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This CL utilizes the existing, but unused, ability to set different histogram thresholds for early and late delay estimation. It does so by tuning the parameters for these. On top of that, some corrections are added to correctly handle resets and the use of the hysteresis thresholds. Bug: webrtc:19886,chromium:896334 Change-Id: I950ac107c124541af8f02b4403f477dda71cc1a1 Reviewed-on: https://webrtc-review.googlesource.com/c/106706 Reviewed-by: Sam Zackrisson <saza@webrtc.org> Reviewed-by: Gustaf Ullberg <gustaf@webrtc.org> Commit-Queue: Per Åhgren <peah@webrtc.org> Cr-Commit-Position: refs/heads/master@{#25443}
50 lines
2 KiB
C++
50 lines
2 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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};
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} // namespace webrtc
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#endif // MODULES_AUDIO_PROCESSING_AEC3_RENDER_DELAY_CONTROLLER_H_
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