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docs/manual/: Update for 0.10: s/0.9/0.10/; s/audioscale/audiorsample/; add converters in front of pipelines; remove ...
Original commit message from CVS: * docs/manual/advanced-dataaccess.xml: * docs/manual/appendix-checklist.xml: * docs/manual/appendix-programs.xml: * docs/manual/basics-pads.xml: * docs/manual/highlevel-components.xml: * docs/manual/manual.xml: Update for 0.10: s/0.9/0.10/; s/audioscale/audiorsample/; add converters in front of pipelines; remove curly brackets for threads stuff, they no longer exist; use GST_TYPE_FRACTION for framerates; update some pieces of code to 0.10, but there's plenty more to do. * docs/manual/appendix-porting.xml: Expand on asynchroneous state changes; s/0.9/0.10/; mention disappearance of gst_init_get_popt_table() (fixes #322916).
This commit is contained in:
Tim-Philipp Müller
parent
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commit
30cfab511a
8 changed files with 126 additions and 47 deletions
19
ChangeLog
19
ChangeLog
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@ -1,3 +1,22 @@
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2005-12-11 Tim-Philipp Müller <tim at centricular dot net>
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* docs/manual/advanced-dataaccess.xml:
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* docs/manual/appendix-checklist.xml:
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* docs/manual/appendix-programs.xml:
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* docs/manual/basics-pads.xml:
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* docs/manual/highlevel-components.xml:
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* docs/manual/manual.xml:
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Update for 0.10: s/0.9/0.10/; s/audioscale/audiorsample/;
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add converters in front of pipelines; remove curly
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brackets for threads stuff, they no longer exist; use
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GST_TYPE_FRACTION for framerates; update some pieces of
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code to 0.10, but there's plenty more to do.
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* docs/manual/appendix-porting.xml:
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Expand on asynchroneous state changes; s/0.9/0.10/;
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mention disappearance of gst_init_get_popt_table()
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(fixes #322916).
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2005-12-11 Tim-Philipp Müller <tim at centricular dot net>
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* docs/faq/using.xml:
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@ -66,6 +66,7 @@ main (gint argc,
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{
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GMainLoop *loop;
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GstElement *pipeline, *src, *sink, *filter, *csp;
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GstCaps *caps;
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GstPad *pad;
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/* init GStreamer */
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@ -80,15 +81,17 @@ main (gint argc,
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sink = gst_element_factory_make ("xvimagesink", "sink");
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gst_bin_add_many (GST_BIN (pipeline), src, filter, csp, sink, NULL);
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gst_element_link_many (src, filter, csp, sink, NULL);
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g_object_set (G_OBJECT (filter), "caps",
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gst_caps_new_simple ("video/x-raw-rgb",
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filtercaps = gst_caps_new_simple ("video/x-raw-rgb",
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"width", G_TYPE_INT, 384,
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"height", G_TYPE_INT, 288,
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"framerate", G_TYPE_DOUBLE, (gdouble) 25.0,
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"framerate", GST_TYPE_FRACTION, 25, 1,
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"bpp", G_TYPE_INT, 16,
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"depth", G_TYPE_INT, 16,
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"endianness", G_TYPE_INT, G_BYTE_ORDER,
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NULL), NULL);
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NULL);
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g_object_set (G_OBJECT (filter), "caps", filtercaps, NULL);
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gst_caps_unref (filtercaps);
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pad = gst_element_get_pad (src, "src");
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gst_pad_add_buffer_probe (pad, G_CALLBACK (cb_have_data), NULL);
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gst_object_unref (pad);
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@ -105,7 +108,7 @@ main (gint argc,
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}
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<!-- example-end probe.c --></programlisting>
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<para>
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Compare that output with the output of <quote>gst-launch-0.9
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Compare that output with the output of <quote>gst-launch-0.10
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videotestsrc ! xvimagesink</quote>, just so you know what you're
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looking for.
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</para>
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@ -239,7 +242,7 @@ main (gint argc,
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gst_caps_new_simple ("video/x-raw-rgb",
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"width", G_TYPE_INT, 384,
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"height", G_TYPE_INT, 288,
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"framerate", G_TYPE_DOUBLE, (gdouble) 1.0,
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"framerate", GST_TYPE_FRACTION, 1, 1,
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"bpp", G_TYPE_INT, 16,
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"depth", G_TYPE_INT, 16,
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"endianness", G_TYPE_INT, G_BYTE_ORDER,
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@ -105,12 +105,14 @@
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</listitem>
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<listitem>
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<para>
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<option>--gst-debug-no-color</option> will disable color debugging.
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<option>--gst-debug-no-color</option> will disable color debugging
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(you can also set the GST_DEBUG_NO_COLOR environment variable to 1
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if you want to disable colored debug output permanently)
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</para>
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</listitem>
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<listitem>
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<para>
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<option>--gst-debug-disable</option> disables debugging alltogether.
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<option>--gst-debug-disable</option> disables debugging altogether.
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</para>
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</listitem>
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<listitem>
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@ -130,7 +132,7 @@
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applications will find useful. Specifically, those are videoscalers
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(videoscale), colorspace convertors (ffmpegcolorspace), audio format
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convertors and channel resamplers (audioconvert) and audio samplerate
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convertors (audioscale). Those convertors don't do anything when not
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convertors (audioresample). Those convertors don't do anything when not
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required, they will act in passthrough mode. They will activate when
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the hardware doesn't support a specific request, though. All
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applications are recommended to use those elements.
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@ -150,16 +152,20 @@
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<para>
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<command>gst-launch</command> is a simple script-like commandline
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application that can be used to test pipelines. For example, the
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command <command>gst-launch audiotestsrc ! alsasink</command> will run
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command <command>gst-launch audiotestsrc ! audioconvert !
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audio/x-raw-int,channels=2 ! alsasink</command> will run
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a pipeline which generates a sine-wave audio stream and plays it
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to your ALSA audio card. <command>gst-launch</command> also allows
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the use of threads (using curly brackets, so <quote>{</quote>
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and <quote>}</quote>) and bins (using brackets, so <quote>(</quote>
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and <quote>)</quote>). You can use dots to imply padnames on elements,
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the use of threads (will be used automatically as required or as queue
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elements are inserted in the pipeline) and bins (using brackets, so
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<quote>(</quote> and <quote>)</quote>). You can use dots to imply
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padnames on elements,
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or even omit the padname to automatically select a pad. Using
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all this, the pipeline <command>gst-launch filesrc location=file.ogg
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! oggdemux name=d { d. ! theoradec ! ffmpegcolorspace ! xvimagesink
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} { d. ! vorbisdec ! alsasink }</command> will play an Ogg file
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all this, the pipeline
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<command>gst-launch filesrc location=file.ogg ! oggdemux name=d
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d. ! queue ! theoradec ! ffmpegcolorspace ! xvimagesink
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d. ! queue ! vorbisdec ! audioconvert ! audioresample ! alsasink
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</command> will play an Ogg file
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containing a Theora video-stream and a Vorbis audio-stream. You can
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also use autopluggers such as decodebin on the commandline. See the
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manual page of <command>gst-launch</command> for more information.
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@ -1,12 +1,12 @@
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<chapter id="chapter-porting">
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<title>Porting 0.8 applications to 0.9</title>
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<title>Porting 0.8 applications to 0.10</title>
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<para>
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This section of the appendix will discuss shortly what changes to
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applications will be needed to quickly and conveniently port most
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applications from &GStreamer;-0.8 to &GStreamer;-0.9, with references
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applications from &GStreamer;-0.8 to &GStreamer;-0.10, with references
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to the relevant sections in this Application Development Manual
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where needed. With this list, it should be possible to port simple
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applications to &GStreamer;-0.9 in less than a day.
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applications to &GStreamer;-0.10 in less than a day.
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</para>
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<sect1 id="section-porting-objects">
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@ -17,13 +17,15 @@
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Most functions returning an object or an object property have
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been changed to return its own reference rather than a constant
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reference of the one owned by the object itself. The reason for
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this change is primarily threadsafety. This means, effectively,
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this change is primarily thread safety. This means, effectively,
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that return values of functions such as
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<function>gst_element_get_pad ()</function>,
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<function>gst_pad_get_name ()</function> and many more like these
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have to be free'ed or unreferenced after use. Check the API
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references of each function to know for sure whether return
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values should be free'ed or not.
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values should be free'ed or not. It is important that all objects
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derived from GstObject are ref'ed/unref'ed using gst_object_ref()
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and gst_object_unref() respectively (instead of g_object_ref/unref).
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</para>
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</listitem>
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<listitem>
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@ -32,8 +34,8 @@
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of errors, end-of-stream and other similar pipeline events.
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Instead, they should use the <classname>GstBus</classname>, which
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has been discussed in <xref linkend="chapter-bus"/>. The bus will
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take care that the messages will be delivered in the context of
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mainloop, which is almost certainly the application's main thread.
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take care that the messages will be delivered in the context of a
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main loop, which is almost certainly the application's main thread.
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The big advantage of this is that applications no longer need to
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be thread-aware; they don't need to use <function>g_idle_add
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()</function> in the signal handler and do the actual real work
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@ -51,14 +53,27 @@
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</listitem>
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<listitem>
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<para>
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State changes can be delayed; ASYNC.
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State changes can be delayed (ASYNC). Due to the new fully threaded
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nature of GStreamer-0.10, state changes are not always immediate,
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in particular changes including the transition from READY to PAUSED
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state. This means two things in the context of porting applications:
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first of all, it is no longer always possible to do
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<function>gst_element_set_state ()</function> and check for a return
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value of GST_STATE_CHANGE_SUCCESS, as the state change might be
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delayed (ASYNC) and the result will not be known until later. You
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should still check for GST_STATE_CHANGE_FAILURE right away, it is
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just no longer possible to assume that everything that is not SUCCESS
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means failure. Secondly, state changes might not be immediate, so
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your code needs to take that into account. You can wait for a state
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change to complete if you use GST_CLOCK_TIME_NONE as timeout interval
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with <function>gst_element_get_state ()</function>.
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</para>
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</listitem>
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<listitem>
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<para>
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In 0.8, events and queries had to manually be sent to sinks in
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pipelines (unless you were using playbin). This is no longer
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the case in 0.9. In 0.9, queries and events can be sent to
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the case in 0.10. In 0.10, queries and events can be sent to
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toplevel pipelines, and the pipeline will do the dispatching
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internally for you. This means less bookkeeping in your
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application. For a short code example, see <xref
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@ -82,12 +97,13 @@
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</listitem>
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<listitem>
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<para>
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Filtered caps -> caps-filter.
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Filtered caps -> capsfilter element (the pipeline syntax for
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gst-launch has not changed though).
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</para>
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</listitem>
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<listitem>
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<para>
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libgstgconf-0.9.la does not exist. Use the
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libgstgconf-0.10.la does not exist. Use the
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<quote>gconfvideosink</quote> and <quote>gconfaudiosink</quote>
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elements instead, which will do live-updates and require no library
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linking.
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@ -100,6 +116,15 @@
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<quote>pad-added</quote> and <quote>state-changed</quote>.
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</para>
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</listitem>
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<listitem>
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<para>
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<function>gst_init_get_popt_table ()</function> has been removed
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in favour of the new GOption command line option API that was
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added to GLib 2.6. <function>gst_init_get_option_group ()</function>
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is the new GOption-based equivalent to
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<function>gst_init_get_ptop_table ()</function>.
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</para>
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</listitem>
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</itemizedlist>
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</sect1>
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</chapter>
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@ -3,15 +3,6 @@
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<para>
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</para>
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<sect1 id="section-programs-gst-register">
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<title><command>gst-register</command></title>
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<para>
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<command>gst-register</command> is used to rebuild the database of plugins.
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It is used after a new plugin has been added to the system. The plugin database
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can be found, by default, in <filename>/etc/gstreamer/reg.xml</filename>.
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</para>
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</sect1>
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<sect1 id="section-programs-gst-launch">
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<title><command>gst-launch</command></title>
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<para>
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@ -22,15 +13,15 @@
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A simple commandline looks like:
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<screen>
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gst-launch filesrc location=hello.mp3 ! mad ! osssink
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gst-launch filesrc location=hello.mp3 ! mad ! audioresample ! osssink
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</screen>
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A more complex pipeline looks like:
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<screen>
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gst-launch filesrc location=redpill.vob ! mpegdemux name=demux \
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{ demux.audio_00 ! queue ! a52dec ! audioconvert ! audioscale ! osssink } \
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{ demux.video_00 ! queue ! mpeg2dec ! ffmpegcolorspace ! xvmagesink }
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gst-launch filesrc location=redpill.vob ! dvddemux name=demux \
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demux.audio_00 ! queue ! a52dec ! audioconvert ! audioresample ! osssink \
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demux.video_00 ! queue ! mpeg2dec ! ffmpegcolorspace ! xvimagesink
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</screen>
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</para>
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@ -49,6 +40,8 @@ main (int argc, char *argv[])
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{
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GstElement *pipeline;
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GstElement *filesrc;
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GstMessage *msg;
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GstBus *bus;
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GError *error = NULL;
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gst_init (&argc, &argv);
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@ -65,13 +58,46 @@ main (int argc, char *argv[])
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}
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filesrc = gst_bin_get_by_name (GST_BIN (pipeline), "my_filesrc");
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g_object_set (G_OBJECT (filesrc), "location", argv[1], NULL);
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g_object_set (filesrc, "location", argv[1], NULL);
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gst_element_set_state (pipeline, GST_STATE_PLAYING);
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while (gst_bin_iterate (GST_BIN (pipeline)));
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bus = gst_element_get_bus (pipeline);
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/* wait until we either get an EOS or an ERROR message. Note that in a real
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* program you would probably not use gst_bus_poll(), but rather set up an
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* async signal watch on the bus and run a main loop and connect to the
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* bus's signals to catch certain messages or all messages */
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msg = gst_bus_poll (bus, GST_MESSAGE_EOS | GST_MESSAGE_ERROR, -1);
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switch (GST_MESSAGE_TYPE (msg)) {
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case GST_MESSAGE_EOS: {
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g_print ("EOS\n");
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break;
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}
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case GST_MESSAGE_ERROR: {
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GError *err = NULL; /* error to show to users */
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gchar *dbg = NULL; /* additional debug string for developers */
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gst_message_parse_error (msg, &err, &dbg);
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if (err) {
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g_printerr ("ERROR: %s\n", err->message);
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g_error_free (err);
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}
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if (dbg) {
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g_printerr ("[Debug details: %s]\n", dbg);
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g_free (dbg);
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}
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}
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default:
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g_printerr ("Unexpected message of type %d", GST_MESSAGE_TYPE (msg));
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break;
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}
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gst_message_unref (msg);
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gst_element_set_state (pipeline, GST_STATE_NULL);
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gst_object_unref (pipeline);
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gst_object_unref (bus);
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return 0;
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}
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|
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@ -451,7 +451,7 @@ link_pads_with_filter (GstPad *one,
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caps = gst_caps_new_simple ("video/x-raw-yuv",
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"width", G_TYPE_INT, 384,
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"height", G_TYPE_INT, 288,
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"framerate", G_TYPE_DOUBLE, 25.,
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"framerate", GST_TYPE_FRACTION, 25, 1,
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NULL);
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gst_pad_link_filtered (one, other, caps);
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}
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@ -473,12 +473,12 @@ link_pads_with_filter (GstPad *one,
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gst_structure_new ("video/x-raw-yuv",
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"width", G_TYPE_INT, 384,
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"height", G_TYPE_INT, 288,
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"framerate", G_TYPE_DOUBLE, 25.,
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"framerate", GST_TYPE_FRACTION, 25, 1,
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NULL),
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gst_structure_new ("video/x-raw-rgb",
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"width", G_TYPE_INT, 384,
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"height", G_TYPE_INT, 288,
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"framerate", G_TYPE_DOUBLE, 25.,
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"framerate", GST_TYPE_FRACTION, 25, 1,
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NULL),
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NULL);
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|
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|
|
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|
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@ -168,7 +168,7 @@ main (gint argc,
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</itemizedlist>
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<para>
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For convenience, it is possible to test <quote>playbin</quote> on
|
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the commandline, using the command <quote>gst-launch-0.9 playbin
|
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the commandline, using the command <quote>gst-launch-0.10 playbin
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uri=file:///path/to/file</quote>.
|
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</para>
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</sect1>
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|
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|
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@ -257,7 +257,7 @@
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</para>
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<para>
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In addition, we also provide a porting guide which will explain
|
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easily how to port &GStreamer;-0.8 applications to &GStreamer;-0.9.
|
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easily how to port &GStreamer;-0.8 applications to &GStreamer;-0.10.
|
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</para>
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</partintro>
|
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|
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|
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