Clean up the media stream audio track code

content/renderer/media/webrtc_local_audio_track_unittest.cc

 // Copyright 2013 The Chromium Authors. All rights reserved.
 // Use of this source code is governed by a BSD-style license that can be
 // found in the LICENSE file.
 
 #include "base/synchronization/waitable_event.h"
 #include "base/test/test_timeouts.h"
+#include "content/public/renderer/media_stream_audio_sink.h"
 #include "content/renderer/media/media_stream_audio_source.h"
 #include "content/renderer/media/mock_media_constraint_factory.h"
 #include "content/renderer/media/webrtc/webrtc_local_audio_track_adapter.h"
 #include "content/renderer/media/webrtc_audio_capturer.h"
-#include "content/renderer/media/webrtc_audio_device_impl.h"
 #include "content/renderer/media/webrtc_local_audio_track.h"
 #include "media/audio/audio_parameters.h"
 #include "media/base/audio_bus.h"
 #include "media/base/audio_capturer_source.h"
 #include "testing/gmock/include/gmock/gmock.h"
 #include "testing/gtest/include/gtest/gtest.h"
 #include "third_party/WebKit/public/platform/WebMediaConstraints.h"
 #include "third_party/WebKit/public/web/WebHeap.h"
 #include "third_party/libjingle/source/talk/app/webrtc/mediastreaminterface.h"
 
@@ skipping 93 matching lines @@
   }
  protected:
   virtual ~MockCapturerSource() {}
 
  private:
   scoped_ptr<FakeAudioThread> audio_thread_;
   WebRtcAudioCapturer* capturer_;
   media::AudioParameters params_;
 };
 
-// TODO(xians): Use MediaStreamAudioSink.
-class MockMediaStreamAudioSink : public PeerConnectionAudioSink {
+class MockMediaStreamAudioSink : public MediaStreamAudioSink {
  public:
   MockMediaStreamAudioSink() {}
   ~MockMediaStreamAudioSink() {}
-  int OnData(const int16* audio_data,
+  void 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 {
+             int number_of_frames) override {
     EXPECT_EQ(params_.sample_rate(), sample_rate);
     EXPECT_EQ(params_.channels(), number_of_channels);
     EXPECT_EQ(params_.frames_per_buffer(), number_of_frames);
-    CaptureData(channels.size(),
-                audio_delay_milliseconds,
-                current_volume,
-                need_audio_processing,
-                key_pressed);
-    return 0;
+    CaptureData();
   }
-  MOCK_METHOD5(CaptureData,
-               void(int number_of_network_channels,
-                    int audio_delay_milliseconds,
-                    int current_volume,
-                    bool need_audio_processing,
-                    bool key_pressed));
+  MOCK_METHOD0(CaptureData, void());
   void OnSetFormat(const media::AudioParameters& params) {
     params_ = params;
     FormatIsSet();
   }
   MOCK_METHOD0(FormatIsSet, void());
 
   const media::AudioParameters& audio_params() const { return params_; }
 
  private:
   media::AudioParameters params_;
@@ skipping 46 matching lines @@
       WebRtcLocalAudioTrackAdapter::Create(std::string(), NULL));
   scoped_ptr<WebRtcLocalAudioTrack> track(
       new WebRtcLocalAudioTrack(adapter.get(), capturer_, NULL));
   track->Start();
   EXPECT_TRUE(track->GetAudioAdapter()->enabled());
 
   scoped_ptr<MockMediaStreamAudioSink> sink(new MockMediaStreamAudioSink());
   base::WaitableEvent event(false, false);
   EXPECT_CALL(*sink, FormatIsSet());
   EXPECT_CALL(*sink,
-      CaptureData(0,
-                  0,
-                  0,
-                  _,
-                  false)).Times(AtLeast(1))
+      CaptureData()).Times(AtLeast(1))
       .WillRepeatedly(SignalEvent(&event));
   track->AddSink(sink.get());
   EXPECT_TRUE(event.TimedWait(TestTimeouts::tiny_timeout()));
   track->RemoveSink(sink.get());
 
   EXPECT_CALL(*capturer_source_.get(), OnStop()).WillOnce(Return());
   capturer_->Stop();
 }
 
 // The same setup as ConnectAndDisconnectOneSink, but enable and disable the
 // audio track on the fly. When the audio track is disabled, there is no data
 // callback to the sink; when the audio track is enabled, there comes data
 // callback.
 // TODO(xians): Enable this test after resolving the racing issue that TSAN
 // reports on MediaStreamTrack::enabled();
 TEST_F(WebRtcLocalAudioTrackTest,  DISABLED_DisableEnableAudioTrack) {
   EXPECT_CALL(*capturer_source_.get(), SetAutomaticGainControl(true));
   EXPECT_CALL(*capturer_source_.get(), OnStart());
   scoped_refptr<WebRtcLocalAudioTrackAdapter> adapter(
       WebRtcLocalAudioTrackAdapter::Create(std::string(), NULL));
   scoped_ptr<WebRtcLocalAudioTrack> track(
       new WebRtcLocalAudioTrack(adapter.get(), capturer_, NULL));
   track->Start();
   EXPECT_TRUE(track->GetAudioAdapter()->enabled());
   EXPECT_TRUE(track->GetAudioAdapter()->set_enabled(false));
   scoped_ptr<MockMediaStreamAudioSink> sink(new MockMediaStreamAudioSink());
   const media::AudioParameters params = capturer_->source_audio_parameters();
   base::WaitableEvent event(false, false);
   EXPECT_CALL(*sink, FormatIsSet()).Times(1);
-  EXPECT_CALL(*sink,
-              CaptureData(0, 0, 0, _, false)).Times(0);
+  EXPECT_CALL(*sink, CaptureData()).Times(0);
   EXPECT_EQ(sink->audio_params().frames_per_buffer(),
             params.sample_rate() / 100);
   track->AddSink(sink.get());
   EXPECT_FALSE(event.TimedWait(TestTimeouts::tiny_timeout()));
 
   event.Reset();
-  EXPECT_CALL(*sink, CaptureData(0, 0, 0, _, false)).Times(AtLeast(1))
+  EXPECT_CALL(*sink, CaptureData()).Times(AtLeast(1))
       .WillRepeatedly(SignalEvent(&event));
   EXPECT_TRUE(track->GetAudioAdapter()->set_enabled(true));
   EXPECT_TRUE(event.TimedWait(TestTimeouts::tiny_timeout()));
   track->RemoveSink(sink.get());
 
   EXPECT_CALL(*capturer_source_.get(), OnStop()).WillOnce(Return());
   capturer_->Stop();
   track.reset();
 }
 
 // Create multiple audio tracks and enable/disable them, verify that the audio
 // callbacks appear/disappear.
 // Flaky due to a data race, see http://crbug.com/295418
 TEST_F(WebRtcLocalAudioTrackTest, DISABLED_MultipleAudioTracks) {
   scoped_refptr<WebRtcLocalAudioTrackAdapter> adapter_1(
       WebRtcLocalAudioTrackAdapter::Create(std::string(), NULL));
   scoped_ptr<WebRtcLocalAudioTrack> track_1(
       new WebRtcLocalAudioTrack(adapter_1.get(), capturer_, NULL));
   track_1->Start();
   EXPECT_TRUE(track_1->GetAudioAdapter()->enabled());
   scoped_ptr<MockMediaStreamAudioSink> sink_1(new MockMediaStreamAudioSink());
   const media::AudioParameters params = capturer_->source_audio_parameters();
   base::WaitableEvent event_1(false, false);
   EXPECT_CALL(*sink_1, FormatIsSet()).WillOnce(Return());
-  EXPECT_CALL(*sink_1,
-      CaptureData(0, 0, 0, _, false)).Times(AtLeast(1))
+  EXPECT_CALL(*sink_1, CaptureData()).Times(AtLeast(1))
       .WillRepeatedly(SignalEvent(&event_1));
   EXPECT_EQ(sink_1->audio_params().frames_per_buffer(),
             params.sample_rate() / 100);
   track_1->AddSink(sink_1.get());
   EXPECT_TRUE(event_1.TimedWait(TestTimeouts::tiny_timeout()));
 
   scoped_refptr<WebRtcLocalAudioTrackAdapter> adapter_2(
       WebRtcLocalAudioTrackAdapter::Create(std::string(), NULL));
   scoped_ptr<WebRtcLocalAudioTrack> track_2(
       new WebRtcLocalAudioTrack(adapter_2.get(), capturer_, NULL));
   track_2->Start();
   EXPECT_TRUE(track_2->GetAudioAdapter()->enabled());
 
   // Verify both |sink_1| and |sink_2| get data.
   event_1.Reset();
   base::WaitableEvent event_2(false, false);
 
   scoped_ptr<MockMediaStreamAudioSink> sink_2(new MockMediaStreamAudioSink());
   EXPECT_CALL(*sink_2, FormatIsSet()).WillOnce(Return());
-  EXPECT_CALL(*sink_1, CaptureData(0, 0, 0, _, false)).Times(AtLeast(1))
+  EXPECT_CALL(*sink_1, CaptureData()).Times(AtLeast(1))
       .WillRepeatedly(SignalEvent(&event_1));
   EXPECT_EQ(sink_1->audio_params().frames_per_buffer(),
             params.sample_rate() / 100);
-  EXPECT_CALL(*sink_2, CaptureData(0, 0, 0, _, false)).Times(AtLeast(1))
+  EXPECT_CALL(*sink_2, CaptureData()).Times(AtLeast(1))
       .WillRepeatedly(SignalEvent(&event_2));
   EXPECT_EQ(sink_2->audio_params().frames_per_buffer(),
             params.sample_rate() / 100);
   track_2->AddSink(sink_2.get());
   EXPECT_TRUE(event_1.TimedWait(TestTimeouts::tiny_timeout()));
   EXPECT_TRUE(event_2.TimedWait(TestTimeouts::tiny_timeout()));
 
   track_1->RemoveSink(sink_1.get());
   track_1->Stop();
   track_1.reset();
@@ skipping 50 matching lines @@
   scoped_refptr<WebRtcLocalAudioTrackAdapter> adapter_1(
       WebRtcLocalAudioTrackAdapter::Create(std::string(), NULL));
   scoped_ptr<WebRtcLocalAudioTrack> track_1(
       new WebRtcLocalAudioTrack(adapter_1.get(), capturer_, NULL));
   track_1->Start();
 
   // Verify the data flow by connecting the sink to |track_1|.
   scoped_ptr<MockMediaStreamAudioSink> sink(new MockMediaStreamAudioSink());
   event.Reset();
   EXPECT_CALL(*sink, FormatIsSet()).WillOnce(SignalEvent(&event));
-  EXPECT_CALL(*sink, CaptureData(_, 0, 0, _, false))
+  EXPECT_CALL(*sink, CaptureData())
       .Times(AnyNumber()).WillRepeatedly(Return());
   track_1->AddSink(sink.get());
   EXPECT_TRUE(event.TimedWait(TestTimeouts::tiny_timeout()));
 
   // Start the second audio track will not start the |capturer_source_|
   // since it has been started.
   EXPECT_CALL(*capturer_source_.get(), OnStart()).Times(0);
   scoped_refptr<WebRtcLocalAudioTrackAdapter> adapter_2(
       WebRtcLocalAudioTrackAdapter::Create(std::string(), NULL));
   scoped_ptr<WebRtcLocalAudioTrack> track_2(
@@ skipping 27 matching lines @@
        DISABLE_ON_TSAN(ConnectTracksToDifferentCapturers)) {
   // Setup the first audio track and start it.
   scoped_refptr<WebRtcLocalAudioTrackAdapter> adapter_1(
       WebRtcLocalAudioTrackAdapter::Create(std::string(), NULL));
   scoped_ptr<WebRtcLocalAudioTrack> track_1(
       new WebRtcLocalAudioTrack(adapter_1.get(), capturer_, NULL));
   track_1->Start();
 
   // Verify the data flow by connecting the |sink_1| to |track_1|.
   scoped_ptr<MockMediaStreamAudioSink> sink_1(new MockMediaStreamAudioSink());
-  EXPECT_CALL(*sink_1.get(), CaptureData(0, 0, 0, _, false))
+  EXPECT_CALL(*sink_1.get(), CaptureData())
       .Times(AnyNumber()).WillRepeatedly(Return());
   EXPECT_CALL(*sink_1.get(), FormatIsSet()).Times(AnyNumber());
   track_1->AddSink(sink_1.get());
 
   // Create a new capturer with new source with different audio format.
   MockMediaConstraintFactory constraint_factory;
   StreamDeviceInfo device(MEDIA_DEVICE_AUDIO_CAPTURE,
                           std::string(), std::string());
   scoped_refptr<WebRtcAudioCapturer> new_capturer(
       WebRtcAudioCapturer::CreateCapturer(
@@ skipping 13 matching lines @@
   // Setup the second audio track, connect it to the new capturer and start it.
   scoped_refptr<WebRtcLocalAudioTrackAdapter> adapter_2(
       WebRtcLocalAudioTrackAdapter::Create(std::string(), NULL));
   scoped_ptr<WebRtcLocalAudioTrack> track_2(
       new WebRtcLocalAudioTrack(adapter_2.get(), new_capturer, NULL));
   track_2->Start();
 
   // Verify the data flow by connecting the |sink_2| to |track_2|.
   scoped_ptr<MockMediaStreamAudioSink> sink_2(new MockMediaStreamAudioSink());
   base::WaitableEvent event(false, false);
-  EXPECT_CALL(*sink_2, CaptureData(0, 0, 0, _, false))
+  EXPECT_CALL(*sink_2, CaptureData())
       .Times(AnyNumber()).WillRepeatedly(Return());
   EXPECT_CALL(*sink_2, FormatIsSet()).WillOnce(SignalEvent(&event));
   track_2->AddSink(sink_2.get());
   EXPECT_TRUE(event.TimedWait(TestTimeouts::tiny_timeout()));
 
   // Stopping the new source will stop the second track.
   event.Reset();
   EXPECT_CALL(*new_source.get(), OnStop())
       .Times(1).WillOnce(SignalEvent(&event));
   new_capturer->Stop();
@@ skipping 40 matching lines @@
   // Verify the data flow by connecting the |sink| to |track|.
   scoped_ptr<MockMediaStreamAudioSink> sink(new MockMediaStreamAudioSink());
   base::WaitableEvent event(false, false);
   EXPECT_CALL(*sink, FormatIsSet()).Times(1);
   // Verify the sinks are getting the packets with an expecting buffer size.
 #if defined(OS_ANDROID)
   const int expected_buffer_size = params.sample_rate() / 100;
 #else
   const int expected_buffer_size = params.frames_per_buffer();
 #endif
-  EXPECT_CALL(*sink, CaptureData(
-      0, 0, 0, _, false))
+  EXPECT_CALL(*sink, CaptureData())
       .Times(AtLeast(1)).WillRepeatedly(SignalEvent(&event));
   track->AddSink(sink.get());
   EXPECT_TRUE(event.TimedWait(TestTimeouts::tiny_timeout()));
   EXPECT_EQ(expected_buffer_size, sink->audio_params().frames_per_buffer());
 
   // Stopping the new source will stop the second track.
   EXPECT_CALL(*source.get(), OnStop()).Times(1);
   capturer->Stop();
 
   // Even though this test don't use |capturer_source_| it will be stopped
   // during teardown of the test harness.
   EXPECT_CALL(*capturer_source_.get(), OnStop());
 }
 
 }  // namespace content