Chromium Code Reviews| Index: webrtc/voice_engine/utility.cc |
| diff --git a/webrtc/voice_engine/utility.cc b/webrtc/voice_engine/utility.cc |
| index 595c71182d9ca192eace4a4640f6f6cc20bdf222..9e6743c56ec853e51973e6bf1543d2c526144d83 100644 |
| --- a/webrtc/voice_engine/utility.cc |
| +++ b/webrtc/voice_engine/utility.cc |
| @@ -41,14 +41,15 @@ void RemixAndResample(const int16_t* src_data, |
| AudioFrame* dst_frame) { |
| const int16_t* audio_ptr = src_data; |
| size_t audio_ptr_num_channels = num_channels; |
| - int16_t mono_audio[AudioFrame::kMaxDataSizeSamples]; |
| + int16_t downsampled_audio[AudioFrame::kMaxDataSizeSamples]; |
| // Downmix before resampling. |
| - if (num_channels == 2 && dst_frame->num_channels_ == 1) { |
| - AudioFrameOperations::StereoToMono(src_data, samples_per_channel, |
| - mono_audio); |
| - audio_ptr = mono_audio; |
| - audio_ptr_num_channels = 1; |
| + if (num_channels > dst_frame->num_channels_) { |
| + AudioFrameOperations::DownmixChannels( |
|
jens.nielsen
2017/02/23 16:29:18
Maybe needs some error handling here for unsupport
hlundin-webrtc
2017/02/24 09:46:36
I think this leaves the field a little too open. I
jens.nielsen
2017/02/24 14:55:49
Done.
|
| + src_data, num_channels, samples_per_channel,downsampled_audio, |
| + dst_frame->num_channels_); |
| + audio_ptr = downsampled_audio; |
| + audio_ptr_num_channels = dst_frame->num_channels_; |
| } |
| if (resampler->InitializeIfNeeded(sample_rate_hz, dst_frame->sample_rate_hz_, |