Index: content/renderer/media/webrtc_local_audio_track_unittest.cc |
diff --git a/content/renderer/media/webrtc_local_audio_track_unittest.cc b/content/renderer/media/webrtc_local_audio_track_unittest.cc |
index 6bdd034ad624becd1fcc33b48cdc9b23c5c9ff2f..e77660e8025dffd06fb08df0181363e03bf32d40 100644 |
--- a/content/renderer/media/webrtc_local_audio_track_unittest.cc |
+++ b/content/renderer/media/webrtc_local_audio_track_unittest.cc |
@@ -181,7 +181,7 @@ class WebRtcLocalAudioTrackTest : public ::testing::Test { |
capturer_ = WebRtcAudioCapturer::CreateCapturer( |
-1, device, constraint_factory.CreateWebMediaConstraints(), NULL, |
audio_source); |
- audio_source->SetAudioCapturer(capturer_); |
+ audio_source->SetAudioCapturer(capturer_.get()); |
capturer_source_ = new MockCapturerSource(capturer_.get()); |
EXPECT_CALL(*capturer_source_.get(), OnInitialize(_, capturer_.get(), -1)) |
.WillOnce(Return()); |
@@ -204,7 +204,7 @@ TEST_F(WebRtcLocalAudioTrackTest, ConnectAndDisconnectOneSink) { |
scoped_refptr<WebRtcLocalAudioTrackAdapter> adapter( |
WebRtcLocalAudioTrackAdapter::Create(std::string(), NULL)); |
scoped_ptr<WebRtcLocalAudioTrack> track( |
- new WebRtcLocalAudioTrack(adapter, capturer_, NULL)); |
+ new WebRtcLocalAudioTrack(adapter.get(), capturer_, NULL)); |
track->Start(); |
EXPECT_TRUE(track->GetAudioAdapter()->enabled()); |
@@ -238,7 +238,7 @@ TEST_F(WebRtcLocalAudioTrackTest, DISABLED_DisableEnableAudioTrack) { |
scoped_refptr<WebRtcLocalAudioTrackAdapter> adapter( |
WebRtcLocalAudioTrackAdapter::Create(std::string(), NULL)); |
scoped_ptr<WebRtcLocalAudioTrack> track( |
- new WebRtcLocalAudioTrack(adapter, capturer_, NULL)); |
+ new WebRtcLocalAudioTrack(adapter.get(), capturer_, NULL)); |
track->Start(); |
EXPECT_TRUE(track->GetAudioAdapter()->enabled()); |
EXPECT_TRUE(track->GetAudioAdapter()->set_enabled(false)); |
@@ -272,7 +272,7 @@ TEST_F(WebRtcLocalAudioTrackTest, DISABLED_MultipleAudioTracks) { |
scoped_refptr<WebRtcLocalAudioTrackAdapter> adapter_1( |
WebRtcLocalAudioTrackAdapter::Create(std::string(), NULL)); |
scoped_ptr<WebRtcLocalAudioTrack> track_1( |
- new WebRtcLocalAudioTrack(adapter_1, capturer_, NULL)); |
+ new WebRtcLocalAudioTrack(adapter_1.get(), capturer_, NULL)); |
track_1->Start(); |
EXPECT_TRUE(track_1->GetAudioAdapter()->enabled()); |
scoped_ptr<MockMediaStreamAudioSink> sink_1(new MockMediaStreamAudioSink()); |
@@ -290,7 +290,7 @@ TEST_F(WebRtcLocalAudioTrackTest, DISABLED_MultipleAudioTracks) { |
scoped_refptr<WebRtcLocalAudioTrackAdapter> adapter_2( |
WebRtcLocalAudioTrackAdapter::Create(std::string(), NULL)); |
scoped_ptr<WebRtcLocalAudioTrack> track_2( |
- new WebRtcLocalAudioTrack(adapter_2, capturer_, NULL)); |
+ new WebRtcLocalAudioTrack(adapter_2.get(), capturer_, NULL)); |
track_2->Start(); |
EXPECT_TRUE(track_2->GetAudioAdapter()->enabled()); |
@@ -329,7 +329,7 @@ TEST_F(WebRtcLocalAudioTrackTest, StartOneAudioTrack) { |
scoped_refptr<WebRtcLocalAudioTrackAdapter> adapter( |
WebRtcLocalAudioTrackAdapter::Create(std::string(), NULL)); |
scoped_ptr<WebRtcLocalAudioTrack> track( |
- new WebRtcLocalAudioTrack(adapter, capturer_, NULL)); |
+ new WebRtcLocalAudioTrack(adapter.get(), capturer_, NULL)); |
track->Start(); |
// When the track goes away, it will automatically stop the |
@@ -345,13 +345,13 @@ TEST_F(WebRtcLocalAudioTrackTest, StartTwoAudioTracks) { |
scoped_refptr<WebRtcLocalAudioTrackAdapter> adapter1( |
WebRtcLocalAudioTrackAdapter::Create(std::string(), NULL)); |
scoped_ptr<WebRtcLocalAudioTrack> track1( |
- new WebRtcLocalAudioTrack(adapter1, capturer_, NULL)); |
+ new WebRtcLocalAudioTrack(adapter1.get(), capturer_, NULL)); |
track1->Start(); |
scoped_refptr<WebRtcLocalAudioTrackAdapter> adapter2( |
WebRtcLocalAudioTrackAdapter::Create(std::string(), NULL)); |
scoped_ptr<WebRtcLocalAudioTrack> track2( |
- new WebRtcLocalAudioTrack(adapter2, capturer_, NULL)); |
+ new WebRtcLocalAudioTrack(adapter2.get(), capturer_, NULL)); |
track2->Start(); |
track1->Stop(); |
@@ -368,7 +368,7 @@ TEST_F(WebRtcLocalAudioTrackTest, StartAndStopAudioTracks) { |
scoped_refptr<WebRtcLocalAudioTrackAdapter> adapter_1( |
WebRtcLocalAudioTrackAdapter::Create(std::string(), NULL)); |
scoped_ptr<WebRtcLocalAudioTrack> track_1( |
- new WebRtcLocalAudioTrack(adapter_1, capturer_, NULL)); |
+ new WebRtcLocalAudioTrack(adapter_1.get(), capturer_, NULL)); |
track_1->Start(); |
// Verify the data flow by connecting the sink to |track_1|. |
@@ -386,7 +386,7 @@ TEST_F(WebRtcLocalAudioTrackTest, StartAndStopAudioTracks) { |
scoped_refptr<WebRtcLocalAudioTrackAdapter> adapter_2( |
WebRtcLocalAudioTrackAdapter::Create(std::string(), NULL)); |
scoped_ptr<WebRtcLocalAudioTrack> track_2( |
- new WebRtcLocalAudioTrack(adapter_2, capturer_, NULL)); |
+ new WebRtcLocalAudioTrack(adapter_2.get(), capturer_, NULL)); |
track_2->Start(); |
// Stop the capturer will clear up the track lists in the capturer. |
@@ -418,7 +418,7 @@ TEST_F(WebRtcLocalAudioTrackTest, |
scoped_refptr<WebRtcLocalAudioTrackAdapter> adapter_1( |
WebRtcLocalAudioTrackAdapter::Create(std::string(), NULL)); |
scoped_ptr<WebRtcLocalAudioTrack> track_1( |
- new WebRtcLocalAudioTrack(adapter_1, capturer_, NULL)); |
+ new WebRtcLocalAudioTrack(adapter_1.get(), capturer_, NULL)); |
track_1->Start(); |
// Verify the data flow by connecting the |sink_1| to |track_1|. |
@@ -451,7 +451,7 @@ TEST_F(WebRtcLocalAudioTrackTest, |
scoped_refptr<WebRtcLocalAudioTrackAdapter> adapter_2( |
WebRtcLocalAudioTrackAdapter::Create(std::string(), NULL)); |
scoped_ptr<WebRtcLocalAudioTrack> track_2( |
- new WebRtcLocalAudioTrack(adapter_2, new_capturer, NULL)); |
+ new WebRtcLocalAudioTrack(adapter_2.get(), new_capturer, NULL)); |
track_2->Start(); |
// Verify the data flow by connecting the |sink_2| to |track_2|. |
@@ -505,7 +505,7 @@ TEST_F(WebRtcLocalAudioTrackTest, TrackWorkWithSmallBufferSize) { |
scoped_refptr<WebRtcLocalAudioTrackAdapter> adapter( |
WebRtcLocalAudioTrackAdapter::Create(std::string(), NULL)); |
scoped_ptr<WebRtcLocalAudioTrack> track( |
- new WebRtcLocalAudioTrack(adapter, capturer, NULL)); |
+ new WebRtcLocalAudioTrack(adapter.get(), capturer, NULL)); |
track->Start(); |
// Verify the data flow by connecting the |sink| to |track|. |
@@ -526,7 +526,7 @@ TEST_F(WebRtcLocalAudioTrackTest, TrackWorkWithSmallBufferSize) { |
EXPECT_EQ(expected_buffer_size, sink->audio_params().frames_per_buffer()); |
// Stopping the new source will stop the second track. |
- EXPECT_CALL(*source, OnStop()).Times(1); |
+ EXPECT_CALL(*source.get(), OnStop()).Times(1); |
capturer->Stop(); |
// Even though this test don't use |capturer_source_| it will be stopped |