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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.
48 lines
1.7 KiB
C++
48 lines
1.7 KiB
C++
/*
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* Copyright (c) 2016 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_BLOCK_FRAMER_H_
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#define MODULES_AUDIO_PROCESSING_AEC3_BLOCK_FRAMER_H_
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#include <vector>
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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 "rtc_base/constructormagic.h"
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namespace webrtc {
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// Class for producing frames consisting of 1 or 2 subframes of 80 samples each
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// from 64 sample blocks. The class is designed to work together with the
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// FrameBlocker class which performs the reverse conversion. Used together with
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// that, this class produces output frames are the same rate as frames are
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// received by the FrameBlocker class. Note that the internal buffers will
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// overrun if any other rate of packets insertion is used.
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class BlockFramer {
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public:
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explicit BlockFramer(size_t num_bands);
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~BlockFramer();
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// Adds a 64 sample block into the data that will form the next output frame.
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void InsertBlock(const std::vector<std::vector<float>>& block);
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// Adds a 64 sample block and extracts an 80 sample subframe.
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void InsertBlockAndExtractSubFrame(
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const std::vector<std::vector<float>>& block,
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std::vector<rtc::ArrayView<float>>* sub_frame);
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private:
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const size_t num_bands_;
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std::vector<std::vector<float>> buffer_;
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RTC_DISALLOW_IMPLICIT_CONSTRUCTORS(BlockFramer);
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};
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} // namespace webrtc
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#endif // MODULES_AUDIO_PROCESSING_AEC3_BLOCK_FRAMER_H_
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