Timestamp playback buffers with appropriate flags

 * GST_BUFFER_FLAG_DISCONT helps to play immediately after a pause
   (i.e. no talking)
 TODO: Not sure GST_BUFFER_FLAG_RESYNC does really help, but
       from the decoumentation it does what we want.

4 files changed, 69 insertions, 10 deletions

diff --git a/src/audio.c b/src/audio.c
index 01a5dab..e40b1c2 100644
--- a/src/audio.c
+++ b/src/audio.c
@@ -15,13 +15,73 @@
 
 void
 cmumble_audio_push(struct cmumble *cm, struct cmumble_user *user,
-		   const guint8 *data, gsize size)
+		   const guint8 *data, gsize size, gint64 sequence)
 {
 	GstBuffer *gstbuf;
+	GstClock *clock;
+	GstClockTime time = 0;
+	GstClockTime base, now = 0;
+
+	if (cm->verbose)
+		g_print("%s: sequence: %ld\n", __func__, sequence);
+
+	base = gst_element_get_base_time(GST_ELEMENT(user->src));
+
+	clock = gst_element_get_clock(GST_ELEMENT(user->src));
+	if (clock) {
+		now = gst_clock_get_time(clock);
+		g_object_unref(clock);
+	}
+
+	/* FIXME: What to do when sequence is a bad value?
+	 * e.g to little in value, to be uptodate?
+	 *  - just drop?
+	 *  - enqueue as now?
+	 */
 
 	gstbuf = gst_buffer_new_wrapped(g_memdup(data, size), size);
 	GST_BUFFER_FLAG_SET(gstbuf, GST_BUFFER_FLAG_LIVE);
 
+	/* Asume packets are in order, since we're using tcp tunnel only atm.
+	 * FIXME: This assumption is probably wrong, since the packets may have
+	 *        been received out of order at the server? */
+
+	if (user->last_sequence < 0 || sequence == 0 ||
+		   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;
+		if (cm->verbose)
+			g_print("%s: set time to now\n", __func__);
+	} else if (sequence >= user->last_sequence + 1) {
+		gint64 num = sequence - (user->last_sequence + 1);
+		time = user->last_time_end;
+		if (num > 0) {
+			time += gst_util_uint64_scale_int(num, GST_SECOND, 100);
+			GST_BUFFER_FLAG_SET(gstbuf, GST_BUFFER_FLAG_DISCONT);
+		}
+		if (cm->verbose)
+			g_print("%s: set time by sequence: %lu, now: %lu\n",
+				__func__, time, now - base);
+
+	}
+
+	if (time < (now - base)) {
+		GST_BUFFER_FLAG_SET(gstbuf, GST_BUFFER_FLAG_DISCONT);
+		GST_BUFFER_FLAG_SET(gstbuf, GST_BUFFER_FLAG_RESYNC);
+		time = now - base;
+		if (cm->verbose)
+			g_print("%s: time is in the past, setting to now\n",
+				__func__);
+	}
+
+	GST_BUFFER_DTS(gstbuf) = now - base;
+	GST_BUFFER_PTS(gstbuf) = time;
+	GST_BUFFER_DURATION(gstbuf) = gst_util_uint64_scale_int (1, GST_SECOND, 100);
+
+	user->last_time_end = time + GST_BUFFER_DURATION(gstbuf);
+	user->last_sequence = sequence;
+
 	gst_app_src_push_buffer(user->src, gstbuf);
 }
 
@@ -123,6 +183,7 @@ setup_recording_gst_pipeline(struct cmumble *cm)
 	cm->audio.record_pipeline = pipeline;
 
 	cutter = gst_bin_get_by_name(GST_BIN(pipeline), "cutter");
+	/* FIXME: The threshold should be configurable. */
 	g_object_set(G_OBJECT(cutter),
 		     "threshold_dB", -45.0, "leaky", TRUE, NULL);
 
@@ -229,15 +290,8 @@ 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"));
-
-	/* 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);
 	add_celt_streamheader(cm, user->src);
 
-	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:
@@ -247,6 +301,8 @@ cmumble_audio_create_playback_pipeline(struct cmumble *cm,
 	gst_iterator_foreach(it, set_pulse_states, user);
 	gst_iterator_free(it);
 
+	user->last_sequence = -1;
+
 	return 0;
 }
 
diff --git a/src/audio.h b/src/audio.h
index 2fd86c4..8e47ca9 100644
--- a/src/audio.h
+++ b/src/audio.h
@@ -36,6 +36,6 @@ cmumble_audio_create_playback_pipeline(struct cmumble *cm,
 
 void
 cmumble_audio_push(struct cmumble *cm, struct cmumble_user *user,
-		   const guint8 *data, gsize size);
+		   const guint8 *data, gsize size, gint64 sequence);
 
 #endif /* _AUDIO_H_ */
diff --git a/src/cmumble.c b/src/cmumble.c
index 977e30a..1ca18bb 100644
--- a/src/cmumble.c
+++ b/src/cmumble.c
@@ -51,7 +51,7 @@ recv_udp_tunnel(mumble_udptunnel_t *tunnel, struct cmumble *cm)
 		if (frame_len < 2)
 			break;
 
-		cmumble_audio_push(cm, user, &data[pos], frame_len);
+		cmumble_audio_push(cm, user, &data[pos], frame_len, sequence);
 
 		pos += frame_len;
 		sequence++;
diff --git a/src/cmumble.h b/src/cmumble.h
index b796d50..3997c8a 100644
--- a/src/cmumble.h
+++ b/src/cmumble.h
@@ -47,6 +47,9 @@ struct cmumble_user {
 
 	GstElement *pipeline;
 	GstAppSrc *src;
+
+	gint64 last_sequence;
+	guint64 last_time_end;
 };
 
 struct cmumble_channel {