50 lines
1.8 KiB
C
50 lines
1.8 KiB
C
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/*
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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 "modules/audio_processing/aec3/block.h"
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namespace webrtc {
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// Class for producing frames consisting of 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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BlockFramer(size_t num_bands, size_t num_channels);
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~BlockFramer();
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BlockFramer(const BlockFramer&) = delete;
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BlockFramer& operator=(const BlockFramer&) = delete;
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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 Block& 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 Block& block,
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std::vector<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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const size_t num_channels_;
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std::vector<std::vector<std::vector<float>>> buffer_;
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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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