*
* 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.
*/
/**
* #GstRtpBin is configured with a number of request pads that define the
* functionality that is activated, similar to the #GstRtpSession element.
*
- * To use #GstRtpBin as an RTP receiver, request a recv_rtp_sink_\%d pad. The session
+ * To use #GstRtpBin as an RTP receiver, request a recv_rtp_sink_\%u pad. The session
* number must be specified in the pad name.
- * Data received on the recv_rtp_sink_\%d pad will be processed in the #GstRtpSession
+ * Data received on the recv_rtp_sink_\%u pad will be processed in the #GstRtpSession
* manager and after being validated forwarded on #GstRtpSsrcDemux element. Each
* RTP stream is demuxed based on the SSRC and send to a #GstRtpJitterBuffer. After
* the packets are released from the jitterbuffer, they will be forwarded to a
* #GstRtpPtDemux element. The #GstRtpPtDemux element will demux the packets based
- * on the payload type and will create a unique pad recv_rtp_src_\%d_\%d_\%d on
+ * on the payload type and will create a unique pad recv_rtp_src_\%u_\%u_\%u on
* gstrtpbin with the session number, SSRC and payload type respectively as the pad
* name.
*
- * To also use #GstRtpBin as an RTCP receiver, request a recv_rtcp_sink_\%d pad. The
+ * To also use #GstRtpBin as an RTCP receiver, request a recv_rtcp_sink_\%u pad. The
* session number must be specified in the pad name.
*
* If you want the session manager to generate and send RTCP packets, request
- * the send_rtcp_src_\%d pad with the session number in the pad name. Packet pushed
+ * the send_rtcp_src_\%u pad with the session number in the pad name. Packet pushed
* on this pad contain SR/RR RTCP reports that should be sent to all participants
* in the session.
*
- * To use #GstRtpBin as a sender, request a send_rtp_sink_\%d pad, which will
- * automatically create a send_rtp_src_\%d pad. If the session number is not provided,
+ * To use #GstRtpBin as a sender, request a send_rtp_sink_\%u pad, which will
+ * automatically create a send_rtp_src_\%u pad. If the session number is not provided,
* the pad from the lowest available session will be returned. The session manager will modify the
* SSRC in the RTP packets to its own SSRC and wil forward the packets on the
- * send_rtp_src_\%d pad after updating its internal state.
+ * send_rtp_src_\%u pad after updating its internal state.
*
* The session manager needs the clock-rate of the payload types it is handling
* and will signal the #GstRtpSession::request-pt-map signal when it needs such a
* <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 \
#include "gstrtpsession.h"
#include "gstrtpjitterbuffer.h"
+#include <gst/glib-compat-private.h>
+
GST_DEBUG_CATEGORY_STATIC (gst_rtp_bin_debug);
#define GST_CAT_DEFAULT gst_rtp_bin_debug
/* sink pads */
static GstStaticPadTemplate rtpbin_recv_rtp_sink_template =
-GST_STATIC_PAD_TEMPLATE ("recv_rtp_sink_%d",
+GST_STATIC_PAD_TEMPLATE ("recv_rtp_sink_%u",
GST_PAD_SINK,
GST_PAD_REQUEST,
GST_STATIC_CAPS ("application/x-rtp")
);
static GstStaticPadTemplate rtpbin_recv_rtcp_sink_template =
-GST_STATIC_PAD_TEMPLATE ("recv_rtcp_sink_%d",
+GST_STATIC_PAD_TEMPLATE ("recv_rtcp_sink_%u",
GST_PAD_SINK,
GST_PAD_REQUEST,
GST_STATIC_CAPS ("application/x-rtcp")
);
static GstStaticPadTemplate rtpbin_send_rtp_sink_template =
-GST_STATIC_PAD_TEMPLATE ("send_rtp_sink_%d",
+GST_STATIC_PAD_TEMPLATE ("send_rtp_sink_%u",
GST_PAD_SINK,
GST_PAD_REQUEST,
GST_STATIC_CAPS ("application/x-rtp")
/* src pads */
static GstStaticPadTemplate rtpbin_recv_rtp_src_template =
-GST_STATIC_PAD_TEMPLATE ("recv_rtp_src_%d_%d_%d",
+GST_STATIC_PAD_TEMPLATE ("recv_rtp_src_%u_%u_%u",
GST_PAD_SRC,
GST_PAD_SOMETIMES,
GST_STATIC_CAPS ("application/x-rtp")
);
static GstStaticPadTemplate rtpbin_send_rtcp_src_template =
-GST_STATIC_PAD_TEMPLATE ("send_rtcp_src_%d",
+GST_STATIC_PAD_TEMPLATE ("send_rtcp_src_%u",
GST_PAD_SRC,
GST_PAD_REQUEST,
GST_STATIC_CAPS ("application/x-rtcp")
);
static GstStaticPadTemplate rtpbin_send_rtp_src_template =
-GST_STATIC_PAD_TEMPLATE ("send_rtp_src_%d",
+GST_STATIC_PAD_TEMPLATE ("send_rtp_src_%u",
GST_PAD_SRC,
GST_PAD_SOMETIMES,
GST_STATIC_CAPS ("application/x-rtp")
#define GST_RTP_BIN_GET_PRIVATE(obj) \
(G_TYPE_INSTANCE_GET_PRIVATE ((obj), GST_TYPE_RTP_BIN, GstRtpBinPrivate))
-#define GST_RTP_BIN_LOCK(bin) g_mutex_lock ((bin)->priv->bin_lock)
-#define GST_RTP_BIN_UNLOCK(bin) g_mutex_unlock ((bin)->priv->bin_lock)
+#define GST_RTP_BIN_LOCK(bin) g_mutex_lock (&(bin)->priv->bin_lock)
+#define GST_RTP_BIN_UNLOCK(bin) g_mutex_unlock (&(bin)->priv->bin_lock)
/* lock to protect dynamic callbacks, like pad-added and new ssrc. */
-#define GST_RTP_BIN_DYN_LOCK(bin) g_mutex_lock ((bin)->priv->dyn_lock)
-#define GST_RTP_BIN_DYN_UNLOCK(bin) g_mutex_unlock ((bin)->priv->dyn_lock)
+#define GST_RTP_BIN_DYN_LOCK(bin) g_mutex_lock (&(bin)->priv->dyn_lock)
+#define GST_RTP_BIN_DYN_UNLOCK(bin) g_mutex_unlock (&(bin)->priv->dyn_lock)
/* lock for shutdown */
#define GST_RTP_BIN_SHUTDOWN_LOCK(bin,label) \
struct _GstRtpBinPrivate
{
- GMutex *bin_lock;
+ GMutex bin_lock;
/* lock protecting dynamic adding/removing */
- GMutex *dyn_lock;
+ GMutex dyn_lock;
/* if we are shutting down or not */
gint shutdown;
};
#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.
*
gint64 clock_base;
};
-#define GST_RTP_SESSION_LOCK(sess) g_mutex_lock ((sess)->lock)
-#define GST_RTP_SESSION_UNLOCK(sess) g_mutex_unlock ((sess)->lock)
+#define GST_RTP_SESSION_LOCK(sess) g_mutex_lock (&(sess)->lock)
+#define GST_RTP_SESSION_UNLOCK(sess) g_mutex_unlock (&(sess)->lock)
/* Manages the receiving end of the packets.
*
gulong demux_newpad_sig;
gulong demux_padremoved_sig;
- GMutex *lock;
+ GMutex lock;
/* list of GstRtpBinStream */
GSList *streams;
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 */
GstElement *session, *demux;
GstState target;
- if (!(session = gst_element_factory_make ("gstrtpsession", NULL)))
+ if (!(session = gst_element_factory_make ("rtpsession", NULL)))
goto no_session;
- if (!(demux = gst_element_factory_make ("gstrtpssrcdemux", NULL)))
+ if (!(demux = gst_element_factory_make ("rtpssrcdemux", NULL)))
goto no_demux;
sess = g_new0 (GstRtpBinSession, 1);
- sess->lock = g_mutex_new ();
+ g_mutex_init (&sess->lock);
sess->id = id;
sess->bin = rtpbin;
sess->session = session;
/* ERRORS */
no_session:
{
- g_warning ("gstrtpbin: could not create gstrtpsession element");
+ g_warning ("rtpbin: could not create gstrtpsession element");
return NULL;
}
no_demux:
{
gst_object_unref (session);
- g_warning ("gstrtpbin: could not create gstrtpssrcdemux element");
+ g_warning ("rtpbin: could not create gstrtpssrcdemux element");
return NULL;
}
}
+/* 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_free (sess->lock);
+ g_mutex_clear (&sess->lock);
g_hash_table_destroy (sess->ptmap);
g_free (sess);
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;
* now equating rtptime obtained from RTP-Info,
* where the large time represent the otherwise irrelevant npt/ntp time */
stream->rtp_delta = (GST_SECOND << 28) - rtp_clock_base;
+ } else {
+ clock_base = rtp_clock_base;
}
+ all_sync = TRUE;
for (walk = client->streams; walk; walk = g_slist_next (walk)) {
GstRtpBinStream *ostream = (GstRtpBinStream *) walk->data;
/* arrange to re-sync for each stream upon significant change,
* e.g. post-seek */
- all_sync = (stream->clock_base == clock_base);
+ all_sync = all_sync && (stream->clock_base == clock_base);
stream->clock_base = clock_base;
/* may need init performed above later on, but nothing more to do now */
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;
guint64 clock_base;
guint64 extrtptime;
GstBuffer *buffer;
- GstRTCPBuffer rtcp = { NULL };
+ GstRTCPBuffer rtcp = { NULL, };
bin = stream->bin;
rtpbin = session->bin;
- if (!(buffer = gst_element_factory_make ("gstrtpjitterbuffer", NULL)))
+ if (!(buffer = gst_element_factory_make ("rtpjitterbuffer", NULL)))
goto no_jitterbuffer;
if (!rtpbin->ignore_pt)
- if (!(demux = gst_element_factory_make ("gstrtpptdemux", NULL)))
+ if (!(demux = gst_element_factory_make ("rtpptdemux", NULL)))
goto no_demux;
/* 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);
/* ERRORS */
no_jitterbuffer:
{
- g_warning ("gstrtpbin: could not create gstrtpjitterbuffer element");
+ g_warning ("rtpbin: could not create gstrtpjitterbuffer element");
return NULL;
}
no_demux:
{
gst_object_unref (buffer);
- g_warning ("gstrtpbin: could not create gstrtpptdemux element");
+ g_warning ("rtpbin: could not create gstrtpptdemux element");
return 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>");
static void
gst_rtp_bin_init (GstRtpBin * rtpbin)
{
- gchar *str;
+ gchar *cname;
rtpbin->priv = GST_RTP_BIN_GET_PRIVATE (rtpbin);
- rtpbin->priv->bin_lock = g_mutex_new ();
- rtpbin->priv->dyn_lock = g_mutex_new ();
+ g_mutex_init (&rtpbin->priv->bin_lock);
+ g_mutex_init (&rtpbin->priv->dyn_lock);
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 */
- str = g_strdup_printf ("%s@%s", g_get_user_name (), g_get_host_name ());
+ 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, str,
- "name", G_TYPE_STRING, g_get_real_name (),
- "tool", G_TYPE_STRING, "GStreamer", NULL);
- g_free (str);
+ "cname", G_TYPE_STRING, cname, "tool", G_TYPE_STRING, "GStreamer", NULL);
+ g_free (cname);
}
static void
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);
}
if (rtpbin->sdes)
gst_structure_free (rtpbin->sdes);
- g_mutex_free (rtpbin->priv->bin_lock);
- g_mutex_free (rtpbin->priv->dyn_lock);
+ g_mutex_clear (&rtpbin->priv->bin_lock);
+ g_mutex_clear (&rtpbin->priv->dyn_lock);
G_OBJECT_CLASS (parent_class)->finalize (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;
/* ghost the pad to the parent */
klass = GST_ELEMENT_GET_CLASS (rtpbin);
- templ = gst_element_class_get_pad_template (klass, "recv_rtp_src_%d_%d_%d");
- padname = g_strdup_printf ("recv_rtp_src_%d_%u_%d",
+ templ = gst_element_class_get_pad_template (klass, "recv_rtp_src_%u_%u_%u");
+ padname = g_strdup_printf ("recv_rtp_src_%u_%u_%u",
stream->session->id, stream->ssrc, pt);
gpad = gst_ghost_pad_new_from_template (padname, pad, templ);
g_free (padname);
/* get pad and link */
GST_DEBUG_OBJECT (rtpbin, "linking jitterbuffer RTP");
- padname = g_strdup_printf ("src_%d", ssrc);
+ padname = g_strdup_printf ("src_%u", ssrc);
srcpad = gst_element_get_static_pad (element, padname);
g_free (padname);
sinkpad = gst_element_get_static_pad (stream->buffer, "sink");
gst_object_unref (srcpad);
GST_DEBUG_OBJECT (rtpbin, "linking jitterbuffer RTCP");
- padname = g_strdup_printf ("rtcp_src_%d", ssrc);
+ padname = g_strdup_printf ("rtcp_src_%u", ssrc);
srcpad = gst_element_get_static_pad (element, padname);
g_free (padname);
sinkpad = gst_element_get_request_pad (stream->buffer, "sink_rtcp");
/* ghost the pad to the parent */
klass = GST_ELEMENT_GET_CLASS (rtpbin);
- templ = gst_element_class_get_pad_template (klass, "recv_rtp_src_%d_%d_%d");
- padname = g_strdup_printf ("recv_rtp_src_%d_%u_%d",
+ templ = gst_element_class_get_pad_template (klass, "recv_rtp_src_%u_%u_%u");
+ padname = g_strdup_printf ("recv_rtp_src_%u_%u_%u",
stream->session->id, stream->ssrc, 255);
gpad = gst_ghost_pad_new_from_template (padname, pad, templ);
g_free (padname);
GstPadLinkReturn lres;
/* first get the session number */
- if (name == NULL || sscanf (name, "recv_rtp_sink_%d", &sessid) != 1)
+ if (name == NULL || sscanf (name, "recv_rtp_sink_%u", &sessid) != 1)
goto no_name;
GST_DEBUG_OBJECT (rtpbin, "finding session %d", sessid);
/* ERRORS */
no_name:
{
- g_warning ("gstrtpbin: invalid name given");
+ g_warning ("rtpbin: invalid name given");
return NULL;
}
create_error:
}
pad_failed:
{
- g_warning ("gstrtpbin: failed to get session pad");
+ g_warning ("rtpbin: failed to get session pad");
return NULL;
}
link_failed:
{
- g_warning ("gstrtpbin: failed to link pads");
+ g_warning ("rtpbin: failed to link pads");
return NULL;
}
}
GstPadLinkReturn lres;
/* first get the session number */
- if (name == NULL || sscanf (name, "recv_rtcp_sink_%d", &sessid) != 1)
+ if (name == NULL || sscanf (name, "recv_rtcp_sink_%u", &sessid) != 1)
goto no_name;
GST_DEBUG_OBJECT (rtpbin, "finding session %d", sessid);
/* ERRORS */
no_name:
{
- g_warning ("gstrtpbin: invalid name given");
+ g_warning ("rtpbin: invalid name given");
return NULL;
}
create_error:
}
pad_failed:
{
- g_warning ("gstrtpbin: failed to get session pad");
+ g_warning ("rtpbin: failed to get session pad");
return NULL;
}
link_failed:
{
- g_warning ("gstrtpbin: failed to link pads");
+ g_warning ("rtpbin: failed to link pads");
return NULL;
}
}
GstElementClass *klass;
/* first get the session number */
- if (name == NULL || sscanf (name, "send_rtp_sink_%d", &sessid) != 1)
+ if (name == NULL || sscanf (name, "send_rtp_sink_%u", &sessid) != 1)
goto no_name;
/* get or create session */
/* ghost the new source pad */
klass = GST_ELEMENT_GET_CLASS (rtpbin);
- gname = g_strdup_printf ("send_rtp_src_%d", sessid);
- templ = gst_element_class_get_pad_template (klass, "send_rtp_src_%d");
+ gname = g_strdup_printf ("send_rtp_src_%u", sessid);
+ templ = gst_element_class_get_pad_template (klass, "send_rtp_src_%u");
session->send_rtp_src_ghost =
gst_ghost_pad_new_from_template (gname, session->send_rtp_src, templ);
gst_pad_set_active (session->send_rtp_src_ghost, TRUE);
/* ERRORS */
no_name:
{
- g_warning ("gstrtpbin: invalid name given");
+ g_warning ("rtpbin: invalid name given");
return NULL;
}
create_error:
}
pad_failed:
{
- g_warning ("gstrtpbin: failed to get session pad for session %d", sessid);
+ g_warning ("rtpbin: failed to get session pad for session %d", sessid);
return NULL;
}
no_srcpad:
{
- g_warning ("gstrtpbin: failed to get rtp source pad for session %d",
- sessid);
+ g_warning ("rtpbin: failed to get rtp source pad for session %d", sessid);
return NULL;
}
}
GstRtpBinSession *session;
/* first get the session number */
- if (name == NULL || sscanf (name, "send_rtcp_src_%d", &sessid) != 1)
+ if (name == NULL || sscanf (name, "send_rtcp_src_%u", &sessid) != 1)
goto no_name;
/* get or create session */
/* ERRORS */
no_name:
{
- g_warning ("gstrtpbin: invalid name given");
+ g_warning ("rtpbin: invalid name given");
return NULL;
}
no_session:
{
- g_warning ("gstrtpbin: session with id %d does not exist", sessid);
+ g_warning ("rtpbin: session with id %d does not exist", sessid);
return NULL;
}
pad_failed:
{
- g_warning ("gstrtpbin: failed to get rtcp pad for session %d", sessid);
+ g_warning ("rtpbin: failed to get rtcp pad for session %d", sessid);
return NULL;
}
}
GST_DEBUG_OBJECT (rtpbin, "Trying to request a pad with name %s", pad_name);
/* figure out the template */
- if (templ == gst_element_class_get_pad_template (klass, "recv_rtp_sink_%d")) {
+ if (templ == gst_element_class_get_pad_template (klass, "recv_rtp_sink_%u")) {
result = create_recv_rtp (rtpbin, templ, pad_name);
} else if (templ == gst_element_class_get_pad_template (klass,
- "recv_rtcp_sink_%d")) {
+ "recv_rtcp_sink_%u")) {
result = create_recv_rtcp (rtpbin, templ, pad_name);
} else if (templ == gst_element_class_get_pad_template (klass,
- "send_rtp_sink_%d")) {
+ "send_rtp_sink_%u")) {
result = create_send_rtp (rtpbin, templ, pad_name);
} else if (templ == gst_element_class_get_pad_template (klass,
- "send_rtcp_src_%d")) {
+ "send_rtcp_src_%u")) {
result = create_rtcp (rtpbin, templ, pad_name);
} else
goto wrong_template;
{
g_free (pad_name);
GST_RTP_BIN_UNLOCK (rtpbin);
- g_warning ("gstrtpbin: this is not our template");
+ g_warning ("rtpbin: this is not our template");
return NULL;
}
}
unknown_pad:
{
GST_RTP_BIN_UNLOCK (rtpbin);
- g_warning ("gstrtpbin: %s:%s is not one of our request pads",
+ g_warning ("rtpbin: %s:%s is not one of our request pads",
GST_DEBUG_PAD_NAME (pad));
return;
}
projects / platform / upstream / gstreamer.git / blobdiff