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libtgvoip/webrtc_dsp/modules/audio_processing/aec3/fft_data.h
Grishka 5caaaafa42 Updated WebRTC APM
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.
2018-11-23 04:02:53 +03:00

99 lines
3.1 KiB
C++

/*
* Copyright (c) 2017 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_AEC3_FFT_DATA_H_
#define MODULES_AUDIO_PROCESSING_AEC3_FFT_DATA_H_
// Defines WEBRTC_ARCH_X86_FAMILY, used below.
#include "rtc_base/system/arch.h"
#if defined(WEBRTC_ARCH_X86_FAMILY)
#include <emmintrin.h>
#endif
#include <algorithm>
#include <array>
#include "api/array_view.h"
#include "modules/audio_processing/aec3/aec3_common.h"
namespace webrtc {
// Struct that holds imaginary data produced from 128 point real-valued FFTs.
struct FftData {
// Copies the data in src.
void Assign(const FftData& src) {
std::copy(src.re.begin(), src.re.end(), re.begin());
std::copy(src.im.begin(), src.im.end(), im.begin());
im[0] = im[kFftLengthBy2] = 0;
}
// Clears all the imaginary.
void Clear() {
re.fill(0.f);
im.fill(0.f);
}
// Computes the power spectrum of the data.
void Spectrum(Aec3Optimization optimization,
rtc::ArrayView<float> power_spectrum) const {
RTC_DCHECK_EQ(kFftLengthBy2Plus1, power_spectrum.size());
switch (optimization) {
#if defined(WEBRTC_ARCH_X86_FAMILY)
case Aec3Optimization::kSse2: {
constexpr int kNumFourBinBands = kFftLengthBy2 / 4;
constexpr int kLimit = kNumFourBinBands * 4;
for (size_t k = 0; k < kLimit; k += 4) {
const __m128 r = _mm_loadu_ps(&re[k]);
const __m128 i = _mm_loadu_ps(&im[k]);
const __m128 ii = _mm_mul_ps(i, i);
const __m128 rr = _mm_mul_ps(r, r);
const __m128 rrii = _mm_add_ps(rr, ii);
_mm_storeu_ps(&power_spectrum[k], rrii);
}
power_spectrum[kFftLengthBy2] = re[kFftLengthBy2] * re[kFftLengthBy2] +
im[kFftLengthBy2] * im[kFftLengthBy2];
} break;
#endif
default:
std::transform(re.begin(), re.end(), im.begin(), power_spectrum.begin(),
[](float a, float b) { return a * a + b * b; });
}
}
// Copy the data from an interleaved array.
void CopyFromPackedArray(const std::array<float, kFftLength>& v) {
re[0] = v[0];
re[kFftLengthBy2] = v[1];
im[0] = im[kFftLengthBy2] = 0;
for (size_t k = 1, j = 2; k < kFftLengthBy2; ++k) {
re[k] = v[j++];
im[k] = v[j++];
}
}
// Copies the data into an interleaved array.
void CopyToPackedArray(std::array<float, kFftLength>* v) const {
RTC_DCHECK(v);
(*v)[0] = re[0];
(*v)[1] = re[kFftLengthBy2];
for (size_t k = 1, j = 2; k < kFftLengthBy2; ++k) {
(*v)[j++] = re[k];
(*v)[j++] = im[k];
}
}
std::array<float, kFftLengthBy2Plus1> re;
std::array<float, kFftLengthBy2Plus1> im;
};
} // namespace webrtc
#endif // MODULES_AUDIO_PROCESSING_AEC3_FFT_DATA_H_