gstreamer/docs/design/part-synchronisation.txt

Synchronisation
---------------

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 NEW_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)
 - 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.


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.


Timestamps
~~~~~~~~~~
 
The GstBuffer timestamps and the preceeding NEW_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_TIMESTAMP)

 NS:  NEWSEGMENT event preceeding the buffers.
  - NS.start: start field in the NEWSEGMENT event
  - NS.stop: stop field in the NEWSEGMENT event
  - NS.rate: rate field of NEWSEGMENT event
  - NS.abs_rate: absolute value of rate field of NEWSEGMENT event
  - NS.time: time field in the NEWSEGMENT event
  - NS.accum: total accumulated time of all previous NEWSEGMENT events. This
              field is kept in the GstSegment structure.

Valid buffers for synchronisation are those with B.timestamp between NS.start
and NS.stop. 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 (NS.rate > 0.0)
      B.running_time = (B.timestamp - NS.start) / NS.abs_rate + NS.accum
    else
      B.running_time = (NS.stop - B.timestamp) / NS.abs_rate + NS.accum

We write B.running_time as the running_time obtained from the NEWSEGMENT event
and the buffers of that segment.

The first displayable buffer will yield a value of 0 (since B.timestamp ==
NS.start and NS.accum == 0).

For NS.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 NS.stop to NS.start so that the
first buffer received will be transformed into B.running_time of 0 (B.timestamp ==
NS.stop and NS.accum == 0).


Synchronisation
~~~~~~~~~~~~~~~

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 preceeding NEWSEGMENT event as (assuming
   positive playback rate):

    B.running_time = (B.timestamp - NS.start) / NS.abs_rate + NS.accum

We prefix C. and B. before the two running times to note how they were
calculated.

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

expaning:

   B.running_time = absolute_time - base_time

or:

   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 NEWSEGMENT it emits on its output pads yield
the same running_time for buffers that should be played synchronized. This
usually means sending the same NEWSEGMENT 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

Stream time is calculated using the buffer times and the preceeding NEWSEGMENT
event as follows:

    stream_time = (B.timestamp - NS.start) * NS.abs_applied_rate + NS.time
 
For negative rates, B.timestamp will go backwards from NS.stop to NS.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 - NS.start) / NS.abs_rate + NS.accum

  =>
    (B.timestamp - NS.start) / NS.abs_rate + NS.accum = absolute_time - base_time;

  =>
    (B.timestamp - NS.start) / NS.abs_rate = absolute_time - base_time - NS.accum;

  =>
    (B.timestamp - NS.start) = (absolute_time - base_time - NS.accum) * NS.abs_rate

  filling (B.timestamp - NS.start) in the above formule for stream time

  =>
    stream_time = (absolute_time - base_time - NS.accum) * NS.abs_rate * NS.abs_applied_rate + NS.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.