mirror of
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.
57 lines
1.5 KiB
C++
57 lines
1.5 KiB
C++
/*
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* Copyright (c) 2017 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 "modules/audio_processing/aec3/aec3_common.h"
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#include <stdint.h>
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#include "rtc_base/checks.h"
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#include "rtc_base/system/arch.h"
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#include "system_wrappers/include/cpu_features_wrapper.h"
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namespace webrtc {
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Aec3Optimization DetectOptimization() {
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#if defined(WEBRTC_ARCH_X86_FAMILY)
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if (WebRtc_GetCPUInfo(kSSE2) != 0) {
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return Aec3Optimization::kSse2;
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}
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#endif
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#if defined(WEBRTC_HAS_NEON)
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return Aec3Optimization::kNeon;
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#endif
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return Aec3Optimization::kNone;
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}
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float FastApproxLog2f(const float in) {
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RTC_DCHECK_GT(in, .0f);
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// Read and interpret float as uint32_t and then cast to float.
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// This is done to extract the exponent (bits 30 - 23).
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// "Right shift" of the exponent is then performed by multiplying
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// with the constant (1/2^23). Finally, we subtract a constant to
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// remove the bias (https://en.wikipedia.org/wiki/Exponent_bias).
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union {
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float dummy;
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uint32_t a;
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} x = {in};
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float out = x.a;
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out *= 1.1920929e-7f; // 1/2^23
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out -= 126.942695f; // Remove bias.
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return out;
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}
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float Log2TodB(const float in_log2) {
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return 3.0102999566398121 * in_log2;
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}
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} // namespace webrtc
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