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) { |