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 b75ca79b920f230ac68f68b1658808df5daba2b3..534784b9831239f00886582162ba5cbafd4f29b8 100644 |
--- a/content/renderer/media/webrtc_local_audio_track_unittest.cc |
+++ b/content/renderer/media/webrtc_local_audio_track_unittest.cc |
@@ -181,14 +181,15 @@ TEST_F(WebRtcLocalAudioTrackTest, ConnectAndDisconnectOneSink) { |
base::WaitableEvent event(false, false); |
EXPECT_CALL(*sink, SetCaptureFormat(_)).WillOnce(Return()); |
EXPECT_CALL(*sink, |
- CaptureData(kNumberOfNetworkChannels, |
- params.sample_rate(), |
- params.channels(), |
- params.frames_per_buffer(), |
- 0, |
- 0, |
- false, |
- false)).Times(AtLeast(1)) |
+ CaptureData(kNumberOfNetworkChannels, |
+ params.sample_rate(), |
+ params.channels(), |
+ params.frames_per_buffer(), |
+ 0, |
+ 0, |
+ // TODO(tommi): Change to |false| when issue 277134 is fixed. |
+ true, |
+ false)).Times(AtLeast(1)) |
.WillRepeatedly(SignalEvent(&event)); |
track->AddSink(sink.get()); |
@@ -268,14 +269,15 @@ TEST_F(WebRtcLocalAudioTrackTest, MultipleAudioTracks) { |
base::WaitableEvent event_1(false, false); |
EXPECT_CALL(*sink_1, SetCaptureFormat(_)).WillOnce(Return()); |
EXPECT_CALL(*sink_1, |
- CaptureData(1, |
- params.sample_rate(), |
- params.channels(), |
- params.frames_per_buffer(), |
- 0, |
- 0, |
- false, |
- false)).Times(AtLeast(1)) |
+ CaptureData(1, |
+ params.sample_rate(), |
+ params.channels(), |
+ params.frames_per_buffer(), |
+ 0, |
+ 0, |
+ // TODO(tommi): Change to |false| when issue 277134 is fixed. |
+ true, |
+ false)).Times(AtLeast(1)) |
.WillRepeatedly(SignalEvent(&event_1)); |
track_1->AddSink(sink_1.get()); |
EXPECT_TRUE(event_1.TimedWait(TestTimeouts::tiny_timeout())); |
@@ -295,24 +297,26 @@ TEST_F(WebRtcLocalAudioTrackTest, MultipleAudioTracks) { |
new MockWebRtcAudioCapturerSink()); |
EXPECT_CALL(*sink_2, SetCaptureFormat(_)).WillOnce(Return()); |
EXPECT_CALL(*sink_1, |
- CaptureData(1, |
- params.sample_rate(), |
- params.channels(), |
- params.frames_per_buffer(), |
- 0, |
- 0, |
- false, |
- false)).Times(AtLeast(1)) |
+ CaptureData(1, |
+ params.sample_rate(), |
+ params.channels(), |
+ params.frames_per_buffer(), |
+ 0, |
+ 0, |
+ // TODO(tommi): Change to |false| when issue 277134 is fixed. |
+ true, |
+ false)).Times(AtLeast(1)) |
.WillRepeatedly(SignalEvent(&event_1)); |
EXPECT_CALL(*sink_2, |
- CaptureData(1, |
- params.sample_rate(), |
- params.channels(), |
- params.frames_per_buffer(), |
- 0, |
- 0, |
- false, |
- false)).Times(AtLeast(1)) |
+ CaptureData(1, |
+ params.sample_rate(), |
+ params.channels(), |
+ params.frames_per_buffer(), |
+ 0, |
+ 0, |
+ // TODO(tommi): Change to |false| when issue 277134 is fixed. |
+ true, |
+ false)).Times(AtLeast(1)) |
.WillRepeatedly(SignalEvent(&event_2)); |
track_2->AddSink(sink_2.get()); |
EXPECT_TRUE(event_1.TimedWait(TestTimeouts::tiny_timeout())); |
@@ -361,7 +365,10 @@ TEST_F(WebRtcLocalAudioTrackTest, StartAndStopAudioTracks) { |
scoped_ptr<MockWebRtcAudioCapturerSink> sink( |
new MockWebRtcAudioCapturerSink()); |
event.Reset(); |
- EXPECT_CALL(*sink, CaptureData(_, _, _, _, 0, 0, false, false)) |
+ EXPECT_CALL(*sink, CaptureData(_, _, _, _, 0, 0, |
+ // TODO(tommi): Change to |false| when issue 277134 is fixed. |
+ true, |
+ false)) |
.Times(AnyNumber()).WillRepeatedly(Return()); |
EXPECT_CALL(*sink, SetCaptureFormat(_)).Times(1); |
track_1->AddSink(sink.get()); |
@@ -441,7 +448,10 @@ TEST_F(WebRtcLocalAudioTrackTest, ConnectTracksToDifferentCapturers) { |
EXPECT_CALL( |
*sink_1.get(), |
CaptureData( |
- kNumberOfNetworkChannelsForTrack1, 48000, 2, _, 0, 0, false, false)) |
+ kNumberOfNetworkChannelsForTrack1, 48000, 2, _, 0, 0, |
+ // TODO(tommi): Change to |false| when issue 277134 is fixed. |
+ true, |
+ false)) |
.Times(AnyNumber()).WillRepeatedly(Return()); |
EXPECT_CALL(*sink_1.get(), SetCaptureFormat(_)).Times(1); |
track_1->AddSink(sink_1.get()); |