Adding key press detection in the browser process.

content/renderer/media/webrtc_local_audio_track_unittest.cc

Index: content/renderer/media/webrtc_local_audio_track_unittest.cc
diff --git a/content/renderer/media/webrtc_local_audio_track_unittest.cc b/content/renderer/media/webrtc_local_audio_track_unittest.cc
index 1638feb86744a30bd493fb029ffdb27c42bf01ae..638d3747ac8ccd4dc60a411bd9bdde409fa4275b 100644
--- a/content/renderer/media/webrtc_local_audio_track_unittest.cc
+++ b/content/renderer/media/webrtc_local_audio_track_unittest.cc
@@ -49,7 +49,7 @@ class FakeAudioThread : public base::PlatformThread::Delegate {
           static_cast<media::AudioCapturerSource::CaptureCallback*>(
               capturer_.get());
       audio_bus_->Zero();
-      callback->Capture(audio_bus_.get(), 0, 0);
+      callback->Capture(audio_bus_.get(), 0, 0, false);
 
       // Sleep 1ms to yield the resource for the main thread.
       base::PlatformThread::Sleep(base::TimeDelta::FromMilliseconds(1));
@@ -95,11 +95,13 @@ class MockWebRtcAudioCapturerSink : public WebRtcAudioCapturerSink {
  public:
   MockWebRtcAudioCapturerSink() {}
   ~MockWebRtcAudioCapturerSink() {}
-  MOCK_METHOD5(CaptureData, void(const int16* audio_data,
-                                 int number_of_channels,
-                                 int number_of_frames,
-                                 int audio_delay_milliseconds,
-                                 double volume));
+  MOCK_METHOD6(CaptureData,
+               void(const int16* audio_data,
+                    int number_of_channels,
+                    int number_of_frames,
+                    int audio_delay_milliseconds,
+                    double volume,
+                    bool key_pressed));
   MOCK_METHOD1(SetCaptureFormat, void(const media::AudioParameters& params));
 };
 
@@ -147,8 +149,10 @@ TEST_F(WebRtcLocalAudioTrackTest, ConnectAndDisconnectOneSink) {
   const media::AudioParameters params = capturer_->audio_parameters();
   base::WaitableEvent event(false, false);
   EXPECT_CALL(*sink, SetCaptureFormat(_)).WillOnce(Return());
-  EXPECT_CALL(*sink, CaptureData(
-      _, params.channels(), params.frames_per_buffer(), 0, 0))
+  EXPECT_CALL(
+      *sink,
+      CaptureData(
+          _, params.channels(), params.frames_per_buffer(), 0, 0, false))
       .Times(AtLeast(1)).WillRepeatedly(SignalEvent(&event));
   track->AddSink(sink.get());
 
@@ -178,15 +182,19 @@ TEST_F(WebRtcLocalAudioTrackTest, DISABLED_DisableEnableAudioTrack) {
   const media::AudioParameters params = capturer_->audio_parameters();
   base::WaitableEvent event(false, false);
   EXPECT_CALL(*sink, SetCaptureFormat(_)).WillOnce(Return());
-  EXPECT_CALL(*sink, CaptureData(
-      _, params.channels(), params.frames_per_buffer(), 0, 0))
+  EXPECT_CALL(
+      *sink,
+      CaptureData(
+          _, params.channels(), params.frames_per_buffer(), 0, 0, false))
       .Times(0);
   track->AddSink(sink.get());
   EXPECT_FALSE(event.TimedWait(TestTimeouts::tiny_timeout()));
 
   event.Reset();
-  EXPECT_CALL(*sink, CaptureData(
-      _, params.channels(), params.frames_per_buffer(), 0, 0))
+  EXPECT_CALL(
+      *sink,
+      CaptureData(
+          _, params.channels(), params.frames_per_buffer(), 0, 0, false))
       .Times(AtLeast(1)).WillRepeatedly(SignalEvent(&event));
   EXPECT_TRUE(track->set_enabled(true));
   EXPECT_TRUE(event.TimedWait(TestTimeouts::tiny_timeout()));
@@ -210,8 +218,10 @@ TEST_F(WebRtcLocalAudioTrackTest, MultipleAudioTracks) {
   const media::AudioParameters params = capturer_->audio_parameters();
   base::WaitableEvent event_1(false, false);
   EXPECT_CALL(*sink_1, SetCaptureFormat(_)).WillOnce(Return());
-  EXPECT_CALL(*sink_1, CaptureData(
-      _, params.channels(), params.frames_per_buffer(), 0, 0))
+  EXPECT_CALL(
+      *sink_1,
+      CaptureData(
+          _, params.channels(), params.frames_per_buffer(), 0, 0, false))
       .Times(AtLeast(1)).WillRepeatedly(SignalEvent(&event_1));
   track_1->AddSink(sink_1.get());
   EXPECT_TRUE(event_1.TimedWait(TestTimeouts::tiny_timeout()));
@@ -228,11 +238,15 @@ TEST_F(WebRtcLocalAudioTrackTest, MultipleAudioTracks) {
   scoped_ptr<MockWebRtcAudioCapturerSink> sink_2(
         new MockWebRtcAudioCapturerSink());
   EXPECT_CALL(*sink_2, SetCaptureFormat(_)).WillOnce(Return());
-  EXPECT_CALL(*sink_1, CaptureData(
-      _, params.channels(), params.frames_per_buffer(), 0, 0))
+  EXPECT_CALL(
+      *sink_1,
+      CaptureData(
+          _, params.channels(), params.frames_per_buffer(), 0, 0, false))
       .Times(AtLeast(1)).WillRepeatedly(SignalEvent(&event_1));
-  EXPECT_CALL(*sink_2, CaptureData(
-      _, params.channels(), params.frames_per_buffer(), 0, 0))
+  EXPECT_CALL(
+      *sink_2,
+      CaptureData(
+          _, params.channels(), params.frames_per_buffer(), 0, 0, false))
       .Times(AtLeast(1)).WillRepeatedly(SignalEvent(&event_2));
   track_2->AddSink(sink_2.get());
   EXPECT_TRUE(event_1.TimedWait(TestTimeouts::tiny_timeout()));
@@ -271,7 +285,7 @@ TEST_F(WebRtcLocalAudioTrackTest, StartAndStopAudioTracks) {
   scoped_ptr<MockWebRtcAudioCapturerSink> default_sink(
       new MockWebRtcAudioCapturerSink());
   EXPECT_CALL(*default_sink, SetCaptureFormat(_)).WillOnce(Return());
-  EXPECT_CALL(*default_sink, CaptureData(_, _, _, 0, 0))
+  EXPECT_CALL(*default_sink, CaptureData(_, _, _, 0, 0, false))
       .Times(AnyNumber()).WillRepeatedly(Return());
   capturer_->SetDefaultSink(default_sink.get());
   EXPECT_CALL(*capturer_source_.get(), Start()).Times(0);