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 f5b668a2fedfe304c32aae43994da0f21c14aa34..3c9caa7d90fa38def765bfa0020413dee4df6301 100644 |
--- a/content/renderer/media/webrtc_local_audio_track_unittest.cc |
+++ b/content/renderer/media/webrtc_local_audio_track_unittest.cc |
@@ -166,9 +166,11 @@ class WebRtcLocalAudioTrackTest : public ::testing::Test { |
capturer_source_ = new MockCapturerSource(capturer_.get()); |
EXPECT_CALL(*capturer_source_.get(), OnInitialize(_, capturer_.get(), 0)) |
.WillOnce(Return()); |
+ media::AudioParameters::PlatformEffects effects; |
capturer_->SetCapturerSource(capturer_source_, |
params_.channel_layout(), |
- params_.sample_rate()); |
+ params_.sample_rate(), |
+ effects); |
} |
media::AudioParameters params_; |
@@ -458,9 +460,11 @@ TEST_F(WebRtcLocalAudioTrackTest, SetNewSourceForCapturerAfterStartTrack) { |
EXPECT_CALL(*new_source.get(), OnInitialize(_, capturer_.get(), 0)) |
.WillOnce(Return()); |
EXPECT_CALL(*new_source.get(), OnStart()); |
+ media::AudioParameters::PlatformEffects effects; |
capturer_->SetCapturerSource(new_source, |
params_.channel_layout(), |
- params_.sample_rate()); |
+ params_.sample_rate(), |
+ effects); |
// Stop the track. |
EXPECT_CALL(*new_source.get(), OnStop()); |
@@ -502,9 +506,11 @@ TEST_F(WebRtcLocalAudioTrackTest, ConnectTracksToDifferentCapturers) { |
scoped_refptr<MockCapturerSource> new_source( |
new MockCapturerSource(new_capturer.get())); |
EXPECT_CALL(*new_source.get(), OnInitialize(_, new_capturer.get(), 0)); |
+ media::AudioParameters::PlatformEffects effects; |
new_capturer->SetCapturerSource(new_source, |
media::CHANNEL_LAYOUT_MONO, |
- 44100); |
+ 44100, |
+ effects); |
// Setup the second audio track, connect it to the new capturer and start it. |
EXPECT_CALL(*new_source.get(), SetAutomaticGainControl(true)); |
@@ -559,12 +565,14 @@ TEST_F(WebRtcLocalAudioTrackTest, TrackWorkWithSmallBufferSize) { |
WebRtcAudioCapturer::CreateCapturer()); |
scoped_refptr<MockCapturerSource> source( |
new MockCapturerSource(capturer.get())); |
+ media::AudioParameters::PlatformEffects effects; |
capturer->Initialize(-1, params.channel_layout(), params.sample_rate(), |
- params.frames_per_buffer(), 0, std::string(), 0, 0); |
+ params.frames_per_buffer(), 0, std::string(), 0, 0, |
+ effects); |
EXPECT_CALL(*source.get(), OnInitialize(_, capturer.get(), 0)); |
capturer->SetCapturerSource(source, params.channel_layout(), |
- params.sample_rate()); |
+ params.sample_rate(), effects); |
// Setup a audio track, connect it to the capturer and start it. |
EXPECT_CALL(*source.get(), SetAutomaticGainControl(true)); |