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}
diff --git a/talk/app/webrtc/audiotrackrenderer.cc b/talk/app/webrtc/audiotrackrenderer.cc
new file mode 100644
index 0000000..264a3cb
--- /dev/null
+++ b/talk/app/webrtc/audiotrackrenderer.cc
@@ -0,0 +1,49 @@
+/*
+ * libjingle
+ * Copyright 2013 Google Inc.
+ *
+ * Redistribution and use in source and binary forms, with or without
+ * modification, are permitted provided that the following conditions are met:
+ *
+ *  1. Redistributions of source code must retain the above copyright notice,
+ *     this list of conditions and the following disclaimer.
+ *  2. Redistributions in binary form must reproduce the above copyright notice,
+ *     this list of conditions and the following disclaimer in the documentation
+ *     and/or other materials provided with the distribution.
+ *  3. The name of the author may not be used to endorse or promote products
+ *     derived from this software without specific prior written permission.
+ *
+ * THIS SOFTWARE IS PROVIDED BY THE AUTHOR ``AS IS'' AND ANY EXPRESS OR IMPLIED
+ * WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE IMPLIED WARRANTIES OF
+ * MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO
+ * EVENT SHALL THE AUTHOR BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL,
+ * SPECIAL, EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO,
+ * PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS;
+ * OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY,
+ * WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR
+ * OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF
+ * ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
+ */
+
+#include "talk/app/webrtc/audiotrackrenderer.h"
+#include "webrtc/base/common.h"
+
+namespace webrtc {
+
+AudioTrackRenderer::AudioTrackRenderer() : channel_id_(-1) {
+}
+
+AudioTrackRenderer::~AudioTrackRenderer() {
+}
+
+void AudioTrackRenderer::AddChannel(int channel_id) {
+  ASSERT(channel_id_ == -1 || channel_id_ == channel_id);
+  channel_id_ = channel_id;
+}
+
+void AudioTrackRenderer::RemoveChannel(int channel_id) {
+  ASSERT(channel_id_ == -1 || channel_id_ == channel_id);
+  channel_id_ = -1;
+}
+
+}  // namespace webrtc
diff --git a/talk/app/webrtc/audiotrackrenderer.h b/talk/app/webrtc/audiotrackrenderer.h
new file mode 100644
index 0000000..e22805f
--- /dev/null
+++ b/talk/app/webrtc/audiotrackrenderer.h
@@ -0,0 +1,59 @@
+/*
+ * libjingle
+ * Copyright 2013 Google Inc.
+ *
+ * Redistribution and use in source and binary forms, with or without
+ * modification, are permitted provided that the following conditions are met:
+ *
+ *  1. Redistributions of source code must retain the above copyright notice,
+ *     this list of conditions and the following disclaimer.
+ *  2. Redistributions in binary form must reproduce the above copyright notice,
+ *     this list of conditions and the following disclaimer in the documentation
+ *     and/or other materials provided with the distribution.
+ *  3. The name of the author may not be used to endorse or promote products
+ *     derived from this software without specific prior written permission.
+ *
+ * THIS SOFTWARE IS PROVIDED BY THE AUTHOR ``AS IS'' AND ANY EXPRESS OR IMPLIED
+ * WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE IMPLIED WARRANTIES OF
+ * MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO
+ * EVENT SHALL THE AUTHOR BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL,
+ * SPECIAL, EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO,
+ * PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS;
+ * OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY,
+ * WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR
+ * OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF
+ * ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
+ */
+
+#ifndef TALK_APP_WEBRTC_AUDIOTRACKRENDERER_H_
+#define TALK_APP_WEBRTC_AUDIOTRACKRENDERER_H_
+
+#include "talk/media/base/audiorenderer.h"
+#include "webrtc/base/thread.h"
+
+namespace webrtc {
+
+// Class used for AudioTrack to get the ID of WebRtc voice channel that
+// the AudioTrack is connecting to.
+// Each AudioTrack owns a AudioTrackRenderer instance.
+// AddChannel() will be called when an AudioTrack is added to a MediaStream.
+// RemoveChannel will be called when the AudioTrack or WebRtc VoE channel is
+// going away.
+// This implementation only supports one channel, and it is only used by
+// Chrome for remote audio tracks."
+class AudioTrackRenderer : public cricket::AudioRenderer {
+ public:
+  AudioTrackRenderer();
+  ~AudioTrackRenderer();
+
+  // Implements cricket::AudioRenderer.
+  void AddChannel(int channel_id) override;
+  void RemoveChannel(int channel_id) override;
+
+ private:
+  int channel_id_;
+};
+
+}  // namespace webrtc
+
+#endif  // TALK_APP_WEBRTC_AUDIOTRACKRENDERER_H_
diff --git a/talk/app/webrtc/webrtcsession_unittest.cc b/talk/app/webrtc/webrtcsession_unittest.cc
index ff41383..2853ca4 100644
--- a/talk/app/webrtc/webrtcsession_unittest.cc
+++ b/talk/app/webrtc/webrtcsession_unittest.cc
@@ -330,16 +330,26 @@
 
 class FakeAudioRenderer : public cricket::AudioRenderer {
  public:
-  FakeAudioRenderer() : sink_(NULL) {}
+  FakeAudioRenderer() : channel_id_(-1), sink_(NULL) {}
   virtual ~FakeAudioRenderer() {
     if (sink_)
       sink_->OnClose();
   }
 
+  void AddChannel(int channel_id) override {
+    ASSERT(channel_id_ == -1);
+    channel_id_ = channel_id;
+  }
+  void RemoveChannel(int channel_id) override {
+    ASSERT(channel_id == channel_id_);
+    channel_id_ = -1;
+  }
   void SetSink(Sink* sink) override { sink_ = sink; }
 
+  int channel_id() const { return channel_id_; }
   cricket::AudioRenderer::Sink* sink() const { return sink_; }
  private:
+  int channel_id_;
   cricket::AudioRenderer::Sink* sink_;
 };
 
@@ -3107,10 +3117,12 @@
   EXPECT_TRUE(channel->GetOutputScaling(receive_ssrc, &left_vol, &right_vol));
   EXPECT_EQ(0, left_vol);
   EXPECT_EQ(0, right_vol);
+  EXPECT_EQ(0, renderer->channel_id());
   session_->SetAudioPlayout(receive_ssrc, true, NULL);
   EXPECT_TRUE(channel->GetOutputScaling(receive_ssrc, &left_vol, &right_vol));
   EXPECT_EQ(1, left_vol);
   EXPECT_EQ(1, right_vol);
+  EXPECT_EQ(-1, renderer->channel_id());
 }
 
 TEST_F(WebRtcSessionTest, SetAudioSend) {
@@ -3130,6 +3142,7 @@
   session_->SetAudioSend(send_ssrc, false, options, renderer.get());
   EXPECT_TRUE(channel->IsStreamMuted(send_ssrc));
   EXPECT_FALSE(channel->options().echo_cancellation.IsSet());
+  EXPECT_EQ(0, renderer->channel_id());
   EXPECT_TRUE(renderer->sink() != NULL);
 
   // This will trigger SetSink(NULL) to the |renderer|.
@@ -3138,6 +3151,7 @@
   bool value;
   EXPECT_TRUE(channel->options().echo_cancellation.Get(&value));
   EXPECT_TRUE(value);
+  EXPECT_EQ(-1, renderer->channel_id());
   EXPECT_TRUE(renderer->sink() == NULL);
 }
 
diff --git a/talk/libjingle.gyp b/talk/libjingle.gyp
index 3268d01..fd2d969 100755
--- a/talk/libjingle.gyp
+++ b/talk/libjingle.gyp
@@ -712,6 +712,8 @@
       'sources': [
         'app/webrtc/audiotrack.cc',
         'app/webrtc/audiotrack.h',
+        'app/webrtc/audiotrackrenderer.cc',
+        'app/webrtc/audiotrackrenderer.h',
         'app/webrtc/datachannel.cc',
         'app/webrtc/datachannel.h',
         'app/webrtc/datachannelinterface.h',
diff --git a/talk/media/base/audiorenderer.h b/talk/media/base/audiorenderer.h
index 229c36e..5c03576 100644
--- a/talk/media/base/audiorenderer.h
+++ b/talk/media/base/audiorenderer.h
@@ -55,6 +55,20 @@
   // 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 e5c4c53..7325667 100644
--- a/talk/media/base/fakemediaengine.h
+++ b/talk/media/base/fakemediaengine.h
@@ -308,9 +308,11 @@
         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 @@
    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 db22b41..54fac22 100644
--- a/talk/media/webrtc/webrtcvoiceengine.cc
+++ b/talk/media/webrtc/webrtcvoiceengine.cc
@@ -1305,6 +1305,10 @@
       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 @@
   // 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.