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/*
* Copyright (c) 2021 The WebRTC project authors. All Rights Reserved.
*
* Use of this source code is governed by a BSD-style license
* that can be found in the LICENSE file in the root of the source
* tree. An additional intellectual property rights grant can be found
* in the file PATENTS. All contributing project authors may
* be found in the AUTHORS file in the root of the source tree.
*/
#ifndef MODULES_AUDIO_PROCESSING_AGC2_CLIPPING_PREDICTOR_H_
#define MODULES_AUDIO_PROCESSING_AGC2_CLIPPING_PREDICTOR_H_
#include <memory>
#include <optional>
#include <vector>
#include "api/audio/audio_processing.h"
#include "modules/audio_processing/include/audio_frame_view.h"
namespace webrtc {
// Frame-wise clipping prediction and clipped level step estimation. Analyzes
// 10 ms multi-channel frames and estimates an analog mic level decrease step
// to possibly avoid clipping when predicted. `Analyze()` and
// `EstimateClippedLevelStep()` can be called in any order.
class ClippingPredictor {
public:
virtual ~ClippingPredictor() = default;
virtual void Reset() = 0;
// Analyzes a 10 ms multi-channel audio frame.
virtual void Analyze(const AudioFrameView<const float>& frame) = 0;
// Predicts if clipping is going to occur for the specified `channel` in the
// near-future and, if so, it returns a recommended analog mic level decrease
// step. Returns std::nullopt if clipping is not predicted.
// `level` is the current analog mic level, `default_step` is the amount the
// mic level is lowered by the analog controller with every clipping event and
// `min_mic_level` and `max_mic_level` is the range of allowed analog mic
// levels.
virtual std::optional<int> EstimateClippedLevelStep(
int channel,
int level,
int default_step,
int min_mic_level,
int max_mic_level) const = 0;
};
// Creates a ClippingPredictor based on the provided `config`. When enabled,
// the following must hold for `config`:
// `window_length < reference_window_length + reference_window_delay`.
// Returns `nullptr` if `config.enabled` is false.
std::unique_ptr<ClippingPredictor> CreateClippingPredictor(
int num_channels,
const AudioProcessing::Config::GainController1::AnalogGainController::
ClippingPredictor& config);
} // namespace webrtc
#endif // MODULES_AUDIO_PROCESSING_AGC2_CLIPPING_PREDICTOR_H_
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