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Reland "Remove PlayoutDelayOracle and make RtpSenderVideo guarantee delivery"

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This is a reland of 4f68f5398d7fa3d47c449e99893c9bea07bf7ca2

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=stefan@webrtc.org

Bug: webrtc:11340
Change-Id: I2fdd0004121b13b96497b21e052359e31d0c477a
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/168305
Commit-Queue: Erik Språng <sprang@webrtc.org>
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30479}
This commit is contained in:
Erik Språng
2020-02-06 17:10:08 +01:00
committed by Commit Bot
parent ecd6fc84cf
commit 56e611bbda
18 changed files with 163 additions and 277 deletions
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5
modules/rtp_rtcp/source/rtp_rtcp_impl.cc
View File

@ -68,7 +68,6 @@ ModuleRtpRtcpImpl::ModuleRtpRtcpImpl(const Configuration& configuration)
nack_last_time_sent_full_ms_(0),
nack_last_seq_number_sent_(0),
remote_bitrate_(configuration.remote_bitrate_estimator),
ack_observer_(configuration.ack_observer),
rtt_stats_(configuration.rtt_stats),
rtt_ms_(0) {
if (!configuration.receiver_only) {
@ -736,7 +735,7 @@ void ModuleRtpRtcpImpl::OnReceivedNack(
void ModuleRtpRtcpImpl::OnReceivedRtcpReportBlocks(
const ReportBlockList& report_blocks) {
if (ack_observer_) {
if (rtp_sender_) {
uint32_t ssrc = SSRC();
absl::optional<uint32_t> rtx_ssrc;
if (rtp_sender_->packet_generator.RtxStatus() != kRtxOff) {
@ -747,8 +746,6 @@ void ModuleRtpRtcpImpl::OnReceivedRtcpReportBlocks(
if (ssrc == report_block.source_ssrc) {
rtp_sender_->packet_generator.OnReceivedAckOnSsrc(
report_block.extended_highest_sequence_number);
ack_observer_->OnReceivedAck(
report_block.extended_highest_sequence_number);
} else if (rtx_ssrc && *rtx_ssrc == report_block.source_ssrc) {
rtp_sender_->packet_generator.OnReceivedAckOnRtxSsrc(
report_block.extended_highest_sequence_number);