cf02cf13a7
This CL adds major render pipeline changes to the AEC3 code. The reason for these are that 1) It allows the echo removal unit to receive information about the content in bands beyond band 0, thereby allowing removal of high-frequency echoes 2) It allows more controlled handling of the render buffers, allowing proper buffer behaviour during capture glitches and clock-drift. Unfortunately, the render pipeline caused a lot of related changes in much of the rest of the AEC3 files. Most of these are, however, caused by a change of class name. Another unfortunate effect of this CL, is that a number of unittest cease to compile. I chose to temporarily solve that by removing them from the build using #if/#endif. The reason for that is that those will anyway again need to be changed in the next review, and doing like this avoids them having to be reviewed twice. BUG=webrtc:6018 Review-Url: https://codereview.webrtc.org/2784023002 Cr-Commit-Position: refs/heads/master@{#17547}
57 lines
2.0 KiB
C++
57 lines
2.0 KiB
C++
/*
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* Copyright (c) 2016 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 WEBRTC_MODULES_AUDIO_PROCESSING_AEC3_BLOCK_PROCESSOR_H_
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#define WEBRTC_MODULES_AUDIO_PROCESSING_AEC3_BLOCK_PROCESSOR_H_
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#include <memory>
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#include <vector>
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#include "webrtc/modules/audio_processing/aec3/echo_remover.h"
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#include "webrtc/modules/audio_processing/aec3/render_delay_buffer.h"
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#include "webrtc/modules/audio_processing/aec3/render_delay_controller.h"
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namespace webrtc {
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// Class for performing echo cancellation on 64 sample blocks of audio data.
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class BlockProcessor {
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public:
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static BlockProcessor* Create(int sample_rate_hz);
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// Only used for testing purposes.
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static BlockProcessor* Create(
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int sample_rate_hz,
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std::unique_ptr<RenderDelayBuffer> render_buffer);
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static BlockProcessor* Create(
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int sample_rate_hz,
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std::unique_ptr<RenderDelayBuffer> render_buffer,
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std::unique_ptr<RenderDelayController> delay_controller,
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std::unique_ptr<EchoRemover> echo_remover);
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virtual ~BlockProcessor() = default;
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// Processes a block of capture data.
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virtual void ProcessCapture(
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bool echo_path_gain_change,
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bool capture_signal_saturation,
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std::vector<std::vector<float>>* capture_block) = 0;
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// Buffers a block of render data supplied by a FrameBlocker object.
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virtual void BufferRender(
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const std::vector<std::vector<float>>& render_block) = 0;
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// Reports whether echo leakage has been detected in the echo canceller
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// output.
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virtual void UpdateEchoLeakageStatus(bool leakage_detected) = 0;
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};
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} // namespace webrtc
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#endif // WEBRTC_MODULES_AUDIO_PROCESSING_AEC3_BLOCK_PROCESSOR_H_
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