*
* You should have received a copy of the GNU Library General Public
* License along with this library; if not, write to the
- * Free Software Foundation, Inc., 59 Temple Place - Suite 330,
- * Boston, MA 02111-1307, USA.
+ * Free Software Foundation, Inc., 51 Franklin St, Fifth Floor,
+ * Boston, MA 02110-1301, USA.
*/
/**
* <refsect2>
* <title>Example pipelines</title>
* |[
- * gst-launch udpsrc port=5000 caps="application/x-rtp, ..." ! .recv_rtp_sink_0 \
+ * gst-launch-1.0 udpsrc port=5000 caps="application/x-rtp, ..." ! .recv_rtp_sink_0 \
* gstrtpbin ! rtptheoradepay ! theoradec ! xvimagesink
* ]| Receive RTP data from port 5000 and send to the session 0 in gstrtpbin.
* |[
- * gst-launch gstrtpbin name=rtpbin \
- * v4l2src ! ffmpegcolorspace ! ffenc_h263 ! rtph263ppay ! rtpbin.send_rtp_sink_0 \
+ * gst-launch-1.0 gstrtpbin name=rtpbin \
+ * v4l2src ! videoconvert ! ffenc_h263 ! rtph263ppay ! rtpbin.send_rtp_sink_0 \
* rtpbin.send_rtp_src_0 ! udpsink port=5000 \
* rtpbin.send_rtcp_src_0 ! udpsink port=5001 sync=false async=false \
* udpsrc port=5005 ! rtpbin.recv_rtcp_sink_0 \
* as soon as possible and do not participate in preroll, sync=false and
* async=false is configured on udpsink
* |[
- * gst-launch -v gstrtpbin name=rtpbin \
+ * gst-launch-1.0 -v gstrtpbin name=rtpbin \
* udpsrc caps="application/x-rtp,media=(string)video,clock-rate=(int)90000,encoding-name=(string)H263-1998" \
* port=5000 ! rtpbin.recv_rtp_sink_0 \
* rtpbin. ! rtph263pdepay ! ffdec_h263 ! xvimagesink \
};
#define DEFAULT_LATENCY_MS 200
+#define DEFAULT_DROP_ON_LATENCY FALSE
#define DEFAULT_SDES NULL
#define DEFAULT_DO_LOST FALSE
#define DEFAULT_IGNORE_PT FALSE
{
PROP_0,
PROP_LATENCY,
+ PROP_DROP_ON_LATENCY,
PROP_SDES,
PROP_DO_LOST,
PROP_IGNORE_PT,
GstRtpBinSession * session);
static void payload_type_change (GstElement * element, guint pt,
GstRtpBinSession * session);
+static void remove_recv_rtp (GstRtpBin * rtpbin, GstRtpBinSession * session);
+static void remove_recv_rtcp (GstRtpBin * rtpbin, GstRtpBinSession * session);
+static void remove_send_rtp (GstRtpBin * rtpbin, GstRtpBinSession * session);
+static void remove_rtcp (GstRtpBin * rtpbin, GstRtpBinSession * session);
static void free_client (GstRtpBinClient * client, GstRtpBin * bin);
-static void free_stream (GstRtpBinStream * stream);
+static void free_stream (GstRtpBinStream * stream, GstRtpBin * bin);
/* Manages the RTP stream for one SSRC.
*
GstRtpBinSession * session)
{
GstRtpBinStream *stream = NULL;
+ GstRtpBin *rtpbin;
+
+ rtpbin = session->bin;
+
+ GST_RTP_BIN_LOCK (rtpbin);
GST_RTP_SESSION_LOCK (session);
if ((stream = find_stream_by_ssrc (session, ssrc)))
GST_RTP_SESSION_UNLOCK (session);
if (stream)
- free_stream (stream);
+ free_stream (stream, rtpbin);
+
+ GST_RTP_BIN_UNLOCK (rtpbin);
}
/* create a session with the given id. Must be called with RTP_BIN_LOCK */
}
}
+/* called with RTP_BIN_LOCK */
static void
free_session (GstRtpBinSession * sess, GstRtpBin * bin)
{
- GSList *client_walk;
-
GST_DEBUG_OBJECT (bin, "freeing session %p", sess);
gst_element_set_locked_state (sess->demux, TRUE);
gst_element_set_state (sess->demux, GST_STATE_NULL);
gst_element_set_state (sess->session, GST_STATE_NULL);
- if (sess->recv_rtp_sink != NULL) {
- gst_element_release_request_pad (sess->session, sess->recv_rtp_sink);
- gst_object_unref (sess->recv_rtp_sink);
- }
- if (sess->recv_rtp_src != NULL)
- gst_object_unref (sess->recv_rtp_src);
- if (sess->recv_rtcp_sink != NULL) {
- gst_element_release_request_pad (sess->session, sess->recv_rtcp_sink);
- gst_object_unref (sess->recv_rtcp_sink);
- }
- if (sess->sync_src != NULL)
- gst_object_unref (sess->sync_src);
- if (sess->send_rtp_sink != NULL) {
- gst_element_release_request_pad (sess->session, sess->send_rtp_sink);
- gst_object_unref (sess->send_rtp_sink);
- }
- if (sess->send_rtp_src != NULL)
- gst_object_unref (sess->send_rtp_src);
- if (sess->send_rtcp_src != NULL) {
- gst_element_release_request_pad (sess->session, sess->send_rtcp_src);
- gst_object_unref (sess->send_rtcp_src);
- }
+ remove_recv_rtp (bin, sess);
+ remove_recv_rtcp (bin, sess);
+ remove_send_rtp (bin, sess);
+ remove_rtcp (bin, sess);
gst_bin_remove (GST_BIN_CAST (bin), sess->session);
gst_bin_remove (GST_BIN_CAST (bin), sess->demux);
- /* remove any references in bin->clients to the streams in sess->streams */
- client_walk = bin->clients;
- while (client_walk) {
- GSList *client_node = client_walk;
- GstRtpBinClient *client = (GstRtpBinClient *) client_node->data;
- GSList *stream_walk = client->streams;
-
- while (stream_walk) {
- GSList *stream_node = stream_walk;
- GstRtpBinStream *stream = (GstRtpBinStream *) stream_node->data;
- GSList *inner_walk;
-
- stream_walk = g_slist_next (stream_walk);
-
- for (inner_walk = sess->streams; inner_walk;
- inner_walk = g_slist_next (inner_walk)) {
- if ((GstRtpBinStream *) inner_walk->data == stream) {
- client->streams = g_slist_delete_link (client->streams, stream_node);
- --client->nstreams;
- break;
- }
- }
- }
- client_walk = g_slist_next (client_walk);
-
- g_assert ((client->streams && client->nstreams > 0) || (!client->streams
- && client->streams == 0));
- if (client->nstreams == 0) {
- free_client (client, bin);
- bin->clients = g_slist_delete_link (bin->clients, client_node);
- }
- }
-
- g_slist_foreach (sess->streams, (GFunc) free_stream, NULL);
+ g_slist_foreach (sess->streams, (GFunc) free_stream, bin);
g_slist_free (sess->streams);
g_mutex_clear (&sess->lock);
clock_time = gst_clock_get_time (clock);
if (bin->use_pipeline_clock) {
- ntpns = clock_time;
+ ntpns = clock_time - base_time;
} else {
GTimeVal current;
static void
stream_set_ts_offset (GstRtpBin * bin, GstRtpBinStream * stream,
- gint64 ts_offset)
+ gint64 ts_offset, gboolean check)
{
gint64 prev_ts_offset;
"ts-offset %" G_GINT64_FORMAT ", prev %" G_GINT64_FORMAT
", diff: %" G_GINT64_FORMAT, ts_offset, prev_ts_offset, diff);
- /* only change diff when it changed more than 4 milliseconds. This
- * compensates for rounding errors in NTP to RTP timestamp
- * conversions */
- if (ABS (diff) > 4 * GST_MSECOND) {
- if (ABS (diff) < (3 * GST_SECOND)) {
- g_object_set (stream->buffer, "ts-offset", ts_offset, NULL);
- } else {
+ if (check) {
+ /* only change diff when it changed more than 4 milliseconds. This
+ * compensates for rounding errors in NTP to RTP timestamp
+ * conversions */
+ if (ABS (diff) < 4 * GST_MSECOND) {
+ GST_DEBUG_OBJECT (bin, "offset too small, ignoring");
+ return;
+ }
+ if (ABS (diff) > (3 * GST_SECOND)) {
GST_WARNING_OBJECT (bin, "offset unusually large, ignoring");
+ return;
}
- } else {
- GST_DEBUG_OBJECT (bin, "offset too small, ignoring");
}
+ g_object_set (stream->buffer, "ts-offset", ts_offset, NULL);
}
GST_DEBUG_OBJECT (bin, "stream SSRC %08x, delta %" G_GINT64_FORMAT,
stream->ssrc, ts_offset);
/* combine to get the final diff to apply to the running_time */
stream->rt_delta = rtdiff - ntpdiff;
- stream_set_ts_offset (bin, stream, stream->rt_delta);
+ stream_set_ts_offset (bin, stream, stream->rt_delta, FALSE);
} else {
gint64 min, rtp_min, clock_base = stream->clock_base;
gboolean all_sync, use_rtp;
else
ts_offset = ostream->rt_delta - min;
- stream_set_ts_offset (bin, ostream, ts_offset);
+ stream_set_ts_offset (bin, ostream, ts_offset, TRUE);
}
}
return;
/* configure latency and packet lost */
g_object_set (buffer, "latency", rtpbin->latency_ms, NULL);
+ g_object_set (buffer, "drop-on-latency", rtpbin->drop_on_latency, NULL);
g_object_set (buffer, "do-lost", rtpbin->do_lost, NULL);
g_object_set (buffer, "mode", rtpbin->buffer_mode, NULL);
}
}
+/* called with RTP_BIN_LOCK */
static void
-free_stream (GstRtpBinStream * stream)
+free_stream (GstRtpBinStream * stream, GstRtpBin * bin)
{
- GstRtpBinSession *session;
+ GSList *clients, *next_client;
- session = stream->session;
+ GST_DEBUG_OBJECT (bin, "freeing stream %p", stream);
if (stream->demux) {
g_signal_handler_disconnect (stream->demux, stream->demux_newpad_sig);
if (stream->demux)
g_signal_handler_disconnect (stream->demux, stream->demux_padremoved_sig);
- gst_bin_remove (GST_BIN_CAST (session->bin), stream->buffer);
+ gst_bin_remove (GST_BIN_CAST (bin), stream->buffer);
if (stream->demux)
- gst_bin_remove (GST_BIN_CAST (session->bin), stream->demux);
+ gst_bin_remove (GST_BIN_CAST (bin), stream->demux);
+
+ for (clients = bin->clients; clients; clients = next_client) {
+ GstRtpBinClient *client = (GstRtpBinClient *) clients->data;
+ GSList *streams, *next_stream;
+
+ next_client = g_slist_next (clients);
+ for (streams = client->streams; streams; streams = next_stream) {
+ GstRtpBinStream *ostream = (GstRtpBinStream *) streams->data;
+
+ next_stream = g_slist_next (streams);
+
+ if (ostream == stream) {
+ client->streams = g_slist_delete_link (client->streams, streams);
+ /* If this was the last stream belonging to this client,
+ * clean up the client. */
+ if (--client->nstreams == 0) {
+ bin->clients = g_slist_delete_link (bin->clients, clients);
+ free_client (client, bin);
+ break;
+ }
+ }
+ }
+ }
g_free (stream);
}
G_MAXUINT, DEFAULT_LATENCY_MS,
G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
+ g_object_class_install_property (gobject_class, PROP_DROP_ON_LATENCY,
+ g_param_spec_boolean ("drop-on-latency",
+ "Drop buffers when maximum latency is reached",
+ "Tells the jitterbuffer to never exceed the given latency in size",
+ DEFAULT_DROP_ON_LATENCY, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
+
/**
* GstRtpBin::request-pt-map:
* @rtpbin: the object which received the signal
g_object_class_install_property (gobject_class, PROP_USE_PIPELINE_CLOCK,
g_param_spec_boolean ("use-pipeline-clock", "Use pipeline clock",
- "Use the pipeline clock to set the NTP time in the RTCP SR messages",
+ "Use the pipeline running-time to set the NTP time in the RTCP SR messages",
DEFAULT_USE_PIPELINE_CLOCK,
G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
/**
/**
* GstRtpBin::ntp-sync:
*
- * Synchronize received streams to the NTP clock. When the NTP clock is shared
- * between the receivers and the senders (such as when using ntpd) this option
- * can be used to synchronize receivers on multiple machines.
+ * Set the NTP time from the sender reports as the running-time on the
+ * buffers. When both the sender and receiver have sychronized
+ * running-time, i.e. when the clock and base-time is shared
+ * between the receivers and the and the senders, this option can be
+ * used to synchronize receivers on multiple machines.
*
* Since: 0.10.21
*/
gst_element_class_add_pad_template (gstelement_class,
gst_static_pad_template_get (&rtpbin_send_rtp_src_template));
- gst_element_class_set_details_simple (gstelement_class, "RTP Bin",
+ gst_element_class_set_static_metadata (gstelement_class, "RTP Bin",
"Filter/Network/RTP",
"Real-Time Transport Protocol bin",
"Wim Taymans <wim.taymans@gmail.com>");
rtpbin->latency_ms = DEFAULT_LATENCY_MS;
rtpbin->latency_ns = DEFAULT_LATENCY_MS * GST_MSECOND;
+ rtpbin->drop_on_latency = DEFAULT_DROP_ON_LATENCY;
rtpbin->do_lost = DEFAULT_DO_LOST;
rtpbin->ignore_pt = DEFAULT_IGNORE_PT;
rtpbin->ntp_sync = DEFAULT_NTP_SYNC;
rtpbin->use_pipeline_clock = DEFAULT_USE_PIPELINE_CLOCK;
/* some default SDES entries */
- cname = g_strdup_printf ("user%u@x-%u.net", g_random_int (), g_random_int ());
+ cname = g_strdup_printf ("user%u@host-%x", g_random_int (), g_random_int ());
rtpbin->sdes = gst_structure_new ("application/x-rtp-source-sdes",
"cname", G_TYPE_STRING, cname, "tool", G_TYPE_STRING, "GStreamer", NULL);
g_free (cname);
rtpbin = GST_RTP_BIN (object);
+ GST_RTP_BIN_LOCK (rtpbin);
GST_DEBUG_OBJECT (object, "freeing sessions");
g_slist_foreach (rtpbin->sessions, (GFunc) free_session, rtpbin);
g_slist_free (rtpbin->sessions);
rtpbin->sessions = NULL;
- GST_DEBUG_OBJECT (object, "freeing clients");
- g_slist_foreach (rtpbin->clients, (GFunc) free_client, rtpbin);
- g_slist_free (rtpbin->clients);
- rtpbin->clients = NULL;
+ GST_RTP_BIN_UNLOCK (rtpbin);
G_OBJECT_CLASS (parent_class)->dispose (object);
}
/* propagate the property down to the jitterbuffer */
gst_rtp_bin_propagate_property_to_jitterbuffer (rtpbin, "latency", value);
break;
+ case PROP_DROP_ON_LATENCY:
+ GST_RTP_BIN_LOCK (rtpbin);
+ rtpbin->drop_on_latency = g_value_get_boolean (value);
+ GST_RTP_BIN_UNLOCK (rtpbin);
+ /* propagate the property down to the jitterbuffer */
+ gst_rtp_bin_propagate_property_to_jitterbuffer (rtpbin,
+ "drop-on-latency", value);
+ break;
case PROP_SDES:
gst_rtp_bin_set_sdes_struct (rtpbin, g_value_get_boxed (value));
break;
g_value_set_uint (value, rtpbin->latency_ms);
GST_RTP_BIN_UNLOCK (rtpbin);
break;
+ case PROP_DROP_ON_LATENCY:
+ GST_RTP_BIN_LOCK (rtpbin);
+ g_value_set_boolean (value, rtpbin->drop_on_latency);
+ GST_RTP_BIN_UNLOCK (rtpbin);
+ break;
case PROP_SDES:
g_value_take_boxed (value, gst_rtp_bin_get_sdes_struct (rtpbin));
break;
projects / platform / upstream / gstreamer.git / blobdiff