/* * 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 #include 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>* 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>& 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_