| Index: webrtc/voice_engine/utility.cc
|
| diff --git a/webrtc/voice_engine/utility.cc b/webrtc/voice_engine/utility.cc
|
| index 595c71182d9ca192eace4a4640f6f6cc20bdf222..f41cef5888521ca2201f04848f0b8f7bf25e6f83 100644
|
| --- a/webrtc/voice_engine/utility.cc
|
| +++ b/webrtc/voice_engine/utility.cc
|
| @@ -41,14 +41,20 @@ 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_) {
|
| + RTC_DCHECK(num_channels == 2 || num_channels == 4) << "num_channels: "
|
| + << num_channels;
|
| + RTC_DCHECK(dst_frame->num_channels_ == 1 || dst_frame->num_channels_ == 2)
|
| + << "dst_frame->num_channels_: " << dst_frame->num_channels_;
|
| +
|
| + AudioFrameOperations::DownmixChannels(
|
| + src_data, num_channels, samples_per_channel, dst_frame->num_channels_,
|
| + downsampled_audio);
|
| + audio_ptr = downsampled_audio;
|
| + audio_ptr_num_channels = dst_frame->num_channels_;
|
| }
|
|
|
| if (resampler->InitializeIfNeeded(sample_rate_hz, dst_frame->sample_rate_hz_,
|
|
|