Use ChannelProxy for most calls on voe::Channel in Audio[Receive|Send]Stream.

webrtc/voice_engine/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 3397 matching lines @@
   CriticalSectionScoped cs(video_sync_lock_.get());
   if (_average_jitter_buffer_delay_us == 0) {
     return false;
   }
   *jitter_buffer_delay_ms = (_average_jitter_buffer_delay_us + 500) / 1000 +
       _recPacketDelayMs;
   *playout_buffer_delay_ms = playout_delay_ms_;
   return true;
 }
 
+uint32_t Channel::GetDelayEstimate() const {
+  int jitter_buffer_delay_ms = 0;
+  int playout_buffer_delay_ms = 0;
+  GetDelayEstimate(&jitter_buffer_delay_ms, &playout_buffer_delay_ms);
+  return jitter_buffer_delay_ms + playout_buffer_delay_ms;
+}
+
 int Channel::LeastRequiredDelayMs() const {
   return audio_coding_->LeastRequiredDelayMs();
 }
 
 int
 Channel::SetMinimumPlayoutDelay(int delayMs)
 {
     WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId,_channelId),
                  "Channel::SetMinimumPlayoutDelay()");
     if ((delayMs < kVoiceEngineMinMinPlayoutDelayMs) ||
@@ skipping 507 matching lines @@
   int64_t min_rtt = 0;
   if (_rtpRtcpModule->RTT(remoteSSRC, &rtt, &avg_rtt, &min_rtt, &max_rtt)
       != 0) {
     return 0;
   }
   return rtt;
 }
 
 }  // namespace voe
 }  // namespace webrtc