Extract ViESyncModule outside ViEChannel.

webrtc/video/vie_channel.cc

 /*
  *  Copyright (c) 2012 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 77 matching lines @@
                        PacedSender* paced_sender,
                        PacketRouter* packet_router,
                        size_t max_rtp_streams,
                        bool sender)
     : sender_(sender),
       module_process_thread_(module_process_thread),
       send_payload_router_(send_payload_router),
       vcm_protection_callback_(new ViEChannelProtectionCallback(this)),
       vcm_(vcm),
       vie_receiver_(vcm_, remote_bitrate_estimator, this),
-      vie_sync_(vcm_),
       stats_observer_(new ChannelStatsObserver(this)),
       receive_stats_callback_(nullptr),
       incoming_video_stream_(nullptr),
       intra_frame_observer_(intra_frame_observer),
       rtt_stats_(rtt_stats),
       paced_sender_(paced_sender),
       packet_router_(packet_router),
       bandwidth_observer_(bandwidth_observer),
       transport_feedback_observer_(transport_feedback_observer),
       nack_history_size_sender_(kMinSendSidePacketHistorySize),
@@ skipping 50 matching lines @@
         std::vector<RtpRtcp*>(1, rtp_rtcp_modules_[0]));
     RTC_DCHECK(!send_payload_router_->active());
   } else {
     if (vcm_->RegisterReceiveCallback(this) != 0) {
       return -1;
     }
     vcm_->RegisterFrameTypeCallback(this);
     vcm_->RegisterReceiveStatisticsCallback(this);
     vcm_->RegisterDecoderTimingCallback(this);
     vcm_->SetRenderDelay(kDefaultRenderDelayMs);
-    module_process_thread_->RegisterModule(&vie_sync_);
   }
   return 0;
 }
 
 ViEChannel::~ViEChannel() {
   UpdateHistograms();
   // Make sure we don't get more callbacks from the RTP module.
   module_process_thread_->DeRegisterModule(
       vie_receiver_.GetReceiveStatistics());
   if (sender_) {
     send_payload_router_->SetSendingRtpModules(std::vector<RtpRtcp*>());
-  } else {
-    module_process_thread_->DeRegisterModule(&vie_sync_);
   }
   for (size_t i = 0; i < num_active_rtp_rtcp_modules_; ++i)
     packet_router_->RemoveRtpModule(rtp_rtcp_modules_[i], sender_);
   for (RtpRtcp* rtp_rtcp : rtp_rtcp_modules_) {
     module_process_thread_->DeRegisterModule(rtp_rtcp);
     delete rtp_rtcp;
   }
 }
 
 void ViEChannel::UpdateHistograms() {
@@ skipping 800 matching lines @@
     rtp_rtcp->SetRTCPStatus(RtcpMode::kCompound);
     modules.push_back(rtp_rtcp);
     // Receive statistics and remote bitrate estimator should only be set for
     // the primary (first) module.
     configuration.receive_statistics = null_receive_statistics;
     configuration.remote_bitrate_estimator = nullptr;
   }
   return modules;
 }
 
-int32_t ViEChannel::SetVoiceChannel(int32_t ve_channel_id,
-                                    VoEVideoSync* ve_sync_interface) {
-  RTC_DCHECK(!sender_);
-  return vie_sync_.ConfigureSync(ve_channel_id, ve_sync_interface,
-                                 rtp_rtcp_modules_[0],
-                                 vie_receiver_.GetRtpReceiver());
-}
-
-int32_t ViEChannel::VoiceChannel() {
-  RTC_DCHECK(!sender_);
-  return vie_sync_.VoiceChannel();
-}
-
 void ViEChannel::RegisterPreRenderCallback(
     I420FrameCallback* pre_render_callback) {
   RTC_DCHECK(!sender_);
   rtc::CritScope lock(&crit_);
   pre_render_callback_ = pre_render_callback;
 }
 
 // TODO(pbos): Remove as soon as audio can handle a changing payload type
 // without this callback.
 int32_t ViEChannel::OnInitializeDecoder(
@@ skipping 22 matching lines @@
   rtc::CritScope lock(&crit_);
   receive_stats_callback_ = receive_statistics_proxy;
 }
 
 void ViEChannel::SetIncomingVideoStream(
     IncomingVideoStream* incoming_video_stream) {
   rtc::CritScope lock(&crit_);
   incoming_video_stream_ = incoming_video_stream;
 }
 }  // namespace webrtc