8328e7c44d
This restores the conversion changes to AudioProcessing originally added in r7561, with minor alterations to ensure it passes all tests. TBR=kwiberg Review URL: https://webrtc-codereview.appspot.com/28899004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@7574 4adac7df-926f-26a2-2b94-8c16560cd09d
103 lines
3.7 KiB
C++
103 lines
3.7 KiB
C++
/*
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* Copyright (c) 2013 The WebRTC project authors. All Rights Reserved.
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*
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* Use of this source code is governed by a BSD-style license
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* that can be found in the LICENSE file in the root of the source
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* tree. An additional intellectual property rights grant can be found
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* in the file PATENTS. All contributing project authors may
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* be found in the AUTHORS file in the root of the source tree.
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*/
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#ifndef WEBRTC_COMMON_AUDIO_INCLUDE_AUDIO_UTIL_H_
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#define WEBRTC_COMMON_AUDIO_INCLUDE_AUDIO_UTIL_H_
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#include <limits>
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#include "webrtc/system_wrappers/interface/scoped_ptr.h"
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#include "webrtc/typedefs.h"
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namespace webrtc {
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typedef std::numeric_limits<int16_t> limits_int16;
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// The conversion functions use the following naming convention:
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// S16: int16_t [-32768, 32767]
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// Float: float [-1.0, 1.0]
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// FloatS16: float [-32768.0, 32767.0]
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static inline int16_t FloatToS16(float v) {
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if (v > 0)
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return v >= 1 ? limits_int16::max() :
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static_cast<int16_t>(v * limits_int16::max() + 0.5f);
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return v <= -1 ? limits_int16::min() :
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static_cast<int16_t>(-v * limits_int16::min() - 0.5f);
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}
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static inline float S16ToFloat(int16_t v) {
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static const float kMaxInt16Inverse = 1.f / limits_int16::max();
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static const float kMinInt16Inverse = 1.f / limits_int16::min();
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return v * (v > 0 ? kMaxInt16Inverse : -kMinInt16Inverse);
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}
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static inline int16_t FloatS16ToS16(float v) {
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static const float kMaxRound = limits_int16::max() - 0.5f;
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static const float kMinRound = limits_int16::min() + 0.5f;
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if (v > 0)
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return v >= kMaxRound ? limits_int16::max() :
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static_cast<int16_t>(v + 0.5f);
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return v <= kMinRound ? limits_int16::min() :
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static_cast<int16_t>(v - 0.5f);
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}
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static inline float FloatToFloatS16(float v) {
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return v * (v > 0 ? limits_int16::max() : -limits_int16::min());
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}
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static inline float FloatS16ToFloat(float v) {
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static const float kMaxInt16Inverse = 1.f / limits_int16::max();
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static const float kMinInt16Inverse = 1.f / limits_int16::min();
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return v * (v > 0 ? kMaxInt16Inverse : -kMinInt16Inverse);
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}
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void FloatToS16(const float* src, size_t size, int16_t* dest);
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void S16ToFloat(const int16_t* src, size_t size, float* dest);
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void FloatS16ToS16(const float* src, size_t size, int16_t* dest);
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void FloatToFloatS16(const float* src, size_t size, float* dest);
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void FloatS16ToFloat(const float* src, size_t size, float* dest);
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// Deinterleave audio from |interleaved| to the channel buffers pointed to
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// by |deinterleaved|. There must be sufficient space allocated in the
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// |deinterleaved| buffers (|num_channel| buffers with |samples_per_channel|
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// per buffer).
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template <typename T>
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void Deinterleave(const T* interleaved, int samples_per_channel,
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int num_channels, T* const* deinterleaved) {
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for (int i = 0; i < num_channels; ++i) {
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T* channel = deinterleaved[i];
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int interleaved_idx = i;
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for (int j = 0; j < samples_per_channel; ++j) {
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channel[j] = interleaved[interleaved_idx];
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interleaved_idx += num_channels;
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}
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}
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}
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// Interleave audio from the channel buffers pointed to by |deinterleaved| to
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// |interleaved|. There must be sufficient space allocated in |interleaved|
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// (|samples_per_channel| * |num_channels|).
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template <typename T>
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void Interleave(const T* const* deinterleaved, int samples_per_channel,
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int num_channels, T* interleaved) {
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for (int i = 0; i < num_channels; ++i) {
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const T* channel = deinterleaved[i];
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int interleaved_idx = i;
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for (int j = 0; j < samples_per_channel; ++j) {
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interleaved[interleaved_idx] = channel[j];
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interleaved_idx += num_channels;
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}
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}
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}
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} // namespace webrtc
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#endif // WEBRTC_COMMON_AUDIO_INCLUDE_AUDIO_UTIL_H_
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