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..184ba0155ab39c7f488b6fc75ccc6e584ac769eb 100644 |
--- a/content/renderer/media/webrtc_audio_capturer_unittest.cc |
+++ b/content/renderer/media/webrtc_audio_capturer_unittest.cc |
@@ -96,12 +96,14 @@ class WebRtcAudioCapturerTest : public testing::Test { |
#endif |
capturer_ = WebRtcAudioCapturer::CreateCapturer(); |
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, |
+ params_.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(), |
+ params_.effects()); |
EXPECT_CALL(*capturer_source_.get(), SetAutomaticGainControl(true)); |
EXPECT_CALL(*capturer_source_.get(), Start()); |