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https://gitlab.freedesktop.org/gstreamer/gstreamer.git
synced 2024-10-03 01:02:19 +00:00
rtpbin: add request-jitterbuffer signal
This can be used to pass the threadsharing jitterbuffer from gst-plugins-rs for example.
This commit is contained in:
Mathieu Duponchelle
Sebastian Dröge
parent
5ffd733317
commit
b5e414cdc2
2 changed files with 134 additions and 43 deletions
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@ -85,8 +85,12 @@
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* An AUX receiver has 1 src_\%u pad that much match the sessionid in the signal
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* and 1 or more sink_\%u pads. A session will be made for each sink_\%u pad
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* when the corresponding recv_rtp_sink_\%u pad is requested on #GstRtpBin.
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* The #GstRtpBin::request-jitterbuffer signal can be used to provide a custom
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* element to perform arrival time smoothing, reordering and optionally packet
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* loss detection and retransmission requests.
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*
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* ## Example pipelines
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*
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* |[
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* gst-launch-1.0 udpsrc port=5000 caps="application/x-rtp, ..." ! .recv_rtp_sink_0 \
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* rtpbin ! rtptheoradepay ! theoradec ! xvimagesink
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@ -274,6 +278,8 @@ enum
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SIGNAL_REQUEST_FEC_DECODER,
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SIGNAL_REQUEST_FEC_ENCODER,
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SIGNAL_REQUEST_JITTERBUFFER,
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SIGNAL_NEW_JITTERBUFFER,
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SIGNAL_NEW_STORAGE,
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@ -383,6 +389,8 @@ complete_session_receiver (GstRtpBin * rtpbin, GstRtpBinSession * session,
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guint sessid);
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static GstPad *complete_session_rtcp (GstRtpBin * rtpbin,
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GstRtpBinSession * session, guint sessid);
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static GstElement *session_request_element (GstRtpBinSession * session,
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guint signal);
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/* Manages the RTP stream for one SSRC.
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*
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@ -990,7 +998,8 @@ gst_rtp_bin_clear_pt_map (GstRtpBin * bin)
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GstRtpBinStream *stream = (GstRtpBinStream *) streams->data;
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GST_DEBUG_OBJECT (bin, "clearing stream %p", stream);
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g_signal_emit_by_name (stream->buffer, "clear-pt-map", NULL);
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if (g_signal_lookup ("clear-pt-map", G_OBJECT_TYPE (stream->buffer)) != 0)
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g_signal_emit_by_name (stream->buffer, "clear-pt-map", NULL);
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if (stream->demux)
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g_signal_emit_by_name (stream->demux, "clear-pt-map", NULL);
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}
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@ -1089,6 +1098,12 @@ gst_rtp_bin_request_decoder (GstRtpBin * bin, guint session_id)
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return NULL;
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}
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static GstElement *
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gst_rtp_bin_request_jitterbuffer (GstRtpBin * bin, guint session_id)
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{
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return gst_element_factory_make ("rtpjitterbuffer", NULL);
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}
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static void
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gst_rtp_bin_propagate_property_to_jitterbuffer (GstRtpBin * bin,
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const gchar * name, const GValue * value)
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@ -1102,8 +1117,14 @@ gst_rtp_bin_propagate_property_to_jitterbuffer (GstRtpBin * bin,
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GST_RTP_SESSION_LOCK (session);
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for (streams = session->streams; streams; streams = g_slist_next (streams)) {
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GstRtpBinStream *stream = (GstRtpBinStream *) streams->data;
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GObjectClass *jb_class;
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g_object_set_property (G_OBJECT (stream->buffer), name, value);
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jb_class = G_OBJECT_GET_CLASS (G_OBJECT (stream->buffer));
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if (g_object_class_find_property (jb_class, name))
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g_object_set_property (G_OBJECT (stream->buffer), name, value);
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else
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GST_WARNING_OBJECT (bin,
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"Stream jitterbuffer does not expose property %s", name);
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}
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GST_RTP_SESSION_UNLOCK (session);
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}
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@ -1235,6 +1256,15 @@ stream_set_ts_offset (GstRtpBin * bin, GstRtpBinStream * stream,
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gboolean allow_positive_ts_offset)
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{
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gint64 prev_ts_offset;
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GObjectClass *jb_class;
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jb_class = G_OBJECT_GET_CLASS (G_OBJECT (stream->buffer));
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if (!g_object_class_find_property (jb_class, "ts-offset")) {
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GST_LOG_OBJECT (bin,
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"stream's jitterbuffer does not expose ts-offset property");
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return;
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}
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g_object_get (stream->buffer, "ts-offset", &prev_ts_offset, NULL);
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@ -1693,13 +1723,15 @@ create_stream (GstRtpBinSession * session, guint32 ssrc)
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GstRtpBinStream *stream;
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GstRtpBin *rtpbin;
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GstState target;
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GObjectClass *jb_class;
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rtpbin = session->bin;
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if (g_slist_length (session->streams) >= rtpbin->max_streams)
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goto max_streams;
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if (!(buffer = gst_element_factory_make ("rtpjitterbuffer", NULL)))
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if (!(buffer =
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session_request_element (session, SIGNAL_REQUEST_JITTERBUFFER)))
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goto no_jitterbuffer;
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if (!rtpbin->ignore_pt) {
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@ -1711,7 +1743,7 @@ create_stream (GstRtpBinSession * session, guint32 ssrc)
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stream->ssrc = ssrc;
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stream->bin = rtpbin;
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stream->session = session;
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stream->buffer = buffer;
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stream->buffer = gst_object_ref (buffer);
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stream->demux = demux;
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stream->have_sync = FALSE;
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@ -1721,28 +1753,44 @@ create_stream (GstRtpBinSession * session, guint32 ssrc)
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stream->clock_base = -100 * GST_SECOND;
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session->streams = g_slist_prepend (session->streams, stream);
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/* provide clock_rate to the jitterbuffer when needed */
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stream->buffer_ptreq_sig = g_signal_connect (buffer, "request-pt-map",
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(GCallback) pt_map_requested, session);
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stream->buffer_ntpstop_sig = g_signal_connect (buffer, "on-npt-stop",
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(GCallback) on_npt_stop, stream);
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jb_class = G_OBJECT_GET_CLASS (G_OBJECT (buffer));
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if (g_signal_lookup ("request-pt-map", G_OBJECT_TYPE (buffer)) != 0) {
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/* provide clock_rate to the jitterbuffer when needed */
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stream->buffer_ptreq_sig = g_signal_connect (buffer, "request-pt-map",
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(GCallback) pt_map_requested, session);
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}
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if (g_signal_lookup ("on-npt-stop", G_OBJECT_TYPE (buffer)) != 0) {
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stream->buffer_ntpstop_sig = g_signal_connect (buffer, "on-npt-stop",
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(GCallback) on_npt_stop, stream);
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}
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g_object_set_data (G_OBJECT (buffer), "GstRTPBin.session", session);
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g_object_set_data (G_OBJECT (buffer), "GstRTPBin.stream", stream);
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/* configure latency and packet lost */
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g_object_set (buffer, "latency", rtpbin->latency_ms, NULL);
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g_object_set (buffer, "drop-on-latency", rtpbin->drop_on_latency, NULL);
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g_object_set (buffer, "do-lost", rtpbin->do_lost, NULL);
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g_object_set (buffer, "mode", rtpbin->buffer_mode, NULL);
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g_object_set (buffer, "do-retransmission", rtpbin->do_retransmission, NULL);
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g_object_set (buffer, "max-rtcp-rtp-time-diff",
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rtpbin->max_rtcp_rtp_time_diff, NULL);
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g_object_set (buffer, "max-dropout-time", rtpbin->max_dropout_time,
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"max-misorder-time", rtpbin->max_misorder_time, NULL);
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g_object_set (buffer, "rfc7273-sync", rtpbin->rfc7273_sync, NULL);
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g_object_set (buffer, "max-ts-offset-adjustment",
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rtpbin->max_ts_offset_adjustment, NULL);
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if (g_object_class_find_property (jb_class, "drop-on-latency"))
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g_object_set (buffer, "drop-on-latency", rtpbin->drop_on_latency, NULL);
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if (g_object_class_find_property (jb_class, "do-lost"))
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g_object_set (buffer, "do-lost", rtpbin->do_lost, NULL);
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if (g_object_class_find_property (jb_class, "mode"))
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g_object_set (buffer, "mode", rtpbin->buffer_mode, NULL);
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if (g_object_class_find_property (jb_class, "do-retransmission"))
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g_object_set (buffer, "do-retransmission", rtpbin->do_retransmission, NULL);
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if (g_object_class_find_property (jb_class, "max-rtcp-rtp-time-diff"))
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g_object_set (buffer, "max-rtcp-rtp-time-diff",
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rtpbin->max_rtcp_rtp_time_diff, NULL);
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if (g_object_class_find_property (jb_class, "max-dropout-time"))
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g_object_set (buffer, "max-dropout-time", rtpbin->max_dropout_time, NULL);
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if (g_object_class_find_property (jb_class, "max-misorder-time"))
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g_object_set (buffer, "max-dropout-time", rtpbin->max_misorder_time, NULL);
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if (g_object_class_find_property (jb_class, "rfc7273-sync"))
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g_object_set (buffer, "rfc7273-sync", rtpbin->rfc7273_sync, NULL);
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if (g_object_class_find_property (jb_class, "max-ts-offset-adjustment"))
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g_object_set (buffer, "max-ts-offset-adjustment",
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rtpbin->max_ts_offset_adjustment, NULL);
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/* need to sink the jitterbufer or otherwise signal handlers from bindings will
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* take ownership of it and we don't own it anymore */
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@ -1752,7 +1800,6 @@ create_stream (GstRtpBinSession * session, guint32 ssrc)
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if (!rtpbin->ignore_pt)
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gst_bin_add (GST_BIN_CAST (rtpbin), demux);
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gst_bin_add (GST_BIN_CAST (rtpbin), buffer);
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/* unref the jitterbuffer again, the bin has a reference now and
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* we don't need it anymore */
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@ -1766,9 +1813,12 @@ create_stream (GstRtpBinSession * session, guint32 ssrc)
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if (rtpbin->buffering) {
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guint64 last_out;
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GST_INFO_OBJECT (rtpbin,
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"bin is buffering, set jitterbuffer as not active");
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g_signal_emit_by_name (buffer, "set-active", FALSE, (gint64) 0, &last_out);
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if (g_signal_lookup ("set-active", G_OBJECT_TYPE (buffer)) != 0) {
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GST_INFO_OBJECT (rtpbin,
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"bin is buffering, set jitterbuffer as not active");
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g_signal_emit_by_name (buffer, "set-active", FALSE, (gint64) 0,
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&last_out);
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}
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}
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@ -1817,24 +1867,27 @@ free_stream (GstRtpBinStream * stream, GstRtpBin * bin)
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g_signal_handler_disconnect (stream->demux, stream->demux_ptreq_sig);
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g_signal_handler_disconnect (stream->demux, stream->demux_ptchange_sig);
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}
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g_signal_handler_disconnect (stream->buffer, stream->buffer_handlesync_sig);
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g_signal_handler_disconnect (stream->buffer, stream->buffer_ptreq_sig);
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g_signal_handler_disconnect (stream->buffer, stream->buffer_ntpstop_sig);
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if (stream->buffer_handlesync_sig)
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g_signal_handler_disconnect (stream->buffer, stream->buffer_handlesync_sig);
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if (stream->buffer_ptreq_sig)
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g_signal_handler_disconnect (stream->buffer, stream->buffer_ptreq_sig);
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if (stream->buffer_ntpstop_sig)
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g_signal_handler_disconnect (stream->buffer, stream->buffer_ntpstop_sig);
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gst_object_unref (stream->buffer);
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if (stream->demux)
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gst_element_set_locked_state (stream->demux, TRUE);
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gst_element_set_locked_state (stream->buffer, TRUE);
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if (stream->demux)
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gst_element_set_state (stream->demux, GST_STATE_NULL);
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gst_element_set_state (stream->buffer, GST_STATE_NULL);
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/* now remove this signal, we need this while going to NULL because it to
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* do some cleanups */
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if (stream->demux)
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g_signal_handler_disconnect (stream->demux, stream->demux_padremoved_sig);
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gst_bin_remove (GST_BIN_CAST (bin), stream->buffer);
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if (stream->demux)
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gst_bin_remove (GST_BIN_CAST (bin), stream->demux);
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@ -2263,6 +2316,34 @@ gst_rtp_bin_class_init (GstRtpBinClass * klass)
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request_rtcp_decoder), _gst_element_accumulator, NULL,
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g_cclosure_marshal_generic, GST_TYPE_ELEMENT, 1, G_TYPE_UINT);
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/**
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* GstRtpBin::request-jitterbuffer:
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* @rtpbin: the object which received the signal
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* @session: the session
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*
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* Request a jitterbuffer element for the given @session.
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*
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* If no handler is connected, the default jitterbuffer will be used.
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*
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* Note: The provided element is expected to conform to the API exposed
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* by the standard #GstRtpJitterBuffer. Runtime checks will be made to
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* determine whether it exposes properties and signals before attempting
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* to set, call or connect to them, and some functionalities of #GstRtpBin
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* may not be available when that is not the case.
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*
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* This should be considered experimental API, as the standard jitterbuffer
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* API is susceptible to change, provided elements will have to update their
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* custom jitterbuffer's API to match the API of #GstRtpJitterBuffer if and
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* when it changes.
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*
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* Since: 1.18
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*/
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gst_rtp_bin_signals[SIGNAL_REQUEST_JITTERBUFFER] =
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g_signal_new ("request-jitterbuffer", G_TYPE_FROM_CLASS (klass),
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G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (GstRtpBinClass,
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request_jitterbuffer), _gst_element_accumulator, NULL,
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g_cclosure_marshal_generic, GST_TYPE_ELEMENT, 1, G_TYPE_UINT);
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/**
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* GstRtpBin::new-jitterbuffer:
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* @rtpbin: the object which received the signal
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@ -2622,6 +2703,8 @@ gst_rtp_bin_class_init (GstRtpBinClass * klass)
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klass->request_rtp_decoder = GST_DEBUG_FUNCPTR (gst_rtp_bin_request_decoder);
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klass->request_rtcp_encoder = GST_DEBUG_FUNCPTR (gst_rtp_bin_request_encoder);
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klass->request_rtcp_decoder = GST_DEBUG_FUNCPTR (gst_rtp_bin_request_decoder);
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klass->request_jitterbuffer =
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GST_DEBUG_FUNCPTR (gst_rtp_bin_request_jitterbuffer);
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GST_DEBUG_CATEGORY_INIT (gst_rtp_bin_debug, "rtpbin", 0, "RTP bin");
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}
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@ -3157,8 +3240,10 @@ gst_rtp_bin_handle_message (GstBin * bin, GstMessage * message)
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GstElement *element = stream->buffer;
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guint64 last_out;
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g_signal_emit_by_name (element, "set-active", active, offset,
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&last_out);
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if (g_signal_lookup ("set-active", G_OBJECT_TYPE (element)) != 0) {
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g_signal_emit_by_name (element, "set-active", active, offset,
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&last_out);
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}
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if (!active) {
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g_object_get (element, "percent", &stream->percent, NULL);
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@ -3535,19 +3620,23 @@ new_ssrc_pad_found (GstElement * element, guint ssrc, GstPad * pad,
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gst_object_unref (sinkpad);
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gst_object_unref (srcpad);
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GST_DEBUG_OBJECT (rtpbin, "linking jitterbuffer RTCP");
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padname = g_strdup_printf ("rtcp_src_%u", ssrc);
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srcpad = gst_element_get_static_pad (element, padname);
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g_free (padname);
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sinkpad = gst_element_get_request_pad (stream->buffer, "sink_rtcp");
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gst_pad_link_full (srcpad, sinkpad, GST_PAD_LINK_CHECK_NOTHING);
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gst_object_unref (sinkpad);
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gst_object_unref (srcpad);
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if (sinkpad) {
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GST_DEBUG_OBJECT (rtpbin, "linking jitterbuffer RTCP");
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padname = g_strdup_printf ("rtcp_src_%u", ssrc);
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srcpad = gst_element_get_static_pad (element, padname);
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g_free (padname);
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gst_pad_link_full (srcpad, sinkpad, GST_PAD_LINK_CHECK_NOTHING);
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gst_object_unref (sinkpad);
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gst_object_unref (srcpad);
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}
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/* connect to the RTCP sync signal from the jitterbuffer */
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GST_DEBUG_OBJECT (rtpbin, "connecting sync signal");
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stream->buffer_handlesync_sig = g_signal_connect (stream->buffer,
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"handle-sync", (GCallback) gst_rtp_bin_handle_sync, stream);
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if (g_signal_lookup ("handle-sync", G_OBJECT_TYPE (stream->buffer)) != 0) {
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/* connect to the RTCP sync signal from the jitterbuffer */
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GST_DEBUG_OBJECT (rtpbin, "connecting sync signal");
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stream->buffer_handlesync_sig = g_signal_connect (stream->buffer,
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"handle-sync", (GCallback) gst_rtp_bin_handle_sync, stream);
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}
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if (stream->demux) {
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/* connect to the new-pad signal of the payload demuxer, this will expose the
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@ -135,6 +135,8 @@ struct _GstRtpBinClass {
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GstElement* (*request_fec_encoder) (GstRtpBin *rtpbin, guint session);
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GstElement* (*request_fec_decoder) (GstRtpBin *rtpbin, guint session);
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GstElement* (*request_jitterbuffer) (GstRtpBin *rtpbin, guint session);
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void (*on_new_sender_ssrc) (GstRtpBin *rtpbin, guint session, guint32 ssrc);
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void (*on_sender_ssrc_active) (GstRtpBin *rtpbin, guint session, guint32 ssrc);
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};
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