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 1151dcef6ac1d15706625a42b1ba6862a3785ef9..a9f99a73baf9101f718a2d35c27ba6078ed9341d 100644 |
--- a/content/renderer/media/webrtc_audio_capturer_unittest.cc |
+++ b/content/renderer/media/webrtc_audio_capturer_unittest.cc |
@@ -85,7 +85,7 @@ class WebRtcAudioCapturerTest : public testing::Test { |
EXPECT_CALL(*capturer_source_.get(), Initialize(_, capturer_.get(), -1)); |
EXPECT_CALL(*capturer_source_.get(), SetAutomaticGainControl(true)); |
EXPECT_CALL(*capturer_source_.get(), Start()); |
- capturer_->SetCapturerSourceForTesting(capturer_source_, params_); |
+ capturer_->SetCapturerSource(capturer_source_, params_); |
scoped_refptr<WebRtcLocalAudioTrackAdapter> adapter( |
WebRtcLocalAudioTrackAdapter::Create(std::string(), NULL)); |