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libtgvoip/webrtc_dsp/webrtc/common_audio/audio_util.cc
Grishka 333c4a1101 Added working audio i/o for OS X
Added simple audio resampler
Replaced prebuilt static libs with their sources & added that to all project files (closes #5)
2017-04-09 19:14:33 +03:00

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1.6 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.
*/
#include "webrtc/common_audio/include/audio_util.h"
#include "webrtc/typedefs.h"
namespace webrtc {
void FloatToS16(const float* src, size_t size, int16_t* dest) {
for (size_t i = 0; i < size; ++i)
dest[i] = FloatToS16(src[i]);
}
void S16ToFloat(const int16_t* src, size_t size, float* dest) {
for (size_t i = 0; i < size; ++i)
dest[i] = S16ToFloat(src[i]);
}
void FloatS16ToS16(const float* src, size_t size, int16_t* dest) {
for (size_t i = 0; i < size; ++i)
dest[i] = FloatS16ToS16(src[i]);
}
void FloatToFloatS16(const float* src, size_t size, float* dest) {
for (size_t i = 0; i < size; ++i)
dest[i] = FloatToFloatS16(src[i]);
}
void FloatS16ToFloat(const float* src, size_t size, float* dest) {
for (size_t i = 0; i < size; ++i)
dest[i] = FloatS16ToFloat(src[i]);
}
template <>
void DownmixInterleavedToMono<int16_t>(const int16_t* interleaved,
size_t num_frames,
int num_channels,
int16_t* deinterleaved) {
DownmixInterleavedToMonoImpl<int16_t, int32_t>(interleaved, num_frames,
num_channels, deinterleaved);
}
} // namespace webrtc