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 ead1d701db8f19a424b5454f3366b368ac49a12a..f92a8dcb28603f977cf2723f348cdaaf49b4a488 100644 |
--- a/content/renderer/media/webrtc_audio_device_unittest.cc |
+++ b/content/renderer/media/webrtc_audio_device_unittest.cc |
@@ -199,7 +199,8 @@ class MockWebRtcAudioCapturerSink : public WebRtcAudioCapturerSink { |
int number_of_channels, |
int number_of_frames, |
int audio_delay_milliseconds, |
- double volume) OVERRIDE { |
+ double volume, |
+ bool key_pressed) OVERRIDE { |
// Signal that a callback has been received. |
event_->Signal(); |
} |
@@ -357,8 +358,11 @@ int RunWebRtcLoopbackTimeTest(media::AudioManager* manager, |
// Sending fake capture data to WebRtc. |
capturer_sink->CaptureData( |
reinterpret_cast<int16*>(capture_data.get() + input_packet_size * j), |
- num_input_channels, webrtc_audio_device->input_buffer_size(), |
- kHardwareLatencyInMs, 1.0); |
+ num_input_channels, |
+ webrtc_audio_device->input_buffer_size(), |
+ kHardwareLatencyInMs, |
+ 1.0, |
+ false); |
// Receiving data from WebRtc. |
renderer_source->RenderData( |