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-rw-r--r--src/audio.c168
-rw-r--r--src/audio.h10
2 files changed, 151 insertions, 27 deletions
diff --git a/src/audio.c b/src/audio.c
index e40b1c2..99d9cbe 100644
--- a/src/audio.c
+++ b/src/audio.c
@@ -8,6 +8,7 @@
#define SAMPLERATE 48000
#define FRAMESIZE 480 /* SAMPLERATE/100 */
#define CHANNELS 1
+#define BUFFER_TIME (gst_util_uint64_scale_int(1, GST_SECOND, 100))
#define CELT_CAPS "audio/x-celt,channels=" G_STRINGIFY(CHANNELS) "," \
"rate=" G_STRINGIFY(SAMPLERATE) ",frame-size=" G_STRINGIFY(FRAMESIZE)
#define AUDIO_CAPS "audio/x-raw,format=S16LE,channels=" \
@@ -47,7 +48,7 @@ cmumble_audio_push(struct cmumble *cm, struct cmumble_user *user,
* been received out of order at the server? */
if (user->last_sequence < 0 || sequence == 0 ||
- sequence < (user->last_sequence + 1)) {
+ sequence < (user->last_sequence + 1)) {
GST_BUFFER_FLAG_SET(gstbuf, GST_BUFFER_FLAG_DISCONT);
GST_BUFFER_FLAG_SET(gstbuf, GST_BUFFER_FLAG_RESYNC);
time = now - base;
@@ -77,7 +78,7 @@ cmumble_audio_push(struct cmumble *cm, struct cmumble_user *user,
GST_BUFFER_DTS(gstbuf) = now - base;
GST_BUFFER_PTS(gstbuf) = time;
- GST_BUFFER_DURATION(gstbuf) = gst_util_uint64_scale_int (1, GST_SECOND, 100);
+ GST_BUFFER_DURATION(gstbuf) = BUFFER_TIME;
user->last_time_end = time + GST_BUFFER_DURATION(gstbuf);
user->last_sequence = sequence;
@@ -86,20 +87,53 @@ cmumble_audio_push(struct cmumble *cm, struct cmumble_user *user,
}
static GstFlowReturn
+send_queued_celt_buffers(struct cmumble *cm)
+{
+ uint8_t data[1024];
+ uint32_t written = 0, pos = 0;
+ mumble_udptunnel_t tunnel;
+ GstSample *sample;
+ GstBuffer *buf;
+ int i;
+
+ if (g_queue_is_empty(cm->audio.buffer_queue))
+ return GST_FLOW_ERROR;
+
+ data[pos++] = (udp_voice_celt_alpha << 5) | (udp_normal_talking);
+ encode_varint(&data[pos], &written, cm->sequence, sizeof(data)-pos);
+ pos += written;
+
+ for (i = 0; !g_queue_is_empty(cm->audio.buffer_queue); ++i) {
+ sample = g_queue_pop_head(cm->audio.buffer_queue);
+ buf = gst_sample_get_buffer(sample);
+
+ data[pos] = gst_buffer_get_size(buf) & 0x7F;
+ if (!g_queue_is_empty(cm->audio.buffer_queue))
+ data[pos] |= 0x80;
+ pos += 1;
+ gst_buffer_extract(buf, 0, &data[pos], gst_buffer_get_size(buf));
+ pos += gst_buffer_get_size(buf);
+ gst_sample_unref(sample);
+ }
+
+ cm->sequence += i;
+
+ cmumble_init_udptunnel(&tunnel);
+ tunnel.packet.data = data;
+ tunnel.packet.len = pos;
+ cmumble_send_udptunnel(cm, &tunnel);
+
+ return GST_FLOW_OK;
+}
+
+static GstFlowReturn
pull_buffer(GstAppSink *sink, gpointer user_data)
{
struct cmumble *cm = user_data;
GstSample *sample;
GstBuffer *buf;
- uint8_t data[1024];
- uint32_t write = 0, pos = 0;
- mumble_udptunnel_t tunnel;
+ GstClockTime *silence;
- /* FIXME: Make this more generic/disable pulling
- * the pipeline completely if not connected?
- */
- if (cm->con.conn == NULL)
- return GST_FLOW_OK;
sample = gst_app_sink_pull_sample(cm->audio.sink);
if (sample == NULL)
@@ -112,27 +146,42 @@ pull_buffer(GstAppSink *sink, gpointer user_data)
return GST_FLOW_OK;
}
+ /* FIXME: Make this more generic/disable pulling
+ * the pipeline completely if not connected?
+ */
+ if (cm->con.conn == NULL) {
+ gst_sample_unref(sample);
+ return GST_FLOW_OK;
+ }
+
if (gst_buffer_get_size(buf) > 127) {
g_printerr("error: unexpected buffer size\n");
gst_sample_unref(sample);
return GST_FLOW_ERROR;
}
- data[pos++] = (udp_voice_celt_alpha << 5) | (udp_normal_talking);
+ if (cm->audio.last_time < GST_BUFFER_PTS(buf) - BUFFER_TIME) {
+ if (!g_queue_is_empty(cm->audio.buffer_queue))
+ send_queued_celt_buffers(cm);
+ cm->sequence = 0;
+ }
+ cm->audio.last_time = GST_BUFFER_PTS(buf);
- encode_varint(&data[pos], &write, ++cm->sequence, sizeof(data)-pos);
- pos += write;
+ silence = g_queue_peek_head(cm->audio.silence_timestamps);
+ while (silence && GST_BUFFER_PTS(buf) > *silence) {
+ g_queue_remove(cm->audio.silence_timestamps, silence);
+ g_free(silence);
+ silence = g_queue_peek_head(cm->audio.silence_timestamps);
+ }
- data[pos++] = 0x00 /*: 0x80 */ | (gst_buffer_get_size(buf) & 0x7F);
- gst_buffer_extract(buf, 0, &data[pos], gst_buffer_get_size(buf));
- pos += gst_buffer_get_size(buf);
+ g_queue_push_tail(cm->audio.buffer_queue, sample);
- gst_sample_unref(sample);
+ if (silence && *silence == (GST_BUFFER_PTS(buf) + BUFFER_TIME))
+ return send_queued_celt_buffers(cm);
- cmumble_init_udptunnel(&tunnel);
- tunnel.packet.data = data;
- tunnel.packet.len = pos;
- cmumble_send_udptunnel(cm, &tunnel);
+ /* FIXME: This should not be hardcoded, but derived from bitrate */
+ if (g_queue_get_length(cm->audio.buffer_queue) == 4)
+ return send_queued_celt_buffers(cm);
return GST_FLOW_OK;
}
@@ -164,14 +213,60 @@ GstAppSinkCallbacks sink_cbs = {
.new_sample = new_sample
};
+static void
+handle_cutter_message(struct cmumble *cm, GstMessage *message)
+{
+ const GstStructure *s;
+ gboolean above;
+ GstClockTime *time;
+
+ s = gst_message_get_structure(message);
+ if (!gst_structure_get_boolean(s, "above", &above))
+ return;
+
+ /* We are only intrested in below state */
+ if (above)
+ return;
+
+ time = g_new(GstClockTime, 1);
+ if (!time)
+ return;
+
+ if (!gst_structure_get_clock_time(s, "timestamp", time))
+ return;
+
+ if (*time == cm->audio.last_time + BUFFER_TIME)
+ send_queued_celt_buffers(cm);
+ else
+ g_queue_push_tail(cm->audio.silence_timestamps, time);
+}
+
+static gboolean
+record_pipe_bus_message(GstBus *bus, GstMessage *message, gpointer data)
+{
+ struct cmumble *cm = data;
+
+ switch (GST_MESSAGE_TYPE(message)) {
+ case GST_MESSAGE_ELEMENT:
+ if (GST_MESSAGE_SRC(message) == GST_OBJECT(cm->audio.cutter))
+ handle_cutter_message(cm, message);
+ default:
+ break;
+ }
+
+ return TRUE;
+}
+
static int
setup_recording_gst_pipeline(struct cmumble *cm)
{
- GstElement *pipeline, *cutter, *sink;
+ GstElement *pipeline, *sink;
GError *error = NULL;
+ GstBus *bus;
char *desc = "autoaudiosrc name=src ! cutter name=cutter ! "
"audioresample ! audioconvert ! "AUDIO_CAPS" ! "
- "celtenc ! appsink name=sink caps="CELT_CAPS;
+ "celtenc name=enc perfect-timestamp=true hard-resync=true" " ! "
+ "appsink name=sink caps="CELT_CAPS;
pipeline = gst_parse_launch(desc, &error);
if (error) {
@@ -182,16 +277,33 @@ setup_recording_gst_pipeline(struct cmumble *cm)
cm->audio.sink = GST_APP_SINK(sink);
cm->audio.record_pipeline = pipeline;
- cutter = gst_bin_get_by_name(GST_BIN(pipeline), "cutter");
+ cm->audio.src = gst_bin_get_by_name(GST_BIN(pipeline), "src");
+
+ cm->audio.cutter = gst_bin_get_by_name(GST_BIN(pipeline), "cutter");
/* FIXME: The threshold should be configurable. */
- g_object_set(G_OBJECT(cutter),
+ g_object_set(G_OBJECT(cm->audio.cutter),
"threshold_dB", -45.0, "leaky", TRUE, NULL);
gst_app_sink_set_callbacks(cm->audio.sink, &sink_cbs, cm, NULL);
- gst_app_sink_set_drop(cm->audio.sink, FALSE);;
+ gst_app_sink_set_drop(cm->audio.sink, TRUE);
+
+ bus = gst_pipeline_get_bus(GST_PIPELINE(pipeline));
+ cm->audio.bus_watch_id =
+ gst_bus_add_watch(bus, record_pipe_bus_message, cm);
+ g_object_unref(bus);
+
+ cm->audio.buffer_queue = g_queue_new();
+ if (!cm->audio.buffer_queue)
+ return -1;
+ cm->audio.silence_timestamps = g_queue_new();
+ if (!cm->audio.silence_timestamps)
+ return -1;
gst_element_set_state(pipeline, GST_STATE_PLAYING);
+
+ cm->audio.enc = gst_bin_get_by_name(GST_BIN(pipeline), "enc");
+
cm->sequence = 0;
return 0;
@@ -336,7 +448,9 @@ cmumble_audio_init(struct cmumble *cm)
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);
return 0;
}
-
diff --git a/src/audio.h b/src/audio.h
index 8e47ca9..a54f269 100644
--- a/src/audio.h
+++ b/src/audio.h
@@ -13,11 +13,21 @@
struct cmumble_audio {
GstElement *record_pipeline;
GstAppSink *sink;
+ GstElement *src;
+ GstElement *cutter;
+ GstElement *enc;
+
+ guint bus_watch_id;
+
+ GstClockTime last_time;
guint8 celt_header_packet[sizeof(CELTHeader)];
CELTHeader celt_header;
CELTMode *celt_mode;
+ GQueue *buffer_queue;
+ GQueue *silence_timestamps;
+
gint32 celt_bitstream_version;
};
generated by cgit (git 2.34.1) at 2024-11-22 00:39:43 +0000