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..6750af08714a14312916ba88c3ae07b4c35c0ad5 100644 |
--- a/webrtc/modules/audio_processing/audio_buffer.h |
+++ b/webrtc/modules/audio_processing/audio_buffer.h |
@@ -112,12 +112,8 @@ class AudioBuffer { |
void InterleaveTo(AudioFrame* frame, bool data_changed) const; |
// Use for float deinterleaved data. |
- void CopyFrom(const float* const* data, |
- int num_frames, |
- AudioProcessing::ChannelLayout layout); |
- void CopyTo(int num_frames, |
- AudioProcessing::ChannelLayout layout, |
- float* const* data); |
+ void CopyFrom(const float* const* data, const StreamConfig& stream_config); |
+ void CopyTo(const StreamConfig& stream_config, float* const* data); |
void CopyLowPassToReference(); |
// Splits the signal into different bands. |