Revert "Remove PlayoutDelayOracle and make RtpSenderVideo guarantee delivery"
This reverts commit 4f68f5398d7fa3d47c449e99893c9bea07bf7ca2. Reason for revert: Breaks downstream project Original change's description: > Remove PlayoutDelayOracle and make RtpSenderVideo guarantee delivery > > The PlayoutDelayOracle was responsible for making sure the PlayoutDelay > header extension was successfully propagated to the receiving side. Once > it was determined that the receiver had received a frame with the new > delay tag, it's no longer necessary to propagate. > > The issue with this implementation is that it is based on max > extended sequence number reported via RTCP, which makes it often slow > to react, could theoretically fail to produce desired outcome (max > received > X does not guarantee X was fully received and decoded), and > added a lot of code complexity. > > The guarantee of delivery can in fact be accomplished more reliably and > with less code by making sure to tag each frame until an undiscardable > frame is sent. > > This allows containing the logic fully within RTPSenderVideo. > > Bug: webrtc:11340 > Change-Id: I2d1d2b6b67f4f07b8b33336f8fcfcde724243eef > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/168221 > Reviewed-by: Stefan Holmer <stefan@webrtc.org> > Reviewed-by: Sebastian Jansson <srte@webrtc.org> > Commit-Queue: Erik Språng <sprang@webrtc.org> > Cr-Commit-Position: refs/heads/master@{#30473} TBR=sprang@webrtc.org,stefan@webrtc.org,srte@webrtc.org Change-Id: Ide922e680ae36bb69b95e58002482cf5ed57e254 No-Presubmit: true No-Tree-Checks: true No-Try: true Bug: webrtc:11340 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/168304 Reviewed-by: Erik Språng <sprang@webrtc.org> Commit-Queue: Erik Språng <sprang@webrtc.org> Cr-Commit-Position: refs/heads/master@{#30475}
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@ -68,6 +68,7 @@ ModuleRtpRtcpImpl::ModuleRtpRtcpImpl(const Configuration& configuration)
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nack_last_time_sent_full_ms_(0),
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nack_last_seq_number_sent_(0),
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remote_bitrate_(configuration.remote_bitrate_estimator),
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ack_observer_(configuration.ack_observer),
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rtt_stats_(configuration.rtt_stats),
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rtt_ms_(0) {
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if (!configuration.receiver_only) {
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@ -735,7 +736,7 @@ void ModuleRtpRtcpImpl::OnReceivedNack(
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void ModuleRtpRtcpImpl::OnReceivedRtcpReportBlocks(
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const ReportBlockList& report_blocks) {
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if (rtp_sender_) {
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if (ack_observer_) {
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uint32_t ssrc = SSRC();
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absl::optional<uint32_t> rtx_ssrc;
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if (rtp_sender_->packet_generator.RtxStatus() != kRtxOff) {
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@ -746,6 +747,8 @@ void ModuleRtpRtcpImpl::OnReceivedRtcpReportBlocks(
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if (ssrc == report_block.source_ssrc) {
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rtp_sender_->packet_generator.OnReceivedAckOnSsrc(
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report_block.extended_highest_sequence_number);
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ack_observer_->OnReceivedAck(
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report_block.extended_highest_sequence_number);
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} else if (rtx_ssrc && *rtx_ssrc == report_block.source_ssrc) {
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rtp_sender_->packet_generator.OnReceivedAckOnRtxSsrc(
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report_block.extended_highest_sequence_number);
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