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authorLinus Torvalds <torvalds@linux-foundation.org>2008-07-14 13:26:07 -0700
committerLinus Torvalds <torvalds@linux-foundation.org>2008-07-14 13:26:07 -0700
commitb5cf43c47b05c8deb10f9674d541dddbdec0e341 (patch)
tree41c9b71c40f5f0d3cd702f0b602254867630e6a1 /include
parentb7f80afa28866c257876c272d6c013e0dbed3c31 (diff)
parentfe0a3fe324811385b64790d42079bf534798a0cd (diff)
Merge branch 'for-linus' of git://git.alsa-project.org/alsa-kernel
* 'for-linus' of git://git.alsa-project.org/alsa-kernel: (179 commits) ALSA: Release v1.0.17 ALSA: correct kcalloc usage ALSA: ALSA driver for SGI O2 audio board ALSA: asoc: kbuild - only show menus for the current ASoC CPU platform. ALSA: ALSA driver for SGI HAL2 audio device ALSA: hda - Fix FSC V5505 model ALSA: hda - Fix missing init for unsol events on micsense model ALSA: hda - Fix internal mic vref pin setup ALSA: hda: 92hd71bxx PC Beep ALSA: HDA - HP dc7600 with pci sub IDs 0x103c/0x3011 belongs to hp-3013 model ALSA: usb-audio: add some Yamaha USB MIDI quirks ALSA: usb-audio: fix Yamaha KX quirk ALSA: ASoC: Au12x0/Au1550 PSC Audio support ALSA: Add Yamaha KX49 (USB MIDI controller) to usbquirks.h ALSA: ASoC: pxa2xx-ac97: fix warning due to missing argument in fuction declaration ALSA: tosa: fix compilation with new DAPM API ALSA: wavefront - add const ALSA: remove CONFIG_KMOD from sound ALSA: Fix a const to non-const assignment in the Digigram VXpocket sound driver ALSA: Fix a const pointer usage warning in the Digigram VX soundcard driver ...
Diffstat (limited to 'include')
-rw-r--r--include/asm-mips/mach-au1x00/au1xxx_psc.h8
-rw-r--r--include/sound/ad1843.h46
-rw-r--r--include/sound/control.h3
-rw-r--r--include/sound/core.h8
-rw-r--r--include/sound/cs4231-regs.h8
-rw-r--r--include/sound/cs4231.h3
-rw-r--r--include/sound/emu10k1.h1
-rw-r--r--include/sound/seq_kernel.h2
-rw-r--r--include/sound/soc-dapm.h42
-rw-r--r--include/sound/soc.h175
-rw-r--r--include/sound/uda1341.h2
-rw-r--r--include/sound/version.h4
12 files changed, 209 insertions, 93 deletions
diff --git a/include/asm-mips/mach-au1x00/au1xxx_psc.h b/include/asm-mips/mach-au1x00/au1xxx_psc.h
index dae4eca2417..892b7f168eb 100644
--- a/include/asm-mips/mach-au1x00/au1xxx_psc.h
+++ b/include/asm-mips/mach-au1x00/au1xxx_psc.h
@@ -204,6 +204,14 @@ typedef struct psc_i2s {
u32 psc_i2sudf;
} psc_i2s_t;
+#define PSC_I2SCFG_OFFSET 0x08
+#define PSC_I2SMASK_OFFSET 0x0C
+#define PSC_I2SPCR_OFFSET 0x10
+#define PSC_I2SSTAT_OFFSET 0x14
+#define PSC_I2SEVENT_OFFSET 0x18
+#define PSC_I2SRXTX_OFFSET 0x1C
+#define PSC_I2SUDF_OFFSET 0x20
+
/* I2S Config Register. */
#define PSC_I2SCFG_RT_MASK (3 << 30)
#define PSC_I2SCFG_RT_FIFO1 (0 << 30)
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 ""