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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.
55 lines
1.7 KiB
C++
55 lines
1.7 KiB
C++
/*
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* Copyright (c) 2018 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_REVERB_FREQUENCY_RESPONSE_H_
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#define MODULES_AUDIO_PROCESSING_AEC3_REVERB_FREQUENCY_RESPONSE_H_
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#include <array>
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#include <vector>
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#include "absl/types/optional.h"
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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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namespace webrtc {
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// Class for updating the frequency response for the reverb.
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class ReverbFrequencyResponse {
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public:
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ReverbFrequencyResponse();
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~ReverbFrequencyResponse();
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// Updates the frequency response estimate of the reverb.
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void Update(const std::vector<std::array<float, kFftLengthBy2Plus1>>&
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frequency_response,
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int filter_delay_blocks,
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const absl::optional<float>& linear_filter_quality,
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bool stationary_block);
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// Returns the estimated frequency response for the reverb.
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rtc::ArrayView<const float> FrequencyResponse() const {
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return tail_response_;
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}
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private:
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void Update(const std::vector<std::array<float, kFftLengthBy2Plus1>>&
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frequency_response,
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int filter_delay_blocks,
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float linear_filter_quality);
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const bool enable_smooth_tail_response_updates_;
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float average_decay_ = 0.f;
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std::array<float, kFftLengthBy2Plus1> tail_response_;
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};
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} // namespace webrtc
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#endif // MODULES_AUDIO_PROCESSING_AEC3_REVERB_FREQUENCY_RESPONSE_H_
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