mirror of
https://github.com/danog/libtgvoip.git
synced 2025-01-06 04:38:16 +01:00
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.
61 lines
1.5 KiB
C++
61 lines
1.5 KiB
C++
/*
|
|
* Copyright (c) 2016 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 MODULES_AUDIO_PROCESSING_UTILITY_OOURA_FFT_H_
|
|
#define MODULES_AUDIO_PROCESSING_UTILITY_OOURA_FFT_H_
|
|
|
|
#include "rtc_base/system/arch.h"
|
|
|
|
namespace webrtc {
|
|
|
|
#if defined(WEBRTC_ARCH_X86_FAMILY)
|
|
void cft1st_128_SSE2(float* a);
|
|
void cftmdl_128_SSE2(float* a);
|
|
void rftfsub_128_SSE2(float* a);
|
|
void rftbsub_128_SSE2(float* a);
|
|
#endif
|
|
|
|
#if defined(MIPS_FPU_LE)
|
|
void cft1st_128_mips(float* a);
|
|
void cftmdl_128_mips(float* a);
|
|
void rftfsub_128_mips(float* a);
|
|
void rftbsub_128_mips(float* a);
|
|
#endif
|
|
|
|
#if defined(WEBRTC_HAS_NEON)
|
|
void cft1st_128_neon(float* a);
|
|
void cftmdl_128_neon(float* a);
|
|
void rftfsub_128_neon(float* a);
|
|
void rftbsub_128_neon(float* a);
|
|
#endif
|
|
|
|
class OouraFft {
|
|
public:
|
|
OouraFft();
|
|
~OouraFft();
|
|
void Fft(float* a) const;
|
|
void InverseFft(float* a) const;
|
|
|
|
private:
|
|
void cft1st_128(float* a) const;
|
|
void cftmdl_128(float* a) const;
|
|
void rftfsub_128(float* a) const;
|
|
void rftbsub_128(float* a) const;
|
|
|
|
void cftfsub_128(float* a) const;
|
|
void cftbsub_128(float* a) const;
|
|
void bitrv2_128(float* a) const;
|
|
bool use_sse2_;
|
|
};
|
|
|
|
} // namespace webrtc
|
|
|
|
#endif // MODULES_AUDIO_PROCESSING_UTILITY_OOURA_FFT_H_
|