timestamp in RUNNING mode is already taken in update_hw_ptr routine, getting a new timestamp introduces offset between hw_ptr, audio_tstamp and system time

Add else condition to read timestamp as fallback and only when enabled

Signed-off-by: Pierre-Louis Bossart pierre-louis.bossart@linux.intel.com --- sound/core/pcm_native.c | 5 ++++- 1 file changed, 4 insertions(+), 1 deletion(-)

diff --git a/sound/core/pcm_native.c b/sound/core/pcm_native.c index ff3abc3..7bbc34d 100644 --- a/sound/core/pcm_native.c +++ b/sound/core/pcm_native.c @@ -720,8 +720,11 @@ int snd_pcm_status(struct snd_pcm_substream *substream, runtime->status->audio_tstamp; goto _tstamp_end; } + } else { + /* get tstamp only in fallback mode and only if enabled */ + if (runtime->tstamp_mode == SNDRV_PCM_TSTAMP_ENABLE) + snd_pcm_gettime(runtime, &status->tstamp); } - snd_pcm_gettime(runtime, &status->tstamp); _tstamp_end: status->appl_ptr = runtime->control->appl_ptr; status->hw_ptr = runtime->status->hw_ptr;

Don't use generic snapshot of trigger_tstamp if low-level driver or hardware can get a more precise value for better audio/system time synchronization.

Signed-off-by: Pierre-Louis Bossart pierre-louis.bossart@linux.intel.com --- include/sound/pcm.h | 1 + sound/core/pcm_native.c | 4 +++- 2 files changed, 4 insertions(+), 1 deletion(-)

diff --git a/include/sound/pcm.h b/include/sound/pcm.h index cd09c1b..e158036 100644 --- a/include/sound/pcm.h +++ b/include/sound/pcm.h @@ -279,6 +279,7 @@ struct snd_pcm_runtime { /* -- Status -- */ struct snd_pcm_substream *trigger_master; struct timespec trigger_tstamp; /* trigger timestamp */ + bool trigger_tstamp_latched; /* trigger timestamp latched in low-level driver/hardware */ int overrange; snd_pcm_uframes_t avail_max; snd_pcm_uframes_t hw_ptr_base; /* Position at buffer restart */ diff --git a/sound/core/pcm_native.c b/sound/core/pcm_native.c index 7bbc34d..d50f3e6 100644 --- a/sound/core/pcm_native.c +++ b/sound/core/pcm_native.c @@ -810,7 +810,8 @@ static void snd_pcm_trigger_tstamp(struct snd_pcm_substream *substream) if (runtime->trigger_master == NULL) return; if (runtime->trigger_master == substream) { - snd_pcm_gettime(runtime, &runtime->trigger_tstamp); + if (!runtime->trigger_tstamp_latched) + snd_pcm_gettime(runtime, &runtime->trigger_tstamp); } else { snd_pcm_trigger_tstamp(runtime->trigger_master); runtime->trigger_tstamp = runtime->trigger_master->runtime->trigger_tstamp; @@ -979,6 +980,7 @@ static int snd_pcm_pre_start(struct snd_pcm_substream *substream, int state) if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK && !snd_pcm_playback_data(substream)) return -EPIPE; + runtime->trigger_tstamp_latched = false; runtime->trigger_master = substream; return 0; }

Make sure wallclock counter and trigger timestamp are read very close to each other for better alignment.

Signed-off-by: Pierre-Louis Bossart pierre-louis.bossart@linux.intel.com --- sound/pci/hda/hda_controller.c | 3 +++ 1 file changed, 3 insertions(+)

diff --git a/sound/pci/hda/hda_controller.c b/sound/pci/hda/hda_controller.c index 0cfc9c8..234208a 100644 --- a/sound/pci/hda/hda_controller.c +++ b/sound/pci/hda/hda_controller.c @@ -657,6 +657,9 @@ static int azx_pcm_trigger(struct snd_pcm_substream *substream, int cmd) azx_writel(chip, SSYNC, azx_readl(chip, SSYNC) & ~sbits); if (start) { azx_timecounter_init(substream, 0, 0); + snd_pcm_gettime(substream->runtime, &substream->runtime->trigger_tstamp); + substream->runtime->trigger_tstamp_latched = true; + if (nsync > 1) { cycle_t cycle_last;

The first URBs are submitted during the prepare stage. When .trigger is called, the ALSA core saves a trigger tstamp that doesn't correspond to the actual time when the samples are submitted. The trigger_tstamp is now updated when the first data are submitted to avoid any time offsets.

A usb-specific trigger_tstamp_pending_update flag is used for now, at some point the flag would need to move to the ALSA core, USB is not the only interface where silent block transfers are programmed as part of the prepare stage, with actual data enabled when .trigger is called.

Signed-off-by: Pierre-Louis Bossart pierre-louis.bossart@linux.intel.com --- sound/usb/card.h | 2 ++ sound/usb/pcm.c | 9 +++++++++ 2 files changed, 11 insertions(+)

diff --git a/sound/usb/card.h b/sound/usb/card.h index 97acb90..ef580b4 100644 --- a/sound/usb/card.h +++ b/sound/usb/card.h @@ -153,6 +153,8 @@ struct snd_usb_substream { int channel; int byte_idx; } dsd_dop; + + bool trigger_tstamp_pending_update; /* trigger timestamp being updated from initial estimate */ };

struct snd_usb_stream { diff --git a/sound/usb/pcm.c b/sound/usb/pcm.c index 0d8aba5..b4ef410 100644 --- a/sound/usb/pcm.c +++ b/sound/usb/pcm.c @@ -1464,6 +1464,14 @@ static void prepare_playback_urb(struct snd_usb_substream *subs, subs->last_frame_number = usb_get_current_frame_number(subs->dev); subs->last_frame_number &= 0xFF; /* keep 8 LSBs */

+ if (subs->trigger_tstamp_pending_update) { + /* this is the first actual URB submitted, + * update trigger timestamp to reflect actual start time + */ + snd_pcm_gettime(runtime, &runtime->trigger_tstamp); + subs->trigger_tstamp_pending_update = false; + } + spin_unlock_irqrestore(&subs->lock, flags); urb->transfer_buffer_length = bytes; if (period_elapsed) @@ -1550,6 +1558,7 @@ static int snd_usb_substream_playback_trigger(struct snd_pcm_substream *substrea

switch (cmd) { case SNDRV_PCM_TRIGGER_START: + subs->trigger_tstamp_pending_update = true; case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: subs->data_endpoint->prepare_data_urb = prepare_playback_urb; subs->data_endpoint->retire_data_urb = retire_playback_urb;

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 pierre-louis.bossart@linux.intel.com --- Documentation/sound/alsa/timestamping.txt | 200 ++++++++++++++++++++++++++++++ include/sound/pcm.h | 59 +++++++++ include/uapi/sound/asound.h | 34 ++++- 3 files changed, 289 insertions(+), 4 deletions(-) create mode 100644 Documentation/sound/alsa/timestamping.txt

diff --git a/Documentation/sound/alsa/timestamping.txt b/Documentation/sound/alsa/timestamping.txt new file mode 100644 index 0000000..0b191a2 --- /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-->|<queued samples>|<---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 e158036..05ad56a 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); @@ -275,6 +278,57 @@ struct snd_pcm_hw_constraint_list {

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, + 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 |= (tmp << 16); + *accuracy = report->accuracy; +} + + struct snd_pcm_runtime { /* -- Status -- */ struct snd_pcm_substream *trigger_master; @@ -355,6 +409,11 @@ 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 0e88e7a..acef4e4 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)

Let userspace select audio timestamp config when the STATUS_EXT ioctl is used, ignore and zero all other fields No change for the existing STATUS ioctl, parameters are treated as read-only.

Signed-off-by: Pierre-Louis Bossart pierre-louis.bossart@linux.intel.com --- sound/core/pcm_native.c | 25 +++++++++++++++++++++---- 1 file changed, 21 insertions(+), 4 deletions(-)

diff --git a/sound/core/pcm_native.c b/sound/core/pcm_native.c index d50f3e6..40d2943 100644 --- a/sound/core/pcm_native.c +++ b/sound/core/pcm_native.c @@ -753,12 +753,27 @@ int snd_pcm_status(struct snd_pcm_substream *substream, }

static int snd_pcm_status_user(struct snd_pcm_substream *substream, - struct snd_pcm_status __user * _status) + struct snd_pcm_status __user * _status, + bool ext) { struct snd_pcm_status status; int res; - - memset(&status, 0, sizeof(status)); + u32 audio_tstamp_data; + u32 __user *_audio_tstamp_data; + + if (ext == false) { + memset(&status, 0, sizeof(status)); + } else { + /* + * parameters are read/write, get audio_tstamp_data from user, + * ignore rest of status structure + */ + _audio_tstamp_data = (u32 __user *)(&_status->audio_tstamp_data); + if (get_user(audio_tstamp_data, _audio_tstamp_data)) + return -EFAULT; + memset(&status, 0, sizeof(status)); + status.audio_tstamp_data = audio_tstamp_data; + } res = snd_pcm_status(substream, &status); if (res < 0) return res; @@ -2723,7 +2738,9 @@ static int snd_pcm_common_ioctl1(struct file *file, case SNDRV_PCM_IOCTL_SW_PARAMS: return snd_pcm_sw_params_user(substream, arg); case SNDRV_PCM_IOCTL_STATUS: - return snd_pcm_status_user(substream, arg); + return snd_pcm_status_user(substream, arg, false); + case SNDRV_PCM_IOCTL_STATUS_EXT: + return snd_pcm_status_user(substream, arg, true); case SNDRV_PCM_IOCTL_CHANNEL_INFO: return snd_pcm_channel_info_user(substream, arg); case SNDRV_PCM_IOCTL_PREPARE:

Let userspace select audio timestamp config, ignore and zero all other fields Use audio_tstamp_data to retrieve config and pass report back to user space

Signed-off-by: Pierre-Louis Bossart pierre-louis.bossart@linux.intel.com --- sound/core/pcm_compat.c | 36 ++++++++++++++++++++++++++++++------ 1 file changed, 30 insertions(+), 6 deletions(-)

diff --git a/sound/core/pcm_compat.c b/sound/core/pcm_compat.c index 2d957ba..188b991 100644 --- a/sound/core/pcm_compat.c +++ b/sound/core/pcm_compat.c @@ -194,18 +194,36 @@ struct snd_pcm_status32 { u32 avail_max; u32 overrange; s32 suspended_state; - u32 reserved_alignment; + u32 audio_tstamp_data; struct compat_timespec audio_tstamp; - unsigned char reserved[56-sizeof(struct compat_timespec)]; + struct compat_timespec driver_tstamp; + u32 audio_tstamp_accuracy; + unsigned char reserved[52-2*sizeof(struct compat_timespec)]; } __attribute__((packed));

static int snd_pcm_status_user_compat(struct snd_pcm_substream *substream, - struct snd_pcm_status32 __user *src) + struct snd_pcm_status32 __user *src, + bool ext) { struct snd_pcm_status status; int err; - + u32 audio_tstamp_data; + u32 __user *_audio_tstamp_data; + + if (ext == false) { + memset(&status, 0, sizeof(status)); + } else { + /* + * parameters are read/write, get audio_tstamp_data from user, + * ignore rest of status structure + */ + _audio_tstamp_data = (u32 __user *)(&src->audio_tstamp_data); + if (get_user(audio_tstamp_data, _audio_tstamp_data)) + return -EFAULT; + memset(&status, 0, sizeof(status)); + status.audio_tstamp_data = audio_tstamp_data; + } err = snd_pcm_status(substream, &status); if (err < 0) return err; @@ -222,7 +240,10 @@ static int snd_pcm_status_user_compat(struct snd_pcm_substream *substream, put_user(status.avail_max, &src->avail_max) || put_user(status.overrange, &src->overrange) || put_user(status.suspended_state, &src->suspended_state) || - compat_put_timespec(&status.audio_tstamp, &src->audio_tstamp)) + put_user(status.audio_tstamp_data, &src->audio_tstamp_data) || + compat_put_timespec(&status.audio_tstamp, &src->audio_tstamp) || + compat_put_timespec(&status.driver_tstamp, &src->driver_tstamp) || + put_user(status.audio_tstamp_accuracy, &src->audio_tstamp_accuracy)) return -EFAULT;

return err; @@ -457,6 +478,7 @@ enum { SNDRV_PCM_IOCTL_HW_PARAMS32 = _IOWR('A', 0x11, struct snd_pcm_hw_params32), SNDRV_PCM_IOCTL_SW_PARAMS32 = _IOWR('A', 0x13, struct snd_pcm_sw_params32), SNDRV_PCM_IOCTL_STATUS32 = _IOR('A', 0x20, struct snd_pcm_status32), + SNDRV_PCM_IOCTL_STATUS_EXT32 = _IOWR('A', 0x24, struct snd_pcm_status32), SNDRV_PCM_IOCTL_DELAY32 = _IOR('A', 0x21, s32), SNDRV_PCM_IOCTL_CHANNEL_INFO32 = _IOR('A', 0x32, struct snd_pcm_channel_info32), SNDRV_PCM_IOCTL_REWIND32 = _IOW('A', 0x46, u32), @@ -517,7 +539,9 @@ static long snd_pcm_ioctl_compat(struct file *file, unsigned int cmd, unsigned l case SNDRV_PCM_IOCTL_SW_PARAMS32: return snd_pcm_ioctl_sw_params_compat(substream, argp); case SNDRV_PCM_IOCTL_STATUS32: - return snd_pcm_status_user_compat(substream, argp); + return snd_pcm_status_user_compat(substream, argp, false); + case SNDRV_PCM_IOCTL_STATUS_EXT32: + return snd_pcm_status_user_compat(substream, argp, true); case SNDRV_PCM_IOCTL_SYNC_PTR32: return snd_pcm_ioctl_sync_ptr_compat(substream, argp); case SNDRV_PCM_IOCTL_CHANNEL_INFO32:

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 pierre-louis.bossart@linux.intel.com --- include/uapi/sound/asound.h | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-)

diff --git a/include/uapi/sound/asound.h b/include/uapi/sound/asound.h index acef4e4..3d46e9a 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;