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

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/renderer/media/rtc_media_constraints.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_source_provider.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/libjingle/source/talk/app/webrtc/mediastreaminterface.h"
 
 using ::testing::_;
@@ skipping 92 matching lines @@
   }
  protected:
   virtual ~MockCapturerSource() {}
 
  private:
   scoped_ptr<FakeAudioThread> audio_thread_;
   WebRtcAudioCapturer* capturer_;
   media::AudioParameters params_;
 };
 
-class MockWebRtcAudioCapturerSink : public WebRtcAudioCapturerSink {
+// TODO(xians): Use MediaStreamAudioSink.
+class MockMediaStreamAudioSink : public PeerConnectionAudioSink {
  public:
-  MockWebRtcAudioCapturerSink() {}
-  ~MockWebRtcAudioCapturerSink() {}
-  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 {
+  MockMediaStreamAudioSink() {}
+  ~MockMediaStreamAudioSink() {}
+  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 {
     CaptureData(channels.size(),
                 sample_rate,
                 number_of_channels,
                 number_of_frames,
                 audio_delay_milliseconds,
                 current_volume,
                 need_audio_processing,
                 key_pressed);
     return 0;
   }
   MOCK_METHOD8(CaptureData,
                void(int number_of_network_channels,
                     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));
-  MOCK_METHOD1(SetCaptureFormat, void(const media::AudioParameters& params));
+  MOCK_METHOD1(OnSetFormat, void(const media::AudioParameters& params));
 };
 
 }  // namespace
 
 class WebRtcLocalAudioTrackTest : public ::testing::Test {
  protected:
   virtual void SetUp() OVERRIDE {
     params_.Reset(media::AudioParameters::AUDIO_PCM_LOW_LATENCY,
                   media::CHANNEL_LAYOUT_STEREO, 2, 0, 48000, 16, 480);
     capturer_ = WebRtcAudioCapturer::CreateCapturer();
@@ skipping 25 matching lines @@
       track->audio_source_provider())->SetSinkParamsForTesting(params_);
   track->Start();
   EXPECT_TRUE(track->enabled());
 
   // Connect a number of network channels to the audio track.
   static const int kNumberOfNetworkChannels = 4;
   for (int i = 0; i < kNumberOfNetworkChannels; ++i) {
     static_cast<webrtc::AudioTrackInterface*>(track.get())->
         GetRenderer()->AddChannel(i);
   }
-  scoped_ptr<MockWebRtcAudioCapturerSink> sink(
-      new MockWebRtcAudioCapturerSink());
+  scoped_ptr<MockMediaStreamAudioSink> sink(new MockMediaStreamAudioSink());
   const media::AudioParameters params = capturer_->audio_parameters();
   base::WaitableEvent event(false, false);
-  EXPECT_CALL(*sink, SetCaptureFormat(_)).WillOnce(Return());
+  EXPECT_CALL(*sink, OnSetFormat(_)).WillOnce(Return());
   EXPECT_CALL(*sink,
       CaptureData(kNumberOfNetworkChannels,
                   params.sample_rate(),
                   params.channels(),
                   params.sample_rate() / 100,
                   0,
                   0,
                   false,
                   false)).Times(AtLeast(1))
       .WillRepeatedly(SignalEvent(&event));
@@ skipping 19 matching lines @@
   scoped_refptr<WebRtcLocalAudioTrack> track =
     WebRtcLocalAudioTrack::Create(std::string(), capturer_, NULL, NULL,
         &constraints);
   static_cast<WebRtcLocalAudioSourceProvider*>(
       track->audio_source_provider())->SetSinkParamsForTesting(params_);
   track->Start();
   static_cast<webrtc::AudioTrackInterface*>(track.get())->
       GetRenderer()->AddChannel(0);
   EXPECT_TRUE(track->enabled());
   EXPECT_TRUE(track->set_enabled(false));
-  scoped_ptr<MockWebRtcAudioCapturerSink> sink(
-      new MockWebRtcAudioCapturerSink());
+  scoped_ptr<MockMediaStreamAudioSink> sink(new MockMediaStreamAudioSink());
   const media::AudioParameters params = capturer_->audio_parameters();
   base::WaitableEvent event(false, false);
-  EXPECT_CALL(*sink, SetCaptureFormat(_)).Times(1);
+  EXPECT_CALL(*sink, OnSetFormat(_)).Times(1);
   EXPECT_CALL(*sink,
               CaptureData(1,
                           params.sample_rate(),
                           params.channels(),
                           params.sample_rate() / 100,
                           0,
                           0,
                           false,
                           false)).Times(0);
   track->AddSink(sink.get());
@@ skipping 28 matching lines @@
   RTCMediaConstraints constraints;
   scoped_refptr<WebRtcLocalAudioTrack> track_1 =
     WebRtcLocalAudioTrack::Create(std::string(), capturer_, NULL, NULL,
         &constraints);
   static_cast<WebRtcLocalAudioSourceProvider*>(
       track_1->audio_source_provider())->SetSinkParamsForTesting(params_);
   track_1->Start();
   static_cast<webrtc::AudioTrackInterface*>(track_1.get())->
       GetRenderer()->AddChannel(0);
   EXPECT_TRUE(track_1->enabled());
-  scoped_ptr<MockWebRtcAudioCapturerSink> sink_1(
-      new MockWebRtcAudioCapturerSink());
+  scoped_ptr<MockMediaStreamAudioSink> sink_1(new MockMediaStreamAudioSink());
   const media::AudioParameters params = capturer_->audio_parameters();
   base::WaitableEvent event_1(false, false);
-  EXPECT_CALL(*sink_1, SetCaptureFormat(_)).WillOnce(Return());
+  EXPECT_CALL(*sink_1, OnSetFormat(_)).WillOnce(Return());
   EXPECT_CALL(*sink_1,
       CaptureData(1,
                   params.sample_rate(),
                   params.channels(),
                   params.sample_rate() / 100,
                   0,
                   0,
                   false,
                   false)).Times(AtLeast(1))
       .WillRepeatedly(SignalEvent(&event_1));
   track_1->AddSink(sink_1.get());
   EXPECT_TRUE(event_1.TimedWait(TestTimeouts::tiny_timeout()));
 
   scoped_refptr<WebRtcLocalAudioTrack> track_2 =
     WebRtcLocalAudioTrack::Create(std::string(), capturer_, NULL, NULL,
         &constraints);
   static_cast<WebRtcLocalAudioSourceProvider*>(
       track_2->audio_source_provider())->SetSinkParamsForTesting(params_);
   track_2->Start();
   static_cast<webrtc::AudioTrackInterface*>(track_2.get())->
       GetRenderer()->AddChannel(1);
   EXPECT_TRUE(track_2->enabled());
 
   // Verify both |sink_1| and |sink_2| get data.
   event_1.Reset();
   base::WaitableEvent event_2(false, false);
 
-  scoped_ptr<MockWebRtcAudioCapturerSink> sink_2(
-        new MockWebRtcAudioCapturerSink());
-  EXPECT_CALL(*sink_2, SetCaptureFormat(_)).WillOnce(Return());
+  scoped_ptr<MockMediaStreamAudioSink> sink_2(new MockMediaStreamAudioSink());
+  EXPECT_CALL(*sink_2, OnSetFormat(_)).WillOnce(Return());
   EXPECT_CALL(*sink_1,
       CaptureData(1,
                   params.sample_rate(),
                   params.channels(),
                   params.sample_rate() / 100,
                   0,
                   0,
                   false,
                   false)).Times(AtLeast(1))
       .WillRepeatedly(SignalEvent(&event_1));
@@ skipping 56 matching lines @@
       WebRtcLocalAudioTrack::Create(std::string(), capturer_, NULL, NULL,
           &constraints);
   static_cast<webrtc::AudioTrackInterface*>(track_1.get())->
       GetRenderer()->AddChannel(0);
   static_cast<WebRtcLocalAudioSourceProvider*>(
       track_1->audio_source_provider())->SetSinkParamsForTesting(params_);
   track_1->Start();
   EXPECT_TRUE(event.TimedWait(TestTimeouts::tiny_timeout()));
 
   // Verify the data flow by connecting the sink to |track_1|.
-  scoped_ptr<MockWebRtcAudioCapturerSink> sink(
-      new MockWebRtcAudioCapturerSink());
+  scoped_ptr<MockMediaStreamAudioSink> sink(new MockMediaStreamAudioSink());
   event.Reset();
-  EXPECT_CALL(*sink, SetCaptureFormat(_)).WillOnce(SignalEvent(&event));
+  EXPECT_CALL(*sink, OnSetFormat(_)).WillOnce(SignalEvent(&event));
   EXPECT_CALL(*sink, CaptureData(_, _, _, _, 0, 0, false, false))
       .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<WebRtcLocalAudioTrack> track_2 =
       WebRtcLocalAudioTrack::Create(std::string(), capturer_, NULL, NULL,
           &constraints);
   static_cast<WebRtcLocalAudioSourceProvider*>(
       track_2->audio_source_provider())->SetSinkParamsForTesting(params_);
   track_2->Start();
   static_cast<webrtc::AudioTrackInterface*>(track_2.get())->
       GetRenderer()->AddChannel(1);
 
   // Stop the capturer will clear up the track lists in the capturer.
   EXPECT_CALL(*capturer_source_.get(), OnStop());
   capturer_->Stop();
 
   // Adding a new track to the capturer.
   track_2->AddSink(sink.get());
-  EXPECT_CALL(*sink, SetCaptureFormat(_)).Times(0);
+  EXPECT_CALL(*sink, OnSetFormat(_)).Times(0);
 
   // Stop the capturer again will not trigger stopping the source of the
   // capturer again..
   event.Reset();
   EXPECT_CALL(*capturer_source_.get(), OnStop()).Times(0);
   capturer_->Stop();
 }
 
 // Set new source to the existing capturer.
 TEST_F(WebRtcLocalAudioTrackTest, SetNewSourceForCapturerAfterStartTrack) {
@@ skipping 38 matching lines @@
       track_1->audio_source_provider())->SetSinkParamsForTesting(params_);
   track_1->Start();
 
   // Connect a number of network channels to the |track_1|.
   static const int kNumberOfNetworkChannelsForTrack1 = 2;
   for (int i = 0; i < kNumberOfNetworkChannelsForTrack1; ++i) {
     static_cast<webrtc::AudioTrackInterface*>(track_1.get())->
         GetRenderer()->AddChannel(i);
   }
   // Verify the data flow by connecting the |sink_1| to |track_1|.
-  scoped_ptr<MockWebRtcAudioCapturerSink> sink_1(
-      new MockWebRtcAudioCapturerSink());
+  scoped_ptr<MockMediaStreamAudioSink> sink_1(new MockMediaStreamAudioSink());
   EXPECT_CALL(
       *sink_1.get(),
       CaptureData(
           kNumberOfNetworkChannelsForTrack1, 48000, 2, _, 0, 0, false, false))
       .Times(AnyNumber()).WillRepeatedly(Return());
-  EXPECT_CALL(*sink_1.get(), SetCaptureFormat(_)).Times(AnyNumber());
+  EXPECT_CALL(*sink_1.get(), OnSetFormat(_)).Times(AnyNumber());
   track_1->AddSink(sink_1.get());
 
   // Create a new capturer with new source with different audio format.
   scoped_refptr<WebRtcAudioCapturer> new_capturer(
       WebRtcAudioCapturer::CreateCapturer());
   scoped_refptr<MockCapturerSource> new_source(
       new MockCapturerSource(new_capturer.get()));
   EXPECT_CALL(*new_source.get(), OnInitialize(_, new_capturer.get(), 0));
   new_capturer->SetCapturerSource(new_source,
                                   media::CHANNEL_LAYOUT_MONO,
                                   44100);
 
   // Setup the second audio track, connect it to the new capturer and start it.
   EXPECT_CALL(*new_source.get(), SetAutomaticGainControl(true));
   EXPECT_CALL(*new_source.get(), OnStart());
   scoped_refptr<WebRtcLocalAudioTrack> track_2 =
       WebRtcLocalAudioTrack::Create(std::string(), new_capturer, NULL, NULL,
           &constraints);
   static_cast<WebRtcLocalAudioSourceProvider*>(
       track_2->audio_source_provider())->SetSinkParamsForTesting(params_);
   track_2->Start();
 
   // Connect a number of network channels to the |track_2|.
   static const int kNumberOfNetworkChannelsForTrack2 = 3;
   for (int i = 0; i < kNumberOfNetworkChannelsForTrack2; ++i) {
     static_cast<webrtc::AudioTrackInterface*>(track_2.get())->
         GetRenderer()->AddChannel(i);
   }
   // Verify the data flow by connecting the |sink_2| to |track_2|.
-  scoped_ptr<MockWebRtcAudioCapturerSink> sink_2(
-      new MockWebRtcAudioCapturerSink());
+  scoped_ptr<MockMediaStreamAudioSink> sink_2(new MockMediaStreamAudioSink());
   base::WaitableEvent event(false, false);
   EXPECT_CALL(
       *sink_2,
       CaptureData(
           kNumberOfNetworkChannelsForTrack2, 44100, 1, _, 0, 0, false, false))
       .Times(AnyNumber()).WillRepeatedly(Return());
-  EXPECT_CALL(*sink_2, SetCaptureFormat(_)).WillOnce(SignalEvent(&event));
+  EXPECT_CALL(*sink_2, OnSetFormat(_)).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();
   EXPECT_TRUE(event.TimedWait(TestTimeouts::tiny_timeout()));
 
@@ skipping 27 matching lines @@
   EXPECT_CALL(*source.get(), OnStart());
   RTCMediaConstraints constraints;
   scoped_refptr<WebRtcLocalAudioTrack> track =
       WebRtcLocalAudioTrack::Create(std::string(), capturer, NULL, NULL,
                                     &constraints);
   static_cast<WebRtcLocalAudioSourceProvider*>(
       track->audio_source_provider())->SetSinkParamsForTesting(params);
   track->Start();
 
   // Verify the data flow by connecting the |sink| to |track|.
-  scoped_ptr<MockWebRtcAudioCapturerSink> sink(
-      new MockWebRtcAudioCapturerSink());
+  scoped_ptr<MockMediaStreamAudioSink> sink(new MockMediaStreamAudioSink());
   base::WaitableEvent event(false, false);
-  EXPECT_CALL(*sink, SetCaptureFormat(_)).Times(1);
+  EXPECT_CALL(*sink, OnSetFormat(_)).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, params.sample_rate(), params.channels(), expected_buffer_size,
       0, 0, false, false))
       .Times(AtLeast(1)).WillRepeatedly(SignalEvent(&event));
   track->AddSink(sink.get());
   EXPECT_TRUE(event.TimedWait(TestTimeouts::tiny_timeout()));
 
   // Stopping the new source will stop the second track.
   EXPECT_CALL(*source, OnStop()).Times(1);
   capturer->Stop();
 }
 
 }  // namespace content