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This document outlines the techniques used for doing synchronised playback of
multiple streams.
Synchronisation in a GstPipeline is achieved using the following 3 components:
- a GstClock, which is global for all elements in a GstPipeline.
- Timestamps on a GstBuffer.
- the SEGMENT event preceding the buffers.
A GstClock
This object provides a counter that represents the current time in nanoseconds.
This value is called the absolute_time.
Different sources exist for this counter:
- the system time (with g_get_current_time() and with microsecond accuracy)
- monotonic time (with g_get_monotonic_time () with microsecond accuracy)
- an audio device (based on number of samples played)
- a network source based on packets received + timestamps in those packets (a
typical example is an RTP source)
- ...
In GStreamer any element can provide a GstClock object that can be used in the
pipeline. The GstPipeline object will select a clock from all the providers and
will distribute it to all other elements (see part-gstpipeline.txt).
A GstClock always counts time upwards and does not necessarily start at 0.
While it is possible, it is not recommended to create a clock derived from the
contents of a stream (for example, create a clock from the PCR in an mpeg-ts
Running time
After a pipeline selected a clock it will maintain the running_time based on the
selected clock. This running_time represents the total time spent in the PLAYING
state and is calculated as follows:
- If the pipeline is NULL/READY, the running_time is undefined.
- In PAUSED, the running_time remains at the time when it was last
PAUSED. When the stream is PAUSED for the first time, the running_time
is 0.
- In PLAYING, the running_time is the delta between the absolute_time
and the base time. The base time is defined as the absolute_time minus
the running_time at the time when the pipeline is set to PLAYING.
- after a flushing seek, the running_time is set to 0 (see part-seeking.txt).
This is accomplished by redistributing a new base_time to the elements that
got flushed.
This algorithm captures the running_time when the pipeline is set from PLAYING
to PAUSED and restores this time based on the current absolute_time when going
back to PLAYING. This allows for both clocks that progress when in the PAUSED
state (systemclock) and clocks that don't (audioclock).
The clock and pipeline now provide a running_time to all elements that want to
perform synchronisation. Indeed, the running time can be observed in each
element (during the PLAYING state) as:
C.running_time = absolute_time - base_time
We note C.running_time as the running_time obtained by looking at the clock.
This value is monotonically increasing at the rate of the clock.
The GstBuffer timestamps and the preceding SEGMENT event (See
part-streams.txt) define a transformation of the buffer timestamps to
running_time as follows:
The following notation is used:
B: GstBuffer
- B.timestamp = buffer timestamp (GST_BUFFER_PTS or GST_BUFFER_DTS)
S: SEGMENT event preceding the buffers.
- S.start: start field in the SEGMENT event. This is the lowest allowed
- S.stop: stop field in the SEGMENT event. This is the highers allowed
- S.rate: rate field of SEGMENT event. This is the desired playback rate.
- S.base: a base time for the time. This is the total elapsed running_time of any
previous segments.
- S.offset: an offset to apply to S.start or S.stop. This is the amount that
has already been elapsed in the segment.
Valid buffers for synchronisation are those with B.timestamp between S.start
and S.stop (after applying the S.offset). All other buffers outside this range
should be dropped or clipped to these boundaries (see also part-segments.txt).
The following transformation to running_time exist:
if (S.rate > 0.0)
B.running_time = (B.timestamp - (S.start + S.offset)) / ABS (S.rate) + S.base
B.running_time = ((S.stop - S.offset) - B.timestamp) / ABS (S.rate) + S.base
We write B.running_time as the running_time obtained from the SEGMENT event
and the buffers of that segment.
The first displayable buffer will yield a value of 0 (since B.timestamp ==
S.start and S.offset and S.base == 0).
For S.rate > 1.0, the timestamps will be scaled down to increase the playback
rate. Likewise, a rate between 0.0 and 1.0 will slow down playback.
For negative rates, timestamps are received stop S.stop to S.start so that the
first buffer received will be transformed into B.running_time of 0
(B.timestamp == S.stop and S.base == 0).
This makes it so that B.running_time is always monotonically increasing
starting from 0 with both positive and negative rates.
As we have seen, we can get a running_time:
- using the clock and the element's base_time with:
C.running_time = absolute_time - base_time
- using the buffer timestamp and the preceding SEGMENT event as (assuming
positive playback rate):
B.running_time = (B.timestamp - (S.start + S.offset)) / ABS (S.rate) + S.base
We prefix C. and B. before the two running times to note how they were
The task of synchronized playback is to make sure that we play a buffer with
B.running_time at the moment when the clock reaches the same C.running_time.
Thus the following must hold:
B.running_time = C.running_time
B.running_time = absolute_time - base_time
absolute_time = B.running_time + base_time
The absolute_time when a buffer with B.running_time should be played is noted
with B.sync_time. Thus:
B.sync_time = B.running_time + base_time
One then waits for the clock to reach B.sync_time before rendering the buffer in
the sink (See also part-clocks.txt).
For multiple streams this means that buffers with the same running_time are to
be displayed at the same time.
A demuxer must make sure that the SEGMENT it emits on its output pads yield
the same running_time for buffers that should be played synchronized. This
usually means sending the same SEGMENT on all pads and making sure that the
synchronized buffers have the same timestamps.
Stream time
The stream time is also known as the position in the stream and is a value
between 0 and the total duration of the media file.
It is the stream time that is used for:
- report the POSITION query in the pipeline
- the position used in seek events/queries
- the position used to synchronize controller values
Additional fields in the SEGMENT are used:
- S.time: time field in the SEGMENT event. This the stream-time of S.start
- S.applied_rate: The rate already applied to the stream.
Stream time is calculated using the buffer times and the preceding SEGMENT
event as follows:
stream_time = (B.timestamp - S.start) * ABS (S.applied_rate) + S.time
For negative rates, B.timestamp will go backwards from S.stop to S.start,
making the stream time go backwards.
In the PLAYING state, it is also possible to use the pipeline clock to derive
the current stream_time.
Give the two formulas above to match the clock times with buffer timestamps
allows us to rewrite the above formula for stream_time (and for positive rates).
C.running_time = absolute_time - base_time
B.running_time = (B.timestamp - (S.start + S.offset)) / ABS (S.rate) + S.base
(B.timestamp - (S.start + S.offset)) / ABS (S.rate) + S.base = absolute_time - base_time;
(B.timestamp - (S.start + S.offset)) / ABS (S.rate) = absolute_time - base_time - S.base;
(B.timestamp - (S.start + S.offset)) = (absolute_time - base_time - S.base) * ABS (S.rate)
(B.timestamp - S.start) = S.offset + (absolute_time - base_time - S.base) * ABS (S.rate)
filling (B.timestamp - S.start) in the above formule for stream time
stream_time = (S.offset + (absolute_time - base_time - S.base) * ABS (S.rate)) * ABS (S.applied_rate) + S.time
This last formula is typically used in sinks to report the current position in
an accurate and efficient way.
Note that the stream time is never used for synchronisation against the clock.