Inform jitter buffer about FlexFEC protection.

webrtc/video/video_receive_stream.cc

 /*
  *  Copyright (c) 2013 The WebRTC project authors. All Rights Reserved.
  *
  *  Use of this source code is governed by a BSD-style license
  *  that can be found in the LICENSE file in the root of the source
  *  tree. An additional intellectual property rights grant can be found
  *  in the file PATENTS.  All contributing project authors may
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
@@ skipping 168 matching lines @@
       kDefaultStartBitrate;
 
   return codec;
 }
 }  // namespace
 
 namespace internal {
 
 VideoReceiveStream::VideoReceiveStream(
     int num_cpu_cores,
+    bool protected_by_flexfec,
     CongestionController* congestion_controller,
     PacketRouter* packet_router,
     VideoReceiveStream::Config config,
     webrtc::VoiceEngine* voice_engine,
     ProcessThread* process_thread,
     CallStats* call_stats,
     VieRemb* remb)
     : transport_adapter_(config.rtcp_send_transport),
       config_(std::move(config)),
       num_cpu_cores_(num_cpu_cores),
+      protected_by_flexfec_(protected_by_flexfec),
       process_thread_(process_thread),
       clock_(Clock::GetRealTimeClock()),
       decode_thread_(DecodeThreadFunction, this, "DecodingThread"),
       congestion_controller_(congestion_controller),
       call_stats_(call_stats),
       timing_(new VCMTiming(clock_)),
       video_receiver_(clock_, nullptr, this, timing_.get(), this, this),
       stats_proxy_(&config_, clock_),
-      rtp_stream_receiver_(
-          &video_receiver_,
-          congestion_controller_->GetRemoteBitrateEstimator(
-              UseSendSideBwe(config_)),
-          &transport_adapter_,
-          call_stats_->rtcp_rtt_stats(),
-          packet_router,
-          remb,
-          &config_,
-          &stats_proxy_,
-          process_thread_,
-          this,  // NackSender
-          this,  // KeyFrameRequestSender
-          this,  // OnCompleteFrameCallback
-          timing_.get()),
+      rtp_stream_receiver_(&video_receiver_,
+                           congestion_controller_->GetRemoteBitrateEstimator(
+                               UseSendSideBwe(config_)),
+                           &transport_adapter_,
+                           call_stats_->rtcp_rtt_stats(),
+                           packet_router,
+                           remb,
+                           &config_,
+                           &stats_proxy_,
+                           process_thread_,
+                           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_);
   RTC_DCHECK(congestion_controller_);
   RTC_DCHECK(call_stats_);
 
@@ skipping 58 matching lines @@
     rtp_stream_sync_.ConfigureSync(audio_channel_id, voe_sync_interface);
     voe_sync_interface->Release();
   } else {
     rtp_stream_sync_.ConfigureSync(-1, nullptr);
   }
 }
 
 void VideoReceiveStream::Start() {
   if (decode_thread_.IsRunning())
     return;
+
+  bool protected_by_fec =
+      protected_by_flexfec_ || rtp_stream_receiver_.IsUlpfecEnabled();
+
   if (jitter_buffer_experiment_) {
     frame_buffer_->Start();
     call_stats_->RegisterStatsObserver(&rtp_stream_receiver_);
 
-    if (rtp_stream_receiver_.IsRetransmissionsEnabled() &&
-        rtp_stream_receiver_.IsUlpfecEnabled()) {
+    if (rtp_stream_receiver_.IsRetransmissionsEnabled() && protected_by_fec) {
       frame_buffer_->SetProtectionMode(kProtectionNackFEC);
     }
   }
   transport_adapter_.Enable();
   rtc::VideoSinkInterface<VideoFrame>* renderer = nullptr;
   if (config_.renderer) {
     if (config_.disable_prerenderer_smoothing) {
       renderer = this;
     } else {
       incoming_video_stream_.reset(
           new IncomingVideoStream(config_.render_delay_ms, this));
       renderer = incoming_video_stream_.get();
     }
   }
   RTC_DCHECK(renderer != nullptr);
 
   for (const Decoder& decoder : config_.decoders) {
     video_receiver_.RegisterExternalDecoder(decoder.decoder,
                                             decoder.payload_type);
     VideoCodec codec = CreateDecoderVideoCodec(decoder);
     RTC_CHECK(
         rtp_stream_receiver_.AddReceiveCodec(codec, decoder.codec_params));
     RTC_CHECK_EQ(VCM_OK, video_receiver_.RegisterReceiveCodec(
                              &codec, num_cpu_cores_, false));
   }
 
   video_stream_decoder_.reset(new VideoStreamDecoder(
       &video_receiver_, &rtp_stream_receiver_, &rtp_stream_receiver_,
-      rtp_stream_receiver_.IsRetransmissionsEnabled(),
-      rtp_stream_receiver_.IsUlpfecEnabled(), &stats_proxy_, renderer,
-      config_.pre_render_callback));
+      rtp_stream_receiver_.IsRetransmissionsEnabled(), protected_by_fec,
+      &stats_proxy_, renderer, config_.pre_render_callback));
   // Register the channel to receive stats updates.
   call_stats_->RegisterStatsObserver(video_stream_decoder_.get());
   // Start the decode thread
   decode_thread_.Start();
   decode_thread_.SetPriority(rtc::kHighestPriority);
   rtp_stream_receiver_.StartReceive();
 }
 
 void VideoReceiveStream::Stop() {
   rtp_stream_receiver_.StopReceive();
@@ skipping 135 matching lines @@
                       << " ms, requesting keyframe.";
       RequestKeyFrame();
     }
   } else {
     video_receiver_.Decode(kMaxDecodeWaitTimeMs);
   }
 }
 
 }  // namespace internal
 }  // namespace webrtc