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 <pierre-louis.bossart@linux.intel.com> Signed-off-by: Takashi Iwai <tiwai@suse.de>hifive-unleashed-5.1
Pierre-Louis Bossart
Takashi Iwai
parent
0c8d948565
commit
229d043096
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@ -0,0 +1,200 @@
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The ALSA API can provide two different system timestamps:
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- Trigger_tstamp is the system time snapshot taken when the .trigger
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callback is invoked. This snapshot is taken by the ALSA core in the
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general case, but specific hardware may have synchronization
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capabilities or conversely may only be able to provide a correct
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estimate with a delay. In the latter two cases, the low-level driver
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is responsible for updating the trigger_tstamp at the most appropriate
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and precise moment. Applications should not rely solely on the first
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trigger_tstamp but update their internal calculations if the driver
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provides a refined estimate with a delay.
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- tstamp is the current system timestamp updated during the last
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event or application query.
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The difference (tstamp - trigger_tstamp) defines the elapsed time.
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The ALSA API provides reports two basic pieces of information, avail
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and delay, which combined with the trigger and current system
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timestamps allow for applications to keep track of the 'fullness' of
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the ring buffer and the amount of queued samples.
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The use of these different pointers and time information depends on
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the application needs:
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- 'avail' reports how much can be written in the ring buffer
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- 'delay' reports the time it will take to hear a new sample after all
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queued samples have been played out.
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When timestamps are enabled, the avail/delay information is reported
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along with a snapshot of system time. Applications can select from
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CLOCK_REALTIME (NTP corrections including going backwards),
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CLOCK_MONOTONIC (NTP corrections but never going backwards),
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CLOCK_MONOTIC_RAW (without NTP corrections) and change the mode
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dynamically with sw_params
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The ALSA API also provide an audio_tstamp which reflects the passage
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of time as measured by different components of audio hardware. In
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ascii-art, this could be represented as follows (for the playback
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case):
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--------------------------------------------------------------> time
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^ ^ ^ ^ ^
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analog link dma app FullBuffer
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time time time time time
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|< codec delay >|<--hw delay-->|<queued samples>|<---avail->|
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|<----------------- delay---------------------->| |
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|<----ring buffer length---->|
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The analog time is taken at the last stage of the playback, as close
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as possible to the actual transducer
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The link time is taken at the output of the SOC/chipset as the samples
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are pushed on a link. The link time can be directly measured if
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supported in hardware by sample counters or wallclocks (e.g. with
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HDAudio 24MHz or PTP clock for networked solutions) or indirectly
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estimated (e.g. with the frame counter in USB).
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The DMA time is measured using counters - typically the least reliable
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of all measurements due to the bursty natured of DMA transfers.
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The app time corresponds to the time tracked by an application after
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writing in the ring buffer.
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The application can query what the hardware supports, define which
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audio time it wants reported by selecting the relevant settings in
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audio_tstamp_config fields, get an estimate of the timestamp
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accuracy. It can also request the delay-to-analog be included in the
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measurement. Direct access to the link time is very interesting on
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platforms that provide an embedded DSP; measuring directly the link
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time with dedicated hardware, possibly synchronized with system time,
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removes the need to keep track of internal DSP processing times and
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latency.
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In case the application requests an audio tstamp that is not supported
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in hardware/low-level driver, the type is overridden as DEFAULT and the
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timestamp will report the DMA time based on the hw_pointer value.
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For backwards compatibility with previous implementations that did not
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provide timestamp selection, with a zero-valued COMPAT timestamp type
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the results will default to the HDAudio wall clock for playback
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streams and to the DMA time (hw_ptr) in all other cases.
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The audio timestamp accuracy can be returned to user-space, so that
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appropriate decisions are made:
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- for dma time (default), the granularity of the transfers can be
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inferred from the steps between updates and in turn provide
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information on how much the application pointer can be rewound
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safely.
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- the link time can be used to track long-term drifts between audio
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and system time using the (tstamp-trigger_tstamp)/audio_tstamp
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ratio, the precision helps define how much smoothing/low-pass
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filtering is required. The link time can be either reset on startup
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or reported as is (the latter being useful to compare progress of
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different streams - but may require the wallclock to be always
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running and not wrap-around during idle periods). If supported in
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hardware, the absolute link time could also be used to define a
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precise start time (patches WIP)
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- including the delay in the audio timestamp may
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counter-intuitively not increase the precision of timestamps, e.g. if a
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codec includes variable-latency DSP processing or a chain of
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hardware components the delay is typically not known with precision.
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The accuracy is reported in nanosecond units (using an unsigned 32-bit
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word), which gives a max precision of 4.29s, more than enough for
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audio applications...
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Due to the varied nature of timestamping needs, even for a single
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application, the audio_tstamp_config can be changed dynamically. In
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the STATUS ioctl, the parameters are read-only and do not allow for
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any application selection. To work around this limitation without
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impacting legacy applications, a new STATUS_EXT ioctl is introduced
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with read/write parameters. ALSA-lib will be modified to make use of
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STATUS_EXT and effectively deprecate STATUS.
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The ALSA API only allows for a single audio timestamp to be reported
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at a time. This is a conscious design decision, reading the audio
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timestamps from hardware registers or from IPC takes time, the more
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timestamps are read the more imprecise the combined measurements
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are. To avoid any interpretation issues, a single (system, audio)
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timestamp is reported. Applications that need different timestamps
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will be required to issue multiple queries and perform an
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interpolation of the results
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In some hardware-specific configuration, the system timestamp is
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latched by a low-level audio subsytem, and the information provided
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back to the driver. Due to potential delays in the communication with
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the hardware, there is a risk of misalignment with the avail and delay
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information. To make sure applications are not confused, a
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driver_timestamp field is added in the snd_pcm_status structure; this
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timestamp shows when the information is put together by the driver
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before returning from the STATUS and STATUS_EXT ioctl. in most cases
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this driver_timestamp will be identical to the regular system tstamp.
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Examples of typestamping with HDaudio:
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1. DMA timestamp, no compensation for DMA+analog delay
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$ ./audio_time -p --ts_type=1
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playback: systime: 341121338 nsec, audio time 342000000 nsec, systime delta -878662
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playback: systime: 426236663 nsec, audio time 427187500 nsec, systime delta -950837
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playback: systime: 597080580 nsec, audio time 598000000 nsec, systime delta -919420
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playback: systime: 682059782 nsec, audio time 683020833 nsec, systime delta -961051
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playback: systime: 852896415 nsec, audio time 853854166 nsec, systime delta -957751
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playback: systime: 937903344 nsec, audio time 938854166 nsec, systime delta -950822
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2. DMA timestamp, compensation for DMA+analog delay
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$ ./audio_time -p --ts_type=1 -d
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playback: systime: 341053347 nsec, audio time 341062500 nsec, systime delta -9153
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playback: systime: 426072447 nsec, audio time 426062500 nsec, systime delta 9947
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playback: systime: 596899518 nsec, audio time 596895833 nsec, systime delta 3685
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playback: systime: 681915317 nsec, audio time 681916666 nsec, systime delta -1349
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playback: systime: 852741306 nsec, audio time 852750000 nsec, systime delta -8694
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3. link timestamp, compensation for DMA+analog delay
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$ ./audio_time -p --ts_type=2 -d
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playback: systime: 341060004 nsec, audio time 341062791 nsec, systime delta -2787
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playback: systime: 426242074 nsec, audio time 426244875 nsec, systime delta -2801
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playback: systime: 597080992 nsec, audio time 597084583 nsec, systime delta -3591
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playback: systime: 682084512 nsec, audio time 682088291 nsec, systime delta -3779
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playback: systime: 852936229 nsec, audio time 852940916 nsec, systime delta -4687
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playback: systime: 938107562 nsec, audio time 938112708 nsec, systime delta -5146
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Example 1 shows that the timestamp at the DMA level is close to 1ms
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ahead of the actual playback time (as a side time this sort of
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measurement can help define rewind safeguards). Compensating for the
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DMA-link delay in example 2 helps remove the hardware buffering abut
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the information is still very jittery, with up to one sample of
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error. In example 3 where the timestamps are measured with the link
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wallclock, the timestamps show a monotonic behavior and a lower
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dispersion.
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Example 3 and 4 are with USB audio class. Example 3 shows a high
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offset between audio time and system time due to buffering. Example 4
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shows how compensating for the delay exposes a 1ms accuracy (due to
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the use of the frame counter by the driver)
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Example 3: DMA timestamp, no compensation for delay, delta of ~5ms
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$ ./audio_time -p -Dhw:1 -t1
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playback: systime: 120174019 nsec, audio time 125000000 nsec, systime delta -4825981
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playback: systime: 245041136 nsec, audio time 250000000 nsec, systime delta -4958864
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playback: systime: 370106088 nsec, audio time 375000000 nsec, systime delta -4893912
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playback: systime: 495040065 nsec, audio time 500000000 nsec, systime delta -4959935
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playback: systime: 620038179 nsec, audio time 625000000 nsec, systime delta -4961821
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playback: systime: 745087741 nsec, audio time 750000000 nsec, systime delta -4912259
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playback: systime: 870037336 nsec, audio time 875000000 nsec, systime delta -4962664
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Example 4: DMA timestamp, compensation for delay, delay of ~1ms
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$ ./audio_time -p -Dhw:1 -t1 -d
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playback: systime: 120190520 nsec, audio time 120000000 nsec, systime delta 190520
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playback: systime: 245036740 nsec, audio time 244000000 nsec, systime delta 1036740
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playback: systime: 370034081 nsec, audio time 369000000 nsec, systime delta 1034081
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playback: systime: 495159907 nsec, audio time 494000000 nsec, systime delta 1159907
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playback: systime: 620098824 nsec, audio time 619000000 nsec, systime delta 1098824
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playback: systime: 745031847 nsec, audio time 744000000 nsec, systime delta 1031847
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@ -60,6 +60,9 @@ struct snd_pcm_hardware {
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struct snd_pcm_substream;
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struct snd_pcm_audio_tstamp_config; /* definitions further down */
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struct snd_pcm_audio_tstamp_report;
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struct snd_pcm_ops {
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int (*open)(struct snd_pcm_substream *substream);
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int (*close)(struct snd_pcm_substream *substream);
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@ -281,6 +284,58 @@ struct snd_pcm_hw_constraint_ranges {
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struct snd_pcm_hwptr_log;
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/*
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* userspace-provided audio timestamp config to kernel,
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* structure is for internal use only and filled with dedicated unpack routine
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*/
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struct snd_pcm_audio_tstamp_config {
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/* 5 of max 16 bits used */
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u32 type_requested:4;
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u32 report_delay:1; /* add total delay to A/D or D/A */
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};
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static inline void snd_pcm_unpack_audio_tstamp_config(__u32 data,
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struct snd_pcm_audio_tstamp_config *config)
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{
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config->type_requested = data & 0xF;
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config->report_delay = (data >> 4) & 1;
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}
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/*
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* kernel-provided audio timestamp report to user-space
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* structure is for internal use only and read by dedicated pack routine
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*/
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struct snd_pcm_audio_tstamp_report {
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/* 6 of max 16 bits used for bit-fields */
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/* for backwards compatibility */
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u32 valid:1;
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/* actual type if hardware could not support requested timestamp */
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u32 actual_type:4;
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/* accuracy represented in ns units */
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u32 accuracy_report:1; /* 0 if accuracy unknown, 1 if accuracy field is valid */
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u32 accuracy; /* up to 4.29s, will be packed in separate field */
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};
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static inline void snd_pcm_pack_audio_tstamp_report(__u32 *data, __u32 *accuracy,
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const struct snd_pcm_audio_tstamp_report *report)
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{
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u32 tmp;
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tmp = report->accuracy_report;
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tmp <<= 4;
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tmp |= report->actual_type;
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tmp <<= 1;
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tmp |= report->valid;
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*data &= 0xffff; /* zero-clear MSBs */
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*data |= (tmp << 16);
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*accuracy = report->accuracy;
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}
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struct snd_pcm_runtime {
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/* -- Status -- */
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struct snd_pcm_substream *trigger_master;
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@ -361,6 +416,11 @@ struct snd_pcm_runtime {
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struct snd_dma_buffer *dma_buffer_p; /* allocated buffer */
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/* -- audio timestamp config -- */
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struct snd_pcm_audio_tstamp_config audio_tstamp_config;
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struct snd_pcm_audio_tstamp_report audio_tstamp_report;
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struct timespec driver_tstamp;
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#if defined(CONFIG_SND_PCM_OSS) || defined(CONFIG_SND_PCM_OSS_MODULE)
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/* -- OSS things -- */
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struct snd_pcm_oss_runtime oss;
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@ -267,10 +267,17 @@ typedef int __bitwise snd_pcm_subformat_t;
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#define SNDRV_PCM_INFO_JOINT_DUPLEX 0x00200000 /* playback and capture stream are somewhat correlated */
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#define SNDRV_PCM_INFO_SYNC_START 0x00400000 /* pcm support some kind of sync go */
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#define SNDRV_PCM_INFO_NO_PERIOD_WAKEUP 0x00800000 /* period wakeup can be disabled */
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#define SNDRV_PCM_INFO_HAS_WALL_CLOCK 0x01000000 /* has audio wall clock for audio/system time sync */
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#define SNDRV_PCM_INFO_HAS_WALL_CLOCK 0x01000000 /* (Deprecated)has audio wall clock for audio/system time sync */
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#define SNDRV_PCM_INFO_HAS_LINK_ATIME 0x01000000 /* report hardware link audio time, reset on startup */
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#define SNDRV_PCM_INFO_HAS_LINK_ABSOLUTE_ATIME 0x02000000 /* report absolute hardware link audio time, not reset on startup */
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#define SNDRV_PCM_INFO_HAS_LINK_ESTIMATED_ATIME 0x04000000 /* report estimated link audio time */
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#define SNDRV_PCM_INFO_HAS_LINK_SYNCHRONIZED_ATIME 0x08000000 /* report synchronized audio/system time */
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#define SNDRV_PCM_INFO_DRAIN_TRIGGER 0x40000000 /* internal kernel flag - trigger in drain */
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#define SNDRV_PCM_INFO_FIFO_IN_FRAMES 0x80000000 /* internal kernel flag - FIFO size is in frames */
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typedef int __bitwise snd_pcm_state_t;
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#define SNDRV_PCM_STATE_OPEN ((__force snd_pcm_state_t) 0) /* stream is open */
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#define SNDRV_PCM_STATE_SETUP ((__force snd_pcm_state_t) 1) /* stream has a setup */
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@ -408,6 +415,22 @@ struct snd_pcm_channel_info {
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unsigned int step; /* samples distance in bits */
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};
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enum {
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/*
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* first definition for backwards compatibility only,
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* maps to wallclock/link time for HDAudio playback and DEFAULT/DMA time for everything else
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*/
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SNDRV_PCM_AUDIO_TSTAMP_TYPE_COMPAT = 0,
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/* timestamp definitions */
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SNDRV_PCM_AUDIO_TSTAMP_TYPE_DEFAULT = 1, /* DMA time, reported as per hw_ptr */
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SNDRV_PCM_AUDIO_TSTAMP_TYPE_LINK = 2, /* link time reported by sample or wallclock counter, reset on startup */
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SNDRV_PCM_AUDIO_TSTAMP_TYPE_LINK_ABSOLUTE = 3, /* link time reported by sample or wallclock counter, not reset on startup */
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SNDRV_PCM_AUDIO_TSTAMP_TYPE_LINK_ESTIMATED = 4, /* link time estimated indirectly */
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SNDRV_PCM_AUDIO_TSTAMP_TYPE_LINK_SYNCHRONIZED = 5, /* link time synchronized with system time */
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SNDRV_PCM_AUDIO_TSTAMP_TYPE_LAST = SNDRV_PCM_AUDIO_TSTAMP_TYPE_LINK_SYNCHRONIZED
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};
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struct snd_pcm_status {
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snd_pcm_state_t state; /* stream state */
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struct timespec trigger_tstamp; /* time when stream was started/stopped/paused */
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@ -419,9 +442,11 @@ struct snd_pcm_status {
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snd_pcm_uframes_t avail_max; /* max frames available on hw since last status */
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snd_pcm_uframes_t overrange; /* count of ADC (capture) overrange detections from last status */
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snd_pcm_state_t suspended_state; /* suspended stream state */
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__u32 reserved_alignment; /* must be filled with zero */
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struct timespec audio_tstamp; /* from sample counter or wall clock */
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unsigned char reserved[56-sizeof(struct timespec)]; /* must be filled with zero */
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__u32 audio_tstamp_data; /* needed for 64-bit alignment, used for configs/report to/from userspace */
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struct timespec audio_tstamp; /* sample counter, wall clock, PHC or on-demand sync'ed */
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struct timespec driver_tstamp; /* useful in case reference system tstamp is reported with delay */
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__u32 audio_tstamp_accuracy; /* in ns units, only valid if indicated in audio_tstamp_data */
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unsigned char reserved[52-2*sizeof(struct timespec)]; /* must be filled with zero */
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};
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struct snd_pcm_mmap_status {
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@ -534,6 +559,7 @@ enum {
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#define SNDRV_PCM_IOCTL_DELAY _IOR('A', 0x21, snd_pcm_sframes_t)
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#define SNDRV_PCM_IOCTL_HWSYNC _IO('A', 0x22)
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#define SNDRV_PCM_IOCTL_SYNC_PTR _IOWR('A', 0x23, struct snd_pcm_sync_ptr)
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#define SNDRV_PCM_IOCTL_STATUS_EXT _IOWR('A', 0x24, struct snd_pcm_status)
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#define SNDRV_PCM_IOCTL_CHANNEL_INFO _IOR('A', 0x32, struct snd_pcm_channel_info)
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#define SNDRV_PCM_IOCTL_PREPARE _IO('A', 0x40)
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#define SNDRV_PCM_IOCTL_RESET _IO('A', 0x41)
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