webrtc/modules/audio_processing/aec3/block_processor.h
Per Åhgren 8ee1ec82e4 Allowing reduced computations in the AEC3 when the output is not used
This CL adds functionality in AEC3 that allows the computational
complexity to be reduced when the output of APM is not used.

Bug: b/177830919
Change-Id: I08121364bf966f34311f54ffa5affbfd8b4db1e2
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/211341
Reviewed-by: Gustaf Ullberg <gustaf@webrtc.org>
Commit-Queue: Per Åhgren <peah@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33476}
2021-03-16 09:16:32 +00:00

82 lines
3 KiB
C++

/*
* Copyright (c) 2016 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_BLOCK_PROCESSOR_H_
#define MODULES_AUDIO_PROCESSING_AEC3_BLOCK_PROCESSOR_H_
#include <stddef.h>
#include <memory>
#include <vector>
#include "api/audio/echo_canceller3_config.h"
#include "api/audio/echo_control.h"
#include "modules/audio_processing/aec3/echo_remover.h"
#include "modules/audio_processing/aec3/render_delay_buffer.h"
#include "modules/audio_processing/aec3/render_delay_controller.h"
namespace webrtc {
// Class for performing echo cancellation on 64 sample blocks of audio data.
class BlockProcessor {
public:
static BlockProcessor* Create(const EchoCanceller3Config& config,
int sample_rate_hz,
size_t num_render_channels,
size_t num_capture_channels);
// Only used for testing purposes.
static BlockProcessor* Create(
const EchoCanceller3Config& config,
int sample_rate_hz,
size_t num_render_channels,
size_t num_capture_channels,
std::unique_ptr<RenderDelayBuffer> render_buffer);
static BlockProcessor* Create(
const EchoCanceller3Config& config,
int sample_rate_hz,
size_t num_render_channels,
size_t num_capture_channels,
std::unique_ptr<RenderDelayBuffer> render_buffer,
std::unique_ptr<RenderDelayController> delay_controller,
std::unique_ptr<EchoRemover> echo_remover);
virtual ~BlockProcessor() = default;
// Get current metrics.
virtual void GetMetrics(EchoControl::Metrics* metrics) const = 0;
// Provides an optional external estimate of the audio buffer delay.
virtual void SetAudioBufferDelay(int delay_ms) = 0;
// Processes a block of capture data.
virtual void ProcessCapture(
bool echo_path_gain_change,
bool capture_signal_saturation,
std::vector<std::vector<std::vector<float>>>* linear_output,
std::vector<std::vector<std::vector<float>>>* capture_block) = 0;
// Buffers a block of render data supplied by a FrameBlocker object.
virtual void BufferRender(
const std::vector<std::vector<std::vector<float>>>& render_block) = 0;
// Reports whether echo leakage has been detected in the echo canceller
// output.
virtual void UpdateEchoLeakageStatus(bool leakage_detected) = 0;
// Specifies whether the capture output will be used. The purpose of this is
// to allow the block processor to deactivate some of the processing when the
// resulting output is anyway not used, for instance when the endpoint is
// muted.
virtual void SetCaptureOutputUsage(bool capture_output_used) = 0;
};
} // namespace webrtc
#endif // MODULES_AUDIO_PROCESSING_AEC3_BLOCK_PROCESSOR_H_