audio: Set pulsestate on element-added event

1 files changed, 41 insertions, 45 deletions

diff --git a/src/audio.c b/src/audio.c
index e0bb6b0..37e746b 100644
--- a/src/audio.c
+++ b/src/audio.c
@@ -15,42 +15,6 @@
 	G_STRINGIFY(CHANNELS) ",rate=" G_STRINGIFY(SAMPLERATE)
 
 static void
-set_pulse_states(const GValue *item, gpointer user_data)
-{
-	GstElement *elm = g_value_get_object(item);
-	struct cmumble_user *user = user_data;
-	GstStructure *props;
-	gchar *name;
-
-	if (g_strcmp0(G_OBJECT_TYPE_NAME(elm), "GstPulseSink") != 0 ||
-	    g_object_class_find_property(G_OBJECT_GET_CLASS(elm),
-					 "stream-properties") == NULL)
-		goto out;
-
-	/* FIXME: Move this into a man-page or so:
-	 * Dear User: Add the following to the pulseaudio configuration:
-	 * load-module module-device-manager "do_routing=1"
-	 * This is to let new join users default to e.g. a headset output.
-	 * Also consider setting device.intended_roles = "phone" for your
-	 * output to be marked as headset (if you dont have a usb headset dev). */
-
-	name = g_strdup_printf("cmumble [%s]", user->name);
-
-	props = gst_structure_new("props",
-				  "application.name", G_TYPE_STRING, name,
-				  "media.role", G_TYPE_STRING, "phone",
-				  NULL);
-
-	g_object_set(elm, "stream-properties", props, NULL);
-	gst_structure_free(props);
-	g_free(name);
-
-out:
-	g_object_unref(G_OBJECT(elm));
-}
-
-
-static void
 add_celt_streamheader(struct cmumble *cm, GstAppSrc *src)
 {
 	GValue streamheader = { 0, }, val = { 0, };
@@ -89,6 +53,44 @@ add_celt_streamheader(struct cmumble *cm, GstAppSrc *src)
 	gst_caps_unref(caps);
 }
 
+static void
+set_pulse_sink_states(GstElement *el, struct cmumble_user *user)
+{
+	GstStructure *props;
+	gchar *name;
+
+	/* FIXME: Move this into a man-page or so:
+	 * Dear User: Add the following to the pulseaudio configuration:
+	 * load-module module-device-manager "do_routing=1"
+	 * This is to let new join users default to e.g. a headset output.
+	 * Also consider setting device.intended_roles = "phone" for your
+	 * output to be marked as headset (if you dont have a usb headset dev). */
+
+	name = g_strdup_printf("cmumble [%s]", user->name);
+
+	props = gst_structure_new("props",
+				  "application.name", G_TYPE_STRING, name,
+				  "media.role", G_TYPE_STRING, "phone",
+				  NULL);
+
+	g_object_set(el, "stream-properties", props, NULL);
+	gst_structure_free(props);
+	g_free(name);
+}
+
+static void
+on_sink_bin_element_added (GstBin *bin, GstElement *el, gpointer user_data)
+{
+	struct cmumble_user *user = user_data;
+
+	if (g_strcmp0(G_OBJECT_TYPE_NAME(el), "GstPulseSink") == 0 &&
+	    g_object_class_find_property(G_OBJECT_GET_CLASS(el),
+					 "stream-properties") != NULL) {
+		set_pulse_sink_states(el, user);
+	}
+}
+
+
 int
 cmumble_audio_create_playback_pipeline(struct cmumble *cm,
 				       struct cmumble_user *user)
@@ -98,7 +100,6 @@ cmumble_audio_create_playback_pipeline(struct cmumble *cm,
 	char *desc = "appsrc name=src format=GST_FORMAT_TIME caps="CELT_CAPS" "
 		     "! celtdec name=dec "
 		     "! audioresample ! audioconvert ! autoaudiosink name=sink";
-	GstIterator *it;
 
 	pipeline = gst_parse_launch(desc, &error);
 	if (error) {
@@ -109,16 +110,11 @@ cmumble_audio_create_playback_pipeline(struct cmumble *cm,
 	user->pipeline = pipeline;
 	user->src = GST_APP_SRC(gst_bin_get_by_name(GST_BIN(pipeline), "src"));
 	add_celt_streamheader(cm, user->src);
-
-	gst_element_set_state(pipeline, GST_STATE_PLAYING);
-
-	/* FIXME: Use a recursive name for sink-actual-sink-pluse instead? like:
-	 * gst_bin_get_by_name(GST_BIN(pipeline), "sink-actual-sink-pulse"); */
 	sink_bin = gst_bin_get_by_name(GST_BIN(pipeline), "sink");
-	it = gst_bin_iterate_sinks(GST_BIN(sink_bin));
-	gst_iterator_foreach(it, set_pulse_states, user);
-	gst_iterator_free(it);
+	g_signal_connect(G_OBJECT(sink_bin), "element-added",
+			 G_CALLBACK(on_sink_bin_element_added), user);
 
+	gst_element_set_state(pipeline, GST_STATE_PLAYING);
 	user->last_sequence = -1;
 
 	return 0;