diff options
author | James Bottomley <James.Bottomley@HansenPartnership.com> | 2009-06-12 10:02:03 -0500 |
---|---|---|
committer | James Bottomley <James.Bottomley@HansenPartnership.com> | 2009-06-12 10:02:03 -0500 |
commit | 82681a318f9f028ea64e61f24bbd9ac535531921 (patch) | |
tree | 529b6a5b4fd040fb54b7672b1a224ebd47445876 /sound | |
parent | 3860c97bd60a4525bb62eb90e3e7d2f02662ac59 (diff) | |
parent | 8ebf975608aaebd7feb33d77f07ba21a6380e086 (diff) |
[SCSI] Merge branch 'linus'
Conflicts:
drivers/message/fusion/mptsas.c
fixed up conflict between req->data_len accessors and mptsas driver updates.
Signed-off-by: James Bottomley <James.Bottomley@HansenPartnership.com>
Diffstat (limited to 'sound')
-rw-r--r-- | sound/arm/aaci.c | 2 | ||||
-rw-r--r-- | sound/core/pcm_lib.c | 10 | ||||
-rw-r--r-- | sound/core/pcm_native.c | 6 | ||||
-rw-r--r-- | sound/drivers/pcsp/pcsp_mixer.c | 2 | ||||
-rw-r--r-- | sound/oss/Kconfig | 2 | ||||
-rw-r--r-- | sound/oss/sh_dac_audio.c | 85 | ||||
-rw-r--r-- | sound/pci/ac97/ac97_patch.c | 7 | ||||
-rw-r--r-- | sound/pci/ca0106/ca0106_mixer.c | 2 | ||||
-rw-r--r-- | sound/pci/hda/hda_intel.c | 1 | ||||
-rw-r--r-- | sound/pci/hda/patch_conexant.c | 1 | ||||
-rw-r--r-- | sound/pci/hda/patch_realtek.c | 7 | ||||
-rw-r--r-- | sound/pci/hda/patch_sigmatel.c | 10 | ||||
-rw-r--r-- | sound/usb/usbaudio.c | 2 | ||||
-rw-r--r-- | sound/usb/usbaudio.h | 2 | ||||
-rw-r--r-- | sound/usb/usbmidi.c | 12 | ||||
-rw-r--r-- | sound/usb/usbquirks.h | 2 |
16 files changed, 88 insertions, 65 deletions
diff --git a/sound/arm/aaci.c b/sound/arm/aaci.c index 7fbd68fab94..5c48e36038f 100644 --- a/sound/arm/aaci.c +++ b/sound/arm/aaci.c @@ -1074,7 +1074,7 @@ static unsigned int __devinit aaci_size_fifo(struct aaci *aaci) return i; } -static int __devinit aaci_probe(struct amba_device *dev, void *id) +static int __devinit aaci_probe(struct amba_device *dev, struct amba_id *id) { struct aaci *aaci; int ret, i; diff --git a/sound/core/pcm_lib.c b/sound/core/pcm_lib.c index a2a792c18c4..d659995ac3a 100644 --- a/sound/core/pcm_lib.c +++ b/sound/core/pcm_lib.c @@ -249,6 +249,11 @@ static int snd_pcm_update_hw_ptr_interrupt(struct snd_pcm_substream *substream) new_hw_ptr = hw_base + pos; } } + + /* Do jiffies check only in xrun_debug mode */ + if (!xrun_debug(substream)) + goto no_jiffies_check; + /* Skip the jiffies check for hardwares with BATCH flag. * Such hardware usually just increases the position at each IRQ, * thus it can't give any strange position. @@ -336,7 +341,9 @@ int snd_pcm_update_hw_ptr(struct snd_pcm_substream *substream) hw_base = 0; new_hw_ptr = hw_base + pos; } - if (((delta * HZ) / runtime->rate) > jdelta + HZ/100) { + /* Do jiffies check only in xrun_debug mode */ + if (xrun_debug(substream) && + ((delta * HZ) / runtime->rate) > jdelta + HZ/100) { hw_ptr_error(substream, "hw_ptr skipping! " "(pos=%ld, delta=%ld, period=%ld, jdelta=%lu/%lu)\n", @@ -1478,7 +1485,6 @@ static int snd_pcm_lib_ioctl_reset(struct snd_pcm_substream *substream, runtime->status->hw_ptr %= runtime->buffer_size; else runtime->status->hw_ptr = 0; - runtime->hw_ptr_jiffies = jiffies; snd_pcm_stream_unlock_irqrestore(substream, flags); return 0; } diff --git a/sound/core/pcm_native.c b/sound/core/pcm_native.c index fc6f98e257d..b5da656d1ec 100644 --- a/sound/core/pcm_native.c +++ b/sound/core/pcm_native.c @@ -848,6 +848,7 @@ static void snd_pcm_post_start(struct snd_pcm_substream *substream, int state) { struct snd_pcm_runtime *runtime = substream->runtime; snd_pcm_trigger_tstamp(substream); + runtime->hw_ptr_jiffies = jiffies; runtime->status->state = state; if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK && runtime->silence_size > 0) @@ -961,6 +962,11 @@ static int snd_pcm_do_pause(struct snd_pcm_substream *substream, int push) { if (substream->runtime->trigger_master != substream) return 0; + /* The jiffies check in snd_pcm_update_hw_ptr*() is done by + * a delta betwen the current jiffies, this gives a large enough + * delta, effectively to skip the check once. + */ + substream->runtime->hw_ptr_jiffies = jiffies - HZ * 1000; return substream->ops->trigger(substream, push ? SNDRV_PCM_TRIGGER_PAUSE_PUSH : SNDRV_PCM_TRIGGER_PAUSE_RELEASE); diff --git a/sound/drivers/pcsp/pcsp_mixer.c b/sound/drivers/pcsp/pcsp_mixer.c index 771955a9be7..199b0337714 100644 --- a/sound/drivers/pcsp/pcsp_mixer.c +++ b/sound/drivers/pcsp/pcsp_mixer.c @@ -51,7 +51,7 @@ static int pcsp_treble_info(struct snd_kcontrol *kcontrol, if (uinfo->value.enumerated.item > chip->max_treble) uinfo->value.enumerated.item = chip->max_treble; sprintf(uinfo->value.enumerated.name, "%lu", - PCSP_CALC_RATE(uinfo->value.enumerated.item)); + (unsigned long)PCSP_CALC_RATE(uinfo->value.enumerated.item)); return 0; } diff --git a/sound/oss/Kconfig b/sound/oss/Kconfig index 1ca7427c4b6..bcf2a0698d5 100644 --- a/sound/oss/Kconfig +++ b/sound/oss/Kconfig @@ -561,7 +561,7 @@ endif # SOUND_OSS config SOUND_SH_DAC_AUDIO tristate "SuperH DAC audio support" - depends on CPU_SH3 + depends on CPU_SH3 && HIGH_RES_TIMERS config SOUND_SH_DAC_AUDIO_CHANNEL int "DAC channel" diff --git a/sound/oss/sh_dac_audio.c b/sound/oss/sh_dac_audio.c index 78cfb66e4c5..b2ed8757542 100644 --- a/sound/oss/sh_dac_audio.c +++ b/sound/oss/sh_dac_audio.c @@ -18,47 +18,36 @@ #include <linux/sound.h> #include <linux/soundcard.h> #include <linux/interrupt.h> +#include <linux/hrtimer.h> #include <asm/io.h> #include <asm/uaccess.h> #include <asm/irq.h> #include <asm/delay.h> #include <asm/clock.h> -#include <asm/cpu/dac.h> -#include <asm/cpu/timer.h> +#include <cpu/dac.h> #include <asm/machvec.h> #include <mach/hp6xx.h> #include <asm/hd64461.h> #define MODNAME "sh_dac_audio" -#define TMU_TOCR_INIT 0x00 - -#define TMU1_TCR_INIT 0x0020 /* Clock/4, rising edge; interrupt on */ -#define TMU1_TSTR_INIT 0x02 /* Bit to turn on TMU1 */ - #define BUFFER_SIZE 48000 static int rate; static int empty; static char *data_buffer, *buffer_begin, *buffer_end; static int in_use, device_major; +static struct hrtimer hrtimer; +static ktime_t wakeups_per_second; static void dac_audio_start_timer(void) { - u8 tstr; - - tstr = ctrl_inb(TMU_TSTR); - tstr |= TMU1_TSTR_INIT; - ctrl_outb(tstr, TMU_TSTR); + hrtimer_start(&hrtimer, wakeups_per_second, HRTIMER_MODE_REL); } static void dac_audio_stop_timer(void) { - u8 tstr; - - tstr = ctrl_inb(TMU_TSTR); - tstr &= ~TMU1_TSTR_INIT; - ctrl_outb(tstr, TMU_TSTR); + hrtimer_cancel(&hrtimer); } static void dac_audio_reset(void) @@ -77,38 +66,30 @@ static void dac_audio_sync(void) static void dac_audio_start(void) { if (mach_is_hp6xx()) { - u16 v = inw(HD64461_GPADR); + u16 v = __raw_readw(HD64461_GPADR); v &= ~HD64461_GPADR_SPEAKER; - outw(v, HD64461_GPADR); + __raw_writew(v, HD64461_GPADR); } sh_dac_enable(CONFIG_SOUND_SH_DAC_AUDIO_CHANNEL); - ctrl_outw(TMU1_TCR_INIT, TMU1_TCR); } static void dac_audio_stop(void) { dac_audio_stop_timer(); if (mach_is_hp6xx()) { - u16 v = inw(HD64461_GPADR); + u16 v = __raw_readw(HD64461_GPADR); v |= HD64461_GPADR_SPEAKER; - outw(v, HD64461_GPADR); + __raw_writew(v, HD64461_GPADR); } - sh_dac_output(0, CONFIG_SOUND_SH_DAC_AUDIO_CHANNEL); + sh_dac_output(0, CONFIG_SOUND_SH_DAC_AUDIO_CHANNEL); sh_dac_disable(CONFIG_SOUND_SH_DAC_AUDIO_CHANNEL); } static void dac_audio_set_rate(void) { - unsigned long interval; - struct clk *clk; - - clk = clk_get(NULL, "module_clk"); - interval = (clk_get_rate(clk) / 4) / rate; - clk_put(clk); - ctrl_outl(interval, TMU1_TCOR); - ctrl_outl(interval, TMU1_TCNT); + wakeups_per_second = ktime_set(0, 1000000000 / rate); } static int dac_audio_ioctl(struct inode *inode, struct file *file, @@ -265,32 +246,26 @@ const struct file_operations dac_audio_fops = { .release = dac_audio_release, }; -static irqreturn_t timer1_interrupt(int irq, void *dev) +static enum hrtimer_restart sh_dac_audio_timer(struct hrtimer *handle) { - unsigned long timer_status; - - timer_status = ctrl_inw(TMU1_TCR); - timer_status &= ~0x100; - ctrl_outw(timer_status, TMU1_TCR); - if (!empty) { sh_dac_output(*buffer_begin, CONFIG_SOUND_SH_DAC_AUDIO_CHANNEL); buffer_begin++; if (buffer_begin == data_buffer + BUFFER_SIZE) buffer_begin = data_buffer; - if (buffer_begin == buffer_end) { + if (buffer_begin == buffer_end) empty = 1; - dac_audio_stop_timer(); - } } - return IRQ_HANDLED; + + if (!empty) + hrtimer_start(&hrtimer, wakeups_per_second, HRTIMER_MODE_REL); + + return HRTIMER_NORESTART; } static int __init dac_audio_init(void) { - int retval; - if ((device_major = register_sound_dsp(&dac_audio_fops, -1)) < 0) { printk(KERN_ERR "Cannot register dsp device"); return device_major; @@ -306,21 +281,25 @@ static int __init dac_audio_init(void) rate = 8000; dac_audio_set_rate(); - retval = - request_irq(TIMER1_IRQ, timer1_interrupt, IRQF_DISABLED, MODNAME, 0); - if (retval < 0) { - printk(KERN_ERR "sh_dac_audio: IRQ %d request failed\n", - TIMER1_IRQ); - return retval; - } + /* Today: High Resolution Timer driven DAC playback. + * The timer callback gets called once per sample. Ouch. + * + * Future: A much better approach would be to use the + * SH7720 CMT+DMAC+DAC hardware combination like this: + * - Program sample rate using CMT0 or CMT1 + * - Program DMAC to use CMT for timing and output to DAC + * - Play sound using DMAC, let CPU sleep. + * - While at it, rewrite this driver to use ALSA. + */ + + hrtimer_init(&hrtimer, CLOCK_MONOTONIC, HRTIMER_MODE_REL); + hrtimer.function = sh_dac_audio_timer; return 0; } static void __exit dac_audio_exit(void) { - free_irq(TIMER1_IRQ, 0); - unregister_sound_dsp(device_major); kfree((void *)data_buffer); } diff --git a/sound/pci/ac97/ac97_patch.c b/sound/pci/ac97/ac97_patch.c index 81bc93e5f1e..7337abdbe4e 100644 --- a/sound/pci/ac97/ac97_patch.c +++ b/sound/pci/ac97/ac97_patch.c @@ -958,10 +958,13 @@ static int patch_sigmatel_stac9708_3d(struct snd_ac97 * ac97) } static const struct snd_kcontrol_new snd_ac97_sigmatel_4speaker = -AC97_SINGLE("Sigmatel 4-Speaker Stereo Playback Switch", AC97_SIGMATEL_DAC2INVERT, 2, 1, 0); +AC97_SINGLE("Sigmatel 4-Speaker Stereo Playback Switch", + AC97_SIGMATEL_DAC2INVERT, 2, 1, 0); +/* "Sigmatel " removed due to excessive name length: */ static const struct snd_kcontrol_new snd_ac97_sigmatel_phaseinvert = -AC97_SINGLE("Sigmatel Surround Phase Inversion Playback Switch", AC97_SIGMATEL_DAC2INVERT, 3, 1, 0); +AC97_SINGLE("Surround Phase Inversion Playback Switch", + AC97_SIGMATEL_DAC2INVERT, 3, 1, 0); static const struct snd_kcontrol_new snd_ac97_sigmatel_controls[] = { AC97_SINGLE("Sigmatel DAC 6dB Attenuate", AC97_SIGMATEL_ANALOG, 1, 1, 0), diff --git a/sound/pci/ca0106/ca0106_mixer.c b/sound/pci/ca0106/ca0106_mixer.c index ad2888705d2..c111efe61c3 100644 --- a/sound/pci/ca0106/ca0106_mixer.c +++ b/sound/pci/ca0106/ca0106_mixer.c @@ -800,7 +800,7 @@ int __devinit snd_ca0106_mixer(struct snd_ca0106 *emu) "Capture Volume", "External Amplifier", "Sigmatel 4-Speaker Stereo Playback Switch", - "Sigmatel Surround Phase Inversion Playback ", + "Surround Phase Inversion Playback Switch", NULL }; static char *ca0106_rename_ctls[] = { diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index 21e99cfa8c4..3128e1a6bc6 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -2141,6 +2141,7 @@ static struct snd_pci_quirk probe_mask_list[] __devinitdata = { /* including bogus ALC268 in slot#2 that conflicts with ALC888 */ SND_PCI_QUIRK(0x17c0, 0x4085, "Medion MD96630", 0x01), /* forced codec slots */ + SND_PCI_QUIRK(0x1043, 0x1262, "ASUS W5Fm", 0x103), SND_PCI_QUIRK(0x1046, 0x1262, "ASUS W5F", 0x103), {} }; diff --git a/sound/pci/hda/patch_conexant.c b/sound/pci/hda/patch_conexant.c index 56ce19e68cb..4fcbe21829a 100644 --- a/sound/pci/hda/patch_conexant.c +++ b/sound/pci/hda/patch_conexant.c @@ -1848,6 +1848,7 @@ static const char *cxt5051_models[CXT5051_MODELS] = { static struct snd_pci_quirk cxt5051_cfg_tbl[] = { SND_PCI_QUIRK(0x103c, 0x30cf, "HP DV6736", CXT5051_HP_DV6736), + SND_PCI_QUIRK(0x103c, 0x360b, "Compaq Presario CQ60", CXT5051_HP), SND_PCI_QUIRK(0x14f1, 0x0101, "Conexant Reference board", CXT5051_LAPTOP), SND_PCI_QUIRK(0x14f1, 0x5051, "HP Spartan 1.1", CXT5051_HP), diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index b8a0d3e7927..0fd258eba3a 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -776,6 +776,12 @@ static void alc_set_input_pin(struct hda_codec *codec, hda_nid_t nid, pincap = (pincap & AC_PINCAP_VREF) >> AC_PINCAP_VREF_SHIFT; if (pincap & AC_PINCAP_VREF_80) val = PIN_VREF80; + else if (pincap & AC_PINCAP_VREF_50) + val = PIN_VREF50; + else if (pincap & AC_PINCAP_VREF_100) + val = PIN_VREF100; + else if (pincap & AC_PINCAP_VREF_GRD) + val = PIN_VREFGRD; } snd_hda_codec_write(codec, nid, 0, AC_VERB_SET_PIN_WIDGET_CONTROL, val); } @@ -12058,6 +12064,7 @@ static struct snd_pci_quirk alc268_cfg_tbl[] = { SND_PCI_QUIRK(0x1028, 0x0253, "Dell OEM", ALC268_DELL), SND_PCI_QUIRK(0x1028, 0x02b0, "Dell Inspiron Mini9", ALC268_DELL), SND_PCI_QUIRK(0x103c, 0x30cc, "TOSHIBA", ALC268_TOSHIBA), + SND_PCI_QUIRK(0x103c, 0x30f1, "HP TX25xx series", ALC268_TOSHIBA), SND_PCI_QUIRK(0x1043, 0x1205, "ASUS W7J", ALC268_3ST), SND_PCI_QUIRK(0x1179, 0xff10, "TOSHIBA A205", ALC268_TOSHIBA), SND_PCI_QUIRK(0x1179, 0xff50, "TOSHIBA A305", ALC268_TOSHIBA), diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c index 03b3646018a..d2fd8ef6aef 100644 --- a/sound/pci/hda/patch_sigmatel.c +++ b/sound/pci/hda/patch_sigmatel.c @@ -150,6 +150,7 @@ enum { STAC_D965_REF, STAC_D965_3ST, STAC_D965_5ST, + STAC_D965_5ST_NO_FP, STAC_DELL_3ST, STAC_DELL_BIOS, STAC_927X_MODELS @@ -2154,6 +2155,13 @@ static unsigned int d965_5st_pin_configs[14] = { 0x40000100, 0x40000100 }; +static unsigned int d965_5st_no_fp_pin_configs[14] = { + 0x40000100, 0x40000100, 0x0181304e, 0x01014010, + 0x01a19040, 0x01011012, 0x01016011, 0x40000100, + 0x40000100, 0x40000100, 0x40000100, 0x01442070, + 0x40000100, 0x40000100 +}; + static unsigned int dell_3st_pin_configs[14] = { 0x02211230, 0x02a11220, 0x01a19040, 0x01114210, 0x01111212, 0x01116211, 0x01813050, 0x01112214, @@ -2166,6 +2174,7 @@ static unsigned int *stac927x_brd_tbl[STAC_927X_MODELS] = { [STAC_D965_REF] = ref927x_pin_configs, [STAC_D965_3ST] = d965_3st_pin_configs, [STAC_D965_5ST] = d965_5st_pin_configs, + [STAC_D965_5ST_NO_FP] = d965_5st_no_fp_pin_configs, [STAC_DELL_3ST] = dell_3st_pin_configs, [STAC_DELL_BIOS] = NULL, }; @@ -2176,6 +2185,7 @@ static const char *stac927x_models[STAC_927X_MODELS] = { [STAC_D965_REF] = "ref", [STAC_D965_3ST] = "3stack", [STAC_D965_5ST] = "5stack", + [STAC_D965_5ST_NO_FP] = "5stack-no-fp", [STAC_DELL_3ST] = "dell-3stack", [STAC_DELL_BIOS] = "dell-bios", }; diff --git a/sound/usb/usbaudio.c b/sound/usb/usbaudio.c index 823296d7d57..a6b88482637 100644 --- a/sound/usb/usbaudio.c +++ b/sound/usb/usbaudio.c @@ -3347,7 +3347,7 @@ static int snd_usb_create_quirk(struct snd_usb_audio *chip, [QUIRK_MIDI_YAMAHA] = snd_usb_create_midi_interface, [QUIRK_MIDI_MIDIMAN] = snd_usb_create_midi_interface, [QUIRK_MIDI_NOVATION] = snd_usb_create_midi_interface, - [QUIRK_MIDI_RAW] = snd_usb_create_midi_interface, + [QUIRK_MIDI_FASTLANE] = snd_usb_create_midi_interface, [QUIRK_MIDI_EMAGIC] = snd_usb_create_midi_interface, [QUIRK_MIDI_CME] = snd_usb_create_midi_interface, [QUIRK_AUDIO_STANDARD_INTERFACE] = create_standard_audio_quirk, diff --git a/sound/usb/usbaudio.h b/sound/usb/usbaudio.h index 36e4f7a29ad..8e7f78941ba 100644 --- a/sound/usb/usbaudio.h +++ b/sound/usb/usbaudio.h @@ -153,7 +153,7 @@ enum quirk_type { QUIRK_MIDI_YAMAHA, QUIRK_MIDI_MIDIMAN, QUIRK_MIDI_NOVATION, - QUIRK_MIDI_RAW, + QUIRK_MIDI_FASTLANE, QUIRK_MIDI_EMAGIC, QUIRK_MIDI_CME, QUIRK_MIDI_US122L, diff --git a/sound/usb/usbmidi.c b/sound/usb/usbmidi.c index 26bad373fe6..2fb35cc22a3 100644 --- a/sound/usb/usbmidi.c +++ b/sound/usb/usbmidi.c @@ -1778,8 +1778,18 @@ int snd_usb_create_midi_interface(struct snd_usb_audio* chip, umidi->usb_protocol_ops = &snd_usbmidi_novation_ops; err = snd_usbmidi_detect_per_port_endpoints(umidi, endpoints); break; - case QUIRK_MIDI_RAW: + case QUIRK_MIDI_FASTLANE: umidi->usb_protocol_ops = &snd_usbmidi_raw_ops; + /* + * Interface 1 contains isochronous endpoints, but with the same + * numbers as in interface 0. Since it is interface 1 that the + * USB core has most recently seen, these descriptors are now + * associated with the endpoint numbers. This will foul up our + * attempts to submit bulk/interrupt URBs to the endpoints in + * interface 0, so we have to make sure that the USB core looks + * again at interface 0 by calling usb_set_interface() on it. + */ + usb_set_interface(umidi->chip->dev, 0, 0); err = snd_usbmidi_detect_per_port_endpoints(umidi, endpoints); break; case QUIRK_MIDI_EMAGIC: diff --git a/sound/usb/usbquirks.h b/sound/usb/usbquirks.h index 647ef502965..5d955aaad85 100644 --- a/sound/usb/usbquirks.h +++ b/sound/usb/usbquirks.h @@ -1868,7 +1868,7 @@ YAMAHA_DEVICE(0x7010, "UB99"), .data = & (const struct snd_usb_audio_quirk[]) { { .ifnum = 0, - .type = QUIRK_MIDI_RAW + .type = QUIRK_MIDI_FASTLANE }, { .ifnum = 1, |