Revert of Remove AudioTrackRenderer (patchset #3 id:40001 of https://codereview.webrtc.org/1399553003/ )

Reason for revert:
Breaks Chrome since its build files were not updated prior to file removal.

Original issue's description:
> - Remove AudioTrackRenderer.
> - Remove AddChannel/RemoveChannel from AudioRenderer interface.
>
> BUG=webrtc:4690
>
> Committed: https://crrev.com/1c0bb386b67835feb5934f503dddfe0912bce3ac
> Cr-Commit-Position: refs/heads/master@{#10226}

TBR=tommi@webrtc.org,solenberg@webrtc.org
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=webrtc:4690

Review URL: https://codereview.webrtc.org/1393343003

Cr-Commit-Position: refs/heads/master@{#10228}

3 files changed, 28 insertions, 4 deletions

diff --git a/talk/media/base/audiorenderer.h b/talk/media/base/audiorenderer.h
index 229c36e8b1..5c03576fa7 100644
--- a/talk/media/base/audiorenderer.h
+++ b/talk/media/base/audiorenderer.h
@@ -55,6 +55,20 @@ class AudioRenderer {
   // to the renderer at a time.
   virtual void SetSink(Sink* sink) {}
 
+  // Add the WebRtc VoE channel to the renderer.
+  // For local stream, multiple WebRtc VoE channels can be connected to the
+  // renderer. While for remote stream, only one WebRtc VoE channel can be
+  // connected to the renderer.
+  // TODO(xians): Remove this interface after Chrome switches to the
+  // AudioRenderer::Sink interface.
+  virtual void AddChannel(int channel_id) {}
+
+  // Remove the WebRtc VoE channel from the renderer.
+  // This method is called when the VoE channel is going away.
+  // TODO(xians): Remove this interface after Chrome switches to the
+  // AudioRenderer::Sink interface.
+  virtual void RemoveChannel(int channel_id) {}
+
  protected:
   virtual ~AudioRenderer() {}
 };
diff --git a/talk/media/base/fakemediaengine.h b/talk/media/base/fakemediaengine.h
index e5c4c53241..7325667aa5 100644
--- a/talk/media/base/fakemediaengine.h
+++ b/talk/media/base/fakemediaengine.h
@@ -308,9 +308,11 @@ class FakeVoiceMediaChannel : public RtpHelper<VoiceMediaChannel> {
         ASSERT(it->second == renderer);
       } else {
         remote_renderers_.insert(std::make_pair(ssrc, renderer));
+        renderer->AddChannel(0);
       }
     } else {
       if (it != remote_renderers_.end()) {
+        it->second->RemoveChannel(0);
         remote_renderers_.erase(it);
       } else {
         return false;
@@ -380,10 +382,12 @@ class FakeVoiceMediaChannel : public RtpHelper<VoiceMediaChannel> {
    public:
     explicit VoiceChannelAudioSink(AudioRenderer* renderer)
         : renderer_(renderer) {
+      renderer_->AddChannel(0);
       renderer_->SetSink(this);
     }
     virtual ~VoiceChannelAudioSink() {
       if (renderer_) {
+        renderer_->RemoveChannel(0);
         renderer_->SetSink(NULL);
       }
     }
diff --git a/talk/media/webrtc/webrtcvoiceengine.cc b/talk/media/webrtc/webrtcvoiceengine.cc
index db22b41fc0..54fac221d8 100644
--- a/talk/media/webrtc/webrtcvoiceengine.cc
+++ b/talk/media/webrtc/webrtcvoiceengine.cc
@@ -1305,6 +1305,10 @@ class WebRtcVoiceMediaChannel::WebRtcVoiceChannelRenderer
       RTC_DCHECK(renderer_ == renderer);
       return;
     }
+
+    // TODO(xians): Remove AddChannel() call after Chrome turns on APM
+    // in getUserMedia by default.
+    renderer->AddChannel(channel_);
     renderer->SetSink(this);
     renderer_ = renderer;
   }
@@ -1314,10 +1318,12 @@ class WebRtcVoiceMediaChannel::WebRtcVoiceChannelRenderer
   // This method is called on the libjingle worker thread.
   void Stop() {
     rtc::CritScope lock(&lock_);
-    if (renderer_ != NULL) {
-      renderer_->SetSink(NULL);
-      renderer_ = NULL;
-    }
+    if (renderer_ == NULL)
+      return;
+
+    renderer_->RemoveChannel(channel_);
+    renderer_->SetSink(NULL);
+    renderer_ = NULL;
   }
 
   // AudioRenderer::Sink implementation.