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/*
* Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
*
* Use of this source code is governed by a BSD-style license
* that can be found in the LICENSE file in the root of the source
* tree. An additional intellectual property rights grant can be found
* in the file PATENTS. All contributing project authors may
* be found in the AUTHORS file in the root of the source tree.
*/
#ifndef WEBRTC_MODULES_UTILITY_INCLUDE_AUDIO_FRAME_OPERATIONS_H_
#define WEBRTC_MODULES_UTILITY_INCLUDE_AUDIO_FRAME_OPERATIONS_H_
#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);
// Zeros out the audio and sets |frame.energy| to zero.
static void Mute(AudioFrame& frame);
static int Scale(float left, float right, AudioFrame& frame);
static int ScaleWithSat(float scale, AudioFrame& frame);
};
} // namespace webrtc
#endif // #ifndef WEBRTC_MODULES_UTILITY_INCLUDE_AUDIO_FRAME_OPERATIONS_H_
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