mirror of
https://github.com/mollyim/webrtc.git
synced 2025-05-19 16:47:50 +01:00

Add implementation for clipping prediction and clipped level step estimation. Bug: webrtc:12774 Change-Id: I855d22980302aac7d49078ca29755f9422af9cb5 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/220935 Commit-Queue: Hanna Silen <silen@webrtc.org> Reviewed-by: Minyue Li <minyue@webrtc.org> Cr-Commit-Position: refs/heads/master@{#34206}
70 lines
2.7 KiB
C++
70 lines
2.7 KiB
C++
/*
|
|
* 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_AGC_CLIPPING_PREDICTOR_H_
|
|
#define MODULES_AUDIO_PROCESSING_AGC_CLIPPING_PREDICTOR_H_
|
|
|
|
#include <memory>
|
|
#include <vector>
|
|
|
|
#include "absl/types/optional.h"
|
|
#include "modules/audio_processing/include/audio_frame_view.h"
|
|
#include "modules/audio_processing/include/audio_processing.h"
|
|
|
|
namespace webrtc {
|
|
|
|
// Frame-wise clipping prediction and clipped level step estimation. Processing
|
|
// is done in two steps: Calling `Process` analyses a frame of audio and stores
|
|
// the frame metrics and `EstimateClippedLevelStep` produces an estimate for the
|
|
// required analog gain level decrease if clipping is predicted.
|
|
class ClippingPredictor {
|
|
public:
|
|
virtual ~ClippingPredictor() = default;
|
|
|
|
virtual void Reset() = 0;
|
|
|
|
// Estimates the analog gain clipped level step for channel `channel`.
|
|
// Returns absl::nullopt if clipping is not predicted, otherwise returns the
|
|
// suggested decrease in the analog gain level.
|
|
virtual absl::optional<int> EstimateClippedLevelStep(
|
|
int channel,
|
|
int level,
|
|
int default_step,
|
|
int min_mic_level,
|
|
int max_mic_level) const = 0;
|
|
|
|
// Analyses a frame of audio and stores the resulting metrics in `data_`.
|
|
virtual void Process(const AudioFrameView<const float>& frame) = 0;
|
|
};
|
|
|
|
// Creates a ClippingPredictor based on crest factor-based clipping event
|
|
// prediction.
|
|
std::unique_ptr<ClippingPredictor> CreateClippingEventPredictor(
|
|
int num_channels,
|
|
const AudioProcessing::Config::GainController1 ::AnalogGainController::
|
|
ClippingPredictor& config);
|
|
|
|
// Creates a ClippingPredictor based on crest factor-based peak estimation and
|
|
// fixed-step clipped level step estimation.
|
|
std::unique_ptr<ClippingPredictor> CreateFixedStepClippingPeakPredictor(
|
|
int num_channels,
|
|
const AudioProcessing::Config::GainController1 ::AnalogGainController::
|
|
ClippingPredictor& config);
|
|
|
|
// Creates a ClippingPredictor based on crest factor-based peak estimation and
|
|
// adaptive-step clipped level step estimation.
|
|
std::unique_ptr<ClippingPredictor> CreateAdaptiveStepClippingPeakPredictor(
|
|
int num_channels,
|
|
const AudioProcessing::Config::GainController1 ::AnalogGainController::
|
|
ClippingPredictor& config);
|
|
|
|
} // namespace webrtc
|
|
|
|
#endif // MODULES_AUDIO_PROCESSING_AGC_CLIPPING_PREDICTOR_H_
|