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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>
201 lines
10 KiB
Plaintext
201 lines
10 KiB
Plaintext
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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