| 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_; | 
|  |