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rtpsource: Don't store invalid running times and calculate with it
If we end up with GST_CLOCK_TIME_NONE as running time for an RTP packet then this can't be used for bitrate estimation, and also not for constructing the next RTCP SR. Both would end up with completely wrong values, and an RTCP SR with wrong values can easily break synchronization in receivers. Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/5329>
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57bfbf51b2
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2a2ef23829
1 changed files with 11 additions and 6 deletions
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@ -1415,7 +1415,8 @@ rtp_source_send_rtp (RTPSource * src, RTPPacketInfo * pinfo)
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running_time = pinfo->running_time;
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do_bitrate_estimation (src, running_time, &src->bytes_sent);
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if (GST_CLOCK_TIME_IS_VALID (running_time))
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do_bitrate_estimation (src, running_time, &src->bytes_sent);
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rtptime = pinfo->rtptime;
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@ -1427,7 +1428,9 @@ rtp_source_send_rtp (RTPSource * src, RTPPacketInfo * pinfo)
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if (ext_rtptime > src->last_rtptime) {
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rtp_diff = ext_rtptime - src->last_rtptime;
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rt_diff = running_time - src->last_rtime;
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rt_diff =
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GST_CLOCK_TIME_IS_VALID (running_time) ? running_time -
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src->last_rtime : GST_CLOCK_TIME_NONE;
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/* calc the diff so we can detect drift at the sender. This can also be used
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* to guestimate the clock rate if the NTP time is locked to the RTP
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@ -1436,10 +1439,12 @@ rtp_source_send_rtp (RTPSource * src, RTPPacketInfo * pinfo)
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GST_TIME_FORMAT, src->ssrc, rtp_diff, GST_TIME_ARGS (rt_diff));
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}
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/* we keep track of the last received RTP timestamp and the corresponding
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* buffer running_time so that we can use this info when constructing SR reports */
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src->last_rtime = running_time;
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src->last_rtptime = ext_rtptime;
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if (GST_CLOCK_TIME_IS_VALID (running_time)) {
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/* we keep track of the last received RTP timestamp and the corresponding
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* buffer running_time so that we can use this info when constructing SR reports */
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src->last_rtime = running_time;
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src->last_rtptime = ext_rtptime;
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}
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/* push packet */
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if (!src->callbacks.push_rtp)
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