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