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Revert of Fix the video buffer size should take rtt into consideration (patchset #3 id:40001 of https://codereview.chromium.org/2980413002/ )

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Reason for revert:
We are not certain this is the behavior we want.

Original issue's description:
> Fix the video buffer size should take rtt into consideration
>
> BUG=webrtc:8010
>
> Review-Url: https://codereview.webrtc.org/2980413002
> Cr-Commit-Position: refs/heads/master@{#19285}
> Committed: f1e08d0b58

TBR=sprang@webrtc.org,gustavogb@gmail.com
# Not skipping CQ checks because original CL landed more than 1 days ago.
BUG=webrtc:8010

Review-Url: https://codereview.webrtc.org/3002033002
Cr-Commit-Position: refs/heads/master@{#19442}
This commit is contained in:
philipel
2017-08-22 02:08:51 -07:00
committed by Commit Bot
parent 3d95a53c96
commit bdbc8895f3
5 changed files with 1 additions and 22 deletions
Download Patch File Download Diff File Expand all files Collapse all files

1
AUTHORS
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@ -61,7 +61,6 @@ Agora IO <*@agora.io>
ARM Holdings <*@arm.com>
BroadSoft Inc. <*@broadsoft.com>
Google Inc. <*@google.com>
Life On Air Inc. <*@lifeonair.com>
Intel Corporation <*@intel.com>
MIPS Technologies <*@mips.com>
Mozilla Foundation <*@mozilla.com>

7
webrtc/modules/video_coding/frame_buffer2.cc
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@ -147,8 +147,6 @@ FrameBuffer::ReturnReason FrameBuffer::NextFrame(
float rtt_mult = protection_mode_ == kProtectionNackFEC ? 0.0 : 1.0;
timing_->SetJitterDelay(jitter_estimator_->GetJitterEstimate(rtt_mult));
timing_->UpdateCurrentDelay(frame->RenderTime(), now_ms);
} else {
jitter_estimator_->FrameNacked();
}
// Gracefully handle bad RTP timestamps and render time issues.
@ -249,11 +247,6 @@ void FrameBuffer::Stop() {
new_continuous_frame_event_.Set();
}
void FrameBuffer::UpdateRtt(int64_t rtt_ms) {
rtc::CritScope lock(&crit_);
jitter_estimator_->UpdateRtt(rtt_ms);
}
bool FrameBuffer::ValidReferences(const FrameObject& frame) const {
for (size_t i = 0; i < frame.num_references; ++i) {
if (AheadOrAt(frame.references[i], frame.picture_id))

3
webrtc/modules/video_coding/frame_buffer2.h
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@ -74,9 +74,6 @@ class FrameBuffer {
// return immediately.
void Stop();
// Updates the RTT for jitter buffer estimation.
void UpdateRtt(int64_t rtt_ms);
private:
struct FrameKey {
FrameKey() : picture_id(0), spatial_layer(0) {}

6
webrtc/video/video_receive_stream.cc
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@ -267,7 +267,6 @@ void VideoReceiveStream::Start() {
frame_buffer_->Start();
call_stats_->RegisterStatsObserver(&rtp_video_stream_receiver_);
call_stats_->RegisterStatsObserver(this);
if (rtp_video_stream_receiver_.IsRetransmissionsEnabled() &&
protected_by_fec) {
@ -317,7 +316,6 @@ void VideoReceiveStream::Stop() {
rtp_video_stream_receiver_.StopReceive();
frame_buffer_->Stop();
call_stats_->DeregisterStatsObserver(this);
call_stats_->DeregisterStatsObserver(&rtp_video_stream_receiver_);
process_thread_->DeRegisterModule(&video_receiver_);
@ -445,10 +443,6 @@ void VideoReceiveStream::OnCompleteFrame(
rtp_video_stream_receiver_.FrameContinuous(last_continuous_pid);
}
void VideoReceiveStream::OnRttUpdate(int64_t avg_rtt_ms, int64_t max_rtt_ms) {
frame_buffer_->UpdateRtt(max_rtt_ms);
}
int VideoReceiveStream::id() const {
RTC_DCHECK_CALLED_SEQUENTIALLY(&worker_sequence_checker_);
return config_.rtp.remote_ssrc;

6
webrtc/video/video_receive_stream.h
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@ -49,8 +49,7 @@ class VideoReceiveStream : public webrtc::VideoReceiveStream,
public NackSender,
public KeyFrameRequestSender,
public video_coding::OnCompleteFrameCallback,
public Syncable,
public CallStatsObserver {
public Syncable {
public:
VideoReceiveStream(RtpStreamReceiverControllerInterface* receiver_controller,
int num_cpu_cores,
@ -105,9 +104,6 @@ class VideoReceiveStream : public webrtc::VideoReceiveStream,
void OnCompleteFrame(
std::unique_ptr<video_coding::FrameObject> frame) override;
// Implements CallStatsObserver::OnRttUpdate
void OnRttUpdate(int64_t avg_rtt_ms, int64_t max_rtt_ms) override;
// Implements Syncable.
int id() const override;
rtc::Optional<Syncable::Info> GetInfo() const override;