-/* GStreamer
- * Copyright (C) <2007> Wim Taymans <wim.taymans@gmail.com>
- *
- * This library is free software; you can redistribute it and/or
- * modify it under the terms of the GNU Library General Public
- * License as published by the Free Software Foundation; either
- * version 2 of the License, or (at your option) any later version.
- *
- * This library is distributed in the hope that it will be useful,
- * but WITHOUT ANY WARRANTY; without even the implied warranty of
- * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
- * Library General Public License for more details.
- *
- * 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.
- */
+ /* GStreamer
+ * Copyright (C) <2007> Wim Taymans <wim.taymans@gmail.com>
+ *
+ * This library is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU Library General Public
+ * License as published by the Free Software Foundation; either
+ * version 2 of the License, or (at your option) any later version.
+ *
+ * This library is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
+ * Library General Public License for more details.
+ *
+ * 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., 51 Franklin St, Fifth Floor,
+ * Boston, MA 02110-1301, USA.
+ */
/**
- * SECTION:element-gstrtpbin
- * @see_also: gstrtpjitterbuffer, gstrtpsession, gstrtpptdemux, gstrtpssrcdemux
+ * SECTION:element-rtpbin
+ * @see_also: rtpjitterbuffer, rtpsession, rtpptdemux, rtpssrcdemux
*
* RTP bin combines the functions of #GstRtpSession, #GstRtpSsrcDemux,
* #GstRtpJitterBuffer and #GstRtpPtDemux in one element. It allows for multiple
* 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_\%u_\%u_\%u on
- * gstrtpbin with the session number, SSRC and payload type respectively as the pad
+ * rtpbin 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_\%u pad. The
* mapping. One can clear the cached values with the #GstRtpSession::clear-pt-map
* signal.
*
- * Access to the internal statistics of gstrtpbin is provided with the
+ * Access to the internal statistics of rtpbin is provided with the
* get-internal-session property. This action signal gives access to the
* RTPSession object which further provides action signals to retrieve the
* internal source and other sources.
*
+ * #GstRtpBin also has signals (#GstRtpBin::request-rtp-encoder,
+ * #GstRtpBin::request-rtp-decoder, #GstRtpBin::request-rtcp-encoder and
+ * #GstRtpBin::request-rtp-decoder) to dynamically request for RTP and RTCP encoders
+ * and decoders in order to support SRTP. The encoders must provide the pads
+ * rtp_sink_\%u and rtp_src_\%u for RTP and rtcp_sink_\%u and rtcp_src_\%u for
+ * RTCP. The session number will be used in the pad name. The decoders must provide
+ * rtp_sink and rtp_src for RTP and rtcp_sink and rtcp_src for RTCP. The decoders will
+ * be placed before the #GstRtpSession element, thus they must support SSRC demuxing
+ * internally.
+ *
+ * #GstRtpBin has signals (#GstRtpBin::request-aux-sender and
+ * #GstRtpBin::request-aux-receiver to dynamically request an element that can be
+ * used to create or merge additional RTP streams. AUX elements are needed to
+ * implement FEC or retransmission (such as RFC 4588). An AUX sender must have one
+ * sink_\%u pad that matches the sessionid in the signal and it should have 1 or
+ * more src_\%u pads. For each src_%\u pad, a session will be made (if needed)
+ * and the pad will be linked to the session send_rtp_sink pad. Each session will
+ * then expose its source pad as send_rtp_src_\%u on #GstRtpBin.
+ * An AUX receiver has 1 src_\%u pad that much match the sessionid in the signal
+ * and 1 or more sink_\%u pads. A session will be made for each sink_\%u pad
+ * when the corresponding recv_rtp_sink_\%u pad is requested on #GstRtpBin.
+ *
* <refsect2>
* <title>Example pipelines</title>
* |[
* 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.
+ * rtpbin ! rtptheoradepay ! theoradec ! xvimagesink
+ * ]| Receive RTP data from port 5000 and send to the session 0 in rtpbin.
* |[
- * gst-launch-1.0 gstrtpbin name=rtpbin \
+ * gst-launch-1.0 rtpbin 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 \
* as soon as possible and do not participate in preroll, sync=false and
* async=false is configured on udpsink
* |[
- * gst-launch-1.0 -v gstrtpbin name=rtpbin \
+ * gst-launch-1.0 -v rtpbin 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 \
* Send RTCP reports for session 0 on port 5005 and RTCP reports for session 1
* on port 5007.
* </refsect2>
- *
- * Last reviewed on 2007-08-30 (0.10.6)
*/
#ifdef HAVE_CONFIG_H
#include <gst/rtp/gstrtpbuffer.h>
#include <gst/rtp/gstrtcpbuffer.h>
-#include "gstrtpbin-marshal.h"
#include "gstrtpbin.h"
#include "rtpsession.h"
#include "gstrtpsession.h"
/* sink pads */
static GstStaticPadTemplate rtpbin_recv_rtp_sink_template =
-GST_STATIC_PAD_TEMPLATE ("recv_rtp_sink_%u",
+ GST_STATIC_PAD_TEMPLATE ("recv_rtp_sink_%u",
GST_PAD_SINK,
GST_PAD_REQUEST,
- GST_STATIC_CAPS ("application/x-rtp")
+ GST_STATIC_CAPS ("application/x-rtp;application/x-srtp")
);
static GstStaticPadTemplate rtpbin_recv_rtcp_sink_template =
-GST_STATIC_PAD_TEMPLATE ("recv_rtcp_sink_%u",
+ GST_STATIC_PAD_TEMPLATE ("recv_rtcp_sink_%u",
GST_PAD_SINK,
GST_PAD_REQUEST,
- GST_STATIC_CAPS ("application/x-rtcp")
+ GST_STATIC_CAPS ("application/x-rtcp;application/x-srtcp")
);
static GstStaticPadTemplate rtpbin_send_rtp_sink_template =
);
static GstStaticPadTemplate rtpbin_send_rtcp_src_template =
-GST_STATIC_PAD_TEMPLATE ("send_rtcp_src_%u",
+ GST_STATIC_PAD_TEMPLATE ("send_rtcp_src_%u",
GST_PAD_SRC,
GST_PAD_REQUEST,
- GST_STATIC_CAPS ("application/x-rtcp")
+ GST_STATIC_CAPS ("application/x-rtcp;application/x-srtcp")
);
static GstStaticPadTemplate rtpbin_send_rtp_src_template =
-GST_STATIC_PAD_TEMPLATE ("send_rtp_src_%u",
+ GST_STATIC_PAD_TEMPLATE ("send_rtp_src_%u",
GST_PAD_SRC,
GST_PAD_SOMETIMES,
- GST_STATIC_CAPS ("application/x-rtp")
+ GST_STATIC_CAPS ("application/x-rtp;application/x-srtp")
);
-#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_SHUTDOWN_UNLOCK(bin) \
GST_RTP_BIN_DYN_UNLOCK (bin); \
+/* Minimum time offset to apply. This compensates for rounding errors in NTP to
+ * RTP timestamp conversions */
+#define MIN_TS_OFFSET (4 * GST_MSECOND)
+
struct _GstRtpBinPrivate
{
GMutex bin_lock;
gboolean autoremove;
- /* UNIX (ntp) time of last SR sync used */
- guint64 last_unix;
+ /* NTP time in ns of last SR sync used */
+ guint64 last_ntpnstime;
+
+ /* list of extra elements */
+ GList *elements;
};
/* signals and args */
SIGNAL_PAYLOAD_TYPE_CHANGE,
SIGNAL_CLEAR_PT_MAP,
SIGNAL_RESET_SYNC,
+ SIGNAL_GET_SESSION,
SIGNAL_GET_INTERNAL_SESSION,
+ SIGNAL_GET_STORAGE,
+ SIGNAL_GET_INTERNAL_STORAGE,
SIGNAL_ON_NEW_SSRC,
SIGNAL_ON_SSRC_COLLISION,
SIGNAL_ON_TIMEOUT,
SIGNAL_ON_SENDER_TIMEOUT,
SIGNAL_ON_NPT_STOP,
+
+ SIGNAL_REQUEST_RTP_ENCODER,
+ SIGNAL_REQUEST_RTP_DECODER,
+ SIGNAL_REQUEST_RTCP_ENCODER,
+ SIGNAL_REQUEST_RTCP_DECODER,
+
+ SIGNAL_REQUEST_FEC_DECODER,
+ SIGNAL_REQUEST_FEC_ENCODER,
+
+ SIGNAL_NEW_JITTERBUFFER,
+ SIGNAL_NEW_STORAGE,
+
+ SIGNAL_REQUEST_AUX_SENDER,
+ SIGNAL_REQUEST_AUX_RECEIVER,
+
+ SIGNAL_ON_NEW_SENDER_SSRC,
+ SIGNAL_ON_SENDER_SSRC_ACTIVE,
+
+ SIGNAL_ON_BUNDLED_SSRC,
+
LAST_SIGNAL
};
#define DEFAULT_USE_PIPELINE_CLOCK FALSE
#define DEFAULT_RTCP_SYNC GST_RTP_BIN_RTCP_SYNC_ALWAYS
#define DEFAULT_RTCP_SYNC_INTERVAL 0
+#define DEFAULT_DO_SYNC_EVENT FALSE
+#define DEFAULT_DO_RETRANSMISSION FALSE
+#define DEFAULT_RTP_PROFILE GST_RTP_PROFILE_AVP
+#define DEFAULT_NTP_TIME_SOURCE GST_RTP_NTP_TIME_SOURCE_NTP
+#define DEFAULT_RTCP_SYNC_SEND_TIME TRUE
+#define DEFAULT_MAX_RTCP_RTP_TIME_DIFF 1000
+#define DEFAULT_MAX_DROPOUT_TIME 60000
+#define DEFAULT_MAX_MISORDER_TIME 2000
+#define DEFAULT_RFC7273_SYNC FALSE
+#define DEFAULT_MAX_STREAMS G_MAXUINT
+#define DEFAULT_MAX_TS_OFFSET_ADJUSTMENT G_GUINT64_CONSTANT(0)
+#define DEFAULT_MAX_TS_OFFSET G_GINT64_CONSTANT(3000000000)
enum
{
PROP_AUTOREMOVE,
PROP_BUFFER_MODE,
PROP_USE_PIPELINE_CLOCK,
- PROP_LAST
-};
-
-enum
-{
- GST_RTP_BIN_RTCP_SYNC_ALWAYS,
- GST_RTP_BIN_RTCP_SYNC_INITIAL,
- GST_RTP_BIN_RTCP_SYNC_RTP
+ PROP_DO_SYNC_EVENT,
+ PROP_DO_RETRANSMISSION,
+ PROP_RTP_PROFILE,
+ PROP_NTP_TIME_SOURCE,
+ PROP_RTCP_SYNC_SEND_TIME,
+ PROP_MAX_RTCP_RTP_TIME_DIFF,
+ PROP_MAX_DROPOUT_TIME,
+ PROP_MAX_MISORDER_TIME,
+ PROP_RFC7273_SYNC,
+ PROP_MAX_STREAMS,
+ PROP_MAX_TS_OFFSET_ADJUSTMENT,
+ PROP_MAX_TS_OFFSET,
};
#define GST_RTP_BIN_RTCP_SYNC_TYPE (gst_rtp_bin_rtcp_sync_get_type())
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);
+static GstRtpBinSession *create_session (GstRtpBin * rtpbin, gint id);
+static GstPad *complete_session_sink (GstRtpBin * rtpbin,
+ GstRtpBinSession * session);
+static void
+complete_session_receiver (GstRtpBin * rtpbin, GstRtpBinSession * session,
+ guint sessid);
+static GstPad *complete_session_rtcp (GstRtpBin * rtpbin,
+ GstRtpBinSession * session, guint sessid);
/* Manages the RTP stream for one SSRC.
*
gulong buffer_ptreq_sig;
gulong buffer_ntpstop_sig;
gint percent;
-
+#ifdef TIZEN_FEATURE_RTSP_MODIFICATION
+ gint prev_percent;
+#endif
/* the PT demuxer of the SSRC */
GstElement *demux;
gulong demux_newpad_sig;
* there they are pushed into an SSRC demuxer that splits the stream based on
* SSRC. Each of the SSRC streams go into their own jitterbuffer (managed with
* the GstRtpBinStream above).
+ *
+ * Before the SSRC demuxer, a storage element may be inserted for the purpose
+ * of Forward Error Correction.
*/
struct _GstRtpBinSession
{
gulong demux_newpad_sig;
gulong demux_padremoved_sig;
+ /* Fec support */
+ GstElement *storage;
+
GMutex lock;
/* list of GstRtpBinStream */
GSList *streams;
+ /* list of elements */
+ GSList *elements;
+
/* mapping of payload type to caps */
GHashTable *ptmap;
GstPad *sync_src;
GstPad *send_rtp_sink;
GstPad *send_rtp_sink_ghost;
- GstPad *send_rtp_src;
GstPad *send_rtp_src_ghost;
GstPad *send_rtcp_src;
GstPad *send_rtcp_src_ghost;
stream->session->id, stream->ssrc);
}
+static void
+on_new_sender_ssrc (GstElement * session, guint32 ssrc, GstRtpBinSession * sess)
+{
+ g_signal_emit (sess->bin, gst_rtp_bin_signals[SIGNAL_ON_NEW_SENDER_SSRC], 0,
+ sess->id, ssrc);
+}
+
+static void
+on_sender_ssrc_active (GstElement * session, guint32 ssrc,
+ GstRtpBinSession * sess)
+{
+ g_signal_emit (sess->bin, gst_rtp_bin_signals[SIGNAL_ON_SENDER_SSRC_ACTIVE],
+ 0, sess->id, ssrc);
+}
+
/* must be called with the SESSION lock */
static GstRtpBinStream *
find_stream_by_ssrc (GstRtpBinSession * session, guint32 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 */
{
GstRtpBinSession *sess;
GstElement *session, *demux;
+ GstElement *storage = NULL;
GstState target;
if (!(session = gst_element_factory_make ("rtpsession", NULL)))
if (!(demux = gst_element_factory_make ("rtpssrcdemux", NULL)))
goto no_demux;
+ if (!(storage = gst_element_factory_make ("rtpstorage", NULL)))
+ goto no_storage;
+
+ /* need to sink the storage or otherwise signal handlers from bindings will
+ * take ownership of it and we don't own it anymore */
+ gst_object_ref_sink (storage);
+ g_signal_emit (rtpbin, gst_rtp_bin_signals[SIGNAL_NEW_STORAGE], 0, storage,
+ id);
+
sess = g_new0 (GstRtpBinSession, 1);
g_mutex_init (&sess->lock);
sess->id = id;
sess->bin = rtpbin;
sess->session = session;
sess->demux = demux;
+ sess->storage = storage;
+
sess->ptmap = g_hash_table_new_full (NULL, NULL, NULL,
(GDestroyNotify) gst_caps_unref);
rtpbin->sessions = g_slist_prepend (rtpbin->sessions, sess);
/* configure SDES items */
GST_OBJECT_LOCK (rtpbin);
- g_object_set (session, "sdes", rtpbin->sdes, "use-pipeline-clock",
- rtpbin->use_pipeline_clock, NULL);
+ g_object_set (session, "sdes", rtpbin->sdes, "rtp-profile",
+ rtpbin->rtp_profile, "rtcp-sync-send-time", rtpbin->rtcp_sync_send_time,
+ NULL);
+ if (rtpbin->use_pipeline_clock)
+ g_object_set (session, "use-pipeline-clock", rtpbin->use_pipeline_clock,
+ NULL);
+ else
+ g_object_set (session, "ntp-time-source", rtpbin->ntp_time_source, NULL);
+
+ g_object_set (session, "max-dropout-time", rtpbin->max_dropout_time,
+ "max-misorder-time", rtpbin->max_misorder_time, NULL);
GST_OBJECT_UNLOCK (rtpbin);
/* provide clock_rate to the session manager when needed */
g_signal_connect (sess->session, "on-timeout", (GCallback) on_timeout, sess);
g_signal_connect (sess->session, "on-sender-timeout",
(GCallback) on_sender_timeout, sess);
+ g_signal_connect (sess->session, "on-new-sender-ssrc",
+ (GCallback) on_new_sender_ssrc, sess);
+ g_signal_connect (sess->session, "on-sender-ssrc-active",
+ (GCallback) on_sender_ssrc_active, sess);
gst_bin_add (GST_BIN_CAST (rtpbin), session);
gst_bin_add (GST_BIN_CAST (rtpbin), demux);
+ gst_bin_add (GST_BIN_CAST (rtpbin), storage);
+
+ /* unref the storage again, the bin has a reference now and
+ * we don't need it anymore */
+ gst_object_unref (storage);
GST_OBJECT_LOCK (rtpbin);
target = GST_STATE_TARGET (rtpbin);
/* change state only to what's needed */
gst_element_set_state (demux, target);
gst_element_set_state (session, target);
+ gst_element_set_state (storage, target);
return sess;
/* ERRORS */
no_session:
{
- g_warning ("rtpbin: could not create gstrtpsession element");
+ g_warning ("rtpbin: could not create rtpsession element");
return NULL;
}
no_demux:
{
gst_object_unref (session);
- g_warning ("rtpbin: could not create gstrtpssrcdemux element");
+ g_warning ("rtpbin: could not create rtpssrcdemux element");
+ return NULL;
+ }
+no_storage:
+ {
+ gst_object_unref (session);
+ gst_object_unref (demux);
+ g_warning ("rtpbin: could not create rtpstorage element");
return NULL;
}
}
+static gboolean
+bin_manage_element (GstRtpBin * bin, GstElement * element)
+{
+ GstRtpBinPrivate *priv = bin->priv;
+
+ if (g_list_find (priv->elements, element)) {
+ GST_DEBUG_OBJECT (bin, "requested element %p already in bin", element);
+ } else {
+ GST_DEBUG_OBJECT (bin, "adding requested element %p", element);
+
+ if (g_object_is_floating (element))
+ element = gst_object_ref_sink (element);
+
+ if (!gst_bin_add (GST_BIN_CAST (bin), element))
+ goto add_failed;
+ if (!gst_element_sync_state_with_parent (element))
+ GST_WARNING_OBJECT (bin, "unable to sync element state with rtpbin");
+ }
+ /* we add the element multiple times, each we need an equal number of
+ * removes to really remove the element from the bin */
+ priv->elements = g_list_prepend (priv->elements, element);
+
+ return TRUE;
+
+ /* ERRORS */
+add_failed:
+ {
+ GST_WARNING_OBJECT (bin, "unable to add element");
+ gst_object_unref (element);
+ return FALSE;
+ }
+}
+
+static void
+remove_bin_element (GstElement * element, GstRtpBin * bin)
+{
+ GstRtpBinPrivate *priv = bin->priv;
+ GList *find;
+
+ find = g_list_find (priv->elements, element);
+ if (find) {
+ priv->elements = g_list_delete_link (priv->elements, find);
+
+ if (!g_list_find (priv->elements, element)) {
+ gst_element_set_locked_state (element, TRUE);
+ gst_bin_remove (GST_BIN_CAST (bin), element);
+ gst_element_set_state (element, GST_STATE_NULL);
+ }
+
+ gst_object_unref (element);
+ }
+}
+
/* 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_locked_state (sess->session, TRUE);
+ gst_element_set_locked_state (sess->storage, TRUE);
gst_element_set_state (sess->demux, GST_STATE_NULL);
gst_element_set_state (sess->session, GST_STATE_NULL);
+ gst_element_set_state (sess->storage, GST_STATE_NULL);
remove_recv_rtp (bin, sess);
remove_recv_rtcp (bin, sess);
gst_bin_remove (GST_BIN_CAST (bin), sess->session);
gst_bin_remove (GST_BIN_CAST (bin), sess->demux);
+ gst_bin_remove (GST_BIN_CAST (bin), sess->storage);
- /* 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->elements, (GFunc) remove_bin_element, bin);
+ g_slist_free (sess->elements);
- 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);
GValue ret = { 0 };
GValue args[3] = { {0}, {0}, {0} };
- GST_DEBUG ("searching pt %d in cache", pt);
+ GST_DEBUG ("searching pt %u in cache", pt);
GST_RTP_SESSION_LOCK (session);
bin = session->bin;
- GST_DEBUG ("emiting signal for pt %d in session %d", pt, session->id);
+ GST_DEBUG ("emiting signal for pt %u in session %u", pt, session->id);
/* not in cache, send signal to request caps */
g_value_init (&args[0], GST_TYPE_ELEMENT);
if (!caps)
goto no_caps;
- GST_DEBUG ("caching pt %d as %" GST_PTR_FORMAT, pt, caps);
+ GST_DEBUG ("caching pt %u as %" GST_PTR_FORMAT, pt, caps);
/* store in cache, take additional ref */
g_hash_table_insert (session->ptmap, GINT_TO_POINTER (pt),
gst_rtp_bin_reset_sync (bin);
}
+static GstElement *
+gst_rtp_bin_get_session (GstRtpBin * bin, guint session_id)
+{
+ GstRtpBinSession *session;
+ GstElement *ret = NULL;
+
+ GST_RTP_BIN_LOCK (bin);
+ GST_DEBUG_OBJECT (bin, "retrieving GstRtpSession, index: %u", session_id);
+ session = find_session_by_id (bin, (gint) session_id);
+ if (session) {
+ ret = gst_object_ref (session->session);
+ }
+ GST_RTP_BIN_UNLOCK (bin);
+
+ return ret;
+}
+
static RTPSession *
gst_rtp_bin_get_internal_session (GstRtpBin * bin, guint session_id)
{
GstRtpBinSession *session;
GST_RTP_BIN_LOCK (bin);
- GST_DEBUG_OBJECT (bin, "retrieving internal RTPSession object, index: %d",
+ GST_DEBUG_OBJECT (bin, "retrieving internal RTPSession object, index: %u",
session_id);
session = find_session_by_id (bin, (gint) session_id);
if (session) {
return internal_session;
}
+static GstElement *
+gst_rtp_bin_get_storage (GstRtpBin * bin, guint session_id)
+{
+ GstRtpBinSession *session;
+ GstElement *res = NULL;
+
+ GST_RTP_BIN_LOCK (bin);
+ GST_DEBUG_OBJECT (bin, "retrieving internal storage object, index: %u",
+ session_id);
+ session = find_session_by_id (bin, (gint) session_id);
+ if (session && session->storage) {
+ res = gst_object_ref (session->storage);
+ }
+ GST_RTP_BIN_UNLOCK (bin);
+
+ return res;
+}
+
+static GObject *
+gst_rtp_bin_get_internal_storage (GstRtpBin * bin, guint session_id)
+{
+ GObject *internal_storage = NULL;
+ GstRtpBinSession *session;
+
+ GST_RTP_BIN_LOCK (bin);
+ GST_DEBUG_OBJECT (bin, "retrieving internal storage object, index: %u",
+ session_id);
+ session = find_session_by_id (bin, (gint) session_id);
+ if (session && session->storage) {
+ g_object_get (session->storage, "internal-storage", &internal_storage,
+ NULL);
+ }
+ GST_RTP_BIN_UNLOCK (bin);
+
+ return internal_storage;
+}
+
+static GstElement *
+gst_rtp_bin_request_encoder (GstRtpBin * bin, guint session_id)
+{
+ GST_DEBUG_OBJECT (bin, "return NULL encoder");
+ return NULL;
+}
+
+static GstElement *
+gst_rtp_bin_request_decoder (GstRtpBin * bin, guint session_id)
+{
+ GST_DEBUG_OBJECT (bin, "return NULL decoder");
+ return NULL;
+}
+
static void
gst_rtp_bin_propagate_property_to_jitterbuffer (GstRtpBin * bin,
const gchar * name, const GValue * value)
GST_RTP_BIN_UNLOCK (bin);
}
+static void
+gst_rtp_bin_propagate_property_to_session (GstRtpBin * bin,
+ const gchar * name, const GValue * value)
+{
+ GSList *sessions;
+
+ GST_RTP_BIN_LOCK (bin);
+ for (sessions = bin->sessions; sessions; sessions = g_slist_next (sessions)) {
+ GstRtpBinSession *sess = (GstRtpBinSession *) sessions->data;
+
+ g_object_set_property (G_OBJECT (sess->session), name, value);
+ }
+ GST_RTP_BIN_UNLOCK (bin);
+}
+
/* get a client with the given SDES name. Must be called with RTP_BIN_LOCK */
static GstRtpBinClient *
get_client (GstRtpBin * bin, guint8 len, guint8 * data, gboolean * created)
get_current_times (GstRtpBin * bin, GstClockTime * running_time,
guint64 * ntpnstime)
{
- guint64 ntpns;
+ guint64 ntpns = -1;
GstClock *clock;
GstClockTime base_time, rt, clock_time;
gst_object_ref (clock);
GST_OBJECT_UNLOCK (bin);
+ /* get current clock time and convert to running time */
clock_time = gst_clock_get_time (clock);
+ rt = clock_time - base_time;
if (bin->use_pipeline_clock) {
- ntpns = clock_time;
+ ntpns = rt;
+ /* add constant to convert from 1970 based time to 1900 based time */
+ ntpns += (2208988800LL * GST_SECOND);
} else {
- GTimeVal current;
-
- /* get current NTP time */
- g_get_current_time (¤t);
- ntpns = GST_TIMEVAL_TO_TIME (current);
+ switch (bin->ntp_time_source) {
+ case GST_RTP_NTP_TIME_SOURCE_NTP:
+ case GST_RTP_NTP_TIME_SOURCE_UNIX:{
+ GTimeVal current;
+
+ /* get current NTP time */
+ g_get_current_time (¤t);
+ ntpns = GST_TIMEVAL_TO_TIME (current);
+
+ /* add constant to convert from 1970 based time to 1900 based time */
+ if (bin->ntp_time_source == GST_RTP_NTP_TIME_SOURCE_NTP)
+ ntpns += (2208988800LL * GST_SECOND);
+ break;
+ }
+ case GST_RTP_NTP_TIME_SOURCE_RUNNING_TIME:
+ ntpns = rt;
+ break;
+ case GST_RTP_NTP_TIME_SOURCE_CLOCK_TIME:
+ ntpns = clock_time;
+ break;
+ default:
+ ntpns = -1; /* Fix uninited compiler warning */
+ g_assert_not_reached ();
+ break;
+ }
}
- /* add constant to convert from 1970 based time to 1900 based time */
- ntpns += (2208988800LL * GST_SECOND);
-
- /* get current clock time and convert to running time */
- rt = clock_time - base_time;
-
gst_object_unref (clock);
} else {
GST_OBJECT_UNLOCK (bin);
static void
stream_set_ts_offset (GstRtpBin * bin, GstRtpBinStream * stream,
- gint64 ts_offset)
+ gint64 ts_offset, gint64 max_ts_offset, gint64 min_ts_offset,
+ gboolean allow_positive_ts_offset)
{
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 {
- GST_WARNING_OBJECT (bin, "offset unusually large, ignoring");
- }
- } else {
+ /* ignore minor offsets */
+ if (ABS (diff) < min_ts_offset) {
GST_DEBUG_OBJECT (bin, "offset too small, ignoring");
+ return;
}
+
+ /* sanity check offset */
+ if (max_ts_offset > 0) {
+ if (ts_offset > 0 && !allow_positive_ts_offset) {
+ GST_DEBUG_OBJECT (bin,
+ "offset is positive (clocks are out of sync), ignoring");
+ return;
+ }
+ if (ABS (ts_offset) > max_ts_offset) {
+ GST_DEBUG_OBJECT (bin, "offset too large, ignoring");
+ return;
+ }
+ }
+
+ 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_send_sync_event (GstRtpBinStream * stream)
+{
+ if (stream->bin->send_sync_event) {
+ GstEvent *event;
+ GstPad *srcpad;
+
+ GST_DEBUG_OBJECT (stream->bin,
+ "sending GstRTCPSRReceived event downstream");
+
+ event = gst_event_new_custom (GST_EVENT_CUSTOM_DOWNSTREAM,
+ gst_structure_new_empty ("GstRTCPSRReceived"));
+
+ srcpad = gst_element_get_static_pad (stream->buffer, "src");
+ gst_pad_push_event (srcpad, event);
+ gst_object_unref (srcpad);
+ }
+}
+
/* associate a stream to the given CNAME. This will make sure all streams for
* that CNAME are synchronized together.
* Must be called with GST_RTP_BIN_LOCK */
GstRtpBinClient *client;
gboolean created;
GSList *walk;
- guint64 local_rt;
- guint64 local_rtp;
- GstClockTime running_time;
+ GstClockTime running_time, running_time_rtp;
guint64 ntpnstime;
- gint64 ntpdiff, rtdiff;
- guint64 last_unix;
/* first find or create the CNAME */
client = get_client (bin, len, data, &created);
* 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
* timestamp in the SR packet. */
- local_rtp = last_extrtptime - base_rtptime;
+ running_time_rtp = last_extrtptime - base_rtptime;
GST_DEBUG_OBJECT (bin,
"base %" G_GUINT64_FORMAT ", extrtptime %" G_GUINT64_FORMAT
", local RTP %" G_GUINT64_FORMAT ", clock-rate %d, "
"clock-base %" G_GINT64_FORMAT, base_rtptime,
- last_extrtptime, local_rtp, clock_rate, rtp_clock_base);
+ last_extrtptime, running_time_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
* into a corresponding gstreamer timestamp. Note that the base_time also
* contains the drift between sender and receiver. */
- local_rt = gst_util_uint64_scale_int (local_rtp, GST_SECOND, clock_rate);
- local_rt += base_time;
+ running_time =
+ gst_util_uint64_scale_int (running_time_rtp, GST_SECOND, clock_rate);
+ running_time += base_time;
- /* convert ntptime to unix time since 1900 */
- last_unix = gst_util_uint64_scale (ntptime, GST_SECOND,
+ /* convert ntptime to nanoseconds */
+ ntpnstime = gst_util_uint64_scale (ntptime, GST_SECOND,
(G_GINT64_CONSTANT (1) << 32));
stream->have_sync = TRUE;
GST_DEBUG_OBJECT (bin,
- "local UNIX %" G_GUINT64_FORMAT ", remote UNIX %" G_GUINT64_FORMAT,
- local_rt, last_unix);
+ "SR RTP running time %" G_GUINT64_FORMAT ", SR NTP %" G_GUINT64_FORMAT,
+ running_time, ntpnstime);
/* recalc inter stream playout offset, but only if there is more than one
* stream or we're doing NTP sync. */
if (bin->ntp_sync) {
+ gint64 ntpdiff, rtdiff;
+ guint64 local_ntpnstime;
+ GstClockTime local_running_time;
+
/* For NTP sync we need to first get a snapshot of running_time and NTP
* time. We know at what running_time we play a certain RTP time, we also
* calculated when we would play the RTP time in the SR packet. Now we need
* to know how the running_time and the NTP time relate to eachother. */
- get_current_times (bin, &running_time, &ntpnstime);
+ get_current_times (bin, &local_running_time, &local_ntpnstime);
/* see how far away the NTP time is. This is the difference between the
* current NTP time and the NTP time in the last SR packet. */
- ntpdiff = ntpnstime - last_unix;
+ ntpdiff = local_ntpnstime - ntpnstime;
/* see how far away the running_time is. This is the difference between the
* current running_time and the running_time of the RTP timestamp in the
* last SR packet. */
- rtdiff = running_time - local_rt;
+ rtdiff = local_running_time - running_time;
GST_DEBUG_OBJECT (bin,
- "NTP time %" G_GUINT64_FORMAT ", last unix %" G_GUINT64_FORMAT,
- ntpnstime, last_unix);
+ "local NTP time %" G_GUINT64_FORMAT ", SR NTP time %" G_GUINT64_FORMAT,
+ local_ntpnstime, ntpnstime);
+ GST_DEBUG_OBJECT (bin,
+ "local running time %" G_GUINT64_FORMAT ", SR RTP running time %"
+ G_GUINT64_FORMAT, local_running_time, running_time);
GST_DEBUG_OBJECT (bin,
"NTP diff %" G_GINT64_FORMAT ", RT diff %" G_GINT64_FORMAT, ntpdiff,
rtdiff);
/* 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, bin->max_ts_offset,
+ 0, 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
+ /* calculate delta between server and receiver. ntpnstime is created by
* converting the ntptime in the last SR packet to a gstreamer timestamp. This
* delta expresses the difference to our timeline and the server timeline. The
* difference in itself doesn't mean much but we can combine the delta of
* multiple streams to create a stream specific offset. */
- stream->rt_delta = last_unix - local_rt;
+ stream->rt_delta = ntpnstime - running_time;
/* calculate the min of all deltas, ignoring streams that did not yet have a
* valid rt_delta because we did not yet receive an SR packet for those
}
/* bail out if we adjusted recently enough */
- if (all_sync && (last_unix - bin->priv->last_unix) <
+ if (all_sync && (ntpnstime - bin->priv->last_ntpnstime) <
bin->rtcp_sync_interval * GST_MSECOND) {
GST_DEBUG_OBJECT (bin, "discarding RTCP sender packet for sync; "
"previous sender info too recent "
- "(previous UNIX %" G_GUINT64_FORMAT ")", bin->priv->last_unix);
+ "(previous NTP %" G_GUINT64_FORMAT ")", bin->priv->last_ntpnstime);
return;
}
- bin->priv->last_unix = last_unix;
+ bin->priv->last_ntpnstime = ntpnstime;
/* calculate offsets for each stream */
for (walk = client->streams; walk; walk = g_slist_next (walk)) {
else
ts_offset = ostream->rt_delta - min;
- stream_set_ts_offset (bin, ostream, ts_offset);
+ stream_set_ts_offset (bin, ostream, ts_offset, bin->max_ts_offset,
+ MIN_TS_OFFSET, TRUE);
}
}
+ gst_rtp_bin_send_sync_event (stream);
+
return;
}
create_stream (GstRtpBinSession * session, guint32 ssrc)
{
GstElement *buffer, *demux = NULL;
+#ifdef TIZEN_FEATURE_RTSP_MODIFICATION
+ GstElement *queue2 = NULL;
+#endif
GstRtpBinStream *stream;
GstRtpBin *rtpbin;
GstState target;
rtpbin = session->bin;
+ if (g_slist_length (session->streams) >= rtpbin->max_streams)
+ goto max_streams;
+
if (!(buffer = gst_element_factory_make ("rtpjitterbuffer", NULL)))
goto no_jitterbuffer;
- if (!rtpbin->ignore_pt)
+ if (!rtpbin->ignore_pt) {
if (!(demux = gst_element_factory_make ("rtpptdemux", NULL)))
goto no_demux;
-
-
+ }
+#ifdef TIZEN_FEATURE_RTSP_MODIFICATION
+ if (session->bin->buffer_mode == RTP_JITTER_BUFFER_MODE_SLAVE)
+ if (!(queue2 = gst_element_factory_make ("queue2", NULL)))
+ goto no_queue2;
+#endif
stream = g_new0 (GstRtpBinStream, 1);
stream->ssrc = ssrc;
stream->bin = rtpbin;
stream->rt_delta = 0;
stream->rtp_delta = 0;
stream->percent = 100;
+#ifdef TIZEN_FEATURE_RTSP_MODIFICATION
+ stream->prev_percent = 0;
+#endif
stream->clock_base = -100 * GST_SECOND;
session->streams = g_slist_prepend (session->streams, stream);
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);
+ g_object_set (buffer, "do-retransmission", rtpbin->do_retransmission, NULL);
+ g_object_set (buffer, "max-rtcp-rtp-time-diff",
+ rtpbin->max_rtcp_rtp_time_diff, NULL);
+ g_object_set (buffer, "max-dropout-time", rtpbin->max_dropout_time,
+ "max-misorder-time", rtpbin->max_misorder_time, NULL);
+ g_object_set (buffer, "rfc7273-sync", rtpbin->rfc7273_sync, NULL);
+ g_object_set (buffer, "max-ts-offset-adjustment",
+ rtpbin->max_ts_offset_adjustment, NULL);
+
+#ifdef TIZEN_FEATURE_RTSP_MODIFICATION
+ /* configure queue2 to use live buffering */
+ if (queue2) {
+ g_object_set_data (G_OBJECT (queue2), "GstRTPBin.stream", stream);
+ g_object_set (queue2, "use-buffering", TRUE, NULL);
+ g_object_set (queue2, "buffer-mode", GST_BUFFERING_LIVE, NULL);
+ }
+#endif
+ /* need to sink the jitterbufer or otherwise signal handlers from bindings will
+ * take ownership of it and we don't own it anymore */
+ gst_object_ref_sink (buffer);
+ g_signal_emit (rtpbin, gst_rtp_bin_signals[SIGNAL_NEW_JITTERBUFFER], 0,
+ buffer, session->id, ssrc);
if (!rtpbin->ignore_pt)
gst_bin_add (GST_BIN_CAST (rtpbin), demux);
+
+#ifdef TIZEN_FEATURE_RTSP_MODIFICATION
+ if (queue2)
+ gst_bin_add (GST_BIN_CAST (rtpbin), queue2);
+#endif
+
gst_bin_add (GST_BIN_CAST (rtpbin), buffer);
+ /* unref the jitterbuffer again, the bin has a reference now and
+ * we don't need it anymore */
+ gst_object_unref (buffer);
+
/* link stuff */
+#ifdef TIZEN_FEATURE_RTSP_MODIFICATION
+ if (queue2) {
+ gst_element_link_pads_full (buffer, "src", queue2, "sink",
+ GST_PAD_LINK_CHECK_NOTHING);
+ if (demux) {
+ gst_element_link_pads_full (queue2, "src", demux, "sink",
+ GST_PAD_LINK_CHECK_NOTHING);
+ }
+ } else if (demux) {
+ gst_element_link_pads_full (buffer, "src", demux, "sink",
+ GST_PAD_LINK_CHECK_NOTHING);
+ }
+#else
if (demux)
- gst_element_link (buffer, demux);
+ gst_element_link_pads_full (buffer, "src", demux, "sink",
+ GST_PAD_LINK_CHECK_NOTHING);
+#endif
if (rtpbin->buffering) {
guint64 last_out;
gst_element_set_state (buffer, target);
+#ifdef TIZEN_FEATURE_RTSP_MODIFICATION
+ if (queue2)
+ gst_element_set_state (queue2, target);
+#endif
+
return stream;
/* ERRORS */
+max_streams:
+ {
+ GST_WARNING_OBJECT (rtpbin, "stream exeeds maximum (%d)",
+ rtpbin->max_streams);
+ return NULL;
+ }
no_jitterbuffer:
{
- g_warning ("rtpbin: could not create gstrtpjitterbuffer element");
+ g_warning ("rtpbin: could not create rtpjitterbuffer element");
return NULL;
}
no_demux:
{
gst_object_unref (buffer);
- g_warning ("rtpbin: could not create gstrtpptdemux element");
+ g_warning ("rtpbin: could not create rtpptdemux element");
return NULL;
}
+#ifdef TIZEN_FEATURE_RTSP_MODIFICATION
+no_queue2:
+ {
+ gst_object_unref (buffer);
+ gst_object_unref (demux);
+ g_warning ("rtpbin: could not create queue2 element");
+ return NULL;
+ }
+#endif
}
+/* 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 void gst_rtp_bin_handle_message (GstBin * bin, GstMessage * message);
#define gst_rtp_bin_parent_class parent_class
-G_DEFINE_TYPE (GstRtpBin, gst_rtp_bin, GST_TYPE_BIN);
+G_DEFINE_TYPE_WITH_PRIVATE (GstRtpBin, gst_rtp_bin, GST_TYPE_BIN);
+
+static gboolean
+_gst_element_accumulator (GSignalInvocationHint * ihint,
+ GValue * return_accu, const GValue * handler_return, gpointer dummy)
+{
+ GstElement *element;
+
+ element = g_value_get_object (handler_return);
+ GST_DEBUG ("got element %" GST_PTR_FORMAT, element);
+
+ if (!(ihint->run_type & G_SIGNAL_RUN_CLEANUP))
+ g_value_set_object (return_accu, element);
+
+ /* stop emission if we have an element */
+ return (element == NULL);
+}
+
+static gboolean
+_gst_caps_accumulator (GSignalInvocationHint * ihint,
+ GValue * return_accu, const GValue * handler_return, gpointer dummy)
+{
+ GstCaps *caps;
+
+ caps = g_value_get_boxed (handler_return);
+ GST_DEBUG ("got caps %" GST_PTR_FORMAT, caps);
+
+ if (!(ihint->run_type & G_SIGNAL_RUN_CLEANUP))
+ g_value_set_boxed (return_accu, caps);
+
+ /* stop emission if we have a caps */
+ return (caps == NULL);
+}
static void
gst_rtp_bin_class_init (GstRtpBinClass * klass)
gstelement_class = (GstElementClass *) klass;
gstbin_class = (GstBinClass *) klass;
- g_type_class_add_private (klass, sizeof (GstRtpBinPrivate));
-
gobject_class->dispose = gst_rtp_bin_dispose;
gobject_class->finalize = gst_rtp_bin_finalize;
gobject_class->set_property = gst_rtp_bin_set_property;
gst_rtp_bin_signals[SIGNAL_REQUEST_PT_MAP] =
g_signal_new ("request-pt-map", G_TYPE_FROM_CLASS (klass),
G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (GstRtpBinClass, request_pt_map),
- NULL, NULL, gst_rtp_bin_marshal_BOXED__UINT_UINT, GST_TYPE_CAPS, 2,
- G_TYPE_UINT, G_TYPE_UINT);
+ _gst_caps_accumulator, NULL, g_cclosure_marshal_generic, GST_TYPE_CAPS,
+ 2, G_TYPE_UINT, G_TYPE_UINT);
/**
* GstRtpBin::payload-type-change:
* @pt: the pt
*
* Signal that the current payload type changed to @pt in @session.
- *
- * Since: 0.10.17
*/
gst_rtp_bin_signals[SIGNAL_PAYLOAD_TYPE_CHANGE] =
g_signal_new ("payload-type-change", G_TYPE_FROM_CLASS (klass),
G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (GstRtpBinClass, payload_type_change),
- NULL, NULL, gst_rtp_bin_marshal_VOID__UINT_UINT, G_TYPE_NONE, 2,
- G_TYPE_UINT, G_TYPE_UINT);
+ NULL, NULL, g_cclosure_marshal_generic, G_TYPE_NONE, 2, G_TYPE_UINT,
+ G_TYPE_UINT);
/**
* GstRtpBin::clear-pt-map:
0, G_TYPE_NONE);
/**
+ * GstRtpBin::get-session:
+ * @rtpbin: the object which received the signal
+ * @id: the session id
+ *
+ * Request the related GstRtpSession as #GstElement related with session @id.
+ *
+ * Since: 1.8
+ */
+ gst_rtp_bin_signals[SIGNAL_GET_SESSION] =
+ g_signal_new ("get-session", G_TYPE_FROM_CLASS (klass),
+ G_SIGNAL_RUN_LAST | G_SIGNAL_ACTION, G_STRUCT_OFFSET (GstRtpBinClass,
+ get_session), NULL, NULL, g_cclosure_marshal_generic,
+ GST_TYPE_ELEMENT, 1, G_TYPE_UINT);
+
+ /**
* GstRtpBin::get-internal-session:
* @rtpbin: the object which received the signal
* @id: the session id
gst_rtp_bin_signals[SIGNAL_GET_INTERNAL_SESSION] =
g_signal_new ("get-internal-session", G_TYPE_FROM_CLASS (klass),
G_SIGNAL_RUN_LAST | G_SIGNAL_ACTION, G_STRUCT_OFFSET (GstRtpBinClass,
- get_internal_session), NULL, NULL, gst_rtp_bin_marshal_OBJECT__UINT,
+ get_internal_session), NULL, NULL, g_cclosure_marshal_generic,
RTP_TYPE_SESSION, 1, G_TYPE_UINT);
/**
- * GstRtpBin::on-new-ssrc:
+ * GstRtpBin::get-internal-storage:
* @rtpbin: the object which received the signal
- * @session: the session
- * @ssrc: the SSRC
+ * @id: the session id
*
- * Notify of a new SSRC that entered @session.
+ * Request the internal RTPStorage object as #GObject in session @id.
+ *
+ * Since: 1.14
*/
- gst_rtp_bin_signals[SIGNAL_ON_NEW_SSRC] =
- g_signal_new ("on-new-ssrc", G_TYPE_FROM_CLASS (klass),
- G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (GstRtpBinClass, on_new_ssrc),
- NULL, NULL, gst_rtp_bin_marshal_VOID__UINT_UINT, G_TYPE_NONE, 2,
- G_TYPE_UINT, G_TYPE_UINT);
+ gst_rtp_bin_signals[SIGNAL_GET_INTERNAL_STORAGE] =
+ g_signal_new ("get-internal-storage", G_TYPE_FROM_CLASS (klass),
+ G_SIGNAL_RUN_LAST | G_SIGNAL_ACTION, G_STRUCT_OFFSET (GstRtpBinClass,
+ get_internal_storage), NULL, NULL, g_cclosure_marshal_generic,
+ G_TYPE_OBJECT, 1, G_TYPE_UINT);
+
/**
- * GstRtpBin::on-ssrc-collision:
+ * GstRtpBin::get-storage:
* @rtpbin: the object which received the signal
- * @session: the session
- * @ssrc: the SSRC
+ * @id: the session id
+ *
+ * Request the RTPStorage element as #GObject in session @id.
+ *
+ * Since: 1.16
+ */
+ gst_rtp_bin_signals[SIGNAL_GET_STORAGE] =
+ g_signal_new ("get-storage", G_TYPE_FROM_CLASS (klass),
+ G_SIGNAL_RUN_LAST | G_SIGNAL_ACTION, G_STRUCT_OFFSET (GstRtpBinClass,
+ get_storage), NULL, NULL, g_cclosure_marshal_generic,
+ GST_TYPE_ELEMENT, 1, G_TYPE_UINT);
+
+ /**
+ * GstRtpBin::on-new-ssrc:
+ * @rtpbin: the object which received the signal
+ * @session: the session
+ * @ssrc: the SSRC
+ *
+ * Notify of a new SSRC that entered @session.
+ */
+ gst_rtp_bin_signals[SIGNAL_ON_NEW_SSRC] =
+ g_signal_new ("on-new-ssrc", G_TYPE_FROM_CLASS (klass),
+ G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (GstRtpBinClass, on_new_ssrc),
+ NULL, NULL, g_cclosure_marshal_generic, G_TYPE_NONE, 2, G_TYPE_UINT,
+ G_TYPE_UINT);
+ /**
+ * GstRtpBin::on-ssrc-collision:
+ * @rtpbin: the object which received the signal
+ * @session: the session
+ * @ssrc: the SSRC
*
* Notify when we have an SSRC collision
*/
gst_rtp_bin_signals[SIGNAL_ON_SSRC_COLLISION] =
g_signal_new ("on-ssrc-collision", G_TYPE_FROM_CLASS (klass),
G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (GstRtpBinClass, on_ssrc_collision),
- NULL, NULL, gst_rtp_bin_marshal_VOID__UINT_UINT, G_TYPE_NONE, 2,
- G_TYPE_UINT, G_TYPE_UINT);
+ NULL, NULL, g_cclosure_marshal_generic, G_TYPE_NONE, 2, G_TYPE_UINT,
+ G_TYPE_UINT);
/**
* GstRtpBin::on-ssrc-validated:
* @rtpbin: the object which received the signal
gst_rtp_bin_signals[SIGNAL_ON_SSRC_VALIDATED] =
g_signal_new ("on-ssrc-validated", G_TYPE_FROM_CLASS (klass),
G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (GstRtpBinClass, on_ssrc_validated),
- NULL, NULL, gst_rtp_bin_marshal_VOID__UINT_UINT, G_TYPE_NONE, 2,
- G_TYPE_UINT, G_TYPE_UINT);
+ NULL, NULL, g_cclosure_marshal_generic, G_TYPE_NONE, 2, G_TYPE_UINT,
+ G_TYPE_UINT);
/**
* GstRtpBin::on-ssrc-active:
* @rtpbin: the object which received the signal
gst_rtp_bin_signals[SIGNAL_ON_SSRC_ACTIVE] =
g_signal_new ("on-ssrc-active", G_TYPE_FROM_CLASS (klass),
G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (GstRtpBinClass, on_ssrc_active),
- NULL, NULL, gst_rtp_bin_marshal_VOID__UINT_UINT, G_TYPE_NONE, 2,
- G_TYPE_UINT, G_TYPE_UINT);
+ NULL, NULL, g_cclosure_marshal_generic, G_TYPE_NONE, 2, G_TYPE_UINT,
+ G_TYPE_UINT);
/**
* GstRtpBin::on-ssrc-sdes:
* @rtpbin: the object which received the signal
gst_rtp_bin_signals[SIGNAL_ON_SSRC_SDES] =
g_signal_new ("on-ssrc-sdes", G_TYPE_FROM_CLASS (klass),
G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (GstRtpBinClass, on_ssrc_sdes),
- NULL, NULL, gst_rtp_bin_marshal_VOID__UINT_UINT, G_TYPE_NONE, 2,
- G_TYPE_UINT, G_TYPE_UINT);
+ NULL, NULL, g_cclosure_marshal_generic, G_TYPE_NONE, 2, G_TYPE_UINT,
+ G_TYPE_UINT);
/**
* GstRtpBin::on-bye-ssrc:
gst_rtp_bin_signals[SIGNAL_ON_BYE_SSRC] =
g_signal_new ("on-bye-ssrc", G_TYPE_FROM_CLASS (klass),
G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (GstRtpBinClass, on_bye_ssrc),
- NULL, NULL, gst_rtp_bin_marshal_VOID__UINT_UINT, G_TYPE_NONE, 2,
- G_TYPE_UINT, G_TYPE_UINT);
+ NULL, NULL, g_cclosure_marshal_generic, G_TYPE_NONE, 2, G_TYPE_UINT,
+ G_TYPE_UINT);
/**
* GstRtpBin::on-bye-timeout:
* @rtpbin: the object which received the signal
gst_rtp_bin_signals[SIGNAL_ON_BYE_TIMEOUT] =
g_signal_new ("on-bye-timeout", G_TYPE_FROM_CLASS (klass),
G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (GstRtpBinClass, on_bye_timeout),
- NULL, NULL, gst_rtp_bin_marshal_VOID__UINT_UINT, G_TYPE_NONE, 2,
- G_TYPE_UINT, G_TYPE_UINT);
+ NULL, NULL, g_cclosure_marshal_generic, G_TYPE_NONE, 2, G_TYPE_UINT,
+ G_TYPE_UINT);
/**
* GstRtpBin::on-timeout:
* @rtpbin: the object which received the signal
gst_rtp_bin_signals[SIGNAL_ON_TIMEOUT] =
g_signal_new ("on-timeout", G_TYPE_FROM_CLASS (klass),
G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (GstRtpBinClass, on_timeout),
- NULL, NULL, gst_rtp_bin_marshal_VOID__UINT_UINT, G_TYPE_NONE, 2,
- G_TYPE_UINT, G_TYPE_UINT);
+ NULL, NULL, g_cclosure_marshal_generic, G_TYPE_NONE, 2, G_TYPE_UINT,
+ G_TYPE_UINT);
/**
* GstRtpBin::on-sender-timeout:
* @rtpbin: the object which received the signal
gst_rtp_bin_signals[SIGNAL_ON_SENDER_TIMEOUT] =
g_signal_new ("on-sender-timeout", G_TYPE_FROM_CLASS (klass),
G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (GstRtpBinClass, on_sender_timeout),
- NULL, NULL, gst_rtp_bin_marshal_VOID__UINT_UINT, G_TYPE_NONE, 2,
- G_TYPE_UINT, G_TYPE_UINT);
+ NULL, NULL, g_cclosure_marshal_generic, G_TYPE_NONE, 2, G_TYPE_UINT,
+ G_TYPE_UINT);
/**
* GstRtpBin::on-npt-stop:
gst_rtp_bin_signals[SIGNAL_ON_NPT_STOP] =
g_signal_new ("on-npt-stop", G_TYPE_FROM_CLASS (klass),
G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (GstRtpBinClass, on_npt_stop),
- NULL, NULL, gst_rtp_bin_marshal_VOID__UINT_UINT, G_TYPE_NONE, 2,
- G_TYPE_UINT, G_TYPE_UINT);
+ NULL, NULL, g_cclosure_marshal_generic, G_TYPE_NONE, 2, G_TYPE_UINT,
+ G_TYPE_UINT);
+
+ /**
+ * GstRtpBin::request-rtp-encoder:
+ * @rtpbin: the object which received the signal
+ * @session: the session
+ *
+ * Request an RTP encoder element for the given @session. The encoder
+ * element will be added to the bin if not previously added.
+ *
+ * If no handler is connected, no encoder will be used.
+ *
+ * Since: 1.4
+ */
+ gst_rtp_bin_signals[SIGNAL_REQUEST_RTP_ENCODER] =
+ g_signal_new ("request-rtp-encoder", G_TYPE_FROM_CLASS (klass),
+ G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (GstRtpBinClass,
+ request_rtp_encoder), _gst_element_accumulator, NULL,
+ g_cclosure_marshal_generic, GST_TYPE_ELEMENT, 1, G_TYPE_UINT);
+
+ /**
+ * GstRtpBin::request-rtp-decoder:
+ * @rtpbin: the object which received the signal
+ * @session: the session
+ *
+ * Request an RTP decoder element for the given @session. The decoder
+ * element will be added to the bin if not previously added.
+ *
+ * If no handler is connected, no encoder will be used.
+ *
+ * Since: 1.4
+ */
+ gst_rtp_bin_signals[SIGNAL_REQUEST_RTP_DECODER] =
+ g_signal_new ("request-rtp-decoder", G_TYPE_FROM_CLASS (klass),
+ G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (GstRtpBinClass,
+ request_rtp_decoder), _gst_element_accumulator, NULL,
+ g_cclosure_marshal_generic, GST_TYPE_ELEMENT, 1, G_TYPE_UINT);
+
+ /**
+ * GstRtpBin::request-rtcp-encoder:
+ * @rtpbin: the object which received the signal
+ * @session: the session
+ *
+ * Request an RTCP encoder element for the given @session. The encoder
+ * element will be added to the bin if not previously added.
+ *
+ * If no handler is connected, no encoder will be used.
+ *
+ * Since: 1.4
+ */
+ gst_rtp_bin_signals[SIGNAL_REQUEST_RTCP_ENCODER] =
+ g_signal_new ("request-rtcp-encoder", G_TYPE_FROM_CLASS (klass),
+ G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (GstRtpBinClass,
+ request_rtcp_encoder), _gst_element_accumulator, NULL,
+ g_cclosure_marshal_generic, GST_TYPE_ELEMENT, 1, G_TYPE_UINT);
+
+ /**
+ * GstRtpBin::request-rtcp-decoder:
+ * @rtpbin: the object which received the signal
+ * @session: the session
+ *
+ * Request an RTCP decoder element for the given @session. The decoder
+ * element will be added to the bin if not previously added.
+ *
+ * If no handler is connected, no encoder will be used.
+ *
+ * Since: 1.4
+ */
+ gst_rtp_bin_signals[SIGNAL_REQUEST_RTCP_DECODER] =
+ g_signal_new ("request-rtcp-decoder", G_TYPE_FROM_CLASS (klass),
+ G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (GstRtpBinClass,
+ request_rtcp_decoder), _gst_element_accumulator, NULL,
+ g_cclosure_marshal_generic, GST_TYPE_ELEMENT, 1, G_TYPE_UINT);
+
+ /**
+ * GstRtpBin::new-jitterbuffer:
+ * @rtpbin: the object which received the signal
+ * @jitterbuffer: the new jitterbuffer
+ * @session: the session
+ * @ssrc: the SSRC
+ *
+ * Notify that a new @jitterbuffer was created for @session and @ssrc.
+ * This signal can, for example, be used to configure @jitterbuffer.
+ *
+ * Since: 1.4
+ */
+ gst_rtp_bin_signals[SIGNAL_NEW_JITTERBUFFER] =
+ g_signal_new ("new-jitterbuffer", G_TYPE_FROM_CLASS (klass),
+ G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (GstRtpBinClass,
+ new_jitterbuffer), NULL, NULL, g_cclosure_marshal_generic,
+ G_TYPE_NONE, 3, GST_TYPE_ELEMENT, G_TYPE_UINT, G_TYPE_UINT);
+
+ /**
+ * GstRtpBin::new-storage:
+ * @rtpbin: the object which received the signal
+ * @storage: the new storage
+ * @session: the session
+ *
+ * Notify that a new @storage was created for @session.
+ * This signal can, for example, be used to configure @storage.
+ *
+ * Since: 1.14
+ */
+ gst_rtp_bin_signals[SIGNAL_NEW_STORAGE] =
+ g_signal_new ("new-storage", G_TYPE_FROM_CLASS (klass),
+ G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (GstRtpBinClass,
+ new_storage), NULL, NULL, g_cclosure_marshal_generic,
+ G_TYPE_NONE, 2, GST_TYPE_ELEMENT, G_TYPE_UINT);
+
+ /**
+ * GstRtpBin::request-aux-sender:
+ * @rtpbin: the object which received the signal
+ * @session: the session
+ *
+ * Request an AUX sender element for the given @session. The AUX
+ * element will be added to the bin.
+ *
+ * If no handler is connected, no AUX element will be used.
+ *
+ * Since: 1.4
+ */
+ gst_rtp_bin_signals[SIGNAL_REQUEST_AUX_SENDER] =
+ g_signal_new ("request-aux-sender", G_TYPE_FROM_CLASS (klass),
+ G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (GstRtpBinClass,
+ request_aux_sender), _gst_element_accumulator, NULL,
+ g_cclosure_marshal_generic, GST_TYPE_ELEMENT, 1, G_TYPE_UINT);
+
+ /**
+ * GstRtpBin::request-aux-receiver:
+ * @rtpbin: the object which received the signal
+ * @session: the session
+ *
+ * Request an AUX receiver element for the given @session. The AUX
+ * element will be added to the bin.
+ *
+ * If no handler is connected, no AUX element will be used.
+ *
+ * Since: 1.4
+ */
+ gst_rtp_bin_signals[SIGNAL_REQUEST_AUX_RECEIVER] =
+ g_signal_new ("request-aux-receiver", G_TYPE_FROM_CLASS (klass),
+ G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (GstRtpBinClass,
+ request_aux_receiver), _gst_element_accumulator, NULL,
+ g_cclosure_marshal_generic, GST_TYPE_ELEMENT, 1, G_TYPE_UINT);
+
+ /**
+ * GstRtpBin::request-fec-decoder:
+ * @rtpbin: the object which received the signal
+ * @session: the session index
+ *
+ * Request a FEC decoder element for the given @session. The element
+ * will be added to the bin after the pt demuxer.
+ *
+ * If no handler is connected, no FEC decoder will be used.
+ *
+ * Since: 1.14
+ */
+ gst_rtp_bin_signals[SIGNAL_REQUEST_FEC_DECODER] =
+ g_signal_new ("request-fec-decoder", G_TYPE_FROM_CLASS (klass),
+ G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (GstRtpBinClass,
+ request_fec_decoder), _gst_element_accumulator, NULL,
+ g_cclosure_marshal_generic, GST_TYPE_ELEMENT, 1, G_TYPE_UINT);
+
+ /**
+ * GstRtpBin::request-fec-encoder:
+ * @rtpbin: the object which received the signal
+ * @session: the session index
+ *
+ * Request a FEC encoder element for the given @session. The element
+ * will be added to the bin after the RTPSession.
+ *
+ * If no handler is connected, no FEC encoder will be used.
+ *
+ * Since: 1.14
+ */
+ gst_rtp_bin_signals[SIGNAL_REQUEST_FEC_ENCODER] =
+ g_signal_new ("request-fec-encoder", G_TYPE_FROM_CLASS (klass),
+ G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (GstRtpBinClass,
+ request_fec_encoder), _gst_element_accumulator, NULL,
+ g_cclosure_marshal_generic, GST_TYPE_ELEMENT, 1, G_TYPE_UINT);
+
+ /**
+ * GstRtpBin::on-new-sender-ssrc:
+ * @rtpbin: the object which received the signal
+ * @session: the session
+ * @ssrc: the sender SSRC
+ *
+ * Notify of a new sender SSRC that entered @session.
+ *
+ * Since: 1.8
+ */
+ gst_rtp_bin_signals[SIGNAL_ON_NEW_SENDER_SSRC] =
+ g_signal_new ("on-new-sender-ssrc", G_TYPE_FROM_CLASS (klass),
+ G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (GstRtpBinClass, on_new_sender_ssrc),
+ NULL, NULL, g_cclosure_marshal_generic, G_TYPE_NONE, 2, G_TYPE_UINT,
+ G_TYPE_UINT);
+ /**
+ * GstRtpBin::on-sender-ssrc-active:
+ * @rtpbin: the object which received the signal
+ * @session: the session
+ * @ssrc: the sender SSRC
+ *
+ * Notify of a sender SSRC that is active, i.e., sending RTCP.
+ *
+ * Since: 1.8
+ */
+ gst_rtp_bin_signals[SIGNAL_ON_SENDER_SSRC_ACTIVE] =
+ g_signal_new ("on-sender-ssrc-active", G_TYPE_FROM_CLASS (klass),
+ G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (GstRtpBinClass,
+ on_sender_ssrc_active), NULL, NULL, g_cclosure_marshal_generic,
+ G_TYPE_NONE, 2, G_TYPE_UINT, G_TYPE_UINT);
g_object_class_install_property (gobject_class, PROP_SDES,
g_param_spec_boxed ("sdes", "SDES",
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 "
+ "(DEPRECATED: Use ntp-time-source property)",
DEFAULT_USE_PIPELINE_CLOCK,
- G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
+ G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS | G_PARAM_DEPRECATED));
/**
- * GstRtpBin::buffer-mode:
+ * GstRtpBin:buffer-mode:
*
* Control the buffering and timestamping mode used by the jitterbuffer.
- *
- * Since: 0.10.17
*/
g_object_class_install_property (gobject_class, PROP_BUFFER_MODE,
g_param_spec_enum ("buffer-mode", "Buffer Mode",
"Control the buffering algorithm in use", RTP_TYPE_JITTER_BUFFER_MODE,
DEFAULT_BUFFER_MODE, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
/**
- * GstRtpBin::ntp-sync:
+ * 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.
- *
- * Since: 0.10.21
+ * 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.
*/
g_object_class_install_property (gobject_class, PROP_NTP_SYNC,
g_param_spec_boolean ("ntp-sync", "Sync on NTP clock",
G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
/**
- * GstRtpBin::rtcp-sync:
+ * 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",
DEFAULT_RTCP_SYNC, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
/**
- * GstRtpBin::rtcp-sync-interval:
+ * GstRtpBin:rtcp-sync-interval:
*
* Determines how often to sync streams using RTCP data.
- *
- * Since: 0.10.31
*/
g_object_class_install_property (gobject_class, PROP_RTCP_SYNC_INTERVAL,
g_param_spec_uint ("rtcp-sync-interval", "RTCP Sync Interval",
0, G_MAXUINT, DEFAULT_RTCP_SYNC_INTERVAL,
G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
+ g_object_class_install_property (gobject_class, PROP_DO_SYNC_EVENT,
+ g_param_spec_boolean ("do-sync-event", "Do Sync Event",
+ "Send event downstream when a stream is synchronized to the sender",
+ DEFAULT_DO_SYNC_EVENT, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
+
+ /**
+ * GstRtpBin:do-retransmission:
+ *
+ * Enables RTP retransmission on all streams. To control retransmission on
+ * a per-SSRC basis, connect to the #GstRtpBin::new-jitterbuffer signal and
+ * set the #GstRtpJitterBuffer::do-retransmission property on the
+ * #GstRtpJitterBuffer object instead.
+ */
+ g_object_class_install_property (gobject_class, PROP_DO_RETRANSMISSION,
+ g_param_spec_boolean ("do-retransmission", "Do retransmission",
+ "Enable retransmission on all streams",
+ DEFAULT_DO_RETRANSMISSION,
+ G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
+
+ /**
+ * GstRtpBin:rtp-profile:
+ *
+ * Sets the default RTP profile of newly created RTP sessions. The
+ * profile can be changed afterwards on a per-session basis.
+ */
+ g_object_class_install_property (gobject_class, PROP_RTP_PROFILE,
+ g_param_spec_enum ("rtp-profile", "RTP Profile",
+ "Default RTP profile of newly created sessions",
+ GST_TYPE_RTP_PROFILE, DEFAULT_RTP_PROFILE,
+ G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
+
+ g_object_class_install_property (gobject_class, PROP_NTP_TIME_SOURCE,
+ g_param_spec_enum ("ntp-time-source", "NTP Time Source",
+ "NTP time source for RTCP packets",
+ gst_rtp_ntp_time_source_get_type (), DEFAULT_NTP_TIME_SOURCE,
+ G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
+
+ g_object_class_install_property (gobject_class, PROP_RTCP_SYNC_SEND_TIME,
+ g_param_spec_boolean ("rtcp-sync-send-time", "RTCP Sync Send Time",
+ "Use send time or capture time for RTCP sync "
+ "(TRUE = send time, FALSE = capture time)",
+ DEFAULT_RTCP_SYNC_SEND_TIME,
+ G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
+
+ g_object_class_install_property (gobject_class, PROP_MAX_RTCP_RTP_TIME_DIFF,
+ g_param_spec_int ("max-rtcp-rtp-time-diff", "Max RTCP RTP Time Diff",
+ "Maximum amount of time in ms that the RTP time in RTCP SRs "
+ "is allowed to be ahead (-1 disabled)", -1, G_MAXINT,
+ DEFAULT_MAX_RTCP_RTP_TIME_DIFF,
+ G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
+
+ g_object_class_install_property (gobject_class, PROP_MAX_DROPOUT_TIME,
+ g_param_spec_uint ("max-dropout-time", "Max dropout time",
+ "The maximum time (milliseconds) of missing packets tolerated.",
+ 0, G_MAXUINT, DEFAULT_MAX_DROPOUT_TIME,
+ G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
+
+ g_object_class_install_property (gobject_class, PROP_MAX_MISORDER_TIME,
+ g_param_spec_uint ("max-misorder-time", "Max misorder time",
+ "The maximum time (milliseconds) of misordered packets tolerated.",
+ 0, G_MAXUINT, DEFAULT_MAX_MISORDER_TIME,
+ G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
+
+ g_object_class_install_property (gobject_class, PROP_RFC7273_SYNC,
+ g_param_spec_boolean ("rfc7273-sync", "Sync on RFC7273 clock",
+ "Synchronize received streams to the RFC7273 clock "
+ "(requires clock and offset to be provided)", DEFAULT_RFC7273_SYNC,
+ G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
+
+ g_object_class_install_property (gobject_class, PROP_MAX_STREAMS,
+ g_param_spec_uint ("max-streams", "Max Streams",
+ "The maximum number of streams to create for one session",
+ 0, G_MAXUINT, DEFAULT_MAX_STREAMS,
+ G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
+
+ /**
+ * GstRtpBin:max-ts-offset-adjustment:
+ *
+ * Syncing time stamps to NTP time adds a time offset. This parameter
+ * specifies the maximum number of nanoseconds per frame that this time offset
+ * may be adjusted with. This is used to avoid sudden large changes to time
+ * stamps.
+ *
+ * Since: 1.14
+ */
+ g_object_class_install_property (gobject_class, PROP_MAX_TS_OFFSET_ADJUSTMENT,
+ g_param_spec_uint64 ("max-ts-offset-adjustment",
+ "Max Timestamp Offset Adjustment",
+ "The maximum number of nanoseconds per frame that time stamp offsets "
+ "may be adjusted (0 = no limit).", 0, G_MAXUINT64,
+ DEFAULT_MAX_TS_OFFSET_ADJUSTMENT, G_PARAM_READWRITE |
+ G_PARAM_STATIC_STRINGS));
+
+ /**
+ * GstRtpBin:max-ts-offset:
+ *
+ * Used to set an upper limit of how large a time offset may be. This
+ * is used to protect against unrealistic values as a result of either
+ * client,server or clock issues.
+ *
+ * Since: 1.14
+ */
+ g_object_class_install_property (gobject_class, PROP_MAX_TS_OFFSET,
+ g_param_spec_int64 ("max-ts-offset", "Max TS Offset",
+ "The maximum absolute value of the time offset in (nanoseconds). "
+ "Note, if the ntp-sync parameter is set the default value is "
+ "changed to 0 (no limit)", 0, G_MAXINT64, DEFAULT_MAX_TS_OFFSET,
+ G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
+
gstelement_class->change_state = GST_DEBUG_FUNCPTR (gst_rtp_bin_change_state);
gstelement_class->request_new_pad =
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));
+ gst_element_class_add_static_pad_template (gstelement_class,
+ &rtpbin_recv_rtp_sink_template);
+ gst_element_class_add_static_pad_template (gstelement_class,
+ &rtpbin_recv_rtcp_sink_template);
+ gst_element_class_add_static_pad_template (gstelement_class,
+ &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_add_static_pad_template (gstelement_class,
+ &rtpbin_recv_rtp_src_template);
+ gst_element_class_add_static_pad_template (gstelement_class,
+ &rtpbin_send_rtcp_src_template);
+ gst_element_class_add_static_pad_template (gstelement_class,
+ &rtpbin_send_rtp_src_template);
gst_element_class_set_static_metadata (gstelement_class, "RTP Bin",
"Filter/Network/RTP",
klass->clear_pt_map = GST_DEBUG_FUNCPTR (gst_rtp_bin_clear_pt_map);
klass->reset_sync = GST_DEBUG_FUNCPTR (gst_rtp_bin_reset_sync);
+ klass->get_session = GST_DEBUG_FUNCPTR (gst_rtp_bin_get_session);
klass->get_internal_session =
GST_DEBUG_FUNCPTR (gst_rtp_bin_get_internal_session);
+ klass->get_storage = GST_DEBUG_FUNCPTR (gst_rtp_bin_get_storage);
+ klass->get_internal_storage =
+ GST_DEBUG_FUNCPTR (gst_rtp_bin_get_internal_storage);
+ klass->request_rtp_encoder = GST_DEBUG_FUNCPTR (gst_rtp_bin_request_encoder);
+ klass->request_rtp_decoder = GST_DEBUG_FUNCPTR (gst_rtp_bin_request_decoder);
+ klass->request_rtcp_encoder = GST_DEBUG_FUNCPTR (gst_rtp_bin_request_encoder);
+ klass->request_rtcp_decoder = GST_DEBUG_FUNCPTR (gst_rtp_bin_request_decoder);
GST_DEBUG_CATEGORY_INIT (gst_rtp_bin_debug, "rtpbin", 0, "RTP bin");
}
{
gchar *cname;
- rtpbin->priv = GST_RTP_BIN_GET_PRIVATE (rtpbin);
+ rtpbin->priv = gst_rtp_bin_get_instance_private (rtpbin);
g_mutex_init (&rtpbin->priv->bin_lock);
g_mutex_init (&rtpbin->priv->dyn_lock);
rtpbin->priv->autoremove = DEFAULT_AUTOREMOVE;
rtpbin->buffer_mode = DEFAULT_BUFFER_MODE;
rtpbin->use_pipeline_clock = DEFAULT_USE_PIPELINE_CLOCK;
+ rtpbin->send_sync_event = DEFAULT_DO_SYNC_EVENT;
+ rtpbin->do_retransmission = DEFAULT_DO_RETRANSMISSION;
+ rtpbin->rtp_profile = DEFAULT_RTP_PROFILE;
+ rtpbin->ntp_time_source = DEFAULT_NTP_TIME_SOURCE;
+ rtpbin->rtcp_sync_send_time = DEFAULT_RTCP_SYNC_SEND_TIME;
+ rtpbin->max_rtcp_rtp_time_diff = DEFAULT_MAX_RTCP_RTP_TIME_DIFF;
+ rtpbin->max_dropout_time = DEFAULT_MAX_DROPOUT_TIME;
+ rtpbin->max_misorder_time = DEFAULT_MAX_MISORDER_TIME;
+ rtpbin->rfc7273_sync = DEFAULT_RFC7273_SYNC;
+ rtpbin->max_streams = DEFAULT_MAX_STREAMS;
+ rtpbin->max_ts_offset_adjustment = DEFAULT_MAX_TS_OFFSET_ADJUSTMENT;
+ rtpbin->max_ts_offset = DEFAULT_MAX_TS_OFFSET;
+ rtpbin->max_ts_offset_is_set = FALSE;
/* some default SDES entries */
cname = g_strdup_printf ("user%u@host-%x", g_random_int (), g_random_int ());
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);
break;
case PROP_NTP_SYNC:
rtpbin->ntp_sync = g_value_get_boolean (value);
+ /* The default value of max_ts_offset depends on ntp_sync. If user
+ * hasn't set it then change default value */
+ if (!rtpbin->max_ts_offset_is_set) {
+ if (rtpbin->ntp_sync) {
+ rtpbin->max_ts_offset = 0;
+ } else {
+ rtpbin->max_ts_offset = DEFAULT_MAX_TS_OFFSET;
+ }
+ }
break;
case PROP_RTCP_SYNC:
g_atomic_int_set (&rtpbin->rtcp_sync, g_value_get_enum (value));
GST_RTP_BIN_UNLOCK (rtpbin);
}
break;
+ case PROP_DO_SYNC_EVENT:
+ rtpbin->send_sync_event = g_value_get_boolean (value);
+ break;
case PROP_BUFFER_MODE:
GST_RTP_BIN_LOCK (rtpbin);
rtpbin->buffer_mode = g_value_get_enum (value);
/* propagate the property down to the jitterbuffer */
gst_rtp_bin_propagate_property_to_jitterbuffer (rtpbin, "mode", value);
break;
+ case PROP_DO_RETRANSMISSION:
+ GST_RTP_BIN_LOCK (rtpbin);
+ rtpbin->do_retransmission = g_value_get_boolean (value);
+ GST_RTP_BIN_UNLOCK (rtpbin);
+ gst_rtp_bin_propagate_property_to_jitterbuffer (rtpbin,
+ "do-retransmission", value);
+ break;
+ case PROP_RTP_PROFILE:
+ rtpbin->rtp_profile = g_value_get_enum (value);
+ break;
+ case PROP_NTP_TIME_SOURCE:{
+ GSList *sessions;
+ GST_RTP_BIN_LOCK (rtpbin);
+ rtpbin->ntp_time_source = g_value_get_enum (value);
+ for (sessions = rtpbin->sessions; sessions;
+ sessions = g_slist_next (sessions)) {
+ GstRtpBinSession *session = (GstRtpBinSession *) sessions->data;
+
+ g_object_set (G_OBJECT (session->session),
+ "ntp-time-source", rtpbin->ntp_time_source, NULL);
+ }
+ GST_RTP_BIN_UNLOCK (rtpbin);
+ break;
+ }
+ case PROP_RTCP_SYNC_SEND_TIME:{
+ GSList *sessions;
+ GST_RTP_BIN_LOCK (rtpbin);
+ rtpbin->rtcp_sync_send_time = g_value_get_boolean (value);
+ for (sessions = rtpbin->sessions; sessions;
+ sessions = g_slist_next (sessions)) {
+ GstRtpBinSession *session = (GstRtpBinSession *) sessions->data;
+
+ g_object_set (G_OBJECT (session->session),
+ "rtcp-sync-send-time", rtpbin->rtcp_sync_send_time, NULL);
+ }
+ GST_RTP_BIN_UNLOCK (rtpbin);
+ break;
+ }
+ case PROP_MAX_RTCP_RTP_TIME_DIFF:
+ GST_RTP_BIN_LOCK (rtpbin);
+ rtpbin->max_rtcp_rtp_time_diff = g_value_get_int (value);
+ GST_RTP_BIN_UNLOCK (rtpbin);
+ gst_rtp_bin_propagate_property_to_jitterbuffer (rtpbin,
+ "max-rtcp-rtp-time-diff", value);
+ break;
+ case PROP_MAX_DROPOUT_TIME:
+ GST_RTP_BIN_LOCK (rtpbin);
+ rtpbin->max_dropout_time = g_value_get_uint (value);
+ GST_RTP_BIN_UNLOCK (rtpbin);
+ gst_rtp_bin_propagate_property_to_jitterbuffer (rtpbin,
+ "max-dropout-time", value);
+ gst_rtp_bin_propagate_property_to_session (rtpbin, "max-dropout-time",
+ value);
+ break;
+ case PROP_MAX_MISORDER_TIME:
+ GST_RTP_BIN_LOCK (rtpbin);
+ rtpbin->max_misorder_time = g_value_get_uint (value);
+ GST_RTP_BIN_UNLOCK (rtpbin);
+ gst_rtp_bin_propagate_property_to_jitterbuffer (rtpbin,
+ "max-misorder-time", value);
+ gst_rtp_bin_propagate_property_to_session (rtpbin, "max-misorder-time",
+ value);
+ break;
+ case PROP_RFC7273_SYNC:
+ rtpbin->rfc7273_sync = g_value_get_boolean (value);
+ gst_rtp_bin_propagate_property_to_jitterbuffer (rtpbin,
+ "rfc7273-sync", value);
+ break;
+ case PROP_MAX_STREAMS:
+ rtpbin->max_streams = g_value_get_uint (value);
+ break;
+ case PROP_MAX_TS_OFFSET_ADJUSTMENT:
+ rtpbin->max_ts_offset_adjustment = g_value_get_uint64 (value);
+ gst_rtp_bin_propagate_property_to_jitterbuffer (rtpbin,
+ "max-ts-offset-adjustment", value);
+ break;
+ case PROP_MAX_TS_OFFSET:
+ rtpbin->max_ts_offset = g_value_get_int64 (value);
+ rtpbin->max_ts_offset_is_set = TRUE;
+ break;
default:
G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
break;
case PROP_USE_PIPELINE_CLOCK:
g_value_set_boolean (value, rtpbin->use_pipeline_clock);
break;
+ case PROP_DO_SYNC_EVENT:
+ g_value_set_boolean (value, rtpbin->send_sync_event);
+ break;
+ case PROP_DO_RETRANSMISSION:
+ GST_RTP_BIN_LOCK (rtpbin);
+ g_value_set_boolean (value, rtpbin->do_retransmission);
+ GST_RTP_BIN_UNLOCK (rtpbin);
+ break;
+ case PROP_RTP_PROFILE:
+ g_value_set_enum (value, rtpbin->rtp_profile);
+ break;
+ case PROP_NTP_TIME_SOURCE:
+ g_value_set_enum (value, rtpbin->ntp_time_source);
+ break;
+ case PROP_RTCP_SYNC_SEND_TIME:
+ g_value_set_boolean (value, rtpbin->rtcp_sync_send_time);
+ break;
+ case PROP_MAX_RTCP_RTP_TIME_DIFF:
+ GST_RTP_BIN_LOCK (rtpbin);
+ g_value_set_int (value, rtpbin->max_rtcp_rtp_time_diff);
+ GST_RTP_BIN_UNLOCK (rtpbin);
+ break;
+ case PROP_MAX_DROPOUT_TIME:
+ g_value_set_uint (value, rtpbin->max_dropout_time);
+ break;
+ case PROP_MAX_MISORDER_TIME:
+ g_value_set_uint (value, rtpbin->max_misorder_time);
+ break;
+ case PROP_RFC7273_SYNC:
+ g_value_set_boolean (value, rtpbin->rfc7273_sync);
+ break;
+ case PROP_MAX_STREAMS:
+ g_value_set_uint (value, rtpbin->max_streams);
+ break;
+ case PROP_MAX_TS_OFFSET_ADJUSTMENT:
+ g_value_set_uint64 (value, rtpbin->max_ts_offset_adjustment);
+ break;
+ case PROP_MAX_TS_OFFSET:
+ g_value_set_int64 (value, rtpbin->max_ts_offset);
+ break;
default:
G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
break;
gint min_percent = 100;
GSList *sessions, *streams;
GstRtpBinStream *stream;
+#ifdef TIZEN_FEATURE_RTSP_MODIFICATION
+ gboolean buffering_flag = FALSE, update_buffering_status = TRUE;
+#endif
gboolean change = FALSE, active = FALSE;
GstClockTime min_out_time;
GstBufferingMode mode;
for (streams = session->streams; streams;
streams = g_slist_next (streams)) {
GstRtpBinStream *stream = (GstRtpBinStream *) streams->data;
-
+#ifdef TIZEN_FEATURE_RTSP_MODIFICATION
+ GstPad *temp_pad_src = NULL;
+ GstCaps *temp_caps_src = NULL;
+ GstStructure *caps_structure;
+ const gchar *caps_str_media = NULL;
+ temp_pad_src = gst_element_get_static_pad (stream->buffer, "src");
+ temp_caps_src = gst_pad_get_current_caps (temp_pad_src);
+ GST_DEBUG_OBJECT (bin,
+ "stream %p percent %d : temp_caps_src=%" GST_PTR_FORMAT,
+ stream, stream->percent, temp_caps_src);
+ if (temp_caps_src) {
+ caps_structure = gst_caps_get_structure (temp_caps_src, 0);
+ caps_str_media =
+ gst_structure_get_string (caps_structure, "media");
+ if (caps_str_media != NULL) {
+ if ((strcmp (caps_str_media, "video") != 0)
+ && (strcmp (caps_str_media, "audio") != 0)) {
+ GST_DEBUG_OBJECT (bin,
+ "Non Audio/Video Stream.. ignoring the same !!");
+ gst_caps_unref (temp_caps_src);
+ gst_object_unref (temp_pad_src);
+ continue;
+ } else if (stream->percent >= 100) {
+ /* Most of the time buffering icon displays in rtsp playback.
+ Optimizing the buffering updation code. Whenever any stream percentage
+ reaches 100 do not post buffering messages. */
+ if (stream->prev_percent < 100)
+ buffering_flag = TRUE;
+ else
+ update_buffering_status = FALSE;
+ }
+ }
+ gst_caps_unref (temp_caps_src);
+ }
+ gst_object_unref (temp_pad_src);
+#else
GST_DEBUG_OBJECT (bin, "stream %p percent %d", stream,
stream->percent);
-
+#endif
/* find min percent */
if (min_percent > stream->percent)
min_percent = stream->percent;
+#ifdef TIZEN_FEATURE_RTSP_MODIFICATION
+ /* Updating prev stream percentage */
+ stream->prev_percent = stream->percent;
+#endif
}
} else {
GST_INFO_OBJECT (bin,
GST_RTP_SESSION_UNLOCK (session);
}
GST_DEBUG_OBJECT (bin, "min percent %d", min_percent);
-
+#ifdef TIZEN_FEATURE_RTSP_MODIFICATION
+ if (rtpbin->buffer_mode != RTP_JITTER_BUFFER_MODE_SLAVE) {
+ if (rtpbin->buffering) {
+ if (min_percent == 100) {
+ rtpbin->buffering = FALSE;
+ active = TRUE;
+ change = TRUE;
+ }
+ } else {
+ if (min_percent < 100) {
+ /* pause the streams */
+ rtpbin->buffering = TRUE;
+ active = FALSE;
+ change = TRUE;
+ }
+ }
+ }
+#else
if (rtpbin->buffering) {
if (min_percent == 100) {
rtpbin->buffering = FALSE;
change = TRUE;
}
}
+#endif
GST_RTP_BIN_UNLOCK (rtpbin);
gst_message_unref (message);
+#ifdef TIZEN_FEATURE_RTSP_MODIFICATION
+ if (rtpbin->buffer_mode == RTP_JITTER_BUFFER_MODE_SLAVE) {
+ if (update_buffering_status == FALSE)
+ break;
+ if (buffering_flag) {
+ min_percent = 100;
+ GST_DEBUG_OBJECT (bin, "forcefully change min_percent to 100!!!");
+ }
+ }
+#endif
/* make a new buffering message with the min value */
message =
gst_message_new_buffering (GST_OBJECT_CAST (bin), min_percent);
gst_message_set_buffering_stats (message, mode, avg_in, avg_out,
buffering_left);
+#ifdef TIZEN_FEATURE_RTSP_MODIFICATION
+ if (rtpbin->buffer_mode == RTP_JITTER_BUFFER_MODE_SLAVE)
+ goto slave_buffering;
+#endif
if (G_UNLIKELY (change)) {
GstClock *clock;
guint64 running_time = 0;
GST_RTP_BIN_UNLOCK (rtpbin);
}
}
+#ifdef TIZEN_FEATURE_RTSP_MODIFICATION
+slave_buffering:
+#endif
GST_BIN_CLASS (parent_class)->handle_message (bin, message);
break;
}
case GST_STATE_CHANGE_NULL_TO_READY:
break;
case GST_STATE_CHANGE_READY_TO_PAUSED:
- priv->last_unix = 0;
+ priv->last_ntpnstime = 0;
GST_LOG_OBJECT (rtpbin, "clearing shutdown flag");
g_atomic_int_set (&priv->shutdown, 0);
break;
return res;
}
-/* a new pad (SSRC) was created in @session. This signal is emited from the
- * payload demuxer. */
-static void
-new_payload_found (GstElement * element, guint pt, GstPad * pad,
- GstRtpBinStream * stream)
+static GstElement *
+session_request_element (GstRtpBinSession * session, guint signal)
{
- GstRtpBin *rtpbin;
- GstElementClass *klass;
- GstPadTemplate *templ;
- gchar *padname;
- GstPad *gpad;
+ GstElement *element = NULL;
+ GstRtpBin *bin = session->bin;
- rtpbin = stream->bin;
+ g_signal_emit (bin, gst_rtp_bin_signals[signal], 0, session->id, &element);
- GST_DEBUG ("new payload pad %d", pt);
+ if (element) {
+ if (!bin_manage_element (bin, element))
+ goto manage_failed;
+ session->elements = g_slist_prepend (session->elements, element);
+ }
+ return element;
- GST_RTP_BIN_SHUTDOWN_LOCK (rtpbin, shutdown);
+ /* ERRORS */
+manage_failed:
+ {
+ GST_WARNING_OBJECT (bin, "unable to manage element");
+ gst_object_unref (element);
+ return NULL;
+ }
+}
- /* ghost the pad to the parent */
+static gboolean
+copy_sticky_events (GstPad * pad, GstEvent ** event, gpointer user_data)
+{
+ GstPad *gpad = GST_PAD_CAST (user_data);
+
+ GST_DEBUG_OBJECT (gpad, "store sticky event %" GST_PTR_FORMAT, *event);
+ gst_pad_store_sticky_event (gpad, *event);
+
+ return TRUE;
+}
+
+static void
+expose_recv_src_pad (GstRtpBin * rtpbin, GstPad * pad, GstRtpBinStream * stream,
+ guint8 pt)
+{
+ GstElementClass *klass;
+ GstPadTemplate *templ;
+ gchar *padname;
+ GstPad *gpad;
+
+ gst_object_ref (pad);
+
+ if (stream->session->storage) {
+ GstElement *fec_decoder =
+ session_request_element (stream->session, SIGNAL_REQUEST_FEC_DECODER);
+
+ if (fec_decoder) {
+ GstPad *sinkpad, *srcpad;
+ GstPadLinkReturn ret;
+
+ sinkpad = gst_element_get_static_pad (fec_decoder, "sink");
+
+ if (!sinkpad)
+ goto fec_decoder_sink_failed;
+
+ ret = gst_pad_link (pad, sinkpad);
+ gst_object_unref (sinkpad);
+
+ if (ret != GST_PAD_LINK_OK)
+ goto fec_decoder_link_failed;
+
+ srcpad = gst_element_get_static_pad (fec_decoder, "src");
+
+ if (!srcpad)
+ goto fec_decoder_src_failed;
+
+ gst_pad_sticky_events_foreach (pad, copy_sticky_events, srcpad);
+ gst_object_unref (pad);
+ pad = srcpad;
+ }
+ }
+
+ GST_RTP_BIN_SHUTDOWN_LOCK (rtpbin, shutdown);
+
+ /* ghost the pad to the parent */
klass = GST_ELEMENT_GET_CLASS (rtpbin);
templ = gst_element_class_get_pad_template (klass, "recv_rtp_src_%u_%u_%u");
padname = g_strdup_printf ("recv_rtp_src_%u_%u_%u",
gst_pad_set_active (gpad, TRUE);
GST_RTP_BIN_SHUTDOWN_UNLOCK (rtpbin);
+ gst_pad_sticky_events_foreach (pad, copy_sticky_events, gpad);
gst_element_add_pad (GST_ELEMENT_CAST (rtpbin), gpad);
+done:
+ gst_object_unref (pad);
+
return;
shutdown:
{
GST_DEBUG ("ignoring, we are shutting down");
- return;
+ goto done;
+ }
+fec_decoder_sink_failed:
+ {
+ g_warning ("rtpbin: failed to get fec encoder sink pad for session %u",
+ stream->session->id);
+ goto done;
+ }
+fec_decoder_src_failed:
+ {
+ g_warning ("rtpbin: failed to get fec encoder src pad for session %u",
+ stream->session->id);
+ goto done;
+ }
+fec_decoder_link_failed:
+ {
+ g_warning ("rtpbin: failed to link fec decoder for session %u",
+ stream->session->id);
+ goto done;
}
}
+/* a new pad (SSRC) was created in @session. This signal is emited from the
+ * payload demuxer. */
+static void
+new_payload_found (GstElement * element, guint pt, GstPad * pad,
+ GstRtpBinStream * stream)
+{
+ GstRtpBin *rtpbin;
+
+ rtpbin = stream->bin;
+
+ GST_DEBUG_OBJECT (rtpbin, "new payload pad %u", pt);
+
+ expose_recv_src_pad (rtpbin, pad, stream, pt);
+}
+
static void
payload_pad_removed (GstElement * element, GstPad * pad,
GstRtpBinStream * stream)
rtpbin = session->bin;
- GST_DEBUG_OBJECT (rtpbin, "payload map requested for pt %d in session %d", pt,
+ GST_DEBUG_OBJECT (rtpbin, "payload map requested for pt %u in session %u", pt,
session->id);
caps = get_pt_map (session, pt);
}
}
+static GstCaps *
+ptdemux_pt_map_requested (GstElement * element, guint pt,
+ GstRtpBinSession * session)
+{
+ GstCaps *ret = pt_map_requested (element, pt, session);
+
+ if (ret && gst_caps_get_size (ret) == 1) {
+ const GstStructure *s = gst_caps_get_structure (ret, 0);
+ gboolean is_fec;
+
+ if (gst_structure_get_boolean (s, "is-fec", &is_fec) && is_fec) {
+ GValue v = G_VALUE_INIT;
+ GValue v2 = G_VALUE_INIT;
+
+ GST_INFO_OBJECT (session->bin, "Will ignore FEC pt %u in session %u", pt,
+ session->id);
+ g_value_init (&v, GST_TYPE_ARRAY);
+ g_value_init (&v2, G_TYPE_INT);
+ g_object_get_property (G_OBJECT (element), "ignored-payload-types", &v);
+ g_value_set_int (&v2, pt);
+ gst_value_array_append_value (&v, &v2);
+ g_value_unset (&v2);
+ g_object_set_property (G_OBJECT (element), "ignored-payload-types", &v);
+ g_value_unset (&v);
+ }
+ }
+
+ return ret;
+}
+
static void
payload_type_change (GstElement * element, guint pt, GstRtpBinSession * session)
{
GST_DEBUG_OBJECT (session->bin,
- "emiting signal for pt type changed to %d in session %d", pt,
+ "emiting signal for pt type changed to %u in session %u", pt,
session->id);
g_signal_emit (session->bin, gst_rtp_bin_signals[SIGNAL_PAYLOAD_TYPE_CHANGE],
0, session->id, pt);
}
-/* emited when caps changed for the session */
+/* emitted when caps changed for the session */
static void
caps_changed (GstPad * pad, GParamSpec * pspec, GstRtpBinSession * session)
{
s = gst_caps_get_structure (caps, 0);
/* get payload, finish when it's not there */
- if (!gst_structure_get_int (s, "payload", &payload))
+ if (!gst_structure_get_int (s, "payload", &payload)) {
+ gst_caps_unref (caps);
return;
+ }
GST_RTP_SESSION_LOCK (session);
GST_DEBUG_OBJECT (bin, "insert caps for payload %d", payload);
srcpad = gst_element_get_static_pad (element, padname);
g_free (padname);
sinkpad = gst_element_get_static_pad (stream->buffer, "sink");
- gst_pad_link (srcpad, sinkpad);
+ gst_pad_link_full (srcpad, sinkpad, GST_PAD_LINK_CHECK_NOTHING);
gst_object_unref (sinkpad);
gst_object_unref (srcpad);
srcpad = gst_element_get_static_pad (element, padname);
g_free (padname);
sinkpad = gst_element_get_request_pad (stream->buffer, "sink_rtcp");
- gst_pad_link (srcpad, sinkpad);
+ gst_pad_link_full (srcpad, sinkpad, GST_PAD_LINK_CHECK_NOTHING);
gst_object_unref (sinkpad);
gst_object_unref (srcpad);
stream->demux_padremoved_sig = g_signal_connect (stream->demux,
"pad-removed", (GCallback) payload_pad_removed, stream);
- /* connect to the request-pt-map signal. This signal will be emited by the
+ /* connect to the request-pt-map signal. This signal will be emitted by the
* demuxer so that it can apply a proper caps on the buffers for the
* depayloaders. */
stream->demux_ptreq_sig = g_signal_connect (stream->demux,
- "request-pt-map", (GCallback) pt_map_requested, session);
+ "request-pt-map", (GCallback) ptdemux_pt_map_requested, session);
/* connect to the signal so it can be forwarded. */
stream->demux_ptchange_sig = g_signal_connect (stream->demux,
"payload-type-change", (GCallback) payload_type_change, session);
+
+ GST_RTP_SESSION_UNLOCK (session);
+ GST_RTP_BIN_SHUTDOWN_UNLOCK (rtpbin);
} else {
- /* add gstrtpjitterbuffer src pad to pads */
- GstElementClass *klass;
- GstPadTemplate *templ;
- gchar *padname;
- GstPad *gpad, *pad;
+ /* add rtpjitterbuffer src pad to pads */
+ GstPad *pad;
pad = gst_element_get_static_pad (stream->buffer, "src");
- /* ghost the pad to the parent */
- klass = GST_ELEMENT_GET_CLASS (rtpbin);
- 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_RTP_SESSION_UNLOCK (session);
+ GST_RTP_BIN_SHUTDOWN_UNLOCK (rtpbin);
- gst_pad_set_active (gpad, TRUE);
- gst_element_add_pad (GST_ELEMENT_CAST (rtpbin), gpad);
+ expose_recv_src_pad (rtpbin, pad, stream, 255);
gst_object_unref (pad);
}
- GST_RTP_SESSION_UNLOCK (session);
- GST_RTP_BIN_SHUTDOWN_UNLOCK (rtpbin);
-
return;
/* ERRORS */
}
}
+static GstPad *
+complete_session_sink (GstRtpBin * rtpbin, GstRtpBinSession * session)
+{
+ guint sessid = session->id;
+ GstPad *recv_rtp_sink;
+ GstElement *decoder;
+
+ g_assert (!session->recv_rtp_sink);
+
+ /* get recv_rtp pad and store */
+ session->recv_rtp_sink =
+ gst_element_get_request_pad (session->session, "recv_rtp_sink");
+ if (session->recv_rtp_sink == NULL)
+ goto pad_failed;
+
+ g_signal_connect (session->recv_rtp_sink, "notify::caps",
+ (GCallback) caps_changed, session);
+
+ GST_DEBUG_OBJECT (rtpbin, "requesting RTP decoder");
+ decoder = session_request_element (session, SIGNAL_REQUEST_RTP_DECODER);
+ if (decoder) {
+ GstPad *decsrc, *decsink;
+ GstPadLinkReturn ret;
+
+ GST_DEBUG_OBJECT (rtpbin, "linking RTP decoder");
+ decsink = gst_element_get_static_pad (decoder, "rtp_sink");
+ if (decsink == NULL)
+ goto dec_sink_failed;
+
+ recv_rtp_sink = decsink;
+
+ decsrc = gst_element_get_static_pad (decoder, "rtp_src");
+ if (decsrc == NULL)
+ goto dec_src_failed;
+
+ ret = gst_pad_link (decsrc, session->recv_rtp_sink);
+
+ gst_object_unref (decsrc);
+
+ if (ret != GST_PAD_LINK_OK)
+ goto dec_link_failed;
+
+ } else {
+ GST_DEBUG_OBJECT (rtpbin, "no RTP decoder given");
+ recv_rtp_sink = gst_object_ref (session->recv_rtp_sink);
+ }
+
+ return recv_rtp_sink;
+
+ /* ERRORS */
+pad_failed:
+ {
+ g_warning ("rtpbin: failed to get session recv_rtp_sink pad");
+ return NULL;
+ }
+dec_sink_failed:
+ {
+ g_warning ("rtpbin: failed to get decoder sink pad for session %u", sessid);
+ return NULL;
+ }
+dec_src_failed:
+ {
+ g_warning ("rtpbin: failed to get decoder src pad for session %u", sessid);
+ gst_object_unref (recv_rtp_sink);
+ return NULL;
+ }
+dec_link_failed:
+ {
+ g_warning ("rtpbin: failed to link rtp decoder for session %u", sessid);
+ gst_object_unref (recv_rtp_sink);
+ return NULL;
+ }
+}
+
+static void
+complete_session_receiver (GstRtpBin * rtpbin, GstRtpBinSession * session,
+ guint sessid)
+{
+ GstElement *aux;
+ GstPad *recv_rtp_src;
+
+ g_assert (!session->recv_rtp_src);
+
+ session->recv_rtp_src =
+ gst_element_get_static_pad (session->session, "recv_rtp_src");
+ if (session->recv_rtp_src == NULL)
+ goto pad_failed;
+
+ /* find out if we need AUX elements */
+ aux = session_request_element (session, SIGNAL_REQUEST_AUX_RECEIVER);
+ if (aux) {
+ gchar *pname;
+ GstPad *auxsink;
+ GstPadLinkReturn ret;
+
+ GST_DEBUG_OBJECT (rtpbin, "linking AUX receiver");
+
+ pname = g_strdup_printf ("sink_%u", sessid);
+ auxsink = gst_element_get_static_pad (aux, pname);
+ g_free (pname);
+ if (auxsink == NULL)
+ goto aux_sink_failed;
+
+ ret = gst_pad_link (session->recv_rtp_src, auxsink);
+ gst_object_unref (auxsink);
+ if (ret != GST_PAD_LINK_OK)
+ goto aux_link_failed;
+
+ /* this can be NULL when this AUX element is not to be linked any further */
+ pname = g_strdup_printf ("src_%u", sessid);
+ recv_rtp_src = gst_element_get_static_pad (aux, pname);
+ g_free (pname);
+ } else {
+ recv_rtp_src = gst_object_ref (session->recv_rtp_src);
+ }
+
+ /* Add a storage element if needed */
+ if (recv_rtp_src && session->storage) {
+ GstPadLinkReturn ret;
+ GstPad *sinkpad = gst_element_get_static_pad (session->storage, "sink");
+
+ ret = gst_pad_link (recv_rtp_src, sinkpad);
+
+ gst_object_unref (sinkpad);
+ gst_object_unref (recv_rtp_src);
+
+ if (ret != GST_PAD_LINK_OK)
+ goto storage_link_failed;
+
+ recv_rtp_src = gst_element_get_static_pad (session->storage, "src");
+ }
+
+ if (recv_rtp_src) {
+ GstPad *sinkdpad;
+
+ GST_DEBUG_OBJECT (rtpbin, "getting demuxer RTP sink pad");
+ sinkdpad = gst_element_get_static_pad (session->demux, "sink");
+ GST_DEBUG_OBJECT (rtpbin, "linking demuxer RTP sink pad");
+ gst_pad_link_full (recv_rtp_src, sinkdpad, GST_PAD_LINK_CHECK_NOTHING);
+ gst_object_unref (sinkdpad);
+ gst_object_unref (recv_rtp_src);
+
+ /* connect to the new-ssrc-pad signal of the SSRC demuxer */
+ session->demux_newpad_sig = g_signal_connect (session->demux,
+ "new-ssrc-pad", (GCallback) new_ssrc_pad_found, session);
+ session->demux_padremoved_sig = g_signal_connect (session->demux,
+ "removed-ssrc-pad", (GCallback) ssrc_demux_pad_removed, session);
+ }
+
+ return;
+
+pad_failed:
+ {
+ g_warning ("rtpbin: failed to get session recv_rtp_src pad");
+ return;
+ }
+aux_sink_failed:
+ {
+ g_warning ("rtpbin: failed to get AUX sink pad for session %u", sessid);
+ return;
+ }
+aux_link_failed:
+ {
+ g_warning ("rtpbin: failed to link AUX pad to session %u", sessid);
+ return;
+ }
+storage_link_failed:
+ {
+ g_warning ("rtpbin: failed to link storage");
+ return;
+ }
+}
+
/* Create a pad for receiving RTP for the session in @name. Must be called with
* RTP_BIN_LOCK.
*/
static GstPad *
create_recv_rtp (GstRtpBin * rtpbin, GstPadTemplate * templ, const gchar * name)
{
- GstPad *sinkdpad;
guint sessid;
GstRtpBinSession *session;
- GstPadLinkReturn lres;
+ GstPad *recv_rtp_sink;
/* first get the session number */
if (name == NULL || sscanf (name, "recv_rtp_sink_%u", &sessid) != 1)
goto no_name;
- GST_DEBUG_OBJECT (rtpbin, "finding session %d", sessid);
+ GST_DEBUG_OBJECT (rtpbin, "finding session %u", sessid);
/* get or create session */
session = find_session_by_id (rtpbin, sessid);
if (!session) {
- GST_DEBUG_OBJECT (rtpbin, "creating session %d", sessid);
+ GST_DEBUG_OBJECT (rtpbin, "creating session %u", sessid);
/* create session now */
session = create_session (rtpbin, sessid);
if (session == NULL)
if (session->recv_rtp_sink_ghost != NULL)
return session->recv_rtp_sink_ghost;
- GST_DEBUG_OBJECT (rtpbin, "getting RTP sink pad");
- /* get recv_rtp pad and store */
- session->recv_rtp_sink =
- gst_element_get_request_pad (session->session, "recv_rtp_sink");
- if (session->recv_rtp_sink == NULL)
- goto pad_failed;
-
- g_signal_connect (session->recv_rtp_sink, "notify::caps",
- (GCallback) caps_changed, session);
-
- GST_DEBUG_OBJECT (rtpbin, "getting RTP src pad");
- /* get srcpad, link to SSRCDemux */
- session->recv_rtp_src =
- gst_element_get_static_pad (session->session, "recv_rtp_src");
- if (session->recv_rtp_src == NULL)
- goto pad_failed;
-
- GST_DEBUG_OBJECT (rtpbin, "getting demuxer RTP sink pad");
- sinkdpad = gst_element_get_static_pad (session->demux, "sink");
- GST_DEBUG_OBJECT (rtpbin, "linking demuxer RTP sink pad");
- lres = gst_pad_link (session->recv_rtp_src, sinkdpad);
- gst_object_unref (sinkdpad);
- if (lres != GST_PAD_LINK_OK)
- goto link_failed;
-
- /* connect to the new-ssrc-pad signal of the SSRC demuxer */
- session->demux_newpad_sig = g_signal_connect (session->demux,
- "new-ssrc-pad", (GCallback) new_ssrc_pad_found, session);
- session->demux_padremoved_sig = g_signal_connect (session->demux,
- "removed-ssrc-pad", (GCallback) ssrc_demux_pad_removed, session);
+ /* setup the session sink pad */
+ recv_rtp_sink = complete_session_sink (rtpbin, session);
+ if (!recv_rtp_sink)
+ goto session_sink_failed;
GST_DEBUG_OBJECT (rtpbin, "ghosting session sink pad");
session->recv_rtp_sink_ghost =
- gst_ghost_pad_new_from_template (name, session->recv_rtp_sink, templ);
+ gst_ghost_pad_new_from_template (name, recv_rtp_sink, templ);
+ gst_object_unref (recv_rtp_sink);
gst_pad_set_active (session->recv_rtp_sink_ghost, TRUE);
gst_element_add_pad (GST_ELEMENT_CAST (rtpbin), session->recv_rtp_sink_ghost);
+ complete_session_receiver (rtpbin, session, sessid);
+
return session->recv_rtp_sink_ghost;
/* ERRORS */
/* create_session already warned */
return NULL;
}
-pad_failed:
+session_sink_failed:
{
- g_warning ("rtpbin: failed to get session pad");
- return NULL;
- }
-link_failed:
- {
- g_warning ("rtpbin: failed to link pads");
+ /* warning already done */
return NULL;
}
}
}
}
+static GstPad *
+complete_session_rtcp (GstRtpBin * rtpbin, GstRtpBinSession * session,
+ guint sessid)
+{
+ GstElement *decoder;
+ GstPad *sinkdpad;
+ GstPad *decsink = NULL;
+
+ /* get recv_rtp pad and store */
+ GST_DEBUG_OBJECT (rtpbin, "getting RTCP sink pad");
+ session->recv_rtcp_sink =
+ gst_element_get_request_pad (session->session, "recv_rtcp_sink");
+ if (session->recv_rtcp_sink == NULL)
+ goto pad_failed;
+
+ GST_DEBUG_OBJECT (rtpbin, "getting RTCP decoder");
+ decoder = session_request_element (session, SIGNAL_REQUEST_RTCP_DECODER);
+ if (decoder) {
+ GstPad *decsrc;
+ GstPadLinkReturn ret;
+
+ GST_DEBUG_OBJECT (rtpbin, "linking RTCP decoder");
+ decsink = gst_element_get_static_pad (decoder, "rtcp_sink");
+ decsrc = gst_element_get_static_pad (decoder, "rtcp_src");
+
+ if (decsink == NULL)
+ goto dec_sink_failed;
+
+ if (decsrc == NULL)
+ goto dec_src_failed;
+
+ ret = gst_pad_link (decsrc, session->recv_rtcp_sink);
+
+ gst_object_unref (decsrc);
+
+ if (ret != GST_PAD_LINK_OK)
+ goto dec_link_failed;
+ } else {
+ GST_DEBUG_OBJECT (rtpbin, "no RTCP decoder given");
+ decsink = gst_object_ref (session->recv_rtcp_sink);
+ }
+
+ /* get srcpad, link to SSRCDemux */
+ GST_DEBUG_OBJECT (rtpbin, "getting sync src pad");
+ session->sync_src = gst_element_get_static_pad (session->session, "sync_src");
+ if (session->sync_src == NULL)
+ goto src_pad_failed;
+
+ GST_DEBUG_OBJECT (rtpbin, "getting demuxer RTCP sink pad");
+ sinkdpad = gst_element_get_static_pad (session->demux, "rtcp_sink");
+ gst_pad_link_full (session->sync_src, sinkdpad, GST_PAD_LINK_CHECK_NOTHING);
+ gst_object_unref (sinkdpad);
+
+ return decsink;
+
+pad_failed:
+ {
+ g_warning ("rtpbin: failed to get session rtcp_sink pad");
+ return NULL;
+ }
+dec_sink_failed:
+ {
+ g_warning ("rtpbin: failed to get decoder sink pad for session %u", sessid);
+ return NULL;
+ }
+dec_src_failed:
+ {
+ g_warning ("rtpbin: failed to get decoder src pad for session %u", sessid);
+ goto cleanup;
+ }
+dec_link_failed:
+ {
+ g_warning ("rtpbin: failed to link rtcp decoder for session %u", sessid);
+ goto cleanup;
+ }
+src_pad_failed:
+ {
+ g_warning ("rtpbin: failed to get session sync_src pad");
+ }
+
+cleanup:
+ gst_object_unref (decsink);
+ return NULL;
+}
+
/* Create a pad for receiving RTCP for the session in @name. Must be called with
* RTP_BIN_LOCK.
*/
{
guint sessid;
GstRtpBinSession *session;
- GstPad *sinkdpad;
- GstPadLinkReturn lres;
+ GstPad *decsink = NULL;
/* first get the session number */
if (name == NULL || sscanf (name, "recv_rtcp_sink_%u", &sessid) != 1)
goto no_name;
- GST_DEBUG_OBJECT (rtpbin, "finding session %d", sessid);
+ GST_DEBUG_OBJECT (rtpbin, "finding session %u", sessid);
/* get or create the session */
session = find_session_by_id (rtpbin, sessid);
if (!session) {
- GST_DEBUG_OBJECT (rtpbin, "creating session %d", sessid);
+ GST_DEBUG_OBJECT (rtpbin, "creating session %u", sessid);
/* create session now */
session = create_session (rtpbin, sessid);
if (session == NULL)
if (session->recv_rtcp_sink_ghost != NULL)
return session->recv_rtcp_sink_ghost;
- /* get recv_rtp pad and store */
- GST_DEBUG_OBJECT (rtpbin, "getting RTCP sink pad");
- session->recv_rtcp_sink =
- gst_element_get_request_pad (session->session, "recv_rtcp_sink");
- if (session->recv_rtcp_sink == NULL)
- goto pad_failed;
-
- /* get srcpad, link to SSRCDemux */
- GST_DEBUG_OBJECT (rtpbin, "getting sync src pad");
- session->sync_src = gst_element_get_static_pad (session->session, "sync_src");
- if (session->sync_src == NULL)
- goto pad_failed;
-
- GST_DEBUG_OBJECT (rtpbin, "getting demuxer RTCP sink pad");
- sinkdpad = gst_element_get_static_pad (session->demux, "rtcp_sink");
- lres = gst_pad_link (session->sync_src, sinkdpad);
- gst_object_unref (sinkdpad);
- if (lres != GST_PAD_LINK_OK)
- goto link_failed;
+ decsink = complete_session_rtcp (rtpbin, session, sessid);
+ if (!decsink)
+ goto create_error;
session->recv_rtcp_sink_ghost =
- gst_ghost_pad_new_from_template (name, session->recv_rtcp_sink, templ);
+ gst_ghost_pad_new_from_template (name, decsink, templ);
+ gst_object_unref (decsink);
gst_pad_set_active (session->recv_rtcp_sink_ghost, TRUE);
gst_element_add_pad (GST_ELEMENT_CAST (rtpbin),
session->recv_rtcp_sink_ghost);
/* create_session already warned */
return NULL;
}
-pad_failed:
- {
- g_warning ("rtpbin: failed to get session pad");
- return NULL;
- }
-link_failed:
- {
- g_warning ("rtpbin: failed to link pads");
- return NULL;
- }
}
static void
}
}
+static gboolean
+complete_session_src (GstRtpBin * rtpbin, GstRtpBinSession * session)
+{
+ gchar *gname;
+ guint sessid = session->id;
+ GstPad *send_rtp_src;
+ GstElement *encoder;
+ GstElementClass *klass;
+ GstPadTemplate *templ;
+ gboolean ret = FALSE;
+
+ /* get srcpad */
+ send_rtp_src = gst_element_get_static_pad (session->session, "send_rtp_src");
+
+ if (send_rtp_src == NULL)
+ goto no_srcpad;
+
+ GST_DEBUG_OBJECT (rtpbin, "getting RTP encoder");
+ encoder = session_request_element (session, SIGNAL_REQUEST_RTP_ENCODER);
+ if (encoder) {
+ gchar *ename;
+ GstPad *encsrc, *encsink;
+ GstPadLinkReturn ret;
+
+ GST_DEBUG_OBJECT (rtpbin, "linking RTP encoder");
+ ename = g_strdup_printf ("rtp_src_%u", sessid);
+ encsrc = gst_element_get_static_pad (encoder, ename);
+ g_free (ename);
+
+ if (encsrc == NULL)
+ goto enc_src_failed;
+
+ ename = g_strdup_printf ("rtp_sink_%u", sessid);
+ encsink = gst_element_get_static_pad (encoder, ename);
+ g_free (ename);
+ if (encsink == NULL)
+ goto enc_sink_failed;
+
+ ret = gst_pad_link (send_rtp_src, encsink);
+ gst_object_unref (encsink);
+ gst_object_unref (send_rtp_src);
+
+ send_rtp_src = encsrc;
+
+ if (ret != GST_PAD_LINK_OK)
+ goto enc_link_failed;
+ } else {
+ GST_DEBUG_OBJECT (rtpbin, "no RTP encoder given");
+ }
+
+ /* ghost the new source pad */
+ klass = GST_ELEMENT_GET_CLASS (rtpbin);
+ 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, send_rtp_src, templ);
+ gst_pad_set_active (session->send_rtp_src_ghost, TRUE);
+ gst_pad_sticky_events_foreach (send_rtp_src, copy_sticky_events,
+ session->send_rtp_src_ghost);
+ gst_element_add_pad (GST_ELEMENT_CAST (rtpbin), session->send_rtp_src_ghost);
+ g_free (gname);
+
+ ret = TRUE;
+
+done:
+ if (send_rtp_src)
+ gst_object_unref (send_rtp_src);
+
+ return ret;
+
+ /* ERRORS */
+no_srcpad:
+ {
+ g_warning ("rtpbin: failed to get rtp source pad for session %u", sessid);
+ goto done;
+ }
+enc_src_failed:
+ {
+ g_warning ("rtpbin: failed to get %" GST_PTR_FORMAT
+ " src pad for session %u", encoder, sessid);
+ goto done;
+ }
+enc_sink_failed:
+ {
+ g_warning ("rtpbin: failed to get %" GST_PTR_FORMAT
+ " sink pad for session %u", encoder, sessid);
+ goto done;
+ }
+enc_link_failed:
+ {
+ g_warning ("rtpbin: failed to link %" GST_PTR_FORMAT " for session %u",
+ encoder, sessid);
+ goto done;
+ }
+}
+
+static gboolean
+setup_aux_sender_fold (const GValue * item, GValue * result, gpointer user_data)
+{
+ GstPad *pad;
+ gchar *name;
+ guint sessid;
+ GstRtpBinSession *session = user_data, *newsess;
+ GstRtpBin *rtpbin = session->bin;
+ GstPadLinkReturn ret;
+
+ pad = g_value_get_object (item);
+ name = gst_pad_get_name (pad);
+
+ if (name == NULL || sscanf (name, "src_%u", &sessid) != 1)
+ goto no_name;
+
+ g_free (name);
+
+ newsess = find_session_by_id (rtpbin, sessid);
+ if (newsess == NULL) {
+ /* create new session */
+ newsess = create_session (rtpbin, sessid);
+ if (newsess == NULL)
+ goto create_error;
+ } else if (newsess->send_rtp_sink != NULL)
+ goto existing_session;
+
+ /* get send_rtp pad and store */
+ newsess->send_rtp_sink =
+ gst_element_get_request_pad (newsess->session, "send_rtp_sink");
+ if (newsess->send_rtp_sink == NULL)
+ goto pad_failed;
+
+ ret = gst_pad_link (pad, newsess->send_rtp_sink);
+ if (ret != GST_PAD_LINK_OK)
+ goto aux_link_failed;
+
+ if (!complete_session_src (rtpbin, newsess))
+ goto session_src_failed;
+
+ return TRUE;
+
+ /* ERRORS */
+no_name:
+ {
+ GST_WARNING ("ignoring invalid pad name %s", GST_STR_NULL (name));
+ g_free (name);
+ return TRUE;
+ }
+create_error:
+ {
+ /* create_session already warned */
+ return FALSE;
+ }
+existing_session:
+ {
+ GST_DEBUG_OBJECT (rtpbin,
+ "skipping src_%i setup, since it is already configured.", sessid);
+ return TRUE;
+ }
+pad_failed:
+ {
+ g_warning ("rtpbin: failed to get session pad for session %u", sessid);
+ return FALSE;
+ }
+aux_link_failed:
+ {
+ g_warning ("rtpbin: failed to link AUX for session %u", sessid);
+ return FALSE;
+ }
+session_src_failed:
+ {
+ g_warning ("rtpbin: failed to complete AUX for session %u", sessid);
+ return FALSE;
+ }
+}
+
+static gboolean
+setup_aux_sender (GstRtpBin * rtpbin, GstRtpBinSession * session,
+ GstElement * aux)
+{
+ GstIterator *it;
+ GValue result = { 0, };
+ GstIteratorResult res;
+
+ it = gst_element_iterate_src_pads (aux);
+ res = gst_iterator_fold (it, setup_aux_sender_fold, &result, session);
+ gst_iterator_free (it);
+
+ return res == GST_ITERATOR_DONE;
+}
+
/* Create a pad for sending RTP for the session in @name. Must be called with
* RTP_BIN_LOCK.
*/
static GstPad *
create_send_rtp (GstRtpBin * rtpbin, GstPadTemplate * templ, const gchar * name)
{
- gchar *gname;
+ gchar *pname;
guint sessid;
+ GstPad *send_rtp_sink;
+ GstElement *aux;
+ GstElement *encoder;
+ GstElement *prev = NULL;
GstRtpBinSession *session;
- GstElementClass *klass;
/* first get the session number */
if (name == NULL || sscanf (name, "send_rtp_sink_%u", &sessid) != 1)
if (session->send_rtp_sink_ghost != NULL)
return session->send_rtp_sink_ghost;
- /* get send_rtp pad and store */
- session->send_rtp_sink =
- gst_element_get_request_pad (session->session, "send_rtp_sink");
- if (session->send_rtp_sink == NULL)
- goto pad_failed;
+ /* check if we are already using this session as a sender */
+ if (session->send_rtp_sink != NULL)
+ goto existing_session;
+
+ encoder = session_request_element (session, SIGNAL_REQUEST_FEC_ENCODER);
+
+ if (encoder) {
+ GST_DEBUG_OBJECT (rtpbin, "Linking FEC encoder");
+
+ send_rtp_sink = gst_element_get_static_pad (encoder, "sink");
+
+ if (!send_rtp_sink)
+ goto enc_sink_failed;
+
+ prev = encoder;
+ }
+
+ GST_DEBUG_OBJECT (rtpbin, "getting RTP AUX sender");
+ aux = session_request_element (session, SIGNAL_REQUEST_AUX_SENDER);
+ if (aux) {
+ GstPad *sinkpad;
+ GST_DEBUG_OBJECT (rtpbin, "linking AUX sender");
+ if (!setup_aux_sender (rtpbin, session, aux))
+ goto aux_session_failed;
+
+ pname = g_strdup_printf ("sink_%u", sessid);
+ sinkpad = gst_element_get_static_pad (aux, pname);
+ g_free (pname);
+
+ if (sinkpad == NULL)
+ goto aux_sink_failed;
+
+ if (!prev) {
+ send_rtp_sink = sinkpad;
+ } else {
+ GstPad *srcpad = gst_element_get_static_pad (prev, "src");
+ GstPadLinkReturn ret;
+
+ ret = gst_pad_link (srcpad, sinkpad);
+ gst_object_unref (srcpad);
+ if (ret != GST_PAD_LINK_OK) {
+ goto aux_link_failed;
+ }
+ }
+ prev = aux;
+ } else {
+ /* get send_rtp pad and store */
+ session->send_rtp_sink =
+ gst_element_get_request_pad (session->session, "send_rtp_sink");
+ if (session->send_rtp_sink == NULL)
+ goto pad_failed;
+
+ if (!complete_session_src (rtpbin, session))
+ goto session_src_failed;
+
+ if (!prev) {
+ send_rtp_sink = gst_object_ref (session->send_rtp_sink);
+ } else {
+ GstPad *srcpad = gst_element_get_static_pad (prev, "src");
+ GstPadLinkReturn ret;
+
+ ret = gst_pad_link (srcpad, session->send_rtp_sink);
+ gst_object_unref (srcpad);
+ if (ret != GST_PAD_LINK_OK)
+ goto session_link_failed;
+ }
+ }
session->send_rtp_sink_ghost =
- gst_ghost_pad_new_from_template (name, session->send_rtp_sink, templ);
+ gst_ghost_pad_new_from_template (name, send_rtp_sink, templ);
+ gst_object_unref (send_rtp_sink);
gst_pad_set_active (session->send_rtp_sink_ghost, TRUE);
gst_element_add_pad (GST_ELEMENT_CAST (rtpbin), session->send_rtp_sink_ghost);
- /* get srcpad */
- session->send_rtp_src =
- gst_element_get_static_pad (session->session, "send_rtp_src");
- if (session->send_rtp_src == NULL)
- goto no_srcpad;
-
- /* ghost the new source pad */
- klass = GST_ELEMENT_GET_CLASS (rtpbin);
- 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);
- gst_element_add_pad (GST_ELEMENT_CAST (rtpbin), session->send_rtp_src_ghost);
- g_free (gname);
-
return session->send_rtp_sink_ghost;
/* ERRORS */
/* create_session already warned */
return NULL;
}
+existing_session:
+ {
+ g_warning ("rtpbin: session %u is already in use", sessid);
+ return NULL;
+ }
+aux_session_failed:
+ {
+ g_warning ("rtpbin: failed to get AUX sink pad for session %u", sessid);
+ return NULL;
+ }
+aux_sink_failed:
+ {
+ g_warning ("rtpbin: failed to get AUX sink pad for session %u", sessid);
+ return NULL;
+ }
+aux_link_failed:
+ {
+ g_warning ("rtpbin: failed to link %" GST_PTR_FORMAT " for session %u",
+ aux, sessid);
+ return NULL;
+ }
pad_failed:
{
- g_warning ("rtpbin: failed to get session pad for session %d", sessid);
+ g_warning ("rtpbin: failed to get session pad for session %u", sessid);
return NULL;
}
-no_srcpad:
+session_src_failed:
+ {
+ g_warning ("rtpbin: failed to setup source pads for session %u", sessid);
+ return NULL;
+ }
+session_link_failed:
{
- g_warning ("rtpbin: failed to get rtp source pad for session %d", sessid);
+ g_warning ("rtpbin: failed to link %" GST_PTR_FORMAT " for session %u",
+ session, sessid);
+ return NULL;
+ }
+enc_sink_failed:
+ {
+ g_warning ("rtpbin: failed to get %" GST_PTR_FORMAT
+ " sink pad for session %u", encoder, sessid);
return NULL;
}
}
session->send_rtp_src_ghost);
session->send_rtp_src_ghost = NULL;
}
- if (session->send_rtp_src) {
- gst_object_unref (session->send_rtp_src);
- session->send_rtp_src = NULL;
- }
if (session->send_rtp_sink) {
gst_element_release_request_pad (GST_ELEMENT_CAST (session->session),
session->send_rtp_sink);
* RTP_BIN_LOCK.
*/
static GstPad *
-create_rtcp (GstRtpBin * rtpbin, GstPadTemplate * templ, const gchar * name)
+create_send_rtcp (GstRtpBin * rtpbin, GstPadTemplate * templ,
+ const gchar * name)
{
guint sessid;
+ GstPad *encsrc;
+ GstElement *encoder;
GstRtpBinSession *session;
/* first get the session number */
/* get or create session */
session = find_session_by_id (rtpbin, sessid);
- if (!session)
- goto no_session;
+ if (!session) {
+ GST_DEBUG_OBJECT (rtpbin, "creating session %u", sessid);
+ /* create session now */
+ session = create_session (rtpbin, sessid);
+ if (session == NULL)
+ goto create_error;
+ }
/* check if pad was requested */
if (session->send_rtcp_src_ghost != NULL)
if (session->send_rtcp_src == NULL)
goto pad_failed;
+ GST_DEBUG_OBJECT (rtpbin, "getting RTCP encoder");
+ encoder = session_request_element (session, SIGNAL_REQUEST_RTCP_ENCODER);
+ if (encoder) {
+ gchar *ename;
+ GstPad *encsink;
+ GstPadLinkReturn ret;
+
+ GST_DEBUG_OBJECT (rtpbin, "linking RTCP encoder");
+
+ ename = g_strdup_printf ("rtcp_src_%u", sessid);
+ encsrc = gst_element_get_static_pad (encoder, ename);
+ g_free (ename);
+ if (encsrc == NULL)
+ goto enc_src_failed;
+
+ ename = g_strdup_printf ("rtcp_sink_%u", sessid);
+ encsink = gst_element_get_static_pad (encoder, ename);
+ g_free (ename);
+ if (encsink == NULL)
+ goto enc_sink_failed;
+
+ ret = gst_pad_link (session->send_rtcp_src, encsink);
+ gst_object_unref (encsink);
+
+ if (ret != GST_PAD_LINK_OK)
+ goto enc_link_failed;
+ } else {
+ GST_DEBUG_OBJECT (rtpbin, "no RTCP encoder given");
+ encsrc = gst_object_ref (session->send_rtcp_src);
+ }
+
session->send_rtcp_src_ghost =
- gst_ghost_pad_new_from_template (name, session->send_rtcp_src, templ);
+ gst_ghost_pad_new_from_template (name, encsrc, templ);
+ gst_object_unref (encsrc);
gst_pad_set_active (session->send_rtcp_src_ghost, TRUE);
gst_element_add_pad (GST_ELEMENT_CAST (rtpbin), session->send_rtcp_src_ghost);
g_warning ("rtpbin: invalid name given");
return NULL;
}
-no_session:
+create_error:
{
- g_warning ("rtpbin: session with id %d does not exist", sessid);
+ /* create_session already warned */
return NULL;
}
pad_failed:
{
- g_warning ("rtpbin: failed to get rtcp pad for session %d", sessid);
+ g_warning ("rtpbin: failed to get rtcp pad for session %u", sessid);
+ return NULL;
+ }
+enc_src_failed:
+ {
+ g_warning ("rtpbin: failed to get encoder src pad for session %u", sessid);
+ return NULL;
+ }
+enc_sink_failed:
+ {
+ g_warning ("rtpbin: failed to get encoder sink pad for session %u", sessid);
+ gst_object_unref (encsrc);
+ return NULL;
+ }
+enc_link_failed:
+ {
+ g_warning ("rtpbin: failed to link rtcp encoder for session %u", sessid);
+ gst_object_unref (encsrc);
return NULL;
}
}
result = create_send_rtp (rtpbin, templ, pad_name);
} else if (templ == gst_element_class_get_pad_template (klass,
"send_rtcp_src_%u")) {
- result = create_rtcp (rtpbin, templ, pad_name);
+ result = create_send_rtcp (rtpbin, templ, pad_name);
} else
goto wrong_template;
projects / platform / upstream / gst-plugins-good.git / blobdiff