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examples: webrtc: mp-sendrecv: add bus handler
Without this bus handler, messages posted to the bus will keep a ref to their source elements, preventing them from being disposed. Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3219>
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1 changed files with 38 additions and 0 deletions
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@ -119,6 +119,39 @@ get_string_from_json_object (JsonObject * object)
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return text;
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}
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static gboolean
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bus_watch_cb (GstBus * bus, GstMessage * message, gpointer user_data)
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{
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switch (GST_MESSAGE_TYPE (message)) {
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case GST_MESSAGE_ERROR:
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{
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GError *error = NULL;
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gchar *debug = NULL;
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gst_message_parse_error (message, &error, &debug);
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g_error ("Error on bus: %s (debug: %s)", error->message, debug);
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g_error_free (error);
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g_free (debug);
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break;
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}
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case GST_MESSAGE_WARNING:
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{
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GError *error = NULL;
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gchar *debug = NULL;
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gst_message_parse_warning (message, &error, &debug);
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g_warning ("Warning on bus: %s (debug: %s)", error->message, debug);
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g_error_free (error);
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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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return G_SOURCE_CONTINUE;
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}
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static void
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handle_media_stream (GstPad * pad, GstElement * pipe, const char *convert_name,
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const char *sink_name)
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@ -413,6 +446,7 @@ start_pipeline (void)
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{
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GstStateChangeReturn ret;
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GError *error = NULL;
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GstBus *bus = NULL;
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/* NOTE: webrtcbin currently does not support dynamic addition/removal of
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* streams, so we use a separate webrtcbin for each peer, but all of them are
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@ -428,6 +462,10 @@ start_pipeline (void)
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goto err;
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}
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bus = gst_pipeline_get_bus (GST_PIPELINE (pipeline));
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gst_bus_add_watch (bus, bus_watch_cb, NULL);
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gst_object_unref (bus);
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gst_print ("Starting pipeline, not transmitting yet\n");
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ret = gst_element_set_state (GST_ELEMENT (pipeline), GST_STATE_PLAYING);
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if (ret == GST_STATE_CHANGE_FAILURE)
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