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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"
+#include "modules/audio_processing/aec3/block.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 Block& block);
+ // Adds a 64 sample block and extracts an 80 sample subframe.
+ void InsertBlockAndExtractSubFrame(
+ const Block& 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_