blob: 43b5828d35386847bea5902049d6ad600ef6a32e [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.
#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/noise_level_estimator.h"
#include "modules/audio_processing/agc2/saturation_protector.h"
#include "modules/audio_processing/agc2/speech_level_estimator.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 {
// Ctor. If `use_internal_vad` is true, an internal voice activity
// detector is used for digital adaptive gain.
const AudioProcessing::Config::GainController2& config,
const InputVolumeController::Config& input_volume_controller_config,
int sample_rate_hz,
int num_channels,
bool use_internal_vad);
GainController2(const GainController2&) = delete;
GainController2& operator=(const GainController2&) = delete;
// 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);
// Updates the recommended input volume, applies the adaptive digital and the
// fixed digital gains and runs a limiter on `audio`.
// When the internal VAD is not used, `speech_probability` should be specified
// and in the [0, 1] range. Otherwise ignores `speech_probability` and
// computes the speech probability via `vad_`.
// 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_; }
absl::optional<int> recommended_input_volume() const {
return recommended_input_volume_;
static std::atomic<int> instance_count_;
const AvailableCpuFeatures cpu_features_;
ApmDataDumper data_dumper_;
GainApplier fixed_gain_applier_;
std::unique_ptr<NoiseLevelEstimator> noise_level_estimator_;
std::unique_ptr<VoiceActivityDetectorWrapper> vad_;
std::unique_ptr<SpeechLevelEstimator> speech_level_estimator_;
std::unique_ptr<InputVolumeController> input_volume_controller_;
// TODO( Rename to `CrestFactorEstimator`.
std::unique_ptr<SaturationProtector> saturation_protector_;
std::unique_ptr<AdaptiveDigitalGainController> adaptive_digital_controller_;
Limiter limiter_;
int calls_since_last_limiter_log_;
// TODO( Remove intermediate storing at this level once
// APM refactoring is completed.
// Recommended input volume from `InputVolumecontroller`. Non-empty after
// `Process()` if input volume controller is enabled and
// `InputVolumeController::Process()` has returned a non-empty value.
absl::optional<int> recommended_input_volume_;
} // namespace webrtc