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 885ac030fdb06e9e73e7cac5cf5f4664a27c0c38..2c5d23fb534be0631f1cbd15c5fdb9878fd4f1f5 100644 |
--- a/content/renderer/media/audio_track_recorder_unittest.cc |
+++ b/content/renderer/media/audio_track_recorder_unittest.cc |
@@ -11,6 +11,7 @@ |
#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" |
@@ -208,10 +209,15 @@ |
// 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())); |
+ new WebRtcLocalAudioTrack(adapter.get(), capturer, NULL)); |
blink::WebMediaStreamSource audio_source; |
audio_source.initialize(base::UTF8ToUTF16("dummy_source_id"), |
blink::WebMediaStreamSource::TypeAudio, |