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/*
* Copyright (c) 2016 The WebRTC project authors. All Rights Reserved.
*
* Use of this source code is governed by a BSD-style license
* that can be found in the LICENSE file in the root of the source
* tree. An additional intellectual property rights grant can be found
* in the file PATENTS. All contributing project authors may
* be found in the AUTHORS file in the root of the source tree.
*/
#ifndef MODULES_AUDIO_PROCESSING_AEC3_BLOCK_FRAMER_H_
#define MODULES_AUDIO_PROCESSING_AEC3_BLOCK_FRAMER_H_
#include <vector>
#include "api/array_view.h"
#include "modules/audio_processing/aec3/aec3_common.h"
namespace webrtc {
// Class for producing frames consisting of 2 subframes of 80 samples each
// from 64 sample blocks. The class is designed to work together with the
// FrameBlocker class which performs the reverse conversion. Used together with
// that, this class produces output frames are the same rate as frames are
// received by the FrameBlocker class. Note that the internal buffers will
// overrun if any other rate of packets insertion is used.
class BlockFramer {
public:
BlockFramer(size_t num_bands, size_t num_channels);
~BlockFramer();
BlockFramer(const BlockFramer&) = delete;
BlockFramer& operator=(const BlockFramer&) = delete;
// Adds a 64 sample block into the data that will form the next output frame.
void InsertBlock(const std::vector<std::vector<std::vector<float>>>& block);
// Adds a 64 sample block and extracts an 80 sample subframe.
void InsertBlockAndExtractSubFrame(
const std::vector<std::vector<std::vector<float>>>& block,
std::vector<std::vector<rtc::ArrayView<float>>>* sub_frame);
private:
const size_t num_bands_;
const size_t num_channels_;
std::vector<std::vector<std::vector<float>>> buffer_;
};
} // namespace webrtc
#endif // MODULES_AUDIO_PROCESSING_AEC3_BLOCK_FRAMER_H_
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