| 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));
|
|
|