| Index: content/renderer/media/webrtc_audio_capturer_unittest.cc
|
| diff --git a/content/renderer/media/webrtc_audio_capturer_unittest.cc b/content/renderer/media/webrtc_audio_capturer_unittest.cc
|
| index cc7d528912567736b513c9aaf1d0a5f754b60882..8ad12a3237a45d6ed8be94bddf00dc1f5fd84fc0 100644
|
| --- a/content/renderer/media/webrtc_audio_capturer_unittest.cc
|
| +++ b/content/renderer/media/webrtc_audio_capturer_unittest.cc
|
| @@ -95,13 +95,16 @@ class WebRtcAudioCapturerTest : public testing::Test {
|
| media::CHANNEL_LAYOUT_STEREO, 48000, 16, 128) {
|
| #endif
|
| capturer_ = WebRtcAudioCapturer::CreateCapturer();
|
| + 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);
|
| capturer_source_ = new MockCapturerSource();
|
| EXPECT_CALL(*capturer_source_.get(), Initialize(_, capturer_.get(), 0));
|
| capturer_->SetCapturerSource(capturer_source_,
|
| params_.channel_layout(),
|
| - params_.sample_rate());
|
| + params_.sample_rate(),
|
| + effects);
|
|
|
| EXPECT_CALL(*capturer_source_.get(), SetAutomaticGainControl(true));
|
| EXPECT_CALL(*capturer_source_.get(), Start());
|
|
|