rtpbin: Handle ntp-sync=true before everything else
This simplifies the code as it's a much simpler case than the normal inter-stream synchronization, and interleaving it with that only reduces readability of the code. Also improve some debug output in this code path. Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6543>
This commit is contained in:
Sebastian Dröge
GStreamer Marge Bot
parent
4b0e75a094
commit
4421c3de75
@ -1481,6 +1481,88 @@ gst_rtp_bin_associate (GstRtpBin * bin, GstRtpBinStream * stream, guint8 len,
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return;
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return;
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}
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}
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/* In case of NTP sync we can directly calculate the offset for this stream
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* here and return immediately */
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if (bin->ntp_sync) {
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GstClockTime rtp_running_time, diff_rtp;
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GstClockTime local_running_time, local_ntpnstime;
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gint64 ntpdiff, rtdiff;
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GST_DEBUG_OBJECT (bin, "Doing NTP sync");
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if (!GST_CLOCK_TIME_IS_VALID (extrtptime)
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|| !GST_CLOCK_TIME_IS_VALID (ntpnstime)
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|| extrtptime < base_rtptime) {
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GST_DEBUG_OBJECT (bin, "invalidated sync data, bailing out");
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return;
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}
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/* Take the extended rtptime we found in the SR packet and map it to the
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* local rtptime. The local rtp time is used to construct timestamps on the
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* buffers so we will calculate what running_time corresponds to the RTP
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* timestamp in the SR packet. */
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diff_rtp = extrtptime - base_rtptime;
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GST_DEBUG_OBJECT (bin,
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"base RTP time %" G_GUINT64_FORMAT ", SR RTP time %" G_GUINT64_FORMAT
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", RTP time difference %" G_GUINT64_FORMAT ", clock-rate %d",
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base_rtptime, extrtptime, diff_rtp, clock_rate);
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/* calculate local RTP time in GStreamer timestamp units, we essentially
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* perform the same conversion that a jitterbuffer would use to convert an
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* rtp timestamp into a corresponding gstreamer timestamp. Note that the
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* base_time also contains the drift between sender and receiver. */
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rtp_running_time =
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gst_util_uint64_scale_int (diff_rtp, GST_SECOND, clock_rate);
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rtp_running_time += base_time;
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GST_DEBUG_OBJECT (bin,
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"RTP running time %" GST_TIME_FORMAT ", SR NTP time %" GST_TIME_FORMAT,
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GST_TIME_ARGS (rtp_running_time), GST_TIME_ARGS (ntpnstime));
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/* For NTP sync we need to first get a snapshot of running_time and NTP
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* time. We know at what running_time we play a certain RTP time, we also
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* calculated when we would play the RTP time in the SR packet. Now we need
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* to know how the running_time and the NTP time relate to each other. */
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get_current_times (bin, &local_running_time, &local_ntpnstime);
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/* see how far away the NTP time is. This is the difference between the
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* current NTP time and the NTP time in the last SR packet. */
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ntpdiff = local_ntpnstime - ntpnstime;
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/* see how far away the running_time is. This is the difference between the
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* current running_time and the running_time of the RTP timestamp in the
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* last SR packet. */
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rtdiff = local_running_time - rtp_running_time;
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GST_DEBUG_OBJECT (bin,
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"local NTP time %" G_GUINT64_FORMAT ", SR NTP time %" G_GUINT64_FORMAT,
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local_ntpnstime, ntpnstime);
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GST_DEBUG_OBJECT (bin,
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"local running time %" G_GUINT64_FORMAT ", SR RTP running time %"
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G_GUINT64_FORMAT, local_running_time, rtp_running_time);
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GST_DEBUG_OBJECT (bin,
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"NTP diff %" G_GINT64_FORMAT ", RT diff %" G_GINT64_FORMAT, ntpdiff,
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rtdiff);
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/* combine to get the final diff to apply to the running_time */
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stream->rt_delta = rtdiff - ntpdiff;
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GST_DEBUG_OBJECT (bin,
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"Calculated ts-offset %" GST_STIME_FORMAT " for SSRC %08x",
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GST_STIME_ARGS (stream->rt_delta), stream->ssrc);
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stream_set_ts_offset (bin, stream, stream->rt_delta, bin->max_ts_offset,
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bin->min_ts_offset, FALSE);
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gst_rtp_bin_send_sync_event (stream);
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return;
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}
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/* For all other cases (not RFC7273 and not NTP sync) we have to look how
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* all streams of a client relate to each other */
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GstRtpBinClient *client;
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GstRtpBinClient *client;
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gboolean created;
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gboolean created;
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GSList *walk;
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GSList *walk;
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@ -1548,195 +1630,159 @@ gst_rtp_bin_associate (GstRtpBin * bin, GstRtpBinStream * stream, guint8 len,
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"SR RTP running time %" G_GUINT64_FORMAT ", SR NTP %" G_GUINT64_FORMAT,
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"SR RTP running time %" G_GUINT64_FORMAT ", SR NTP %" G_GUINT64_FORMAT,
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running_time, ntpnstime);
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running_time, ntpnstime);
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/* recalc inter stream playout offset, but only if there is more than one
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/* recalc inter stream playout offset, but only if there is more than one stream. */
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* stream or we're doing NTP sync. */
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gint64 min, rtp_min, clock_base;
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if (bin->ntp_sync) {
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gboolean all_sync, use_rtp;
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gint64 ntpdiff, rtdiff;
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gboolean rtcp_sync = g_atomic_int_get (&bin->rtcp_sync);
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guint64 local_ntpnstime;
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GstClockTime local_running_time;
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/* For NTP sync we need to first get a snapshot of running_time and NTP
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/* calculate delta between server and receiver. ntpnstime is created by
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* time. We know at what running_time we play a certain RTP time, we also
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* converting the ntptime in the last SR packet to a gstreamer timestamp. This
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* calculated when we would play the RTP time in the SR packet. Now we need
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* delta expresses the difference to our timeline and the server timeline. The
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* to know how the running_time and the NTP time relate to each other. */
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* difference in itself doesn't mean much but we can combine the delta of
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get_current_times (bin, &local_running_time, &local_ntpnstime);
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* multiple streams to create a stream specific offset. */
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stream->rt_delta = ntpnstime - running_time;
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/* see how far away the NTP time is. This is the difference between the
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/* calculate the min of all deltas, ignoring streams that did not yet have a
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* current NTP time and the NTP time in the last SR packet. */
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* valid rt_delta because we did not yet receive an SR packet for those
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ntpdiff = local_ntpnstime - ntpnstime;
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* streams.
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/* see how far away the running_time is. This is the difference between the
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* We calculate the minimum because we would like to only apply positive
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* current running_time and the running_time of the RTP timestamp in the
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* offsets to streams, delaying their playback instead of trying to speed up
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* last SR packet. */
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* other streams (which might be impossible when we have to create negative
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rtdiff = local_running_time - running_time;
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* latencies).
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* The stream that has the smallest diff is selected as the reference stream,
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* all other streams will have a positive offset to this difference. */
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GST_DEBUG_OBJECT (bin,
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/* some alternative setting allow ignoring RTCP as much as possible,
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"local NTP time %" G_GUINT64_FORMAT ", SR NTP time %" G_GUINT64_FORMAT,
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* for servers generating bogus ntp timeline */
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local_ntpnstime, ntpnstime);
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min = rtp_min = G_MAXINT64;
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GST_DEBUG_OBJECT (bin,
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use_rtp = FALSE;
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"local running time %" G_GUINT64_FORMAT ", SR RTP running time %"
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if (rtcp_sync == GST_RTP_BIN_RTCP_SYNC_RTP) {
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G_GUINT64_FORMAT, local_running_time, running_time);
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guint64 ext_base = -1;
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GST_DEBUG_OBJECT (bin,
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gint64 rtp_delta = 0;
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"NTP diff %" G_GINT64_FORMAT ", RT diff %" G_GINT64_FORMAT, ntpdiff,
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rtdiff);
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/* combine to get the final diff to apply to the running_time */
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use_rtp = TRUE;
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stream->rt_delta = rtdiff - ntpdiff;
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/* convert to extended RTP time */
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rtp_clock_base = gst_rtp_buffer_ext_timestamp (&ext_base, rtp_clock_base);
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stream_set_ts_offset (bin, stream, stream->rt_delta, bin->max_ts_offset,
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/* sanity check; base rtp and provided clock_base should be close */
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bin->min_ts_offset, FALSE);
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if (rtp_clock_base >= base_rtptime) {
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} else {
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if (rtp_clock_base - base_rtptime < 10 * clock_rate) {
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gint64 min, rtp_min, clock_base;
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rtp_delta = base_time +
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gboolean all_sync, use_rtp;
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gst_util_uint64_scale_int (rtp_clock_base - base_rtptime,
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gboolean rtcp_sync = g_atomic_int_get (&bin->rtcp_sync);
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GST_SECOND, clock_rate);
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/* calculate delta between server and receiver. ntpnstime is created by
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* converting the ntptime in the last SR packet to a gstreamer timestamp. This
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* delta expresses the difference to our timeline and the server timeline. The
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* difference in itself doesn't mean much but we can combine the delta of
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* multiple streams to create a stream specific offset. */
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stream->rt_delta = ntpnstime - running_time;
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/* calculate the min of all deltas, ignoring streams that did not yet have a
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* valid rt_delta because we did not yet receive an SR packet for those
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* streams.
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* We calculate the minimum because we would like to only apply positive
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* offsets to streams, delaying their playback instead of trying to speed up
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* other streams (which might be impossible when we have to create negative
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* latencies).
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* The stream that has the smallest diff is selected as the reference stream,
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* all other streams will have a positive offset to this difference. */
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/* some alternative setting allow ignoring RTCP as much as possible,
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* for servers generating bogus ntp timeline */
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min = rtp_min = G_MAXINT64;
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use_rtp = FALSE;
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if (rtcp_sync == GST_RTP_BIN_RTCP_SYNC_RTP) {
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guint64 ext_base = -1;
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gint64 rtp_delta = 0;
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use_rtp = TRUE;
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/* convert to extended RTP time */
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rtp_clock_base = gst_rtp_buffer_ext_timestamp (&ext_base, rtp_clock_base);
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/* sanity check; base rtp and provided clock_base should be close */
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if (rtp_clock_base >= base_rtptime) {
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if (rtp_clock_base - base_rtptime < 10 * clock_rate) {
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rtp_delta = base_time +
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gst_util_uint64_scale_int (rtp_clock_base - base_rtptime,
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GST_SECOND, clock_rate);
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} else {
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use_rtp = FALSE;
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}
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} else {
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} else {
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if (base_rtptime - rtp_clock_base < 10 * clock_rate) {
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use_rtp = FALSE;
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rtp_delta = base_time -
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gst_util_uint64_scale_int (base_rtptime - rtp_clock_base,
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GST_SECOND, clock_rate);
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} else {
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use_rtp = FALSE;
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}
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}
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}
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/* warn and bail for clarity out if no sane values */
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} else {
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if (!use_rtp) {
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if (base_rtptime - rtp_clock_base < 10 * clock_rate) {
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GST_WARNING_OBJECT (bin, "unable to sync to provided rtptime");
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rtp_delta = base_time -
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gst_util_uint64_scale_int (base_rtptime - rtp_clock_base,
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GST_SECOND, clock_rate);
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} else {
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use_rtp = FALSE;
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}
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}
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/* warn and bail for clarity out if no sane values */
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if (!use_rtp) {
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GST_WARNING_OBJECT (bin, "unable to sync to provided rtptime");
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return;
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}
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/* store to track changes */
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clock_base = rtp_delta;
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/* generate a fake as before,
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* now equating rtptime obtained from RTP-Info,
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* where the large time represent the otherwise irrelevant npt/ntp time */
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stream->rtp_delta = (GST_SECOND << 28) - rtp_delta;
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} else {
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clock_base = rtp_clock_base;
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}
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all_sync = TRUE;
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for (walk = client->streams; walk; walk = g_slist_next (walk)) {
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GstRtpBinStream *ostream = (GstRtpBinStream *) walk->data;
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if (!ostream->have_sync) {
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all_sync = FALSE;
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continue;
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}
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/* change in current stream's base from previously init'ed value
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* leads to reset of all stream's base */
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if (stream != ostream && stream->clock_base >= 0 &&
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(stream->clock_base != clock_base)) {
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GST_DEBUG_OBJECT (bin, "reset upon clock base change");
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ostream->clock_base = -100 * GST_SECOND;
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ostream->rtp_delta = 0;
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}
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if (ostream->rt_delta < min)
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min = ostream->rt_delta;
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if (ostream->rtp_delta < rtp_min)
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rtp_min = ostream->rtp_delta;
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}
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/* arrange to re-sync for each stream upon significant change,
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* e.g. post-seek */
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all_sync = all_sync && (stream->clock_base == clock_base);
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stream->clock_base = clock_base;
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/* may need init performed above later on, but nothing more to do now */
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if (client->nstreams <= 1)
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return;
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GST_DEBUG_OBJECT (bin, "client %p min delta %" G_GINT64_FORMAT
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" all sync %d", client, min, all_sync);
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GST_DEBUG_OBJECT (bin, "rtcp sync mode %d, use_rtp %d", rtcp_sync, use_rtp);
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switch (rtcp_sync) {
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case GST_RTP_BIN_RTCP_SYNC_RTP:
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if (!use_rtp)
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break;
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GST_DEBUG_OBJECT (bin, "using rtp generated reports; "
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"client %p min rtp delta %" G_GINT64_FORMAT, client, rtp_min);
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/* fall-through */
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case GST_RTP_BIN_RTCP_SYNC_INITIAL:
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/* if all have been synced already, do not bother further */
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if (all_sync) {
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GST_DEBUG_OBJECT (bin, "all streams already synced; done");
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return;
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return;
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}
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}
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/* store to track changes */
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break;
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clock_base = rtp_delta;
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default:
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/* generate a fake as before,
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break;
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* now equating rtptime obtained from RTP-Info,
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* where the large time represent the otherwise irrelevant npt/ntp time */
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stream->rtp_delta = (GST_SECOND << 28) - rtp_delta;
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} else {
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clock_base = rtp_clock_base;
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}
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all_sync = TRUE;
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for (walk = client->streams; walk; walk = g_slist_next (walk)) {
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GstRtpBinStream *ostream = (GstRtpBinStream *) walk->data;
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if (!ostream->have_sync) {
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all_sync = FALSE;
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continue;
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}
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/* change in current stream's base from previously init'ed value
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* leads to reset of all stream's base */
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if (stream != ostream && stream->clock_base >= 0 &&
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(stream->clock_base != clock_base)) {
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GST_DEBUG_OBJECT (bin, "reset upon clock base change");
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ostream->clock_base = -100 * GST_SECOND;
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ostream->rtp_delta = 0;
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}
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if (ostream->rt_delta < min)
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min = ostream->rt_delta;
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if (ostream->rtp_delta < rtp_min)
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rtp_min = ostream->rtp_delta;
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}
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/* arrange to re-sync for each stream upon significant change,
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* e.g. post-seek */
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all_sync = all_sync && (stream->clock_base == clock_base);
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stream->clock_base = clock_base;
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/* may need init performed above later on, but nothing more to do now */
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if (client->nstreams <= 1)
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return;
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GST_DEBUG_OBJECT (bin, "client %p min delta %" G_GINT64_FORMAT
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" all sync %d", client, min, all_sync);
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GST_DEBUG_OBJECT (bin, "rtcp sync mode %d, use_rtp %d", rtcp_sync, use_rtp);
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switch (rtcp_sync) {
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case GST_RTP_BIN_RTCP_SYNC_RTP:
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if (!use_rtp)
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break;
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GST_DEBUG_OBJECT (bin, "using rtp generated reports; "
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|
||||||
"client %p min rtp delta %" G_GINT64_FORMAT, client, rtp_min);
|
|
||||||
/* fall-through */
|
|
||||||
case GST_RTP_BIN_RTCP_SYNC_INITIAL:
|
|
||||||
/* if all have been synced already, do not bother further */
|
|
||||||
if (all_sync) {
|
|
||||||
GST_DEBUG_OBJECT (bin, "all streams already synced; done");
|
|
||||||
return;
|
|
||||||
}
|
|
||||||
break;
|
|
||||||
default:
|
|
||||||
break;
|
|
||||||
}
|
|
||||||
|
|
||||||
/* bail out if we adjusted recently enough */
|
|
||||||
if (all_sync && (ntpnstime - bin->priv->last_ntpnstime) <
|
|
||||||
bin->rtcp_sync_interval * GST_MSECOND) {
|
|
||||||
GST_DEBUG_OBJECT (bin, "discarding RTCP sender packet for sync; "
|
|
||||||
"previous sender info too recent "
|
|
||||||
"(previous NTP %" G_GUINT64_FORMAT ")", bin->priv->last_ntpnstime);
|
|
||||||
return;
|
|
||||||
}
|
|
||||||
bin->priv->last_ntpnstime = ntpnstime;
|
|
||||||
|
|
||||||
/* calculate offsets for each stream */
|
|
||||||
for (walk = client->streams; walk; walk = g_slist_next (walk)) {
|
|
||||||
GstRtpBinStream *ostream = (GstRtpBinStream *) walk->data;
|
|
||||||
gint64 ts_offset;
|
|
||||||
|
|
||||||
/* ignore streams for which we didn't receive an SR packet yet, we
|
|
||||||
* can't synchronize them yet. We can however sync other streams just
|
|
||||||
* fine. */
|
|
||||||
if (!ostream->have_sync)
|
|
||||||
continue;
|
|
||||||
|
|
||||||
/* calculate offset to our reference stream, this should always give a
|
|
||||||
* positive number. */
|
|
||||||
if (use_rtp)
|
|
||||||
ts_offset = ostream->rtp_delta - rtp_min;
|
|
||||||
else
|
|
||||||
ts_offset = ostream->rt_delta - min;
|
|
||||||
|
|
||||||
stream_set_ts_offset (bin, ostream, ts_offset, bin->max_ts_offset,
|
|
||||||
bin->min_ts_offset, TRUE);
|
|
||||||
}
|
|
||||||
}
|
}
|
||||||
|
|
||||||
|
/* bail out if we adjusted recently enough */
|
||||||
|
if (all_sync && (ntpnstime - bin->priv->last_ntpnstime) <
|
||||||
|
bin->rtcp_sync_interval * GST_MSECOND) {
|
||||||
|
GST_DEBUG_OBJECT (bin, "discarding RTCP sender packet for sync; "
|
||||||
|
"previous sender info too recent "
|
||||||
|
"(previous NTP %" G_GUINT64_FORMAT ")", bin->priv->last_ntpnstime);
|
||||||
|
return;
|
||||||
|
}
|
||||||
|
bin->priv->last_ntpnstime = ntpnstime;
|
||||||
|
|
||||||
|
/* calculate offsets for each stream */
|
||||||
|
for (walk = client->streams; walk; walk = g_slist_next (walk)) {
|
||||||
|
GstRtpBinStream *ostream = (GstRtpBinStream *) walk->data;
|
||||||
|
gint64 ts_offset;
|
||||||
|
|
||||||
|
/* ignore streams for which we didn't receive an SR packet yet, we
|
||||||
|
* can't synchronize them yet. We can however sync other streams just
|
||||||
|
* fine. */
|
||||||
|
if (!ostream->have_sync)
|
||||||
|
continue;
|
||||||
|
|
||||||
|
/* calculate offset to our reference stream, this should always give a
|
||||||
|
* positive number. */
|
||||||
|
if (use_rtp)
|
||||||
|
ts_offset = ostream->rtp_delta - rtp_min;
|
||||||
|
else
|
||||||
|
ts_offset = ostream->rt_delta - min;
|
||||||
|
|
||||||
|
stream_set_ts_offset (bin, ostream, ts_offset, bin->max_ts_offset,
|
||||||
|
bin->min_ts_offset, TRUE);
|
||||||
|
}
|
||||||
|
|
||||||
gst_rtp_bin_send_sync_event (stream);
|
gst_rtp_bin_send_sync_event (stream);
|
||||||
|
|
||||||
return;
|
return;
|
||||||
|
|||||||
Loading…
x
Reference in New Issue
Block a user