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https://gitlab.freedesktop.org/gstreamer/gstreamer.git
synced 2024-10-03 01:02:19 +00:00
session: Rearrange RTCP reporting a little
Make a function to generate an RTCP packet for a source, pass the source as a parameter. Move timeout of collisions to session cleanup phase.
This commit is contained in:
Wim Taymans
parent
a3bf374351
commit
2163355a47
1 changed files with 85 additions and 81 deletions
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@ -2649,6 +2649,7 @@ typedef struct
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{
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GstRTCPBuffer rtcpbuf;
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RTPSession *sess;
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RTPSource *source;
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GstBuffer *rtcp;
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GstClockTime current_time;
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guint64 ntpnstime;
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@ -2659,16 +2660,19 @@ typedef struct
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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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} ReportData;
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static void
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session_start_rtcp (RTPSession * sess, ReportData * data)
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{
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GstRTCPPacket *packet = &data->packet;
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RTPSource *own = sess->source;
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RTPSource *own = data->source;
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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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@ -2703,19 +2707,11 @@ session_start_rtcp (RTPSession * sess, ReportData * data)
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static void
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session_report_blocks (const gchar * key, RTPSource * source, ReportData * data)
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{
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RTPSession *sess = data->sess;
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GstRTCPPacket *packet = &data->packet;
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/* create a new buffer if needed */
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if (data->rtcp == NULL) {
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session_start_rtcp (sess, data);
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} else if (data->is_early) {
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/* Put a single RR or SR in minimal compound packets */
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return;
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}
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if (gst_rtcp_packet_get_rb_count (packet) < GST_RTCP_MAX_RB_COUNT) {
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/* only report about other sender sources */
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if (source != sess->source && RTP_SOURCE_IS_SENDER (source)) {
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if (source != data->source && RTP_SOURCE_IS_SENDER (source)) {
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guint8 fractionlost;
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gint32 packetslost;
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guint32 exthighestseq, jitter;
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@ -2753,13 +2749,22 @@ session_cleanup (const gchar * key, RTPSource * source, ReportData * data)
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GstClockTime interval, binterval;
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GstClockTime btime;
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is_sender = RTP_SOURCE_IS_SENDER (source);
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is_active = RTP_SOURCE_IS_ACTIVE (source);
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/* check for outdated collisions */
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if (source->internal) {
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GST_DEBUG ("Timing out collisions");
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rtp_source_timeout (source, data->current_time,
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/* "a relatively long time" -- RFC 3550 section 8.2 */
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RTP_STATS_MIN_INTERVAL * GST_SECOND * 10,
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data->running_time - sess->rtcp_feedback_retention_window);
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}
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/* nothing to do when without RTCP */
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/* nothing else to do when without RTCP */
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if (data->interval == GST_CLOCK_TIME_NONE)
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return;
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is_sender = RTP_SOURCE_IS_SENDER (source);
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is_active = RTP_SOURCE_IS_ACTIVE (source);
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/* our own rtcp interval may have been forced low by secondary configuration,
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* while sender side may still operate with higher interval,
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* so do not just take our interval to decide on timing out sender,
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@ -2856,9 +2861,9 @@ session_sdes (RTPSession * sess, ReportData * data)
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/* add SDES packet */
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gst_rtcp_buffer_add_packet (rtcp, GST_RTCP_TYPE_SDES, packet);
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gst_rtcp_packet_sdes_add_item (packet, sess->source->ssrc);
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gst_rtcp_packet_sdes_add_item (packet, data->source->ssrc);
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sdes = rtp_source_get_sdes_struct (sess->source);
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sdes = rtp_source_get_sdes_struct (data->source);
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/* add all fields in the structure, the order is not important. */
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n_fields = gst_structure_n_fields (sdes);
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@ -2912,18 +2917,13 @@ session_sdes (RTPSession * sess, ReportData * data)
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/* schedule a BYE packet */
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static void
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session_bye (RTPSession * sess, ReportData * data)
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make_source_bye (RTPSession * sess, RTPSource * source, ReportData * data)
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{
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GstRTCPPacket *packet = &data->packet;
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GstRTCPBuffer *rtcp = &data->rtcpbuf;
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RTPSource *source = sess->source;
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/* open packet */
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session_start_rtcp (sess, data);
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/* add SDES */
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session_sdes (sess, data);
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/* add a BYE packet */
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gst_rtcp_buffer_add_packet (rtcp, GST_RTCP_TYPE_BYE, packet);
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gst_rtcp_packet_bye_add_ssrc (packet, source->ssrc);
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@ -2932,6 +2932,7 @@ session_bye (RTPSession * sess, ReportData * data)
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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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static gboolean
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@ -3027,6 +3028,51 @@ remove_closing_sources (const gchar * key, RTPSource * source, gpointer * data)
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return source->closing;
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}
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static void
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generate_rtcp (RTPSource * source, ReportData * data)
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{
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RTPSession *sess = data->sess;
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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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return;
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data->source = source;
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/* open packet */
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session_start_rtcp (sess, 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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} 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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g_hash_table_foreach (sess->ssrcs[sess->mask_idx],
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(GHFunc) session_report_blocks, data);
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}
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if (!data->has_sdes)
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session_sdes (sess, data);
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gst_rtcp_buffer_unmap (&data->rtcpbuf);
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if (sess->change_ssrc) {
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GST_DEBUG ("need to change our SSRC (%08x)", source->ssrc);
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g_hash_table_steal (sess->ssrcs[sess->mask_idx],
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GINT_TO_POINTER (source->ssrc));
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source->ssrc = rtp_session_create_new_ssrc (sess);
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rtp_source_reset (source);
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g_hash_table_insert (sess->ssrcs[sess->mask_idx],
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GINT_TO_POINTER (source->ssrc), source);
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sess->change_ssrc = FALSE;
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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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}
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/**
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* rtp_session_on_timeout:
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* @sess: an #RTPSession
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@ -3053,7 +3099,6 @@ rtp_session_on_timeout (RTPSession * sess, GstClockTime current_time,
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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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gboolean notify = FALSE;
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g_return_val_if_fail (RTP_IS_SESSION (sess), GST_FLOW_ERROR);
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@ -3062,13 +3107,11 @@ rtp_session_on_timeout (RTPSession * sess, GstClockTime current_time,
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GST_TIME_ARGS (ntpnstime), GST_TIME_ARGS (running_time));
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data.sess = sess;
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data.rtcp = NULL;
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data.current_time = current_time;
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data.ntpnstime = ntpnstime;
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data.is_bye = FALSE;
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data.has_sdes = FALSE;
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data.may_suppress = FALSE;
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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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@ -3093,88 +3136,49 @@ rtp_session_on_timeout (RTPSession * sess, GstClockTime current_time,
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(GHRFunc) remove_closing_sources, NULL);
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/* see if we need to generate SR or RR packets */
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if (is_rtcp_time (sess, current_time, &data)) {
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if (own->marked_bye) {
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/* generate BYE instead */
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GST_DEBUG ("generating BYE message");
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session_bye (sess, &data);
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own->sent_bye = TRUE;
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} else {
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/* loop over all known sources and do something */
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g_hash_table_foreach (sess->ssrcs[sess->mask_idx],
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(GHFunc) session_report_blocks, &data);
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}
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}
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if (!is_rtcp_time (sess, current_time, &data))
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goto done;
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if (data.rtcp) {
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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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if (!data.is_early && !data.may_suppress)
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sess->last_rtcp_send_time = data.current_time;
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sess->first_rtcp = FALSE;
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sess->next_early_rtcp_time = GST_CLOCK_TIME_NONE;
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generate_rtcp (own, &data);
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/* add SDES for this source when not already added */
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if (!data.has_sdes)
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session_sdes (sess, &data);
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}
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/* check for outdated collisions */
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GST_DEBUG ("Timing out collisions");
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rtp_source_timeout (sess->source, current_time,
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/* "a relatively long time" -- RFC 3550 section 8.2 */
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RTP_STATS_MIN_INTERVAL * GST_SECOND * 10,
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running_time - sess->rtcp_feedback_retention_window);
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if (sess->change_ssrc) {
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GST_DEBUG ("need to change our SSRC (%08x)", own->ssrc);
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g_hash_table_steal (sess->ssrcs[sess->mask_idx],
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GINT_TO_POINTER (own->ssrc));
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own->ssrc = rtp_session_create_new_ssrc (sess);
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rtp_source_reset (own);
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g_hash_table_insert (sess->ssrcs[sess->mask_idx],
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GINT_TO_POINTER (own->ssrc), own);
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sess->change_ssrc = FALSE;
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notify = TRUE;
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GST_DEBUG ("changed our SSRC to %08x", own->ssrc);
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}
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sess->allow_early = TRUE;
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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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if (!data.is_early && !data.may_suppress)
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sess->last_rtcp_send_time = data.current_time;
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sess->first_rtcp = FALSE;
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sess->next_early_rtcp_time = GST_CLOCK_TIME_NONE;
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done:
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RTP_SESSION_UNLOCK (sess);
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if (notify)
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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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gboolean do_not_suppress;
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gst_rtcp_buffer_unmap (&data.rtcpbuf);
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GstBuffer *buffer = data.rtcp;
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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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data.rtcp, data.is_early, &do_not_suppress);
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buffer, data.is_early, &do_not_suppress);
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if (sess->callbacks.send_rtcp && (do_not_suppress || !data.may_suppress)) {
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guint packet_size;
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packet_size = gst_buffer_get_size (data.rtcp) + sess->header_len;
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packet_size = gst_buffer_get_size (buffer) + sess->header_len;
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UPDATE_AVG (sess->stats.avg_rtcp_packet_size, packet_size);
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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, data.rtcp, own->sent_bye,
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sess->callbacks.send_rtcp (sess, own, buffer, data.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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" do_not_suppress: %d may_suppress: %d",
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sess->callbacks.send_rtcp, do_not_suppress, data.may_suppress);
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gst_buffer_unref (data.rtcp);
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gst_buffer_unref (buffer);
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}
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}
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