diff options
Diffstat (limited to 'content/renderer/media/webrtc_audio_device_unittest.cc')
| -rw-r--r-- | content/renderer/media/webrtc_audio_device_unittest.cc | 24 |
1 files changed, 12 insertions, 12 deletions
diff --git a/content/renderer/media/webrtc_audio_device_unittest.cc b/content/renderer/media/webrtc_audio_device_unittest.cc index 222271d..37d5401 100644 --- a/content/renderer/media/webrtc_audio_device_unittest.cc +++ b/content/renderer/media/webrtc_audio_device_unittest.cc @@ -105,7 +105,7 @@ bool HardwareSampleRatesAreValid() { // HardwareSampleRatesAreValid() has been called and returned true. bool InitializeCapturer(WebRtcAudioDeviceImpl* webrtc_audio_device) { // Access the capturer owned and created by the audio device. - WebRtcAudioCapturer* capturer = webrtc_audio_device->capturer(); + WebRtcAudioCapturer* capturer = webrtc_audio_device->capturer().get(); if (!capturer) return false; @@ -297,7 +297,7 @@ int RunWebRtcLoopbackTimeTest(media::AudioManager* manager, EXPECT_TRUE(engine.valid()); ScopedWebRTCPtr<webrtc::VoEBase> base(engine.get()); EXPECT_TRUE(base.valid()); - int err = base->Init(webrtc_audio_device); + int err = base->Init(webrtc_audio_device.get()); EXPECT_EQ(0, err); // We use SetCaptureFormat() and SetRenderFormat() to configure the audio @@ -454,7 +454,7 @@ TEST_F(WebRTCAudioDeviceTest, Construct) { ASSERT_TRUE(engine.valid()); ScopedWebRTCPtr<webrtc::VoEBase> base(engine.get()); - int err = base->Init(webrtc_audio_device); + int err = base->Init(webrtc_audio_device.get()); EXPECT_TRUE(InitializeCapturer(webrtc_audio_device.get())); EXPECT_EQ(0, err); EXPECT_EQ(0, base->Terminate()); @@ -493,14 +493,14 @@ TEST_F(WebRTCAudioDeviceTest, DISABLED_StartPlayout) { new WebRtcAudioRenderer(kRenderViewId); scoped_refptr<WebRtcAudioDeviceImpl> webrtc_audio_device( new WebRtcAudioDeviceImpl()); - EXPECT_TRUE(webrtc_audio_device->SetAudioRenderer(renderer)); + EXPECT_TRUE(webrtc_audio_device->SetAudioRenderer(renderer.get())); WebRTCAutoDelete<webrtc::VoiceEngine> engine(webrtc::VoiceEngine::Create()); ASSERT_TRUE(engine.valid()); ScopedWebRTCPtr<webrtc::VoEBase> base(engine.get()); ASSERT_TRUE(base.valid()); - int err = base->Init(webrtc_audio_device); + int err = base->Init(webrtc_audio_device.get()); ASSERT_EQ(0, err); int ch = base->CreateChannel(); @@ -578,7 +578,7 @@ TEST_F(WebRTCAudioDeviceTest, MAYBE_StartRecording) { ScopedWebRTCPtr<webrtc::VoEBase> base(engine.get()); ASSERT_TRUE(base.valid()); - int err = base->Init(webrtc_audio_device); + int err = base->Init(webrtc_audio_device.get()); ASSERT_EQ(0, err); EXPECT_TRUE(InitializeCapturer(webrtc_audio_device.get())); @@ -656,14 +656,14 @@ TEST_F(WebRTCAudioDeviceTest, DISABLED_PlayLocalFile) { new WebRtcAudioRenderer(kRenderViewId); scoped_refptr<WebRtcAudioDeviceImpl> webrtc_audio_device( new WebRtcAudioDeviceImpl()); - EXPECT_TRUE(webrtc_audio_device->SetAudioRenderer(renderer)); + EXPECT_TRUE(webrtc_audio_device->SetAudioRenderer(renderer.get())); WebRTCAutoDelete<webrtc::VoiceEngine> engine(webrtc::VoiceEngine::Create()); ASSERT_TRUE(engine.valid()); ScopedWebRTCPtr<webrtc::VoEBase> base(engine.get()); ASSERT_TRUE(base.valid()); - int err = base->Init(webrtc_audio_device); + int err = base->Init(webrtc_audio_device.get()); ASSERT_EQ(0, err); int ch = base->CreateChannel(); @@ -734,14 +734,14 @@ TEST_F(WebRTCAudioDeviceTest, MAYBE_FullDuplexAudioWithAGC) { new WebRtcAudioRenderer(kRenderViewId); scoped_refptr<WebRtcAudioDeviceImpl> webrtc_audio_device( new WebRtcAudioDeviceImpl()); - EXPECT_TRUE(webrtc_audio_device->SetAudioRenderer(renderer)); + EXPECT_TRUE(webrtc_audio_device->SetAudioRenderer(renderer.get())); WebRTCAutoDelete<webrtc::VoiceEngine> engine(webrtc::VoiceEngine::Create()); ASSERT_TRUE(engine.valid()); ScopedWebRTCPtr<webrtc::VoEBase> base(engine.get()); ASSERT_TRUE(base.valid()); - int err = base->Init(webrtc_audio_device); + int err = base->Init(webrtc_audio_device.get()); ASSERT_EQ(0, err); EXPECT_TRUE(InitializeCapturer(webrtc_audio_device.get())); @@ -811,7 +811,7 @@ TEST_F(WebRTCAudioDeviceTest, WebRtcRecordingSetupTime) { ScopedWebRTCPtr<webrtc::VoEBase> base(engine.get()); ASSERT_TRUE(base.valid()); - int err = base->Init(webrtc_audio_device); + int err = base->Init(webrtc_audio_device.get()); ASSERT_EQ(0, err); EXPECT_TRUE(InitializeCapturer(webrtc_audio_device.get())); @@ -820,7 +820,7 @@ TEST_F(WebRTCAudioDeviceTest, WebRtcRecordingSetupTime) { base::WaitableEvent event(false, false); scoped_ptr<MockWebRtcAudioCapturerSink> capturer_sink( new MockWebRtcAudioCapturerSink(&event)); - WebRtcAudioCapturer* capturer = webrtc_audio_device->capturer(); + WebRtcAudioCapturer* capturer = webrtc_audio_device->capturer().get(); capturer->AddSink(capturer_sink.get()); int ch = base->CreateChannel(); |