Index: content/renderer/media/webrtc_audio_device_unittest.cc |
diff --git a/content/renderer/media/webrtc_audio_device_unittest.cc b/content/renderer/media/webrtc_audio_device_unittest.cc |
index 6746ab650aa87657b60f91d8248b9d6a7f516976..a08ea93365b2f401579982c9ead8a4dd0b7710d8 100644 |
--- a/content/renderer/media/webrtc_audio_device_unittest.cc |
+++ b/content/renderer/media/webrtc_audio_device_unittest.cc |
@@ -221,7 +221,8 @@ class MockWebRtcAudioCapturerSink : public WebRtcAudioCapturerSink { |
int number_of_frames, |
int audio_delay_milliseconds, |
int current_volume, |
- bool need_audio_processing) OVERRIDE { |
+ bool need_audio_processing, |
+ bool key_pressed) OVERRIDE { |
// Signal that a callback has been received. |
event_->Signal(); |
return 0; |
@@ -381,8 +382,13 @@ int RunWebRtcLoopbackTimeTest(media::AudioManager* manager, |
capturer_sink->CaptureData( |
voe_channels, |
reinterpret_cast<int16*>(capture_data.get() + input_packet_size * j), |
- params.sample_rate(), params.channels(), params.frames_per_buffer(), |
- kHardwareLatencyInMs, 1.0, enable_apm); |
+ params.sample_rate(), |
+ params.channels(), |
+ params.frames_per_buffer(), |
+ kHardwareLatencyInMs, |
+ 1.0, |
+ enable_apm, |
+ false); |
// Receiving data from WebRtc. |
renderer_source->RenderData( |