- /* 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.
- */
+/* 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-rtpbin
+ * @title: rtpbin
* @see_also: rtpjitterbuffer, rtpsession, rtpptdemux, rtpssrcdemux
*
* RTP bin combines the functions of #GstRtpSession, #GstRtpSsrcDemux,
* To use #GstRtpBin as a sender, request a send_rtp_sink_\%u pad, which will
* automatically create a send_rtp_src_\%u pad. If the session number is not provided,
* the pad from the lowest available session will be returned. The session manager will modify the
- * SSRC in the RTP packets to its own SSRC and wil forward the packets on the
+ * SSRC in the RTP packets to its own SSRC and will forward the packets on the
* send_rtp_src_\%u pad after updating its internal state.
*
* The session manager needs the clock-rate of the payload types it is handling
* 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.
+ * The #GstRtpBin::request-jitterbuffer signal can be used to provide a custom
+ * element to perform arrival time smoothing, reordering and optionally packet
+ * loss detection and retransmission requests.
+ *
+ * ## Example pipelines
*
- * <refsect2>
- * <title>Example pipelines</title>
* |[
* gst-launch-1.0 udpsrc port=5000 caps="application/x-rtp, ..." ! .recv_rtp_sink_0 \
* rtpbin ! rtptheoradepay ! theoradec ! xvimagesink
* synchronisation.
* Send RTCP reports for session 0 on port 5005 and RTCP reports for session 1
* on port 5007.
- * </refsect2>
+ *
*/
#ifdef HAVE_CONFIG_H
SIGNAL_REQUEST_FEC_DECODER,
SIGNAL_REQUEST_FEC_ENCODER,
+ SIGNAL_REQUEST_JITTERBUFFER,
+
SIGNAL_NEW_JITTERBUFFER,
SIGNAL_NEW_STORAGE,
guint sessid);
static GstPad *complete_session_rtcp (GstRtpBin * rtpbin,
GstRtpBinSession * session, guint sessid);
+static GstElement *session_request_element (GstRtpBinSession * session,
+ guint signal);
/* Manages the RTP stream for one SSRC.
*
- * We pipe the stream (comming from the SSRC demuxer) into a jitterbuffer.
+ * We pipe the stream (coming from the SSRC demuxer) into a jitterbuffer.
* If we see an SDES RTCP packet that links multiple SSRCs together based on a
* common CNAME, we create a GstRtpBinClient structure to group the SSRCs
* together (see below).
/* configure SDES items */
GST_OBJECT_LOCK (rtpbin);
+ g_object_set (demux, "max-streams", rtpbin->max_streams, NULL);
g_object_set (session, "sdes", rtpbin->sdes, "rtp-profile",
rtpbin->rtp_profile, "rtcp-sync-send-time", rtpbin->rtcp_sync_send_time,
NULL);
g_slist_foreach (sess->elements, (GFunc) remove_bin_element, bin);
g_slist_free (sess->elements);
+ sess->elements = NULL;
g_slist_foreach (sess->streams, (GFunc) free_stream, bin);
g_slist_free (sess->streams);
bin = session->bin;
- GST_DEBUG ("emiting signal for pt %u in session %u", pt, session->id);
+ GST_DEBUG ("emitting 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);
GstRtpBinStream *stream = (GstRtpBinStream *) streams->data;
GST_DEBUG_OBJECT (bin, "clearing stream %p", stream);
- g_signal_emit_by_name (stream->buffer, "clear-pt-map", NULL);
+ if (g_signal_lookup ("clear-pt-map", G_OBJECT_TYPE (stream->buffer)) != 0)
+ g_signal_emit_by_name (stream->buffer, "clear-pt-map", NULL);
if (stream->demux)
g_signal_emit_by_name (stream->demux, "clear-pt-map", NULL);
}
return NULL;
}
+static GstElement *
+gst_rtp_bin_request_jitterbuffer (GstRtpBin * bin, guint session_id)
+{
+ return gst_element_factory_make ("rtpjitterbuffer", NULL);
+}
+
static void
gst_rtp_bin_propagate_property_to_jitterbuffer (GstRtpBin * bin,
const gchar * name, const GValue * value)
GST_RTP_SESSION_LOCK (session);
for (streams = session->streams; streams; streams = g_slist_next (streams)) {
GstRtpBinStream *stream = (GstRtpBinStream *) streams->data;
+ GObjectClass *jb_class;
- g_object_set_property (G_OBJECT (stream->buffer), name, value);
+ jb_class = G_OBJECT_GET_CLASS (G_OBJECT (stream->buffer));
+ if (g_object_class_find_property (jb_class, name))
+ g_object_set_property (G_OBJECT (stream->buffer), name, value);
+ else
+ GST_WARNING_OBJECT (bin,
+ "Stream jitterbuffer does not expose property %s", name);
}
GST_RTP_SESSION_UNLOCK (session);
}
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);
+ ntpns = g_get_real_time () * GST_USECOND;
/* add constant to convert from 1970 based time to 1900 based time */
if (bin->ntp_time_source == GST_RTP_NTP_TIME_SOURCE_NTP)
gboolean allow_positive_ts_offset)
{
gint64 prev_ts_offset;
+ GObjectClass *jb_class;
+
+ jb_class = G_OBJECT_GET_CLASS (G_OBJECT (stream->buffer));
+
+ if (!g_object_class_find_property (jb_class, "ts-offset")) {
+ GST_LOG_OBJECT (bin,
+ "stream's jitterbuffer does not expose ts-offset property");
+ return;
+ }
g_object_get (stream->buffer, "ts-offset", &prev_ts_offset, NULL);
/* 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. */
+ * to know how the running_time and the NTP time relate to each other. */
get_current_times (bin, &local_running_time, &local_ntpnstime);
/* see how far away the NTP time is. This is the difference between the
/* 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
* streams.
- * We calculate the mininum because we would like to only apply positive
+ * We calculate the minimum because we would like to only apply positive
* offsets to streams, delaying their playback instead of trying to speed up
- * other streams (which might be imposible when we have to create negative
+ * other streams (which might be impossible when we have to create negative
* latencies).
* The stream that has the smallest diff is selected as the reference stream,
* all other streams will have a positive offset to this difference. */
guint64 ext_base;
use_rtp = TRUE;
- /* signed version for convienience */
+ /* signed version for convenience */
clock_base = base_rtptime;
/* deal with possible wrap-around */
ext_base = base_rtptime;
GstRtpBinStream *stream;
GstRtpBin *rtpbin;
GstState target;
+ GObjectClass *jb_class;
rtpbin = session->bin;
if (g_slist_length (session->streams) >= rtpbin->max_streams)
goto max_streams;
- if (!(buffer = gst_element_factory_make ("rtpjitterbuffer", NULL)))
+ if (!(buffer =
+ session_request_element (session, SIGNAL_REQUEST_JITTERBUFFER)))
goto no_jitterbuffer;
if (!rtpbin->ignore_pt) {
goto no_queue2;
}
#endif
+
stream = g_new0 (GstRtpBinStream, 1);
stream->ssrc = ssrc;
stream->bin = rtpbin;
stream->session = session;
- stream->buffer = buffer;
+ stream->buffer = gst_object_ref (buffer);
stream->demux = demux;
stream->have_sync = FALSE;
stream->clock_base = -100 * GST_SECOND;
session->streams = g_slist_prepend (session->streams, stream);
- /* provide clock_rate to the jitterbuffer when needed */
- stream->buffer_ptreq_sig = g_signal_connect (buffer, "request-pt-map",
- (GCallback) pt_map_requested, session);
- stream->buffer_ntpstop_sig = g_signal_connect (buffer, "on-npt-stop",
- (GCallback) on_npt_stop, stream);
+ jb_class = G_OBJECT_GET_CLASS (G_OBJECT (buffer));
+
+ if (g_signal_lookup ("request-pt-map", G_OBJECT_TYPE (buffer)) != 0) {
+ /* provide clock_rate to the jitterbuffer when needed */
+ stream->buffer_ptreq_sig = g_signal_connect (buffer, "request-pt-map",
+ (GCallback) pt_map_requested, session);
+ }
+ if (g_signal_lookup ("on-npt-stop", G_OBJECT_TYPE (buffer)) != 0) {
+ stream->buffer_ntpstop_sig = g_signal_connect (buffer, "on-npt-stop",
+ (GCallback) on_npt_stop, stream);
+ }
g_object_set_data (G_OBJECT (buffer), "GstRTPBin.session", session);
g_object_set_data (G_OBJECT (buffer), "GstRTPBin.stream", stream);
/* configure latency and packet lost */
g_object_set (buffer, "latency", rtpbin->latency_ms, NULL);
- g_object_set (buffer, "drop-on-latency", rtpbin->drop_on_latency, NULL);
- g_object_set (buffer, "do-lost", rtpbin->do_lost, NULL);
- g_object_set (buffer, "mode", rtpbin->buffer_mode, NULL);
- 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);
+
+ if (g_object_class_find_property (jb_class, "drop-on-latency"))
+ g_object_set (buffer, "drop-on-latency", rtpbin->drop_on_latency, NULL);
+ if (g_object_class_find_property (jb_class, "do-lost"))
+ g_object_set (buffer, "do-lost", rtpbin->do_lost, NULL);
+ if (g_object_class_find_property (jb_class, "mode"))
+ g_object_set (buffer, "mode", rtpbin->buffer_mode, NULL);
+ if (g_object_class_find_property (jb_class, "do-retransmission"))
+ g_object_set (buffer, "do-retransmission", rtpbin->do_retransmission, NULL);
+ if (g_object_class_find_property (jb_class, "max-rtcp-rtp-time-diff"))
+ g_object_set (buffer, "max-rtcp-rtp-time-diff",
+ rtpbin->max_rtcp_rtp_time_diff, NULL);
+ if (g_object_class_find_property (jb_class, "max-dropout-time"))
+ g_object_set (buffer, "max-dropout-time", rtpbin->max_dropout_time, NULL);
+ if (g_object_class_find_property (jb_class, "max-misorder-time"))
+ g_object_set (buffer, "max-misorder-time", rtpbin->max_misorder_time, NULL);
+ if (g_object_class_find_property (jb_class, "rfc7273-sync"))
+ g_object_set (buffer, "rfc7273-sync", rtpbin->rfc7273_sync, NULL);
+ if (g_object_class_find_property (jb_class, "max-ts-offset-adjustment"))
+ 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 */
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);
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) {
if (rtpbin->buffering) {
guint64 last_out;
- GST_INFO_OBJECT (rtpbin,
- "bin is buffering, set jitterbuffer as not active");
- g_signal_emit_by_name (buffer, "set-active", FALSE, (gint64) 0, &last_out);
+ if (g_signal_lookup ("set-active", G_OBJECT_TYPE (buffer)) != 0) {
+ GST_INFO_OBJECT (rtpbin,
+ "bin is buffering, set jitterbuffer as not active");
+ g_signal_emit_by_name (buffer, "set-active", FALSE, (gint64) 0,
+ &last_out);
+ }
}
/* ERRORS */
max_streams:
{
- GST_WARNING_OBJECT (rtpbin, "stream exeeds maximum (%d)",
+ GST_WARNING_OBJECT (rtpbin, "stream exceeds maximum (%d)",
rtpbin->max_streams);
return NULL;
}
static void
free_stream (GstRtpBinStream * stream, GstRtpBin * bin)
{
+ GstRtpBinSession *sess = stream->session;
GSList *clients, *next_client;
GST_DEBUG_OBJECT (bin, "freeing stream %p", stream);
- if (stream->demux) {
- g_signal_handler_disconnect (stream->demux, stream->demux_newpad_sig);
- g_signal_handler_disconnect (stream->demux, stream->demux_ptreq_sig);
- g_signal_handler_disconnect (stream->demux, stream->demux_ptchange_sig);
- }
- g_signal_handler_disconnect (stream->buffer, stream->buffer_handlesync_sig);
- g_signal_handler_disconnect (stream->buffer, stream->buffer_ptreq_sig);
- g_signal_handler_disconnect (stream->buffer, stream->buffer_ntpstop_sig);
-
+ gst_element_set_locked_state (stream->buffer, TRUE);
if (stream->demux)
gst_element_set_locked_state (stream->demux, TRUE);
- gst_element_set_locked_state (stream->buffer, TRUE);
+ gst_element_set_state (stream->buffer, GST_STATE_NULL);
if (stream->demux)
gst_element_set_state (stream->demux, GST_STATE_NULL);
- gst_element_set_state (stream->buffer, GST_STATE_NULL);
- /* now remove this signal, we need this while going to NULL because it to
- * do some cleanups */
- if (stream->demux)
+ if (stream->demux) {
+ g_signal_handler_disconnect (stream->demux, stream->demux_newpad_sig);
+ g_signal_handler_disconnect (stream->demux, stream->demux_ptreq_sig);
+ g_signal_handler_disconnect (stream->demux, stream->demux_ptchange_sig);
g_signal_handler_disconnect (stream->demux, stream->demux_padremoved_sig);
+ }
+
+ if (stream->buffer_handlesync_sig)
+ g_signal_handler_disconnect (stream->buffer, stream->buffer_handlesync_sig);
+ if (stream->buffer_ptreq_sig)
+ g_signal_handler_disconnect (stream->buffer, stream->buffer_ptreq_sig);
+ if (stream->buffer_ntpstop_sig)
+ g_signal_handler_disconnect (stream->buffer, stream->buffer_ntpstop_sig);
+
+ sess->elements = g_slist_remove (sess->elements, stream->buffer);
+ remove_bin_element (stream->buffer, bin);
+ gst_object_unref (stream->buffer);
- gst_bin_remove (GST_BIN_CAST (bin), stream->buffer);
if (stream->demux)
gst_bin_remove (GST_BIN_CAST (bin), stream->demux);
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),
- _gst_caps_accumulator, NULL, g_cclosure_marshal_generic, GST_TYPE_CAPS,
- 2, G_TYPE_UINT, G_TYPE_UINT);
+ _gst_caps_accumulator, NULL, NULL, GST_TYPE_CAPS, 2, G_TYPE_UINT,
+ G_TYPE_UINT);
/**
* GstRtpBin::payload-type-change:
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, g_cclosure_marshal_generic, G_TYPE_NONE, 2, G_TYPE_UINT,
- G_TYPE_UINT);
+ NULL, NULL, NULL, G_TYPE_NONE, 2, G_TYPE_UINT, G_TYPE_UINT);
/**
* GstRtpBin::clear-pt-map:
gst_rtp_bin_signals[SIGNAL_CLEAR_PT_MAP] =
g_signal_new ("clear-pt-map", G_TYPE_FROM_CLASS (klass),
G_SIGNAL_RUN_LAST | G_SIGNAL_ACTION, G_STRUCT_OFFSET (GstRtpBinClass,
- clear_pt_map), NULL, NULL, g_cclosure_marshal_VOID__VOID, G_TYPE_NONE,
- 0, G_TYPE_NONE);
+ clear_pt_map), NULL, NULL, NULL, G_TYPE_NONE, 0, G_TYPE_NONE);
/**
* GstRtpBin::reset-sync:
gst_rtp_bin_signals[SIGNAL_RESET_SYNC] =
g_signal_new ("reset-sync", G_TYPE_FROM_CLASS (klass),
G_SIGNAL_RUN_LAST | G_SIGNAL_ACTION, G_STRUCT_OFFSET (GstRtpBinClass,
- reset_sync), NULL, NULL, g_cclosure_marshal_VOID__VOID, G_TYPE_NONE,
- 0, G_TYPE_NONE);
+ reset_sync), NULL, NULL, NULL, G_TYPE_NONE, 0, G_TYPE_NONE);
/**
* GstRtpBin::get-session:
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);
+ get_session), NULL, NULL, NULL, GST_TYPE_ELEMENT, 1, G_TYPE_UINT);
/**
* GstRtpBin::get-internal-session:
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, g_cclosure_marshal_generic,
- RTP_TYPE_SESSION, 1, G_TYPE_UINT);
+ get_internal_session), NULL, NULL, NULL, RTP_TYPE_SESSION, 1,
+ G_TYPE_UINT);
/**
* GstRtpBin::get-internal-storage:
* @rtpbin: the object which received the signal
* @id: the session id
*
- * Request the internal RTPStorage object as #GObject in session @id.
+ * Request the internal RTPStorage object as #GObject in session @id. This
+ * is the internal storage used by the RTPStorage element, which is used to
+ * keep a backlog of received RTP packets for the session @id.
*
* Since: 1.14
*/
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);
+ get_internal_storage), NULL, NULL, NULL, G_TYPE_OBJECT, 1,
+ G_TYPE_UINT);
/**
* GstRtpBin::get-storage:
* @rtpbin: the object which received the signal
* @id: the session id
*
- * Request the RTPStorage element as #GObject in session @id.
+ * Request the RTPStorage element as #GObject in session @id. This element
+ * is used to keep a backlog of received RTP packets for the 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);
+ get_storage), NULL, NULL, NULL, GST_TYPE_ELEMENT, 1, G_TYPE_UINT);
/**
* GstRtpBin::on-new-ssrc:
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);
+ NULL, NULL, NULL, G_TYPE_NONE, 2, G_TYPE_UINT, G_TYPE_UINT);
/**
* GstRtpBin::on-ssrc-collision:
* @rtpbin: the object which received the signal
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, g_cclosure_marshal_generic, G_TYPE_NONE, 2, G_TYPE_UINT,
- G_TYPE_UINT);
+ NULL, NULL, NULL, 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, g_cclosure_marshal_generic, G_TYPE_NONE, 2, G_TYPE_UINT,
- G_TYPE_UINT);
+ NULL, NULL, NULL, 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, g_cclosure_marshal_generic, G_TYPE_NONE, 2, G_TYPE_UINT,
- G_TYPE_UINT);
+ NULL, NULL, NULL, 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, g_cclosure_marshal_generic, G_TYPE_NONE, 2, G_TYPE_UINT,
- G_TYPE_UINT);
+ NULL, NULL, NULL, 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, g_cclosure_marshal_generic, G_TYPE_NONE, 2, G_TYPE_UINT,
- G_TYPE_UINT);
+ NULL, NULL, NULL, 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, g_cclosure_marshal_generic, G_TYPE_NONE, 2, G_TYPE_UINT,
- G_TYPE_UINT);
+ NULL, NULL, NULL, 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, g_cclosure_marshal_generic, G_TYPE_NONE, 2, G_TYPE_UINT,
- G_TYPE_UINT);
+ NULL, NULL, NULL, 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, g_cclosure_marshal_generic, G_TYPE_NONE, 2, G_TYPE_UINT,
- G_TYPE_UINT);
+ NULL, NULL, NULL, 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, g_cclosure_marshal_generic, G_TYPE_NONE, 2, G_TYPE_UINT,
- G_TYPE_UINT);
+ NULL, NULL, NULL, G_TYPE_NONE, 2, G_TYPE_UINT, G_TYPE_UINT);
/**
* GstRtpBin::request-rtp-encoder:
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);
+ request_rtp_encoder), _gst_element_accumulator, NULL, NULL,
+ GST_TYPE_ELEMENT, 1, G_TYPE_UINT);
/**
* GstRtpBin::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);
+ NULL, GST_TYPE_ELEMENT, 1, G_TYPE_UINT);
/**
* GstRtpBin::request-rtcp-encoder:
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);
+ request_rtcp_encoder), _gst_element_accumulator, NULL, NULL,
+ GST_TYPE_ELEMENT, 1, G_TYPE_UINT);
/**
* GstRtpBin::request-rtcp-decoder:
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,
+ request_rtcp_decoder), _gst_element_accumulator, NULL, NULL,
+ GST_TYPE_ELEMENT, 1, G_TYPE_UINT);
+
+ /**
+ * GstRtpBin::request-jitterbuffer:
+ * @rtpbin: the object which received the signal
+ * @session: the session
+ *
+ * Request a jitterbuffer element for the given @session.
+ *
+ * If no handler is connected, the default jitterbuffer will be used.
+ *
+ * Note: The provided element is expected to conform to the API exposed
+ * by the standard #GstRtpJitterBuffer. Runtime checks will be made to
+ * determine whether it exposes properties and signals before attempting
+ * to set, call or connect to them, and some functionalities of #GstRtpBin
+ * may not be available when that is not the case.
+ *
+ * This should be considered experimental API, as the standard jitterbuffer
+ * API is susceptible to change, provided elements will have to update their
+ * custom jitterbuffer's API to match the API of #GstRtpJitterBuffer if and
+ * when it changes.
+ *
+ * Since: 1.18
+ */
+ gst_rtp_bin_signals[SIGNAL_REQUEST_JITTERBUFFER] =
+ g_signal_new ("request-jitterbuffer", G_TYPE_FROM_CLASS (klass),
+ G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (GstRtpBinClass,
+ request_jitterbuffer), _gst_element_accumulator, NULL,
g_cclosure_marshal_generic, GST_TYPE_ELEMENT, 1, G_TYPE_UINT);
/**
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,
+ new_jitterbuffer), NULL, NULL, NULL,
G_TYPE_NONE, 3, GST_TYPE_ELEMENT, G_TYPE_UINT, G_TYPE_UINT);
/**
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,
+ new_storage), NULL, NULL, NULL,
G_TYPE_NONE, 2, GST_TYPE_ELEMENT, G_TYPE_UINT);
/**
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);
+ request_aux_sender), _gst_element_accumulator, NULL, NULL,
+ GST_TYPE_ELEMENT, 1, G_TYPE_UINT);
/**
* GstRtpBin::request-aux-receiver:
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);
+ request_aux_receiver), _gst_element_accumulator, NULL, NULL,
+ GST_TYPE_ELEMENT, 1, G_TYPE_UINT);
/**
* GstRtpBin::request-fec-decoder:
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);
+ request_fec_decoder), _gst_element_accumulator, NULL, NULL,
+ GST_TYPE_ELEMENT, 1, G_TYPE_UINT);
/**
* GstRtpBin::request-fec-encoder:
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);
+ request_fec_encoder), _gst_element_accumulator, NULL, NULL,
+ GST_TYPE_ELEMENT, 1, G_TYPE_UINT);
/**
* GstRtpBin::on-new-sender-ssrc:
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);
+ NULL, NULL, NULL, G_TYPE_NONE, 2, G_TYPE_UINT, G_TYPE_UINT);
/**
* GstRtpBin::on-sender-ssrc-active:
* @rtpbin: the object which received the signal
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,
+ on_sender_ssrc_active), NULL, NULL, NULL,
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",
"The SDES items of this session",
- GST_TYPE_STRUCTURE, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
+ GST_TYPE_STRUCTURE, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS
+#ifndef TIZEN_FEATURE_GST_UPSTREAM_AVOID_BUILD_BREAK
+ | GST_PARAM_DOC_SHOW_DEFAULT));
+#else
+ ));
+#endif
g_object_class_install_property (gobject_class, PROP_DO_LOST,
g_param_spec_boolean ("do-lost", "Do Lost",
*
* 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
+ * set the #GstRtpJitterBuffer:do-retransmission property on the
* #GstRtpJitterBuffer object instead.
*/
g_object_class_install_property (gobject_class, PROP_DO_RETRANSMISSION,
"changed to 0 (no limit)", 0, G_MAXINT64, DEFAULT_MAX_TS_OFFSET,
G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
-#ifdef TIZEN_FEATURE_RTSP_MODIFICATION
- g_object_class_install_property (gobject_class, PROP_USE_RTSP_BUFFERING,
- g_param_spec_boolean ("use-rtsp-buffering", "Use RTSP buffering",
- "Use RTSP buffering in RTP_JITTER_BUFFER_MODE_SLAVE buffer mode",
- DEFAULT_RTSP_USE_BUFFERING,
- G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
-#endif
-
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);
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);
+ klass->request_jitterbuffer =
+ GST_DEBUG_FUNCPTR (gst_rtp_bin_request_jitterbuffer);
GST_DEBUG_CATEGORY_INIT (gst_rtp_bin_debug, "rtpbin", 0, "RTP bin");
+
+#ifndef TIZEN_FEATURE_GST_UPSTREAM_AVOID_BUILD_BREAK
+ gst_type_mark_as_plugin_api (GST_RTP_BIN_RTCP_SYNC_TYPE, 0);
+#endif
}
static void
for (streams = session->streams; streams;
streams = g_slist_next (streams)) {
GstRtpBinStream *stream = (GstRtpBinStream *) streams->data;
+
#ifdef TIZEN_FEATURE_RTSP_MODIFICATION
if (rtpbin->use_rtsp_buffering &&
rtpbin->buffer_mode == RTP_JITTER_BUFFER_MODE_SLAVE) {
#else
GST_DEBUG_OBJECT (bin, "stream %p percent %d", stream,
stream->percent);
+
#endif
/* find min percent */
if (min_percent > stream->percent)
GST_RTP_SESSION_UNLOCK (session);
}
GST_DEBUG_OBJECT (bin, "min percent %d", min_percent);
+
#ifdef TIZEN_FEATURE_RTSP_MODIFICATION
if (!(rtpbin->use_rtsp_buffering &&
rtpbin->buffer_mode == RTP_JITTER_BUFFER_MODE_SLAVE)) {
streams = g_slist_next (streams)) {
GstRtpBinStream *stream = (GstRtpBinStream *) streams->data;
GstElement *element = stream->buffer;
- guint64 last_out;
+ guint64 last_out = -1;
- g_signal_emit_by_name (element, "set-active", active, offset,
- &last_out);
+ if (g_signal_lookup ("set-active", G_OBJECT_TYPE (element)) != 0) {
+ g_signal_emit_by_name (element, "set-active", active, offset,
+ &last_out);
+ }
if (!active) {
g_object_get (element, "percent", &stream->percent, NULL);
}
}
-/* a new pad (SSRC) was created in @session. This signal is emited from the
+/* a new pad (SSRC) was created in @session. This signal is emitted from the
* payload demuxer. */
static void
new_payload_found (GstElement * element, guint pt, GstPad * pad,
payload_type_change (GstElement * element, guint pt, GstRtpBinSession * session)
{
GST_DEBUG_OBJECT (session->bin,
- "emiting signal for pt type changed to %u in session %u", pt,
+ "emitting 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],
gst_object_unref (sinkpad);
gst_object_unref (srcpad);
- GST_DEBUG_OBJECT (rtpbin, "linking jitterbuffer RTCP");
- padname = g_strdup_printf ("rtcp_src_%u", ssrc);
- srcpad = gst_element_get_static_pad (element, padname);
- g_free (padname);
sinkpad = gst_element_get_request_pad (stream->buffer, "sink_rtcp");
- gst_pad_link_full (srcpad, sinkpad, GST_PAD_LINK_CHECK_NOTHING);
- gst_object_unref (sinkpad);
- gst_object_unref (srcpad);
+ if (sinkpad) {
+ GST_DEBUG_OBJECT (rtpbin, "linking jitterbuffer RTCP");
+ padname = g_strdup_printf ("rtcp_src_%u", ssrc);
+ srcpad = gst_element_get_static_pad (element, padname);
+ g_free (padname);
+ gst_pad_link_full (srcpad, sinkpad, GST_PAD_LINK_CHECK_NOTHING);
+ gst_object_unref (sinkpad);
+ gst_object_unref (srcpad);
+ }
- /* connect to the RTCP sync signal from the jitterbuffer */
- GST_DEBUG_OBJECT (rtpbin, "connecting sync signal");
- stream->buffer_handlesync_sig = g_signal_connect (stream->buffer,
- "handle-sync", (GCallback) gst_rtp_bin_handle_sync, stream);
+ if (g_signal_lookup ("handle-sync", G_OBJECT_TYPE (stream->buffer)) != 0) {
+ /* connect to the RTCP sync signal from the jitterbuffer */
+ GST_DEBUG_OBJECT (rtpbin, "connecting sync signal");
+ stream->buffer_handlesync_sig = g_signal_connect (stream->buffer,
+ "handle-sync", (GCallback) gst_rtp_bin_handle_sync, stream);
+ }
if (stream->demux) {
/* connect to the new-pad signal of the payload demuxer, this will expose the
/* ERRORS */
no_name:
{
- g_warning ("rtpbin: invalid name given");
+ g_warning ("rtpbin: cannot find session id for pad: %s",
+ GST_STR_NULL (name));
return NULL;
}
create_error:
/* ERRORS */
no_name:
{
- g_warning ("rtpbin: invalid name given");
+ g_warning ("rtpbin: cannot find session id for pad: %s",
+ GST_STR_NULL (name));
return NULL;
}
create_error:
/* ERRORS */
no_name:
{
- g_warning ("rtpbin: invalid name given");
+ g_warning ("rtpbin: cannot find session id for pad: %s",
+ GST_STR_NULL (name));
return NULL;
}
create_error:
/* ERRORS */
no_name:
{
- g_warning ("rtpbin: invalid name given");
+ g_warning ("rtpbin: cannot find session id for pad: %s",
+ GST_STR_NULL (name));
return NULL;
}
create_error:
}
/* If the requested name is NULL we should create a name with
- * the session number assuming we want the lowest posible session
+ * the session number assuming we want the lowest possible session
* with a free pad like the template */
static gchar *
gst_rtp_bin_get_free_pad_name (GstElement * element, GstPadTemplate * templ)
projects / platform / upstream / gst-plugins-good.git / blobdiff