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https://gitlab.freedesktop.org/gstreamer/gstreamer.git
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session: produce RTCP for all internal sources
Loop over all the internal sources and produce RTCP. We also need to queue the RTCP packets and send them when we are finished.
This commit is contained in:
parent
9505fd4150
commit
495d43c089
2 changed files with 32 additions and 13 deletions
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@ -2674,6 +2674,13 @@ early_exit:
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return result;
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}
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typedef struct
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{
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RTPSource *source;
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gboolean is_bye;
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GstBuffer *buffer;
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} ReportOutput;
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typedef struct
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{
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GstRTCPBuffer rtcpbuf;
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@ -2685,11 +2692,11 @@ typedef struct
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GstClockTime running_time;
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GstClockTime interval;
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GstRTCPPacket packet;
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gboolean is_bye;
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gboolean has_sdes;
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gboolean is_early;
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gboolean may_suppress;
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gboolean notify;
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GQueue output;
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} ReportData;
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static void
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@ -2700,7 +2707,6 @@ session_start_rtcp (RTPSession * sess, ReportData * data)
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GstRTCPBuffer *rtcp = &data->rtcpbuf;
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data->rtcp = gst_rtcp_buffer_new (sess->mtu);
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data->is_bye = FALSE;
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data->has_sdes = FALSE;
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gst_rtcp_buffer_map (data->rtcp, GST_MAP_READWRITE, rtcp);
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@ -2968,7 +2974,6 @@ make_source_bye (RTPSession * sess, RTPSource * source, ReportData * data)
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gst_rtcp_packet_bye_set_reason (packet, source->bye_reason);
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/* we have a BYE packet now */
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data->is_bye = TRUE;
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source->sent_bye = TRUE;
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}
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@ -3066,9 +3071,11 @@ remove_closing_sources (const gchar * key, RTPSource * source, gpointer * data)
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}
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static void
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generate_rtcp (RTPSource * source, ReportData * data)
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generate_rtcp (const gchar * key, RTPSource * source, ReportData * data)
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{
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RTPSession *sess = data->sess;
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gboolean is_bye = FALSE;
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ReportOutput *output;
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/* only generate RTCP for active internal sources */
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if (!source->internal || source->sent_bye)
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@ -3082,6 +3089,7 @@ generate_rtcp (RTPSource * source, ReportData * data)
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if (source->marked_bye) {
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/* send BYE */
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make_source_bye (sess, source, data);
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is_bye = TRUE;
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} else if (!data->is_early) {
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/* loop over all known sources and add report blocks. If we are ealy, we
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* just make a minimal RTCP packet and skip this step */
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@ -3108,6 +3116,13 @@ generate_rtcp (RTPSource * source, ReportData * data)
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data->notify = TRUE;
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GST_DEBUG ("changed our SSRC to %08x", source->ssrc);
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}
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output = g_slice_new (ReportOutput);
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output->source = g_object_ref (source);
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output->is_bye = is_bye;
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output->buffer = data->rtcp;
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/* queue the RTCP packet to push later */
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g_queue_push_tail (&data->output, output);
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}
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/**
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@ -3134,8 +3149,8 @@ rtp_session_on_timeout (RTPSession * sess, GstClockTime current_time,
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{
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GstFlowReturn result = GST_FLOW_OK;
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ReportData data = { GST_RTCP_BUFFER_INIT };
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RTPSource *own;
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GHashTable *table_copy;
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ReportOutput *output;
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g_return_val_if_fail (RTP_IS_SESSION (sess), GST_FLOW_ERROR);
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@ -3149,8 +3164,7 @@ rtp_session_on_timeout (RTPSession * sess, GstClockTime current_time,
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data.running_time = running_time;
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data.may_suppress = FALSE;
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data.notify = FALSE;
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own = sess->source;
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g_queue_init (&data.output);
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RTP_SESSION_LOCK (sess);
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/* get a new interval, we need this for various cleanups etc */
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@ -3176,7 +3190,9 @@ rtp_session_on_timeout (RTPSession * sess, GstClockTime current_time,
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if (!is_rtcp_time (sess, current_time, &data))
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goto done;
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generate_rtcp (own, &data);
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/* generate RTCP for all internal sources */
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g_hash_table_foreach (sess->ssrcs[sess->mask_idx],
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(GHFunc) generate_rtcp, &data);
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/* we keep track of the last report time in order to timeout inactive
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* receivers or senders */
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@ -3191,10 +3207,11 @@ done:
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if (data.notify)
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g_object_notify (G_OBJECT (sess), "internal-ssrc");
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/* push out the RTCP packet */
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if (data.rtcp) {
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/* push out the RTCP packets */
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while ((output = g_queue_pop_head (&data.output))) {
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gboolean do_not_suppress;
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GstBuffer *buffer = data.rtcp;
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GstBuffer *buffer = output->buffer;
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RTPSource *source = output->source;
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/* Give the user a change to add its own packet */
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g_signal_emit (sess, rtp_session_signals[SIGNAL_ON_SENDING_RTCP], 0,
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@ -3209,7 +3226,7 @@ done:
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GST_DEBUG ("%p, sending RTCP packet, avg size %u, %u", &sess->stats,
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sess->stats.avg_rtcp_packet_size, packet_size);
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result =
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sess->callbacks.send_rtcp (sess, own, buffer, data.is_bye,
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sess->callbacks.send_rtcp (sess, source, buffer, output->is_bye,
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sess->send_rtcp_user_data);
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} else {
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GST_DEBUG ("freeing packet callback: %p"
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@ -3217,6 +3234,8 @@ done:
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sess->callbacks.send_rtcp, do_not_suppress, data.may_suppress);
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gst_buffer_unref (buffer);
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}
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g_object_unref (source);
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g_slice_free (ReportOutput, output);
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}
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return result;
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@ -328,7 +328,7 @@ void rtp_session_update_send_caps (RTPSession *sess, GstCaps *c
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GstFlowReturn rtp_session_send_rtp (RTPSession *sess, gpointer data, gboolean is_list,
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GstClockTime current_time, GstClockTime running_time);
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/* stopping the session */
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/* scheduling bye */
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void rtp_session_mark_all_bye (RTPSession *sess, const gchar *reason);
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GstFlowReturn rtp_session_schedule_bye (RTPSession *sess, GstClockTime current_time);
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