| Index: webrtc/modules/audio_processing/aec3/render_delay_buffer.h
|
| diff --git a/webrtc/modules/audio_processing/aec3/render_delay_buffer.h b/webrtc/modules/audio_processing/aec3/render_delay_buffer.h
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| new file mode 100644
|
| index 0000000000000000000000000000000000000000..0fe2c9e677b8d5eeda95e91dfe2d17b6726785f6
|
| --- /dev/null
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| +++ b/webrtc/modules/audio_processing/aec3/render_delay_buffer.h
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| @@ -0,0 +1,57 @@
|
| +/*
|
| + * 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.
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| +class RenderDelayBuffer {
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| + public:
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| + static RenderDelayBuffer* Create(size_t size_blocks,
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| + size_t num_bands,
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| + 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.
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| + virtual bool Insert(std::vector<std::vector<float>>* block) = 0;
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| +
|
| + // Gets a reference to the next block (having the specified buffer delay) to
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| + // read in the buffer. This method can only be called if a block is
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| + // available which means that that must be checked prior to the call using
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| + // the method IsBlockAvailable().
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| + virtual const std::vector<std::vector<float>>& GetNext() = 0;
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| +
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| + // Sets the buffer delay. The delay set must be lower than the delay reported
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| + // by MaxDelay().
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| + virtual void SetDelay(size_t delay) = 0;
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| +
|
| + // Gets the buffer delay.
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| + virtual size_t Delay() const = 0;
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| +
|
| + // Returns the maximum allowed buffer delay increase.
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| + virtual size_t MaxDelay() const = 0;
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| +
|
| + // Returns whether a block is available for reading.
|
| + virtual bool IsBlockAvailable() const = 0;
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| +
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| + // Returns the maximum allowed api call jitter in blocks.
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| + virtual size_t MaxApiJitter() const = 0;
|
| +};
|
| +
|
| +} // namespace webrtc
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| +
|
| +#endif // WEBRTC_MODULES_AUDIO_PROCESSING_AEC3_RENDER_DELAY_BUFFER_H_
|
|
|