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This change introduces a clockdrift detector operating on the estimated delay of the echo path delay estimator. Each time the delay estimate changes it is compared to previous estimates. If the estimates are slowly increasing or decreasing, clockdrift is detected. Four different patterns are considered clockdrift: - k, k+1, k+2, k+3 - k, k+2, k+1, k+3 - k, k-1, k-2, k-3 - k, k-2, k-1, k-3 A delay estimate history matching the three last elements in one of the patterns is considered probable clockdrift. Matching all four elements is considered verified clockdrift. If the delay is constant for some time after clockdrift is detected the clockdrift detector will revert to no detected clockdrift. The level of clockdrift is reported via an UMA histogram. Bug: webrtc:10014 Change-Id: I1cce4d593e101a8b3fa99df6935e59b4243cb97a Reviewed-on: https://webrtc-review.googlesource.com/c/111381 Commit-Queue: Gustaf Ullberg <gustaf@webrtc.org> Reviewed-by: Per Åhgren <peah@webrtc.org> Cr-Commit-Position: refs/heads/master@{#25758}
53 lines
2.1 KiB
C++
53 lines
2.1 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 non_causal_offset,
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int sample_rate_hz);
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static RenderDelayController* Create2(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 absl::optional<int>& echo_remover_delay,
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rtc::ArrayView<const 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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