
size_t is more appropriate for array lengths, since int might theoretically be too small for a really large array. But more importantly, if the caller's value is naturally of type size_t and the function requires an int, VC++ will trigger warning C4267 (http://msdn.microsoft.com/en-us/library/6kck0s93.aspx) because the implicit cast might be lossy, forcing the caller to do a manual cast. Typing the function with size_t in the first place resolves the problem. R=aluebs@webrtc.org, andrew@webrtc.org, minyue@webrtc.org Review URL: https://webrtc-codereview.appspot.com/21909004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@6702 4adac7df-926f-26a2-2b94-8c16560cd09d
95 lines
3.4 KiB
C++
95 lines
3.4 KiB
C++
/*
|
|
* 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/system_wrappers/interface/scoped_ptr.h"
|
|
#include "webrtc/typedefs.h"
|
|
|
|
namespace webrtc {
|
|
|
|
typedef std::numeric_limits<int16_t> limits_int16;
|
|
|
|
static inline int16_t RoundToInt16(float v) {
|
|
const float kMaxRound = limits_int16::max() - 0.5f;
|
|
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);
|
|
}
|
|
|
|
// Scale (from [-1, 1]) and round to full-range int16 with clamping.
|
|
static inline int16_t ScaleAndRoundToInt16(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);
|
|
}
|
|
|
|
// Scale to float [-1, 1].
|
|
static inline float ScaleToFloat(int16_t v) {
|
|
const float kMaxInt16Inverse = 1.f / limits_int16::max();
|
|
const float kMinInt16Inverse = 1.f / limits_int16::min();
|
|
return v * (v > 0 ? kMaxInt16Inverse : -kMinInt16Inverse);
|
|
}
|
|
|
|
// Round |size| elements of |src| to int16 with clamping and write to |dest|.
|
|
void RoundToInt16(const float* src, size_t size, int16_t* dest);
|
|
|
|
// Scale (from [-1, 1]) and round |size| elements of |src| to full-range int16
|
|
// with clamping and write to |dest|.
|
|
void ScaleAndRoundToInt16(const float* src, size_t size, int16_t* dest);
|
|
|
|
// Scale |size| elements of |src| to float [-1, 1] and write to |dest|.
|
|
void ScaleToFloat(const int16_t* 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** 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_
|