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authorLinus Torvalds <torvalds@linux-foundation.org>2013-05-03 09:10:23 -0700
committerLinus Torvalds <torvalds@linux-foundation.org>2013-05-03 09:10:23 -0700
commit9992ba72327fa0d8bdc9fb624e80f5cce338a711 (patch)
treee0bf31ae53cb19c44674df7e0d0343a26037ad34 /include/sound
parent00fdffb5131125dce0702bf61e24a806ec3aed80 (diff)
parent4ca231b2e6ed171107c5b21f9e92d1965fd6fd9e (diff)
Merge tag 'sound-3.10' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound
Pull sound updates from Takashi Iwai: "Mostly many small changes spread as seen in diffstat in sound/* directory by this update. A significant change in the subsystem level is the introduction of snd_soc_component, which will help more generic handling of SoC and off-SoC components. Also, snd_BUG_ON() macro is enabled unconditionally now due to its misuses, so people might hit kernel warnings (it's a good thing for us). - compress-offload: support for capture by Charles Keepax - HD-audio: codec delay support by Dylan Reid - HD-audio: improvements/fixes in generic parser: better headphone mic and headset mic support, jack_modes hint consolidation, proper beep attach/detachment, generalized power filter controls by David Henningsson, et al - HD-audio: Improved management of HDMI codec pins/converters - HD-audio: Better pin/DAC assignment for VIA codecs - HD-audio: Haswell HDMI workarounds - HD-audio: ALC268 codec support, a few new quirks for Chromebooks - USB: regression fixes: USB-MIDI autopm fix, the recent ISO latency fix by Clemens Ladisch - USB: support for DSD formats by Daniel Mack - USB: A few UAC2 device endian/cock fixes by Eldad Zack - USB: quirks for Emu 192kHz support, Novation Twitch DJ controller, Yamaha THRxx devices - HDSPM: updates for TCO controls by Adrian Knoth - ASoC: Add a snd_soc_component object type for generic handling of SoC and off-SoC components by Kuninori Morimoto, - dmaengine: a large set of cleanups and conversions by Lars-Peter Clausen - ASoC DAPM: performance optimizations from Ryo Tsutsui - ASoC DAPM: support for mixer control sharing by Stephen Warren - ASoC: multiplatform ARM cleanups from Arnd Bergmann - ASoC: new codec drivers for AK5385 and TAS5086 from Daniel Mack" * tag 'sound-3.10' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound: (315 commits) ALSA: usb-audio: caiaq: fix endianness bug in snd_usb_caiaq_maschine_dispatch ALSA: asihpi: add format support check in snd_card_asihpi_capture_formats ALSA: pcm_format_to_bits strong-typed conversion ALSA: compress: fix the states to check for allowing read ALSA: hda - Move Thinkpad X220 to use auto parser ALSA: USB: adjust for changed 3.8 USB API ALSA: usb - Avoid unnecessary sample rate changes on USB 2.0 clock sources sound: oss/dmabuf: use dma_map_single ALSA: ali5451: use mdelay instead of large udelay constants ALSA: hda - Add the support for ALC286 codec ALSA: usb-audio: USB quirk for Yamaha THR10C ALSA: usb-audio: USB quirk for Yamaha THR5A ALSA: usb-audio: USB quirk for Yamaha THR10 ALSA: usb-audio: Fix autopm error during probing ALSA: snd-usb: try harder to find USB_DT_CS_ENDPOINT ALSA: sound kconfig typo ALSA: emu10k1: Fix dock firmware loading ASoC: ux500: forward declare msp_i2s_platform_data ASoC: davinci-mcasp: Add Support BCLK-to-LRCLK ratio for TDM modes ASoC: davinci-pcm, davinci-mcasp: Clean up active_serializers ...
Diffstat (limited to 'include/sound')
-rw-r--r--include/sound/compress_driver.h4
-rw-r--r--include/sound/control.h5
-rw-r--r--include/sound/core.h26
-rw-r--r--include/sound/dmaengine_pcm.h97
-rw-r--r--include/sound/emu10k1.h1
-rw-r--r--include/sound/pcm.h31
-rw-r--r--include/sound/soc-dai.h8
-rw-r--r--include/sound/soc-dapm.h1
-rw-r--r--include/sound/soc.h33
-rw-r--r--include/sound/tas5086.h7
-rw-r--r--include/sound/tegra_wm8903.h26
11 files changed, 165 insertions, 74 deletions
diff --git a/include/sound/compress_driver.h b/include/sound/compress_driver.h
index ff6c74153fa1..9031a26249b5 100644
--- a/include/sound/compress_driver.h
+++ b/include/sound/compress_driver.h
@@ -56,8 +56,6 @@ struct snd_compr_runtime {
u64 buffer_size;
u32 fragment_size;
u32 fragments;
- u64 hw_pointer;
- u64 app_pointer;
u64 total_bytes_available;
u64 total_bytes_transferred;
wait_queue_head_t sleep;
@@ -121,7 +119,7 @@ struct snd_compr_ops {
int (*trigger)(struct snd_compr_stream *stream, int cmd);
int (*pointer)(struct snd_compr_stream *stream,
struct snd_compr_tstamp *tstamp);
- int (*copy)(struct snd_compr_stream *stream, const char __user *buf,
+ int (*copy)(struct snd_compr_stream *stream, char __user *buf,
size_t count);
int (*mmap)(struct snd_compr_stream *stream,
struct vm_area_struct *vma);
diff --git a/include/sound/control.h b/include/sound/control.h
index 8332e865c759..34bc93d80d55 100644
--- a/include/sound/control.h
+++ b/include/sound/control.h
@@ -189,7 +189,6 @@ int _snd_ctl_add_slave(struct snd_kcontrol *master, struct snd_kcontrol *slave,
*
* Add a virtual slave control to the given master element created via
* snd_ctl_create_virtual_master() beforehand.
- * Returns zero if successful or a negative error code.
*
* All slaves must be the same type (returning the same information
* via info callback). The function doesn't check it, so it's your
@@ -199,6 +198,8 @@ int _snd_ctl_add_slave(struct snd_kcontrol *master, struct snd_kcontrol *slave,
* at most two channels,
* logarithmic volume control (dB level) thus no linear volume,
* master can only attenuate the volume without gain
+ *
+ * Return: Zero if successful or a negative error code.
*/
static inline int
snd_ctl_add_slave(struct snd_kcontrol *master, struct snd_kcontrol *slave)
@@ -219,6 +220,8 @@ snd_ctl_add_slave(struct snd_kcontrol *master, struct snd_kcontrol *slave)
* When the control peeks the hardware values directly and the value
* can be changed by other means than the put callback of the element,
* this function should be used to keep the value always up-to-date.
+ *
+ * Return: Zero if successful or a negative error code.
*/
static inline int
snd_ctl_add_slave_uncached(struct snd_kcontrol *master,
diff --git a/include/sound/core.h b/include/sound/core.h
index 7cede2d6aa86..5bfe5136441c 100644
--- a/include/sound/core.h
+++ b/include/sound/core.h
@@ -229,7 +229,7 @@ int snd_register_device_for_dev(int type, struct snd_card *card,
* This function uses the card's device pointer to link to the
* correct &struct device.
*
- * Returns zero if successful, or a negative error code on failure.
+ * Return: Zero if successful, or a negative error code on failure.
*/
static inline int snd_register_device(int type, struct snd_card *card, int dev,
const struct file_operations *f_ops,
@@ -379,18 +379,10 @@ void __snd_printk(unsigned int level, const char *file, int line,
* snd_BUG_ON - debugging check macro
* @cond: condition to evaluate
*
- * When CONFIG_SND_DEBUG is set, this macro evaluates the given condition,
- * and call WARN() and returns the value if it's non-zero.
- *
- * When CONFIG_SND_DEBUG is not set, this just returns zero, and the given
- * condition is ignored.
- *
- * NOTE: the argument won't be evaluated at all when CONFIG_SND_DEBUG=n.
- * Thus, don't put any statement that influences on the code behavior,
- * such as pre/post increment, to the argument of this macro.
- * If you want to evaluate and give a warning, use standard WARN_ON().
+ * Has the same behavior as WARN_ON when CONFIG_SND_DEBUG is set,
+ * otherwise just evaluates the conditional and returns the value.
*/
-#define snd_BUG_ON(cond) WARN((cond), "BUG? (%s)\n", __stringify(cond))
+#define snd_BUG_ON(cond) WARN_ON((cond))
#else /* !CONFIG_SND_DEBUG */
@@ -400,11 +392,11 @@ __printf(2, 3)
static inline void _snd_printd(int level, const char *format, ...) {}
#define snd_BUG() do { } while (0)
-static inline int __snd_bug_on(int cond)
-{
- return 0;
-}
-#define snd_BUG_ON(cond) __snd_bug_on(0 && (cond)) /* always false */
+
+#define snd_BUG_ON(condition) ({ \
+ int __ret_warn_on = !!(condition); \
+ unlikely(__ret_warn_on); \
+})
#endif /* CONFIG_SND_DEBUG */
diff --git a/include/sound/dmaengine_pcm.h b/include/sound/dmaengine_pcm.h
index b877334bbb0f..f11c35cd5532 100644
--- a/include/sound/dmaengine_pcm.h
+++ b/include/sound/dmaengine_pcm.h
@@ -16,6 +16,7 @@
#define __SOUND_DMAENGINE_PCM_H__
#include <sound/pcm.h>
+#include <sound/soc.h>
#include <linux/dmaengine.h>
/**
@@ -32,9 +33,6 @@ snd_pcm_substream_to_dma_direction(const struct snd_pcm_substream *substream)
return DMA_DEV_TO_MEM;
}
-void snd_dmaengine_pcm_set_data(struct snd_pcm_substream *substream, void *data);
-void *snd_dmaengine_pcm_get_data(struct snd_pcm_substream *substream);
-
int snd_hwparams_to_dma_slave_config(const struct snd_pcm_substream *substream,
const struct snd_pcm_hw_params *params, struct dma_slave_config *slave_config);
int snd_dmaengine_pcm_trigger(struct snd_pcm_substream *substream, int cmd);
@@ -42,9 +40,100 @@ snd_pcm_uframes_t snd_dmaengine_pcm_pointer(struct snd_pcm_substream *substream)
snd_pcm_uframes_t snd_dmaengine_pcm_pointer_no_residue(struct snd_pcm_substream *substream);
int snd_dmaengine_pcm_open(struct snd_pcm_substream *substream,
- dma_filter_fn filter_fn, void *filter_data);
+ struct dma_chan *chan);
int snd_dmaengine_pcm_close(struct snd_pcm_substream *substream);
+int snd_dmaengine_pcm_open_request_chan(struct snd_pcm_substream *substream,
+ dma_filter_fn filter_fn, void *filter_data);
+int snd_dmaengine_pcm_close_release_chan(struct snd_pcm_substream *substream);
+
+struct dma_chan *snd_dmaengine_pcm_request_channel(dma_filter_fn filter_fn,
+ void *filter_data);
struct dma_chan *snd_dmaengine_pcm_get_chan(struct snd_pcm_substream *substream);
+/**
+ * struct snd_dmaengine_dai_dma_data - DAI DMA configuration data
+ * @addr: Address of the DAI data source or destination register.
+ * @addr_width: Width of the DAI data source or destination register.
+ * @maxburst: Maximum number of words(note: words, as in units of the
+ * src_addr_width member, not bytes) that can be send to or received from the
+ * DAI in one burst.
+ * @slave_id: Slave requester id for the DMA channel.
+ * @filter_data: Custom DMA channel filter data, this will usually be used when
+ * requesting the DMA channel.
+ */
+struct snd_dmaengine_dai_dma_data {
+ dma_addr_t addr;
+ enum dma_slave_buswidth addr_width;
+ u32 maxburst;
+ unsigned int slave_id;
+ void *filter_data;
+};
+
+void snd_dmaengine_pcm_set_config_from_dai_data(
+ const struct snd_pcm_substream *substream,
+ const struct snd_dmaengine_dai_dma_data *dma_data,
+ struct dma_slave_config *config);
+
+
+/*
+ * Try to request the DMA channel using compat_request_channel or
+ * compat_filter_fn if it couldn't be requested through devicetree.
+ */
+#define SND_DMAENGINE_PCM_FLAG_COMPAT BIT(0)
+/*
+ * Don't try to request the DMA channels through devicetree. This flag only
+ * makes sense if SND_DMAENGINE_PCM_FLAG_COMPAT is set as well.
+ */
+#define SND_DMAENGINE_PCM_FLAG_NO_DT BIT(1)
+/*
+ * The platforms dmaengine driver does not support reporting the amount of
+ * bytes that are still left to transfer.
+ */
+#define SND_DMAENGINE_PCM_FLAG_NO_RESIDUE BIT(2)
+/*
+ * The PCM is half duplex and the DMA channel is shared between capture and
+ * playback.
+ */
+#define SND_DMAENGINE_PCM_FLAG_HALF_DUPLEX BIT(3)
+
+/**
+ * struct snd_dmaengine_pcm_config - Configuration data for dmaengine based PCM
+ * @prepare_slave_config: Callback used to fill in the DMA slave_config for a
+ * PCM substream. Will be called from the PCM drivers hwparams callback.
+ * @compat_request_channel: Callback to request a DMA channel for platforms
+ * which do not use devicetree.
+ * @compat_filter_fn: Will be used as the filter function when requesting a
+ * channel for platforms which do not use devicetree. The filter parameter
+ * will be the DAI's DMA data.
+ * @pcm_hardware: snd_pcm_hardware struct to be used for the PCM.
+ * @prealloc_buffer_size: Size of the preallocated audio buffer.
+ *
+ * Note: If both compat_request_channel and compat_filter_fn are set
+ * compat_request_channel will be used to request the channel and
+ * compat_filter_fn will be ignored. Otherwise the channel will be requested
+ * using dma_request_channel with compat_filter_fn as the filter function.
+ */
+struct snd_dmaengine_pcm_config {
+ int (*prepare_slave_config)(struct snd_pcm_substream *substream,
+ struct snd_pcm_hw_params *params,
+ struct dma_slave_config *slave_config);
+ struct dma_chan *(*compat_request_channel)(
+ struct snd_soc_pcm_runtime *rtd,
+ struct snd_pcm_substream *substream);
+ dma_filter_fn compat_filter_fn;
+
+ const struct snd_pcm_hardware *pcm_hardware;
+ unsigned int prealloc_buffer_size;
+};
+
+int snd_dmaengine_pcm_register(struct device *dev,
+ const struct snd_dmaengine_pcm_config *config,
+ unsigned int flags);
+void snd_dmaengine_pcm_unregister(struct device *dev);
+
+int snd_dmaengine_pcm_prepare_slave_config(struct snd_pcm_substream *substream,
+ struct snd_pcm_hw_params *params,
+ struct dma_slave_config *slave_config);
+
#endif
diff --git a/include/sound/emu10k1.h b/include/sound/emu10k1.h
index f841ba4bacb8..dfb42ca6d043 100644
--- a/include/sound/emu10k1.h
+++ b/include/sound/emu10k1.h
@@ -1787,6 +1787,7 @@ struct snd_emu10k1 {
unsigned int next_free_voice;
const struct firmware *firmware;
+ const struct firmware *dock_fw;
#ifdef CONFIG_PM_SLEEP
unsigned int *saved_ptr;
diff --git a/include/sound/pcm.h b/include/sound/pcm.h
index 5ec42dbd2308..b48792fe386b 100644
--- a/include/sound/pcm.h
+++ b/include/sound/pcm.h
@@ -181,6 +181,8 @@ struct snd_pcm_ops {
#define SNDRV_PCM_FMTBIT_G723_24_1B _SNDRV_PCM_FMTBIT(G723_24_1B)
#define SNDRV_PCM_FMTBIT_G723_40 _SNDRV_PCM_FMTBIT(G723_40)
#define SNDRV_PCM_FMTBIT_G723_40_1B _SNDRV_PCM_FMTBIT(G723_40_1B)
+#define SNDRV_PCM_FMTBIT_DSD_U8 _SNDRV_PCM_FMTBIT(DSD_U8)
+#define SNDRV_PCM_FMTBIT_DSD_U16_LE _SNDRV_PCM_FMTBIT(DSD_U16_LE)
#ifdef SNDRV_LITTLE_ENDIAN
#define SNDRV_PCM_FMTBIT_S16 SNDRV_PCM_FMTBIT_S16_LE
@@ -659,7 +661,7 @@ static inline snd_pcm_sframes_t snd_pcm_capture_hw_avail(struct snd_pcm_runtime
*
* Checks whether enough free space is available on the playback buffer.
*
- * Returns non-zero if available, or zero if not.
+ * Return: Non-zero if available, or zero if not.
*/
static inline int snd_pcm_playback_ready(struct snd_pcm_substream *substream)
{
@@ -673,7 +675,7 @@ static inline int snd_pcm_playback_ready(struct snd_pcm_substream *substream)
*
* Checks whether enough capture data is available on the capture buffer.
*
- * Returns non-zero if available, or zero if not.
+ * Return: Non-zero if available, or zero if not.
*/
static inline int snd_pcm_capture_ready(struct snd_pcm_substream *substream)
{
@@ -685,10 +687,10 @@ static inline int snd_pcm_capture_ready(struct snd_pcm_substream *substream)
* snd_pcm_playback_data - check whether any data exists on the playback buffer
* @substream: the pcm substream instance
*
- * Checks whether any data exists on the playback buffer. If stop_threshold
- * is bigger or equal to boundary, then this function returns always non-zero.
+ * Checks whether any data exists on the playback buffer.
*
- * Returns non-zero if exists, or zero if not.
+ * Return: Non-zero if any data exists, or zero if not. If stop_threshold
+ * is bigger or equal to boundary, then this function returns always non-zero.
*/
static inline int snd_pcm_playback_data(struct snd_pcm_substream *substream)
{
@@ -705,7 +707,7 @@ static inline int snd_pcm_playback_data(struct snd_pcm_substream *substream)
*
* Checks whether the playback buffer is empty.
*
- * Returns non-zero if empty, or zero if not.
+ * Return: Non-zero if empty, or zero if not.
*/
static inline int snd_pcm_playback_empty(struct snd_pcm_substream *substream)
{
@@ -719,7 +721,7 @@ static inline int snd_pcm_playback_empty(struct snd_pcm_substream *substream)
*
* Checks whether the capture buffer is empty.
*
- * Returns non-zero if empty, or zero if not.
+ * Return: Non-zero if empty, or zero if not.
*/
static inline int snd_pcm_capture_empty(struct snd_pcm_substream *substream)
{
@@ -852,7 +854,7 @@ int snd_pcm_format_big_endian(snd_pcm_format_t format);
* snd_pcm_format_cpu_endian - Check the PCM format is CPU-endian
* @format: the format to check
*
- * Returns 1 if the given PCM format is CPU-endian, 0 if
+ * Return: 1 if the given PCM format is CPU-endian, 0 if
* opposite, or a negative error code if endian not specified.
*/
int snd_pcm_format_cpu_endian(snd_pcm_format_t format);
@@ -963,7 +965,7 @@ struct page *snd_pcm_lib_get_vmalloc_page(struct snd_pcm_substream *substream,
* contiguous in kernel virtual space, but not in physical memory. Use this
* if the buffer is accessed by kernel code but not by device DMA.
*
- * Returns 1 if the buffer was changed, 0 if not changed, or a negative error
+ * Return: 1 if the buffer was changed, 0 if not changed, or a negative error
* code.
*/
static int snd_pcm_lib_alloc_vmalloc_buffer
@@ -975,6 +977,9 @@ static int snd_pcm_lib_alloc_vmalloc_buffer
*
* This function works like snd_pcm_lib_alloc_vmalloc_buffer(), but uses
* vmalloc_32(), i.e., the pages are allocated from 32-bit-addressable memory.
+ *
+ * Return: 1 if the buffer was changed, 0 if not changed, or a negative error
+ * code.
*/
static int snd_pcm_lib_alloc_vmalloc_32_buffer
(struct snd_pcm_substream *substream, size_t size);
@@ -1070,6 +1075,8 @@ const char *snd_pcm_format_name(snd_pcm_format_t format);
/**
* snd_pcm_stream_str - Get a string naming the direction of a stream
* @substream: the pcm substream instance
+ *
+ * Return: A string naming the direction of the stream.
*/
static inline const char *snd_pcm_stream_str(struct snd_pcm_substream *substream)
{
@@ -1126,4 +1133,10 @@ int snd_pcm_add_chmap_ctls(struct snd_pcm *pcm, int stream,
unsigned long private_value,
struct snd_pcm_chmap **info_ret);
+/* Strong-typed conversion of pcm_format to bitwise */
+static inline u64 pcm_format_to_bits(snd_pcm_format_t pcm_format)
+{
+ return 1ULL << (__force int) pcm_format;
+}
+
#endif /* __SOUND_PCM_H */
diff --git a/include/sound/soc-dai.h b/include/sound/soc-dai.h
index 3d84808952b9..ae9a227d35d3 100644
--- a/include/sound/soc-dai.h
+++ b/include/sound/soc-dai.h
@@ -95,14 +95,6 @@ struct snd_soc_dai_driver;
struct snd_soc_dai;
struct snd_ac97_bus_ops;
-/* Digital Audio Interface registration */
-int snd_soc_register_dai(struct device *dev,
- struct snd_soc_dai_driver *dai_drv);
-void snd_soc_unregister_dai(struct device *dev);
-int snd_soc_register_dais(struct device *dev,
- struct snd_soc_dai_driver *dai_drv, size_t count);
-void snd_soc_unregister_dais(struct device *dev, size_t count);
-
/* Digital Audio Interface clocking API.*/
int snd_soc_dai_set_sysclk(struct snd_soc_dai *dai, int clk_id,
unsigned int freq, int dir);
diff --git a/include/sound/soc-dapm.h b/include/sound/soc-dapm.h
index 44a30b108683..d4609029f014 100644
--- a/include/sound/soc-dapm.h
+++ b/include/sound/soc-dapm.h
@@ -566,7 +566,6 @@ struct snd_soc_dapm_update {
/* DAPM context */
struct snd_soc_dapm_context {
- int n_widgets; /* number of widgets in this context */
enum snd_soc_bias_level bias_level;
enum snd_soc_bias_level suspend_bias_level;
struct delayed_work delayed_work;
diff --git a/include/sound/soc.h b/include/sound/soc.h
index a6a059ca3874..85c15226103b 100644
--- a/include/sound/soc.h
+++ b/include/sound/soc.h
@@ -324,6 +324,8 @@ struct snd_soc_dai_link;
struct snd_soc_platform_driver;
struct snd_soc_codec;
struct snd_soc_codec_driver;
+struct snd_soc_component;
+struct snd_soc_component_driver;
struct soc_enum;
struct snd_soc_jack;
struct snd_soc_jack_zone;
@@ -371,12 +373,20 @@ int snd_soc_suspend(struct device *dev);
int snd_soc_resume(struct device *dev);
int snd_soc_poweroff(struct device *dev);
int snd_soc_register_platform(struct device *dev,
- struct snd_soc_platform_driver *platform_drv);
+ const struct snd_soc_platform_driver *platform_drv);
void snd_soc_unregister_platform(struct device *dev);
+int snd_soc_add_platform(struct device *dev, struct snd_soc_platform *platform,
+ const struct snd_soc_platform_driver *platform_drv);
+void snd_soc_remove_platform(struct snd_soc_platform *platform);
+struct snd_soc_platform *snd_soc_lookup_platform(struct device *dev);
int snd_soc_register_codec(struct device *dev,
const struct snd_soc_codec_driver *codec_drv,
struct snd_soc_dai_driver *dai_drv, int num_dai);
void snd_soc_unregister_codec(struct device *dev);
+int snd_soc_register_component(struct device *dev,
+ const struct snd_soc_component_driver *cmpnt_drv,
+ struct snd_soc_dai_driver *dai_drv, int num_dai);
+void snd_soc_unregister_component(struct device *dev);
int snd_soc_codec_volatile_register(struct snd_soc_codec *codec,
unsigned int reg);
int snd_soc_codec_readable_register(struct snd_soc_codec *codec,
@@ -801,10 +811,10 @@ struct snd_soc_platform_driver {
struct snd_soc_dai *);
/* platform stream pcm ops */
- struct snd_pcm_ops *ops;
+ const struct snd_pcm_ops *ops;
/* platform stream compress ops */
- struct snd_compr_ops *compr_ops;
+ const struct snd_compr_ops *compr_ops;
/* platform stream completion event */
int (*stream_event)(struct snd_soc_dapm_context *dapm, int event);
@@ -823,7 +833,7 @@ struct snd_soc_platform {
const char *name;
int id;
struct device *dev;
- struct snd_soc_platform_driver *driver;
+ const struct snd_soc_platform_driver *driver;
struct mutex mutex;
unsigned int suspended:1; /* platform is suspended */
@@ -841,6 +851,20 @@ struct snd_soc_platform {
#endif
};
+struct snd_soc_component_driver {
+ const char *name;
+};
+
+struct snd_soc_component {
+ const char *name;
+ int id;
+ int num_dai;
+ struct device *dev;
+ struct list_head list;
+
+ const struct snd_soc_component_driver *driver;
+};
+
struct snd_soc_dai_link {
/* config - must be set by machine driver */
const char *name; /* Codec name */
@@ -1086,7 +1110,6 @@ struct soc_enum {
unsigned int mask;
const char * const *texts;
const unsigned int *values;
- void *dapm;
};
/* codec IO */
diff --git a/include/sound/tas5086.h b/include/sound/tas5086.h
new file mode 100644
index 000000000000..aac481b7db8f
--- /dev/null
+++ b/include/sound/tas5086.h
@@ -0,0 +1,7 @@
+#ifndef _SND_SOC_CODEC_TAS5086_H_
+#define _SND_SOC_CODEC_TAS5086_H_
+
+#define TAS5086_CLK_IDX_MCLK 0
+#define TAS5086_CLK_IDX_SCLK 1
+
+#endif /* _SND_SOC_CODEC_TAS5086_H_ */
diff --git a/include/sound/tegra_wm8903.h b/include/sound/tegra_wm8903.h
deleted file mode 100644
index 57b202ee97c3..000000000000
--- a/include/sound/tegra_wm8903.h
+++ /dev/null
@@ -1,26 +0,0 @@
-/*
- * Copyright 2011 NVIDIA, Inc.
- *
- * This software is licensed under the terms of the GNU General Public
- * License version 2, as published by the Free Software Foundation, and
- * may be copied, distributed, and modified under those terms.
- *
- * 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.
- *
- */
-
-#ifndef __SOUND_TEGRA_WM38903_H
-#define __SOUND_TEGRA_WM38903_H
-
-struct tegra_wm8903_platform_data {
- int gpio_spkr_en;
- int gpio_hp_det;
- int gpio_hp_mute;
- int gpio_int_mic_en;
- int gpio_ext_mic_en;
-};
-
-#endif