blob: 3341cd22d002427eb3935aa85ec081afc670bed5 [file] [log] [blame]
/*
* Copyright (c) 2017 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_GAIN_CONTROLLER2_H_
#define MODULES_AUDIO_PROCESSING_GAIN_CONTROLLER2_H_
#include <atomic>
#include <memory>
#include <string>
#include "modules/audio_processing/agc2/adaptive_digital_gain_controller.h"
#include "modules/audio_processing/agc2/cpu_features.h"
#include "modules/audio_processing/agc2/gain_applier.h"
#include "modules/audio_processing/agc2/input_volume_controller.h"
#include "modules/audio_processing/agc2/limiter.h"
#include "modules/audio_processing/agc2/vad_wrapper.h"
#include "modules/audio_processing/include/audio_processing.h"
#include "modules/audio_processing/logging/apm_data_dumper.h"
namespace webrtc {
class AudioBuffer;
// Gain Controller 2 aims to automatically adjust levels by acting on the
// microphone gain and/or applying digital gain.
class GainController2 {
public:
// Ctor. If `use_internal_vad` is true, an internal voice activity
// detector is used for digital adaptive gain.
GainController2(const AudioProcessing::Config::GainController2& config,
int sample_rate_hz,
int num_channels,
bool use_internal_vad);
GainController2(const GainController2&) = delete;
GainController2& operator=(const GainController2&) = delete;
~GainController2();
// Sets the fixed digital gain.
void SetFixedGainDb(float gain_db);
// Updates the input volume controller about whether the capture output is
// used or not.
void SetCaptureOutputUsed(bool capture_output_used);
// Analyzes `audio_buffer` before `Process()` is called so that the analysis
// can be performed before digital processing operations take place (e.g.,
// echo cancellation). The analysis consists of input clipping detection and
// prediction (if enabled). The value of `applied_input_volume` is limited to
// [0, 255].
void Analyze(int applied_input_volume, const AudioBuffer& audio_buffer);
// Applies fixed and adaptive digital gains to `audio` and runs a limiter.
// If the internal VAD is used, `speech_probability` is ignored. Otherwise
// `speech_probability` is used for digital adaptive gain if it's available
// (limited to values [0.0, 1.0]). Handles input volume changes; if the caller
// cannot determine whether an input volume change occurred, set
// `input_volume_changed` to false.
void Process(absl::optional<float> speech_probability,
bool input_volume_changed,
AudioBuffer* audio);
static bool Validate(const AudioProcessing::Config::GainController2& config);
AvailableCpuFeatures GetCpuFeatures() const { return cpu_features_; }
// Returns the recommended input volume if input volume controller is enabled
// and if a volume recommendation is available.
absl::optional<int> GetRecommendedInputVolume() const;
private:
static std::atomic<int> instance_count_;
const AvailableCpuFeatures cpu_features_;
ApmDataDumper data_dumper_;
GainApplier fixed_gain_applier_;
std::unique_ptr<VoiceActivityDetectorWrapper> vad_;
std::unique_ptr<AdaptiveDigitalGainController> adaptive_digital_controller_;
std::unique_ptr<InputVolumeController> input_volume_controller_;
Limiter limiter_;
int calls_since_last_limiter_log_;
};
} // namespace webrtc
#endif // MODULES_AUDIO_PROCESSING_GAIN_CONTROLLER2_H_