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 4291fb3eb99832a416d691cecab9c8b8ff9a041e..49791e5dd06ebcb537a1527663de1211a8aee3b6 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, |
@@ -156,6 +156,7 @@ 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_; |