docs/pwg/advanced-clock.xml

<chapter id="chapter-advanced-clock">
  <title>Clocking</title>

  <para>
    When playing complex media, each sound and video sample must be played in a
    specific order at a specific time. For this purpose, GStreamer provides a
    synchronization mechanism.
  </para>

  <sect1 id="section-clocks" xreflabel="Clocks">
    <title>Clocks</title>
    <para>
      Time in &GStreamer; is defined as the value returned from a particular 
      <classname>GstClock</classname> object from the method
      <function>gst_clock_get_time ()</function>. 
    </para>
    <para>
      In a typical computer, there are many sources that can be used as a
      time source, e.g., the system time, soundcards, CPU performance
      counters, ... For this reason, there are many
      <classname>GstClock</classname> implementations available in &GStreamer;.
      The clock time doesn't always start from 0 or from some known value.
      Some clocks start counting from some known start date, other clocks start
      counting since last reboot, etc...
    </para>
    <para>
      As clocks return an absolute measure of time, they are not usually used
      directly. Instead, differences between two clock times are used to
      measure elapsed time according to a clock.
    </para>
  </sect1>

  <sect1 id="section-clock-time-types" xreflabel="Clock running-time"> 
    <title> Clock running-time </title>
    <para>
      A clock returns the <emphasis role="strong">absolute-time</emphasis>
      according to that clock with <function>gst_clock_get_time ()</function>.
      From the absolute-time is a <emphasis role="strong">running-time</emphasis>
      calculated, which is simply the difference between a previous snapshot
      of the absolute-time called the <emphasis role="strong">base-time</emphasis>.
      So:
    </para>
    <para>
      running-time = absolute-time - base-time
    </para>
    <para>
      A &GStreamer; <classname>GstPipeline</classname> object maintains a 
      <classname>GstClock</classname> object and a base-time when it goes
      to the PLAYING state.  The pipeline gives a handle to the selected
      <classname>GstClock</classname> to each element in the pipeline along
      with selected base-time. The pipeline will select a base-time in such
      a way that the running-time reflects the total time spent in the
      PLAYING state. As a result, when the pipeline is PAUSED, the
      running-time stands still.
    </para>
    <para>
      Because all objects in the pipeline have the same clock and base-time,
      they can thus all calculate the running-time according to the pipeline
      clock.
    </para>
  </sect1>

  <sect1 id="section-buffer-time-types" xreflabel="Buffer running-time"> 
    <title> Buffer running-time </title>
    <para>
      To calculate a buffer running-time, we need a buffer timestamp and
      the SEGMENT event that preceded the buffer. First we can convert
      the SEGMENT event into a <classname>GstSegment</classname> object
      and then we can use the
      <function>gst_segment_to_running_time ()</function> function to
      perform the calculation of the buffer running-time.
    </para>
    <para>
      Synchronization is now a matter of making sure that a buffer with a
      certain running-time is played when the clock reaches the same
      running-time. Usually this task is done by sink elements. Sink also
      have to take into account the latency configured in the pipeline and
      add this to the buffer running-time before synchronizing to the
      pipeline clock.
    </para>
  </sect1>
  
  
  <sect1 id="section-clock-obligations-of-each-element" xreflabel="Obligations
    of each element">
    <title>
      Obligations of each element.
    </title>
    
    <para>
      Let us clarify the contract between GStreamer and each element in the
      pipeline.
    </para>
    
    <sect2>
      <title>Non-live source elements </title>
      <para>
        Non-live source elements must place a timestamp in each buffer that
        they deliver when this is possible.  They must choose the timestamps
        and the values of the SEGMENT event in such a way that the
        running-time of the buffer starts from 0.
      </para>
      <para>
        Some sources, such as filesrc, is not able to generate timestamps
        on all buffers. It can and must however create a timestamp on the
        first buffer (with a running-time of 0).
      </para>
      <para>
        The source then pushes out the SEGMENT event followed by the
        timestamped buffers.
      </para>
    </sect2>

    <sect2>
      <title>Live source elements </title>
      <para>
        Live source elements must place a timestamp in each buffer that
        they deliver. They must choose the timestamps and the values of the
        SEGMENT event in such a way that the running-time of the buffer
        matches exactly the running-time of the pipeline clock when the first 
        byte in the buffer was captured.
      </para>
    </sect2>

    <sect2>
      <title>Parser/Decoder/Encoder elements </title>
      <para>
        Parser/Decoder elements must use the incoming timestamps and transfer
        those to the resulting output buffers. They are allowed to interpolate
        or reconstruct timestamps on missing input buffers when they can.
      </para>
    </sect2>

    <sect2>
      <title>Demuxer elements </title>
      <para>
        Demuxer elements can usually set the timestamps stored inside the media
        file onto the outgoing buffers. They need to make sure that outgoing
        buffers that are to be played at the same time have the same
        running-time. Demuxers also need to take into account the incoming
        timestamps on buffers and use that to calculate an offset on the outgoing
        buffer timestamps.
      </para>
    </sect2>

    <sect2>
      <title>Muxer elements</title>
      <para>
        Muxer elements should use the incoming buffer running-time to mux the
        different streams together. They should copy the incoming running-time
        to the outgoing buffers.
      </para>
    </sect2>
    
    <sect2>
      <title>Sink elements</title>
      <para>
	If the element is intended to emit samples at a specific time (real time
	playing), the element should require a clock, and thus implement the
	method <function>set_clock</function>.
      </para>
      <para>
        The sink should then make sure that the sample with running-time is played
        exactly when the pipeline clock reaches that running-time + latency.
        Some elements might use the clock API such as
        <function>gst_clock_id_wait()</function>
        to perform this action. Other sinks might need to use other means of
        scheduling timely playback of the data.
      </para>
    </sect2>
  </sect1>
  
</chapter>