1 files changed, 247 insertions, 0 deletions
diff --git a/src/audio_old.c b/src/audio_old.c
new file mode 100644
index 0000000..cf18c78
--- /dev/null
+++ b/src/audio_old.c
@@ -0,0 +1,247 @@
+#include "audio.h"
+#include "varint.h"
+#include "cmumble.h"
+#include <string.h>
+
+#define SAMPLERATE 48000
+#define CHANNELS 1
+
+void
+cmumble_audio_push(struct cmumble *cm, struct cmumble_user *user,
+		   const guint8 *data, gsize size)
+{
+	GstBuffer *gstbuf;
+
+	gstbuf = gst_app_buffer_new(g_memdup(data, size), size, g_free, NULL);
+	gst_app_src_push_buffer(user->src, gstbuf);
+}
+
+static GstFlowReturn
+pull_buffer(GstAppSink *sink, gpointer user_data)
+{
+	struct cmumble *cm = user_data;
+	GstBuffer *buf;
+	uint8_t data[1024];
+	uint32_t write = 0, pos = 0;
+	mumble_udptunnel_t tunnel;
+	static int seq = 0;
+
+	/* FIXME: Make this more generic/disable pulling
+	 * the pipeline completely if not connected?
+	 */
+	if (cm->con.conn == NULL)
+		return GST_FLOW_OK;
+
+	buf = gst_app_sink_pull_buffer(cm->audio.sink);
+
+	if (++seq <= 2) {
+		gst_buffer_unref(buf);
+		return GST_FLOW_OK;
+	}
+	if (GST_BUFFER_SIZE(buf) > 127) {
+		g_printerr("GOT TOO BIG BUFFER\n");
+		return GST_FLOW_ERROR;
+	}
+
+	data[pos++] = (udp_voice_celt_alpha << 5) | (udp_normal_talking);
+
+	encode_varint(&data[pos], &write, ++cm->sequence, sizeof(data)-pos);
+	pos += write;
+
+	data[pos++] = 0x00 /*: 0x80 */ | (GST_BUFFER_SIZE(buf) & 0x7F);
+	memcpy(&data[pos], GST_BUFFER_DATA(buf), GST_BUFFER_SIZE(buf));
+	pos += GST_BUFFER_SIZE(buf);
+
+	gst_buffer_unref(buf);
+
+	cmumble_init_udptunnel(&tunnel);
+	tunnel.packet.data = data;
+	tunnel.packet.len = pos;
+	cmumble_send_udptunnel(cm, &tunnel);
+
+	return GST_FLOW_OK;
+}
+
+static gboolean
+idle(gpointer user_data)
+{
+	struct cmumble *cm = user_data;
+	GstAppSink *sink;
+
+	while ((sink = g_async_queue_try_pop(cm->async_queue)) != NULL)
+		pull_buffer(sink, cm);
+
+	return FALSE;
+}
+
+static GstFlowReturn
+new_buffer(GstAppSink *sink, gpointer user_data)
+{
+	struct cmumble *cm = user_data;
+
+	g_async_queue_push(cm->async_queue, sink);
+	g_idle_add(idle, cm);
+
+	return GST_FLOW_OK;
+}
+
+static int
+setup_recording_gst_pipeline(struct cmumble *cm)
+{
+	GstElement *pipeline, *cutter, *sink;
+	GError *error = NULL;
+	GstCaps *caps;
+
+	char *desc = "autoaudiosrc ! cutter name=cutter ! audioresample ! audioconvert ! "
+		"audio/x-raw-int,channels=1,depth=16,rate=48000,signed=TRUE,width=16 ! "
+		"celtenc ! appsink name=sink";
+
+	pipeline = gst_parse_launch(desc, &error);
+	if (error) {
+		g_printerr("Failed to create pipeline: %s\n", error->message);
+		return -1;
+	}
+	sink = gst_bin_get_by_name(GST_BIN(pipeline), "sink");
+	cm->audio.sink = GST_APP_SINK(sink);
+	cm->audio.record_pipeline = pipeline;
+
+	cutter = gst_bin_get_by_name(GST_BIN(pipeline), "cutter");
+	g_object_set(G_OBJECT(cutter),
+		     "threshold_dB", -45.0, "leaky", TRUE, NULL);
+
+	gst_app_sink_set_emit_signals(cm->audio.sink, TRUE);
+	gst_app_sink_set_drop(cm->audio.sink, FALSE);;
+	g_signal_connect(sink, "new-buffer", G_CALLBACK(new_buffer), cm);
+
+	caps = gst_caps_new_simple("audio/x-celt",
+				   "rate", G_TYPE_INT, SAMPLERATE,
+				   "channels", G_TYPE_INT, 1,
+				   "frame-size", G_TYPE_INT, SAMPLERATE/100,
+				   NULL);
+	gst_app_sink_set_caps(cm->audio.sink, caps);
+	gst_caps_unref(caps);
+
+	gst_element_set_state(pipeline, GST_STATE_PLAYING);
+
+	cm->sequence = 0;
+
+	return 0;
+}
+
+static void
+set_pulse_states(gpointer data, gpointer user_data)
+{
+	GstElement *elm = data;
+	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));
+}
+
+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 ! celtdec ! 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"));
+
+	/* Important! */
+	gst_base_src_set_live(GST_BASE_SRC(user->src), TRUE); 
+	gst_base_src_set_do_timestamp(GST_BASE_SRC(user->src), TRUE);
+	gst_base_src_set_format(GST_BASE_SRC(user->src), GST_FORMAT_TIME);
+
+	gst_app_src_set_stream_type(user->src, GST_APP_STREAM_TYPE_STREAM); 
+
+	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);
+
+	/* Setup Celt Decoder */
+	cmumble_audio_push(cm, user,
+			   cm->audio.celt_header_packet, sizeof(CELTHeader));
+	/* fake vorbiscomment buffer */
+	cmumble_audio_push(cm, user, NULL, 0);
+
+	return 0;
+}
+
+static int
+setup_playback_gst_pipeline(struct cmumble *cm)
+{
+	cm->audio.celt_mode = celt_mode_create(SAMPLERATE,
+					       SAMPLERATE / 100, NULL);
+
+#ifdef HAVE_CELT_071
+	celt_header_init(&cm->audio.celt_header, cm->audio.celt_mode, CHANNELS);
+#else
+	celt_header_init(&cm->audio.celt_header, cm->audio.celt_mode, SAMPLERATE/100, CHANNELS);
+#endif
+	celt_header_to_packet(&cm->audio.celt_header,
+			      cm->audio.celt_header_packet, sizeof(CELTHeader));
+
+	celt_mode_info(cm->audio.celt_mode, CELT_GET_BITSTREAM_VERSION,
+		       &cm->audio.celt_bitstream_version);
+
+	return 0;
+}
+
+int
+cmumble_audio_init(struct cmumble *cm)
+{
+	if (setup_playback_gst_pipeline(cm) < 0)
+		return -1;
+
+	if (setup_recording_gst_pipeline(cm) < 0)
+		return -1;
+
+	return 0;
+}
+
+int
+cmumble_audio_fini(struct cmumble *cm)
+{
+
+	return 0;
+}
+