From bdbc8895f3a630a4fe28d4661d8e71877ecaf14d Mon Sep 17 00:00:00 2001 From: philipel Date: Tue, 22 Aug 2017 02:08:51 -0700 Subject: [PATCH] Revert of Fix the video buffer size should take rtt into consideration (patchset #3 id:40001 of https://codereview.chromium.org/2980413002/ ) 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: https://chromium.googlesource.com/external/webrtc/+/f1e08d0b5848d32fd31c5b6e4e570115c32b7ce5 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} --- AUTHORS | 1 - webrtc/modules/video_coding/frame_buffer2.cc | 7 ------- webrtc/modules/video_coding/frame_buffer2.h | 3 --- webrtc/video/video_receive_stream.cc | 6 ------ webrtc/video/video_receive_stream.h | 6 +----- 5 files changed, 1 insertion(+), 22 deletions(-) diff --git a/AUTHORS b/AUTHORS index 3bf6671fb2..ed702b6b66 100644 --- a/AUTHORS +++ b/AUTHORS @@ -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> diff --git a/webrtc/modules/video_coding/frame_buffer2.cc b/webrtc/modules/video_coding/frame_buffer2.cc index 016a24e807..a783657f3e 100644 --- a/webrtc/modules/video_coding/frame_buffer2.cc +++ b/webrtc/modules/video_coding/frame_buffer2.cc @@ -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)) diff --git a/webrtc/modules/video_coding/frame_buffer2.h b/webrtc/modules/video_coding/frame_buffer2.h index 8de0680ef9..13dd341aaa 100644 --- a/webrtc/modules/video_coding/frame_buffer2.h +++ b/webrtc/modules/video_coding/frame_buffer2.h @@ -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) {} diff --git a/webrtc/video/video_receive_stream.cc b/webrtc/video/video_receive_stream.cc index 50b3321ee1..59d193377a 100644 --- a/webrtc/video/video_receive_stream.cc +++ b/webrtc/video/video_receive_stream.cc @@ -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; diff --git a/webrtc/video/video_receive_stream.h b/webrtc/video/video_receive_stream.h index dddb5f0568..bd70d1e5bf 100644 --- a/webrtc/video/video_receive_stream.h +++ b/webrtc/video/video_receive_stream.h @@ -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 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 GetInfo() const override;