Use vcm::VideoReceiver on the receive side.

webrtc/video/vie_sync_module.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.
  */
 
 #include "webrtc/video/vie_sync_module.h"
 
 #include "webrtc/base/checks.h"
 #include "webrtc/base/logging.h"
 #include "webrtc/base/trace_event.h"
 #include "webrtc/modules/rtp_rtcp/include/rtp_receiver.h"
 #include "webrtc/modules/rtp_rtcp/include/rtp_rtcp.h"
-#include "webrtc/modules/video_coding/include/video_coding.h"
+#include "webrtc/modules/video_coding/video_coding_impl.h"
 #include "webrtc/system_wrappers/include/clock.h"
 #include "webrtc/video/stream_synchronization.h"
 #include "webrtc/video_frame.h"
 #include "webrtc/voice_engine/include/voe_video_sync.h"
 
 namespace webrtc {
 namespace {
 int UpdateMeasurements(StreamSynchronization::Measurements* stream,
                        const RtpRtcp& rtp_rtcp, const RtpReceiver& receiver) {
   if (!receiver.Timestamp(&stream->latest_timestamp))
@@ skipping 12 matching lines @@
   bool new_rtcp_sr = false;
   if (!UpdateRtcpList(
       ntp_secs, ntp_frac, rtp_timestamp, &stream->rtcp, &new_rtcp_sr)) {
     return -1;
   }
 
   return 0;
 }
 }  // namespace
 
-ViESyncModule::ViESyncModule(VideoCodingModule* vcm)
-    : vcm_(vcm),
+ViESyncModule::ViESyncModule(vcm::VideoReceiver* video_receiver)
+    : video_receiver_(video_receiver),
       clock_(Clock::GetRealTimeClock()),
-      video_receiver_(nullptr),
+      rtp_receiver_(nullptr),
       video_rtp_rtcp_(nullptr),
       voe_channel_id_(-1),
       voe_sync_interface_(nullptr),
       last_sync_time_(TickTime::Now()),
       sync_() {}
 
 ViESyncModule::~ViESyncModule() {
 }
 
 void ViESyncModule::ConfigureSync(int voe_channel_id,
                                   VoEVideoSync* voe_sync_interface,
                                   RtpRtcp* video_rtcp_module,
-                                  RtpReceiver* video_receiver) {
+                                  RtpReceiver* rtp_receiver) {
   if (voe_channel_id != -1)
     RTC_DCHECK(voe_sync_interface);
   rtc::CritScope lock(&data_cs_);
   // Prevent expensive no-ops.
   if (voe_channel_id_ == voe_channel_id &&
       voe_sync_interface_ == voe_sync_interface &&
-      video_receiver_ == video_receiver &&
-      video_rtp_rtcp_ == video_rtcp_module) {
+      rtp_receiver_ == rtp_receiver && video_rtp_rtcp_ == video_rtcp_module) {
     return;
   }
   voe_channel_id_ = voe_channel_id;
   voe_sync_interface_ = voe_sync_interface;
-  video_receiver_ = video_receiver;
+  rtp_receiver_ = rtp_receiver;
   video_rtp_rtcp_ = video_rtcp_module;
   sync_.reset(
       new StreamSynchronization(video_rtp_rtcp_->SSRC(), voe_channel_id));
 }
 
 int64_t ViESyncModule::TimeUntilNextProcess() {
   const int64_t kSyncIntervalMs = 1000;
   return kSyncIntervalMs - (TickTime::Now() - last_sync_time_).Milliseconds();
 }
 
 void ViESyncModule::Process() {
   rtc::CritScope lock(&data_cs_);
   last_sync_time_ = TickTime::Now();
 
-  const int current_video_delay_ms = vcm_->Delay();
+  const int current_video_delay_ms = video_receiver_->Delay();
 
   if (voe_channel_id_ == -1) {
     return;
   }
   assert(video_rtp_rtcp_ && voe_sync_interface_);
   assert(sync_.get());
 
   int audio_jitter_buffer_delay_ms = 0;
   int playout_buffer_delay_ms = 0;
   if (voe_sync_interface_->GetDelayEstimate(voe_channel_id_,
                                             &audio_jitter_buffer_delay_ms,
                                             &playout_buffer_delay_ms) != 0) {
     return;
   }
   const int current_audio_delay_ms = audio_jitter_buffer_delay_ms +
       playout_buffer_delay_ms;
 
   RtpRtcp* voice_rtp_rtcp = nullptr;
   RtpReceiver* voice_receiver = nullptr;
   if (voe_sync_interface_->GetRtpRtcp(voe_channel_id_, &voice_rtp_rtcp,
                                       &voice_receiver) != 0) {
     return;
   }
   assert(voice_rtp_rtcp);
   assert(voice_receiver);
 
   if (UpdateMeasurements(&video_measurement_, *video_rtp_rtcp_,
-                         *video_receiver_) != 0) {
+                         *rtp_receiver_) != 0) {
     return;
   }
 
   if (UpdateMeasurements(&audio_measurement_, *voice_rtp_rtcp,
                          *voice_receiver) != 0) {
     return;
   }
 
   int relative_delay_ms;
   // Calculate how much later or earlier the audio stream is compared to video.
@@ skipping 13 matching lines @@
                             current_audio_delay_ms,
                             &target_audio_delay_ms,
                             &target_video_delay_ms)) {
     return;
   }
 
   if (voe_sync_interface_->SetMinimumPlayoutDelay(
       voe_channel_id_, target_audio_delay_ms) == -1) {
     LOG(LS_ERROR) << "Error setting voice delay.";
   }
-  vcm_->SetMinimumPlayoutDelay(target_video_delay_ms);
+  video_receiver_->SetMinimumPlayoutDelay(target_video_delay_ms);
 }
 
 bool ViESyncModule::GetStreamSyncOffsetInMs(const VideoFrame& frame,
                                             int64_t* stream_offset_ms) const {
   rtc::CritScope lock(&data_cs_);
   if (voe_channel_id_ == -1)
     return false;
 
   uint32_t playout_timestamp = 0;
   if (voe_sync_interface_->GetPlayoutTimestamp(voe_channel_id_,
@@ skipping 16 matching lines @@
   int64_t time_to_render_ms =
       frame.render_time_ms() - clock_->TimeInMilliseconds();
   if (time_to_render_ms > 0)
     latest_video_ntp += time_to_render_ms;
 
   *stream_offset_ms = latest_audio_ntp - latest_video_ntp;
   return true;
 }
 
 }  // namespace webrtc