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https://gitlab.freedesktop.org/gstreamer/gstreamer.git
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gst/rtpmanager/gstrtpjitterbuffer.c: Report NO_PREROLL when going to PAUSED.
Original commit message from CVS: * gst/rtpmanager/gstrtpjitterbuffer.c: (gst_rtp_jitter_buffer_change_state): Report NO_PREROLL when going to PAUSED. * gst/rtpmanager/gstrtpsession.c: (rtcp_thread): Don't send RTCP right before we are shutting down. * gst/rtpmanager/rtpsession.c: (rtp_session_process_rtp), (rtp_session_process_sr), (session_report_blocks), (rtp_session_perform_reporting): Improve report blocks. * gst/rtpmanager/rtpsource.c: (calculate_jitter), (init_seq), (rtp_source_process_rtp), (rtp_source_process_sr), (rtp_source_process_rb), (rtp_source_get_last_sr), (rtp_source_get_last_rb): * gst/rtpmanager/rtpsource.h: * gst/rtpmanager/rtpstats.h: Cleanups, add methods to access stats.
This commit is contained in:
parent
23883be047
commit
ae536e0c89
6 changed files with 161 additions and 42 deletions
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@ -553,6 +553,8 @@ gst_rtp_jitter_buffer_change_state (GstElement * element,
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async_jitter_queue_set_blocking_unlocked (jitterbuffer->priv->queue,
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async_jitter_queue_set_blocking_unlocked (jitterbuffer->priv->queue,
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TRUE);
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TRUE);
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async_jitter_queue_unlock (priv->queue);
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async_jitter_queue_unlock (priv->queue);
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if (ret != GST_STATE_CHANGE_FAILURE)
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ret = GST_STATE_CHANGE_NO_PREROLL;
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break;
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break;
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case GST_STATE_CHANGE_PAUSED_TO_READY:
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case GST_STATE_CHANGE_PAUSED_TO_READY:
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break;
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break;
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@ -304,21 +304,27 @@ rtcp_thread (GstRTPSession * rtpsession)
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while (!rtpsession->priv->stop_thread) {
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while (!rtpsession->priv->stop_thread) {
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gdouble timeout;
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gdouble timeout;
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GstClockTime target;
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GstClockTime target;
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GstClockReturn res;
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timeout = rtp_session_get_reporting_interval (rtpsession->priv->session);
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timeout = rtp_session_get_reporting_interval (rtpsession->priv->session);
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GST_DEBUG_OBJECT (rtpsession, "next RTCP timeout: %lf", timeout);
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GST_DEBUG_OBJECT (rtpsession, "next RTCP timeout: %lf", timeout);
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target = gst_clock_get_time (clock);
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target = gst_clock_get_time (clock);
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target += GST_SECOND * timeout;
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target += GST_SECOND * timeout;
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id = rtpsession->priv->id = gst_clock_new_single_shot_id (clock, target);
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id = rtpsession->priv->id = gst_clock_new_single_shot_id (clock, target);
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GST_RTP_SESSION_UNLOCK (rtpsession);
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GST_RTP_SESSION_UNLOCK (rtpsession);
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gst_clock_id_wait (id, NULL);
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res = gst_clock_id_wait (id, NULL);
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if (res != GST_CLOCK_UNSCHEDULED) {
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GST_DEBUG_OBJECT (rtpsession, "got RTCP timeout");
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GST_DEBUG_OBJECT (rtpsession, "got RTCP timeout");
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/* make the session manager produce RTCP, we ignore the result. */
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/* make the session manager produce RTCP, we ignore the result. */
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rtp_session_perform_reporting (rtpsession->priv->session);
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rtp_session_perform_reporting (rtpsession->priv->session);
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} else {
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GST_DEBUG_OBJECT (rtpsession, "got unscheduled");
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}
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GST_RTP_SESSION_LOCK (rtpsession);
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GST_RTP_SESSION_LOCK (rtpsession);
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gst_clock_id_unref (id);
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gst_clock_id_unref (id);
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@ -893,6 +893,7 @@ rtp_session_process_rtp (RTPSession * sess, GstBuffer * buffer)
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prevsender = RTP_SOURCE_IS_SENDER (source);
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prevsender = RTP_SOURCE_IS_SENDER (source);
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prevactive = RTP_SOURCE_IS_ACTIVE (source);
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prevactive = RTP_SOURCE_IS_ACTIVE (source);
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/* we need to ref so that we can process the CSRCs later */
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gst_buffer_ref (buffer);
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gst_buffer_ref (buffer);
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/* let source process the packet */
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/* let source process the packet */
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@ -982,7 +983,8 @@ rtp_session_process_sr (RTPSession * sess, GstRTCPPacket * packet,
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prevsender = RTP_SOURCE_IS_SENDER (source);
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prevsender = RTP_SOURCE_IS_SENDER (source);
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/* first update the source */
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/* first update the source */
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rtp_source_process_sr (source, ntptime, rtptime, packet_count, octet_count);
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rtp_source_process_sr (source, ntptime, rtptime, packet_count, octet_count,
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arrival->time);
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if (prevsender != RTP_SOURCE_IS_SENDER (source)) {
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if (prevsender != RTP_SOURCE_IS_SENDER (source)) {
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sess->stats.sender_sources++;
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sess->stats.sender_sources++;
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@ -1004,7 +1006,7 @@ rtp_session_process_sr (RTPSession * sess, GstRTCPPacket * packet,
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if (ssrc == sess->source->ssrc) {
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if (ssrc == sess->source->ssrc) {
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/* only deal with report blocks for our session, we update the stats of
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/* only deal with report blocks for our session, we update the stats of
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* the sender of the TCP message. We could also compare our stats against
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* the sender of the RTCP message. We could also compare our stats against
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* the other sender to see if we are better or worse. */
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* the other sender to see if we are better or worse. */
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rtp_source_process_rb (source, fractionlost, packetslost,
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rtp_source_process_rb (source, fractionlost, packetslost,
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exthighestseq, jitter, lsr, dlsr);
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exthighestseq, jitter, lsr, dlsr);
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@ -1292,6 +1294,7 @@ typedef struct
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{
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{
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RTPSession *sess;
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RTPSession *sess;
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GstBuffer *rtcp;
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GstBuffer *rtcp;
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GstClockTime time;
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GstRTCPPacket packet;
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GstRTCPPacket packet;
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} ReportData;
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} ReportData;
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@ -1322,29 +1325,25 @@ session_report_blocks (const gchar * key, RTPSource * source, ReportData * data)
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}
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}
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}
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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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if (gst_rtcp_packet_get_rb_count (packet) < GST_RTCP_MAX_RB_COUNT) {
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/* only report about other sources */
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/* only report about other sender sources */
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if (source != sess->source) {
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if (source != sess->source && RTP_SOURCE_IS_SENDER (source)) {
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RTPSourceStats *stats;
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RTPSourceStats *stats;
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guint32 extended_max, expected;
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guint64 extended_max, expected;
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guint32 expected_interval, received_interval;
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guint64 expected_interval, received_interval, ntptime;
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guint32 lost, lost_interval, fraction;
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gint64 lost, lost_interval;
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guint32 fraction, LSR, DLSR;
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GstClockTime time;
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stats = &source->stats;
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stats = &source->stats;
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extended_max = (stats->cycles << 16) + stats->max_seq;
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extended_max = stats->cycles + stats->max_seq;
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expected = extended_max - stats->base_seq + 1;
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expected = extended_max - stats->base_seq + 1;
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if (expected > stats->packets_received) {
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GST_DEBUG ("ext_max %d, expected %d, received %d, base_seq %d",
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extended_max, expected, stats->packets_received, stats->base_seq);
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lost = expected - stats->packets_received;
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lost = expected - stats->packets_received;
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if (lost > 0x7fffff)
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lost = CLAMP (lost, -0x800000, 0x7fffff);
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lost = 0x7fffff;
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} else {
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lost = stats->packets_received - expected;
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if (lost > 0x800000)
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lost = 0x800000;
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else
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lost = -lost;
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}
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expected_interval = expected - stats->prev_expected;
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expected_interval = expected - stats->prev_expected;
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stats->prev_expected = expected;
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stats->prev_expected = expected;
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@ -1363,9 +1362,21 @@ session_report_blocks (const gchar * key, RTPSource * source, ReportData * data)
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GST_DEBUG ("fraction %d, lost %d, extseq %u, jitter %d", fraction, lost,
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GST_DEBUG ("fraction %d, lost %d, extseq %u, jitter %d", fraction, lost,
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extended_max, stats->jitter >> 4);
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extended_max, stats->jitter >> 4);
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if (rtp_source_get_last_sr (source, &ntptime, NULL, NULL, NULL, &time)) {
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/* LSR is middle bits of the last ntptime */
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LSR = (ntptime >> 16) & 0xffffffff;
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/* DLSR, delay since last SR is expressed in 1/65536 second units */
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DLSR = gst_util_uint64_scale_int (data->time - time, 65536, GST_SECOND);
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} else {
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/* No valid SR received, LSR/DLSR are set to 0 then */
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LSR = 0;
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DLSR = 0;
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}
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GST_DEBUG ("LSR %08x, DLSR %08x", LSR, DLSR);
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/* packet is not yet filled, add report block for this source. */
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/* packet is not yet filled, add report block for this source. */
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gst_rtcp_packet_add_rb (packet, source->ssrc, fraction, lost,
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gst_rtcp_packet_add_rb (packet, source->ssrc, fraction, lost,
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extended_max, stats->jitter >> 4, 0, 0);
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extended_max, stats->jitter >> 4, LSR, DLSR);
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}
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}
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}
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}
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}
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}
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@ -1413,6 +1424,9 @@ rtp_session_perform_reporting (RTPSession * sess)
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data.sess = sess;
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data.sess = sess;
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data.rtcp = NULL;
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data.rtcp = NULL;
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/* get time so it can be used later */
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data.time = sess->callbacks.get_time (sess, sess->user_data);
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RTP_SESSION_LOCK (sess);
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RTP_SESSION_LOCK (sess);
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/* loop over all known sources and do something */
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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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g_hash_table_foreach (sess->ssrcs[sess->mask_idx],
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@ -248,15 +248,19 @@ calculate_jitter (RTPSource * src, GstBuffer * buffer,
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/* transit time is difference with RTP timestamp */
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/* transit time is difference with RTP timestamp */
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transit = rtparrival - rtptime;
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transit = rtparrival - rtptime;
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/* get diff with previous transit time */
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if (src->stats.transit != -1)
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/* get ABS diff with previous transit time */
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if (src->stats.transit != -1) {
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if (transit > src->stats.transit)
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diff = transit - src->stats.transit;
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diff = transit - src->stats.transit;
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else
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else
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diff = src->stats.transit - transit;
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} else
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diff = 0;
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diff = 0;
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src->stats.transit = transit;
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src->stats.transit = transit;
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if (diff < 0)
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diff = -diff;
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/* update jitter, the value we store is scaled up so we can keep precision. */
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/* update jitter */
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src->stats.jitter += diff - ((src->stats.jitter + 8) >> 4);
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src->stats.jitter += diff - ((src->stats.jitter + 8) >> 4);
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src->stats.prev_rtptime = src->stats.last_rtptime;
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src->stats.prev_rtptime = src->stats.last_rtptime;
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@ -292,6 +296,8 @@ init_seq (RTPSource * src, guint16 seq)
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src->stats.bytes_received = 0;
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src->stats.bytes_received = 0;
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src->stats.prev_received = 0;
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src->stats.prev_received = 0;
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src->stats.prev_expected = 0;
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src->stats.prev_expected = 0;
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GST_DEBUG ("base_seq %d", seq);
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}
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}
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/**
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/**
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@ -319,7 +325,7 @@ rtp_source_process_rtp (RTPSource * src, GstBuffer * buffer,
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seqnr = gst_rtp_buffer_get_seq (buffer);
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seqnr = gst_rtp_buffer_get_seq (buffer);
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if (stats->cycles == -1) {
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if (stats->cycles == -1) {
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GST_DEBUG ("first buffer");
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GST_DEBUG ("received first buffer");
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/* first time we heard of this source */
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/* first time we heard of this source */
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init_seq (src, seqnr);
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init_seq (src, seqnr);
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src->stats.max_seq = seqnr - 1;
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src->stats.max_seq = seqnr - 1;
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@ -366,7 +372,7 @@ rtp_source_process_rtp (RTPSource * src, GstBuffer * buffer,
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/* in order, with permissible gap */
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/* in order, with permissible gap */
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if (seqnr < stats->max_seq) {
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if (seqnr < stats->max_seq) {
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/* sequence number wrapped - count another 64K cycle. */
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/* sequence number wrapped - count another 64K cycle. */
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stats->cycles++;
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stats->cycles += RTP_SEQ_MOD;
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}
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}
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stats->max_seq = seqnr;
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stats->max_seq = seqnr;
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} else if (udelta <= RTP_SEQ_MOD - RTP_MAX_MISORDER) {
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} else if (udelta <= RTP_SEQ_MOD - RTP_MAX_MISORDER) {
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@ -392,8 +398,8 @@ rtp_source_process_rtp (RTPSource * src, GstBuffer * buffer,
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src->is_sender = TRUE;
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src->is_sender = TRUE;
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src->validated = TRUE;
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src->validated = TRUE;
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GST_DEBUG ("PC: %" G_GUINT64_FORMAT ", OC: %" G_GUINT64_FORMAT,
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GST_DEBUG ("seq %d, PC: %" G_GUINT64_FORMAT ", OC: %" G_GUINT64_FORMAT,
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src->stats.packets_received, src->stats.octets_received);
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seqnr, src->stats.packets_received, src->stats.octets_received);
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/* calculate jitter */
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/* calculate jitter */
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calculate_jitter (src, buffer, arrival);
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calculate_jitter (src, buffer, arrival);
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@ -470,20 +476,21 @@ rtp_source_send_rtp (RTPSource * src, GstBuffer * buffer)
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* @rtptime: the RTP time
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* @rtptime: the RTP time
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* @packet_count: the packet count
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* @packet_count: the packet count
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* @octet_count: the octect count
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* @octet_count: the octect count
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* @time: time of packet arrival
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*
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*
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* Update the sender report in @src.
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* Update the sender report in @src.
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*/
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*/
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void
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void
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rtp_source_process_sr (RTPSource * src, guint64 ntptime, guint32 rtptime,
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rtp_source_process_sr (RTPSource * src, guint64 ntptime, guint32 rtptime,
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guint32 packet_count, guint32 octet_count)
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guint32 packet_count, guint32 octet_count, GstClockTime time)
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{
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{
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RTPSenderReport *curr;
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RTPSenderReport *curr;
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gint curridx;
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gint curridx;
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g_return_if_fail (RTP_IS_SOURCE (src));
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g_return_if_fail (RTP_IS_SOURCE (src));
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GST_DEBUG ("got SR packet: SSRC %08x, NTP %" G_GUINT64_FORMAT
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GST_DEBUG ("got SR packet: SSRC %08x, NTP %08x:%08x, RTP %u, PC %u, OC %u",
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", RTP %u, PC %u, OC %u", src->ssrc, ntptime, rtptime, packet_count,
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src->ssrc, ntptime >> 32, ntptime & 0xffffffff, rtptime, packet_count,
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octet_count);
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octet_count);
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curridx = src->stats.curr_sr ^ 1;
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curridx = src->stats.curr_sr ^ 1;
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@ -498,6 +505,7 @@ rtp_source_process_sr (RTPSource * src, guint64 ntptime, guint32 rtptime,
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curr->rtptime = rtptime;
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curr->rtptime = rtptime;
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curr->packet_count = packet_count;
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curr->packet_count = packet_count;
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curr->octet_count = octet_count;
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curr->octet_count = octet_count;
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curr->time = time;
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/* make current */
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/* make current */
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src->stats.curr_sr = curridx;
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src->stats.curr_sr = curridx;
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@ -525,7 +533,7 @@ rtp_source_process_rb (RTPSource * src, guint8 fractionlost, gint32 packetslost,
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g_return_if_fail (RTP_IS_SOURCE (src));
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g_return_if_fail (RTP_IS_SOURCE (src));
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GST_DEBUG ("got RB packet %d: SSRC %08x, FL %u"
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GST_DEBUG ("got RB packet %d: SSRC %08x, FL %u"
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", PL %u, HS %u, JITTER %u, LSR %u, DLSR %u", src->ssrc, fractionlost,
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", PL %u, HS %u, JITTER %u, LSR %08x, DLSR %08x", src->ssrc, fractionlost,
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packetslost, exthighestseq, jitter, lsr, dlsr);
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packetslost, exthighestseq, jitter, lsr, dlsr);
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curridx = src->stats.curr_rr ^ 1;
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curridx = src->stats.curr_rr ^ 1;
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@ -543,3 +551,85 @@ rtp_source_process_rb (RTPSource * src, guint8 fractionlost, gint32 packetslost,
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/* make current */
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/* make current */
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src->stats.curr_rr = curridx;
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src->stats.curr_rr = curridx;
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}
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}
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/**
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* rtp_source_get_last_sr:
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* @src: an #RTPSource
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* @ntptime: the NTP time
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* @rtptime: the RTP time
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* @packet_count: the packet count
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* @octet_count: the octect count
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* @time: time of packet arrival
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*
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* Get the values of the last sender report as set with rtp_source_process_sr().
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*
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* Returns: %TRUE if there was a valid SR report.
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*/
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gboolean
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rtp_source_get_last_sr (RTPSource * src, guint64 * ntptime, guint32 * rtptime,
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||||||
|
guint32 * packet_count, guint32 * octet_count, GstClockTime * time)
|
||||||
|
{
|
||||||
|
RTPSenderReport *curr;
|
||||||
|
|
||||||
|
g_return_val_if_fail (RTP_IS_SOURCE (src), FALSE);
|
||||||
|
|
||||||
|
curr = &src->stats.sr[src->stats.curr_sr];
|
||||||
|
if (!curr->is_valid)
|
||||||
|
return FALSE;
|
||||||
|
|
||||||
|
if (ntptime)
|
||||||
|
*ntptime = curr->ntptime;
|
||||||
|
if (rtptime)
|
||||||
|
*rtptime = curr->rtptime;
|
||||||
|
if (packet_count)
|
||||||
|
*packet_count = curr->packet_count;
|
||||||
|
if (octet_count)
|
||||||
|
*octet_count = curr->octet_count;
|
||||||
|
if (time)
|
||||||
|
*time = curr->time;
|
||||||
|
|
||||||
|
return TRUE;
|
||||||
|
}
|
||||||
|
|
||||||
|
/**
|
||||||
|
* rtp_source_get_last_rb:
|
||||||
|
* @src: an #RTPSource
|
||||||
|
* @fractionlost: fraction lost since last SR/RR
|
||||||
|
* @packetslost: the cumululative number of packets lost
|
||||||
|
* @exthighestseq: the extended last sequence number received
|
||||||
|
* @jitter: the interarrival jitter
|
||||||
|
* @lsr: the last SR packet from this source
|
||||||
|
* @dlsr: the delay since last SR packet
|
||||||
|
*
|
||||||
|
* Get the values of the last RB report set with rtp_source_process_rb().
|
||||||
|
*
|
||||||
|
* Returns: %TRUE if there was a valid SB report.
|
||||||
|
*/
|
||||||
|
gboolean
|
||||||
|
rtp_source_get_last_rb (RTPSource * src, guint8 * fractionlost,
|
||||||
|
gint32 * packetslost, guint32 * exthighestseq, guint32 * jitter,
|
||||||
|
guint32 * lsr, guint32 * dlsr)
|
||||||
|
{
|
||||||
|
RTPReceiverReport *curr;
|
||||||
|
|
||||||
|
g_return_val_if_fail (RTP_IS_SOURCE (src), FALSE);
|
||||||
|
|
||||||
|
curr = &src->stats.rr[src->stats.curr_rr];
|
||||||
|
if (!curr->is_valid)
|
||||||
|
return FALSE;
|
||||||
|
|
||||||
|
if (fractionlost)
|
||||||
|
*fractionlost = curr->fractionlost;
|
||||||
|
if (packetslost)
|
||||||
|
*packetslost = curr->packetslost;
|
||||||
|
if (exthighestseq)
|
||||||
|
*exthighestseq = curr->exthighestseq;
|
||||||
|
if (jitter)
|
||||||
|
*jitter = curr->jitter;
|
||||||
|
if (lsr)
|
||||||
|
*lsr = curr->lsr;
|
||||||
|
if (dlsr)
|
||||||
|
*dlsr = curr->dlsr;
|
||||||
|
|
||||||
|
return TRUE;
|
||||||
|
}
|
||||||
|
|
|
@ -167,9 +167,15 @@ GstFlowReturn rtp_source_send_rtp (RTPSource *src, GstBuffer *buffer);
|
||||||
/* RTCP messages */
|
/* RTCP messages */
|
||||||
void rtp_source_process_bye (RTPSource *src, const gchar *reason);
|
void rtp_source_process_bye (RTPSource *src, const gchar *reason);
|
||||||
void rtp_source_process_sr (RTPSource *src, guint64 ntptime, guint32 rtptime,
|
void rtp_source_process_sr (RTPSource *src, guint64 ntptime, guint32 rtptime,
|
||||||
guint32 packet_count, guint32 octet_count);
|
guint32 packet_count, guint32 octet_count, GstClockTime time);
|
||||||
void rtp_source_process_rb (RTPSource *src, guint8 fractionlost, gint32 packetslost,
|
void rtp_source_process_rb (RTPSource *src, guint8 fractionlost, gint32 packetslost,
|
||||||
guint32 exthighestseq, guint32 jitter,
|
guint32 exthighestseq, guint32 jitter,
|
||||||
guint32 lsr, guint32 dlsr);
|
guint32 lsr, guint32 dlsr);
|
||||||
|
|
||||||
|
gboolean rtp_source_get_last_sr (RTPSource *src, guint64 *ntptime, guint32 *rtptime,
|
||||||
|
guint32 *packet_count, guint32 *octet_count, GstClockTime *time);
|
||||||
|
gboolean rtp_source_get_last_rb (RTPSource *src, guint8 *fractionlost, gint32 *packetslost,
|
||||||
|
guint32 *exthighestseq, guint32 *jitter,
|
||||||
|
guint32 *lsr, guint32 *dlsr);
|
||||||
|
|
||||||
#endif /* __RTP_SOURCE_H__ */
|
#endif /* __RTP_SOURCE_H__ */
|
||||||
|
|
|
@ -34,6 +34,7 @@ typedef struct {
|
||||||
guint32 rtptime;
|
guint32 rtptime;
|
||||||
guint32 packet_count;
|
guint32 packet_count;
|
||||||
guint32 octet_count;
|
guint32 octet_count;
|
||||||
|
GstClockTime time;
|
||||||
} RTPSenderReport;
|
} RTPSenderReport;
|
||||||
|
|
||||||
/**
|
/**
|
||||||
|
@ -100,7 +101,7 @@ typedef struct {
|
||||||
guint32 prev_received;
|
guint32 prev_received;
|
||||||
|
|
||||||
guint16 max_seq;
|
guint16 max_seq;
|
||||||
guint32 cycles;
|
guint64 cycles;
|
||||||
guint32 base_seq;
|
guint32 base_seq;
|
||||||
guint32 bad_seq;
|
guint32 bad_seq;
|
||||||
guint32 transit;
|
guint32 transit;
|
||||||
|
|
Loading…
Reference in a new issue