webrtc/modules/audio_processing/aec3/render_delay_buffer.h
Per Åhgren ce202a0f98 Reland "Reland "Add core multi-channel pipeline in AEC3 This CL adds basic the basic pipeline to support multi-channel processing in AEC3.""
This is a reland of a66395e72f

Original change's description:
> Reland "Add core multi-channel pipeline in AEC3 This CL adds basic the basic pipeline to support multi-channel processing in AEC3."
> 
> This is a reland of f3a197e553
> 
> Original change's description:
> > Add core multi-channel pipeline in AEC3
> > This CL adds basic the basic pipeline to support multi-channel
> > processing in AEC3.
> > 
> > Apart from that, it removes the 8 kHz processing support in several
> > places of the AEC3 code.
> > 
> > Bug: webrtc:10913
> > Change-Id: If5b75fa325ed0071deea94a7546cb4a7adf22137
> > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/150332
> > Commit-Queue: Per Åhgren <peah@webrtc.org>
> > Reviewed-by: Sam Zackrisson <saza@webrtc.org>
> > Cr-Commit-Position: refs/heads/master@{#29017}
> 
> Bug: webrtc:10913
> Change-Id: Ifc4b13bd994cfd22dca8f8755fa5700617cc379d
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/151124
> Reviewed-by: Sam Zackrisson <saza@webrtc.org>
> Commit-Queue: Per Åhgren <peah@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#29034}

Bug: webrtc:10913
Change-Id: Id8da5666df8c86f290c73ad5dc9958199f1a7ebe
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/151127
Commit-Queue: Sam Zackrisson <saza@webrtc.org>
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29042}
2019-09-03 06:12:32 +00:00

83 lines
2.7 KiB
C++

/*
* 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 MODULES_AUDIO_PROCESSING_AEC3_RENDER_DELAY_BUFFER_H_
#define MODULES_AUDIO_PROCESSING_AEC3_RENDER_DELAY_BUFFER_H_
#include <stddef.h>
#include <vector>
#include "api/audio/echo_canceller3_config.h"
#include "modules/audio_processing/aec3/downsampled_render_buffer.h"
#include "modules/audio_processing/aec3/render_buffer.h"
namespace webrtc {
// Class for buffering the incoming render blocks such that these may be
// extracted with a specified delay.
class RenderDelayBuffer {
public:
enum class BufferingEvent {
kNone,
kRenderUnderrun,
kRenderOverrun,
kApiCallSkew
};
static RenderDelayBuffer* Create(const EchoCanceller3Config& config,
int sample_rate_hz,
size_t num_render_channels);
virtual ~RenderDelayBuffer() = default;
// Resets the buffer alignment.
virtual void Reset() = 0;
// Inserts a block into the buffer.
virtual BufferingEvent Insert(
const std::vector<std::vector<std::vector<float>>>& block) = 0;
// Updates the buffers one step based on the specified buffer delay. Returns
// an enum indicating whether there was a special event that occurred.
virtual BufferingEvent PrepareCaptureProcessing() = 0;
// Sets the buffer delay and returns a bool indicating whether the delay
// changed.
virtual bool AlignFromDelay(size_t delay) = 0;
// Sets the buffer delay from the most recently reported external delay.
virtual void AlignFromExternalDelay() = 0;
// Gets the buffer delay.
virtual size_t Delay() const = 0;
// Gets the buffer delay.
virtual size_t MaxDelay() const = 0;
// Returns the render buffer for the echo remover.
virtual RenderBuffer* GetRenderBuffer() = 0;
// Returns the downsampled render buffer.
virtual const DownsampledRenderBuffer& GetDownsampledRenderBuffer() const = 0;
// Returns the maximum non calusal offset that can occur in the delay buffer.
static int DelayEstimatorOffset(const EchoCanceller3Config& config);
// Provides an optional external estimate of the audio buffer delay.
virtual void SetAudioBufferDelay(size_t delay_ms) = 0;
// Returns whether an external delay estimate has been reported via
// SetAudioBufferDelay.
virtual bool HasReceivedBufferDelay() = 0;
};
} // namespace webrtc
#endif // MODULES_AUDIO_PROCESSING_AEC3_RENDER_DELAY_BUFFER_H_