| /* |
| * 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(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_; |
| } |
| |
| 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. |
| absl::optional<int> recommended_input_volume_; |
| }; |
| |
| } // namespace webrtc |
| |
| #endif // MODULES_AUDIO_PROCESSING_GAIN_CONTROLLER2_H_ |