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https://github.com/danog/libtgvoip.git
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5caaaafa42
I'm now using the entire audio processing module from WebRTC as opposed to individual DSP algorithms pulled from there before. Seems to work better this way.
50 lines
1.6 KiB
C++
50 lines
1.6 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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#include "common_audio/include/audio_util.h"
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namespace webrtc {
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void FloatToS16(const float* src, size_t size, int16_t* dest) {
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for (size_t i = 0; i < size; ++i)
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dest[i] = FloatToS16(src[i]);
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}
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void S16ToFloat(const int16_t* src, size_t size, float* dest) {
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for (size_t i = 0; i < size; ++i)
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dest[i] = S16ToFloat(src[i]);
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}
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void FloatS16ToS16(const float* src, size_t size, int16_t* dest) {
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for (size_t i = 0; i < size; ++i)
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dest[i] = FloatS16ToS16(src[i]);
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}
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void FloatToFloatS16(const float* src, size_t size, float* dest) {
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for (size_t i = 0; i < size; ++i)
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dest[i] = FloatToFloatS16(src[i]);
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}
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void FloatS16ToFloat(const float* src, size_t size, float* dest) {
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for (size_t i = 0; i < size; ++i)
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dest[i] = FloatS16ToFloat(src[i]);
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}
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template <>
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void DownmixInterleavedToMono<int16_t>(const int16_t* interleaved,
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size_t num_frames,
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int num_channels,
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int16_t* deinterleaved) {
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DownmixInterleavedToMonoImpl<int16_t, int32_t>(interleaved, num_frames,
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num_channels, deinterleaved);
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}
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} // namespace webrtc
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