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https://github.com/danog/libtgvoip.git
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5caaaafa42
I'm now using the entire audio processing module from WebRTC as opposed to individual DSP algorithms pulled from there before. Seems to work better this way.
80 lines
2.6 KiB
C++
80 lines
2.6 KiB
C++
/*
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* Copyright (c) 2017 The WebRTC project authors. All Rights Reserved.
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*
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* Use of this source code is governed by a BSD-style license
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* that can be found in the LICENSE file in the root of the source
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* tree. An additional intellectual property rights grant can be found
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* in the file PATENTS. All contributing project authors may
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* be found in the AUTHORS file in the root of the source tree.
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*/
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#ifndef MODULES_AUDIO_PROCESSING_AEC3_RENDER_DELAY_BUFFER_H_
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#define MODULES_AUDIO_PROCESSING_AEC3_RENDER_DELAY_BUFFER_H_
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#include <stddef.h>
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#include <vector>
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#include "api/audio/echo_canceller3_config.h"
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#include "modules/audio_processing/aec3/downsampled_render_buffer.h"
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#include "modules/audio_processing/aec3/render_buffer.h"
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namespace webrtc {
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// Class for buffering the incoming render blocks such that these may be
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// extracted with a specified delay.
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class RenderDelayBuffer {
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public:
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enum class BufferingEvent {
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kNone,
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kRenderUnderrun,
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kRenderOverrun,
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kApiCallSkew
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};
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static RenderDelayBuffer* Create(const EchoCanceller3Config& config,
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size_t num_bands);
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static RenderDelayBuffer* Create2(const EchoCanceller3Config& config,
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size_t num_bands);
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virtual ~RenderDelayBuffer() = default;
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// Resets the buffer alignment.
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virtual void Reset() = 0;
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// Inserts a block into the buffer.
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virtual BufferingEvent Insert(
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const std::vector<std::vector<float>>& block) = 0;
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// Updates the buffers one step based on the specified buffer delay. Returns
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// an enum indicating whether there was a special event that occurred.
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virtual BufferingEvent PrepareCaptureProcessing() = 0;
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// Sets the buffer delay and returns a bool indicating whether the delay
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// changed.
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virtual bool 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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// Gets the buffer delay.
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virtual size_t MaxDelay() const = 0;
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// Returns the render buffer for the echo remover.
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virtual RenderBuffer* GetRenderBuffer() = 0;
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// Returns the downsampled render buffer.
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virtual const DownsampledRenderBuffer& GetDownsampledRenderBuffer() const = 0;
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// Returns whether the current delay is noncausal.
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virtual bool CausalDelay(size_t delay) const = 0;
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// Returns the maximum non calusal offset that can occur in the delay buffer.
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static int DelayEstimatorOffset(const EchoCanceller3Config& config);
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// Provides an optional external estimate of the audio buffer delay.
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virtual void SetAudioBufferDelay(size_t delay_ms) = 0;
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};
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} // namespace webrtc
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#endif // MODULES_AUDIO_PROCESSING_AEC3_RENDER_DELAY_BUFFER_H_
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