| Index: webrtc/modules/utility/include/audio_frame_operations.h
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| diff --git a/webrtc/modules/utility/include/audio_frame_operations.h b/webrtc/modules/utility/include/audio_frame_operations.h
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| index e12e3e561be8d439fe5a59709149a875fc2b0509..4bf73df6660cbf4eef7c9c86129df399d8d381a6 100644
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| --- a/webrtc/modules/utility/include/audio_frame_operations.h
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| +++ b/webrtc/modules/utility/include/audio_frame_operations.h
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| @@ -10,54 +10,11 @@
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|  
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|  #ifndef WEBRTC_MODULES_UTILITY_INCLUDE_AUDIO_FRAME_OPERATIONS_H_
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|  #define WEBRTC_MODULES_UTILITY_INCLUDE_AUDIO_FRAME_OPERATIONS_H_
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| +// The contents of this file have moved to
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| +// //webrtc/audio/utility. This file is deprecated.
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|  
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| -#include "webrtc/typedefs.h"
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| -
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| -namespace webrtc {
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| -
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| -class AudioFrame;
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| -
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| -// TODO(andrew): consolidate this with utility.h and audio_frame_manipulator.h.
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| -// Change reference parameters to pointers. Consider using a namespace rather
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| -// than a class.
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| -class AudioFrameOperations {
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| - public:
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| -  // Upmixes mono |src_audio| to stereo |dst_audio|. This is an out-of-place
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| -  // operation, meaning src_audio and dst_audio must point to different
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| -  // buffers. It is the caller's responsibility to ensure that |dst_audio| is
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| -  // sufficiently large.
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| -  static void MonoToStereo(const int16_t* src_audio, size_t samples_per_channel,
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| -                           int16_t* dst_audio);
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| -  // |frame.num_channels_| will be updated. This version checks for sufficient
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| -  // buffer size and that |num_channels_| is mono.
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| -  static int MonoToStereo(AudioFrame* frame);
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| -
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| -  // Downmixes stereo |src_audio| to mono |dst_audio|. This is an in-place
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| -  // operation, meaning |src_audio| and |dst_audio| may point to the same
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| -  // buffer.
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| -  static void StereoToMono(const int16_t* src_audio, size_t samples_per_channel,
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| -                           int16_t* dst_audio);
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| -  // |frame.num_channels_| will be updated. This version checks that
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| -  // |num_channels_| is stereo.
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| -  static int StereoToMono(AudioFrame* frame);
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| -
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| -  // Swap the left and right channels of |frame|. Fails silently if |frame| is
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| -  // not stereo.
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| -  static void SwapStereoChannels(AudioFrame* frame);
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| -
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| -  // Conditionally zero out contents of |frame| for implementing audio mute:
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| -  //  |previous_frame_muted| &&  |current_frame_muted| - Zero out whole frame.
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| -  //  |previous_frame_muted| && !|current_frame_muted| - Fade-in at frame start.
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| -  // !|previous_frame_muted| &&  |current_frame_muted| - Fade-out at frame end.
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| -  // !|previous_frame_muted| && !|current_frame_muted| - Leave frame untouched.
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| -  static void Mute(AudioFrame* frame, bool previous_frame_muted,
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| -                   bool current_frame_muted);
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| -
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| -  static int Scale(float left, float right, AudioFrame& frame);
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| -
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| -  static int ScaleWithSat(float scale, AudioFrame& frame);
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| -};
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| -
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| -}  // namespace webrtc
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| +// TODO(aleloi): Remove this file when clients have updated their
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| +// includes. See bugs.webrtc.org/6548.
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| +#include "webrtc/audio/utility/audio_frame_operations.h"
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|  
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|  #endif  // #ifndef WEBRTC_MODULES_UTILITY_INCLUDE_AUDIO_FRAME_OPERATIONS_H_
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| 
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