| 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(
|
|
|