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d0fa820559
This CL adds support for using any externally reported audio buffer delay to set the initial alignment in AEC3 which is used before the AEC has been able to detect the delay. Bug: chromium:834182,webrtc:9163 Change-Id: Ic71355f69b7c4d5815b78e49987043441e7908fb Reviewed-on: https://webrtc-review.googlesource.com/70580 Reviewed-by: Gustaf Ullberg <gustaf@webrtc.org> Commit-Queue: Gustaf Ullberg <gustaf@webrtc.org> Commit-Queue: Per Åhgren <peah@webrtc.org> Cr-Commit-Position: refs/heads/master@{#22917}
65 lines
2.3 KiB
C++
65 lines
2.3 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 MODULES_AUDIO_PROCESSING_AEC3_BLOCK_PROCESSOR_H_
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#define MODULES_AUDIO_PROCESSING_AEC3_BLOCK_PROCESSOR_H_
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#include <memory>
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#include <vector>
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#include "modules/audio_processing/aec3/echo_remover.h"
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#include "modules/audio_processing/aec3/render_delay_buffer.h"
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#include "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(const EchoCanceller3Config& config,
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int sample_rate_hz);
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// Only used for testing purposes.
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static BlockProcessor* Create(
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const EchoCanceller3Config& config,
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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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const EchoCanceller3Config& config,
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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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// Get current metrics.
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virtual void GetMetrics(EchoControl::Metrics* metrics) const = 0;
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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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// 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 // MODULES_AUDIO_PROCESSING_AEC3_BLOCK_PROCESSOR_H_
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