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| 1 /* | |
| 2 * Copyright (c) 2017 The WebRTC project authors. All Rights Reserved. | |
| 3 * | |
| 4 * Use of this source code is governed by a BSD-style license | |
| 5 * that can be found in the LICENSE file in the root of the source | |
| 6 * tree. An additional intellectual property rights grant can be found | |
| 7 * in the file PATENTS. All contributing project authors may | |
| 8 * be found in the AUTHORS file in the root of the source tree. | |
| 9 */ | |
| 10 | |
| 11 #ifndef WEBRTC_MODULES_AUDIO_PROCESSING_AEC3_RENDER_DELAY_BUFFER_H_ | |
| 12 #define WEBRTC_MODULES_AUDIO_PROCESSING_AEC3_RENDER_DELAY_BUFFER_H_ | |
| 13 | |
| 14 #include <stddef.h> | |
| 15 #include <vector> | |
| 16 | |
| 17 namespace webrtc { | |
| 18 | |
| 19 // Class for buffering the incoming render blocks such that these may be | |
| 20 // extracted with a specified delay. | |
| 21 class RenderDelayBuffer { | |
| 22 public: | |
| 23 static RenderDelayBuffer* Create(size_t size_blocks, | |
| 24 size_t num_bands, | |
| 25 size_t max_api_jitter_blocks); | |
| 26 virtual ~RenderDelayBuffer() = default; | |
| 27 | |
| 28 // Swaps a block into the buffer (the content of block is destroyed) and | |
| 29 // returns true if the insert is successful. | |
| 30 virtual bool Insert(std::vector<std::vector<float>>* block) = 0; | |
| 31 | |
| 32 // Gets a reference to the next block (having the specified buffer delay) to | |
| 33 // read in the buffer. This method can only be called if a block is | |
| 34 // avalailable which means that that must be checked prior to the call using | |
|
hlundin-webrtc
2017/01/23 19:48:28
Close, but no cigar. You still need to drop some l
peah-webrtc
2017/01/23 21:38:22
Argh! Thanks!
Done.
| |
| 35 // the method IsBlockAvailable(). | |
| 36 virtual const std::vector<std::vector<float>>& GetNext() = 0; | |
| 37 | |
| 38 // Sets the buffer delay. The delay set must be lower than the delay reported | |
| 39 // by MaxDelay(). | |
| 40 virtual void SetDelay(size_t delay) = 0; | |
| 41 | |
| 42 // Gets the buffer delay. | |
| 43 virtual size_t Delay() const = 0; | |
| 44 | |
| 45 // Returns the maximum allowed buffer delay increase. | |
| 46 virtual size_t MaxDelay() const = 0; | |
| 47 | |
| 48 // Returns whether a block is available for reading. | |
| 49 virtual bool IsBlockAvailable() const = 0; | |
| 50 | |
| 51 // Returns the maximum allowed api call jitter in blocks. | |
| 52 virtual size_t MaxApiJitter() const = 0; | |
| 53 }; | |
| 54 | |
| 55 } // namespace webrtc | |
| 56 | |
| 57 #endif // WEBRTC_MODULES_AUDIO_PROCESSING_AEC3_RENDER_DELAY_BUFFER_H_ | |
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