Index: content/renderer/media/webrtc_audio_capturer_unittest.cc |
diff --git a/content/renderer/media/webrtc_audio_capturer_unittest.cc b/content/renderer/media/webrtc_audio_capturer_unittest.cc |
index 3abd3cdfcedba44496f0d0db09495c41d0125004..a93aede2bca997dc9877fd076f485beb06d9aacf 100644 |
--- a/content/renderer/media/webrtc_audio_capturer_unittest.cc |
+++ b/content/renderer/media/webrtc_audio_capturer_unittest.cc |
@@ -76,28 +76,25 @@ |
void VerifyAudioParams(const blink::WebMediaConstraints& constraints, |
bool need_audio_processing) { |
- const scoped_ptr<WebRtcAudioCapturer> capturer = |
- WebRtcAudioCapturer::CreateCapturer( |
- -1, StreamDeviceInfo( |
- MEDIA_DEVICE_AUDIO_CAPTURE, "", "", params_.sample_rate(), |
- params_.channel_layout(), params_.frames_per_buffer()), |
- constraints, nullptr, nullptr); |
- const scoped_refptr<MockCapturerSource> capturer_source( |
- new MockCapturerSource()); |
- EXPECT_CALL(*capturer_source.get(), Initialize(_, capturer.get(), -1)); |
- EXPECT_CALL(*capturer_source.get(), SetAutomaticGainControl(true)); |
- EXPECT_CALL(*capturer_source.get(), Start()); |
- capturer->SetCapturerSource(capturer_source, params_); |
+ capturer_ = WebRtcAudioCapturer::CreateCapturer( |
+ -1, StreamDeviceInfo(MEDIA_DEVICE_AUDIO_CAPTURE, "", "", |
+ params_.sample_rate(), params_.channel_layout(), |
+ params_.frames_per_buffer()), |
+ constraints, NULL, NULL); |
+ capturer_source_ = new MockCapturerSource(); |
+ EXPECT_CALL(*capturer_source_.get(), Initialize(_, capturer_.get(), -1)); |
+ EXPECT_CALL(*capturer_source_.get(), SetAutomaticGainControl(true)); |
+ EXPECT_CALL(*capturer_source_.get(), Start()); |
+ capturer_->SetCapturerSource(capturer_source_, params_); |
scoped_refptr<WebRtcLocalAudioTrackAdapter> adapter( |
WebRtcLocalAudioTrackAdapter::Create(std::string(), NULL)); |
- const scoped_ptr<WebRtcLocalAudioTrack> track( |
- new WebRtcLocalAudioTrack(adapter.get())); |
- capturer->AddTrack(track.get()); |
+ track_.reset(new WebRtcLocalAudioTrack(adapter.get(), capturer_, NULL)); |
+ track_->Start(); |
// Connect a mock sink to the track. |
scoped_ptr<MockMediaStreamAudioSink> sink(new MockMediaStreamAudioSink()); |
- track->AddSink(sink.get()); |
+ track_->AddSink(sink.get()); |
int delay_ms = 65; |
bool key_pressed = true; |
@@ -108,19 +105,22 @@ |
media::AudioCapturerSource::CaptureCallback* callback = |
static_cast<media::AudioCapturerSource::CaptureCallback*>( |
- capturer.get()); |
+ capturer_.get()); |
// Verify the sink is getting the correct values. |
EXPECT_CALL(*sink, FormatIsSet()); |
EXPECT_CALL(*sink, OnDataCallback()).Times(AtLeast(1)); |
callback->Capture(audio_bus.get(), delay_ms, volume, key_pressed); |
- track->RemoveSink(sink.get()); |
- EXPECT_CALL(*capturer_source.get(), Stop()); |
- capturer->Stop(); |
+ track_->RemoveSink(sink.get()); |
+ EXPECT_CALL(*capturer_source_.get(), Stop()); |
+ capturer_->Stop(); |
} |
media::AudioParameters params_; |
+ scoped_refptr<MockCapturerSource> capturer_source_; |
+ scoped_refptr<WebRtcAudioCapturer> capturer_; |
+ scoped_ptr<WebRtcLocalAudioTrack> track_; |
}; |
TEST_F(WebRtcAudioCapturerTest, VerifyAudioParamsWithAudioProcessing) { |
@@ -137,11 +137,12 @@ |
// Set a non-audio constraint. |
constraint_factory.basic().width.setExact(240); |
- scoped_ptr<WebRtcAudioCapturer> capturer(WebRtcAudioCapturer::CreateCapturer( |
- 0, StreamDeviceInfo(MEDIA_DEVICE_AUDIO_CAPTURE, "", "", |
- params_.sample_rate(), params_.channel_layout(), |
- params_.frames_per_buffer()), |
- constraint_factory.CreateWebMediaConstraints(), NULL, NULL)); |
+ scoped_refptr<WebRtcAudioCapturer> capturer( |
+ WebRtcAudioCapturer::CreateCapturer( |
+ 0, StreamDeviceInfo(MEDIA_DEVICE_AUDIO_CAPTURE, "", "", |
+ params_.sample_rate(), params_.channel_layout(), |
+ params_.frames_per_buffer()), |
+ constraint_factory.CreateWebMediaConstraints(), NULL, NULL)); |
EXPECT_TRUE(capturer.get() == NULL); |
} |