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 5d9d1957b250df67074dff19e28a0785d9bbb4a2..d227186d2d26e61fb47f0bbc95c4b403a0620cf5 100644 |
--- a/content/renderer/media/webrtc_audio_capturer_unittest.cc |
+++ b/content/renderer/media/webrtc_audio_capturer_unittest.cc |
@@ -62,17 +62,17 @@ class WebRtcAudioCapturerTest : public testing::Test { |
: params_(media::AudioParameters::AUDIO_PCM_LOW_LATENCY, |
media::CHANNEL_LAYOUT_STEREO, 48000, 16, 128) { |
#endif |
- capturer_ = WebRtcAudioCapturer::CreateCapturer(); |
blink::WebMediaConstraints constraints; |
- capturer_->Initialize(-1, params_.channel_layout(), params_.sample_rate(), |
- params_.frames_per_buffer(), 0, std::string(), 0, 0, |
- params_.effects(), constraints); |
+ capturer_ = WebRtcAudioCapturer::CreateCapturer( |
+ -1, StreamDeviceInfo(MEDIA_DEVICE_AUDIO_CAPTURE, |
+ "", "", params_.sample_rate(), |
+ params_.channel_layout(), |
+ params_.frames_per_buffer()), |
+ constraints, |
+ NULL); |
capturer_source_ = new MockCapturerSource(); |
- EXPECT_CALL(*capturer_source_.get(), Initialize(_, capturer_.get(), 0)); |
- capturer_->SetCapturerSource(capturer_source_, |
- params_.channel_layout(), |
- params_.sample_rate(), |
- params_.effects(), constraints); |
+ EXPECT_CALL(*capturer_source_.get(), Initialize(_, capturer_.get(), -1)); |
+ capturer_->SetCapturerSourceForTesting(capturer_source_, params_); |
EXPECT_CALL(*capturer_source_.get(), SetAutomaticGainControl(true)); |
EXPECT_CALL(*capturer_source_.get(), Start()); |