| Index: trunk/src/content/renderer/media/webrtc_audio_capturer_unittest.cc
|
| ===================================================================
|
| --- trunk/src/content/renderer/media/webrtc_audio_capturer_unittest.cc (revision 240588)
|
| +++ trunk/src/content/renderer/media/webrtc_audio_capturer_unittest.cc (working copy)
|
| @@ -96,14 +96,12 @@
|
| #endif
|
| capturer_ = WebRtcAudioCapturer::CreateCapturer();
|
| capturer_->Initialize(-1, params_.channel_layout(), params_.sample_rate(),
|
| - params_.frames_per_buffer(), 0, std::string(), 0, 0,
|
| - params_.effects());
|
| + params_.frames_per_buffer(), 0, std::string(), 0, 0);
|
| capturer_source_ = new MockCapturerSource();
|
| EXPECT_CALL(*capturer_source_.get(), Initialize(_, capturer_.get(), 0));
|
| capturer_->SetCapturerSource(capturer_source_,
|
| params_.channel_layout(),
|
| - params_.sample_rate(),
|
| - params_.effects());
|
| + params_.sample_rate());
|
|
|
| EXPECT_CALL(*capturer_source_.get(), SetAutomaticGainControl(true));
|
| EXPECT_CALL(*capturer_source_.get(), Start());
|
|
|