From 76a3aeac2f6c02ecf065fa9baa279dd54bf2d819 Mon Sep 17 00:00:00 2001 From: Mikko Rapeli Date: Tue, 17 Feb 2015 00:05:27 +0100 Subject: hdspm.h: include stdint.h in userspace Fixes compilation error: sound/hdspm.h:43:2: error: unknown type name ‘uint32_t’ Signed-off-by: Mikko Rapeli Signed-off-by: Takashi Iwai --- include/uapi/sound/hdspm.h | 6 ++++++ 1 file changed, 6 insertions(+) (limited to 'include/uapi') diff --git a/include/uapi/sound/hdspm.h b/include/uapi/sound/hdspm.h index b357f1a5e29c..5737332d38f2 100644 --- a/include/uapi/sound/hdspm.h +++ b/include/uapi/sound/hdspm.h @@ -20,6 +20,12 @@ * Foundation, Inc., 675 Mass Ave, Cambridge, MA 02139, USA. */ +#ifdef __KERNEL__ +#include +#else +#include +#endif + /* Maximum channels is 64 even on 56Mode you have 64playbacks to matrix */ #define HDSPM_MAX_CHANNELS 64 -- cgit v1.2.3-55-g7522 From 4bebf7091aa15ec60edf0dcbc654410a87ca21fe Mon Sep 17 00:00:00 2001 From: Mikko Rapeli Date: Tue, 17 Feb 2015 00:05:38 +0100 Subject: include/uapi/sound/asound.h: include stdlib.h in userspace Fixes compiler errors like: error: field ‘trigger_tstamp’ has incomplete type error: invalid application of ‘sizeof’ to incomplete t ype ‘struct timespec’ Signed-off-by: Mikko Rapeli Signed-off-by: Takashi Iwai --- include/uapi/sound/asound.h | 3 +++ 1 file changed, 3 insertions(+) (limited to 'include/uapi') diff --git a/include/uapi/sound/asound.h b/include/uapi/sound/asound.h index 1f23cd635957..1fe3f4f3d696 100644 --- a/include/uapi/sound/asound.h +++ b/include/uapi/sound/asound.h @@ -25,6 +25,9 @@ #include +#ifndef __KERNEL__ +#include +#endif /* * protocol version -- cgit v1.2.3-55-g7522 From bbf91c1c5bfc00c2961f15657359ee7e87de4269 Mon Sep 17 00:00:00 2001 From: Mikko Rapeli Date: Tue, 17 Feb 2015 00:05:42 +0100 Subject: include/uapi/sound/asequencer.h: include sound/asound.h Fixes userspace compilation error: error: unknown type name ‘snd_seq_client_type_t’ snd_seq_client_type_t type; /* client type */ Signed-off-by: Mikko Rapeli Signed-off-by: Takashi Iwai --- include/uapi/sound/asequencer.h | 1 + 1 file changed, 1 insertion(+) (limited to 'include/uapi') diff --git a/include/uapi/sound/asequencer.h b/include/uapi/sound/asequencer.h index 09c8a00ea503..5a5fa4956ebd 100644 --- a/include/uapi/sound/asequencer.h +++ b/include/uapi/sound/asequencer.h @@ -22,6 +22,7 @@ #ifndef _UAPI__SOUND_ASEQUENCER_H #define _UAPI__SOUND_ASEQUENCER_H +#include /** version of the sequencer */ #define SNDRV_SEQ_VERSION SNDRV_PROTOCOL_VERSION (1, 0, 1) -- cgit v1.2.3-55-g7522 From b9956409c281931c74ba8d0a2b61a98076a58602 Mon Sep 17 00:00:00 2001 From: Mikko Rapeli Date: Tue, 17 Feb 2015 00:05:43 +0100 Subject: include/uapi/sound/emu10k1.h: include sound/asound.h Fixes userspace compilation errors like: error: field ‘id’ has incomplete type struct snd_ctl_elem_id id; /* full control ID definition */ Signed-off-by: Mikko Rapeli Signed-off-by: Takashi Iwai --- include/uapi/sound/emu10k1.h | 3 +-- 1 file changed, 1 insertion(+), 2 deletions(-) (limited to 'include/uapi') diff --git a/include/uapi/sound/emu10k1.h b/include/uapi/sound/emu10k1.h index d1bbaf78457a..ec1535bb6aed 100644 --- a/include/uapi/sound/emu10k1.h +++ b/include/uapi/sound/emu10k1.h @@ -23,8 +23,7 @@ #define _UAPI__SOUND_EMU10K1_H #include - - +#include /* * ---- FX8010 ---- -- cgit v1.2.3-55-g7522 From 229d043096ea8e58829d37d35767afeac15997f5 Mon Sep 17 00:00:00 2001 From: Pierre-Louis Bossart Date: Fri, 13 Feb 2015 15:14:03 -0600 Subject: ALSA: core: selection of audio_tstamp type and accuracy reports Audio timestamps can be extracted from sample counters, wall clocks, PHC clocks (Ethernet AVB), on-demand synchronized snapshots. This patch provides the ability to report timestamping capabilities, select timestamp types and retrieve timestamp accuracy, if supported. Details can be found in Documentations/sound/alsa/timestamping.txt This functionality is introduced by reclaiming the reserved_aligned field introduced by commit9c7066aef4a5eb8e4063de28f06c508bf6f2963a in snd_pcm_status to provide userspace with selection/query capabilities. Additional driver_tstamp and audio_tstamp_accuracy fields are also added. snd_pcm_mmap_status remains a read-only structure with only the audio timestamp value accessible from user space. The selection of audio timestamp type is done through snd_pcm_status only This commit does not impact ABI and does not impact the default behavior. By default audio timestamp is aligned with hw_pointer and reports the DMA position. Backwards compatibility is handled by using the HDAudio wall clock for playback and the hw_ptr for all other cases. For timestamp selection a new STATUS_EXT ioctl is introduced with read/write parameters. Alsa-lib will be modified to make use of STATUS_EXT. Signed-off-by: Pierre-Louis Bossart Signed-off-by: Takashi Iwai --- Documentation/sound/alsa/timestamping.txt | 200 ++++++++++++++++++++++++++++++ include/sound/pcm.h | 60 +++++++++ include/uapi/sound/asound.h | 34 ++++- 3 files changed, 290 insertions(+), 4 deletions(-) create mode 100644 Documentation/sound/alsa/timestamping.txt (limited to 'include/uapi') diff --git a/Documentation/sound/alsa/timestamping.txt b/Documentation/sound/alsa/timestamping.txt new file mode 100644 index 000000000000..0b191a23f534 --- /dev/null +++ b/Documentation/sound/alsa/timestamping.txt @@ -0,0 +1,200 @@ +The ALSA API can provide two different system timestamps: + +- Trigger_tstamp is the system time snapshot taken when the .trigger +callback is invoked. This snapshot is taken by the ALSA core in the +general case, but specific hardware may have synchronization +capabilities or conversely may only be able to provide a correct +estimate with a delay. In the latter two cases, the low-level driver +is responsible for updating the trigger_tstamp at the most appropriate +and precise moment. Applications should not rely solely on the first +trigger_tstamp but update their internal calculations if the driver +provides a refined estimate with a delay. + +- tstamp is the current system timestamp updated during the last +event or application query. +The difference (tstamp - trigger_tstamp) defines the elapsed time. + +The ALSA API provides reports two basic pieces of information, avail +and delay, which combined with the trigger and current system +timestamps allow for applications to keep track of the 'fullness' of +the ring buffer and the amount of queued samples. + +The use of these different pointers and time information depends on +the application needs: + +- 'avail' reports how much can be written in the ring buffer +- 'delay' reports the time it will take to hear a new sample after all +queued samples have been played out. + +When timestamps are enabled, the avail/delay information is reported +along with a snapshot of system time. Applications can select from +CLOCK_REALTIME (NTP corrections including going backwards), +CLOCK_MONOTONIC (NTP corrections but never going backwards), +CLOCK_MONOTIC_RAW (without NTP corrections) and change the mode +dynamically with sw_params + + +The ALSA API also provide an audio_tstamp which reflects the passage +of time as measured by different components of audio hardware. In +ascii-art, this could be represented as follows (for the playback +case): + + +--------------------------------------------------------------> time + ^ ^ ^ ^ ^ + | | | | | + analog link dma app FullBuffer + time time time time time + | | | | | + |< codec delay >|<--hw delay-->||<---avail->| + |<----------------- delay---------------------->| | + |<----ring buffer length---->| + +The analog time is taken at the last stage of the playback, as close +as possible to the actual transducer + +The link time is taken at the output of the SOC/chipset as the samples +are pushed on a link. The link time can be directly measured if +supported in hardware by sample counters or wallclocks (e.g. with +HDAudio 24MHz or PTP clock for networked solutions) or indirectly +estimated (e.g. with the frame counter in USB). + +The DMA time is measured using counters - typically the least reliable +of all measurements due to the bursty natured of DMA transfers. + +The app time corresponds to the time tracked by an application after +writing in the ring buffer. + +The application can query what the hardware supports, define which +audio time it wants reported by selecting the relevant settings in +audio_tstamp_config fields, get an estimate of the timestamp +accuracy. It can also request the delay-to-analog be included in the +measurement. Direct access to the link time is very interesting on +platforms that provide an embedded DSP; measuring directly the link +time with dedicated hardware, possibly synchronized with system time, +removes the need to keep track of internal DSP processing times and +latency. + +In case the application requests an audio tstamp that is not supported +in hardware/low-level driver, the type is overridden as DEFAULT and the +timestamp will report the DMA time based on the hw_pointer value. + +For backwards compatibility with previous implementations that did not +provide timestamp selection, with a zero-valued COMPAT timestamp type +the results will default to the HDAudio wall clock for playback +streams and to the DMA time (hw_ptr) in all other cases. + +The audio timestamp accuracy can be returned to user-space, so that +appropriate decisions are made: + +- for dma time (default), the granularity of the transfers can be + inferred from the steps between updates and in turn provide + information on how much the application pointer can be rewound + safely. + +- the link time can be used to track long-term drifts between audio + and system time using the (tstamp-trigger_tstamp)/audio_tstamp + ratio, the precision helps define how much smoothing/low-pass + filtering is required. The link time can be either reset on startup + or reported as is (the latter being useful to compare progress of + different streams - but may require the wallclock to be always + running and not wrap-around during idle periods). If supported in + hardware, the absolute link time could also be used to define a + precise start time (patches WIP) + +- including the delay in the audio timestamp may + counter-intuitively not increase the precision of timestamps, e.g. if a + codec includes variable-latency DSP processing or a chain of + hardware components the delay is typically not known with precision. + +The accuracy is reported in nanosecond units (using an unsigned 32-bit +word), which gives a max precision of 4.29s, more than enough for +audio applications... + +Due to the varied nature of timestamping needs, even for a single +application, the audio_tstamp_config can be changed dynamically. In +the STATUS ioctl, the parameters are read-only and do not allow for +any application selection. To work around this limitation without +impacting legacy applications, a new STATUS_EXT ioctl is introduced +with read/write parameters. ALSA-lib will be modified to make use of +STATUS_EXT and effectively deprecate STATUS. + +The ALSA API only allows for a single audio timestamp to be reported +at a time. This is a conscious design decision, reading the audio +timestamps from hardware registers or from IPC takes time, the more +timestamps are read the more imprecise the combined measurements +are. To avoid any interpretation issues, a single (system, audio) +timestamp is reported. Applications that need different timestamps +will be required to issue multiple queries and perform an +interpolation of the results + +In some hardware-specific configuration, the system timestamp is +latched by a low-level audio subsytem, and the information provided +back to the driver. Due to potential delays in the communication with +the hardware, there is a risk of misalignment with the avail and delay +information. To make sure applications are not confused, a +driver_timestamp field is added in the snd_pcm_status structure; this +timestamp shows when the information is put together by the driver +before returning from the STATUS and STATUS_EXT ioctl. in most cases +this driver_timestamp will be identical to the regular system tstamp. + +Examples of typestamping with HDaudio: + +1. DMA timestamp, no compensation for DMA+analog delay +$ ./audio_time -p --ts_type=1 +playback: systime: 341121338 nsec, audio time 342000000 nsec, systime delta -878662 +playback: systime: 426236663 nsec, audio time 427187500 nsec, systime delta -950837 +playback: systime: 597080580 nsec, audio time 598000000 nsec, systime delta -919420 +playback: systime: 682059782 nsec, audio time 683020833 nsec, systime delta -961051 +playback: systime: 852896415 nsec, audio time 853854166 nsec, systime delta -957751 +playback: systime: 937903344 nsec, audio time 938854166 nsec, systime delta -950822 + +2. DMA timestamp, compensation for DMA+analog delay +$ ./audio_time -p --ts_type=1 -d +playback: systime: 341053347 nsec, audio time 341062500 nsec, systime delta -9153 +playback: systime: 426072447 nsec, audio time 426062500 nsec, systime delta 9947 +playback: systime: 596899518 nsec, audio time 596895833 nsec, systime delta 3685 +playback: systime: 681915317 nsec, audio time 681916666 nsec, systime delta -1349 +playback: systime: 852741306 nsec, audio time 852750000 nsec, systime delta -8694 + +3. link timestamp, compensation for DMA+analog delay +$ ./audio_time -p --ts_type=2 -d +playback: systime: 341060004 nsec, audio time 341062791 nsec, systime delta -2787 +playback: systime: 426242074 nsec, audio time 426244875 nsec, systime delta -2801 +playback: systime: 597080992 nsec, audio time 597084583 nsec, systime delta -3591 +playback: systime: 682084512 nsec, audio time 682088291 nsec, systime delta -3779 +playback: systime: 852936229 nsec, audio time 852940916 nsec, systime delta -4687 +playback: systime: 938107562 nsec, audio time 938112708 nsec, systime delta -5146 + +Example 1 shows that the timestamp at the DMA level is close to 1ms +ahead of the actual playback time (as a side time this sort of +measurement can help define rewind safeguards). Compensating for the +DMA-link delay in example 2 helps remove the hardware buffering abut +the information is still very jittery, with up to one sample of +error. In example 3 where the timestamps are measured with the link +wallclock, the timestamps show a monotonic behavior and a lower +dispersion. + +Example 3 and 4 are with USB audio class. Example 3 shows a high +offset between audio time and system time due to buffering. Example 4 +shows how compensating for the delay exposes a 1ms accuracy (due to +the use of the frame counter by the driver) + +Example 3: DMA timestamp, no compensation for delay, delta of ~5ms +$ ./audio_time -p -Dhw:1 -t1 +playback: systime: 120174019 nsec, audio time 125000000 nsec, systime delta -4825981 +playback: systime: 245041136 nsec, audio time 250000000 nsec, systime delta -4958864 +playback: systime: 370106088 nsec, audio time 375000000 nsec, systime delta -4893912 +playback: systime: 495040065 nsec, audio time 500000000 nsec, systime delta -4959935 +playback: systime: 620038179 nsec, audio time 625000000 nsec, systime delta -4961821 +playback: systime: 745087741 nsec, audio time 750000000 nsec, systime delta -4912259 +playback: systime: 870037336 nsec, audio time 875000000 nsec, systime delta -4962664 + +Example 4: DMA timestamp, compensation for delay, delay of ~1ms +$ ./audio_time -p -Dhw:1 -t1 -d +playback: systime: 120190520 nsec, audio time 120000000 nsec, systime delta 190520 +playback: systime: 245036740 nsec, audio time 244000000 nsec, systime delta 1036740 +playback: systime: 370034081 nsec, audio time 369000000 nsec, systime delta 1034081 +playback: systime: 495159907 nsec, audio time 494000000 nsec, systime delta 1159907 +playback: systime: 620098824 nsec, audio time 619000000 nsec, systime delta 1098824 +playback: systime: 745031847 nsec, audio time 744000000 nsec, systime delta 1031847 diff --git a/include/sound/pcm.h b/include/sound/pcm.h index c0ddb7e69c28..60f0e48f7905 100644 --- a/include/sound/pcm.h +++ b/include/sound/pcm.h @@ -60,6 +60,9 @@ struct snd_pcm_hardware { struct snd_pcm_substream; +struct snd_pcm_audio_tstamp_config; /* definitions further down */ +struct snd_pcm_audio_tstamp_report; + struct snd_pcm_ops { int (*open)(struct snd_pcm_substream *substream); int (*close)(struct snd_pcm_substream *substream); @@ -281,6 +284,58 @@ struct snd_pcm_hw_constraint_ranges { struct snd_pcm_hwptr_log; +/* + * userspace-provided audio timestamp config to kernel, + * structure is for internal use only and filled with dedicated unpack routine + */ +struct snd_pcm_audio_tstamp_config { + /* 5 of max 16 bits used */ + u32 type_requested:4; + u32 report_delay:1; /* add total delay to A/D or D/A */ +}; + +static inline void snd_pcm_unpack_audio_tstamp_config(__u32 data, + struct snd_pcm_audio_tstamp_config *config) +{ + config->type_requested = data & 0xF; + config->report_delay = (data >> 4) & 1; +} + +/* + * kernel-provided audio timestamp report to user-space + * structure is for internal use only and read by dedicated pack routine + */ +struct snd_pcm_audio_tstamp_report { + /* 6 of max 16 bits used for bit-fields */ + + /* for backwards compatibility */ + u32 valid:1; + + /* actual type if hardware could not support requested timestamp */ + u32 actual_type:4; + + /* accuracy represented in ns units */ + u32 accuracy_report:1; /* 0 if accuracy unknown, 1 if accuracy field is valid */ + u32 accuracy; /* up to 4.29s, will be packed in separate field */ +}; + +static inline void snd_pcm_pack_audio_tstamp_report(__u32 *data, __u32 *accuracy, + const struct snd_pcm_audio_tstamp_report *report) +{ + u32 tmp; + + tmp = report->accuracy_report; + tmp <<= 4; + tmp |= report->actual_type; + tmp <<= 1; + tmp |= report->valid; + + *data &= 0xffff; /* zero-clear MSBs */ + *data |= (tmp << 16); + *accuracy = report->accuracy; +} + + struct snd_pcm_runtime { /* -- Status -- */ struct snd_pcm_substream *trigger_master; @@ -361,6 +416,11 @@ struct snd_pcm_runtime { struct snd_dma_buffer *dma_buffer_p; /* allocated buffer */ + /* -- audio timestamp config -- */ + struct snd_pcm_audio_tstamp_config audio_tstamp_config; + struct snd_pcm_audio_tstamp_report audio_tstamp_report; + struct timespec driver_tstamp; + #if defined(CONFIG_SND_PCM_OSS) || defined(CONFIG_SND_PCM_OSS_MODULE) /* -- OSS things -- */ struct snd_pcm_oss_runtime oss; diff --git a/include/uapi/sound/asound.h b/include/uapi/sound/asound.h index 0e88e7a0f0eb..acef4e4d2735 100644 --- a/include/uapi/sound/asound.h +++ b/include/uapi/sound/asound.h @@ -267,10 +267,17 @@ typedef int __bitwise snd_pcm_subformat_t; #define SNDRV_PCM_INFO_JOINT_DUPLEX 0x00200000 /* playback and capture stream are somewhat correlated */ #define SNDRV_PCM_INFO_SYNC_START 0x00400000 /* pcm support some kind of sync go */ #define SNDRV_PCM_INFO_NO_PERIOD_WAKEUP 0x00800000 /* period wakeup can be disabled */ -#define SNDRV_PCM_INFO_HAS_WALL_CLOCK 0x01000000 /* has audio wall clock for audio/system time sync */ +#define SNDRV_PCM_INFO_HAS_WALL_CLOCK 0x01000000 /* (Deprecated)has audio wall clock for audio/system time sync */ +#define SNDRV_PCM_INFO_HAS_LINK_ATIME 0x01000000 /* report hardware link audio time, reset on startup */ +#define SNDRV_PCM_INFO_HAS_LINK_ABSOLUTE_ATIME 0x02000000 /* report absolute hardware link audio time, not reset on startup */ +#define SNDRV_PCM_INFO_HAS_LINK_ESTIMATED_ATIME 0x04000000 /* report estimated link audio time */ +#define SNDRV_PCM_INFO_HAS_LINK_SYNCHRONIZED_ATIME 0x08000000 /* report synchronized audio/system time */ + #define SNDRV_PCM_INFO_DRAIN_TRIGGER 0x40000000 /* internal kernel flag - trigger in drain */ #define SNDRV_PCM_INFO_FIFO_IN_FRAMES 0x80000000 /* internal kernel flag - FIFO size is in frames */ + + typedef int __bitwise snd_pcm_state_t; #define SNDRV_PCM_STATE_OPEN ((__force snd_pcm_state_t) 0) /* stream is open */ #define SNDRV_PCM_STATE_SETUP ((__force snd_pcm_state_t) 1) /* stream has a setup */ @@ -408,6 +415,22 @@ struct snd_pcm_channel_info { unsigned int step; /* samples distance in bits */ }; +enum { + /* + * first definition for backwards compatibility only, + * maps to wallclock/link time for HDAudio playback and DEFAULT/DMA time for everything else + */ + SNDRV_PCM_AUDIO_TSTAMP_TYPE_COMPAT = 0, + + /* timestamp definitions */ + SNDRV_PCM_AUDIO_TSTAMP_TYPE_DEFAULT = 1, /* DMA time, reported as per hw_ptr */ + SNDRV_PCM_AUDIO_TSTAMP_TYPE_LINK = 2, /* link time reported by sample or wallclock counter, reset on startup */ + SNDRV_PCM_AUDIO_TSTAMP_TYPE_LINK_ABSOLUTE = 3, /* link time reported by sample or wallclock counter, not reset on startup */ + SNDRV_PCM_AUDIO_TSTAMP_TYPE_LINK_ESTIMATED = 4, /* link time estimated indirectly */ + SNDRV_PCM_AUDIO_TSTAMP_TYPE_LINK_SYNCHRONIZED = 5, /* link time synchronized with system time */ + SNDRV_PCM_AUDIO_TSTAMP_TYPE_LAST = SNDRV_PCM_AUDIO_TSTAMP_TYPE_LINK_SYNCHRONIZED +}; + struct snd_pcm_status { snd_pcm_state_t state; /* stream state */ struct timespec trigger_tstamp; /* time when stream was started/stopped/paused */ @@ -419,9 +442,11 @@ struct snd_pcm_status { snd_pcm_uframes_t avail_max; /* max frames available on hw since last status */ snd_pcm_uframes_t overrange; /* count of ADC (capture) overrange detections from last status */ snd_pcm_state_t suspended_state; /* suspended stream state */ - __u32 reserved_alignment; /* must be filled with zero */ - struct timespec audio_tstamp; /* from sample counter or wall clock */ - unsigned char reserved[56-sizeof(struct timespec)]; /* must be filled with zero */ + __u32 audio_tstamp_data; /* needed for 64-bit alignment, used for configs/report to/from userspace */ + struct timespec audio_tstamp; /* sample counter, wall clock, PHC or on-demand sync'ed */ + struct timespec driver_tstamp; /* useful in case reference system tstamp is reported with delay */ + __u32 audio_tstamp_accuracy; /* in ns units, only valid if indicated in audio_tstamp_data */ + unsigned char reserved[52-2*sizeof(struct timespec)]; /* must be filled with zero */ }; struct snd_pcm_mmap_status { @@ -534,6 +559,7 @@ enum { #define SNDRV_PCM_IOCTL_DELAY _IOR('A', 0x21, snd_pcm_sframes_t) #define SNDRV_PCM_IOCTL_HWSYNC _IO('A', 0x22) #define SNDRV_PCM_IOCTL_SYNC_PTR _IOWR('A', 0x23, struct snd_pcm_sync_ptr) +#define SNDRV_PCM_IOCTL_STATUS_EXT _IOWR('A', 0x24, struct snd_pcm_status) #define SNDRV_PCM_IOCTL_CHANNEL_INFO _IOR('A', 0x32, struct snd_pcm_channel_info) #define SNDRV_PCM_IOCTL_PREPARE _IO('A', 0x40) #define SNDRV_PCM_IOCTL_RESET _IO('A', 0x41) -- cgit v1.2.3-55-g7522 From c72638bdaabe9ea4b09003b9db7e1754f472fbed Mon Sep 17 00:00:00 2001 From: Pierre-Louis Bossart Date: Fri, 13 Feb 2015 15:14:09 -0600 Subject: ALSA: bump PCM protocol to 2.0.13 Bump PCM protocol to enable use of STATUS_EXT ioctls, older apps will still use STATUS and audio timestamp configuration is not supported (backwards compatible behavior). Signed-off-by: Pierre-Louis Bossart Signed-off-by: Takashi Iwai --- include/uapi/sound/asound.h | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'include/uapi') diff --git a/include/uapi/sound/asound.h b/include/uapi/sound/asound.h index acef4e4d2735..3d46e9a0cd2e 100644 --- a/include/uapi/sound/asound.h +++ b/include/uapi/sound/asound.h @@ -140,7 +140,7 @@ struct snd_hwdep_dsp_image { * * *****************************************************************************/ -#define SNDRV_PCM_VERSION SNDRV_PROTOCOL_VERSION(2, 0, 12) +#define SNDRV_PCM_VERSION SNDRV_PROTOCOL_VERSION(2, 0, 13) typedef unsigned long snd_pcm_uframes_t; typedef signed long snd_pcm_sframes_t; -- cgit v1.2.3-55-g7522 From 1a6ab46fa9c2bc9399694b4856ab7ea19c036485 Mon Sep 17 00:00:00 2001 From: Masanari Iida Date: Wed, 4 Mar 2015 10:56:13 +0900 Subject: ALSA: Fix spelling typo in Documentation/DocBook/alsa-driver-api.xml This patch fix spelling typo found in alsa-driver-api.xml. It is because this file is generated from comments in source files, I have to fix source files. Signed-off-by: Masanari Iida Signed-off-by: Takashi Iwai --- include/sound/compress_driver.h | 4 ++-- include/sound/control.h | 2 +- include/sound/soc.h | 2 +- include/uapi/sound/compress_offload.h | 2 +- sound/core/pcm_dmaengine.c | 4 ++-- 5 files changed, 7 insertions(+), 7 deletions(-) (limited to 'include/uapi') diff --git a/include/sound/compress_driver.h b/include/sound/compress_driver.h index f48089d364c5..fa1d05512c09 100644 --- a/include/sound/compress_driver.h +++ b/include/sound/compress_driver.h @@ -70,7 +70,7 @@ struct snd_compr_runtime { * @device: device pointer * @direction: stream direction, playback/recording * @metadata_set: metadata set flag, true when set - * @next_track: has userspace signall next track transistion, true when set + * @next_track: has userspace signal next track transition, true when set * @private_data: pointer to DSP private data */ struct snd_compr_stream { @@ -95,7 +95,7 @@ struct snd_compr_stream { * and the stream properties * @get_params: retrieve the codec parameters, mandatory * @set_metadata: Set the metadata values for a stream - * @get_metadata: retreives the requested metadata values from stream + * @get_metadata: retrieves the requested metadata values from stream * @trigger: Trigger operations like start, pause, resume, drain, stop. * This callback is mandatory * @pointer: Retrieve current h/w pointer information. Mandatory diff --git a/include/sound/control.h b/include/sound/control.h index 75f3054023f7..95aad6d3fd1a 100644 --- a/include/sound/control.h +++ b/include/sound/control.h @@ -227,7 +227,7 @@ snd_ctl_add_slave(struct snd_kcontrol *master, struct snd_kcontrol *slave) * Add a virtual slave control to the given master. * Unlike snd_ctl_add_slave(), the element added via this function * is supposed to have volatile values, and get callback is called - * at each time quried from the master. + * at each time queried from the master. * * When the control peeks the hardware values directly and the value * can be changed by other means than the put callback of the element, diff --git a/include/sound/soc.h b/include/sound/soc.h index 0d1ade195628..cf0bb156d6da 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -1469,7 +1469,7 @@ static inline struct snd_soc_codec *snd_soc_kcontrol_codec( } /** - * snd_soc_kcontrol_platform() - Returns the platform that registerd the control + * snd_soc_kcontrol_platform() - Returns the platform that registered the control * @kcontrol: The control for which to get the platform * * Note: This function will only work correctly if the control has been diff --git a/include/uapi/sound/compress_offload.h b/include/uapi/sound/compress_offload.h index 22ed8cb7800b..e00d8cbfc628 100644 --- a/include/uapi/sound/compress_offload.h +++ b/include/uapi/sound/compress_offload.h @@ -75,7 +75,7 @@ struct snd_compr_tstamp { /** * struct snd_compr_avail - avail descriptor * @avail: Number of bytes available in ring buffer for writing/reading - * @tstamp: timestamp infomation + * @tstamp: timestamp information */ struct snd_compr_avail { __u64 avail; diff --git a/sound/core/pcm_dmaengine.c b/sound/core/pcm_dmaengine.c index 6542c4083594..fba365a78390 100644 --- a/sound/core/pcm_dmaengine.c +++ b/sound/core/pcm_dmaengine.c @@ -289,7 +289,7 @@ EXPORT_SYMBOL_GPL(snd_dmaengine_pcm_request_channel); * * The function should usually be called from the pcm open callback. Note that * this function will use private_data field of the substream's runtime. So it - * is not availabe to your pcm driver implementation. + * is not available to your pcm driver implementation. */ int snd_dmaengine_pcm_open(struct snd_pcm_substream *substream, struct dma_chan *chan) @@ -328,7 +328,7 @@ EXPORT_SYMBOL_GPL(snd_dmaengine_pcm_open); * This function will request a DMA channel using the passed filter function and * data. The function should usually be called from the pcm open callback. Note * that this function will use private_data field of the substream's runtime. So - * it is not availabe to your pcm driver implementation. + * it is not available to your pcm driver implementation. */ int snd_dmaengine_pcm_open_request_chan(struct snd_pcm_substream *substream, dma_filter_fn filter_fn, void *filter_data) -- cgit v1.2.3-55-g7522