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 11a017c021d81d634f8ce0e760621cc11dfb459d..d0bcf296c6042413fc3785809ea3fa5c731a1ae3 100644 |
--- a/content/renderer/media/webrtc_audio_capturer_unittest.cc |
+++ b/content/renderer/media/webrtc_audio_capturer_unittest.cc |
@@ -106,8 +106,6 @@ class WebRtcAudioCapturerTest : public testing::Test { |
scoped_refptr<WebRtcLocalAudioTrackAdapter> adapter( |
WebRtcLocalAudioTrackAdapter::Create(std::string(), NULL)); |
track_.reset(new WebRtcLocalAudioTrack(adapter, capturer_, NULL)); |
- static_cast<WebRtcLocalAudioSourceProvider*>( |
- track_->audio_source_provider())->SetSinkParamsForTesting(params_); |
track_->Start(); |
// Connect a mock sink to the track. |