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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@ -89,16 +89,8 @@ typedef SpatialLayer SimulcastStream;
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// Note: Given that this gets embedded in a union, it is up-to the owner to
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// initialize these values.
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struct PlayoutDelay {
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PlayoutDelay(int min_ms, int max_ms) : min_ms(min_ms), max_ms(max_ms) {}
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int min_ms;
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int max_ms;
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static PlayoutDelay Noop() { return PlayoutDelay(-1, -1); }
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bool IsNoop() const { return min_ms == -1 && max_ms == -1; }
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bool operator==(const PlayoutDelay& rhs) const {
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return min_ms == rhs.min_ms && max_ms == rhs.max_ms;
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}
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};
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} // namespace webrtc
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