diff options
Diffstat (limited to 'include')
| -rw-r--r-- | include/linux/mfd/arizona/registers.h | 42 | ||||
| -rw-r--r-- | include/linux/mfd/wm8994/pdata.h | 8 | ||||
| -rw-r--r-- | include/linux/usb/audio-v2.h | 2 | ||||
| -rw-r--r-- | include/sound/compress_driver.h | 4 | ||||
| -rw-r--r-- | include/sound/control.h | 5 | ||||
| -rw-r--r-- | include/sound/core.h | 26 | ||||
| -rw-r--r-- | include/sound/dmaengine_pcm.h | 97 | ||||
| -rw-r--r-- | include/sound/emu10k1.h | 1 | ||||
| -rw-r--r-- | include/sound/pcm.h | 31 | ||||
| -rw-r--r-- | include/sound/soc-dai.h | 8 | ||||
| -rw-r--r-- | include/sound/soc-dapm.h | 1 | ||||
| -rw-r--r-- | include/sound/soc.h | 33 | ||||
| -rw-r--r-- | include/sound/tas5086.h | 7 | ||||
| -rw-r--r-- | include/sound/tegra_wm8903.h | 26 | ||||
| -rw-r--r-- | include/uapi/sound/asound.h | 4 |
15 files changed, 220 insertions, 75 deletions
diff --git a/include/linux/mfd/arizona/registers.h b/include/linux/mfd/arizona/registers.h index f43aa7c8d040..715b6ba3d52a 100644 --- a/include/linux/mfd/arizona/registers.h +++ b/include/linux/mfd/arizona/registers.h @@ -85,12 +85,14 @@ #define ARIZONA_FLL1_CONTROL_6 0x176 #define ARIZONA_FLL1_LOOP_FILTER_TEST_1 0x177 #define ARIZONA_FLL1_NCO_TEST_0 0x178 +#define ARIZONA_FLL1_CONTROL_7 0x179 #define ARIZONA_FLL1_SYNCHRONISER_1 0x181 #define ARIZONA_FLL1_SYNCHRONISER_2 0x182 #define ARIZONA_FLL1_SYNCHRONISER_3 0x183 #define ARIZONA_FLL1_SYNCHRONISER_4 0x184 #define ARIZONA_FLL1_SYNCHRONISER_5 0x185 #define ARIZONA_FLL1_SYNCHRONISER_6 0x186 +#define ARIZONA_FLL1_SYNCHRONISER_7 0x187 #define ARIZONA_FLL1_SPREAD_SPECTRUM 0x189 #define ARIZONA_FLL1_GPIO_CLOCK 0x18A #define ARIZONA_FLL2_CONTROL_1 0x191 @@ -101,12 +103,14 @@ #define ARIZONA_FLL2_CONTROL_6 0x196 #define ARIZONA_FLL2_LOOP_FILTER_TEST_1 0x197 #define ARIZONA_FLL2_NCO_TEST_0 0x198 +#define ARIZONA_FLL2_CONTROL_7 0x199 #define ARIZONA_FLL2_SYNCHRONISER_1 0x1A1 #define ARIZONA_FLL2_SYNCHRONISER_2 0x1A2 #define ARIZONA_FLL2_SYNCHRONISER_3 0x1A3 #define ARIZONA_FLL2_SYNCHRONISER_4 0x1A4 #define ARIZONA_FLL2_SYNCHRONISER_5 0x1A5 #define ARIZONA_FLL2_SYNCHRONISER_6 0x1A6 +#define ARIZONA_FLL2_SYNCHRONISER_7 0x1A7 #define ARIZONA_FLL2_SPREAD_SPECTRUM 0x1A9 #define ARIZONA_FLL2_GPIO_CLOCK 0x1AA #define ARIZONA_MIC_CHARGE_PUMP_1 0x200 @@ -217,6 +221,8 @@ #define ARIZONA_PDM_SPK1_CTRL_2 0x491 #define ARIZONA_PDM_SPK2_CTRL_1 0x492 #define ARIZONA_PDM_SPK2_CTRL_2 0x493 +#define ARIZONA_SPK_CTRL_2 0x4B5 +#define ARIZONA_SPK_CTRL_3 0x4B6 #define ARIZONA_DAC_COMP_1 0x4DC #define ARIZONA_DAC_COMP_2 0x4DD #define ARIZONA_DAC_COMP_3 0x4DE @@ -1682,6 +1688,13 @@ #define ARIZONA_FLL1_FRC_INTEG_VAL_WIDTH 12 /* FLL1_FRC_INTEG_VAL - [11:0] */ /* + * R377 (0x179) - FLL1 Control 7 + */ +#define ARIZONA_FLL1_GAIN_MASK 0x003c /* FLL1_GAIN */ +#define ARIZONA_FLL1_GAIN_SHIFT 2 /* FLL1_GAIN */ +#define ARIZONA_FLL1_GAIN_WIDTH 4 /* FLL1_GAIN */ + +/* * R385 (0x181) - FLL1 Synchroniser 1 */ #define ARIZONA_FLL1_SYNC_ENA 0x0001 /* FLL1_SYNC_ENA */ @@ -1728,6 +1741,17 @@ #define ARIZONA_FLL1_CLK_SYNC_SRC_WIDTH 4 /* FLL1_CLK_SYNC_SRC - [3:0] */ /* + * R391 (0x187) - FLL1 Synchroniser 7 + */ +#define ARIZONA_FLL1_SYNC_GAIN_MASK 0x003c /* FLL1_SYNC_GAIN */ +#define ARIZONA_FLL1_SYNC_GAIN_SHIFT 2 /* FLL1_SYNC_GAIN */ +#define ARIZONA_FLL1_SYNC_GAIN_WIDTH 4 /* FLL1_SYNC_GAIN */ +#define ARIZONA_FLL1_SYNC_BW 0x0001 /* FLL1_SYNC_BW */ +#define ARIZONA_FLL1_SYNC_BW_MASK 0x0001 /* FLL1_SYNC_BW */ +#define ARIZONA_FLL1_SYNC_BW_SHIFT 0 /* FLL1_SYNC_BW */ +#define ARIZONA_FLL1_SYNC_BW_WIDTH 1 /* FLL1_SYNC_BW */ + +/* * R393 (0x189) - FLL1 Spread Spectrum */ #define ARIZONA_FLL1_SS_AMPL_MASK 0x0030 /* FLL1_SS_AMPL - [5:4] */ @@ -1820,6 +1844,13 @@ #define ARIZONA_FLL2_FRC_INTEG_VAL_WIDTH 12 /* FLL2_FRC_INTEG_VAL - [11:0] */ /* + * R409 (0x199) - FLL2 Control 7 + */ +#define ARIZONA_FLL2_GAIN_MASK 0x003c /* FLL2_GAIN */ +#define ARIZONA_FLL2_GAIN_SHIFT 2 /* FLL2_GAIN */ +#define ARIZONA_FLL2_GAIN_WIDTH 4 /* FLL2_GAIN */ + +/* * R417 (0x1A1) - FLL2 Synchroniser 1 */ #define ARIZONA_FLL2_SYNC_ENA 0x0001 /* FLL2_SYNC_ENA */ @@ -1866,6 +1897,17 @@ #define ARIZONA_FLL2_CLK_SYNC_SRC_WIDTH 4 /* FLL2_CLK_SYNC_SRC - [3:0] */ /* + * R423 (0x1A7) - FLL2 Synchroniser 7 + */ +#define ARIZONA_FLL2_SYNC_GAIN_MASK 0x003c /* FLL2_SYNC_GAIN */ +#define ARIZONA_FLL2_SYNC_GAIN_SHIFT 2 /* FLL2_SYNC_GAIN */ +#define ARIZONA_FLL2_SYNC_GAIN_WIDTH 4 /* FLL2_SYNC_GAIN */ +#define ARIZONA_FLL2_SYNC_BW_MASK 0x0001 /* FLL2_SYNC_BW */ +#define ARIZONA_FLL2_SYNC_BW_MASK 0x0001 /* FLL2_SYNC_BW */ +#define ARIZONA_FLL2_SYNC_BW_SHIFT 0 /* FLL2_SYNC_BW */ +#define ARIZONA_FLL2_SYNC_BW_WIDTH 1 /* FLL2_SYNC_BW */ + +/* * R425 (0x1A9) - FLL2 Spread Spectrum */ #define ARIZONA_FLL2_SS_AMPL_MASK 0x0030 /* FLL2_SS_AMPL - [5:4] */ diff --git a/include/linux/mfd/wm8994/pdata.h b/include/linux/mfd/wm8994/pdata.h index 8e21a094836d..68e776594889 100644 --- a/include/linux/mfd/wm8994/pdata.h +++ b/include/linux/mfd/wm8994/pdata.h @@ -17,6 +17,7 @@ #define WM8994_NUM_LDO 2 #define WM8994_NUM_GPIO 11 +#define WM8994_NUM_AIF 3 struct wm8994_ldo_pdata { /** GPIOs to enable regulator, 0 or less if not available */ @@ -215,6 +216,13 @@ struct wm8994_pdata { * system. */ bool spkmode_pu; + + /** + * Maximum number of channels clocks will be generated for, + * useful for systems where and I2S bus with multiple data + * lines is mastered. + */ + int max_channels_clocked[WM8994_NUM_AIF]; }; #endif diff --git a/include/linux/usb/audio-v2.h b/include/linux/usb/audio-v2.h index ed13053153f4..c5f2158ab00e 100644 --- a/include/linux/usb/audio-v2.h +++ b/include/linux/usb/audio-v2.h @@ -170,6 +170,8 @@ struct uac2_as_header_descriptor { __u8 iChannelNames; } __attribute__((packed)); +#define UAC2_FORMAT_TYPE_I_RAW_DATA (1 << 31) + /* 4.10.1.2 Class-Specific AS Isochronous Audio Data Endpoint Descriptor */ struct uac2_iso_endpoint_descriptor { 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 diff --git a/include/uapi/sound/asound.h b/include/uapi/sound/asound.h index 1774a5c3ef10..e3983d508272 100644 --- a/include/uapi/sound/asound.h +++ b/include/uapi/sound/asound.h @@ -214,7 +214,9 @@ typedef int __bitwise snd_pcm_format_t; #define SNDRV_PCM_FORMAT_G723_24_1B ((__force snd_pcm_format_t) 45) /* 1 sample in 1 byte */ #define SNDRV_PCM_FORMAT_G723_40 ((__force snd_pcm_format_t) 46) /* 8 Samples in 5 bytes */ #define SNDRV_PCM_FORMAT_G723_40_1B ((__force snd_pcm_format_t) 47) /* 1 sample in 1 byte */ -#define SNDRV_PCM_FORMAT_LAST SNDRV_PCM_FORMAT_G723_40_1B +#define SNDRV_PCM_FORMAT_DSD_U8 ((__force snd_pcm_format_t) 48) /* DSD, 1-byte samples DSD (x8) */ +#define SNDRV_PCM_FORMAT_DSD_U16_LE ((__force snd_pcm_format_t) 49) /* DSD, 2-byte samples DSD (x16), little endian */ +#define SNDRV_PCM_FORMAT_LAST SNDRV_PCM_FORMAT_DSD_U16_LE #ifdef SNDRV_LITTLE_ENDIAN #define SNDRV_PCM_FORMAT_S16 SNDRV_PCM_FORMAT_S16_LE |
