*
* 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
#define DEFAULT_AUTOREMOVE FALSE
#define DEFAULT_BUFFER_MODE RTP_JITTER_BUFFER_MODE_SLAVE
#define DEFAULT_USE_PIPELINE_CLOCK FALSE
+#define DEFAULT_RTCP_SYNC GST_RTP_BIN_RTCP_SYNC_ALWAYS
#define DEFAULT_RTCP_SYNC_INTERVAL 0
enum
{
PROP_0,
PROP_LATENCY,
+ PROP_DROP_ON_LATENCY,
PROP_SDES,
PROP_DO_LOST,
PROP_IGNORE_PT,
PROP_NTP_SYNC,
+ PROP_RTCP_SYNC,
PROP_RTCP_SYNC_INTERVAL,
PROP_AUTOREMOVE,
PROP_BUFFER_MODE,
PROP_LAST
};
+enum
+{
+ GST_RTP_BIN_RTCP_SYNC_ALWAYS,
+ GST_RTP_BIN_RTCP_SYNC_INITIAL,
+ GST_RTP_BIN_RTCP_SYNC_RTP
+};
+
+#define GST_RTP_BIN_RTCP_SYNC_TYPE (gst_rtp_bin_rtcp_sync_get_type())
+static GType
+gst_rtp_bin_rtcp_sync_get_type (void)
+{
+ static GType rtcp_sync_type = 0;
+ static const GEnumValue rtcp_sync_types[] = {
+ {GST_RTP_BIN_RTCP_SYNC_ALWAYS, "always", "always"},
+ {GST_RTP_BIN_RTCP_SYNC_INITIAL, "initial", "initial"},
+ {GST_RTP_BIN_RTCP_SYNC_RTP, "rtp-info", "rtp-info"},
+ {0, NULL, NULL},
+ };
+
+ if (!rtcp_sync_type) {
+ rtcp_sync_type = g_enum_register_static ("GstRTCPSync", rtcp_sync_types);
+ }
+ return rtcp_sync_type;
+}
+
/* helper objects */
typedef struct _GstRtpBinSession GstRtpBinSession;
typedef struct _GstRtpBinStream GstRtpBinStream;
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.
*
gboolean have_sync;
/* mapping to local RTP and NTP time */
gint64 rt_delta;
+ gint64 rtp_delta;
+ /* base rtptime in gst time */
+ 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);
* lip-sync */
stream->have_sync = FALSE;
stream->rt_delta = 0;
+ stream->rtp_delta = 0;
+ stream->clock_base = -100 * GST_SECOND;
}
}
GST_RTP_BIN_UNLOCK (rtpbin);
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);
static void
gst_rtp_bin_associate (GstRtpBin * bin, GstRtpBinStream * stream, guint8 len,
guint8 * data, guint64 ntptime, guint64 last_extrtptime,
- guint64 base_rtptime, guint64 base_time, guint clock_rate)
+ guint64 base_rtptime, guint64 base_time, guint clock_rate,
+ gint64 rtp_clock_base)
{
GstRtpBinClient *client;
gboolean created;
stream->ssrc, client, client->cname);
}
+ if (!GST_CLOCK_TIME_IS_VALID (last_extrtptime)) {
+ GST_DEBUG_OBJECT (bin, "invalidated sync data");
+ if (bin->rtcp_sync == GST_RTP_BIN_RTCP_SYNC_RTP) {
+ /* we don't need that data, so carry on,
+ * but make some values look saner */
+ last_extrtptime = base_rtptime;
+ } else {
+ /* nothing we can do with this data in this case */
+ GST_DEBUG_OBJECT (bin, "bailing out");
+ return;
+ }
+ }
+
/* Take the extended rtptime we found in the SR packet and map it to the
* local rtptime. The local rtp time is used to construct timestamps on the
* buffers so we will calculate what running_time corresponds to the RTP
GST_DEBUG_OBJECT (bin,
"base %" G_GUINT64_FORMAT ", extrtptime %" G_GUINT64_FORMAT
- ", local RTP %" G_GUINT64_FORMAT ", clock-rate %d", base_rtptime,
- last_extrtptime, local_rtp, clock_rate);
+ ", local RTP %" G_GUINT64_FORMAT ", clock-rate %d, "
+ "clock-base %" G_GINT64_FORMAT, base_rtptime,
+ last_extrtptime, local_rtp, clock_rate, rtp_clock_base);
/* calculate local RTP time in gstreamer timestamp, we essentially perform the
* same conversion that a jitterbuffer would use to convert an rtp timestamp
/* 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);
- } else if (client->nstreams > 1) {
- gint64 min;
+ 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;
+ gboolean rtcp_sync = g_atomic_int_get (&bin->rtcp_sync);
/* calculate delta between server and receiver. last_unix is created by
* converting the ntptime in the last SR packet to a gstreamer timestamp. This
* latencies).
* The stream that has the smallest diff is selected as the reference stream,
* all other streams will have a positive offset to this difference. */
- min = G_MAXINT64;
+
+ /* some alternative setting allow ignoring RTCP as much as possible,
+ * for servers generating bogus ntp timeline */
+ min = rtp_min = G_MAXINT64;
+ use_rtp = FALSE;
+ if (rtcp_sync == GST_RTP_BIN_RTCP_SYNC_RTP) {
+ guint64 ext_base;
+
+ use_rtp = TRUE;
+ /* signed version for convienience */
+ clock_base = base_rtptime;
+ /* deal with possible wrap-around */
+ ext_base = base_rtptime;
+ rtp_clock_base = gst_rtp_buffer_ext_timestamp (&ext_base, rtp_clock_base);
+ /* sanity check; base rtp and provided clock_base should be close */
+ if (rtp_clock_base >= clock_base) {
+ if (rtp_clock_base - clock_base < 10 * clock_rate) {
+ rtp_clock_base = base_time +
+ gst_util_uint64_scale_int (rtp_clock_base - clock_base,
+ GST_SECOND, clock_rate);
+ } else {
+ use_rtp = FALSE;
+ }
+ } else {
+ if (clock_base - rtp_clock_base < 10 * clock_rate) {
+ rtp_clock_base = base_time -
+ gst_util_uint64_scale_int (clock_base - rtp_clock_base,
+ GST_SECOND, clock_rate);
+ } else {
+ use_rtp = FALSE;
+ }
+ }
+ /* warn and bail for clarity out if no sane values */
+ if (!use_rtp) {
+ GST_WARNING_OBJECT (bin, "unable to sync to provided rtptime");
+ return;
+ }
+ /* store to track changes */
+ clock_base = rtp_clock_base;
+ /* generate a fake as before,
+ * 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;
- if (!ostream->have_sync)
+ if (!ostream->have_sync) {
+ all_sync = FALSE;
continue;
+ }
+
+ /* change in current stream's base from previously init'ed value
+ * leads to reset of all stream's base */
+ if (stream != ostream && stream->clock_base >= 0 &&
+ (stream->clock_base != clock_base)) {
+ GST_DEBUG_OBJECT (bin, "reset upon clock base change");
+ ostream->clock_base = -100 * GST_SECOND;
+ ostream->rtp_delta = 0;
+ }
if (ostream->rt_delta < min)
min = ostream->rt_delta;
+ if (ostream->rtp_delta < rtp_min)
+ rtp_min = ostream->rtp_delta;
}
- GST_DEBUG_OBJECT (bin, "client %p min delta %" G_GINT64_FORMAT, client,
- min);
+ /* arrange to re-sync for each stream upon significant change,
+ * e.g. post-seek */
+ 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 */
+ if (client->nstreams <= 1)
+ return;
+
+ GST_DEBUG_OBJECT (bin, "client %p min delta %" G_GINT64_FORMAT
+ " all sync %d", client, min, all_sync);
+ GST_DEBUG_OBJECT (bin, "rtcp sync mode %d, use_rtp %d", rtcp_sync, use_rtp);
+
+ switch (rtcp_sync) {
+ case GST_RTP_BIN_RTCP_SYNC_RTP:
+ if (!use_rtp)
+ break;
+ GST_DEBUG_OBJECT (bin, "using rtp generated reports; "
+ "client %p min rtp delta %" G_GINT64_FORMAT, client, rtp_min);
+ /* fall-through */
+ case GST_RTP_BIN_RTCP_SYNC_INITIAL:
+ /* if all have been synced already, do not bother further */
+ if (all_sync) {
+ GST_DEBUG_OBJECT (bin, "all streams already synced; done");
+ return;
+ }
+ break;
+ default:
+ break;
+ }
/* bail out if we adjusted recently enough */
if (all_sync && (last_unix - bin->priv->last_unix) <
/* calculate offset to our reference stream, this should always give a
* positive number. */
- ts_offset = ostream->rt_delta - min;
+ if (use_rtp)
+ ts_offset = ostream->rtp_delta - rtp_min;
+ 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 base_rtptime;
guint64 base_time;
guint clock_rate;
+ guint64 clock_base;
guint64 extrtptime;
GstBuffer *buffer;
+ GstRTCPBuffer rtcp = { NULL, };
bin = stream->bin;
g_value_get_uint64 (gst_structure_get_value (s, "base-rtptime"));
base_time = g_value_get_uint64 (gst_structure_get_value (s, "base-time"));
clock_rate = g_value_get_uint (gst_structure_get_value (s, "clock-rate"));
+ clock_base = g_value_get_uint64 (gst_structure_get_value (s, "clock-base"));
extrtptime =
g_value_get_uint64 (gst_structure_get_value (s, "sr-ext-rtptime"));
buffer = gst_value_get_buffer (gst_structure_get_value (s, "sr-buffer"));
have_sr = FALSE;
have_sdes = FALSE;
- GST_RTCP_BUFFER_FOR_PACKETS (more, buffer, &packet) {
+
+ gst_rtcp_buffer_map (buffer, GST_MAP_READ, &rtcp);
+
+ GST_RTCP_BUFFER_FOR_PACKETS (more, &rtcp, &packet) {
/* first packet must be SR or RR or else the validate would have failed */
switch (gst_rtcp_packet_get_type (&packet)) {
case GST_RTCP_TYPE_SR:
GST_RTP_BIN_LOCK (bin);
/* associate the stream to CNAME */
gst_rtp_bin_associate (bin, stream, len, data,
- ntptime, extrtptime, base_rtptime, base_time, clock_rate);
+ ntptime, extrtptime, base_rtptime, base_time, clock_rate,
+ clock_base);
GST_RTP_BIN_UNLOCK (bin);
}
}
break;
}
}
+ gst_rtcp_buffer_unmap (&rtcp);
}
/* create a new stream with @ssrc in @session. Must be called with
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;
stream->have_sync = FALSE;
stream->rt_delta = 0;
+ stream->rtp_delta = 0;
stream->percent = 100;
+ stream->clock_base = -100 * GST_SECOND;
session->streams = g_slist_prepend (session->streams, stream);
/* provide clock_rate to the jitterbuffer when needed */
/* 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);
}
static GstStateChangeReturn gst_rtp_bin_change_state (GstElement * element,
GstStateChange transition);
static GstPad *gst_rtp_bin_request_new_pad (GstElement * element,
- GstPadTemplate * templ, const gchar * name);
+ GstPadTemplate * templ, const gchar * name, const GstCaps * caps);
static void gst_rtp_bin_release_pad (GstElement * element, GstPad * pad);
static void gst_rtp_bin_handle_message (GstBin * bin, GstMessage * message);
-GST_BOILERPLATE (GstRtpBin, gst_rtp_bin, GstBin, GST_TYPE_BIN);
-
-static void
-gst_rtp_bin_base_init (gpointer klass)
-{
- GstElementClass *element_class = GST_ELEMENT_CLASS (klass);
-
- /* sink pads */
- gst_element_class_add_pad_template (element_class,
- gst_static_pad_template_get (&rtpbin_recv_rtp_sink_template));
- gst_element_class_add_pad_template (element_class,
- gst_static_pad_template_get (&rtpbin_recv_rtcp_sink_template));
- gst_element_class_add_pad_template (element_class,
- gst_static_pad_template_get (&rtpbin_send_rtp_sink_template));
-
- /* src pads */
- gst_element_class_add_pad_template (element_class,
- gst_static_pad_template_get (&rtpbin_recv_rtp_src_template));
- gst_element_class_add_pad_template (element_class,
- gst_static_pad_template_get (&rtpbin_send_rtcp_src_template));
- gst_element_class_add_pad_template (element_class,
- gst_static_pad_template_get (&rtpbin_send_rtp_src_template));
-
- gst_element_class_set_details_simple (element_class, "RTP Bin",
- "Filter/Network/RTP",
- "Real-Time Transport Protocol bin",
- "Wim Taymans <wim.taymans@gmail.com>");
-}
+#define gst_rtp_bin_parent_class parent_class
+G_DEFINE_TYPE (GstRtpBin, gst_rtp_bin, GST_TYPE_BIN);
static void
gst_rtp_bin_class_init (GstRtpBinClass * klass)
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
*/
G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
/**
+ * GstRtpBin::rtcp-sync:
+ *
+ * If not synchronizing (directly) to the NTP clock, determines how to sync
+ * the various streams.
+ *
+ * Since: 0.10.31
+ */
+ g_object_class_install_property (gobject_class, PROP_RTCP_SYNC,
+ g_param_spec_enum ("rtcp-sync", "RTCP Sync",
+ "Use of RTCP SR in synchronization", GST_RTP_BIN_RTCP_SYNC_TYPE,
+ DEFAULT_RTCP_SYNC, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
+
+ /**
* GstRtpBin::rtcp-sync-interval:
*
* Determines how often to sync streams using RTCP data.
GST_DEBUG_FUNCPTR (gst_rtp_bin_request_new_pad);
gstelement_class->release_pad = GST_DEBUG_FUNCPTR (gst_rtp_bin_release_pad);
+ /* sink pads */
+ gst_element_class_add_pad_template (gstelement_class,
+ gst_static_pad_template_get (&rtpbin_recv_rtp_sink_template));
+ gst_element_class_add_pad_template (gstelement_class,
+ gst_static_pad_template_get (&rtpbin_recv_rtcp_sink_template));
+ gst_element_class_add_pad_template (gstelement_class,
+ gst_static_pad_template_get (&rtpbin_send_rtp_sink_template));
+
+ /* src pads */
+ gst_element_class_add_pad_template (gstelement_class,
+ gst_static_pad_template_get (&rtpbin_recv_rtp_src_template));
+ gst_element_class_add_pad_template (gstelement_class,
+ gst_static_pad_template_get (&rtpbin_send_rtcp_src_template));
+ gst_element_class_add_pad_template (gstelement_class,
+ gst_static_pad_template_get (&rtpbin_send_rtp_src_template));
+
+ gst_element_class_set_static_metadata (gstelement_class, "RTP Bin",
+ "Filter/Network/RTP",
+ "Real-Time Transport Protocol bin",
+ "Wim Taymans <wim.taymans@gmail.com>");
+
gstbin_class->handle_message = GST_DEBUG_FUNCPTR (gst_rtp_bin_handle_message);
klass->clear_pt_map = GST_DEBUG_FUNCPTR (gst_rtp_bin_clear_pt_map);
}
static void
-gst_rtp_bin_init (GstRtpBin * rtpbin, GstRtpBinClass * klass)
+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->rtcp_sync = DEFAULT_RTCP_SYNC;
rtpbin->rtcp_sync_interval = DEFAULT_RTCP_SYNC_INTERVAL;
rtpbin->priv->autoremove = DEFAULT_AUTOREMOVE;
rtpbin->buffer_mode = DEFAULT_BUFFER_MODE;
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;
case PROP_NTP_SYNC:
rtpbin->ntp_sync = g_value_get_boolean (value);
break;
+ case PROP_RTCP_SYNC:
+ g_atomic_int_set (&rtpbin->rtcp_sync, g_value_get_enum (value));
+ break;
case PROP_RTCP_SYNC_INTERVAL:
rtpbin->rtcp_sync_interval = g_value_get_uint (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;
case PROP_NTP_SYNC:
g_value_set_boolean (value, rtpbin->ntp_sync);
break;
+ case PROP_RTCP_SYNC:
+ g_value_set_enum (value, g_atomic_int_get (&rtpbin->rtcp_sync));
+ break;
case PROP_RTCP_SYNC_INTERVAL:
g_value_set_uint (value, rtpbin->rtcp_sync_interval);
break;
now = gst_clock_get_time (clock);
base_time = gst_element_get_base_time (GST_ELEMENT_CAST (bin));
running_time = now - base_time;
+ gst_object_unref (clock);
}
GST_DEBUG_OBJECT (bin,
"running time now %" GST_TIME_FORMAT,
/* 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);
g_object_set_data (G_OBJECT (pad), "GstRTPBin.ghostpad", gpad);
- gst_pad_set_caps (gpad, GST_PAD_CAPS (pad));
gst_pad_set_active (gpad, TRUE);
GST_RTP_BIN_SHUTDOWN_UNLOCK (rtpbin);
/* 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);
- gst_pad_set_caps (gpad, GST_PAD_CAPS (pad));
gst_pad_set_active (gpad, TRUE);
gst_element_add_pad (GST_ELEMENT_CAST (rtpbin), gpad);
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;
}
}
gint session = 0;
GstIterator *pad_it = NULL;
gchar *pad_name = NULL;
+ GValue data = { 0, };
GST_DEBUG_OBJECT (element, "find a free pad name for template");
while (!name_found) {
gboolean done = FALSE;
+
g_free (pad_name);
pad_name = g_strdup_printf (templ->name_template, session++);
pad_it = gst_element_iterate_pads (GST_ELEMENT (element));
name_found = TRUE;
while (!done) {
- gpointer data;
-
switch (gst_iterator_next (pad_it, &data)) {
case GST_ITERATOR_OK:
{
GstPad *pad;
gchar *name;
- pad = GST_PAD_CAST (data);
+ pad = g_value_get_object (&data);
name = gst_pad_get_name (pad);
if (strcmp (name, pad_name) == 0) {
name_found = FALSE;
}
g_free (name);
- gst_object_unref (pad);
+ g_value_reset (&data);
break;
}
case GST_ITERATOR_ERROR:
break;
}
}
+ g_value_unset (&data);
gst_iterator_free (pad_it);
}
*/
static GstPad *
gst_rtp_bin_request_new_pad (GstElement * element,
- GstPadTemplate * templ, const gchar * name)
+ GstPadTemplate * templ, const gchar * name, const GstCaps * caps)
{
GstRtpBin *rtpbin;
GstElementClass *klass;
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