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ee84d39fce
Multichannel signals are downmixed to mono before decimation and delay estimation. This is useful when not all channels play audio content. The feature can be toggled in the AEC3 configuration. Bug: webrtc:10913 Change-Id: I7d40edf7732bb51fec69e7f3ca063d821c5069c4 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/151762 Commit-Queue: Gustaf Ullberg <gustaf@webrtc.org> Reviewed-by: Per Åhgren <peah@webrtc.org> Cr-Commit-Position: refs/heads/master@{#29126}
49 lines
1.9 KiB
C++
49 lines
1.9 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_CONTROLLER_H_
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#define MODULES_AUDIO_PROCESSING_AEC3_RENDER_DELAY_CONTROLLER_H_
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#include "absl/types/optional.h"
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#include "api/array_view.h"
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#include "api/audio/echo_canceller3_config.h"
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#include "modules/audio_processing/aec3/delay_estimate.h"
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#include "modules/audio_processing/aec3/downsampled_render_buffer.h"
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#include "modules/audio_processing/aec3/render_delay_buffer.h"
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#include "modules/audio_processing/logging/apm_data_dumper.h"
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namespace webrtc {
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// Class for aligning the render and capture signal using a RenderDelayBuffer.
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class RenderDelayController {
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public:
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static RenderDelayController* Create(const EchoCanceller3Config& config,
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int sample_rate_hz);
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virtual ~RenderDelayController() = default;
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// Resets the delay controller. If the delay confidence is reset, the reset
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// behavior is as if the call is restarted.
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virtual void Reset(bool reset_delay_confidence) = 0;
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// Logs a render call.
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virtual void LogRenderCall() = 0;
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// Aligns the render buffer content with the capture signal.
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virtual absl::optional<DelayEstimate> GetDelay(
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const DownsampledRenderBuffer& render_buffer,
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size_t render_delay_buffer_delay,
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const std::vector<std::vector<float>>& capture) = 0;
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// Returns true if clockdrift has been detected.
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virtual bool HasClockdrift() const = 0;
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};
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} // namespace webrtc
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#endif // MODULES_AUDIO_PROCESSING_AEC3_RENDER_DELAY_CONTROLLER_H_
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