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_, |