audio: Try to create playback pipelines ondemand

2 files changed, 153 insertions, 117 deletions

diff --git a/src/audio.c b/src/audio.c
index b532870..e0bb6b0 100644
--- a/src/audio.c
+++ b/src/audio.c
@@ -14,6 +14,116 @@
 #define AUDIO_CAPS "audio/x-raw,format=S16LE,channels=" \
 	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, };
+	GstTagList *tags;
+	GstBuffer *buf[2];
+	GstStructure *s;
+	GstCaps *caps;
+	gint i;
+
+	buf[0] = gst_buffer_new_allocate(NULL, sizeof(CELTHeader), NULL);
+	gst_buffer_fill(buf[0], 0, cm->audio.celt_header_packet,
+			sizeof(CELTHeader));
+
+	tags = gst_tag_list_new_empty();
+	buf[1] = gst_tag_list_to_vorbiscomment_buffer(tags, NULL, 0, "mumble");
+	gst_tag_list_unref(tags);
+
+	g_value_init(&streamheader, GST_TYPE_ARRAY);
+	for (i = 0; i < G_N_ELEMENTS(buf); ++i) {
+		GST_BUFFER_FLAG_SET(buf[i], GST_BUFFER_FLAG_HEADER);
+		GST_BUFFER_OFFSET(buf[i]) = 0;
+		GST_BUFFER_OFFSET_END(buf[i]) = 0;
+		g_value_init(&val, GST_TYPE_BUFFER);
+		gst_value_take_buffer(&val, buf[i]);
+		gst_value_array_append_value(&streamheader, &val);
+		g_value_unset(&val);
+	}
+
+	caps = gst_app_src_get_caps(src);
+	caps = gst_caps_make_writable(caps);
+	s = gst_caps_get_structure(caps, 0);
+	gst_structure_set_value(s, "streamheader", &streamheader);
+	g_value_unset(&streamheader);
+
+	gst_app_src_set_caps(src, caps);
+	gst_caps_unref(caps);
+}
+
+int
+cmumble_audio_create_playback_pipeline(struct cmumble *cm,
+				       struct cmumble_user *user)
+{
+	GstElement *pipeline, *sink_bin;
+	GError *error = NULL;
+	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) {
+		g_printerr("Failed to create pipeline: %s\n", error->message);
+		return -1;
+	}
+
+	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);
+
+	user->last_sequence = -1;
+
+	return 0;
+}
+
 void
 cmumble_audio_push(struct cmumble *cm, struct cmumble_user *user,
 		   const guint8 *data, gsize size, gint64 sequence)
@@ -23,6 +133,10 @@ cmumble_audio_push(struct cmumble *cm, struct cmumble_user *user,
 	GstClockTime time = 0;
 	GstClockTime base, now = 0;
 
+	if (user->src == NULL)
+		if (cmumble_audio_create_playback_pipeline(cm, user) < 0)
+			return;
+
 	if (cm->verbose)
 		g_print("%s: sequence: %ld\n", __func__, sequence);
 
@@ -320,115 +434,6 @@ setup_recording_gst_pipeline(struct cmumble *cm)
 	return 0;
 }
 
-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, };
-	GstTagList *tags;
-	GstBuffer *buf[2];
-	GstStructure *s;
-	GstCaps *caps;
-	gint i;
-
-	buf[0] = gst_buffer_new_allocate(NULL, sizeof(CELTHeader), NULL);
-	gst_buffer_fill(buf[0], 0, cm->audio.celt_header_packet,
-			sizeof(CELTHeader));
-
-	tags = gst_tag_list_new_empty();
-	buf[1] = gst_tag_list_to_vorbiscomment_buffer(tags, NULL, 0, "mumble");
-	gst_tag_list_unref(tags);
-
-	g_value_init(&streamheader, GST_TYPE_ARRAY);
-	for (i = 0; i < G_N_ELEMENTS(buf); ++i) {
-		GST_BUFFER_FLAG_SET(buf[i], GST_BUFFER_FLAG_HEADER);
-		GST_BUFFER_OFFSET(buf[i]) = 0;
-		GST_BUFFER_OFFSET_END(buf[i]) = 0;
-		g_value_init(&val, GST_TYPE_BUFFER);
-		gst_value_take_buffer(&val, buf[i]);
-		gst_value_array_append_value(&streamheader, &val);
-		g_value_unset(&val);
-	}
-
-	caps = gst_app_src_get_caps(src);
-	caps = gst_caps_make_writable(caps);
-	s = gst_caps_get_structure(caps, 0);
-	gst_structure_set_value(s, "streamheader", &streamheader);
-	g_value_unset(&streamheader);
-
-	gst_app_src_set_caps(src, caps);
-	gst_caps_unref(caps);
-}
-
-int
-cmumble_audio_create_playback_pipeline(struct cmumble *cm,
-				       struct cmumble_user *user)
-{
-	GstElement *pipeline, *sink_bin;
-	GError *error = NULL;
-	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) {
-		g_printerr("Failed to create pipeline: %s\n", error->message);
-		return -1;
-	}
-
-	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);
-
-	user->last_sequence = -1;
-
-	return 0;
-}
-
 static int
 setup_playback_gst_pipeline(struct cmumble *cm)
 {
@@ -456,12 +461,48 @@ cmumble_audio_init(struct cmumble *cm)
 	return 0;
 }
 
+static void
+destroy_record_pipeline(struct cmumble *cm)
+{
+	if (cm->audio.record_pipeline == NULL)
+		return;
+
+	g_source_remove(cm->audio.bus_watch_id);
+	if (cm->audio.silence_timestamps)
+		g_queue_free_full(cm->audio.silence_timestamps, g_free);
+	if (cm->audio.silence_timestamps)
+		g_queue_free_full(cm->audio.buffer_queue, (GDestroyNotify) gst_sample_unref);
+
+	gst_element_set_state(cm->audio.record_pipeline, GST_STATE_NULL);
+	gst_object_unref(GST_OBJECT(cm->audio.record_pipeline));
+	gst_object_unref(GST_OBJECT(cm->audio.sink));
+	gst_object_unref(GST_OBJECT(cm->audio.src));
+	gst_object_unref(GST_OBJECT(cm->audio.cutter));
+}
+
+static void
+destroy_playback_pipelines(struct cmumble *cm)
+{
+	struct cmumble_user *user = NULL;
+	GList *l;
+
+	for (l = cm->users; l; l = l->next) {
+		user = l->data;
+		if (user->pipeline == NULL || user->src == NULL)
+			continue;
+		gst_element_set_state(user->pipeline, GST_STATE_NULL);
+		gst_object_unref(GST_OBJECT(user->pipeline));
+		user->pipeline = NULL;
+		gst_object_unref(GST_OBJECT(user->src));
+		user->src = NULL;
+	}
+}
+
 int
 cmumble_audio_fini(struct cmumble *cm)
 {
-	g_source_remove(cm->audio.bus_watch_id);
-	g_queue_free_full(cm->audio.silence_timestamps, g_free);
-	g_queue_free_full(cm->audio.buffer_queue, (GDestroyNotify) gst_sample_unref);
+	destroy_record_pipeline(cm);
+	destroy_playback_pipelines(cm);
 
 	return 0;
 }
diff --git a/src/cmumble.c b/src/cmumble.c
index 228f28b..8267834 100644
--- a/src/cmumble.c
+++ b/src/cmumble.c
@@ -197,11 +197,6 @@ recv_user_state(mumble_user_state_t *state, struct cmumble *cm)
 	if (cm->session == user->session)
 		cm->user = user;
 
-	/* FIXME: Rather than doing this ugly check by name here,
-	 * we should rather create the pipeline ondemand?
-	 */
-	if (g_strcmp0(user->name, cm->user_name) != 0)
-		cmumble_audio_create_playback_pipeline(cm, user);
 	if (cm->verbose)
 		g_print("receive user: %s\n", user->name);
 	cm->users = g_list_prepend(cm->users, user);