| Index: webrtc/build/webrtc.gni
|
| diff --git a/webrtc/build/webrtc.gni b/webrtc/build/webrtc.gni
|
| index 30e505d1d796a7852f3d9d9115d7c6018afa0b95..b345d735f1f412e580abc34a43eae39c0e13499c 100644
|
| --- a/webrtc/build/webrtc.gni
|
| +++ b/webrtc/build/webrtc.gni
|
| @@ -112,6 +112,10 @@ declare_args() {
|
| # Determines whether QUIC code will be built.
|
| rtc_use_quic = false
|
|
|
| + # By default, use normal platform audio support or dummy audio, but don't
|
| + # use file-based audio playout and record.
|
| + rtc_use_dummy_audio_file_devices = false
|
| +
|
| # FFmpeg must be initialized for |H264DecoderImpl| to work. This can be done
|
| # by WebRTC during |H264DecoderImpl::InitDecode| or externally. FFmpeg must
|
| # only be initialized once. Projects that initialize FFmpeg externally, such
|
|
|