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- /*
- * Copyright (c) 2018 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_AGC2_ADAPTIVE_AGC_H_
- #define MODULES_AUDIO_PROCESSING_AGC2_ADAPTIVE_AGC_H_
- #include "modules/audio_processing/agc2/adaptive_digital_gain_applier.h"
- #include "modules/audio_processing/agc2/adaptive_mode_level_estimator.h"
- #include "modules/audio_processing/agc2/noise_level_estimator.h"
- #include "modules/audio_processing/agc2/vad_with_level.h"
- #include "modules/audio_processing/include/audio_frame_view.h"
- #include "modules/audio_processing/include/audio_processing.h"
- namespace webrtc {
- class ApmDataDumper;
- // Adaptive digital gain controller.
- // TODO(crbug.com/webrtc/7494): Unify with `AdaptiveDigitalGainApplier`.
- class AdaptiveAgc {
- public:
- explicit AdaptiveAgc(ApmDataDumper* apm_data_dumper);
- // TODO(crbug.com/webrtc/7494): Remove ctor above.
- AdaptiveAgc(ApmDataDumper* apm_data_dumper,
- const AudioProcessing::Config::GainController2& config);
- ~AdaptiveAgc();
- // Analyzes `frame` and applies a digital adaptive gain to it. Takes into
- // account the envelope measured by the limiter.
- // TODO(crbug.com/webrtc/7494): Make the class depend on the limiter.
- void Process(AudioFrameView<float> frame, float limiter_envelope);
- void Reset();
- private:
- AdaptiveModeLevelEstimator speech_level_estimator_;
- VadLevelAnalyzer vad_;
- AdaptiveDigitalGainApplier gain_applier_;
- ApmDataDumper* const apm_data_dumper_;
- NoiseLevelEstimator noise_level_estimator_;
- };
- } // namespace webrtc
- #endif // MODULES_AUDIO_PROCESSING_AGC2_ADAPTIVE_AGC_H_
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