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6a05bb1b12
This CL adds code for doing signal-dependent downmixing before the delay estimation in the multichannel case. As part of the CL, the unittests of the render delay controller are corrected. However, as that caused some of them to fail, the CL (for now) as well disables the failing test. Bug: webrtc:11153,chromium:1029740, webrtc:11161 Change-Id: I0b765c28fa5e547aabd6dfbd24b626ff9a16346f Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/161045 Commit-Queue: Per Åhgren <peah@webrtc.org> Reviewed-by: Sam Zackrisson <saza@webrtc.org> Cr-Commit-Position: refs/heads/master@{#29980}
50 lines
1.9 KiB
C++
50 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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size_t num_capture_channels);
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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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