Bump to WebRTC M120 release
Some API deprecation -- ExperimentalAgc and ExperimentalNs are gone. We're continuing to carry iSAC even though it's gone upstream, but maybe we'll want to drop that soon.
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webrtc/modules/audio_processing/agc2/clipping_predictor.h
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webrtc/modules/audio_processing/agc2/clipping_predictor.h
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/*
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* Copyright (c) 2021 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_AGC2_CLIPPING_PREDICTOR_H_
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#define MODULES_AUDIO_PROCESSING_AGC2_CLIPPING_PREDICTOR_H_
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#include <memory>
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#include <vector>
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#include "absl/types/optional.h"
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#include "modules/audio_processing/include/audio_frame_view.h"
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#include "modules/audio_processing/include/audio_processing.h"
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namespace webrtc {
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// Frame-wise clipping prediction and clipped level step estimation. Analyzes
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// 10 ms multi-channel frames and estimates an analog mic level decrease step
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// to possibly avoid clipping when predicted. `Analyze()` and
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// `EstimateClippedLevelStep()` can be called in any order.
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class ClippingPredictor {
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public:
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virtual ~ClippingPredictor() = default;
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virtual void Reset() = 0;
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// Analyzes a 10 ms multi-channel audio frame.
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virtual void Analyze(const AudioFrameView<const float>& frame) = 0;
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// Predicts if clipping is going to occur for the specified `channel` in the
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// near-future and, if so, it returns a recommended analog mic level decrease
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// step. Returns absl::nullopt if clipping is not predicted.
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// `level` is the current analog mic level, `default_step` is the amount the
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// mic level is lowered by the analog controller with every clipping event and
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// `min_mic_level` and `max_mic_level` is the range of allowed analog mic
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// levels.
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virtual absl::optional<int> EstimateClippedLevelStep(
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int channel,
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int level,
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int default_step,
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int min_mic_level,
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int max_mic_level) const = 0;
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};
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// Creates a ClippingPredictor based on the provided `config`. When enabled,
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// the following must hold for `config`:
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// `window_length < reference_window_length + reference_window_delay`.
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// Returns `nullptr` if `config.enabled` is false.
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std::unique_ptr<ClippingPredictor> CreateClippingPredictor(
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int num_channels,
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const AudioProcessing::Config::GainController1::AnalogGainController::
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ClippingPredictor& config);
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} // namespace webrtc
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#endif // MODULES_AUDIO_PROCESSING_AGC2_CLIPPING_PREDICTOR_H_
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