| Index: webrtc/modules/audio_processing/audio_buffer.h
|
| diff --git a/webrtc/modules/audio_processing/audio_buffer.h b/webrtc/modules/audio_processing/audio_buffer.h
|
| index d0e79db3293c790d7dfd7504f3e6d58f32fff3c1..da75dbf1b80de0217253fb8aca34d823490ea8c6 100644
|
| --- a/webrtc/modules/audio_processing/audio_buffer.h
|
| +++ b/webrtc/modules/audio_processing/audio_buffer.h
|
| @@ -110,7 +110,7 @@ class AudioBuffer {
|
| void DeinterleaveFrom(AudioFrame* audioFrame);
|
| // If |data_changed| is false, only the non-audio data members will be copied
|
| // to |frame|.
|
| - void InterleaveTo(AudioFrame* frame, bool data_changed);
|
| + void InterleaveTo(AudioFrame* frame, bool data_changed) const;
|
|
|
| // Use for float deinterleaved data.
|
| void CopyFrom(const float* const* data, const StreamConfig& stream_config);
|
|
|