mirror of
https://github.com/mollyim/webrtc.git
synced 2025-05-13 22:00:47 +01:00

Usage replaced with stdint.h, rtc_base/system/arch.h and rtc_base/system/unused.h, as appropriate. Bug: webrtc:6854 Change-Id: I97225465d14b969903d92979e2df3c3c05d35f18 Reviewed-on: https://webrtc-review.googlesource.com/90249 Reviewed-by: Niklas Enbom <niklas.enbom@webrtc.org> Reviewed-by: Fredrik Solenberg <solenberg@webrtc.org> Commit-Queue: Niels Moller <nisse@webrtc.org> Cr-Commit-Position: refs/heads/master@{#24100}
49 lines
1.6 KiB
C++
49 lines
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 "common_audio/include/audio_util.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
|