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/*
* Copyright (c) 2013 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_COMMON_AUDIO_INCLUDE_AUDIO_UTIL_H_
#define WEBRTC_COMMON_AUDIO_INCLUDE_AUDIO_UTIL_H_
#include <limits>
#include "webrtc/base/scoped_ptr.h"
#include "webrtc/typedefs.h"
namespace webrtc {
typedef std::numeric_limits<int16_t> limits_int16;
// The conversion functions use the following naming convention:
// S16: int16_t [-32768, 32767]
// Float: float [-1.0, 1.0]
// FloatS16: float [-32768.0, 32767.0]
static inline int16_t FloatToS16(float v) {
if (v > 0)
return v >= 1 ? limits_int16::max() :
static_cast<int16_t>(v * limits_int16::max() + 0.5f);
return v <= -1 ? limits_int16::min() :
static_cast<int16_t>(-v * limits_int16::min() - 0.5f);
}
static inline float S16ToFloat(int16_t v) {
static const float kMaxInt16Inverse = 1.f / limits_int16::max();
static const float kMinInt16Inverse = 1.f / limits_int16::min();
return v * (v > 0 ? kMaxInt16Inverse : -kMinInt16Inverse);
}
static inline int16_t FloatS16ToS16(float v) {
static const float kMaxRound = limits_int16::max() - 0.5f;
static const float kMinRound = limits_int16::min() + 0.5f;
if (v > 0)
return v >= kMaxRound ? limits_int16::max() :
static_cast<int16_t>(v + 0.5f);
return v <= kMinRound ? limits_int16::min() :
static_cast<int16_t>(v - 0.5f);
}
static inline float FloatToFloatS16(float v) {
return v * (v > 0 ? limits_int16::max() : -limits_int16::min());
}
static inline float FloatS16ToFloat(float v) {
static const float kMaxInt16Inverse = 1.f / limits_int16::max();
static const float kMinInt16Inverse = 1.f / limits_int16::min();
return v * (v > 0 ? kMaxInt16Inverse : -kMinInt16Inverse);
}
void FloatToS16(const float* src, size_t size, int16_t* dest);
void S16ToFloat(const int16_t* src, size_t size, float* dest);
void FloatS16ToS16(const float* src, size_t size, int16_t* dest);
void FloatToFloatS16(const float* src, size_t size, float* dest);
void FloatS16ToFloat(const float* src, size_t size, float* dest);
// Deinterleave audio from |interleaved| to the channel buffers pointed to
// by |deinterleaved|. There must be sufficient space allocated in the
// |deinterleaved| buffers (|num_channel| buffers with |samples_per_channel|
// per buffer).
template <typename T>
void Deinterleave(const T* interleaved, int samples_per_channel,
int num_channels, T* const* deinterleaved) {
for (int i = 0; i < num_channels; ++i) {
T* channel = deinterleaved[i];
int interleaved_idx = i;
for (int j = 0; j < samples_per_channel; ++j) {
channel[j] = interleaved[interleaved_idx];
interleaved_idx += num_channels;
}
}
}
// Interleave audio from the channel buffers pointed to by |deinterleaved| to
// |interleaved|. There must be sufficient space allocated in |interleaved|
// (|samples_per_channel| * |num_channels|).
template <typename T>
void Interleave(const T* const* deinterleaved, int samples_per_channel,
int num_channels, T* interleaved) {
for (int i = 0; i < num_channels; ++i) {
const T* channel = deinterleaved[i];
int interleaved_idx = i;
for (int j = 0; j < samples_per_channel; ++j) {
interleaved[interleaved_idx] = channel[j];
interleaved_idx += num_channels;
}
}
}
} // namespace webrtc
#endif // WEBRTC_COMMON_AUDIO_INCLUDE_AUDIO_UTIL_H_
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