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Docs updates
Original commit message from CVS: Docs updates
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@ -44,7 +44,8 @@
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The basic main function of a chain-based element is like:
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</para>
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<programlisting>
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static void chain_function (GstPad *pad, GstBuffer *buffer)
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static void
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chain_function (GstPad *pad, GstBuffer *buffer)
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{
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GstBuffer *outbuffer;
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@ -55,6 +56,16 @@ static void chain_function (GstPad *pad, GstBuffer *buffer)
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gst_pad_push (srcpad, outbuffer);
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}
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</programlisting>
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<para>
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Chain based function are mainly used for elements that have a one to one
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relation between their input and output behaviour. An example of such an
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element can be a simple video blur filter. The filter takes a buffer in, performs
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the blur operation on it and sends out the resulting buffer.
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</para>
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<para>
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Another element, for example, is a volume filter. The filter takes audio samples as
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input, performs the volume effect and sends out the resulting buffer.
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</para>
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</sect1>
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@ -71,6 +82,12 @@ static void chain_function (GstPad *pad, GstBuffer *buffer)
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buffer = gst_pad_pull (sinkpad);
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...
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// process buffer, create outbuffer
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while (!done) {
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....
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// optionally request another buffer
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buffer = gst_pad_pull (sinkpad);
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....
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}
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...
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gst_pad_push (srcpad, outbuffer);
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}
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@ -87,6 +104,13 @@ static void chain_function (GstPad *pad, GstBuffer *buffer)
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complex using cothreads when there are multiple input/output pads for the
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loop-based element.
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</para>
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<para>
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Loop based elements are mainly used for the more complex elements that need a
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specific amount of data before they can start to produce output. An example
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of such an element is the mpeg video decoder. the element will pull a buffer,
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performs some decoding on it and optionally requests more buffers to decode, when
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a complete video frame has been decoded, a buffer is send out.
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</para>
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<para>
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There is no problem in putting cothreaded elements into a
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<classname>GstThread</classname> to create even more complex pipelines with
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@ -42,7 +42,7 @@
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</sect1>
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<sect1>
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<title>MIME Types</title>
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<title>more on MIME Types</title>
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<para>
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GStreamer uses MIME types to indentify the different types of data
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that can be handled by the elements. They are the high level
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@ -77,6 +77,10 @@
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This allows for very dynamic and extensible element creation as we
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will see.
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</para>
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<para>
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As we have seen in the previous chapter, the MIME types are added
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to the Capability structure of a pad.
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</para>
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<para>
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In our helloworld example the elements we constructed would have the
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@ -325,8 +325,8 @@ eos (GstSrc *src)
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you can create a custom MP3 element with a more high level API.
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</para>
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<para>
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It should be clear from the example that we can vary easily replace the
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disksrc element with a httpsrc, giving you instant network streaming.
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It should be clear from the example that we can very easily replace the
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disksrc element with an httpsrc, giving you instant network streaming.
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An element could be build to handle icecast connections, for example.
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</para>
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<para>
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@ -7,7 +7,7 @@
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</para>
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<para>
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In this chapter we will introduce you to autoplugging. Using the MIME
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types of the elements GStreamer can automatically create a pipeline
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types of the elements <application>GStreamer</application> can automatically create a pipeline
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for you.
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</para>
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@ -26,63 +26,63 @@
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static gboolean playing;
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/* eos will be called when the src element has an end of stream */
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void eos(GstSrc *src)
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void
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eos (GstSrc *src)
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{
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g_print("have eos, quitting\n");
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g_print ("have eos, quitting\n");
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playing = FALSE;
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}
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int main(int argc,char *argv[])
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int
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main (int argc, char *argv[])
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{
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GstElement *disksrc, *audiosink;
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GstElement *pipeline;
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if (argc != 2) {
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g_print("usage: %s <filename>\n", argv[0]);
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exit(-1);
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g_print ("usage: %s <filename>\n", argv[0]);
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exit (-1);
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}
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gst_init(&argc,&argv);
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gst_init (&argc, &argv);
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/* create a new bin to hold the elements */
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pipeline = gst_pipeline_new("pipeline");
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pipeline = gst_pipeline_new ("pipeline");
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/* create a disk reader */
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disksrc = gst_elementfactory_make("disksrc", "disk_source");
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gtk_object_set(GTK_OBJECT(disksrc),"location", argv[1],NULL);
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gtk_signal_connect(GTK_OBJECT(disksrc),"eos",
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GTK_SIGNAL_FUNC(eos),NULL);
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disksrc = gst_elementfactory_make ("disksrc", "disk_source");
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gtk_object_set (GTK_OBJECT (disksrc), "location", argv[1], NULL);
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gtk_signal_connect (GTK_OBJECT (disksrc), "eos",
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GTK_SIGNAL_FUNC (eos), NULL);
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/* and an audio sink */
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audiosink = gst_elementfactory_make("audiosink", "play_audio");
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audiosink = gst_elementfactory_make ("audiosink", "play_audio");
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/* add objects to the main pipeline */
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gst_pipeline_add_src(GST_PIPELINE(pipeline), disksrc);
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gst_pipeline_add_sink(GST_PIPELINE(pipeline), audiosink);
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gst_pipeline_add_src (GST_PIPELINE (pipeline), disksrc);
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gst_pipeline_add_sink (GST_PIPELINE (pipeline), audiosink);
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if (!gst_pipeline_autoplug(GST_PIPELINE(pipeline))) {
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g_print("unable to handle stream\n");
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exit(-1);
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if (!gst_pipeline_autoplug (GST_PIPELINE (pipeline))) {
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g_print ("unable to handle stream\n");
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exit (-1);
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}
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/* make it ready */
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gst_element_set_state(GST_ELEMENT(pipeline), GST_STATE_READY);
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/* start playing */
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gst_element_set_state(GST_ELEMENT(pipeline), GST_STATE_PLAYING);
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gst_element_set_state (GST_ELEMENT (pipeline), GST_STATE_PLAYING);
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playing = TRUE;
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while (playing) {
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gst_bin_iterate(GST_BIN(pipeline));
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gst_bin_iterate (GST_BIN (pipeline));
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}
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/* stop the bin */
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gst_element_set_state(GST_ELEMENT(pipeline), GST_STATE_NULL);
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gst_element_set_state (GST_ELEMENT (pipeline), GST_STATE_NULL);
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gst_pipeline_destroy(pipeline);
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gst_pipeline_destroy (pipeline);
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exit(0);
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exit (0);
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}
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</programlisting>
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@ -99,9 +99,9 @@ int main(int argc,char *argv[])
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<programlisting>
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...
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if (!gst_pipeline_autoplug(pipeline)) {
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g_print("unable to handle stream\n");
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exit(-1);
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if (!gst_pipeline_autoplug (pipeline)) {
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g_print ("unable to handle stream\n");
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exit (-1);
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}
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...
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@ -147,7 +147,7 @@ int main(int argc,char *argv[])
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<para>
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If you really want, you can use the GSteamer components to do the
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autoplugging yourself.
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autoplugging yourself. We will cover this topic in the dynamic pipeline chapter.
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</para>
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<para>
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@ -165,9 +165,9 @@ int main(int argc,char *argv[])
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</programlisting>
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<para>
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You can also try to use an AVI or MPEG file as its input. Using autoplugging,
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GStreamer will automatically figure out how to handle the stream. Remember that
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only the audio part will be played because we have only added an audiosink to
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the pipeline.
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<application>GStreamer</application> will automatically figure out how to
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handle the stream. Remember that only the audio part will be played because
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we have only added an audiosink to the pipeline.
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</para>
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<programlisting>
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./helloworld2 mymovie.mpeg
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@ -157,6 +157,6 @@ Single
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4 0 0 50 0 0 12 0.0000 4 135 75 3675 5475 ?\001
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4 0 0 50 0 0 12 0.0000 4 135 735 9825 5475 audio/raw\001
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4 0 0 50 0 0 12 0.0000 4 180 855 4350 5325 audio/mpeg\001
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4 0 0 50 0 0 12 0.0000 4 180 1395 5475 5625 audio/mpeg-frame\001
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4 0 0 50 0 0 12 0.0000 4 135 735 8700 5625 audio/raw\001
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4 0 0 50 0 0 12 0.0000 4 180 1395 6825 5325 audio/mpeg-frame\001
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4 0 0 50 0 0 12 0.0000 4 180 855 5775 5625 audio/mpeg\001
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4 0 0 50 0 0 12 0.0000 4 180 855 7125 5325 audio/mpeg\001
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@ -116,6 +116,26 @@ Pads:
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</itemizedlist>
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</sect2>
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<sect2 id="sec-pads-caps-use">
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<title>What are the capabilities used for?</title>
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<para>
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Capabilities describe in great detail the type of media that is handled by the pads.
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They are mostly used for:
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</para>
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<itemizedlist>
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<listitem>
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<para>
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Autoplugging: automatically finding plugins for a set of capabilities
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</para>
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</listitem>
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<listitem>
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<para>
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Compatibility detection: when two pads are connected, <application>GStreamer</application>
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can verify if the two pads are talking about the same media types.
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</para>
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</listitem>
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</itemizedlist>
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</sect2>
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</sect1>
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</chapter>
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@ -103,6 +103,88 @@ gstreamer-launch disksrc redpill.vob ! css-descramble ! private_stream_1.0 ! \
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</para>
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</sect1>
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<sect1>
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<title><command>gstreamer-inspect</command></title>
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<para>
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This is a tool to query a plugin or an element about its properties.
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</para>
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<para>
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To query the information about the element mpg123, you would specify:
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</para>
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<screen>
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gstreamer-inspect mpg123
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</screen>
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<para>
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Below is the output of a query for the audiosink element:
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</para>
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<screen>
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Factory Details:
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Long name: Audio Sink (OSS)
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Class: Sink/Audio
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Description: Output to a sound card via OSS
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Version: 0.1.0
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Author(s): Erik Walthinsen <omega@cse.ogi.edu>
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Copyright: (C) 1999
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Pad Templates:
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SINK template: 'sink'
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Exists: Always
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Capabilities:
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'audiosink_sink':
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MIME type: 'audio/raw':
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format: Integer: 16
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depth: List:
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Integer: 8
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Integer: 16
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rate: Integer range: 8000 - 48000
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channels: Integer range: 1 - 2
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Element Flags:
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GST_ELEMENT_THREADSUGGESTED
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no flags set
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Element Implementation:
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No loopfunc(), must be chain-based or not configured yet
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Has change_state() function
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Pads:
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SINK: 'sink'
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Implementation:
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Has chainfunc(): 0x4001cde8
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Has default eosfunc() gst_pad_eos_func()
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Pad Template: 'sink'
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Capabilities:
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'audiosink_sink':
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MIME type: 'audio/raw':
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format: Integer: 16
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depth: List:
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Integer: 8
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Integer: 16
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rate: Integer range: 8000 - 48000
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channels: Integer range: 1 - 2
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Element Arguments:
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GstAudioSink::mute: Boolean
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GstAudioSink::format: Enum (default 16)
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(8): 8 Bits
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(16): 16 Bits
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GstAudioSink::channels: Enum (default 2)
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(1): Mono
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(2): Stereo
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GstAudioSink::frequency: Integer
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</screen>
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<para>
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To query the information about a plugin, you would do:
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</para>
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<screen>
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gstreamer-inspect gstelements
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</screen>
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</sect1>
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<sect1>
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<title><command>gstmediaplay</command></title>
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<para>
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|
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@ -1,20 +1,20 @@
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<chapter id="cha-queues">
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<title>Queues</title>
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<para>
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A <classname>GstQueue</classname> is an implementation of a <classname>GstConnection</classname>.
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A <classname>GstQueue</classname> is a filter element.
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Queues can be used to connect two elements in such way that the data can
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be buffered.
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</para>
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<para>
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A buffer that is sinked to a Queue will not automatically be pushed to the
|
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next connected element but will be buffered. It will be pushed to the next
|
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element as soon as gst_connection_push() is called.
|
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element as soon as gst_connection_push () is called.
|
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</para>
|
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<para>
|
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Queues are mostly used in conjunction with a <classname>GstThread</classname> to
|
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provide an external connection for the thread elements. You could have one
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thread feeding buffers into a <classname>GstQueue</classname> and another
|
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thread repeadedly calling gst_connection_push() on the queue to feed its
|
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thread repeadedly calling gst_connection_push () on the queue to feed its
|
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internal elements.
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</para>
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|
|
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@ -14,21 +14,18 @@
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GstElement *my_thread;
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// create the thread object
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my_thread = gst_thread_new("my_thread");
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g_return_if_fail(audio_thread != NULL);
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my_thread = gst_thread_new ("my_thread");
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g_return_if_fail (audio_thread != NULL);
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// add some plugins
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gst_bin_add(GST_BIN(my_thread),GST_ELEMENT(funky_src));
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gst_bin_add(GST_BIN(my_thread),GST_ELEMENT(cool_effect));
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gst_bin_add (GST_BIN (my_thread), GST_ELEMENT (funky_src));
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gst_bin_add (GST_BIN (my_thread), GST_ELEMENT (cool_effect));
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|
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// connect the elements here...
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...
|
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|
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// prepare the thread
|
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gst_element_set_state(GST_ELEMENT(my_thread),GST_STATE_READY);
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|
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// start playing
|
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gst_element_set_state(GST_ELEMENT(my_thread),GST_STATE_PLAYING);
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gst_element_set_state (GST_ELEMENT (my_thread), GST_STATE_PLAYING);
|
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|
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</programlisting>
|
||||
|
||||
|
@ -39,8 +36,8 @@
|
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|
||||
<note>
|
||||
<para>
|
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The thread must contain at least one element of type <classname>GstSrc</classname>
|
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or <classname>GstConnection</classname> in order to work.
|
||||
A thread should normally contain a source element. Most often, the thread
|
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is fed with data from a queue.
|
||||
</para>
|
||||
</note>
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|
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|
@ -60,82 +57,82 @@
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|||
#include <gst/gst.h>
|
||||
|
||||
/* eos will be called when the src element has an end of stream */
|
||||
void eos(GstSrc *src, gpointer data)
|
||||
void
|
||||
eos (GstSrc *src, gpointer data)
|
||||
{
|
||||
GstThread *thread = GST_THREAD(data);
|
||||
g_print("have eos, quitting\n");
|
||||
GstThread *thread = GST_THREAD (data);
|
||||
g_print ("have eos, quitting\n");
|
||||
|
||||
/* stop the bin */
|
||||
gst_element_set_state(GST_ELEMENT(thread), GST_STATE_NULL);
|
||||
gst_element_set_state (GST_ELEMENT (thread), GST_STATE_NULL);
|
||||
|
||||
gst_main_quit();
|
||||
gst_main_quit ();
|
||||
}
|
||||
|
||||
int main(int argc,char *argv[])
|
||||
int
|
||||
main (int argc, char *argv[])
|
||||
{
|
||||
GstElement *disksrc, *audiosink;
|
||||
GstElement *pipeline;
|
||||
GstElement *thread;
|
||||
|
||||
if (argc != 2) {
|
||||
g_print("usage: %s <filename>\n", argv[0]);
|
||||
exit(-1);
|
||||
g_print ("usage: %s <filename>\n", argv[0]);
|
||||
exit (-1);
|
||||
}
|
||||
|
||||
gst_init(&argc,&argv);
|
||||
gst_init (&argc, &argv);
|
||||
|
||||
/* create a new thread to hold the elements */
|
||||
thread = gst_thread_new("thread");
|
||||
g_assert(thread != NULL);
|
||||
thread = gst_thread_new ("thread");
|
||||
g_assert (thread != NULL);
|
||||
|
||||
/* create a new bin to hold the elements */
|
||||
pipeline = gst_pipeline_new("pipeline");
|
||||
g_assert(pipeline != NULL);
|
||||
pipeline = gst_pipeline_new ("pipeline");
|
||||
g_assert (pipeline != NULL);
|
||||
|
||||
/* create a disk reader */
|
||||
disksrc = gst_elementfactory_make("disksrc", "disk_source");
|
||||
g_assert(disksrc != NULL);
|
||||
gtk_object_set(GTK_OBJECT(disksrc),"location", argv[1],NULL);
|
||||
gtk_signal_connect(GTK_OBJECT(disksrc),"eos",
|
||||
GTK_SIGNAL_FUNC(eos), thread);
|
||||
disksrc = gst_elementfactory_make ("disksrc", "disk_source");
|
||||
g_assert (disksrc != NULL);
|
||||
gtk_object_set (GTK_OBJECT (disksrc), "location", argv[1], NULL);
|
||||
gtk_signal_connect (GTK_OBJECT (disksrc), "eos",
|
||||
GTK_SIGNAL_FUNC (eos), thread);
|
||||
|
||||
/* and an audio sink */
|
||||
audiosink = gst_elementfactory_make("audiosink", "play_audio");
|
||||
g_assert(audiosink != NULL);
|
||||
audiosink = gst_elementfactory_make ("audiosink", "play_audio");
|
||||
g_assert (audiosink != NULL);
|
||||
|
||||
/* add objects to the main pipeline */
|
||||
gst_bin_add(GST_BIN(pipeline), disksrc);
|
||||
gst_bin_add(GST_BIN(pipeline), audiosink);
|
||||
gst_bin_add (GST_BIN (pipeline), disksrc);
|
||||
gst_bin_add (GST_BIN (pipeline), audiosink);
|
||||
|
||||
/* automatically setup the pipeline */
|
||||
if (!gst_pipeline_autoplug(GST_PIPELINE(pipeline))) {
|
||||
g_print("unable to handle stream\n");
|
||||
exit(-1);
|
||||
if (!gst_pipeline_autoplug (GST_PIPELINE (pipeline))) {
|
||||
g_print ("unable to handle stream\n");
|
||||
exit (-1);
|
||||
}
|
||||
|
||||
/* remove the source element from the pipeline */
|
||||
gst_bin_remove(GST_BIN(pipeline), disksrc);
|
||||
gst_bin_remove (GST_BIN (pipeline), disksrc);
|
||||
|
||||
/* insert the source element in the thread, remember a thread needs at
|
||||
least one source or connection element */
|
||||
gst_bin_add(GST_BIN(thread), disksrc);
|
||||
gst_bin_add (GST_BIN (thread), disksrc);
|
||||
|
||||
/* add the pipeline to the thread too */
|
||||
gst_bin_add(GST_BIN(thread), GST_ELEMENT(pipeline));
|
||||
gst_bin_add (GST_BIN (thread), GST_ELEMENT (pipeline));
|
||||
|
||||
/* make it ready */
|
||||
gst_element_set_state(GST_ELEMENT(thread), GST_STATE_READY);
|
||||
/* start playing */
|
||||
gst_element_set_state(GST_ELEMENT(thread), GST_STATE_PLAYING);
|
||||
gst_element_set_state (GST_ELEMENT (thread), GST_STATE_PLAYING);
|
||||
|
||||
/* do whatever you want here, the thread will be playing */
|
||||
...
|
||||
|
||||
gst_main();
|
||||
gst_main ();
|
||||
|
||||
gst_pipeline_destroy(thread);
|
||||
gst_pipeline_destroy (thread);
|
||||
|
||||
exit(0);
|
||||
exit (0);
|
||||
}
|
||||
</programlisting>
|
||||
|
||||
|
|
Loading…
Reference in a new issue