Index: webrtc/voice_engine/utility.cc |
diff --git a/webrtc/voice_engine/utility.cc b/webrtc/voice_engine/utility.cc |
index 720817fc0eaf478eccafbe5084c1a7cde80ad637..f394762c5294bc7fe4c7110cadea16271ab5c18e 100644 |
--- a/webrtc/voice_engine/utility.cc |
+++ b/webrtc/voice_engine/utility.cc |
@@ -41,7 +41,7 @@ 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 downsampled_audio[AudioFrame::kMaxDataSizeSamples]; |
+ int16_t downsmixed_audio[AudioFrame::kMaxDataSizeSamples]; |
ossu
2017/07/07 11:14:42
Perhaps take a closer look at what you actually re
|
// Downmix before resampling. |
if (num_channels > dst_frame->num_channels_) { |
@@ -52,8 +52,8 @@ void RemixAndResample(const int16_t* src_data, |
AudioFrameOperations::DownmixChannels( |
src_data, num_channels, samples_per_channel, dst_frame->num_channels_, |
- downsampled_audio); |
- audio_ptr = downsampled_audio; |
+ downsmixed_audio); |
+ audio_ptr = downsmixed_audio; |
audio_ptr_num_channels = dst_frame->num_channels_; |
} |