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d7cfbe3843
Add InputVolumeController as a member in GainController2 (not created by default). Add a method GainController2::Analyze() to update the applied input volume and run the pre-processing steps in InputVolumeController. Add a call InputVolumeController::Process() in GainController2::Process(). Bug: webrtc:7494 Change-Id: Idf4111ac5e19a620b6421c7f23fd642f169c7b5a Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/279822 Reviewed-by: Per Åhgren <peah@webrtc.org> Reviewed-by: Alessio Bazzica <alessiob@webrtc.org> Commit-Queue: Hanna Silen <silen@webrtc.org> Cr-Commit-Position: refs/heads/main@{#38548}
91 lines
3.6 KiB
C++
91 lines
3.6 KiB
C++
/*
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* Copyright (c) 2017 The WebRTC project authors. All Rights Reserved.
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*
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* Use of this source code is governed by a BSD-style license
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* that can be found in the LICENSE file in the root of the source
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* tree. An additional intellectual property rights grant can be found
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* in the file PATENTS. All contributing project authors may
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* be found in the AUTHORS file in the root of the source tree.
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*/
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#ifndef MODULES_AUDIO_PROCESSING_GAIN_CONTROLLER2_H_
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#define MODULES_AUDIO_PROCESSING_GAIN_CONTROLLER2_H_
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#include <atomic>
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#include <memory>
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#include <string>
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#include "modules/audio_processing/agc2/adaptive_digital_gain_controller.h"
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#include "modules/audio_processing/agc2/cpu_features.h"
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#include "modules/audio_processing/agc2/gain_applier.h"
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#include "modules/audio_processing/agc2/input_volume_controller.h"
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#include "modules/audio_processing/agc2/limiter.h"
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#include "modules/audio_processing/agc2/vad_wrapper.h"
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#include "modules/audio_processing/include/audio_processing.h"
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#include "modules/audio_processing/logging/apm_data_dumper.h"
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namespace webrtc {
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class AudioBuffer;
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// Gain Controller 2 aims to automatically adjust levels by acting on the
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// microphone gain and/or applying digital gain.
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class GainController2 {
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public:
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// Ctor. If `use_internal_vad` is true, an internal voice activity
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// detector is used for digital adaptive gain.
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GainController2(const AudioProcessing::Config::GainController2& config,
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int sample_rate_hz,
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int num_channels,
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bool use_internal_vad);
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GainController2(const GainController2&) = delete;
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GainController2& operator=(const GainController2&) = delete;
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~GainController2();
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// Sets the fixed digital gain.
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void SetFixedGainDb(float gain_db);
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// Updates the input volume controller about whether the capture output is
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// used or not.
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void SetCaptureOutputUsed(bool capture_output_used);
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// Analyzes `audio_buffer` before `Process()` is called so that the analysis
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// can be performed before digital processing operations take place (e.g.,
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// echo cancellation). The analysis consists of input clipping detection and
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// prediction (if enabled). The value of `applied_input_volume` is limited to
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// [0, 255].
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void Analyze(int applied_input_volume, const AudioBuffer& audio_buffer);
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// Applies fixed and adaptive digital gains to `audio` and runs a limiter.
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// If the internal VAD is used, `speech_probability` is ignored. Otherwise
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// `speech_probability` is used for digital adaptive gain if it's available
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// (limited to values [0.0, 1.0]). Handles input volume changes; if the caller
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// cannot determine whether an input volume change occurred, set
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// `input_volume_changed` to false.
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void Process(absl::optional<float> speech_probability,
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bool input_volume_changed,
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AudioBuffer* audio);
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static bool Validate(const AudioProcessing::Config::GainController2& config);
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AvailableCpuFeatures GetCpuFeatures() const { return cpu_features_; }
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// Returns the recommended input volume if input volume controller is enabled
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// and if a volume recommendation is available.
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absl::optional<int> GetRecommendedInputVolume() const;
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private:
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static std::atomic<int> instance_count_;
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const AvailableCpuFeatures cpu_features_;
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ApmDataDumper data_dumper_;
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GainApplier fixed_gain_applier_;
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std::unique_ptr<VoiceActivityDetectorWrapper> vad_;
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std::unique_ptr<AdaptiveDigitalGainController> adaptive_digital_controller_;
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std::unique_ptr<InputVolumeController> input_volume_controller_;
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Limiter limiter_;
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int calls_since_last_limiter_log_;
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};
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} // namespace webrtc
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#endif // MODULES_AUDIO_PROCESSING_GAIN_CONTROLLER2_H_
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