| 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_;
|
|
|