Add generic interfaces for the sinks of the media stream audio track

content/renderer/media/webrtc_audio_device_unittest.cc

Index: content/renderer/media/webrtc_audio_device_unittest.cc
diff --git a/content/renderer/media/webrtc_audio_device_unittest.cc b/content/renderer/media/webrtc_audio_device_unittest.cc
index 6dcdda7033405830876fb72be6834f0b4d24adfa..a79176ec19d3d0d5acf3412d281d92f3b77abbdc 100644
--- a/content/renderer/media/webrtc_audio_device_unittest.cc
+++ b/content/renderer/media/webrtc_audio_device_unittest.cc
@@ -135,7 +135,7 @@ bool CreateAndInitializeCapturer(WebRtcAudioDeviceImpl* webrtc_audio_device) {
 // Also, connect the sink to the audio track.
 scoped_refptr<WebRtcLocalAudioTrack>
 CreateAndStartLocalAudioTrack(WebRtcAudioCapturer* capturer,
-                              WebRtcAudioCapturerSink* sink) {
+                              PeerConnectionAudioSink* sink) {
   scoped_refptr<WebRtcLocalAudioTrack> local_audio_track(
       WebRtcLocalAudioTrack::Create(std::string(), capturer, NULL, NULL, NULL));
   local_audio_track->AddSink(sink);
@@ -205,37 +205,38 @@ class WebRTCMediaProcessImpl : public webrtc::VoEMediaProcess {
   DISALLOW_COPY_AND_ASSIGN(WebRTCMediaProcessImpl);
 };
 
-class MockWebRtcAudioCapturerSink : public WebRtcAudioCapturerSink {
+// TODO(xians): Use MediaStreamAudioSink.
+class MockMediaStreamAudioSink : public PeerConnectionAudioSink {
  public:
-  explicit MockWebRtcAudioCapturerSink(base::WaitableEvent* event)
+  explicit MockMediaStreamAudioSink(base::WaitableEvent* event)
       : event_(event) {
     DCHECK(event_);
   }
-  virtual ~MockWebRtcAudioCapturerSink() {}
-
-  // WebRtcAudioCapturerSink implementation.
-  virtual int CaptureData(const std::vector<int>& channels,
-                          const int16* audio_data,
-                          int sample_rate,
-                          int number_of_channels,
-                          int number_of_frames,
-                          int audio_delay_milliseconds,
-                          int current_volume,
-                          bool need_audio_processing,
-                          bool key_pressed) OVERRIDE {
+  virtual ~MockMediaStreamAudioSink() {}
+
+  // PeerConnectionAudioSink implementation.
+  virtual int OnData(const int16* audio_data,
+                     int sample_rate,
+                     int number_of_channels,
+                     int number_of_frames,
+                     const std::vector<int>& channels,
+                     int audio_delay_milliseconds,
+                     int current_volume,
+                     bool need_audio_processing,
+                     bool key_pressed) OVERRIDE {
     // Signal that a callback has been received.
     event_->Signal();
     return 0;
   }
 
   // Set the format for the capture audio parameters.
-  virtual void SetCaptureFormat(
+  virtual void OnSetFormat(
       const media::AudioParameters& params) OVERRIDE {}
 
  private:
    base::WaitableEvent* event_;
 
-   DISALLOW_COPY_AND_ASSIGN(MockWebRtcAudioCapturerSink);
+   DISALLOW_COPY_AND_ASSIGN(MockMediaStreamAudioSink);
 };
 
 class MockWebRtcAudioRendererSource : public WebRtcAudioRendererSource {
@@ -329,13 +330,13 @@ int RunWebRtcLoopbackTimeTest(media::AudioManager* manager,
   int err = base->Init(webrtc_audio_device.get());
   EXPECT_EQ(0, err);
 
-  // We use SetCaptureFormat() and SetRenderFormat() to configure the audio
+  // We use OnSetFormat() and SetRenderFormat() to configure the audio
   // parameters so that this test can run on machine without hardware device.
   const media::AudioParameters params = media::AudioParameters(
       media::AudioParameters::AUDIO_PCM_LOW_LATENCY, CHANNEL_LAYOUT_STEREO,
       48000, 2, 480);
-  WebRtcAudioCapturerSink* capturer_sink =
-      static_cast<WebRtcAudioCapturerSink*>(webrtc_audio_device.get());
+  PeerConnectionAudioSink* capturer_sink =
+      static_cast<PeerConnectionAudioSink*>(webrtc_audio_device.get());
   WebRtcAudioRendererSource* renderer_source =
       static_cast<WebRtcAudioRendererSource*>(webrtc_audio_device.get());
   renderer_source->SetRenderFormat(params);
@@ -379,12 +380,12 @@ int RunWebRtcLoopbackTimeTest(media::AudioManager* manager,
   voe_channels.push_back(channel);
   for (int j = 0; j < kNumberOfPacketsForLoopbackTest; ++j) {
     // Sending fake capture data to WebRtc.
-    capturer_sink->CaptureData(
-        voe_channels,
+    capturer_sink->OnData(
         reinterpret_cast<int16*>(capture_data.get() + input_packet_size * j),
         params.sample_rate(),
         params.channels(),
         params.frames_per_buffer(),
+        voe_channels,
         kHardwareLatencyInMs,
         1.0,
         enable_apm,
@@ -860,8 +861,8 @@ TEST_F(MAYBE_WebRTCAudioDeviceTest, DISABLED_WebRtcRecordingSetupTime) {
 
   EXPECT_TRUE(CreateAndInitializeCapturer(webrtc_audio_device.get()));
   base::WaitableEvent event(false, false);
-  scoped_ptr<MockWebRtcAudioCapturerSink> sink(
-      new MockWebRtcAudioCapturerSink(&event));
+  scoped_ptr<MockMediaStreamAudioSink> sink(
+      new MockMediaStreamAudioSink(&event));
 
   // Create and start a local audio track. Starting the audio track will connect
   // the audio track to the capturer and also start the source of the capturer.