blob: 6d12339888c2e8400513c3bf4e02202b6bc1826b [file] [log] [blame]
* Copyright (c) 2018 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.
#include <stddef.h>
#include <stdint.h>
#include "modules/audio_processing/agc/agc.h"
#include "modules/audio_processing/agc2/adaptive_mode_level_estimator.h"
#include "modules/audio_processing/agc2/saturation_protector.h"
#include "modules/audio_processing/agc2/vad_with_level.h"
namespace webrtc {
class AdaptiveModeLevelEstimatorAgc : public Agc {
explicit AdaptiveModeLevelEstimatorAgc(ApmDataDumper* apm_data_dumper);
// |audio| must be mono; in a multi-channel stream, provide the first (usually
// left) channel.
void Process(const int16_t* audio,
size_t length,
int sample_rate_hz) override;
// Retrieves the difference between the target RMS level and the current
// signal RMS level in dB. Returns true if an update is available and false
// otherwise, in which case |error| should be ignored and no action taken.
bool GetRmsErrorDb(int* error) override;
void Reset() override;
float voice_probability() const override;
static constexpr int kTimeUntilConfidentMs = 700;
static constexpr int kDefaultAgc2LevelHeadroomDbfs = -1;
int32_t time_in_ms_since_last_estimate_ = 0;
AdaptiveModeLevelEstimator level_estimator_;
VadWithLevel agc2_vad_;
float latest_voice_probability_ = 0.f;
} // namespace webrtc