| Index: content/renderer/media/webrtc_audio_renderer_unittest.cc
|
| diff --git a/content/renderer/media/webrtc_audio_renderer_unittest.cc b/content/renderer/media/webrtc_audio_renderer_unittest.cc
|
| index 6c1da64b91415c2cf34171a1ce9f1d77d640f6f6..e9d48ef641b1aa83765b53418fa8cb663c91314e 100644
|
| --- a/content/renderer/media/webrtc_audio_renderer_unittest.cc
|
| +++ b/content/renderer/media/webrtc_audio_renderer_unittest.cc
|
| @@ -14,11 +14,13 @@
|
| #include "content/public/renderer/media_stream_audio_renderer.h"
|
| #include "content/renderer/media/audio_device_factory.h"
|
| #include "content/renderer/media/webrtc_audio_device_impl.h"
|
| +#include "media/base/audio_capturer_source.h"
|
| #include "media/base/mock_audio_renderer_sink.h"
|
| #include "testing/gmock/include/gmock/gmock.h"
|
| #include "testing/gtest/include/gtest/gtest.h"
|
| #include "third_party/WebKit/public/platform/WebMediaStream.h"
|
| #include "third_party/WebKit/public/platform/WebMediaStreamTrack.h"
|
| +#include "third_party/WebKit/public/platform/WebString.h"
|
| #include "third_party/WebKit/public/web/WebHeap.h"
|
| #include "third_party/webrtc/api/mediastreaminterface.h"
|
|
|
| @@ -64,7 +66,8 @@ class WebRtcAudioRendererTest : public testing::Test,
|
| : message_loop_(new base::MessageLoopForIO),
|
| source_(new MockAudioRendererSource()) {
|
| blink::WebVector<blink::WebMediaStreamTrack> dummy_tracks;
|
| - stream_.initialize("new stream", dummy_tracks, dummy_tracks);
|
| + stream_.initialize(blink::WebString::fromUTF8("new stream"), dummy_tracks,
|
| + dummy_tracks);
|
| }
|
|
|
| void SetupRenderer(const std::string& device_id) {
|
|
|