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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.
46 lines
1.4 KiB
C++
46 lines
1.4 KiB
C++
/*
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* Copyright (c) 2013 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_TRANSIENT_WPD_NODE_H_
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#define MODULES_AUDIO_PROCESSING_TRANSIENT_WPD_NODE_H_
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#include <memory>
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namespace webrtc {
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class FIRFilter;
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// A single node of a Wavelet Packet Decomposition (WPD) tree.
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class WPDNode {
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public:
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// Creates a WPDNode. The data vector will contain zeros. The filter will have
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// the coefficients provided.
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WPDNode(size_t length, const float* coefficients, size_t coefficients_length);
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~WPDNode();
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// Updates the node data. |parent_data| / 2 must be equals to |length_|.
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// Returns 0 if correct, and -1 otherwise.
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int Update(const float* parent_data, size_t parent_data_length);
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const float* data() const { return data_.get(); }
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// Returns 0 if correct, and -1 otherwise.
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int set_data(const float* new_data, size_t length);
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size_t length() const { return length_; }
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private:
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std::unique_ptr<float[]> data_;
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size_t length_;
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std::unique_ptr<FIRFilter> filter_;
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};
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} // namespace webrtc
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#endif // MODULES_AUDIO_PROCESSING_TRANSIENT_WPD_NODE_H_
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