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rtpjitterbuffer: Improve accuracy of RFC7273 clock time calculations
Previously the result of the calculations included inaccuracies caused by the NTP clock estimation, which caused the timestamps to jitter +/- 1/clockrate. By reorganizing the calculations it is possible to get rid of this inaccuracy and calculate deterministic and exact packet timestamps based on the actual NTP clock as long as the estimation is not off by more than 2**31 clockrate units. The only remaining inaccuracy that is introduced now is caused by the conversion from the NTP clock to the pipeline clock. Also split up debug output, demote many messages to the trace debug level and output more intermediate results. Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1955>
This commit is contained in:
Sebastian Dröge
GStreamer Marge Bot
parent
8c2ef0f025
commit
5ca39060f4
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@ -870,48 +870,94 @@ rtp_jitter_buffer_calculate_pts (RTPJitterBuffer * jbuf, GstClockTime dts,
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} else if (rfc7273_mode && (jbuf->mode == RTP_JITTER_BUFFER_MODE_SLAVE
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|| jbuf->mode == RTP_JITTER_BUFFER_MODE_SYNCED)
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&& media_clock_offset != -1 && jbuf->rfc7273_sync) {
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GstClockTime ntptime, rtptime_tmp;
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GstClockTime ntptime;
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GstClockTime ntprtptime, rtpsystime;
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GstClockTime internal, external;
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GstClockTime rate_num, rate_denom;
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GstClockTime ntprtptime_period_start;
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/* Don't do any of the dts related adjustments further down */
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dts = -1;
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/* Calculate the actual clock time on the sender side based on the
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* RFC7273 clock and convert it to our pipeline clock
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*/
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* RFC7273 clock and convert it to our pipeline clock. */
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gst_clock_get_calibration (media_clock, &internal, &external, &rate_num,
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&rate_denom);
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/* Current NTP clock estimation */
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ntptime = gst_clock_get_internal_time (media_clock);
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/* Current RTP time based on the estimated NTP clock */
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ntprtptime = gst_util_uint64_scale (ntptime, jbuf->clock_rate, GST_SECOND);
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ntprtptime += media_clock_offset;
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GST_TRACE ("Current NTP time %" GST_TIME_FORMAT " (RTP: %" G_GUINT64_FORMAT
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")", GST_TIME_ARGS (ntptime), ntprtptime);
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/* Start RTP time of the current RTP time period we are in */
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ntprtptime_period_start = ntprtptime & 0xffffffff00000000;
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/* Current RTP timestamp in this RTP time period */
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ntprtptime &= 0xffffffff;
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rtptime_tmp = rtptime;
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GST_TRACE ("Current NTP RTP time period start %" GST_TIME_FORMAT " (RTP: %"
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G_GUINT64_FORMAT ")",
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GST_TIME_ARGS (gst_util_uint64_scale (ntprtptime_period_start,
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GST_SECOND, jbuf->clock_rate)), ntprtptime_period_start);
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GST_TRACE ("Current NTP RTP time related to period start %" GST_TIME_FORMAT
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" (RTP: %" G_GUINT64_FORMAT ")",
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GST_TIME_ARGS (gst_util_uint64_scale (ntprtptime, GST_SECOND,
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jbuf->clock_rate)), ntprtptime);
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/* Check for wraparounds, we assume that the diff between current RTP
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* timestamp and current media clock time can't be bigger than
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* 2**31 clock units */
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if (ntprtptime > rtptime_tmp && ntprtptime - rtptime_tmp >= 0x80000000)
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rtptime_tmp += G_GUINT64_CONSTANT (0x100000000);
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else if (rtptime_tmp > ntprtptime && rtptime_tmp - ntprtptime >= 0x80000000)
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ntprtptime += G_GUINT64_CONSTANT (0x100000000);
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* timestamp and current media clock time can't be bigger than 2**31 clock
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* rate units. If it is bigger then get closer to it by moving one RTP
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* timestamp period into the future or into the past.
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*
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* E.g.
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* current NTP: 0x_______5 fffffffe
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* packet RTP: 0x 00000001
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* => packet NTP: 0x_______6 00000001
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*
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* current NTP: 0x_______5 00000001
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* packet RTP: 0x fffffffe
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* => packet NTP: 0x_______4 fffffffe
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*
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*/
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if (ntprtptime > rtptime && ntprtptime - rtptime >= 0x80000000) {
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ntprtptime_period_start += 0x100000000;
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} else if (rtptime > ntprtptime && rtptime - ntprtptime >= 0x80000000) {
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/* If there was not a full RTP time period yet then assume we're
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* actually that far away. This can only really happen if the epoch of
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* the clock is very close to now. */
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if (ntprtptime_period_start >= 0x100000000)
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ntprtptime_period_start -= 0x100000000;
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}
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if (ntprtptime > rtptime_tmp)
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ntptime -=
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gst_util_uint64_scale (ntprtptime - rtptime_tmp, GST_SECOND,
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jbuf->clock_rate);
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else
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ntptime +=
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gst_util_uint64_scale (rtptime_tmp - ntprtptime, GST_SECOND,
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jbuf->clock_rate);
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GST_TRACE ("Wraparound adjusted NTP RTP time period start %" GST_TIME_FORMAT
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" (RTP: %" G_GUINT64_FORMAT ")",
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GST_TIME_ARGS (gst_util_uint64_scale (ntprtptime_period_start,
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GST_SECOND, jbuf->clock_rate)), ntprtptime_period_start);
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/* Packet timestamp according to the NTP clock in RTP time units.
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* Note that this does not include any inaccuracy caused by the estimation
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* of the NTP clock unless it is more than 2**31 RTP time units off. */
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ntprtptime = ntprtptime_period_start + rtptime;
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/* Packet timestamp in nanoseconds according to the NTP clock. */
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ntptime = gst_util_uint64_scale (ntprtptime, GST_SECOND, jbuf->clock_rate);
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GST_DEBUG ("RFC7273 packet NTP time %" GST_TIME_FORMAT " (RTP: %"
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G_GUINT64_FORMAT ")", GST_TIME_ARGS (ntptime), ntprtptime);
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/* Packet timestamp converted to the pipeline clock.
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* Note that this includes again inaccuracy caused by the estimation of
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* the NTP vs. pipeline clock. */
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rtpsystime =
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gst_clock_adjust_with_calibration (media_clock, ntptime, internal,
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external, rate_num, rate_denom);
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/* All this assumes that the pipeline has enough additional
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* latency to cover for the network delay */
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if (rtpsystime > base_time)
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@ -919,14 +965,8 @@ rtp_jitter_buffer_calculate_pts (RTPJitterBuffer * jbuf, GstClockTime dts,
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else
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pts = 0;
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GST_DEBUG ("RFC7273 clock time %" GST_TIME_FORMAT ", ntptime %"
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GST_TIME_FORMAT ", ntprtptime %" G_GUINT64_FORMAT ", rtptime %"
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G_GUINT32_FORMAT ", base_time %" GST_TIME_FORMAT ", internal %"
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GST_TIME_FORMAT ", external %" GST_TIME_FORMAT ", out %"
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GST_TIME_FORMAT, GST_TIME_ARGS (rtpsystime), GST_TIME_ARGS (ntptime),
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ntprtptime, rtptime, GST_TIME_ARGS (base_time),
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GST_TIME_ARGS (internal), GST_TIME_ARGS (external),
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GST_TIME_ARGS (pts));
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GST_DEBUG ("Packet pipeline clock time %" GST_TIME_FORMAT ", PTS %"
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GST_TIME_FORMAT, GST_TIME_ARGS (rtpsystime), GST_TIME_ARGS (pts));
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} else {
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/* If we used the RFC7273 clock before and not anymore,
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* we need to resync it later again */
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