b5c48b97f6
Ongoing fixes and improvements, transient suppressor is gone. Also, dropping isac because it doesn't seem to be useful, and is just build system deadweight now. Upstream references: Version: 131.0.6778.200 WebRTC: 79aff54b0fa9238ce3518dd9eaf9610cd6f22e82 Chromium: 2a19506ad24af755f2a215a4c61f775393e0db42
112 lines
4.5 KiB
C++
112 lines
4.5 KiB
C++
/*
|
|
* 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 "api/audio/audio_processing.h"
|
|
#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/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,
|
|
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;
|
|
~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);
|
|
|
|
// 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.
|
|
// TODO(bugs.webrtc.org/7494): Remove `speech_probability`.
|
|
void Process(std::optional<float> speech_probability,
|
|
bool input_volume_changed,
|
|
AudioBuffer* audio);
|
|
|
|
static bool Validate(const AudioProcessing::Config::GainController2& config);
|
|
|
|
AvailableCpuFeatures GetCpuFeatures() const { return cpu_features_; }
|
|
|
|
std::optional<int> recommended_input_volume() const {
|
|
return recommended_input_volume_;
|
|
}
|
|
|
|
private:
|
|
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(bugs.webrtc.org/7494): 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(bugs.webrtc.org/7494): 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.
|
|
std::optional<int> recommended_input_volume_;
|
|
};
|
|
|
|
} // namespace webrtc
|
|
|
|
#endif // MODULES_AUDIO_PROCESSING_GAIN_CONTROLLER2_H_
|