| 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 6750af08714a14312916ba88c3ae07b4c35c0ad5..aeb303bf119d1783dc20b3c454eb1fbcaeaaccfb 100644
 | 
| --- a/webrtc/modules/audio_processing/audio_buffer.h
 | 
| +++ b/webrtc/modules/audio_processing/audio_buffer.h
 | 
| @@ -109,7 +109,7 @@ 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) const;
 | 
| +  void InterleaveTo(AudioFrame* frame, bool data_changed);
 | 
|  
 | 
|    // Use for float deinterleaved data.
 | 
|    void CopyFrom(const float* const* data, const StreamConfig& stream_config);
 | 
| @@ -152,6 +152,7 @@ class AudioBuffer {
 | 
|    rtc::scoped_ptr<ChannelBuffer<int16_t> > mixed_low_pass_channels_;
 | 
|    rtc::scoped_ptr<ChannelBuffer<int16_t> > low_pass_reference_channels_;
 | 
|    rtc::scoped_ptr<IFChannelBuffer> input_buffer_;
 | 
| +  rtc::scoped_ptr<IFChannelBuffer> output_buffer_;
 | 
|    rtc::scoped_ptr<ChannelBuffer<float> > process_buffer_;
 | 
|    ScopedVector<PushSincResampler> input_resamplers_;
 | 
|    ScopedVector<PushSincResampler> output_resamplers_;
 | 
| 
 |