Chromium Code Reviews| Index: webrtc/video/video_receive_stream.cc |
| diff --git a/webrtc/video/video_receive_stream.cc b/webrtc/video/video_receive_stream.cc |
| index 5173b2d4796aa5e8e974bfef5046001f671f77ff..44c9011da6c30c9d1b8bbd7ecd7d347ab4ff0ef7 100644 |
| --- a/webrtc/video/video_receive_stream.cc |
| +++ b/webrtc/video/video_receive_stream.cc |
| @@ -18,16 +18,21 @@ |
| #include "webrtc/base/checks.h" |
| #include "webrtc/base/logging.h" |
| +#include "webrtc/base/optional.h" |
| #include "webrtc/common_video/libyuv/include/webrtc_libyuv.h" |
| #include "webrtc/modules/congestion_controller/include/congestion_controller.h" |
| #include "webrtc/modules/utility/include/process_thread.h" |
| #include "webrtc/modules/video_coding/include/video_coding.h" |
| +#include "webrtc/modules/video_coding/jitter_estimator.h" |
| +#include "webrtc/modules/video_coding/timing.h" |
| #include "webrtc/modules/video_coding/utility/ivf_file_writer.h" |
| #include "webrtc/system_wrappers/include/clock.h" |
| +#include "webrtc/system_wrappers/include/field_trial.h" |
| #include "webrtc/video/call_stats.h" |
| #include "webrtc/video/receive_statistics_proxy.h" |
| #include "webrtc/video_receive_stream.h" |
| #include "webrtc/voice_engine/include/voe_video_sync.h" |
| +#include "webrtc/modules/video_coding/frame_object.h" |
|
stefan-webrtc
2016/11/08 10:41:35
Sort
philipel
2016/11/08 12:28:39
Done.
|
| namespace webrtc { |
| @@ -197,6 +202,7 @@ VideoReceiveStream::VideoReceiveStream( |
| call_stats_(call_stats), |
| video_receiver_(clock_, nullptr, this, this, this), |
| stats_proxy_(&config_, clock_), |
| + timing_(new VCMTiming(clock_)), |
| rtp_stream_receiver_( |
| &video_receiver_, |
| congestion_controller_->GetRemoteBitrateEstimator( |
| @@ -209,7 +215,11 @@ VideoReceiveStream::VideoReceiveStream( |
| &config_, |
| &stats_proxy_, |
| process_thread_, |
| - congestion_controller_->GetRetransmissionRateLimiter()), |
| + congestion_controller_->GetRetransmissionRateLimiter(), |
| + this, // NackSender |
| + this, // KeyFrameRequestSender |
| + this, // OnCompleteFrameCallback |
|
stefan-webrtc
2016/11/08 10:41:35
Should we consolidate these interfaces so that we
philipel
2016/11/08 12:28:39
To me it doesn't make much sense to have them in o
stefan-webrtc
2016/11/08 14:49:27
Acknowledged.
|
| + timing_.get()), |
| rtp_stream_sync_(&video_receiver_, &rtp_stream_receiver_) { |
| LOG(LS_INFO) << "VideoReceiveStream: " << config_.ToString(); |
| @@ -230,6 +240,15 @@ VideoReceiveStream::VideoReceiveStream( |
| video_receiver_.SetRenderDelay(config.render_delay_ms); |
| + jitter_buffer_experiment_ = |
| + field_trial::FindFullName("WebRTC-NewVideoJitterBuffer") == "Enabled"; |
| + |
| + if (jitter_buffer_experiment_) { |
| + jitter_estimator_.reset(new VCMJitterEstimator(clock_)); |
| + frame_buffer_.reset(new video_coding::FrameBuffer( |
| + clock_, jitter_estimator_.get(), timing_.get())); |
| + } |
| + |
| process_thread_->RegisterModule(&video_receiver_); |
| process_thread_->RegisterModule(&rtp_stream_sync_); |
| } |
| @@ -268,6 +287,15 @@ bool VideoReceiveStream::OnRecoveredPacket(const uint8_t* packet, |
| void VideoReceiveStream::Start() { |
| if (decode_thread_.IsRunning()) |
| return; |
| + if (jitter_buffer_experiment_) { |
| + frame_buffer_->Start(); |
| + call_stats_->RegisterStatsObserver(&rtp_stream_receiver_); |
| + |
| + if (rtp_stream_receiver_.IsRetransmissionsEnabled() && |
| + rtp_stream_receiver_.IsFecEnabled()) { |
| + frame_buffer_->SetProtectionMode(kProtectionNackFEC); |
|
stefan-webrtc
2016/11/08 10:41:35
No need to set kProtectionNack if fec is disabled?
philipel
2016/11/08 12:28:39
The FrameBuffer only change its behavior if the pr
stefan-webrtc
2016/11/08 14:49:27
Acknowledged.
|
| + } |
| + } |
| transport_adapter_.Enable(); |
| rtc::VideoSinkInterface<VideoFrame>* renderer = nullptr; |
| if (config_.renderer) { |
| @@ -310,6 +338,12 @@ void VideoReceiveStream::Stop() { |
| // stop immediately, instead of waiting for a timeout. Needs to be called |
| // before joining the decoder thread thread. |
| video_receiver_.TriggerDecoderShutdown(); |
| + |
| + if (jitter_buffer_experiment_) { |
| + frame_buffer_->Stop(); |
| + call_stats_->DeregisterStatsObserver(&rtp_stream_receiver_); |
| + } |
| + |
| if (decode_thread_.IsRunning()) { |
| decode_thread_.Stop(); |
| // Deregister external decoders so they are no longer running during |
| @@ -319,6 +353,7 @@ void VideoReceiveStream::Stop() { |
| for (const Decoder& decoder : config_.decoders) |
| video_receiver_.RegisterExternalDecoder(nullptr, decoder.payload_type); |
| } |
| + |
| call_stats_->DeregisterStatsObserver(video_stream_decoder_.get()); |
| video_stream_decoder_.reset(); |
| incoming_video_stream_.reset(); |
| @@ -365,6 +400,13 @@ void VideoReceiveStream::OnFrame(const VideoFrame& video_frame) { |
| stats_proxy_.OnRenderedFrame(video_frame); |
| } |
| +void VideoReceiveStream::OnCompleteFrame( |
| + std::unique_ptr<video_coding::FrameObject> frame) { |
| + int last_continuous_pid = frame_buffer_->InsertFrame(std::move(frame)); |
| + if (last_continuous_pid != -1) |
|
nisse-webrtc
2016/11/08 11:56:30
When using magic values like this, you have to mak
philipel
2016/11/08 12:28:39
Good point, but I have a few other changes I would
|
| + rtp_stream_receiver_.FrameContinuous(last_continuous_pid); |
| +} |
| + |
| // TODO(asapersson): Consider moving callback from video_encoder.h or |
| // creating a different callback. |
| EncodedImageCallback::Result VideoReceiveStream::OnEncodedImage( |
| @@ -397,7 +439,26 @@ bool VideoReceiveStream::DecodeThreadFunction(void* ptr) { |
| void VideoReceiveStream::Decode() { |
| static const int kMaxDecodeWaitTimeMs = 50; |
| - video_receiver_.Decode(kMaxDecodeWaitTimeMs); |
| + if (jitter_buffer_experiment_) { |
| + static const int kMaxWaitForFrameMs = 3000; |
| + std::unique_ptr<video_coding::FrameObject> frame; |
| + video_coding::FrameBuffer::ReturnReason res = |
| + frame_buffer_->NextFrame(kMaxWaitForFrameMs, &frame); |
| + |
| + if (res == video_coding::FrameBuffer::ReturnReason::kStopped) |
| + return; |
| + |
| + if (frame) { |
| + if (video_receiver_.Decode(frame.get()) == VCM_OK) |
| + rtp_stream_receiver_.FrameDecoded(frame->picture_id); |
| + } else { |
| + LOG(LS_WARNING) << "No decodable frame in " << kMaxWaitForFrameMs |
| + << " ms, requesting keyframe."; |
| + RequestKeyFrame(); |
| + } |
| + } else { |
| + video_receiver_.Decode(kMaxDecodeWaitTimeMs); |
| + } |
| } |
| void VideoReceiveStream::SendNack( |