clkdiv(cpu_dai, S3C24XX_DIV_PRESCALER, S3C24XX_PRESCALE(4, 4)); if (ret < 0) return ret; /* codec PLL input is PCLK/4 */ - ret = codec_dai->dai_ops.set_pll(codec_dai, WM8753_PLL1, + ret = snd_soc_dai_set_pll(codec_dai, WM8753_PLL1, iis_clkrate / 4, pll_out); if (ret < 0) return ret; @@ -149,10 +156,12 @@ static int neo1973_hifi_hw_params(struct snd_pcm_substream *substream, static int neo1973_hifi_hw_free(struct snd_pcm_substream *substream) { struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_codec_dai *codec_dai = rtd->dai->codec_dai; + struct snd_soc_dai *codec_dai = rtd->dai->codec_dai; + + DBG("Entered %s\n", __func__); /* disable the PLL */ - return codec_dai->dai_ops.set_pll(codec_dai, WM8753_PLL1, 0, 0); + return snd_soc_dai_set_pll(codec_dai, WM8753_PLL1, 0, 0); } /* @@ -167,11 +176,13 @@ static int neo1973_voice_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params) { struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_codec_dai *codec_dai = rtd->dai->codec_dai; + struct snd_soc_dai *codec_dai = rtd->dai->codec_dai; unsigned int pcmdiv = 0; int ret = 0; unsigned long iis_clkrate; + DBG("Entered %s\n", __func__); + iis_clkrate = s3c24xx_i2s_get_clockrate(); if (params_rate(params) != 8000) @@ -183,24 +194,24 @@ static int neo1973_voice_hw_params(struct snd_pcm_substream *substream, /* todo: gg check mode (DSP_B) against CSR datasheet */ /* set codec DAI configuration */ - ret = codec_dai->dai_ops.set_fmt(codec_dai, SND_SOC_DAIFMT_DSP_B | + ret = snd_soc_dai_set_fmt(codec_dai, SND_SOC_DAIFMT_DSP_B | SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBS_CFS); if (ret < 0) return ret; /* set the codec system clock for DAC and ADC */ - ret = codec_dai->dai_ops.set_sysclk(codec_dai, WM8753_PCMCLK, 12288000, + ret = snd_soc_dai_set_sysclk(codec_dai, WM8753_PCMCLK, 12288000, SND_SOC_CLOCK_IN); if (ret < 0) return ret; /* set codec PCM division for sample rate */ - ret = codec_dai->dai_ops.set_clkdiv(codec_dai, WM8753_PCMDIV, pcmdiv); + ret = snd_soc_dai_set_clkdiv(codec_dai, WM8753_PCMDIV, pcmdiv); if (ret < 0) return ret; /* configue and enable PLL for 12.288MHz output */ - ret = codec_dai->dai_ops.set_pll(codec_dai, WM8753_PLL2, + ret = snd_soc_dai_set_pll(codec_dai, WM8753_PLL2, iis_clkrate / 4, 12288000); if (ret < 0) return ret; @@ -211,10 +222,12 @@ static int neo1973_voice_hw_params(struct snd_pcm_substream *substream, static int neo1973_voice_hw_free(struct snd_pcm_substream *substream) { struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_codec_dai *codec_dai = rtd->dai->codec_dai; + struct snd_soc_dai *codec_dai = rtd->dai->codec_dai; + + DBG("Entered %s\n", __func__); /* disable the PLL */ - return codec_dai->dai_ops.set_pll(codec_dai, WM8753_PLL2, 0, 0); + return snd_soc_dai_set_pll(codec_dai, WM8753_PLL2, 0, 0); } static struct snd_soc_ops neo1973_voice_ops = { @@ -233,79 +246,81 @@ static int neo1973_get_scenario(struct snd_kcontrol *kcontrol, static int set_scenario_endpoints(struct snd_soc_codec *codec, int scenario) { + DBG("Entered %s\n", __func__); + switch (neo1973_scenario) { case NEO_AUDIO_OFF: - snd_soc_dapm_set_endpoint(codec, "Audio Out", 0); - snd_soc_dapm_set_endpoint(codec, "GSM Line Out", 0); - snd_soc_dapm_set_endpoint(codec, "GSM Line In", 0); - snd_soc_dapm_set_endpoint(codec, "Headset Mic", 0); - snd_soc_dapm_set_endpoint(codec, "Call Mic", 0); + snd_soc_dapm_disable_pin(codec, "Audio Out"); + snd_soc_dapm_disable_pin(codec, "GSM Line Out"); + snd_soc_dapm_disable_pin(codec, "GSM Line In"); + snd_soc_dapm_disable_pin(codec, "Headset Mic"); + snd_soc_dapm_disable_pin(codec, "Call Mic"); break; case NEO_GSM_CALL_AUDIO_HANDSET: - snd_soc_dapm_set_endpoint(codec, "Audio Out", 1); - snd_soc_dapm_set_endpoint(codec, "GSM Line Out", 1); - snd_soc_dapm_set_endpoint(codec, "GSM Line In", 1); - snd_soc_dapm_set_endpoint(codec, "Headset Mic", 0); - snd_soc_dapm_set_endpoint(codec, "Call Mic", 1); + snd_soc_dapm_enable_pin(codec, "Audio Out"); + snd_soc_dapm_enable_pin(codec, "GSM Line Out"); + snd_soc_dapm_enable_pin(codec, "GSM Line In"); + snd_soc_dapm_disable_pin(codec, "Headset Mic"); + snd_soc_dapm_enable_pin(codec, "Call Mic"); break; case NEO_GSM_CALL_AUDIO_HEADSET: - snd_soc_dapm_set_endpoint(codec, "Audio Out", 1); - snd_soc_dapm_set_endpoint(codec, "GSM Line Out", 1); - snd_soc_dapm_set_endpoint(codec, "GSM Line In", 1); - snd_soc_dapm_set_endpoint(codec, "Headset Mic", 1); - snd_soc_dapm_set_endpoint(codec, "Call Mic", 0); + snd_soc_dapm_enable_pin(codec, "Audio Out"); + snd_soc_dapm_enable_pin(codec, "GSM Line Out"); + snd_soc_dapm_enable_pin(codec, "GSM Line In"); + snd_soc_dapm_enable_pin(codec, "Headset Mic"); + snd_soc_dapm_disable_pin(codec, "Call Mic"); break; case NEO_GSM_CALL_AUDIO_BLUETOOTH: - snd_soc_dapm_set_endpoint(codec, "Audio Out", 0); - snd_soc_dapm_set_endpoint(codec, "GSM Line Out", 1); - snd_soc_dapm_set_endpoint(codec, "GSM Line In", 1); - snd_soc_dapm_set_endpoint(codec, "Headset Mic", 0); - snd_soc_dapm_set_endpoint(codec, "Call Mic", 0); + snd_soc_dapm_disable_pin(codec, "Audio Out"); + snd_soc_dapm_enable_pin(codec, "GSM Line Out"); + snd_soc_dapm_enable_pin(codec, "GSM Line In"); + snd_soc_dapm_disable_pin(codec, "Headset Mic"); + snd_soc_dapm_disable_pin(codec, "Call Mic"); break; case NEO_STEREO_TO_SPEAKERS: - snd_soc_dapm_set_endpoint(codec, "Audio Out", 1); - snd_soc_dapm_set_endpoint(codec, "GSM Line Out", 0); - snd_soc_dapm_set_endpoint(codec, "GSM Line In", 0); - snd_soc_dapm_set_endpoint(codec, "Headset Mic", 0); - snd_soc_dapm_set_endpoint(codec, "Call Mic", 0); + snd_soc_dapm_enable_pin(codec, "Audio Out"); + snd_soc_dapm_disable_pin(codec, "GSM Line Out"); + snd_soc_dapm_disable_pin(codec, "GSM Line In"); + snd_soc_dapm_disable_pin(codec, "Headset Mic"); + snd_soc_dapm_disable_pin(codec, "Call Mic"); break; case NEO_STEREO_TO_HEADPHONES: - snd_soc_dapm_set_endpoint(codec, "Audio Out", 1); - snd_soc_dapm_set_endpoint(codec, "GSM Line Out", 0); - snd_soc_dapm_set_endpoint(codec, "GSM Line In", 0); - snd_soc_dapm_set_endpoint(codec, "Headset Mic", 0); - snd_soc_dapm_set_endpoint(codec, "Call Mic", 0); + snd_soc_dapm_enable_pin(codec, "Audio Out"); + snd_soc_dapm_disable_pin(codec, "GSM Line Out"); + snd_soc_dapm_disable_pin(codec, "GSM Line In"); + snd_soc_dapm_disable_pin(codec, "Headset Mic"); + snd_soc_dapm_disable_pin(codec, "Call Mic"); break; case NEO_CAPTURE_HANDSET: - snd_soc_dapm_set_endpoint(codec, "Audio Out", 0); - snd_soc_dapm_set_endpoint(codec, "GSM Line Out", 0); - snd_soc_dapm_set_endpoint(codec, "GSM Line In", 0); - snd_soc_dapm_set_endpoint(codec, "Headset Mic", 0); - snd_soc_dapm_set_endpoint(codec, "Call Mic", 1); + snd_soc_dapm_disable_pin(codec, "Audio Out"); + snd_soc_dapm_disable_pin(codec, "GSM Line Out"); + snd_soc_dapm_disable_pin(codec, "GSM Line In"); + snd_soc_dapm_disable_pin(codec, "Headset Mic"); + snd_soc_dapm_enable_pin(codec, "Call Mic"); break; case NEO_CAPTURE_HEADSET: - snd_soc_dapm_set_endpoint(codec, "Audio Out", 0); - snd_soc_dapm_set_endpoint(codec, "GSM Line Out", 0); - snd_soc_dapm_set_endpoint(codec, "GSM Line In", 0); - snd_soc_dapm_set_endpoint(codec, "Headset Mic", 1); - snd_soc_dapm_set_endpoint(codec, "Call Mic", 0); + snd_soc_dapm_disable_pin(codec, "Audio Out"); + snd_soc_dapm_disable_pin(codec, "GSM Line Out"); + snd_soc_dapm_disable_pin(codec, "GSM Line In"); + snd_soc_dapm_enable_pin(codec, "Headset Mic"); + snd_soc_dapm_disable_pin(codec, "Call Mic"); break; case NEO_CAPTURE_BLUETOOTH: - snd_soc_dapm_set_endpoint(codec, "Audio Out", 0); - snd_soc_dapm_set_endpoint(codec, "GSM Line Out", 0); - snd_soc_dapm_set_endpoint(codec, "GSM Line In", 0); - snd_soc_dapm_set_endpoint(codec, "Headset Mic", 0); - snd_soc_dapm_set_endpoint(codec, "Call Mic", 0); + snd_soc_dapm_disable_pin(codec, "Audio Out"); + snd_soc_dapm_disable_pin(codec, "GSM Line Out"); + snd_soc_dapm_disable_pin(codec, "GSM Line In"); + snd_soc_dapm_disable_pin(codec, "Headset Mic"); + snd_soc_dapm_disable_pin(codec, "Call Mic"); break; default: - snd_soc_dapm_set_endpoint(codec, "Audio Out", 0); - snd_soc_dapm_set_endpoint(codec, "GSM Line Out", 0); - snd_soc_dapm_set_endpoint(codec, "GSM Line In", 0); - snd_soc_dapm_set_endpoint(codec, "Headset Mic", 0); - snd_soc_dapm_set_endpoint(codec, "Call Mic", 0); + snd_soc_dapm_disable_pin(codec, "Audio Out"); + snd_soc_dapm_disable_pin(codec, "GSM Line Out"); + snd_soc_dapm_disable_pin(codec, "GSM Line In"); + snd_soc_dapm_disable_pin(codec, "Headset Mic"); + snd_soc_dapm_disable_pin(codec, "Call Mic"); } - snd_soc_dapm_sync_endpoints(codec); + snd_soc_dapm_sync(codec); return 0; } @@ -315,6 +330,8 @@ static int neo1973_set_scenario(struct snd_kcontrol *kcontrol, { struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); + DBG("Entered %s\n", __func__); + if (neo1973_scenario == ucontrol->value.integer.value[0]) return 0; @@ -327,6 +344,8 @@ static u8 lm4857_regs[4] = {0x00, 0x40, 0x80, 0xC0}; static void lm4857_write_regs(void) { + DBG("Entered %s\n", __func__); + if (i2c_master_send(i2c, lm4857_regs, 4) != 4) printk(KERN_ERR "lm4857: i2c write failed\n"); } @@ -338,6 +357,8 @@ static int lm4857_get_reg(struct snd_kcontrol *kcontrol, int shift = (kcontrol->private_value >> 8) & 0x0F; int mask = (kcontrol->private_value >> 16) & 0xFF; + DBG("Entered %s\n", __func__); + ucontrol->value.integer.value[0] = (lm4857_regs[reg] >> shift) & mask; return 0; } @@ -364,6 +385,8 @@ static int lm4857_get_mode(struct snd_kcontrol *kcontrol, { u8 value = lm4857_regs[LM4857_CTRL] & 0x0F; + DBG("Entered %s\n", __func__); + if (value) value -= 5; @@ -376,6 +399,8 @@ static int lm4857_set_mode(struct snd_kcontrol *kcontrol, { u8 value = ucontrol->value.integer.value[0]; + DBG("Entered %s\n", __func__); + if (value) value += 5; @@ -397,8 +422,7 @@ static const struct snd_soc_dapm_widget wm8753_dapm_widgets[] = { }; -/* example machine audio_mapnections */ -static const char *audio_map[][3] = { +static const struct snd_soc_dapm_route dapm_routes[] = { /* Connections to the lm4857 amp */ {"Audio Out", NULL, "LOUT1"}, @@ -421,8 +445,6 @@ static const char *audio_map[][3] = { /* Connect the ALC pins */ {"ACIN", NULL, "ACOP"}, - - {NULL, NULL, NULL}, }; static const char *lm4857_mode[] = { @@ -453,13 +475,16 @@ static const struct soc_enum neo_scenario_enum[] = { SOC_ENUM_SINGLE_EXT(ARRAY_SIZE(neo_scenarios), neo_scenarios), }; +static const DECLARE_TLV_DB_SCALE(stereo_tlv, -4050, 150, 0); +static const DECLARE_TLV_DB_SCALE(mono_tlv, -3450, 150, 0); + static const struct snd_kcontrol_new wm8753_neo1973_controls[] = { - SOC_SINGLE_EXT("Amp Left Playback Volume", LM4857_LVOL, 0, 31, 0, - lm4857_get_reg, lm4857_set_reg), - SOC_SINGLE_EXT("Amp Right Playback Volume", LM4857_RVOL, 0, 31, 0, - lm4857_get_reg, lm4857_set_reg), - SOC_SINGLE_EXT("Amp Mono Playback Volume", LM4857_MVOL, 0, 31, 0, - lm4857_get_reg, lm4857_set_reg), + SOC_SINGLE_EXT_TLV("Amp Left Playback Volume", LM4857_LVOL, 0, 31, 0, + lm4857_get_reg, lm4857_set_reg, stereo_tlv), + SOC_SINGLE_EXT_TLV("Amp Right Playback Volume", LM4857_RVOL, 0, 31, 0, + lm4857_get_reg, lm4857_set_reg, stereo_tlv), + SOC_SINGLE_EXT_TLV("Amp Mono Playback Volume", LM4857_MVOL, 0, 31, 0, + lm4857_get_reg, lm4857_set_reg, mono_tlv), SOC_ENUM_EXT("Amp Mode", lm4857_mode_enum[0], lm4857_get_mode, lm4857_set_mode), SOC_ENUM_EXT("Neo Mode", neo_scenario_enum[0], @@ -483,21 +508,23 @@ static int neo1973_wm8753_init(struct snd_soc_codec *codec) { int i, err; + DBG("Entered %s\n", __func__); + /* set up NC codec pins */ - snd_soc_dapm_set_endpoint(codec, "LOUT2", 0); - snd_soc_dapm_set_endpoint(codec, "ROUT2", 0); - snd_soc_dapm_set_endpoint(codec, "OUT3", 0); - snd_soc_dapm_set_endpoint(codec, "OUT4", 0); - snd_soc_dapm_set_endpoint(codec, "LINE1", 0); - snd_soc_dapm_set_endpoint(codec, "LINE2", 0); + snd_soc_dapm_disable_pin(codec, "LOUT2"); + snd_soc_dapm_disable_pin(codec, "ROUT2"); + snd_soc_dapm_disable_pin(codec, "OUT3"); + snd_soc_dapm_disable_pin(codec, "OUT4"); + snd_soc_dapm_disable_pin(codec, "LINE1"); + snd_soc_dapm_disable_pin(codec, "LINE2"); /* set endpoints to default mode */ set_scenario_endpoints(codec, NEO_AUDIO_OFF); /* Add neo1973 specific widgets */ - for (i = 0; i < ARRAY_SIZE(wm8753_dapm_widgets); i++) - snd_soc_dapm_new_control(codec, &wm8753_dapm_widgets[i]); + snd_soc_dapm_new_controls(codec, wm8753_dapm_widgets, + ARRAY_SIZE(wm8753_dapm_widgets)); /* add neo1973 specific controls */ for (i = 0; i < ARRAY_SIZE(wm8753_neo1973_controls); i++) { @@ -508,20 +535,18 @@ static int neo1973_wm8753_init(struct snd_soc_codec *codec) return err; } - /* set up neo1973 specific audio path audio_mapnects */ - for (i = 0; audio_map[i][0] != NULL; i++) { - snd_soc_dapm_connect_input(codec, audio_map[i][0], - audio_map[i][1], audio_map[i][2]); - } + /* set up neo1973 specific audio routes */ + err = snd_soc_dapm_add_routes(codec, dapm_routes, + ARRAY_SIZE(dapm_routes)); - snd_soc_dapm_sync_endpoints(codec); + snd_soc_dapm_sync(codec); return 0; } /* * BT Codec DAI */ -static struct snd_soc_cpu_dai bt_dai = { +static struct snd_soc_dai bt_dai = { .name = "Bluetooth", .id = 0, .type = SND_SOC_DAI_PCM, @@ -583,6 +608,8 @@ static int lm4857_amp_probe(struct i2c_adapter *adap, int addr, int kind) { int ret; + DBG("Entered %s\n", __func__); + client_template.adapter = adap; client_template.addr = addr; @@ -606,6 +633,8 @@ exit_err: static int lm4857_i2c_detach(struct i2c_client *client) { + DBG("Entered %s\n", __func__); + i2c_detach_client(client); kfree(client); return 0; @@ -613,6 +642,8 @@ static int lm4857_i2c_detach(struct i2c_client *client) static int lm4857_i2c_attach(struct i2c_adapter *adap) { + DBG("Entered %s\n", __func__); + return i2c_probe(adap, &addr_data, lm4857_amp_probe); } @@ -620,6 +651,8 @@ static u8 lm4857_state; static int lm4857_suspend(struct i2c_client *dev, pm_message_t state) { + DBG("Entered %s\n", __func__); + dev_dbg(&dev->dev, "lm4857_suspend\n"); lm4857_state = lm4857_regs[LM4857_CTRL] & 0xf; if (lm4857_state) { @@ -631,6 +664,8 @@ static int lm4857_suspend(struct i2c_client *dev, pm_message_t state) static int lm4857_resume(struct i2c_client *dev) { + DBG("Entered %s\n", __func__); + if (lm4857_state) { lm4857_regs[LM4857_CTRL] |= (lm4857_state & 0x0f); lm4857_write_regs(); @@ -640,6 +675,8 @@ static int lm4857_resume(struct i2c_client *dev) static void lm4857_shutdown(struct i2c_client *dev) { + DBG("Entered %s\n", __func__); + dev_dbg(&dev->dev, "lm4857_shutdown\n"); lm4857_regs[LM4857_CTRL] &= 0xf0; lm4857_write_regs(); @@ -671,6 +708,8 @@ static int __init neo1973_init(void) { int ret; + DBG("Entered %s\n", __func__); + neo1973_snd_device = platform_device_alloc("soc-audio", -1); if (!neo1973_snd_device) return -ENOMEM; @@ -691,6 +730,8 @@ static int __init neo1973_init(void) static void __exit neo1973_exit(void) { + DBG("Entered %s\n", __func__); + i2c_del_driver(&lm4857_i2c_driver); platform_device_unregister(neo1973_snd_device); } diff --git a/sound/soc/s3c24xx/s3c2412-i2s.c b/sound/soc/s3c24xx/s3c2412-i2s.c index c4a46dd589b..ee4676ed128 100644 --- a/sound/soc/s3c24xx/s3c2412-i2s.c +++ b/sound/soc/s3c24xx/s3c2412-i2s.c @@ -295,7 +295,7 @@ static inline int s3c2412_snd_is_clkmaster(void) /* * Set S3C2412 I2S DAI format */ -static int s3c2412_i2s_set_fmt(struct snd_soc_cpu_dai *cpu_dai, +static int s3c2412_i2s_set_fmt(struct snd_soc_dai *cpu_dai, unsigned int fmt) { u32 iismod; @@ -500,7 +500,7 @@ EXPORT_SYMBOL_GPL(s3c2412_iis_calc_rate); /* * Set S3C2412 Clock source */ -static int s3c2412_i2s_set_sysclk(struct snd_soc_cpu_dai *cpu_dai, +static int s3c2412_i2s_set_sysclk(struct snd_soc_dai *cpu_dai, int clk_id, unsigned int freq, int dir) { u32 iismod = readl(s3c2412_i2s.regs + S3C2412_IISMOD); @@ -528,7 +528,7 @@ static int s3c2412_i2s_set_sysclk(struct snd_soc_cpu_dai *cpu_dai, /* * Set S3C2412 Clock dividers */ -static int s3c2412_i2s_set_clkdiv(struct snd_soc_cpu_dai *cpu_dai, +static int s3c2412_i2s_set_clkdiv(struct snd_soc_dai *cpu_dai, int div_id, int div) { struct s3c2412_i2s_info *i2s = &s3c2412_i2s; @@ -601,7 +601,8 @@ struct clk *s3c2412_get_iisclk(void) EXPORT_SYMBOL_GPL(s3c2412_get_iisclk); -static int s3c2412_i2s_probe(struct platform_device *pdev) +static int s3c2412_i2s_probe(struct platform_device *pdev, + struct snd_soc_dai *dai) { DBG("Entered %s\n", __func__); @@ -647,7 +648,7 @@ static int s3c2412_i2s_probe(struct platform_device *pdev) #ifdef CONFIG_PM static int s3c2412_i2s_suspend(struct platform_device *dev, - struct snd_soc_cpu_dai *dai) + struct snd_soc_dai *dai) { struct s3c2412_i2s_info *i2s = &s3c2412_i2s; u32 iismod; @@ -675,7 +676,7 @@ static int s3c2412_i2s_suspend(struct platform_device *dev, } static int s3c2412_i2s_resume(struct platform_device *pdev, - struct snd_soc_cpu_dai *dai) + struct snd_soc_dai *dai) { struct s3c2412_i2s_info *i2s = &s3c2412_i2s; @@ -707,7 +708,7 @@ static int s3c2412_i2s_resume(struct platform_device *pdev, SNDRV_PCM_RATE_22050 | SNDRV_PCM_RATE_32000 | SNDRV_PCM_RATE_44100 | \ SNDRV_PCM_RATE_48000 | SNDRV_PCM_RATE_88200 | SNDRV_PCM_RATE_96000) -struct snd_soc_cpu_dai s3c2412_i2s_dai = { +struct snd_soc_dai s3c2412_i2s_dai = { .name = "s3c2412-i2s", .id = 0, .type = SND_SOC_DAI_I2S, diff --git a/sound/soc/s3c24xx/s3c2412-i2s.h b/sound/soc/s3c24xx/s3c2412-i2s.h index 27f48e1ffa8..aac08a25e54 100644 --- a/sound/soc/s3c24xx/s3c2412-i2s.h +++ b/sound/soc/s3c24xx/s3c2412-i2s.h @@ -24,7 +24,7 @@ extern struct clk *s3c2412_get_iisclk(void); -extern struct snd_soc_cpu_dai s3c2412_i2s_dai; +extern struct snd_soc_dai s3c2412_i2s_dai; struct s3c2412_rate_calc { unsigned int clk_div; /* for prescaler */ diff --git a/sound/soc/s3c24xx/s3c2443-ac97.c b/sound/soc/s3c24xx/s3c2443-ac97.c index e81d9a6c83d..783349b7fed 100644 --- a/sound/soc/s3c24xx/s3c2443-ac97.c +++ b/sound/soc/s3c24xx/s3c2443-ac97.c @@ -10,9 +10,6 @@ * This program is free software; you can redistribute it and/or modify * it under the terms of the GNU General Public License version 2 as * published by the Free Software Foundation. - * - * Revision history - * 21st Mar 2007 Initial Version */ #include <linux/init.h> @@ -212,7 +209,8 @@ static struct s3c24xx_pcm_dma_params s3c2443_ac97_mic_mono_in = { .dma_size = 4, }; -static int s3c2443_ac97_probe(struct platform_device *pdev) +static int s3c2443_ac97_probe(struct platform_device *pdev, + struct snd_soc_dai *dai) { int ret; u32 ac_glbctrl; @@ -263,7 +261,8 @@ static int s3c2443_ac97_probe(struct platform_device *pdev) return ret; } -static void s3c2443_ac97_remove(struct platform_device *pdev) +static void s3c2443_ac97_remove(struct platform_device *pdev, + struct snd_soc_dai *dai) { free_irq(IRQ_S3C244x_AC97, NULL); clk_disable(s3c24xx_ac97.ac97_clk); @@ -275,7 +274,7 @@ static int s3c2443_ac97_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params) { struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_cpu_dai *cpu_dai = rtd->dai->cpu_dai; + struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai; if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) cpu_dai->dma_data = &s3c2443_ac97_pcm_stereo_out; @@ -317,7 +316,7 @@ static int s3c2443_ac97_hw_mic_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params) { struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_cpu_dai *cpu_dai = rtd->dai->cpu_dai; + struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai; if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) return -ENODEV; @@ -353,7 +352,7 @@ static int s3c2443_ac97_mic_trigger(struct snd_pcm_substream *substream, SNDRV_PCM_RATE_16000 | SNDRV_PCM_RATE_22050 | \ SNDRV_PCM_RATE_44100 | SNDRV_PCM_RATE_48000) -struct snd_soc_cpu_dai s3c2443_ac97_dai[] = { +struct snd_soc_dai s3c2443_ac97_dai[] = { { .name = "s3c2443-ac97", .id = 0, diff --git a/sound/soc/s3c24xx/s3c24xx-ac97.h b/sound/soc/s3c24xx/s3c24xx-ac97.h index bf03e8ed16c..a96dcadf28b 100644 --- a/sound/soc/s3c24xx/s3c24xx-ac97.h +++ b/sound/soc/s3c24xx/s3c24xx-ac97.h @@ -26,6 +26,6 @@ #define IRQ_S3C244x_AC97 IRQ_S3C2443_AC97 #endif -extern struct snd_soc_cpu_dai s3c2443_ac97_dai[]; +extern struct snd_soc_dai s3c2443_ac97_dai[]; #endif /*S3C24XXAC97_H_*/ diff --git a/sound/soc/s3c24xx/s3c24xx-i2s.c b/sound/soc/s3c24xx/s3c24xx-i2s.c index 1ed6afd4545..397524282b5 100644 --- a/sound/soc/s3c24xx/s3c24xx-i2s.c +++ b/sound/soc/s3c24xx/s3c24xx-i2s.c @@ -12,11 +12,6 @@ * under the terms of the GNU General Public License as published by the * Free Software Foundation; either version 2 of the License, or (at your * option) any later version. - * - * - * Revision history - * 11th Dec 2006 Merged with Simtec driver - * 10th Nov 2006 Initial version. */ #include <linux/init.h> @@ -180,7 +175,7 @@ static void s3c24xx_snd_rxctrl(int on) static int s3c24xx_snd_lrsync(void) { u32 iiscon; - unsigned long timeout = jiffies + msecs_to_jiffies(5); + int timeout = 50; /* 5ms */ DBG("Entered %s\n", __func__); @@ -189,8 +184,9 @@ static int s3c24xx_snd_lrsync(void) if (iiscon & S3C2410_IISCON_LRINDEX) break; - if (time_after(jiffies, timeout)) + if (!timeout--) return -ETIMEDOUT; + udelay(100); } return 0; @@ -209,7 +205,7 @@ static inline int s3c24xx_snd_is_clkmaster(void) /* * Set S3C24xx I2S DAI format */ -static int s3c24xx_i2s_set_fmt(struct snd_soc_cpu_dai *cpu_dai, +static int s3c24xx_i2s_set_fmt(struct snd_soc_dai *cpu_dai, unsigned int fmt) { u32 iismod; @@ -317,7 +313,7 @@ exit_err: /* * Set S3C24xx Clock source */ -static int s3c24xx_i2s_set_sysclk(struct snd_soc_cpu_dai *cpu_dai, +static int s3c24xx_i2s_set_sysclk(struct snd_soc_dai *cpu_dai, int clk_id, unsigned int freq, int dir) { u32 iismod = readl(s3c24xx_i2s.regs + S3C2410_IISMOD); @@ -343,7 +339,7 @@ static int s3c24xx_i2s_set_sysclk(struct snd_soc_cpu_dai *cpu_dai, /* * Set S3C24xx Clock dividers */ -static int s3c24xx_i2s_set_clkdiv(struct snd_soc_cpu_dai *cpu_dai, +static int s3c24xx_i2s_set_clkdiv(struct snd_soc_dai *cpu_dai, int div_id, int div) { u32 reg; @@ -381,7 +377,8 @@ u32 s3c24xx_i2s_get_clockrate(void) } EXPORT_SYMBOL_GPL(s3c24xx_i2s_get_clockrate); -static int s3c24xx_i2s_probe(struct platform_device *pdev) +static int s3c24xx_i2s_probe(struct platform_device *pdev, + struct snd_soc_dai *dai) { DBG("Entered %s\n", __func__); @@ -414,7 +411,7 @@ static int s3c24xx_i2s_probe(struct platform_device *pdev) #ifdef CONFIG_PM static int s3c24xx_i2s_suspend(struct platform_device *pdev, - struct snd_soc_cpu_dai *cpu_dai) + struct snd_soc_dai *cpu_dai) { DBG("Entered %s\n", __func__); @@ -429,7 +426,7 @@ static int s3c24xx_i2s_suspend(struct platform_device *pdev, } static int s3c24xx_i2s_resume(struct platform_device *pdev, - struct snd_soc_cpu_dai *cpu_dai) + struct snd_soc_dai *cpu_dai) { DBG("Entered %s\n", __func__); clk_enable(s3c24xx_i2s.iis_clk); @@ -452,7 +449,7 @@ static int s3c24xx_i2s_resume(struct platform_device *pdev, SNDRV_PCM_RATE_22050 | SNDRV_PCM_RATE_32000 | SNDRV_PCM_RATE_44100 | \ SNDRV_PCM_RATE_48000 | SNDRV_PCM_RATE_88200 | SNDRV_PCM_RATE_96000) -struct snd_soc_cpu_dai s3c24xx_i2s_dai = { +struct snd_soc_dai s3c24xx_i2s_dai = { .name = "s3c24xx-i2s", .id = 0, .type = SND_SOC_DAI_I2S, diff --git a/sound/soc/s3c24xx/s3c24xx-i2s.h b/sound/soc/s3c24xx/s3c24xx-i2s.h index 537b4ecce8a..726d91cf4e1 100644 --- a/sound/soc/s3c24xx/s3c24xx-i2s.h +++ b/sound/soc/s3c24xx/s3c24xx-i2s.h @@ -32,6 +32,6 @@ u32 s3c24xx_i2s_get_clockrate(void); -extern struct snd_soc_cpu_dai s3c24xx_i2s_dai; +extern struct snd_soc_dai s3c24xx_i2s_dai; #endif /*S3C24XXI2S_H_*/ diff --git a/sound/soc/s3c24xx/s3c24xx-pcm.c b/sound/soc/s3c24xx/s3c24xx-pcm.c index 7806ae61461..cef79b34dc6 100644 --- a/sound/soc/s3c24xx/s3c24xx-pcm.c +++ b/sound/soc/s3c24xx/s3c24xx-pcm.c @@ -12,10 +12,6 @@ * under the terms of the GNU General Public License as published by the * Free Software Foundation; either version 2 of the License, or (at your * option) any later version. - * - * Revision history - * 11th Dec 2006 Merged with Simtec driver - * 10th Nov 2006 Initial version. */ #include <linux/module.h> @@ -433,7 +429,7 @@ static void s3c24xx_pcm_free_dma_buffers(struct snd_pcm *pcm) static u64 s3c24xx_pcm_dmamask = DMA_32BIT_MASK; static int s3c24xx_pcm_new(struct snd_card *card, - struct snd_soc_codec_dai *dai, struct snd_pcm *pcm) + struct snd_soc_dai *dai, struct snd_pcm *pcm) { int ret = 0; diff --git a/sound/soc/s3c24xx/smdk2443_wm9710.c b/sound/soc/s3c24xx/smdk2443_wm9710.c index b4a56302b9a..8515d6ff03f 100644 --- a/sound/soc/s3c24xx/smdk2443_wm9710.c +++ b/sound/soc/s3c24xx/smdk2443_wm9710.c @@ -10,9 +10,6 @@ * Free Software Foundation; either version 2 of the License, or (at your * option) any later version. * - * Revision history - * 8th Mar 2007 Initial version. - * */ #include <linux/module.h> diff --git a/sound/soc/sh/Kconfig b/sound/soc/sh/Kconfig index 4c1e013381c..54bd604012a 100644 --- a/sound/soc/sh/Kconfig +++ b/sound/soc/sh/Kconfig @@ -3,7 +3,7 @@ menu "SoC Audio support for SuperH" config SND_SOC_PCM_SH7760 tristate "SoC Audio support for Renesas SH7760" - depends on CPU_SUBTYPE_SH7760 && SND_SOC && SH_DMABRG + depends on CPU_SUBTYPE_SH7760 && SH_DMABRG help Enable this option for SH7760 AC97/I2S audio support. @@ -13,10 +13,9 @@ config SND_SOC_PCM_SH7760 ## config SND_SOC_SH4_HAC + tristate select AC97_BUS select SND_SOC_AC97_BUS - select SND_AC97_CODEC - tristate config SND_SOC_SH4_SSI tristate diff --git a/sound/soc/sh/dma-sh7760.c b/sound/soc/sh/dma-sh7760.c index 7a3ce80d672..9faa12622d0 100644 --- a/sound/soc/sh/dma-sh7760.c +++ b/sound/soc/sh/dma-sh7760.c @@ -326,7 +326,7 @@ static void camelot_pcm_free(struct snd_pcm *pcm) } static int camelot_pcm_new(struct snd_card *card, - struct snd_soc_codec_dai *dai, + struct snd_soc_dai *dai, struct snd_pcm *pcm) { /* dont use SNDRV_DMA_TYPE_DEV, since it will oops the SH kernel diff --git a/sound/soc/sh/hac.c b/sound/soc/sh/hac.c index b7b676b3d67..df7bc345c32 100644 --- a/sound/soc/sh/hac.c +++ b/sound/soc/sh/hac.c @@ -266,7 +266,7 @@ static int hac_hw_params(struct snd_pcm_substream *substream, #define AC97_FMTS \ SNDRV_PCM_FMTBIT_S16_LE -struct snd_soc_cpu_dai sh4_hac_dai[] = { +struct snd_soc_dai sh4_hac_dai[] = { { .name = "HAC0", .id = 0, diff --git a/sound/soc/sh/sh7760-ac97.c b/sound/soc/sh/sh7760-ac97.c index 2f91de84c5c..92bfaf4774a 100644 --- a/sound/soc/sh/sh7760-ac97.c +++ b/sound/soc/sh/sh7760-ac97.c @@ -20,12 +20,12 @@ #define IPSEL 0xFE400034 /* platform specific structs can be declared here */ -extern struct snd_soc_cpu_dai sh4_hac_dai[2]; +extern struct snd_soc_dai sh4_hac_dai[2]; extern struct snd_soc_platform sh7760_soc_platform; static int machine_init(struct snd_soc_codec *codec) { - snd_soc_dapm_sync_endpoints(codec); + snd_soc_dapm_sync(codec); return 0; } diff --git a/sound/soc/sh/ssi.c b/sound/soc/sh/ssi.c index 3388bc3d62d..55c3464163a 100644 --- a/sound/soc/sh/ssi.c +++ b/sound/soc/sh/ssi.c @@ -208,7 +208,7 @@ static int ssi_hw_params(struct snd_pcm_substream *substream, return 0; } -static int ssi_set_sysclk(struct snd_soc_cpu_dai *cpu_dai, int clk_id, +static int ssi_set_sysclk(struct snd_soc_dai *cpu_dai, int clk_id, unsigned int freq, int dir) { struct ssi_priv *ssi = &ssi_cpu_data[cpu_dai->id]; @@ -222,7 +222,7 @@ static int ssi_set_sysclk(struct snd_soc_cpu_dai *cpu_dai, int clk_id, * This divider is used to generate the SSI_SCK (I2S bitclock) from the * clock at the HAC_BIT_CLK ("oversampling clock") pin. */ -static int ssi_set_clkdiv(struct snd_soc_cpu_dai *dai, int did, int div) +static int ssi_set_clkdiv(struct snd_soc_dai *dai, int did, int div) { struct ssi_priv *ssi = &ssi_cpu_data[dai->id]; unsigned long ssicr; @@ -245,7 +245,7 @@ static int ssi_set_clkdiv(struct snd_soc_cpu_dai *dai, int did, int div) return 0; } -static int ssi_set_fmt(struct snd_soc_cpu_dai *dai, unsigned int fmt) +static int ssi_set_fmt(struct snd_soc_dai *dai, unsigned int fmt) { struct ssi_priv *ssi = &ssi_cpu_data[dai->id]; unsigned long ssicr = SSIREG(SSICR); @@ -332,7 +332,7 @@ static int ssi_set_fmt(struct snd_soc_cpu_dai *dai, unsigned int fmt) SNDRV_PCM_FMTBIT_S24_3LE | SNDRV_PCM_FMTBIT_U24_3LE | \ SNDRV_PCM_FMTBIT_S32_LE | SNDRV_PCM_FMTBIT_U32_LE) -struct snd_soc_cpu_dai sh4_ssi_dai[] = { +struct snd_soc_dai sh4_ssi_dai[] = { { .name = "SSI0", .id = 0, diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index e148db940cf..83f1190293a 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -14,10 +14,6 @@ * Free Software Foundation; either version 2 of the License, or (at your * option) any later version. * - * Revision history - * 12th Aug 2005 Initial version. - * 25th Oct 2005 Working Codec, Interface and Platform registration. - * * TODO: * o Add hw rules to enforce rates, etc. * o More testing with other codecs/machines. @@ -112,9 +108,9 @@ static int soc_ac97_dev_register(struct snd_soc_codec *codec) } #endif -static inline const char* get_dai_name(int type) +static inline const char *get_dai_name(int type) { - switch(type) { + switch (type) { case SND_SOC_DAI_AC97_BUS: case SND_SOC_DAI_AC97: return "AC97"; @@ -138,8 +134,8 @@ static int soc_pcm_open(struct snd_pcm_substream *substream) struct snd_pcm_runtime *runtime = substream->runtime; struct snd_soc_dai_link *machine = rtd->dai; struct snd_soc_platform *platform = socdev->platform; - struct snd_soc_cpu_dai *cpu_dai = machine->cpu_dai; - struct snd_soc_codec_dai *codec_dai = machine->codec_dai; + struct snd_soc_dai *cpu_dai = machine->cpu_dai; + struct snd_soc_dai *codec_dai = machine->codec_dai; int ret = 0; mutex_lock(&pcm_mutex); @@ -182,9 +178,11 @@ static int soc_pcm_open(struct snd_pcm_substream *substream) /* Check that the codec and cpu DAI's are compatible */ if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { runtime->hw.rate_min = - max(codec_dai->playback.rate_min, cpu_dai->playback.rate_min); + max(codec_dai->playback.rate_min, + cpu_dai->playback.rate_min); runtime->hw.rate_max = - min(codec_dai->playback.rate_max, cpu_dai->playback.rate_max); + min(codec_dai->playback.rate_max, + cpu_dai->playback.rate_max); runtime->hw.channels_min = max(codec_dai->playback.channels_min, cpu_dai->playback.channels_min); @@ -197,9 +195,11 @@ static int soc_pcm_open(struct snd_pcm_substream *substream) codec_dai->playback.rates & cpu_dai->playback.rates; } else { runtime->hw.rate_min = - max(codec_dai->capture.rate_min, cpu_dai->capture.rate_min); + max(codec_dai->capture.rate_min, + cpu_dai->capture.rate_min); runtime->hw.rate_max = - min(codec_dai->capture.rate_max, cpu_dai->capture.rate_max); + min(codec_dai->capture.rate_max, + cpu_dai->capture.rate_max); runtime->hw.channels_min = max(codec_dai->capture.channels_min, cpu_dai->capture.channels_min); @@ -229,7 +229,7 @@ static int soc_pcm_open(struct snd_pcm_substream *substream) goto machine_err; } - dbg("asoc: %s <-> %s info:\n",codec_dai->name, cpu_dai->name); + dbg("asoc: %s <-> %s info:\n", codec_dai->name, cpu_dai->name); dbg("asoc: rate mask 0x%x\n", runtime->hw.rates); dbg("asoc: min ch %d max ch %d\n", runtime->hw.channels_min, runtime->hw.channels_max); @@ -272,11 +272,11 @@ static void close_delayed_work(struct work_struct *work) struct snd_soc_device *socdev = container_of(work, struct snd_soc_device, delayed_work.work); struct snd_soc_codec *codec = socdev->codec; - struct snd_soc_codec_dai *codec_dai; + struct snd_soc_dai *codec_dai; int i; mutex_lock(&pcm_mutex); - for(i = 0; i < codec->num_dai; i++) { + for (i = 0; i < codec->num_dai; i++) { codec_dai = &codec->dai[i]; dbg("pop wq checking: %s status: %s waiting: %s\n", @@ -287,12 +287,12 @@ static void close_delayed_work(struct work_struct *work) /* are we waiting on this codec DAI stream */ if (codec_dai->pop_wait == 1) { - /* power down the codec to D1 if no longer active */ + /* Reduce power if no longer active */ if (codec->active == 0) { dbg("pop wq D1 %s %s\n", codec->name, codec_dai->playback.stream_name); - snd_soc_dapm_device_event(socdev, - SNDRV_CTL_POWER_D1); + snd_soc_dapm_set_bias_level(socdev, + SND_SOC_BIAS_PREPARE); } codec_dai->pop_wait = 0; @@ -300,12 +300,12 @@ static void close_delayed_work(struct work_struct *work) codec_dai->playback.stream_name, SND_SOC_DAPM_STREAM_STOP); - /* power down the codec power domain if no longer active */ + /* Fall into standby if no longer active */ if (codec->active == 0) { dbg("pop wq D3 %s %s\n", codec->name, codec_dai->playback.stream_name); - snd_soc_dapm_device_event(socdev, - SNDRV_CTL_POWER_D3hot); + snd_soc_dapm_set_bias_level(socdev, + SND_SOC_BIAS_STANDBY); } } } @@ -323,8 +323,8 @@ static int soc_codec_close(struct snd_pcm_substream *substream) struct snd_soc_device *socdev = rtd->socdev; struct snd_soc_dai_link *machine = rtd->dai; struct snd_soc_platform *platform = socdev->platform; - struct snd_soc_cpu_dai *cpu_dai = machine->cpu_dai; - struct snd_soc_codec_dai *codec_dai = machine->codec_dai; + struct snd_soc_dai *cpu_dai = machine->cpu_dai; + struct snd_soc_dai *codec_dai = machine->codec_dai; struct snd_soc_codec *codec = socdev->codec; mutex_lock(&pcm_mutex); @@ -365,8 +365,8 @@ static int soc_codec_close(struct snd_pcm_substream *substream) SND_SOC_DAPM_STREAM_STOP); if (codec->active == 0 && codec_dai->pop_wait == 0) - snd_soc_dapm_device_event(socdev, - SNDRV_CTL_POWER_D3hot); + snd_soc_dapm_set_bias_level(socdev, + SND_SOC_BIAS_STANDBY); } mutex_unlock(&pcm_mutex); @@ -384,8 +384,8 @@ static int soc_pcm_prepare(struct snd_pcm_substream *substream) struct snd_soc_device *socdev = rtd->socdev; struct snd_soc_dai_link *machine = rtd->dai; struct snd_soc_platform *platform = socdev->platform; - struct snd_soc_cpu_dai *cpu_dai = machine->cpu_dai; - struct snd_soc_codec_dai *codec_dai = machine->codec_dai; + struct snd_soc_dai *cpu_dai = machine->cpu_dai; + struct snd_soc_dai *codec_dai = machine->codec_dai; struct snd_soc_codec *codec = socdev->codec; int ret = 0; @@ -434,14 +434,14 @@ static int soc_pcm_prepare(struct snd_pcm_substream *substream) else { codec_dai->pop_wait = 0; cancel_delayed_work(&socdev->delayed_work); - if (codec_dai->dai_ops.digital_mute) - codec_dai->dai_ops.digital_mute(codec_dai, 0); + snd_soc_dai_digital_mute(codec_dai, 0); } } else { /* no delayed work - do we need to power up codec */ - if (codec->dapm_state != SNDRV_CTL_POWER_D0) { + if (codec->bias_level != SND_SOC_BIAS_ON) { - snd_soc_dapm_device_event(socdev, SNDRV_CTL_POWER_D1); + snd_soc_dapm_set_bias_level(socdev, + SND_SOC_BIAS_PREPARE); if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) snd_soc_dapm_stream_event(codec, @@ -452,9 +452,8 @@ static int soc_pcm_prepare(struct snd_pcm_substream *substream) codec_dai->capture.stream_name, SND_SOC_DAPM_STREAM_START); - snd_soc_dapm_device_event(socdev, SNDRV_CTL_POWER_D0); - if (codec_dai->dai_ops.digital_mute) - codec_dai->dai_ops.digital_mute(codec_dai, 0); + snd_soc_dapm_set_bias_level(socdev, SND_SOC_BIAS_ON); + snd_soc_dai_digital_mute(codec_dai, 0); } else { /* codec already powered - power on widgets */ @@ -466,8 +465,8 @@ static int soc_pcm_prepare(struct snd_pcm_substream *substream) snd_soc_dapm_stream_event(codec, codec_dai->capture.stream_name, SND_SOC_DAPM_STREAM_START); - if (codec_dai->dai_ops.digital_mute) - codec_dai->dai_ops.digital_mute(codec_dai, 0); + + snd_soc_dai_digital_mute(codec_dai, 0); } } @@ -488,8 +487,8 @@ static int soc_pcm_hw_params(struct snd_pcm_substream *substream, struct snd_soc_device *socdev = rtd->socdev; struct snd_soc_dai_link *machine = rtd->dai; struct snd_soc_platform *platform = socdev->platform; - struct snd_soc_cpu_dai *cpu_dai = machine->cpu_dai; - struct snd_soc_codec_dai *codec_dai = machine->codec_dai; + struct snd_soc_dai *cpu_dai = machine->cpu_dai; + struct snd_soc_dai *codec_dai = machine->codec_dai; int ret = 0; mutex_lock(&pcm_mutex); @@ -514,7 +513,7 @@ static int soc_pcm_hw_params(struct snd_pcm_substream *substream, if (cpu_dai->ops.hw_params) { ret = cpu_dai->ops.hw_params(substream, params); if (ret < 0) { - printk(KERN_ERR "asoc: can't set interface %s hw params\n", + printk(KERN_ERR "asoc: interface %s hw params failed\n", cpu_dai->name); goto interface_err; } @@ -523,7 +522,7 @@ static int soc_pcm_hw_params(struct snd_pcm_substream *substream, if (platform->pcm_ops->hw_params) { ret = platform->pcm_ops->hw_params(substream, params); if (ret < 0) { - printk(KERN_ERR "asoc: can't set platform %s hw params\n", + printk(KERN_ERR "asoc: platform %s hw params failed\n", platform->name); goto platform_err; } @@ -542,7 +541,7 @@ interface_err: codec_dai->ops.hw_free(substream); codec_err: - if(machine->ops && machine->ops->hw_free) + if (machine->ops && machine->ops->hw_free) machine->ops->hw_free(substream); mutex_unlock(&pcm_mutex); @@ -558,15 +557,15 @@ static int soc_pcm_hw_free(struct snd_pcm_substream *substream) struct snd_soc_device *socdev = rtd->socdev; struct snd_soc_dai_link *machine = rtd->dai; struct snd_soc_platform *platform = socdev->platform; - struct snd_soc_cpu_dai *cpu_dai = machine->cpu_dai; - struct snd_soc_codec_dai *codec_dai = machine->codec_dai; + struct snd_soc_dai *cpu_dai = machine->cpu_dai; + struct snd_soc_dai *codec_dai = machine->codec_dai; struct snd_soc_codec *codec = socdev->codec; mutex_lock(&pcm_mutex); /* apply codec digital mute */ - if (!codec->active && codec_dai->dai_ops.digital_mute) - codec_dai->dai_ops.digital_mute(codec_dai, 1); + if (!codec->active) + snd_soc_dai_digital_mute(codec_dai, 1); /* free any machine hw params */ if (machine->ops && machine->ops->hw_free) @@ -593,8 +592,8 @@ static int soc_pcm_trigger(struct snd_pcm_substream *substream, int cmd) struct snd_soc_device *socdev = rtd->socdev; struct snd_soc_dai_link *machine = rtd->dai; struct snd_soc_platform *platform = socdev->platform; - struct snd_soc_cpu_dai *cpu_dai = machine->cpu_dai; - struct snd_soc_codec_dai *codec_dai = machine->codec_dai; + struct snd_soc_dai *cpu_dai = machine->cpu_dai; + struct snd_soc_dai *codec_dai = machine->codec_dai; int ret; if (codec_dai->ops.trigger) { @@ -631,16 +630,26 @@ static struct snd_pcm_ops soc_pcm_ops = { /* powers down audio subsystem for suspend */ static int soc_suspend(struct platform_device *pdev, pm_message_t state) { - struct snd_soc_device *socdev = platform_get_drvdata(pdev); - struct snd_soc_machine *machine = socdev->machine; - struct snd_soc_platform *platform = socdev->platform; - struct snd_soc_codec_device *codec_dev = socdev->codec_dev; + struct snd_soc_device *socdev = platform_get_drvdata(pdev); + struct snd_soc_machine *machine = socdev->machine; + struct snd_soc_platform *platform = socdev->platform; + struct snd_soc_codec_device *codec_dev = socdev->codec_dev; struct snd_soc_codec *codec = socdev->codec; int i; + /* Due to the resume being scheduled into a workqueue we could + * suspend before that's finished - wait for it to complete. + */ + snd_power_lock(codec->card); + snd_power_wait(codec->card, SNDRV_CTL_POWER_D0); + snd_power_unlock(codec->card); + + /* we're going to block userspace touching us until resume completes */ + snd_power_change_state(codec->card, SNDRV_CTL_POWER_D3hot); + /* mute any active DAC's */ - for(i = 0; i < machine->num_links; i++) { - struct snd_soc_codec_dai *dai = machine->dai_link[i].codec_dai; + for (i = 0; i < machine->num_links; i++) { + struct snd_soc_dai *dai = machine->dai_link[i].codec_dai; if (dai->dai_ops.digital_mute && dai->playback.active) dai->dai_ops.digital_mute(dai, 1); } @@ -652,8 +661,8 @@ static int soc_suspend(struct platform_device *pdev, pm_message_t state) if (machine->suspend_pre) machine->suspend_pre(pdev, state); - for(i = 0; i < machine->num_links; i++) { - struct snd_soc_cpu_dai *cpu_dai = machine->dai_link[i].cpu_dai; + for (i = 0; i < machine->num_links; i++) { + struct snd_soc_dai *cpu_dai = machine->dai_link[i].cpu_dai; if (cpu_dai->suspend && cpu_dai->type != SND_SOC_DAI_AC97) cpu_dai->suspend(pdev, cpu_dai); if (platform->suspend) @@ -662,9 +671,9 @@ static int soc_suspend(struct platform_device *pdev, pm_message_t state) /* close any waiting streams and save state */ run_delayed_work(&socdev->delayed_work); - codec->suspend_dapm_state = codec->dapm_state; + codec->suspend_bias_level = codec->bias_level; - for(i = 0; i < codec->num_dai; i++) { + for (i = 0; i < codec->num_dai; i++) { char *stream = codec->dai[i].playback.stream_name; if (stream != NULL) snd_soc_dapm_stream_event(codec, stream, @@ -678,8 +687,8 @@ static int soc_suspend(struct platform_device *pdev, pm_message_t state) if (codec_dev->suspend) codec_dev->suspend(pdev, state); - for(i = 0; i < machine->num_links; i++) { - struct snd_soc_cpu_dai *cpu_dai = machine->dai_link[i].cpu_dai; + for (i = 0; i < machine->num_links; i++) { + struct snd_soc_dai *cpu_dai = machine->dai_link[i].cpu_dai; if (cpu_dai->suspend && cpu_dai->type == SND_SOC_DAI_AC97) cpu_dai->suspend(pdev, cpu_dai); } @@ -690,21 +699,32 @@ static int soc_suspend(struct platform_device *pdev, pm_message_t state) return 0; } -/* powers up audio subsystem after a suspend */ -static int soc_resume(struct platform_device *pdev) +/* deferred resume work, so resume can complete before we finished + * setting our codec back up, which can be very slow on I2C + */ +static void soc_resume_deferred(struct work_struct *work) { - struct snd_soc_device *socdev = platform_get_drvdata(pdev); - struct snd_soc_machine *machine = socdev->machine; - struct snd_soc_platform *platform = socdev->platform; - struct snd_soc_codec_device *codec_dev = socdev->codec_dev; + struct snd_soc_device *socdev = container_of(work, + struct snd_soc_device, + deferred_resume_work); + struct snd_soc_machine *machine = socdev->machine; + struct snd_soc_platform *platform = socdev->platform; + struct snd_soc_codec_device *codec_dev = socdev->codec_dev; struct snd_soc_codec *codec = socdev->codec; + struct platform_device *pdev = to_platform_device(socdev->dev); int i; + /* our power state is still SNDRV_CTL_POWER_D3hot from suspend time, + * so userspace apps are blocked from touching us + */ + + dev_info(socdev->dev, "starting resume work\n"); + if (machine->resume_pre) machine->resume_pre(pdev); - for(i = 0; i < machine->num_links; i++) { - struct snd_soc_cpu_dai *cpu_dai = machine->dai_link[i].cpu_dai; + for (i = 0; i < machine->num_links; i++) { + struct snd_soc_dai *cpu_dai = machine->dai_link[i].cpu_dai; if (cpu_dai->resume && cpu_dai->type == SND_SOC_DAI_AC97) cpu_dai->resume(pdev, cpu_dai); } @@ -712,8 +732,8 @@ static int soc_resume(struct platform_device *pdev) if (codec_dev->resume) codec_dev->resume(pdev); - for(i = 0; i < codec->num_dai; i++) { - char* stream = codec->dai[i].playback.stream_name; + for (i = 0; i < codec->num_dai; i++) { + char *stream = codec->dai[i].playback.stream_name; if (stream != NULL) snd_soc_dapm_stream_event(codec, stream, SND_SOC_DAPM_STREAM_RESUME); @@ -723,15 +743,15 @@ static int soc_resume(struct platform_device *pdev) SND_SOC_DAPM_STREAM_RESUME); } - /* unmute any active DAC's */ - for(i = 0; i < machine->num_links; i++) { - struct snd_soc_codec_dai *dai = machine->dai_link[i].codec_dai; + /* unmute any active DACs */ + for (i = 0; i < machine->num_links; i++) { + struct snd_soc_dai *dai = machine->dai_link[i].codec_dai; if (dai->dai_ops.digital_mute && dai->playback.active) dai->dai_ops.digital_mute(dai, 0); } - for(i = 0; i < machine->num_links; i++) { - struct snd_soc_cpu_dai *cpu_dai = machine->dai_link[i].cpu_dai; + for (i = 0; i < machine->num_links; i++) { + struct snd_soc_dai *cpu_dai = machine->dai_link[i].cpu_dai; if (cpu_dai->resume && cpu_dai->type != SND_SOC_DAI_AC97) cpu_dai->resume(pdev, cpu_dai); if (platform->resume) @@ -741,6 +761,22 @@ static int soc_resume(struct platform_device *pdev) if (machine->resume_post) machine->resume_post(pdev); + dev_info(socdev->dev, "resume work completed\n"); + + /* userspace can access us now we are back as we were before */ + snd_power_change_state(codec->card, SNDRV_CTL_POWER_D0); +} + +/* powers up audio subsystem after a suspend */ +static int soc_resume(struct platform_device *pdev) +{ + struct snd_soc_device *socdev = platform_get_drvdata(pdev); + + dev_info(socdev->dev, "scheduling resume work\n"); + + if (!schedule_work(&socdev->deferred_resume_work)) + dev_err(socdev->dev, "work item may be lost\n"); + return 0; } @@ -760,33 +796,38 @@ static int soc_probe(struct platform_device *pdev) if (machine->probe) { ret = machine->probe(pdev); - if(ret < 0) + if (ret < 0) return ret; } for (i = 0; i < machine->num_links; i++) { - struct snd_soc_cpu_dai *cpu_dai = machine->dai_link[i].cpu_dai; + struct snd_soc_dai *cpu_dai = machine->dai_link[i].cpu_dai; if (cpu_dai->probe) { - ret = cpu_dai->probe(pdev); - if(ret < 0) + ret = cpu_dai->probe(pdev, cpu_dai); + if (ret < 0) goto cpu_dai_err; } } if (codec_dev->probe) { ret = codec_dev->probe(pdev); - if(ret < 0) + if (ret < 0) goto cpu_dai_err; } if (platform->probe) { ret = platform->probe(pdev); - if(ret < 0) + if (ret < 0) goto platform_err; } /* DAPM stream work */ INIT_DELAYED_WORK(&socdev->delayed_work, close_delayed_work); +#ifdef CONFIG_PM + /* deferred resume work */ + INIT_WORK(&socdev->deferred_resume_work, soc_resume_deferred); +#endif + return 0; platform_err: @@ -795,9 +836,9 @@ platform_err: cpu_dai_err: for (i--; i >= 0; i--) { - struct snd_soc_cpu_dai *cpu_dai = machine->dai_link[i].cpu_dai; + struct snd_soc_dai *cpu_dai = machine->dai_link[i].cpu_dai; if (cpu_dai->remove) - cpu_dai->remove(pdev); + cpu_dai->remove(pdev, cpu_dai); } if (machine->remove) @@ -824,9 +865,9 @@ static int soc_remove(struct platform_device *pdev) codec_dev->remove(pdev); for (i = 0; i < machine->num_links; i++) { - struct snd_soc_cpu_dai *cpu_dai = machine->dai_link[i].cpu_dai; + struct snd_soc_dai *cpu_dai = machine->dai_link[i].cpu_dai; if (cpu_dai->remove) - cpu_dai->remove(pdev); + cpu_dai->remove(pdev, cpu_dai); } if (machine->remove) @@ -852,8 +893,8 @@ static int soc_new_pcm(struct snd_soc_device *socdev, struct snd_soc_dai_link *dai_link, int num) { struct snd_soc_codec *codec = socdev->codec; - struct snd_soc_codec_dai *codec_dai = dai_link->codec_dai; - struct snd_soc_cpu_dai *cpu_dai = dai_link->cpu_dai; + struct snd_soc_dai *codec_dai = dai_link->codec_dai; + struct snd_soc_dai *cpu_dai = dai_link->cpu_dai; struct snd_soc_pcm_runtime *rtd; struct snd_pcm *pcm; char new_name[64]; @@ -868,7 +909,7 @@ static int soc_new_pcm(struct snd_soc_device *socdev, codec_dai->codec = socdev->codec; /* check client and interface hw capabilities */ - sprintf(new_name, "%s %s-%s-%d",dai_link->stream_name, codec_dai->name, + sprintf(new_name, "%s %s-%s-%d", dai_link->stream_name, codec_dai->name, get_dai_name(cpu_dai->type), num); if (codec_dai->playback.channels_min) @@ -879,7 +920,8 @@ static int soc_new_pcm(struct snd_soc_device *socdev, ret = snd_pcm_new(codec->card, new_name, codec->pcm_devs++, playback, capture, &pcm); if (ret < 0) { - printk(KERN_ERR "asoc: can't create pcm for codec %s\n", codec->name); + printk(KERN_ERR "asoc: can't create pcm for codec %s\n", + codec->name); kfree(rtd); return ret; } @@ -928,8 +970,9 @@ static ssize_t codec_reg_show(struct device *dev, step = codec->reg_cache_step; count += sprintf(buf, "%s registers\n", codec->name); - for(i = 0; i < codec->reg_cache_size; i += step) - count += sprintf(buf + count, "%2x: %4x\n", i, codec->read(codec, i)); + for (i = 0; i < codec->reg_cache_size; i += step) + count += sprintf(buf + count, "%2x: %4x\n", i, + codec->read(codec, i)); return count; } @@ -1072,7 +1115,7 @@ int snd_soc_new_pcms(struct snd_soc_device *socdev, int idx, const char *xid) strncpy(codec->card->driver, codec->name, sizeof(codec->card->driver)); /* create the pcms */ - for(i = 0; i < machine->num_links; i++) { + for (i = 0; i < machine->num_links; i++) { ret = soc_new_pcm(socdev, &machine->dai_link[i], i); if (ret < 0) { printk(KERN_ERR "asoc: can't create pcm %s\n", @@ -1102,7 +1145,7 @@ int snd_soc_register_card(struct snd_soc_device *socdev) struct snd_soc_machine *machine = socdev->machine; int ret = 0, i, ac97 = 0, err = 0; - for(i = 0; i < machine->num_links; i++) { + for (i = 0; i < machine->num_links; i++) { if (socdev->machine->dai_link[i].init) { err = socdev->machine->dai_link[i].init(codec); if (err < 0) { @@ -1111,7 +1154,7 @@ int snd_soc_register_card(struct snd_soc_device *socdev) continue; } } - if (socdev->machine->dai_link[i].codec_dai->type == + if (socdev->machine->dai_link[i].codec_dai->type == SND_SOC_DAI_AC97_BUS) ac97 = 1; } @@ -1122,7 +1165,7 @@ int snd_soc_register_card(struct snd_soc_device *socdev) ret = snd_card_register(codec->card); if (ret < 0) { - printk(KERN_ERR "asoc: failed to register soundcard for codec %s\n", + printk(KERN_ERR "asoc: failed to register soundcard for %s\n", codec->name); goto out; } @@ -1146,7 +1189,7 @@ int snd_soc_register_card(struct snd_soc_device *socdev) err = device_create_file(socdev->dev, &dev_attr_codec_reg); if (err < 0) - printk(KERN_WARNING "asoc: failed to add codec sysfs entries\n"); + printk(KERN_WARNING "asoc: failed to add codec sysfs files\n"); mutex_unlock(&codec->mutex); @@ -1166,13 +1209,13 @@ void snd_soc_free_pcms(struct snd_soc_device *socdev) { struct snd_soc_codec *codec = socdev->codec; #ifdef CONFIG_SND_SOC_AC97_BUS - struct snd_soc_codec_dai *codec_dai; + struct snd_soc_dai *codec_dai; int i; #endif mutex_lock(&codec->mutex); #ifdef CONFIG_SND_SOC_AC97_BUS - for(i = 0; i < codec->num_dai; i++) { + for (i = 0; i < codec->num_dai; i++) { codec_dai = &codec->dai[i]; if (codec_dai->type == SND_SOC_DAI_AC97_BUS && codec->ac97) { soc_ac97_dev_unregister(codec); @@ -1282,7 +1325,8 @@ int snd_soc_get_enum_double(struct snd_kcontrol *kcontrol, for (bitmask = 1; bitmask < e->mask; bitmask <<= 1) ; val = snd_soc_read(codec, e->reg); - ucontrol->value.enumerated.item[0] = (val >> e->shift_l) & (bitmask - 1); + ucontrol->value.enumerated.item[0] + = (val >> e->shift_l) & (bitmask - 1); if (e->shift_l != e->shift_r) ucontrol->value.enumerated.item[1] = (val >> e->shift_r) & (bitmask - 1); @@ -1576,7 +1620,8 @@ int snd_soc_put_volsw_2r(struct snd_kcontrol *kcontrol, val = val << shift; val2 = val2 << shift; - if ((err = snd_soc_update_bits(codec, reg, val_mask, val)) < 0) + err = snd_soc_update_bits(codec, reg, val_mask, val); + if (err < 0) return err; err = snd_soc_update_bits(codec, reg2, val_mask, val2); @@ -1584,6 +1629,204 @@ int snd_soc_put_volsw_2r(struct snd_kcontrol *kcontrol, } EXPORT_SYMBOL_GPL(snd_soc_put_volsw_2r); +/** + * snd_soc_info_volsw_s8 - signed mixer info callback + * @kcontrol: mixer control + * @uinfo: control element information + * + * Callback to provide information about a signed mixer control. + * + * Returns 0 for success. + */ +int snd_soc_info_volsw_s8(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_info *uinfo) +{ + int max = (signed char)((kcontrol->private_value >> 16) & 0xff); + int min = (signed char)((kcontrol->private_value >> 24) & 0xff); + + uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER; + uinfo->count = 2; + uinfo->value.integer.min = 0; + uinfo->value.integer.max = max-min; + return 0; +} +EXPORT_SYMBOL_GPL(snd_soc_info_volsw_s8); + +/** + * snd_soc_get_volsw_s8 - signed mixer get callback + * @kcontrol: mixer control + * @uinfo: control element information + * + * Callback to get the value of a signed mixer control. + * + * Returns 0 for success. + */ +int snd_soc_get_volsw_s8(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); + int reg = kcontrol->private_value & 0xff; + int min = (signed char)((kcontrol->private_value >> 24) & 0xff); + int val = snd_soc_read(codec, reg); + + ucontrol->value.integer.value[0] = + ((signed char)(val & 0xff))-min; + ucontrol->value.integer.value[1] = + ((signed char)((val >> 8) & 0xff))-min; + return 0; +} +EXPORT_SYMBOL_GPL(snd_soc_get_volsw_s8); + +/** + * snd_soc_put_volsw_sgn - signed mixer put callback + * @kcontrol: mixer control + * @uinfo: control element information + * + * Callback to set the value of a signed mixer control. + * + * Returns 0 for success. + */ +int snd_soc_put_volsw_s8(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); + int reg = kcontrol->private_value & 0xff; + int min = (signed char)((kcontrol->private_value >> 24) & 0xff); + unsigned short val; + + val = (ucontrol->value.integer.value[0]+min) & 0xff; + val |= ((ucontrol->value.integer.value[1]+min) & 0xff) << 8; + + return snd_soc_update_bits(codec, reg, 0xffff, val); +} +EXPORT_SYMBOL_GPL(snd_soc_put_volsw_s8); + +/** + * snd_soc_dai_set_sysclk - configure DAI system or master clock. + * @dai: DAI + * @clk_id: DAI specific clock ID + * @freq: new clock frequency in Hz + * @dir: new clock direction - input/output. + * + * Configures the DAI master (MCLK) or system (SYSCLK) clocking. + */ +int snd_soc_dai_set_sysclk(struct snd_soc_dai *dai, int clk_id, + unsigned int freq, int dir) +{ + if (dai->dai_ops.set_sysclk) + return dai->dai_ops.set_sysclk(dai, clk_id, freq, dir); + else + return -EINVAL; +} +EXPORT_SYMBOL_GPL(snd_soc_dai_set_sysclk); + +/** + * snd_soc_dai_set_clkdiv - configure DAI clock dividers. + * @dai: DAI + * @clk_id: DAI specific clock divider ID + * @div: new clock divisor. + * + * Configures the clock dividers. This is used to derive the best DAI bit and + * frame clocks from the system or master clock. It's best to set the DAI bit + * and frame clocks as low as possible to save system power. + */ +int snd_soc_dai_set_clkdiv(struct snd_soc_dai *dai, + int div_id, int div) +{ + if (dai->dai_ops.set_clkdiv) + return dai->dai_ops.set_clkdiv(dai, div_id, div); + else + return -EINVAL; +} +EXPORT_SYMBOL_GPL(snd_soc_dai_set_clkdiv); + +/** + * snd_soc_dai_set_pll - configure DAI PLL. + * @dai: DAI + * @pll_id: DAI specific PLL ID + * @freq_in: PLL input clock frequency in Hz + * @freq_out: requested PLL output clock frequency in Hz + * + * Configures and enables PLL to generate output clock based on input clock. + */ +int snd_soc_dai_set_pll(struct snd_soc_dai *dai, + int pll_id, unsigned int freq_in, unsigned int freq_out) +{ + if (dai->dai_ops.set_pll) + return dai->dai_ops.set_pll(dai, pll_id, freq_in, freq_out); + else + return -EINVAL; +} +EXPORT_SYMBOL_GPL(snd_soc_dai_set_pll); + +/** + * snd_soc_dai_set_fmt - configure DAI hardware audio format. + * @dai: DAI + * @clk_id: DAI specific clock ID + * @fmt: SND_SOC_DAIFMT_ format value. + * + * Configures the DAI hardware format and clocking. + */ +int snd_soc_dai_set_fmt(struct snd_soc_dai *dai, unsigned int fmt) +{ + if (dai->dai_ops.set_fmt) + return dai->dai_ops.set_fmt(dai, fmt); + else + return -EINVAL; +} +EXPORT_SYMBOL_GPL(snd_soc_dai_set_fmt); + +/** + * snd_soc_dai_set_tdm_slot - configure DAI TDM. + * @dai: DAI + * @mask: DAI specific mask representing used slots. + * @slots: Number of slots in use. + * + * Configures a DAI for TDM operation. Both mask and slots are codec and DAI + * specific. + */ +int snd_soc_dai_set_tdm_slot(struct snd_soc_dai *dai, + unsigned int mask, int slots) +{ + if (dai->dai_ops.set_sysclk) + return dai->dai_ops.set_tdm_slot(dai, mask, slots); + else + return -EINVAL; +} +EXPORT_SYMBOL_GPL(snd_soc_dai_set_tdm_slot); + +/** + * snd_soc_dai_set_tristate - configure DAI system or master clock. + * @dai: DAI + * @tristate: tristate enable + * + * Tristates the DAI so that others can use it. + */ +int snd_soc_dai_set_tristate(struct snd_soc_dai *dai, int tristate) +{ + if (dai->dai_ops.set_sysclk) + return dai->dai_ops.set_tristate(dai, tristate); + else + return -EINVAL; +} +EXPORT_SYMBOL_GPL(snd_soc_dai_set_tristate); + +/** + * snd_soc_dai_digital_mute - configure DAI system or master clock. + * @dai: DAI + * @mute: mute enable + * + * Mutes the DAI DAC. + */ +int snd_soc_dai_digital_mute(struct snd_soc_dai *dai, int mute) +{ + if (dai->dai_ops.digital_mute) + return dai->dai_ops.digital_mute(dai, mute); + else + return -EINVAL; +} +EXPORT_SYMBOL_GPL(snd_soc_dai_digital_mute); + static int __devinit snd_soc_init(void) { printk(KERN_INFO "ASoC version %s\n", SND_SOC_VERSION); @@ -1592,7 +1835,7 @@ static int __devinit snd_soc_init(void) static void snd_soc_exit(void) { - platform_driver_unregister(&soc_driver); + platform_driver_unregister(&soc_driver); } module_init(snd_soc_init); diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index af3326c6350..2c87061c2a6 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -10,11 +10,6 @@ * Free Software Foundation; either version 2 of the License, or (at your * option) any later version. * - * Revision history - * 12th Aug 2005 Initial version. - * 25th Oct 2005 Implemented path power domain. - * 18th Dec 2005 Implemented machine and stream level power domain. - * * Features: * o Changes power status of internal codec blocks depending on the * dynamic configuration of codec internal audio paths and active @@ -50,23 +45,10 @@ #include <sound/initval.h> /* debug */ -#define DAPM_DEBUG 0 -#if DAPM_DEBUG +#ifdef DEBUG #define dump_dapm(codec, action) dbg_dump_dapm(codec, action) -#define dbg(format, arg...) printk(format, ## arg) #else #define dump_dapm(codec, action) -#define dbg(format, arg...) -#endif - -#define POP_DEBUG 0 -#if POP_DEBUG -#define POP_TIME 500 /* 500 msecs - change if pop debug is too fast */ -#define pop_wait(time) schedule_timeout_uninterruptible(msecs_to_jiffies(time)) -#define pop_dbg(format, arg...) printk(format, ## arg); pop_wait(POP_TIME) -#else -#define pop_dbg(format, arg...) -#define pop_wait(time) #endif /* dapm power sequences - make this per codec in the future */ @@ -85,6 +67,28 @@ static int dapm_status = 1; module_param(dapm_status, int, 0); MODULE_PARM_DESC(dapm_status, "enable DPM sysfs entries"); +static unsigned int pop_time; + +static void pop_wait(void) +{ + if (pop_time) + schedule_timeout_uninterruptible(msecs_to_jiffies(pop_time)); +} + +static void pop_dbg(const char *fmt, ...) +{ + va_list args; + + va_start(args, fmt); + + if (pop_time) { + vprintk(fmt, args); + pop_wait(); + } + + va_end(args); +} + /* create a new dapm widget */ static inline struct snd_soc_dapm_widget *dapm_cnew_widget( const struct snd_soc_dapm_widget *_widget) @@ -222,11 +226,12 @@ static int dapm_update_bits(struct snd_soc_dapm_widget *widget) change = old != new; if (change) { pop_dbg("pop test %s : %s in %d ms\n", widget->name, - widget->power ? "on" : "off", POP_TIME); + widget->power ? "on" : "off", pop_time); snd_soc_write(codec, widget->reg, new); - pop_wait(POP_TIME); + pop_wait(); } - dbg("reg %x old %x new %x change %d\n", widget->reg, old, new, change); + pr_debug("reg %x old %x new %x change %d\n", widget->reg, + old, new, change); return change; } @@ -448,6 +453,25 @@ static int is_connected_input_ep(struct snd_soc_dapm_widget *widget) } /* + * Handler for generic register modifier widget. + */ +int dapm_reg_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + unsigned int val; + + if (SND_SOC_DAPM_EVENT_ON(event)) + val = w->on_val; + else + val = w->off_val; + + snd_soc_update_bits(w->codec, -(w->reg + 1), + w->mask << w->shift, val << w->shift); + + return 0; +} + +/* * Scan each dapm widget for complete audio path. * A complete path is a route that has valid endpoints i.e.:- * @@ -565,8 +589,8 @@ static int dapm_power_widgets(struct snd_soc_codec *codec, int event) /* call any power change event handlers */ if (power_change) { if (w->event) { - dbg("power %s event for %s flags %x\n", - w->power ? "on" : "off", w->name, w->event_flags); + pr_debug("power %s event for %s flags %x\n", + w->power ? "on" : "off", w->name, w->event_flags); if (power) { /* power up event */ if (w->event_flags & SND_SOC_DAPM_PRE_PMU) { @@ -608,7 +632,7 @@ static int dapm_power_widgets(struct snd_soc_codec *codec, int event) return ret; } -#if DAPM_DEBUG +#ifdef DEBUG static void dbg_dump_dapm(struct snd_soc_codec* codec, const char *action) { struct snd_soc_dapm_widget *w; @@ -693,8 +717,10 @@ static int dapm_mux_update_power(struct snd_soc_dapm_widget *widget, path->connect = 0; /* old connection must be powered down */ } - if (found) + if (found) { dapm_power_widgets(widget->codec, SND_SOC_DAPM_STREAM_NOP); + dump_dapm(widget->codec, "mux power update"); + } return 0; } @@ -730,8 +756,10 @@ static int dapm_mixer_update_power(struct snd_soc_dapm_widget *widget, break; } - if (found) + if (found) { dapm_power_widgets(widget->codec, SND_SOC_DAPM_STREAM_NOP); + dump_dapm(widget->codec, "mixer power update"); + } return 0; } @@ -768,21 +796,18 @@ static ssize_t dapm_widget_show(struct device *dev, } } - switch(codec->dapm_state){ - case SNDRV_CTL_POWER_D0: - state = "D0"; + switch (codec->bias_level) { + case SND_SOC_BIAS_ON: + state = "On"; break; - case SNDRV_CTL_POWER_D1: - state = "D1"; + case SND_SOC_BIAS_PREPARE: + state = "Prepare"; break; - case SNDRV_CTL_POWER_D2: - state = "D2"; + case SND_SOC_BIAS_STANDBY: + state = "Standby"; break; - case SNDRV_CTL_POWER_D3hot: - state = "D3hot"; - break; - case SNDRV_CTL_POWER_D3cold: - state = "D3cold"; + case SND_SOC_BIAS_OFF: + state = "Off"; break; } count += sprintf(buf + count, "PM State: %s\n", state); @@ -792,20 +817,51 @@ static ssize_t dapm_widget_show(struct device *dev, static DEVICE_ATTR(dapm_widget, 0444, dapm_widget_show, NULL); +/* pop/click delay times */ +static ssize_t dapm_pop_time_show(struct device *dev, + struct device_attribute *attr, char *buf) +{ + return sprintf(buf, "%d\n", pop_time); +} + +static ssize_t dapm_pop_time_store(struct device *dev, + struct device_attribute *attr, + const char *buf, size_t count) + +{ + unsigned long val; + + if (strict_strtoul(buf, 10, &val) >= 0) + pop_time = val; + else + printk(KERN_ERR "Unable to parse pop_time setting\n"); + + return count; +} + +static DEVICE_ATTR(dapm_pop_time, 0744, dapm_pop_time_show, + dapm_pop_time_store); + int snd_soc_dapm_sys_add(struct device *dev) { int ret = 0; - if (dapm_status) + if (dapm_status) { ret = device_create_file(dev, &dev_attr_dapm_widget); + if (ret == 0) + ret = device_create_file(dev, &dev_attr_dapm_pop_time); + } + return ret; } static void snd_soc_dapm_sys_remove(struct device *dev) { - if (dapm_status) + if (dapm_status) { + device_remove_file(dev, &dev_attr_dapm_pop_time); device_remove_file(dev, &dev_attr_dapm_widget); + } } /* free all dapm widgets and resources */ @@ -826,8 +882,25 @@ static void dapm_free_widgets(struct snd_soc_codec *codec) } } +static int snd_soc_dapm_set_pin(struct snd_soc_codec *codec, + char *pin, int status) +{ + struct snd_soc_dapm_widget *w; + + list_for_each_entry(w, &codec->dapm_widgets, list) { + if (!strcmp(w->name, pin)) { + pr_debug("dapm: %s: pin %s\n", codec->name, pin); + w->connected = status; + return 0; + } + } + + pr_err("dapm: %s: configuring unknown pin %s\n", codec->name, pin); + return -EINVAL; +} + /** - * snd_soc_dapm_sync_endpoints - scan and power dapm paths + * snd_soc_dapm_sync - scan and power dapm paths * @codec: audio codec * * Walks all dapm audio paths and powers widgets according to their @@ -835,27 +908,16 @@ static void dapm_free_widgets(struct snd_soc_codec *codec) * * Returns 0 for success. */ -int snd_soc_dapm_sync_endpoints(struct snd_soc_codec *codec) +int snd_soc_dapm_sync(struct snd_soc_codec *codec) { - return dapm_power_widgets(codec, SND_SOC_DAPM_STREAM_NOP); + int ret = dapm_power_widgets(codec, SND_SOC_DAPM_STREAM_NOP); + dump_dapm(codec, "sync"); + return ret; } -EXPORT_SYMBOL_GPL(snd_soc_dapm_sync_endpoints); +EXPORT_SYMBOL_GPL(snd_soc_dapm_sync); -/** - * snd_soc_dapm_connect_input - connect dapm widgets - * @codec: audio codec - * @sink: name of target widget - * @control: mixer control name - * @source: name of source name - * - * Connects 2 dapm widgets together via a named audio path. The sink is - * the widget receiving the audio signal, whilst the source is the sender - * of the audio signal. - * - * Returns 0 for success else error. - */ -int snd_soc_dapm_connect_input(struct snd_soc_codec *codec, const char *sink, - const char * control, const char *source) +static int snd_soc_dapm_add_route(struct snd_soc_codec *codec, + const char *sink, const char *control, const char *source) { struct snd_soc_dapm_path *path; struct snd_soc_dapm_widget *wsource = NULL, *wsink = NULL, *w; @@ -957,9 +1019,64 @@ err: kfree(path); return ret; } + +/** + * snd_soc_dapm_connect_input - connect dapm widgets + * @codec: audio codec + * @sink: name of target widget + * @control: mixer control name + * @source: name of source name + * + * Connects 2 dapm widgets together via a named audio path. The sink is + * the widget receiving the audio signal, whilst the source is the sender + * of the audio signal. + * + * This function has been deprecated in favour of snd_soc_dapm_add_routes(). + * + * Returns 0 for success else error. + */ +int snd_soc_dapm_connect_input(struct snd_soc_codec *codec, const char *sink, + const char *control, const char *source) +{ + return snd_soc_dapm_add_route(codec, sink, control, source); +} EXPORT_SYMBOL_GPL(snd_soc_dapm_connect_input); /** + * snd_soc_dapm_add_routes - Add routes between DAPM widgets + * @codec: codec + * @route: audio routes + * @num: number of routes + * + * Connects 2 dapm widgets together via a named audio path. The sink is + * the widget receiving the audio signal, whilst the source is the sender + * of the audio signal. + * + * Returns 0 for success else error. On error all resources can be freed + * with a call to snd_soc_card_free(). + */ +int snd_soc_dapm_add_routes(struct snd_soc_codec *codec, + const struct snd_soc_dapm_route *route, int num) +{ + int i, ret; + + for (i = 0; i < num; i++) { + ret = snd_soc_dapm_add_route(codec, route->sink, + route->control, route->source); + if (ret < 0) { + printk(KERN_ERR "Failed to add route %s->%s\n", + route->source, + route->sink); + return ret; + } + route++; + } + + return 0; +} +EXPORT_SYMBOL_GPL(snd_soc_dapm_add_routes); + +/** * snd_soc_dapm_new_widgets - add new dapm widgets * @codec: audio codec * @@ -1234,6 +1351,33 @@ int snd_soc_dapm_new_control(struct snd_soc_codec *codec, EXPORT_SYMBOL_GPL(snd_soc_dapm_new_control); /** + * snd_soc_dapm_new_controls - create new dapm controls + * @codec: audio codec + * @widget: widget array + * @num: number of widgets + * + * Creates new DAPM controls based upon the templates. + * + * Returns 0 for success else error. + */ +int snd_soc_dapm_new_controls(struct snd_soc_codec *codec, + const struct snd_soc_dapm_widget *widget, + int num) +{ + int i, ret; + + for (i = 0; i < num; i++) { + ret = snd_soc_dapm_new_control(codec, widget); + if (ret < 0) + return ret; + widget++; + } + return 0; +} +EXPORT_SYMBOL_GPL(snd_soc_dapm_new_controls); + + +/** * snd_soc_dapm_stream_event - send a stream event to the dapm core * @codec: audio codec * @stream: stream name @@ -1257,8 +1401,8 @@ int snd_soc_dapm_stream_event(struct snd_soc_codec *codec, { if (!w->sname) continue; - dbg("widget %s\n %s stream %s event %d\n", w->name, w->sname, - stream, event); + pr_debug("widget %s\n %s stream %s event %d\n", + w->name, w->sname, stream, event); if (strstr(w->sname, stream)) { switch(event) { case SND_SOC_DAPM_STREAM_START: @@ -1294,53 +1438,81 @@ int snd_soc_dapm_stream_event(struct snd_soc_codec *codec, EXPORT_SYMBOL_GPL(snd_soc_dapm_stream_event); /** - * snd_soc_dapm_device_event - send a device event to the dapm core + * snd_soc_dapm_set_bias_level - set the bias level for the system * @socdev: audio device - * @event: device event + * @level: level to configure * - * Sends a device event to the dapm core. The core then makes any - * necessary machine or codec power changes.. + * Configure the bias (power) levels for the SoC audio device. * * Returns 0 for success else error. */ -int snd_soc_dapm_device_event(struct snd_soc_device *socdev, int event) +int snd_soc_dapm_set_bias_level(struct snd_soc_device *socdev, + enum snd_soc_bias_level level) { struct snd_soc_codec *codec = socdev->codec; struct snd_soc_machine *machine = socdev->machine; + int ret = 0; - if (machine->dapm_event) - machine->dapm_event(machine, event); - if (codec->dapm_event) - codec->dapm_event(codec, event); - return 0; + if (machine->set_bias_level) + ret = machine->set_bias_level(machine, level); + if (ret == 0 && codec->set_bias_level) + ret = codec->set_bias_level(codec, level); + + return ret; } -EXPORT_SYMBOL_GPL(snd_soc_dapm_device_event); /** - * snd_soc_dapm_set_endpoint - set audio endpoint status + * snd_soc_dapm_enable_pin - enable pin. + * @snd_soc_codec: SoC codec + * @pin: pin name + * + * Enables input/output pin and it's parents or children widgets iff there is + * a valid audio route and active audio stream. + * NOTE: snd_soc_dapm_sync() needs to be called after this for DAPM to + * do any widget power switching. + */ +int snd_soc_dapm_enable_pin(struct snd_soc_codec *codec, char *pin) +{ + return snd_soc_dapm_set_pin(codec, pin, 1); +} +EXPORT_SYMBOL_GPL(snd_soc_dapm_enable_pin); + +/** + * snd_soc_dapm_disable_pin - disable pin. + * @codec: SoC codec + * @pin: pin name + * + * Disables input/output pin and it's parents or children widgets. + * NOTE: snd_soc_dapm_sync() needs to be called after this for DAPM to + * do any widget power switching. + */ +int snd_soc_dapm_disable_pin(struct snd_soc_codec *codec, char *pin) +{ + return snd_soc_dapm_set_pin(codec, pin, 0); +} +EXPORT_SYMBOL_GPL(snd_soc_dapm_disable_pin); + +/** + * snd_soc_dapm_get_pin_status - get audio pin status * @codec: audio codec - * @endpoint: audio signal endpoint (or start point) - * @status: point status + * @pin: audio signal pin endpoint (or start point) * - * Set audio endpoint status - connected or disconnected. + * Get audio pin status - connected or disconnected. * - * Returns 0 for success else error. + * Returns 1 for connected otherwise 0. */ -int snd_soc_dapm_set_endpoint(struct snd_soc_codec *codec, - char *endpoint, int status) +int snd_soc_dapm_get_pin_status(struct snd_soc_codec *codec, char *pin) { struct snd_soc_dapm_widget *w; list_for_each_entry(w, &codec->dapm_widgets, list) { - if (!strcmp(w->name, endpoint)) { - w->connected = status; - return 0; - } + if (!strcmp(w->name, pin)) + return w->connected; } - return -ENODEV; + return 0; } -EXPORT_SYMBOL_GPL(snd_soc_dapm_set_endpoint); +EXPORT_SYMBOL_GPL(snd_soc_dapm_get_pin_status); /** * snd_soc_dapm_free - free dapm resources diff --git a/sound/sparc/Kconfig b/sound/sparc/Kconfig index 079e22af074..d75deba5617 100644 --- a/sound/sparc/Kconfig +++ b/sound/sparc/Kconfig @@ -1,11 +1,17 @@ # ALSA Sparc drivers -menu "ALSA Sparc devices" - depends on SND!=n && SPARC +menuconfig SND_SPARC + bool "Sparc sound devices" + depends on SPARC + default y + help + Support for sound devices specific to Sun SPARC architectures. + +if SND_SPARC config SND_SUN_AMD7930 tristate "Sun AMD7930" - depends on SBUS && SND + depends on SBUS select SND_PCM help Say Y here to include support for AMD7930 sound device on Sun. @@ -15,7 +21,6 @@ config SND_SUN_AMD7930 config SND_SUN_CS4231 tristate "Sun CS4231" - depends on SND select SND_PCM help Say Y here to include support for CS4231 sound device on Sun. @@ -25,7 +30,7 @@ config SND_SUN_CS4231 config SND_SUN_DBRI tristate "Sun DBRI" - depends on SND && SBUS + depends on SBUS select SND_PCM help Say Y here to include support for DBRI sound device on Sun. @@ -33,4 +38,4 @@ config SND_SUN_DBRI To compile this driver as a module, choose M here: the module will be called snd-sun-dbri. -endmenu +endif # SND_SPARC diff --git a/sound/sparc/dbri.c b/sound/sparc/dbri.c index 3d00e0797b1..ee2e1b4f355 100644 --- a/sound/sparc/dbri.c +++ b/sound/sparc/dbri.c @@ -2490,7 +2490,7 @@ static void dbri_debug_read(struct snd_info_entry *entry, } #endif -void __devinit snd_dbri_proc(struct snd_card *card) +static void __devinit snd_dbri_proc(struct snd_card *card) { struct snd_dbri *dbri = card->private_data; struct snd_info_entry *entry; diff --git a/sound/spi/Kconfig b/sound/spi/Kconfig index 0d08c29213c..e6485be2e6f 100644 --- a/sound/spi/Kconfig +++ b/sound/spi/Kconfig @@ -1,7 +1,13 @@ #SPI drivers -menu "SPI devices" - depends on SND != n +menuconfig SND_SPI + bool "SPI sound devices" + depends on SPI + default y + help + Support for sound devices connected via the SPI bus. + +if SND_SPI config SND_AT73C213 tristate "Atmel AT73C213 DAC driver" @@ -28,4 +34,5 @@ config SND_AT73C213_TARGET_BITRATE Set to 48000 Hz by default. -endmenu +endif # SND_SPI + diff --git a/sound/usb/Kconfig b/sound/usb/Kconfig index 9351b8a765b..ffcdc8f4ef6 100644 --- a/sound/usb/Kconfig +++ b/sound/usb/Kconfig @@ -1,11 +1,16 @@ # ALSA USB drivers -menu "USB devices" - depends on SND!=n && USB!=n +menuconfig SND_USB + bool "USB sound devices" + depends on USB + default y + help + Support for sound devices connected via the USB bus. + +if SND_USB && USB config SND_USB_AUDIO tristate "USB Audio/MIDI driver" - depends on SND && USB select SND_HWDEP select SND_RAWMIDI select SND_PCM @@ -18,7 +23,7 @@ config SND_USB_AUDIO config SND_USB_USX2Y tristate "Tascam US-122, US-224 and US-428 USB driver" - depends on SND && USB && (X86 || PPC || ALPHA) + depends on X86 || PPC || ALPHA select SND_HWDEP select SND_RAWMIDI select SND_PCM @@ -31,7 +36,6 @@ config SND_USB_USX2Y config SND_USB_CAIAQ tristate "Native Instruments USB audio devices" - depends on SND && USB select SND_HWDEP select SND_RAWMIDI select SND_PCM @@ -63,5 +67,5 @@ config SND_USB_CAIAQ_INPUT * Native Instruments Kore Controller 2 * Native Instruments Audio Kontrol 1 -endmenu +endif # SND_USB diff --git a/sound/usb/caiaq/caiaq-audio.c b/sound/usb/caiaq/caiaq-audio.c index 24970a5c888..b3a60332583 100644 --- a/sound/usb/caiaq/caiaq-audio.c +++ b/sound/usb/caiaq/caiaq-audio.c @@ -637,6 +637,7 @@ int snd_usb_caiaq_audio_init(struct snd_usb_caiaqdev *dev) switch (dev->chip.usb_id) { case USB_ID(USB_VID_NATIVEINSTRUMENTS, USB_PID_AK1): case USB_ID(USB_VID_NATIVEINSTRUMENTS, USB_PID_RIGKONTROL3): + case USB_ID(USB_VID_NATIVEINSTRUMENTS, USB_PID_SESSIONIO): dev->samplerates |= SNDRV_PCM_RATE_88200; dev->samplerates |= SNDRV_PCM_RATE_192000; break; diff --git a/sound/usb/caiaq/caiaq-device.c b/sound/usb/caiaq/caiaq-device.c index a972f77bd78..83175083e50 100644 --- a/sound/usb/caiaq/caiaq-device.c +++ b/sound/usb/caiaq/caiaq-device.c @@ -42,14 +42,15 @@ #endif MODULE_AUTHOR("Daniel Mack <daniel@caiaq.de>"); -MODULE_DESCRIPTION("caiaq USB audio, version 1.3.6"); +MODULE_DESCRIPTION("caiaq USB audio, version 1.3.8"); MODULE_LICENSE("GPL"); MODULE_SUPPORTED_DEVICE("{{Native Instruments, RigKontrol2}," "{Native Instruments, RigKontrol3}," "{Native Instruments, Kore Controller}," "{Native Instruments, Kore Controller 2}," - "{Native Instruments, Audio Kontrol 1}" - "{Native Instruments, Audio 8 DJ}}"); + "{Native Instruments, Audio Kontrol 1}," + "{Native Instruments, Audio 8 DJ}," + "{Native Instruments, Session I/O}}"); static int index[SNDRV_CARDS] = SNDRV_DEFAULT_IDX; /* Index 0-max */ static char* id[SNDRV_CARDS] = SNDRV_DEFAULT_STR; /* Id for this card */ @@ -110,6 +111,11 @@ static struct usb_device_id snd_usb_id_table[] = { .idVendor = USB_VID_NATIVEINSTRUMENTS, .idProduct = USB_PID_AUDIO8DJ }, + { + .match_flags = USB_DEVICE_ID_MATCH_DEVICE, + .idVendor = USB_VID_NATIVEINSTRUMENTS, + .idProduct = USB_PID_SESSIONIO + }, { /* terminator */ } }; diff --git a/sound/usb/caiaq/caiaq-device.h b/sound/usb/caiaq/caiaq-device.h index 96a491379c6..f9fbdbae269 100644 --- a/sound/usb/caiaq/caiaq-device.h +++ b/sound/usb/caiaq/caiaq-device.h @@ -11,6 +11,7 @@ #define USB_PID_KORECONTROLLER2 0x4712 #define USB_PID_AK1 0x0815 #define USB_PID_AUDIO8DJ 0x1978 +#define USB_PID_SESSIONIO 0x1915 #define EP1_BUFSIZE 64 #define CAIAQ_USB_STR_LEN 0xff diff --git a/sound/usb/usbaudio.c b/sound/usb/usbaudio.c index 410be4aff1b..b8cfb7c2276 100644 --- a/sound/usb/usbaudio.c +++ b/sound/usb/usbaudio.c @@ -819,10 +819,6 @@ static const char *usb_error_string(int err) return "device disabled"; case -EHOSTUNREACH: return "device suspended"; -#ifndef CONFIG_USB_EHCI_SPLIT_ISO - case -ENOSYS: - return "enable CONFIG_USB_EHCI_SPLIT_ISO to play through a hub"; -#endif case -EINVAL: case -EAGAIN: case -EFBIG: diff --git a/sound/usb/usbquirks.h b/sound/usb/usbquirks.h index 82a8d14c26a..9ea726c049c 100644 --- a/sound/usb/usbquirks.h +++ b/sound/usb/usbquirks.h @@ -210,6 +210,11 @@ YAMAHA_DEVICE(0x1042, NULL), YAMAHA_DEVICE(0x1043, NULL), YAMAHA_DEVICE(0x1044, NULL), YAMAHA_DEVICE(0x1045, NULL), +YAMAHA_INTERFACE(0x104e, 0, NULL), +YAMAHA_DEVICE(0x104f, NULL), +YAMAHA_DEVICE(0x1050, NULL), +YAMAHA_DEVICE(0x1051, NULL), +YAMAHA_DEVICE(0x1052, NULL), YAMAHA_DEVICE(0x2000, "DGP-7"), YAMAHA_DEVICE(0x2001, "DGP-5"), YAMAHA_DEVICE(0x2002, NULL), @@ -1379,6 +1384,39 @@ YAMAHA_DEVICE(0x7010, "UB99"), } }, +{ + /* Roland SonicCell */ + USB_DEVICE(0x0582, 0x00c2), + .driver_info = (unsigned long) & (const struct snd_usb_audio_quirk) { + .vendor_name = "Roland", + .product_name = "SonicCell", + .ifnum = QUIRK_ANY_INTERFACE, + .type = QUIRK_COMPOSITE, + .data = (const struct snd_usb_audio_quirk[]) { + { + .ifnum = 0, + .type = QUIRK_AUDIO_STANDARD_INTERFACE + }, + { + .ifnum = 1, + .type = QUIRK_AUDIO_STANDARD_INTERFACE + }, + { + .ifnum = 2, + .type = QUIRK_MIDI_FIXED_ENDPOINT, + .data = & (const struct snd_usb_midi_endpoint_info) { + .out_cables = 0x0001, + .in_cables = 0x0001 + } + }, + { + .ifnum = -1 + } + } + } +}, + + /* Guillemot devices */ { /* |