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https://gitlab.freedesktop.org/gstreamer/gstreamer.git
synced 2024-07-03 05:05:52 +00:00
Handle media bus messages
Handle media bus messages in a custom mainloop and dispatch them to the RTSPMedia objects. Let the default implementation handle some common messages.
This commit is contained in:
Wim Taymans
parent
e1154c92d6
commit
cd29e2a454
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@ -77,7 +77,6 @@ struct _GstRTSPMediaFactory {
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* pay%d to create the streams.
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* @configure: configure the media created with @construct. The default
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* implementation will configure the 'shared' property of the media.
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* @handle_message: Handle a bus message for @media created from @factory.
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*
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* The #GstRTSPMediaFactory class structure.
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*/
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@ -89,9 +88,6 @@ struct _GstRTSPMediaFactoryClass {
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GstElement * (*get_element) (GstRTSPMediaFactory *factory, const GstRTSPUrl *url);
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GstRTSPMedia * (*construct) (GstRTSPMediaFactory *factory, const GstRTSPUrl *url);
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void (*configure) (GstRTSPMediaFactory *factory, GstRTSPMedia *media);
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void (*handle_message) (GstRTSPMediaFactory *factory, GstRTSPMedia *media,
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GstMessage *message);
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};
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GType gst_rtsp_media_factory_get_type (void);
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@ -34,12 +34,16 @@ static void gst_rtsp_media_set_property (GObject *object, guint propid,
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const GValue *value, GParamSpec *pspec);
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static void gst_rtsp_media_finalize (GObject * obj);
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static gpointer do_loop (GstRTSPMediaClass *klass);
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static gboolean default_handle_message (GstRTSPMedia *media, GstMessage *message);
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G_DEFINE_TYPE (GstRTSPMedia, gst_rtsp_media, G_TYPE_OBJECT);
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static void
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gst_rtsp_media_class_init (GstRTSPMediaClass * klass)
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{
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GObjectClass *gobject_class;
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GError *error = NULL;
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gobject_class = G_OBJECT_CLASS (klass);
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@ -50,6 +54,15 @@ gst_rtsp_media_class_init (GstRTSPMediaClass * klass)
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g_object_class_install_property (gobject_class, PROP_SHARED,
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g_param_spec_boolean ("shared", "Shared", "If this media pipeline can be shared",
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DEFAULT_SHARED, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
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klass->context = g_main_context_new ();
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klass->loop = g_main_loop_new (klass->context, TRUE);
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klass->thread = g_thread_create ((GThreadFunc) do_loop, klass, TRUE, &error);
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if (error != NULL) {
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g_critical ("could not start bus thread: %s", error->message);
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}
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klass->handle_message = default_handle_message;
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}
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static void
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@ -57,6 +70,8 @@ gst_rtsp_media_init (GstRTSPMedia * media)
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{
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media->streams = g_array_new (FALSE, TRUE, sizeof (GstRTSPMediaStream *));
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media->complete = FALSE;
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media->is_live = FALSE;
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media->buffering = FALSE;
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}
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static void
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@ -87,6 +102,11 @@ gst_rtsp_media_finalize (GObject * obj)
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}
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g_array_free (media->streams, TRUE);
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if (media->source) {
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g_source_destroy (media->source);
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g_source_unref (media->source);
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}
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if (media->pipeline)
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gst_object_unref (media->pipeline);
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@ -123,6 +143,16 @@ gst_rtsp_media_set_property (GObject *object, guint propid,
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}
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}
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static gpointer
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do_loop (GstRTSPMediaClass *klass)
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{
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g_message ("enter mainloop");
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g_main_loop_run (klass->loop);
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g_message ("exit mainloop");
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return NULL;
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}
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/**
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* gst_rtsp_media_new:
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*
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@ -449,6 +479,23 @@ setup_stream (GstRTSPMediaStream *stream, guint idx, GstRTSPMedia *media)
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return TRUE;
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}
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static void
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unlock_streams (GstRTSPMedia *media)
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{
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guint i, n_streams;
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/* unlock the udp src elements */
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n_streams = gst_rtsp_media_n_streams (media);
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for (i = 0; i < n_streams; i++) {
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GstRTSPMediaStream *stream;
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stream = gst_rtsp_media_get_stream (media, i);
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gst_element_set_locked_state (stream->udpsrc[0], FALSE);
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gst_element_set_locked_state (stream->udpsrc[1], FALSE);
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}
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}
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static void
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collect_media_stats (GstRTSPMedia *media)
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{
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@ -474,6 +521,92 @@ collect_media_stats (GstRTSPMedia *media)
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}
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}
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static gboolean
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default_handle_message (GstRTSPMedia *media, GstMessage *message)
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{
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GstMessageType type;
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type = GST_MESSAGE_TYPE (message);
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switch (type) {
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case GST_MESSAGE_STATE_CHANGED:
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break;
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case GST_MESSAGE_BUFFERING:
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{
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gint percent;
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gst_message_parse_buffering (message, &percent);
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/* no state management needed for live pipelines */
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if (media->is_live)
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break;
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if (percent == 100) {
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/* a 100% message means buffering is done */
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media->buffering = FALSE;
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/* if the desired state is playing, go back */
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if (media->target_state == GST_STATE_PLAYING) {
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g_message ("Buffering done, setting pipeline to PLAYING");
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gst_element_set_state (media->pipeline, GST_STATE_PLAYING);
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}
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else {
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g_message ("Buffering done");
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}
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} else {
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/* buffering busy */
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if (media->buffering == FALSE) {
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if (media->target_state == GST_STATE_PLAYING) {
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/* we were not buffering but PLAYING, PAUSE the pipeline. */
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g_message ("Buffering, setting pipeline to PAUSED ...");
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gst_element_set_state (media->pipeline, GST_STATE_PAUSED);
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}
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else {
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g_message ("Buffering ...");
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}
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}
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media->buffering = TRUE;
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}
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break;
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}
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case GST_MESSAGE_LATENCY:
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{
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gst_bin_recalculate_latency (GST_BIN_CAST (media->pipeline));
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break;
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}
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case GST_MESSAGE_ERROR:
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{
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GError *gerror;
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gchar *debug;
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gst_message_parse_error (message, &gerror, &debug);
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g_warning ("%p: got error %s (%s)", media, gerror->message, debug);
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g_error_free (gerror);
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g_free (debug);
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break;
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}
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default:
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g_message ("%p: got message type %s", media, gst_message_type_get_name (type));
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break;
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}
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return TRUE;
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}
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static gboolean
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bus_message (GstBus *bus, GstMessage *message, GstRTSPMedia *media)
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{
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GstRTSPMediaClass *klass;
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gboolean ret;
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klass = GST_RTSP_MEDIA_GET_CLASS (media);
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if (klass->handle_message)
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ret = klass->handle_message (media, message);
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else
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ret = FALSE;
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return ret;
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}
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/**
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* gst_rtsp_media_prepare:
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* @obj: a #GstRTSPMedia
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@ -488,6 +621,8 @@ gst_rtsp_media_prepare (GstRTSPMedia *media)
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{
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GstStateChangeReturn ret;
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guint i, n_streams;
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GstRTSPMediaClass *klass;
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GstBus *bus;
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if (media->prepared)
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goto was_prepared;
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@ -495,6 +630,7 @@ gst_rtsp_media_prepare (GstRTSPMedia *media)
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g_message ("preparing media %p", media);
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media->pipeline = gst_pipeline_new ("media-pipeline");
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bus = gst_pipeline_get_bus (GST_PIPELINE_CAST (media->pipeline));
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gst_bin_add (GST_BIN_CAST (media->pipeline), media->element);
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@ -515,6 +651,7 @@ gst_rtsp_media_prepare (GstRTSPMedia *media)
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/* first go to PAUSED */
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ret = gst_element_set_state (media->pipeline, GST_STATE_PAUSED);
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media->target_state = GST_STATE_PAUSED;
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switch (ret) {
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case GST_STATE_CHANGE_SUCCESS:
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@ -524,6 +661,7 @@ gst_rtsp_media_prepare (GstRTSPMedia *media)
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case GST_STATE_CHANGE_NO_PREROLL:
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/* we need to go to PLAYING */
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g_message ("live media %p", media);
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media->is_live = TRUE;
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ret = gst_element_set_state (media->pipeline, GST_STATE_PLAYING);
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break;
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case GST_STATE_CHANGE_FAILURE:
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@ -539,18 +677,21 @@ gst_rtsp_media_prepare (GstRTSPMedia *media)
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/* collect stats about the media */
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collect_media_stats (media);
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/* unlock the udp src elements */
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n_streams = gst_rtsp_media_n_streams (media);
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for (i = 0; i < n_streams; i++) {
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GstRTSPMediaStream *stream;
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stream = gst_rtsp_media_get_stream (media, i);
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gst_element_set_locked_state (stream->udpsrc[0], FALSE);
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gst_element_set_locked_state (stream->udpsrc[1], FALSE);
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}
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/* unlock the streams so that they follow the state changes from now on */
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unlock_streams (media);
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g_message ("object %p is prerolled", media);
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/* add the pipeline bus to our custom mainloop */
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bus = gst_pipeline_get_bus (GST_PIPELINE_CAST (media->pipeline));
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media->source = gst_bus_create_watch (bus);
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gst_object_unref (bus);
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g_source_set_callback (media->source, (GSourceFunc) bus_message, media, NULL);
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klass = GST_RTSP_MEDIA_GET_CLASS (media);
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media->id = g_source_attach (media->source, klass->context);
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media->prepared = TRUE;
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return TRUE;
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@ -563,7 +704,33 @@ was_prepared:
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/* ERRORS */
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state_failed:
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{
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GstMessage *message;
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g_message ("state change failed for media %p", media);
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while ((message = gst_bus_pop (bus))) {
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GstMessageType type;
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type = GST_MESSAGE_TYPE (message);
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switch (type) {
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case GST_MESSAGE_ERROR:
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{
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GError *gerror;
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gchar *debug;
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gst_message_parse_error (message, &gerror, &debug);
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g_warning ("%p: got error %s (%s)", media, gerror->message, debug);
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g_error_free (gerror);
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g_free (debug);
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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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gst_message_unref (message);
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}
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unlock_streams (media);
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gst_element_set_state (media->pipeline, GST_STATE_NULL);
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gst_object_unref (bus);
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return FALSE;
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}
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}
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@ -611,6 +778,7 @@ gst_rtsp_media_play (GstRTSPMedia *media, GArray *transports)
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}
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g_message ("playing");
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media->target_state = GST_STATE_PLAYING;
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ret = gst_element_set_state (media->pipeline, GST_STATE_PLAYING);
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return TRUE;
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@ -659,6 +827,7 @@ gst_rtsp_media_pause (GstRTSPMedia *media, GArray *transports)
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}
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g_message ("pause");
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media->target_state = GST_STATE_PAUSED;
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ret = gst_element_set_state (media->pipeline, GST_STATE_PAUSED);
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return TRUE;
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@ -685,6 +854,7 @@ gst_rtsp_media_stop (GstRTSPMedia *media, GArray *transports)
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gst_rtsp_media_pause (media, transports);
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g_message ("stop");
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media->target_state = GST_STATE_NULL;
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ret = gst_element_set_state (media->pipeline, GST_STATE_NULL);
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return TRUE;
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@ -124,6 +124,12 @@ struct _GstRTSPMedia {
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/* the pipeline for the media */
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GstElement *pipeline;
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GSource *source;
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guint id;
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gboolean is_live;
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gboolean buffering;
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GstState target_state;
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/* RTP session manager */
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GstElement *rtpbin;
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@ -135,8 +141,24 @@ struct _GstRTSPMedia {
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GstRTSPTimeRange range;
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};
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/**
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* GstRTSPMediaClass:
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* @context: the main context for dispatching messages
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* @loop: the mainloop for message.
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* @thread: the thread dispatching messages.
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* @handle_message: handle a message
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*
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* The RTSP media class
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*/
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struct _GstRTSPMediaClass {
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GObjectClass parent_class;
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/* thread for the mainloop */
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GMainContext *context;
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GMainLoop *loop;
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GThread *thread;
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gboolean (*handle_message) (GstRTSPMedia *media, GstMessage *message);
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};
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GType gst_rtsp_media_get_type (void);
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