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Revert "Remove PlayoutDelayOracle and make RtpSenderVideo guarantee delivery"

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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}
This commit is contained in:
Erik Språng
2020-02-06 16:04:48 +00:00
committed by Commit Bot
parent 67dba30178
commit 632a03c0cd
18 changed files with 278 additions and 162 deletions
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3
modules/rtp_rtcp/source/rtp_rtcp_impl_unittest.cc
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@ -17,6 +17,7 @@
#include "api/transport/field_trial_based_config.h"
#include "api/video_codecs/video_codec.h"
#include "modules/rtp_rtcp/include/rtp_rtcp_defines.h"
#include "modules/rtp_rtcp/source/playout_delay_oracle.h"
#include "modules/rtp_rtcp/source/rtcp_packet.h"
#include "modules/rtp_rtcp/source/rtcp_packet/nack.h"
#include "modules/rtp_rtcp/source/rtp_packet_received.h"
@ -181,6 +182,7 @@ class RtpRtcpImplTest : public ::testing::Test {
RTPSenderVideo::Config video_config;
video_config.clock = &clock_;
video_config.rtp_sender = sender_.impl_->RtpSender();
video_config.playout_delay_oracle = &playout_delay_oracle_;
video_config.field_trials = &field_trials;
sender_video_ = std::make_unique<RTPSenderVideo>(video_config);
@ -199,6 +201,7 @@ class RtpRtcpImplTest : public ::testing::Test {
SimulatedClock clock_;
RtpRtcpModule sender_;
PlayoutDelayOracle playout_delay_oracle_;
std::unique_ptr<RTPSenderVideo> sender_video_;
RtpRtcpModule receiver_;
VideoCodec codec_;