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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.
52 lines
1.5 KiB
C++
52 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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#ifndef MODULES_AUDIO_PROCESSING_AEC3_SUBTRACTOR_OUTPUT_H_
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#define MODULES_AUDIO_PROCESSING_AEC3_SUBTRACTOR_OUTPUT_H_
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#include <array>
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#include "api/array_view.h"
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#include "modules/audio_processing/aec3/aec3_common.h"
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#include "modules/audio_processing/aec3/fft_data.h"
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namespace webrtc {
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// Stores the values being returned from the echo subtractor.
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struct SubtractorOutput {
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SubtractorOutput();
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~SubtractorOutput();
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std::array<float, kBlockSize> s_main;
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std::array<float, kBlockSize> s_shadow;
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std::array<float, kBlockSize> e_main;
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std::array<float, kBlockSize> e_shadow;
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FftData E_main;
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std::array<float, kFftLengthBy2Plus1> E2_main;
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std::array<float, kFftLengthBy2Plus1> E2_shadow;
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float s2_main = 0.f;
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float s2_shadow = 0.f;
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float e2_main = 0.f;
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float e2_shadow = 0.f;
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float y2 = 0.f;
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float s_main_max_abs = 0.f;
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float s_shadow_max_abs = 0.f;
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// Reset the struct content.
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void Reset();
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// Updates the powers of the signals.
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void ComputeMetrics(rtc::ArrayView<const float> y);
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};
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} // namespace webrtc
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#endif // MODULES_AUDIO_PROCESSING_AEC3_SUBTRACTOR_OUTPUT_H_
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