Index: content/renderer/media/audio_track_recorder_unittest.cc |
diff --git a/content/renderer/media/audio_track_recorder_unittest.cc b/content/renderer/media/audio_track_recorder_unittest.cc |
index 2c5d23fb534be0631f1cbd15c5fdb9878fd4f1f5..885ac030fdb06e9e73e7cac5cf5f4664a27c0c38 100644 |
--- a/content/renderer/media/audio_track_recorder_unittest.cc |
+++ b/content/renderer/media/audio_track_recorder_unittest.cc |
@@ -11,7 +11,6 @@ |
#include "base/stl_util.h" |
#include "base/strings/utf_string_conversions.h" |
#include "content/renderer/media/media_stream_audio_source.h" |
-#include "content/renderer/media/mock_media_constraint_factory.h" |
#include "content/renderer/media/webrtc/webrtc_local_audio_track_adapter.h" |
#include "content/renderer/media/webrtc_local_audio_track.h" |
#include "media/audio/simple_sources.h" |
@@ -209,15 +208,10 @@ class AudioTrackRecorderTest : public TestWithParam<ATRTestParams> { |
// track, which can be used to capture audio data and pass it to the producer. |
// Adapted from media::WebRTCLocalAudioSourceProviderTest. |
void PrepareBlinkTrack() { |
- MockMediaConstraintFactory constraint_factory; |
- scoped_refptr<WebRtcAudioCapturer> capturer( |
- WebRtcAudioCapturer::CreateCapturer( |
- -1, StreamDeviceInfo(), |
- constraint_factory.CreateWebMediaConstraints(), NULL, NULL)); |
scoped_refptr<WebRtcLocalAudioTrackAdapter> adapter( |
WebRtcLocalAudioTrackAdapter::Create(std::string(), NULL)); |
scoped_ptr<WebRtcLocalAudioTrack> native_track( |
- new WebRtcLocalAudioTrack(adapter.get(), capturer, NULL)); |
+ new WebRtcLocalAudioTrack(adapter.get())); |
blink::WebMediaStreamSource audio_source; |
audio_source.initialize(base::UTF8ToUTF16("dummy_source_id"), |
blink::WebMediaStreamSource::TypeAudio, |