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249 lines
8.3 KiB
Markdown
249 lines
8.3 KiB
Markdown
---
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title: Your first application
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...
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# Your first application
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This chapter will summarize everything you've learned in the previous
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chapters. It describes all aspects of a simple GStreamer application,
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including initializing libraries, creating elements, packing elements
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together in a pipeline and playing this pipeline. By doing all this, you
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will be able to build a simple Ogg/Vorbis audio player.
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# Hello world
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We're going to create a simple first application, a simple Ogg/Vorbis
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command-line audio player. For this, we will use only standard GStreamer
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components. The player will read a file specified on the command-line.
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Let's get started\!
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We've learned, in [Initializing GStreamer](manual-init.md), that the
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first thing to do in your application is to initialize GStreamer by
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calling `gst_init ()`. Also, make sure that the application includes
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`gst/gst.h` so all function names and objects are properly defined. Use
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`#include
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<gst/gst.h>` to do that.
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Next, you'll want to create the different elements using
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`gst_element_factory_make ()`. For an Ogg/Vorbis audio player, we'll
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need a source element that reads files from a disk. GStreamer includes
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this element under the name “filesrc”. Next, we'll need something to
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parse the file and decode it into raw audio. GStreamer has two elements
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for this: the first parses Ogg streams into elementary streams (video,
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audio) and is called “oggdemux”. The second is a Vorbis audio decoder,
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it's conveniently called “vorbisdec”. Since “oggdemux” creates dynamic
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pads for each elementary stream, you'll need to set a “pad-added” event
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handler on the “oggdemux” element, like you've learned in [Dynamic (or
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sometimes) pads](manual-pads.md#dynamic-or-sometimes-pads), to link the
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Ogg demuxer and the Vorbis decoder elements together. At last, we'll
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also need an audio output element, we will use “autoaudiosink”, which
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automatically detects your audio device.
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The last thing left to do is to add all elements into a container
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element, a `GstPipeline`, and wait until we've played the whole song.
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We've previously learned how to add elements to a container bin in
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[Bins](manual-bins.md), and we've learned about element states in
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[Element States](manual-elements.md#element-states). We will also
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attach a message handler to the pipeline bus so we can retrieve errors
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and detect the end-of-stream.
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Let's now add all the code together to get our very first audio player:
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``` c
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#include <gst/gst.h>
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#include <glib.h>
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static gboolean
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bus_call (GstBus *bus,
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GstMessage *msg,
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gpointer data)
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{
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GMainLoop *loop = (GMainLoop *) data;
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switch (GST_MESSAGE_TYPE (msg)) {
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case GST_MESSAGE_EOS:
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g_print ("End of stream\n");
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g_main_loop_quit (loop);
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break;
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case GST_MESSAGE_ERROR: {
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gchar *debug;
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GError *error;
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gst_message_parse_error (msg, &error, &debug);
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g_free (debug);
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g_printerr ("Error: %s\n", error->message);
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g_error_free (error);
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g_main_loop_quit (loop);
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break;
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}
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default:
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break;
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}
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return TRUE;
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}
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static void
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on_pad_added (GstElement *element,
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GstPad *pad,
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gpointer data)
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{
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GstPad *sinkpad;
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GstElement *decoder = (GstElement *) data;
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/* We can now link this pad with the vorbis-decoder sink pad */
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g_print ("Dynamic pad created, linking demuxer/decoder\n");
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sinkpad = gst_element_get_static_pad (decoder, "sink");
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gst_pad_link (pad, sinkpad);
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gst_object_unref (sinkpad);
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}
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int
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main (int argc,
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char *argv[])
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{
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GMainLoop *loop;
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GstElement *pipeline, *source, *demuxer, *decoder, *conv, *sink;
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GstBus *bus;
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guint bus_watch_id;
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/* Initialisation */
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gst_init (&argc, &argv);
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loop = g_main_loop_new (NULL, FALSE);
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/* Check input arguments */
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if (argc != 2) {
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g_printerr ("Usage: %s <Ogg/Vorbis filename>\n", argv[0]);
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return -1;
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}
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/* Create gstreamer elements */
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pipeline = gst_pipeline_new ("audio-player");
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source = gst_element_factory_make ("filesrc", "file-source");
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demuxer = gst_element_factory_make ("oggdemux", "ogg-demuxer");
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decoder = gst_element_factory_make ("vorbisdec", "vorbis-decoder");
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conv = gst_element_factory_make ("audioconvert", "converter");
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sink = gst_element_factory_make ("autoaudiosink", "audio-output");
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if (!pipeline || !source || !demuxer || !decoder || !conv || !sink) {
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g_printerr ("One element could not be created. Exiting.\n");
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return -1;
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}
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/* Set up the pipeline */
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/* we set the input filename to the source element */
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g_object_set (G_OBJECT (source), "location", argv[1], NULL);
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/* we add a message handler */
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bus = gst_pipeline_get_bus (GST_PIPELINE (pipeline));
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bus_watch_id = gst_bus_add_watch (bus, bus_call, loop);
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gst_object_unref (bus);
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/* we add all elements into the pipeline */
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/* file-source | ogg-demuxer | vorbis-decoder | converter | alsa-output */
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gst_bin_add_many (GST_BIN (pipeline),
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source, demuxer, decoder, conv, sink, NULL);
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/* we link the elements together */
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/* file-source -> ogg-demuxer ~> vorbis-decoder -> converter -> alsa-output */
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gst_element_link (source, demuxer);
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gst_element_link_many (decoder, conv, sink, NULL);
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g_signal_connect (demuxer, "pad-added", G_CALLBACK (on_pad_added), decoder);
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/* note that the demuxer will be linked to the decoder dynamically.
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The reason is that Ogg may contain various streams (for example
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audio and video). The source pad(s) will be created at run time,
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by the demuxer when it detects the amount and nature of streams.
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Therefore we connect a callback function which will be executed
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when the "pad-added" is emitted.*/
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/* Set the pipeline to "playing" state*/
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g_print ("Now playing: %s\n", argv[1]);
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gst_element_set_state (pipeline, GST_STATE_PLAYING);
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/* Iterate */
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g_print ("Running...\n");
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g_main_loop_run (loop);
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/* Out of the main loop, clean up nicely */
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g_print ("Returned, stopping playback\n");
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gst_element_set_state (pipeline, GST_STATE_NULL);
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g_print ("Deleting pipeline\n");
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gst_object_unref (GST_OBJECT (pipeline));
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g_source_remove (bus_watch_id);
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g_main_loop_unref (loop);
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return 0;
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}
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```
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We now have created a complete pipeline. We can visualise the pipeline
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as follows:
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# Compiling and Running helloworld.c
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To compile the helloworld example, use: `gcc -Wall
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helloworld.c -o helloworld
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$(pkg-config --cflags --libs gstreamer-1.0)`. GStreamer makes use of
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`pkg-config` to get compiler and linker flags needed to compile this
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application.
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If you're running a non-standard installation (ie. you've installed
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GStreamer from source yourself instead of using pre-built packages),
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make sure the `PKG_CONFIG_PATH` environment variable is set to the
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correct location (`$libdir/pkgconfig`).
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In the unlikely case that you are using an uninstalled GStreamer setup
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(ie. gst-uninstalled), you will need to use libtool to build the hello
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world program, like this: `libtool --mode=link gcc -Wall
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helloworld.c -o helloworld
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$(pkg-config --cflags --libs gstreamer-1.0)`.
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You can run this example application with `./helloworld
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file.ogg`. Substitute `file.ogg` with your favourite Ogg/Vorbis file.
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# Conclusion
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This concludes our first example. As you see, setting up a pipeline is
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very low-level but powerful. You will see later in this manual how you
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can create a more powerful media player with even less effort using
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higher-level interfaces. We will discuss all that in [Higher-level
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interfaces for GStreamer applications](manual-highlevel.md). We will
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first, however, go more in-depth into more advanced GStreamer internals.
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It should be clear from the example that we can very easily replace the
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“filesrc” element with some other element that reads data from a
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network, or some other data source element that is better integrated
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with your desktop environment. Also, you can use other decoders and
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parsers/demuxers to support other media types. You can use another audio
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sink if you're not running Linux, but Mac OS X, Windows or FreeBSD, or
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you can instead use a filesink to write audio files to disk instead of
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playing them back. By using an audio card source, you can even do audio
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capture instead of playback. All this shows the reusability of GStreamer
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elements, which is its greatest advantage.
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