Stop using VoEVideoSync in Call/VideoReceiveStream.

webrtc/voice_engine/channel_proxy.cc

 /*
  *  Copyright (c) 2015 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 231 matching lines @@
     const ChannelProxy& send_channel_proxy) {
   RTC_DCHECK(thread_checker_.CalledOnValidThread());
   channel()->set_associate_send_channel(send_channel_proxy.channel_owner_);
 }
 
 void ChannelProxy::DisassociateSendChannel() {
   RTC_DCHECK(thread_checker_.CalledOnValidThread());
   channel()->set_associate_send_channel(ChannelOwner(nullptr));
 }
 
+void ChannelProxy::GetRtpRtcp(RtpRtcp** rtp_rtcp,
+                              RtpReceiver** rtp_receiver) const {
+  RTC_DCHECK(thread_checker_.CalledOnValidThread());
+  RTC_DCHECK(rtp_rtcp);
+  RTC_DCHECK(rtp_receiver);
+  int error = channel()->GetRtpRtcp(rtp_rtcp, rtp_receiver);
+  RTC_DCHECK_EQ(0, error);
+}
+
+void ChannelProxy::GetDelayEstimate(int* jitter_buffer_delay_ms,
+                                    int* playout_buffer_delay_ms) const {
+  // Called on Call's module_process_thread_.
+  RTC_DCHECK(jitter_buffer_delay_ms);
+  RTC_DCHECK(playout_buffer_delay_ms);
+  bool error = channel()->GetDelayEstimate(jitter_buffer_delay_ms,
+                                           playout_buffer_delay_ms);
+  RTC_DCHECK(error);
+}
+
+uint32_t ChannelProxy::GetPlayoutTimestamp() const {
+  // Called on video capture thread.
+  unsigned int timestamp = 0;
+  int error = channel()->GetPlayoutTimestamp(timestamp);
+  RTC_DCHECK(!error || timestamp == 0);
+  return timestamp;
+}
+
+void ChannelProxy::SetMinimumPlayoutDelay(int delay_ms) {
+  // Called on Call's module_process_thread_.
+  // Limit to range accepted by both VoE and ACM, so we're at least getting as
+  // close as possible, instead of failing.
+  delay_ms = std::max(0, std::min(delay_ms, 10000));
+  int error = channel()->SetMinimumPlayoutDelay(delay_ms);
+  RTC_DCHECK_EQ(0, error);
+}
+
 void ChannelProxy::SetRtcpRttStats(RtcpRttStats* rtcp_rtt_stats) {
   RTC_DCHECK(thread_checker_.CalledOnValidThread());
   channel()->SetRtcpRttStats(rtcp_rtt_stats);
 }
 
 Channel* ChannelProxy::channel() const {
   RTC_DCHECK(channel_owner_.channel());
   return channel_owner_.channel();
 }
 
 }  // namespace voe
 }  // namespace webrtc