gstreamer/docs/manual/advanced-clocks.xml

<chapter id="chapter-clocks">
  <title>Clocks in GStreamer</title>

  <para>
    To maintain sync in pipeline playback (which is the only case where this
    really matters), &GStreamer; uses <emphasis>clocks</emphasis>. Clocks
    are exposed by some elements, whereas other elements are merely clock
    slaves. The primary task of a clock is to represent the time progress
    according to the element exposing the clock, based on its own playback
    rate. If no clock provider is available in a pipeline, the system clock
    is used instead.
  </para>

  <sect1 id="section-clocks-providers">
    <title>Clock providers</title>

    <para>
      Clock providers exist because they play back media at some rate, and
      this rate is not necessarily the same as the system clock rate. For
      example, a soundcard may playback at 44,1 kHz, but that doesn't mean
      that after <emphasis>exactly</emphasis> 1 second <emphasis>according
      to the system clock</emphasis>, the soundcard has played back 44.100
      samples. This is only true by approximation. Therefore, generally,
      pipelines with an audio output use the audiosink as clock provider.
      This ensures that one second of video will be played back at the same
      rate as that the soundcard plays back 1 second of audio.
    </para>
    <para>
      Whenever some part of the pipeline requires to know the current clock
      time, it will be requested from the clock through
      <function>gst_clock_get_time ()</function>. The clock-time does not
      need to start at 0. The pipeline, which contains the global clock that
      all elements in the pipeline will use, in addition has a <quote>base
      time</quote>, which is the clock time at the the point where media time
      is starting from zero. This timestamp is subctracted from the clock
      time, and that value is returned by <function>_get_time ()</function>.
    </para>
    <para>
      The clock provider is responsible for making sure that the clock time
      always represents the current media time as closely as possible; it
      has to take care of things such as playback latencies, buffering in
      audio-kernel modules, and so on, since all those could affect a/v sync
      and thus decrease the user experience.
    </para>
  </sect1>

  <sect1 id="section-clocks-slaves">
    <title>Clock slaves</title>
    <para>
      Clock slaves get assigned a clock by their containing pipeline. Their
      task is to make sure that media playback follows the time progress as
      represented by this clock as closely as possible. For most elements,
      that will simply mean to wait until a certain time is reached before
      playing back their current sample; this can be done with the function
      <function>gst_clock_id_wait ()</function>. Some elements may need to
      support dropping samples too, however.
    </para>
    <para>
      For more information on how to write elements that conform to this
      required behaviour, see the Plugin Writer's Guide.
    </para>
  </sect1>
</chapter>