Michael Froman
29783a6209
Upstream commit: https://webrtc.googlesource.com/src/+/a80152e9386192c7465ca123f31e6db5b13b079c Remove speech_probability as input parameter to AGC2 Since the VAD module has been removed from audio processing module (APM) some time ago, there is no need to pass the speech probability as an input parameter. The speech probability is now computed inside AGC2. Bug: webrtc:42232605 Change-Id: If4b25bdfb5c8961d4acc81a8440d4be9c8570973 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/421160 Reviewed-by: Gustaf Ullberg <gustaf@webrtc.org> Commit-Queue: Lionel Koenig <lionelk@webrtc.org> Cr-Commit-Position: refs/heads/main@{#46065}
111 lines
4.5 KiB
C++
111 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 <optional>
|
|
|
|
#include "api/audio/audio_processing.h"
|
|
#include "api/environment/environment.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 Environment& env,
|
|
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.
|
|
void Process(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_
|