diff options
Diffstat (limited to 'include/sound')
-rw-r--r-- | include/sound/ad1843.h | 46 | ||||
-rw-r--r-- | include/sound/control.h | 3 | ||||
-rw-r--r-- | include/sound/core.h | 8 | ||||
-rw-r--r-- | include/sound/cs4231-regs.h | 8 | ||||
-rw-r--r-- | include/sound/cs4231.h | 3 | ||||
-rw-r--r-- | include/sound/emu10k1.h | 1 | ||||
-rw-r--r-- | include/sound/seq_kernel.h | 2 | ||||
-rw-r--r-- | include/sound/soc-dapm.h | 42 | ||||
-rw-r--r-- | include/sound/soc.h | 175 | ||||
-rw-r--r-- | include/sound/uda1341.h | 2 | ||||
-rw-r--r-- | include/sound/version.h | 4 |
11 files changed, 201 insertions, 93 deletions
diff --git a/include/sound/ad1843.h b/include/sound/ad1843.h new file mode 100644 index 00000000000..b236a9d1d6e --- /dev/null +++ b/include/sound/ad1843.h @@ -0,0 +1,46 @@ +/* + * This file is subject to the terms and conditions of the GNU General Public + * License. See the file "COPYING" in the main directory of this archive + * for more details. + * + * Copyright 2003 Vivien Chappelier <vivien.chappelier@linux-mips.org> + * Copyright 2008 Thomas Bogendoerfer <tsbogend@franken.de> + */ + +#ifndef __SOUND_AD1843_H +#define __SOUND_AD1843_H + +struct snd_ad1843 { + void *chip; + int (*read)(void *chip, int reg); + int (*write)(void *chip, int reg, int val); +}; + +#define AD1843_GAIN_RECLEV 0 +#define AD1843_GAIN_LINE 1 +#define AD1843_GAIN_LINE_2 2 +#define AD1843_GAIN_MIC 3 +#define AD1843_GAIN_PCM_0 4 +#define AD1843_GAIN_PCM_1 5 +#define AD1843_GAIN_SIZE (AD1843_GAIN_PCM_1+1) + +int ad1843_get_gain_max(struct snd_ad1843 *ad1843, int id); +int ad1843_get_gain(struct snd_ad1843 *ad1843, int id); +int ad1843_set_gain(struct snd_ad1843 *ad1843, int id, int newval); +int ad1843_get_recsrc(struct snd_ad1843 *ad1843); +int ad1843_set_recsrc(struct snd_ad1843 *ad1843, int newsrc); +void ad1843_setup_dac(struct snd_ad1843 *ad1843, + unsigned int id, + unsigned int framerate, + snd_pcm_format_t fmt, + unsigned int channels); +void ad1843_shutdown_dac(struct snd_ad1843 *ad1843, + unsigned int id); +void ad1843_setup_adc(struct snd_ad1843 *ad1843, + unsigned int framerate, + snd_pcm_format_t fmt, + unsigned int channels); +void ad1843_shutdown_adc(struct snd_ad1843 *ad1843); +int ad1843_init(struct snd_ad1843 *ad1843); + +#endif /* __SOUND_AD1843_H */ diff --git a/include/sound/control.h b/include/sound/control.h index 3dc1291f52d..4721b4bba05 100644 --- a/include/sound/control.h +++ b/include/sound/control.h @@ -129,9 +129,6 @@ int snd_ctl_unregister_ioctl_compat(snd_kctl_ioctl_func_t fcn); #define snd_ctl_unregister_ioctl_compat(fcn) #endif -int snd_ctl_elem_read(struct snd_card *card, struct snd_ctl_elem_value *control); -int snd_ctl_elem_write(struct snd_card *card, struct snd_ctl_file *file, struct snd_ctl_elem_value *control); - static inline unsigned int snd_ctl_get_ioffnum(struct snd_kcontrol *kctl, struct snd_ctl_elem_id *id) { return id->numid - kctl->id.numid; diff --git a/include/sound/core.h b/include/sound/core.h index 695ee53488a..558b96284bd 100644 --- a/include/sound/core.h +++ b/include/sound/core.h @@ -412,13 +412,13 @@ void snd_verbose_printd(const char *file, int line, const char *format, ...) #endif /* CONFIG_SND_DEBUG */ -#ifdef CONFIG_SND_DEBUG_DETECT +#ifdef CONFIG_SND_DEBUG_VERBOSE /** * snd_printdd - debug printk * @format: format string * * Works like snd_printk() for debugging purposes. - * Ignored when CONFIG_SND_DEBUG_DETECT is not set. + * Ignored when CONFIG_SND_DEBUG_VERBOSE is not set. */ #define snd_printdd(format, args...) snd_printk(format, ##args) #else @@ -442,7 +442,7 @@ struct snd_pci_quirk { unsigned short subvendor; /* PCI subvendor ID */ unsigned short subdevice; /* PCI subdevice ID */ int value; /* value */ -#ifdef CONFIG_SND_DEBUG_DETECT +#ifdef CONFIG_SND_DEBUG_VERBOSE const char *name; /* name of the device (optional) */ #endif }; @@ -450,7 +450,7 @@ struct snd_pci_quirk { #define _SND_PCI_QUIRK_ID(vend,dev) \ .subvendor = (vend), .subdevice = (dev) #define SND_PCI_QUIRK_ID(vend,dev) {_SND_PCI_QUIRK_ID(vend, dev)} -#ifdef CONFIG_SND_DEBUG_DETECT +#ifdef CONFIG_SND_DEBUG_VERBOSE #define SND_PCI_QUIRK(vend,dev,xname,val) \ {_SND_PCI_QUIRK_ID(vend, dev), .value = (val), .name = (xname)} #else diff --git a/include/sound/cs4231-regs.h b/include/sound/cs4231-regs.h index e8d1f3e31f9..92647532c45 100644 --- a/include/sound/cs4231-regs.h +++ b/include/sound/cs4231-regs.h @@ -177,4 +177,12 @@ #define CS4236_RIGHT_WAVE 0x1c /* right wavetable serial port volume */ #define CS4236_VERSION 0x9c /* chip version and ID */ +/* definitions for extended registers - OPTI93X */ +#define OPTi931_AUX_LEFT_INPUT 0x10 +#define OPTi931_AUX_RIGHT_INPUT 0x11 +#define OPTi93X_MIC_LEFT_INPUT 0x14 +#define OPTi93X_MIC_RIGHT_INPUT 0x15 +#define OPTi93X_OUT_LEFT 0x16 +#define OPTi93X_OUT_RIGHT 0x17 + #endif /* __SOUND_CS4231_REGS_H */ diff --git a/include/sound/cs4231.h b/include/sound/cs4231.h index 66055d702aa..f0785f9f4ae 100644 --- a/include/sound/cs4231.h +++ b/include/sound/cs4231.h @@ -58,6 +58,7 @@ /* compatible, but clones */ #define CS4231_HW_INTERWAVE 0x1000 /* InterWave chip */ #define CS4231_HW_OPL3SA2 0x1101 /* OPL3-SA2 chip, similar to cs4231 */ +#define CS4231_HW_OPTI93X 0x1102 /* Opti 930/931/933 */ /* defines for codec.hwshare */ #define CS4231_HWSHARE_IRQ (1<<0) @@ -120,6 +121,8 @@ unsigned char snd_cs4236_ext_in(struct snd_cs4231 *chip, unsigned char reg); void snd_cs4231_mce_up(struct snd_cs4231 *chip); void snd_cs4231_mce_down(struct snd_cs4231 *chip); +void snd_cs4231_overrange(struct snd_cs4231 *chip); + irqreturn_t snd_cs4231_interrupt(int irq, void *dev_id); const char *snd_cs4231_chip_id(struct snd_cs4231 *chip); diff --git a/include/sound/emu10k1.h b/include/sound/emu10k1.h index 7b7b9b13b4d..10ee28eac01 100644 --- a/include/sound/emu10k1.h +++ b/include/sound/emu10k1.h @@ -1670,6 +1670,7 @@ struct snd_emu_chip_details { unsigned char spi_dac; /* SPI interface for DAC */ unsigned char i2c_adc; /* I2C interface for ADC */ unsigned char adc_1361t; /* Use Philips 1361T ADC */ + unsigned char invert_shared_spdif; /* analog/digital switch inverted */ const char *driver; const char *name; const char *id; /* for backward compatibility - can be NULL if not needed */ diff --git a/include/sound/seq_kernel.h b/include/sound/seq_kernel.h index f023c1b97f8..3d9afb6a8c9 100644 --- a/include/sound/seq_kernel.h +++ b/include/sound/seq_kernel.h @@ -105,7 +105,7 @@ int snd_seq_event_port_attach(int client, struct snd_seq_port_callback *pcbp, int cap, int type, int midi_channels, int midi_voices, char *portname); int snd_seq_event_port_detach(int client, int port); -#ifdef CONFIG_KMOD +#ifdef CONFIG_MODULES void snd_seq_autoload_lock(void); void snd_seq_autoload_unlock(void); #else diff --git a/include/sound/soc-dapm.h b/include/sound/soc-dapm.h index a105b01e06d..3030fdc6981 100644 --- a/include/sound/soc-dapm.h +++ b/include/sound/soc-dapm.h @@ -130,6 +130,13 @@ { .id = snd_soc_dapm_adc, .name = wname, .sname = stname, .reg = wreg, \ .shift = wshift, .invert = winvert} +/* generic register modifier widget */ +#define SND_SOC_DAPM_REG(wid, wname, wreg, wshift, wmask, won_val, woff_val) \ +{ .id = wid, .name = wname, .kcontrols = NULL, .num_kcontrols = 0, \ + .reg = -((wreg) + 1), .shift = wshift, .mask = wmask, \ + .on_val = won_val, .off_val = woff_val, .event = dapm_reg_event, \ + .event_flags = SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMD} + /* dapm kcontrol types */ #define SOC_DAPM_SINGLE(xname, reg, shift, max, invert) \ { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, \ @@ -193,6 +200,7 @@ struct snd_soc_dapm_widget; enum snd_soc_dapm_type; struct snd_soc_dapm_path; struct snd_soc_dapm_pin; +struct snd_soc_dapm_route; /* dapm controls */ int snd_soc_dapm_put_volsw(struct snd_kcontrol *kcontrol, @@ -205,25 +213,32 @@ int snd_soc_dapm_put_enum_double(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol); int snd_soc_dapm_new_control(struct snd_soc_codec *codec, const struct snd_soc_dapm_widget *widget); +int snd_soc_dapm_new_controls(struct snd_soc_codec *codec, + const struct snd_soc_dapm_widget *widget, + int num); /* dapm path setup */ -int snd_soc_dapm_connect_input(struct snd_soc_codec *codec, +int __deprecated snd_soc_dapm_connect_input(struct snd_soc_codec *codec, const char *sink_name, const char *control_name, const char *src_name); int snd_soc_dapm_new_widgets(struct snd_soc_codec *codec); void snd_soc_dapm_free(struct snd_soc_device *socdev); +int snd_soc_dapm_add_routes(struct snd_soc_codec *codec, + const struct snd_soc_dapm_route *route, int num); /* dapm events */ int snd_soc_dapm_stream_event(struct snd_soc_codec *codec, char *stream, int event); -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); /* dapm sys fs - used by the core */ int snd_soc_dapm_sys_add(struct device *dev); -/* dapm audio endpoint control */ -int snd_soc_dapm_set_endpoint(struct snd_soc_codec *codec, - char *pin, int status); -int snd_soc_dapm_sync_endpoints(struct snd_soc_codec *codec); +/* dapm audio pin control and status */ +int snd_soc_dapm_enable_pin(struct snd_soc_codec *codec, char *pin); +int snd_soc_dapm_disable_pin(struct snd_soc_codec *codec, char *pin); +int snd_soc_dapm_get_pin_status(struct snd_soc_codec *codec, char *pin); +int snd_soc_dapm_sync(struct snd_soc_codec *codec); /* dapm widget types */ enum snd_soc_dapm_type { @@ -245,6 +260,18 @@ enum snd_soc_dapm_type { snd_soc_dapm_post, /* machine specific post widget - exec last */ }; +/* + * DAPM audio route definition. + * + * Defines an audio route originating at source via control and finishing + * at sink. + */ +struct snd_soc_dapm_route { + const char *sink; + const char *control; + const char *source; +}; + /* dapm audio path between two widgets */ struct snd_soc_dapm_path { char *name; @@ -277,6 +304,9 @@ struct snd_soc_dapm_widget { unsigned char shift; /* bits to shift */ unsigned int saved_value; /* widget saved value */ unsigned int value; /* widget current value */ + unsigned int mask; /* non-shifted mask */ + unsigned int on_val; /* on state value */ + unsigned int off_val; /* off state value */ unsigned char power:1; /* block power status */ unsigned char invert:1; /* invert the power bit */ unsigned char active:1; /* active stream on DAC, ADC's */ diff --git a/include/sound/soc.h b/include/sound/soc.h index d3c8c033dff..1890d87c520 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -73,6 +73,15 @@ .get = snd_soc_get_volsw_2r, .put = snd_soc_put_volsw_2r, \ .private_value = (reg_left) | ((shift) << 8) | \ ((max) << 12) | ((invert) << 20) | ((reg_right) << 24) } +#define SOC_DOUBLE_S8_TLV(xname, reg, min, max, 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_s8, .get = snd_soc_get_volsw_s8, \ + .put = snd_soc_put_volsw_s8, \ + .private_value = (reg) | (((signed char)max) << 16) | \ + (((signed char)min) << 24) } #define SOC_ENUM_DOUBLE(xreg, xshift_l, xshift_r, xmask, xtexts) \ { .reg = xreg, .shift_l = xshift_l, .shift_r = xshift_r, \ .mask = xmask, .texts = xtexts } @@ -91,6 +100,15 @@ .info = snd_soc_info_volsw, \ .get = xhandler_get, .put = xhandler_put, \ .private_value = SOC_SINGLE_VALUE(xreg, xshift, xmask, xinvert) } +#define SOC_SINGLE_EXT_TLV(xname, xreg, xshift, xmask, xinvert,\ + xhandler_get, xhandler_put, 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 = xhandler_get, .put = xhandler_put, \ + .private_value = SOC_SINGLE_VALUE(xreg, xshift, xmask, xinvert) } #define SOC_SINGLE_BOOL_EXT(xname, xdata, xhandler_get, xhandler_put) \ { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, \ .info = snd_soc_info_bool_ext, \ @@ -103,6 +121,24 @@ .private_value = (unsigned long)&xenum } /* + * Bias levels + * + * @ON: Bias is fully on for audio playback and capture operations. + * @PREPARE: Prepare for audio operations. Called before DAPM switching for + * stream start and stop operations. + * @STANDBY: Low power standby state when no playback/capture operations are + * in progress. NOTE: The transition time between STANDBY and ON + * should be as fast as possible and no longer than 10ms. + * @OFF: Power Off. No restrictions on transition times. + */ +enum snd_soc_bias_level { + SND_SOC_BIAS_ON, + SND_SOC_BIAS_PREPARE, + SND_SOC_BIAS_STANDBY, + SND_SOC_BIAS_OFF, +}; + +/* * Digital Audio Interface (DAI) types */ #define SND_SOC_DAI_AC97 0x1 @@ -185,8 +221,7 @@ struct snd_soc_pcm_stream; struct snd_soc_ops; struct snd_soc_dai_mode; struct snd_soc_pcm_runtime; -struct snd_soc_codec_dai; -struct snd_soc_cpu_dai; +struct snd_soc_dai; struct snd_soc_codec; struct snd_soc_machine_config; struct soc_enum; @@ -221,6 +256,27 @@ int snd_soc_new_ac97_codec(struct snd_soc_codec *codec, struct snd_ac97_bus_ops *ops, int num); void snd_soc_free_ac97_codec(struct snd_soc_codec *codec); +/* Digital Audio Interface clocking API.*/ +int snd_soc_dai_set_sysclk(struct snd_soc_dai *dai, int clk_id, + unsigned int freq, int dir); + +int snd_soc_dai_set_clkdiv(struct snd_soc_dai *dai, + int div_id, int div); + +int snd_soc_dai_set_pll(struct snd_soc_dai *dai, + int pll_id, unsigned int freq_in, unsigned int freq_out); + +/* Digital Audio interface formatting */ +int snd_soc_dai_set_fmt(struct snd_soc_dai *dai, unsigned int fmt); + +int snd_soc_dai_set_tdm_slot(struct snd_soc_dai *dai, + unsigned int mask, int slots); + +int snd_soc_dai_set_tristate(struct snd_soc_dai *dai, int tristate); + +/* Digital Audio Interface mute */ +int snd_soc_dai_digital_mute(struct snd_soc_dai *dai, int mute); + /* *Controls */ @@ -249,6 +305,12 @@ int snd_soc_get_volsw_2r(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol); int snd_soc_put_volsw_2r(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol); +int snd_soc_info_volsw_s8(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_info *uinfo); +int snd_soc_get_volsw_s8(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol); +int snd_soc_put_volsw_s8(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol); /* SoC PCM stream information */ struct snd_soc_pcm_stream { @@ -272,87 +334,45 @@ struct snd_soc_ops { int (*trigger)(struct snd_pcm_substream *, int); }; -/* ASoC codec DAI ops */ -struct snd_soc_codec_ops { - /* codec DAI clocking configuration */ - int (*set_sysclk)(struct snd_soc_codec_dai *codec_dai, +/* ASoC DAI ops */ +struct snd_soc_dai_ops { + /* DAI clocking configuration */ + int (*set_sysclk)(struct snd_soc_dai *dai, int clk_id, unsigned int freq, int dir); - int (*set_pll)(struct snd_soc_codec_dai *codec_dai, + int (*set_pll)(struct snd_soc_dai *dai, int pll_id, unsigned int freq_in, unsigned int freq_out); - int (*set_clkdiv)(struct snd_soc_codec_dai *codec_dai, - int div_id, int div); + int (*set_clkdiv)(struct snd_soc_dai *dai, int div_id, int div); - /* CPU DAI format configuration */ - int (*set_fmt)(struct snd_soc_codec_dai *codec_dai, - unsigned int fmt); - int (*set_tdm_slot)(struct snd_soc_codec_dai *codec_dai, + /* DAI format configuration */ + int (*set_fmt)(struct snd_soc_dai *dai, unsigned int fmt); + int (*set_tdm_slot)(struct snd_soc_dai *dai, unsigned int mask, int slots); - int (*set_tristate)(struct snd_soc_codec_dai *, int tristate); + int (*set_tristate)(struct snd_soc_dai *dai, int tristate); /* digital mute */ - int (*digital_mute)(struct snd_soc_codec_dai *, int mute); -}; - -/* ASoC cpu DAI ops */ -struct snd_soc_cpu_ops { - /* CPU DAI clocking configuration */ - int (*set_sysclk)(struct snd_soc_cpu_dai *cpu_dai, - int clk_id, unsigned int freq, int dir); - int (*set_clkdiv)(struct snd_soc_cpu_dai *cpu_dai, - int div_id, int div); - int (*set_pll)(struct snd_soc_cpu_dai *cpu_dai, - int pll_id, unsigned int freq_in, unsigned int freq_out); - - /* CPU DAI format configuration */ - int (*set_fmt)(struct snd_soc_cpu_dai *cpu_dai, - unsigned int fmt); - int (*set_tdm_slot)(struct snd_soc_cpu_dai *cpu_dai, - unsigned int mask, int slots); - int (*set_tristate)(struct snd_soc_cpu_dai *, int tristate); -}; - -/* SoC Codec DAI */ -struct snd_soc_codec_dai { - char *name; - int id; - unsigned char type; - - /* DAI capabilities */ - struct snd_soc_pcm_stream playback; - struct snd_soc_pcm_stream capture; - - /* DAI runtime info */ - struct snd_soc_codec *codec; - unsigned int active; - unsigned char pop_wait:1; - - /* ops */ - struct snd_soc_ops ops; - struct snd_soc_codec_ops dai_ops; - - /* DAI private data */ - void *private_data; + int (*digital_mute)(struct snd_soc_dai *dai, int mute); }; -/* SoC CPU DAI */ -struct snd_soc_cpu_dai { - +/* SoC DAI (Digital Audio Interface) */ +struct snd_soc_dai { /* DAI description */ char *name; unsigned int id; unsigned char type; /* DAI callbacks */ - int (*probe)(struct platform_device *pdev); - void (*remove)(struct platform_device *pdev); + int (*probe)(struct platform_device *pdev, + struct snd_soc_dai *dai); + void (*remove)(struct platform_device *pdev, + struct snd_soc_dai *dai); int (*suspend)(struct platform_device *pdev, - struct snd_soc_cpu_dai *cpu_dai); + struct snd_soc_dai *dai); int (*resume)(struct platform_device *pdev, - struct snd_soc_cpu_dai *cpu_dai); + struct snd_soc_dai *dai); /* ops */ struct snd_soc_ops ops; - struct snd_soc_cpu_ops dai_ops; + struct snd_soc_dai_ops dai_ops; /* DAI capabilities */ struct snd_soc_pcm_stream capture; @@ -360,7 +380,9 @@ struct snd_soc_cpu_dai { /* DAI runtime info */ struct snd_pcm_runtime *runtime; - unsigned char active:1; + struct snd_soc_codec *codec; + unsigned int active; + unsigned char pop_wait:1; void *dma_data; /* DAI private data */ @@ -374,7 +396,8 @@ struct snd_soc_codec { struct mutex mutex; /* callbacks */ - int (*dapm_event)(struct snd_soc_codec *codec, int event); + int (*set_bias_level)(struct snd_soc_codec *, + enum snd_soc_bias_level level); /* runtime */ struct snd_card *card; @@ -396,12 +419,12 @@ struct snd_soc_codec { /* dapm */ struct list_head dapm_widgets; struct list_head dapm_paths; - unsigned int dapm_state; - unsigned int suspend_dapm_state; + enum snd_soc_bias_level bias_level; + enum snd_soc_bias_level suspend_bias_level; struct delayed_work delayed_work; /* codec DAI's */ - struct snd_soc_codec_dai *dai; + struct snd_soc_dai *dai; unsigned int num_dai; }; @@ -420,12 +443,12 @@ struct snd_soc_platform { int (*probe)(struct platform_device *pdev); int (*remove)(struct platform_device *pdev); int (*suspend)(struct platform_device *pdev, - struct snd_soc_cpu_dai *cpu_dai); + struct snd_soc_dai *dai); int (*resume)(struct platform_device *pdev, - struct snd_soc_cpu_dai *cpu_dai); + struct snd_soc_dai *dai); /* pcm creation and destruction */ - int (*pcm_new)(struct snd_card *, struct snd_soc_codec_dai *, + int (*pcm_new)(struct snd_card *, struct snd_soc_dai *, struct snd_pcm *); void (*pcm_free)(struct snd_pcm *); @@ -439,8 +462,8 @@ struct snd_soc_dai_link { char *stream_name; /* Stream name */ /* DAI */ - struct snd_soc_codec_dai *codec_dai; - struct snd_soc_cpu_dai *cpu_dai; + struct snd_soc_dai *codec_dai; + struct snd_soc_dai *cpu_dai; /* machine stream operations */ struct snd_soc_ops *ops; @@ -467,7 +490,8 @@ struct snd_soc_machine { int (*resume_post)(struct platform_device *pdev); /* callbacks */ - int (*dapm_event)(struct snd_soc_machine *, int event); + int (*set_bias_level)(struct snd_soc_machine *, + enum snd_soc_bias_level level); /* CPU <--> Codec DAI links */ struct snd_soc_dai_link *dai_link; @@ -482,6 +506,7 @@ struct snd_soc_device { struct snd_soc_codec *codec; struct snd_soc_codec_device *codec_dev; struct delayed_work delayed_work; + struct work_struct deferred_resume_work; void *codec_data; }; diff --git a/include/sound/uda1341.h b/include/sound/uda1341.h index 2e564bfb37f..110d5dc3a2b 100644 --- a/include/sound/uda1341.h +++ b/include/sound/uda1341.h @@ -15,8 +15,6 @@ * features support */ -/* $Id: uda1341.h,v 1.8 2005/11/17 14:17:21 tiwai Exp $ */ - #define UDA1341_ALSA_NAME "snd-uda1341" /* diff --git a/include/sound/version.h b/include/sound/version.h index ed6fb2eb1ea..6b78aff273a 100644 --- a/include/sound/version.h +++ b/include/sound/version.h @@ -1,3 +1,3 @@ -/* include/version.h. Generated by alsa/ksync script. */ -#define CONFIG_SND_VERSION "1.0.16" +/* include/version.h */ +#define CONFIG_SND_VERSION "1.0.17" #define CONFIG_SND_DATE "" |