Revert of Use VoiceChannel/VideoChannel directly from RtpSender/RtpReceiver.

webrtc/api/remoteaudiosource.cc

 /*
  *  Copyright 2014 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/api/remoteaudiosource.h"
 
 #include <algorithm>
 #include <functional>
 #include <memory>
 #include <utility>
 
+#include "webrtc/api/mediastreamprovider.h"
 #include "webrtc/base/checks.h"
 #include "webrtc/base/constructormagic.h"
 #include "webrtc/base/logging.h"
 #include "webrtc/base/thread.h"
 
 namespace webrtc {
 
 class RemoteAudioSource::MessageHandler : public rtc::MessageHandler {
  public:
   explicit MessageHandler(RemoteAudioSource* source) : source_(source) {}
 
  private:
   ~MessageHandler() override {}
 
   void OnMessage(rtc::Message* msg) override {
     source_->OnMessage(msg);
     delete this;
   }
 
   const rtc::scoped_refptr<RemoteAudioSource> source_;
   RTC_DISALLOW_IMPLICIT_CONSTRUCTORS(MessageHandler);
 };
 
 class RemoteAudioSource::Sink : public AudioSinkInterface {
  public:
   explicit Sink(RemoteAudioSource* source) : source_(source) {}
-  ~Sink() override { source_->OnAudioChannelGone(); }
+  ~Sink() override { source_->OnAudioProviderGone(); }
 
  private:
   void OnData(const AudioSinkInterface::Data& audio) override {
     if (source_)
       source_->OnData(audio);
   }
 
   const rtc::scoped_refptr<RemoteAudioSource> source_;
   RTC_DISALLOW_IMPLICIT_CONSTRUCTORS(Sink);
 };
 
 rtc::scoped_refptr<RemoteAudioSource> RemoteAudioSource::Create(
     uint32_t ssrc,
-    cricket::VoiceChannel* channel) {
+    AudioProviderInterface* provider) {
   rtc::scoped_refptr<RemoteAudioSource> ret(
       new rtc::RefCountedObject<RemoteAudioSource>());
-  ret->Initialize(ssrc, channel);
+  ret->Initialize(ssrc, provider);
   return ret;
 }
 
 RemoteAudioSource::RemoteAudioSource()
     : main_thread_(rtc::Thread::Current()),
       state_(MediaSourceInterface::kLive) {
   RTC_DCHECK(main_thread_);
 }
 
 RemoteAudioSource::~RemoteAudioSource() {
   RTC_DCHECK(main_thread_->IsCurrent());
   RTC_DCHECK(audio_observers_.empty());
   RTC_DCHECK(sinks_.empty());
 }
 
 void RemoteAudioSource::Initialize(uint32_t ssrc,
-                                   cricket::VoiceChannel* channel) {
+                                   AudioProviderInterface* provider) {
   RTC_DCHECK(main_thread_->IsCurrent());
-  // To make sure we always get notified when the channel goes out of scope,
+  // To make sure we always get notified when the provider goes out of scope,
   // we register for callbacks here and not on demand in AddSink.
-  if (channel) {  // May be null in tests.
-    channel->SetRawAudioSink(
+  if (provider) {  // May be null in tests.
+    provider->SetRawAudioSink(
         ssrc, std::unique_ptr<AudioSinkInterface>(new Sink(this)));
   }
 }
 
 MediaSourceInterface::SourceState RemoteAudioSource::state() const {
   RTC_DCHECK(main_thread_->IsCurrent());
   return state_;
 }
 
 bool RemoteAudioSource::remote() const {
@@ skipping 43 matching lines @@
 
 void RemoteAudioSource::OnData(const AudioSinkInterface::Data& audio) {
   // Called on the externally-owned audio callback thread, via/from webrtc.
   rtc::CritScope lock(&sink_lock_);
   for (auto* sink : sinks_) {
     sink->OnData(audio.data, 16, audio.sample_rate, audio.channels,
                  audio.samples_per_channel);
   }
 }
 
-void RemoteAudioSource::OnAudioChannelGone() {
-  // Called when the audio channel is deleted.  It may be the worker thread
+void RemoteAudioSource::OnAudioProviderGone() {
+  // Called when the data provider is deleted.  It may be the worker thread
   // in libjingle or may be a different worker thread.
   main_thread_->Post(RTC_FROM_HERE, new MessageHandler(this));
 }
 
 void RemoteAudioSource::OnMessage(rtc::Message* msg) {
   RTC_DCHECK(main_thread_->IsCurrent());
   sinks_.clear();
   state_ = MediaSourceInterface::kEnded;
   FireOnChanged();
 }
 
 }  // namespace webrtc