<chapter id="chapter-other-base" xreflabel="Pre-made base classes">
  <title>Pre-made base classes</title>
  <para>
    So far, we've been looking at low-level concepts of creating any type of
    &GStreamer; element. Now, let's assume that all you want is to create an
    simple audiosink that works exactly the same as, say,
    <quote>esdsink</quote>, or a filter that simply normalizes audio volume.
    Such elements are very general in concept and since they do nothing
    special, they should be easier to code than to provide your own scheduler
    activation functions and doing complex caps negotiation. For this purpose,
    &GStreamer; provides base classes that simplify some types of elements.
    Those base classes will be discussed in this chapter.
  </para>

  <sect1 id="section-base-sink" xreflabel="Writing a sink">
    <title>Writing a sink</title>
    <para>
      Sinks are special elements in &GStreamer;. This is because sink elements
      have to take care of <emphasis>preroll</emphasis>, which is the process
      that takes care that elements going into the
      <classname>GST_STATE_PAUSED</classname> state will have buffers ready
      after the state change. The result of this is that such elements can
      start processing data immediately after going into the
      <classname>GST_STATE_PLAYING</classname> state, without requiring to
      take some time to initialize outputs or set up decoders; all that is done
      already before the state-change to <classname>GST_STATE_PAUSED</classname>
      successfully completes.
    </para>
    <para>
      Preroll, however, is a complex process that would require the same
      code in many elements. Therefore, sink elements can derive from the
      <classname>GstBaseSink</classname> base-class, which does preroll and
      a few other utility functions automatically. The derived class only
      needs to implement a bunch of virtual functions and will work
      automatically.
    </para>
    <para>
      The base class implement much of the synchronization logic that a
      sink has to perform.
    </para>
    <para>
      The <classname>GstBaseSink</classname> base-class specifies some
      limitations on elements, though:
    </para>
    <itemizedlist>
      <listitem>
        <para>
          It requires that the sink only has one sinkpad. Sink elements that
          need more than one sinkpad, must make a manager element with
          multiple GstBaseSink elements inside.
        </para>
      </listitem>
    </itemizedlist>
    <para>
      Sink elements can derive from <classname>GstBaseSink</classname> using
      the usual <classname>GObject</classname> convenience macro
      <function>G_DEFINE_TYPE ()</function>:
    </para>
    <programlisting>
G_DEFINE_TYPE (GstMySink, gst_my_sink, GST_TYPE_BASE_SINK);

[..]

static void
gst_my_sink_class_init (GstMySinkClass * klass)
{
  klass->set_caps = [..];
  klass->render = [..];
[..]
}
    </programlisting>
    <para>
      The advantages of deriving from <classname>GstBaseSink</classname> are
      numerous:
    </para>
    <itemizedlist>
      <listitem>
        <para>
          Derived implementations barely need to be aware of preroll, and do
          not need to know anything about the technical implementation
          requirements of preroll. The base-class does all the hard work.
        </para>
        <para>
          Less code to write in the derived class, shared code (and thus
          shared bugfixes).
        </para>
      </listitem>
    </itemizedlist>
    <para>
      There are also specialized base classes for audio and video, let's look
      at those a bit.
    </para>

    <sect2 id="section-base-audiosink" xreflabel="Writing an audio sink">
      <title>Writing an audio sink</title>
      <para>
        Essentially, audio sink implementations are just a special case of a
        general sink. An audio sink has the added complexity that it needs to
        schedule playback of samples. It must match the clock selected in the
        pipeline against the clock of the audio device and calculate and
        compensate for drift and jitter.
      </para>
      <para>
        There are two audio base classes that you can choose to
        derive from, depending on your needs:
        <classname>GstAudioBasesink</classname> and
        <classname>GstAudioSink</classname>. The audiobasesink provides full
        control over how synchronization and scheduling is handled, by using
        a ringbuffer that the derived class controls and provides. The
        audiosink base-class is a derived class of the audiobasesink,
        implementing a standard ringbuffer implementing default
        synchronization and providing a standard audio-sample clock. Derived
        classes of this base class merely need to provide a <function>_open
        ()</function>, <function>_close ()</function> and a <function>_write
        ()</function> function implementation, and some optional functions.
        This should suffice for many sound-server output elements and even
        most interfaces. More demanding audio systems, such as Jack, would
        want to implement the <classname>GstAudioBaseSink</classname>
        base-class.
      </para>
      <para>
        The <classname>GstBaseAusioSink</classname> has little to no
        limitations and should fit virtually every implementation, but is
        hard to implement. The <classname>GstAudioSink</classname>, on the
        other hand, only fits those systems with a simple <function>open
        ()</function> / <function>close ()</function> / <function>write
        ()</function> API (which practically means pretty much all of them),
        but has the advantage that it is a lot easier to implement. The
        benefits of this second base class are large:
      </para>
      <itemizedlist>
        <listitem>
          <para>
            Automatic synchronization, without any code in the derived class.
          </para>
        </listitem>
        <listitem>
          <para>
            Also automatically provides a clock, so that other sinks (e.g. in
            case of audio/video playback) are synchronized.
          </para>
        </listitem>
        <listitem>
          <para>
            Features can be added to all audiosinks by making a change in the
            base class, which makes maintenance easy.
          </para>
        </listitem>
        <listitem>
          <para>
            Derived classes require only three small functions, plus some
            <classname>GObject</classname> boilerplate code.
          </para>
        </listitem>
      </itemizedlist>
      <para>
        In addition to implementing the audio base-class virtual functions,
        derived classes can (should) also implement the
        <classname>GstBaseSink</classname> <function>set_caps ()</function> and
        <function>get_caps ()</function> virtual functions for negotiation.
      </para>
    </sect2>

    <sect2 id="section-base-videosink" xreflabel="Writing a general sink">
      <title>Writing a video sink</title>
      <para>
        Writing a videosink can be done using the
        <classname>GstVideoSink</classname> base-class, which derives from
        <classname>GstBaseSink</classname> internally. Currently, it does
        nothing yet but add another compile dependency, so derived classes
        will need to implement all base-sink virtual functions. When they do
        this correctly, this will have some positive effects on the end user
        experience with the videosink:
      </para>
      <itemizedlist>
        <listitem>
          <para>
            Because of preroll (and the <function>preroll ()</function> virtual
            function), it is possible to display a video frame already when
            going into the <classname>GST_STATE_PAUSED</classname> state.
          </para>
        </listitem>
        <listitem>
          <para>
            By adding new features to <classname>GstVideoSink</classname>, it
            will be possible to add extensions to videosinks that affect all of
            them, but only need to be coded once, which is a huge maintenance
            benefit.
          </para>
        </listitem>
      </itemizedlist>
    </sect2>
  </sect1>

  <sect1 id="section-base-src" xreflabel="Writing a source">
    <title>Writing a source</title>
    <para>
      In the previous part, particularly <xref
      linkend="section-scheduling-randomxs"/>, we have learned that some types
      of elements can provide random access. This applies most definitely to
      source elements reading from a randomly seekable location, such as file
      sources. However, other source elements may be better described as a
      live source element, such as a camera source, an audio card source and
      such; those are not seekable and do not provide byte-exact access. For
      all such use cases, &GStreamer; provides two base classes:
      <classname>GstBaseSrc</classname> for the basic source functionality, and
      <classname>GstPushSrc</classname>, which is a non-byte exact source
      base-class. The pushsource base class itself derives from basesource as
      well, and thus all statements about the basesource apply to the
      pushsource, too.
    </para>
    <para>
      The basesrc class does several things automatically for derived classes,
      so they no longer have to worry about it:
    </para>
    <itemizedlist>
      <listitem>
        <para>
          Fixes to <classname>GstBaseSrc</classname> apply to all derived
          classes automatically.
        </para>
      </listitem>
      <listitem>
        <para>
          Automatic pad activation handling, and task-wrapping in case we get
          assigned to start a task ourselves.
        </para>
      </listitem>
    </itemizedlist>
    <para>
      The <classname>GstBaseSrc</classname> may not be suitable for all cases,
      though; it has limitations:
    </para>
    <itemizedlist>
      <listitem>
        <para>
          There is one and only one sourcepad. Source elements requiring
          multiple sourcepads must implement a manager bin and use multiple
          source elements internally or make a manager element that uses
          a source element and a demuxer inside.
        </para>
      </listitem>
    </itemizedlist>
    <para>
      It is possible to use special memory, such as X server memory pointers
      or <function>mmap ()</function>'ed memory areas, as data pointers in
      buffers returned from the <function>create()</function> virtual function.
    </para>

    <sect2 id="section-base-audiosrc" xreflabel="Writing an audio source">
      <title>Writing an audio source</title>
      <para>
        An audio source is nothing more but a special case of a pushsource.
        Audio sources would be anything that reads audio, such as a source
        reading from a soundserver, a kernel interface (such as ALSA) or a
        test sound / signal generator. &GStreamer; provides two base classes,
        similar to the two audiosinks described in <xref
        linkend="section-base-audiosink"/>; one is ringbuffer-based, and
        requires the derived class to take care of its own scheduling,
        synchronization and such. The other is based on this
        <classname>GstAudioBaseSrc</classname> and is called
        <classname>GstAudioSrc</classname>, and provides a simple
        <function>open ()</function>, <function>close ()</function> and
        <function>read ()</function> interface, which is rather simple to
        implement and will suffice for most soundserver sources and audio
        interfaces (e.g. ALSA or OSS) out there.
      </para>
      <para>
        The <classname>GstAudioSrc</classname> base-class has several benefits
        for derived classes, on top of the benefits of the
        <classname>GstPushSrc</classname> base-class that it is based on:
      </para>
      <itemizedlist>
        <listitem>
          <para>
            Does syncronization and provides a clock.
          </para>
        </listitem>
        <listitem>
          <para>
            New features can be added to it and will apply to all derived
            classes automatically.
          </para>
        </listitem>
      </itemizedlist>
    </sect2>
  </sect1>

  <sect1 id="section-base-transform"
      xreflabel="Writing a transformation element">
    <title>Writing a transformation element</title>
    <para>
      A third base-class that &GStreamer; provides is the
      <classname>GstBaseTransform</classname>. This is a base class for
      elements with one sourcepad and one sinkpad which act as a filter
      of some sort, such as volume changing, audio resampling, audio format
      conversion, and so on and so on. There is quite a lot of bookkeeping
      that such elements need to do in order for things such as buffer
      allocation forwarding, passthrough, in-place processing and such to all
      work correctly. This base class does all that for you, so that you just
      need to do the actual processing.
    </para>
    <para>
      Since the <classname>GstBaseTransform</classname> is based on the 1-to-1
      model for filters, it may not apply well to elements such as decoders,
      which may have to parse properties from the stream. Also, it will not
      work for elements requiring more than one sourcepad or sinkpad.
    </para>
  </sect1>
</chapter>
