Adding second layer of the echo canceller 3 functionality.

webrtc/modules/audio_processing/aec3/render_delay_buffer.h

+/*
+ *  Copyright (c) 2017 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 WEBRTC_MODULES_AUDIO_PROCESSING_AEC3_RENDER_DELAY_BUFFER_H_
+#define WEBRTC_MODULES_AUDIO_PROCESSING_AEC3_RENDER_DELAY_BUFFER_H_
+
+#include <stddef.h>
+#include <vector>
+
+namespace webrtc {
+
+// Class for buffering the incoming render blocks such that these may be
+// extracted with a specified delay.
+class RenderDelayBuffer {
+ public:
+  static RenderDelayBuffer* Create(size_t size_blocks,
+                                   size_t num_bands,
+                                   size_t max_api_jitter_blocks);
+  virtual ~RenderDelayBuffer() = default;
+
+  // Swaps a block into the buffer (the content of block is destroyed) and
+  // returns true if the insert is successful.
+  virtual bool Insert(std::vector<std::vector<float>>* block) = 0;
+
+  // Gets a reference to the next block (having the specified buffer delay) to
+  // read in the buffer. This method can only be called if a block is
+  // available which means that that must be checked prior to the call using
+  // the method IsBlockAvailable().
+  virtual const std::vector<std::vector<float>>& GetNext() = 0;
+
+  // Sets the buffer delay. The delay set must be lower than the delay reported
+  // by MaxDelay().
+  virtual void SetDelay(size_t delay) = 0;
+
+  // Gets the buffer delay.
+  virtual size_t Delay() const = 0;
+
+  // Returns the maximum allowed buffer delay increase.
+  virtual size_t MaxDelay() const = 0;
+
+  // Returns whether a block is available for reading.
+  virtual bool IsBlockAvailable() const = 0;
+
+  // Returns the maximum allowed api call jitter in blocks.
+  virtual size_t MaxApiJitter() const = 0;
+};
+
+}  // namespace webrtc
+
+#endif  // WEBRTC_MODULES_AUDIO_PROCESSING_AEC3_RENDER_DELAY_BUFFER_H_