Index: webrtc/modules/audio_processing/audio_buffer.cc |
diff --git a/webrtc/modules/audio_processing/audio_buffer.cc b/webrtc/modules/audio_processing/audio_buffer.cc |
index f300bb85cd0ba21e4f1913b872dcf0f962caa2f1..f9b9f631d2b813523ed4a9935ecbc3c0af2ab3bd 100644 |
--- a/webrtc/modules/audio_processing/audio_buffer.cc |
+++ b/webrtc/modules/audio_processing/audio_buffer.cc |
@@ -430,10 +430,10 @@ void AudioBuffer::InterleaveTo(AudioFrame* frame, bool data_changed) { |
} |
if (frame->num_channels_ == num_channels_) { |
- Interleave(data_ptr->ibuf()->channels(), proc_num_frames_, num_channels_, |
+ Interleave(data_ptr->ibuf()->channels(), output_num_frames_, num_channels_, |
frame->data_); |
} else { |
- UpmixMonoToInterleaved(data_ptr->ibuf()->channels()[0], proc_num_frames_, |
+ UpmixMonoToInterleaved(data_ptr->ibuf()->channels()[0], output_num_frames_, |
frame->num_channels_, frame->data_); |
} |
} |