Chromium Code Reviews
chromiumcodereview-hr@appspot.gserviceaccount.com (chromiumcodereview-hr) | Please choose your nickname with Settings | Help | Chromium Project | Gerrit Changes | Sign out
(368)

Unified Diff: webrtc/video/video_receive_stream.cc

Issue 2480293002: New jitter buffer experiment. (Closed)
Patch Set: Nit fix. Created 4 years, 1 month ago
Use n/p to move between diff chunks; N/P to move between comments. Draft comments are only viewable by you.
Jump to:
View side-by-side diff with in-line comments
Download patch
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..ea0289a3d1ffb51b45e979500b70311137607067 100644
--- a/webrtc/video/video_receive_stream.cc
+++ b/webrtc/video/video_receive_stream.cc
@@ -18,12 +18,17 @@
#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/frame_object.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"
@@ -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(
@@ -209,8 +215,15 @@ VideoReceiveStream::VideoReceiveStream(
&config_,
&stats_proxy_,
process_thread_,
- congestion_controller_->GetRetransmissionRateLimiter()),
- rtp_stream_sync_(&video_receiver_, &rtp_stream_receiver_) {
+ congestion_controller_->GetRetransmissionRateLimiter(),
+ this, // NackSender
+ this, // KeyFrameRequestSender
+ this, // OnCompleteFrameCallback
+ timing_.get()),
+ rtp_stream_sync_(&video_receiver_, &rtp_stream_receiver_),
+ jitter_buffer_experiment_(
+ field_trial::FindFullName("WebRTC-NewVideoJitterBuffer") ==
+ "Enabled") {
LOG(LS_INFO) << "VideoReceiveStream: " << config_.ToString();
RTC_DCHECK(process_thread_);
@@ -230,6 +243,12 @@ VideoReceiveStream::VideoReceiveStream(
video_receiver_.SetRenderDelay(config.render_delay_ms);
+ 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);
+ }
+ }
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)
+ 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(
« webrtc/modules/video_coding/packet_buffer.cc ('K') | « webrtc/video/video_receive_stream.h ('k') | no next file » | no next file with comments »

Powered by Google App Engine
This is Rietveld 408576698