Stop using VoEVideoSync in Call/VideoReceiveStream.

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 15 matching lines @@
 #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"
-#include "webrtc/voice_engine/include/voe_video_sync.h"
 
 namespace webrtc {
 
 static bool UseSendSideBwe(const VideoReceiveStream::Config& config) {
   if (!config.rtp.transport_cc)
     return false;
   for (const auto& extension : config.rtp.extensions) {
     if (extension.uri == RtpExtension::kTransportSequenceNumberUri)
       return true;
   }
@@ skipping 137 matching lines @@
 }
 }  // namespace
 
 namespace internal {
 
 VideoReceiveStream::VideoReceiveStream(
     int num_cpu_cores,
     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),
       process_thread_(process_thread),
       clock_(Clock::GetRealTimeClock()),
       decode_thread_(DecodeThreadFunction, this, "DecodingThread"),
       congestion_controller_(congestion_controller),
@@ skipping 145 matching lines @@
     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();
   transport_adapter_.Disable();
 }
 
-void VideoReceiveStream::SetSyncChannel(VoiceEngine* voice_engine,
-                                        int audio_channel_id) {
-  if (voice_engine && audio_channel_id != -1) {
-    VoEVideoSync* voe_sync_interface = VoEVideoSync::GetInterface(voice_engine);
-    rtp_stream_sync_.ConfigureSync(audio_channel_id, voe_sync_interface);
-    voe_sync_interface->Release();
-  } else {
-    rtp_stream_sync_.ConfigureSync(-1, nullptr);
-  }
+void VideoReceiveStream::SetSync(Syncable* syncable) {
+  rtp_stream_sync_.ConfigureSync(syncable);
 }
 
 VideoReceiveStream::Stats VideoReceiveStream::GetStats() const {
   return stats_proxy_.GetStats();
 }
 
 // TODO(tommi): This method grabs a lock 6 times.
 void VideoReceiveStream::OnFrame(const VideoFrame& video_frame) {
   // TODO(tommi): OnDecodedFrame grabs a lock, incidentally the same lock
   // that OnSyncOffsetUpdated() and OnRenderedFrame() below grab.
@@ skipping 96 matching lines @@
     RequestKeyFrame();
   }
 }
 
 void VideoReceiveStream::RequestKeyFrame() {
   rtp_stream_receiver_.RequestKeyFrame();
 }
 
 }  // namespace internal
 }  // namespace webrtc