Support for remote audio into tracks

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.
  */
 
 #include "webrtc/voice_engine/channel_proxy.h"
 
+#include <utility>
+
+#include "webrtc/audio/audio_sink.h"
 #include "webrtc/base/checks.h"
 #include "webrtc/voice_engine/channel.h"
 
 namespace webrtc {
 namespace voe {
 ChannelProxy::ChannelProxy() : channel_owner_(nullptr) {}
 
 ChannelProxy::ChannelProxy(const ChannelOwner& channel_owner) :
     channel_owner_(channel_owner) {
   RTC_CHECK(channel_owner_.channel());
 }
 
+ChannelProxy::~ChannelProxy() {}
+
 void ChannelProxy::SetRTCPStatus(bool enable) {
   channel()->SetRTCPStatus(enable);
 }
 
 void ChannelProxy::SetLocalSSRC(uint32_t ssrc) {
   RTC_DCHECK(thread_checker_.CalledOnValidThread());
   int error = channel()->SetLocalSSRC(ssrc);
   RTC_DCHECK_EQ(0, error);
 }
 
@@ skipping 92 matching lines @@
   return channel()->SetSendTelephoneEventPayloadType(payload_type) == 0;
 }
 
 bool ChannelProxy::SendTelephoneEventOutband(uint8_t event,
                                              uint32_t duration_ms) {
   RTC_DCHECK(thread_checker_.CalledOnValidThread());
   return
       channel()->SendTelephoneEventOutband(event, duration_ms, 10, false) == 0;
 }
 
+void ChannelProxy::SetSink(rtc::scoped_ptr<AudioSinkInterface> sink) {
+  RTC_DCHECK(thread_checker_.CalledOnValidThread());
+  channel()->SetSink(std::move(sink));
+}
+
 Channel* ChannelProxy::channel() const {
   RTC_DCHECK(channel_owner_.channel());
   return channel_owner_.channel();
 }
 
 }  // namespace voe
 }  // namespace webrtc