| 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 3ee59eacf70f6f96b4d71878be59c64a9545e810..dc2f39fdc197b761cfe67abc82ae64df606245dd 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(
|
|
|