Index: content/renderer/media/webrtc_local_audio_source_provider_unittest.cc |
diff --git a/content/renderer/media/webrtc_local_audio_source_provider_unittest.cc b/content/renderer/media/webrtc_local_audio_source_provider_unittest.cc |
index 8b0fd6b27c157e9ce14da82c82b8d83f0135de39..664904e2da5735f0481d7413e7486a0a91ec419c 100644 |
--- a/content/renderer/media/webrtc_local_audio_source_provider_unittest.cc |
+++ b/content/renderer/media/webrtc_local_audio_source_provider_unittest.cc |
@@ -4,6 +4,7 @@ |
#include "base/logging.h" |
#include "base/strings/utf_string_conversions.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_audio_capturer.h" |
#include "content/renderer/media/webrtc_local_audio_source_provider.h" |
@@ -11,7 +12,6 @@ |
#include "media/audio/audio_parameters.h" |
#include "media/base/audio_bus.h" |
#include "testing/gtest/include/gtest/gtest.h" |
-#include "third_party/WebKit/public/platform/WebMediaConstraints.h" |
#include "third_party/WebKit/public/platform/WebMediaStreamTrack.h" |
namespace content { |
@@ -29,10 +29,11 @@ class WebRtcLocalAudioSourceProviderTest : public testing::Test { |
source_params_.frames_per_buffer() * source_params_.channels(); |
source_data_.reset(new int16[length]); |
sink_bus_ = media::AudioBus::Create(sink_params_); |
- blink::WebMediaConstraints constraints; |
- scoped_refptr<WebRtcAudioCapturer> capturer( |
- WebRtcAudioCapturer::CreateCapturer(-1, StreamDeviceInfo(), |
- constraints, NULL)); |
+ MockMediaConstraintFactory constraint_factory; |
+ scoped_refptr<WebRtcAudioCapturer> capturer( |
+ WebRtcAudioCapturer::CreateCapturer( |
+ -1, StreamDeviceInfo(), |
+ constraint_factory.CreateWebMediaConstraints(), NULL)); |
scoped_refptr<WebRtcLocalAudioTrackAdapter> adapter( |
WebRtcLocalAudioTrackAdapter::Create(std::string(), NULL)); |
scoped_ptr<WebRtcLocalAudioTrack> native_track( |