1 /* 2 * Copyright (c) 2018 The WebRTC project authors. All Rights Reserved. 3 * 4 * Use of this source code is governed by a BSD-style license 5 * that can be found in the LICENSE file in the root of the source 6 * tree. An additional intellectual property rights grant can be found 7 * in the file PATENTS. All contributing project authors may 8 * be found in the AUTHORS file in the root of the source tree. 9 */ 10 11 #ifndef MODULES_AUDIO_PROCESSING_AGC2_ADAPTIVE_DIGITAL_GAIN_APPLIER_H_ 12 #define MODULES_AUDIO_PROCESSING_AGC2_ADAPTIVE_DIGITAL_GAIN_APPLIER_H_ 13 14 #include <vector> 15 16 #include "modules/audio_processing/agc2/gain_applier.h" 17 #include "modules/audio_processing/include/audio_frame_view.h" 18 #include "modules/audio_processing/include/audio_processing.h" 19 20 namespace webrtc { 21 22 class ApmDataDumper; 23 24 // TODO(bugs.webrtc.org/7494): Split into `GainAdaptor` and `GainApplier`. 25 // Selects the target digital gain, decides when and how quickly to adapt to the 26 // target and applies the current gain to 10 ms frames. 27 class AdaptiveDigitalGainApplier { 28 public: 29 // Information about a frame to process. 30 struct FrameInfo { 31 float speech_probability; // Probability of speech in the [0, 1] range. 32 float speech_level_dbfs; // Estimated speech level (dBFS). 33 bool speech_level_reliable; // True with reliable speech level estimation. 34 float noise_rms_dbfs; // Estimated noise RMS level (dBFS). 35 float headroom_db; // Headroom (dB). 36 float limiter_envelope_dbfs; // Envelope level from the limiter (dBFS). 37 }; 38 39 AdaptiveDigitalGainApplier( 40 ApmDataDumper* apm_data_dumper, 41 const AudioProcessing::Config::GainController2::AdaptiveDigital& config, 42 int sample_rate_hz, 43 int num_channels); 44 AdaptiveDigitalGainApplier(const AdaptiveDigitalGainApplier&) = delete; 45 AdaptiveDigitalGainApplier& operator=(const AdaptiveDigitalGainApplier&) = 46 delete; 47 48 void Initialize(int sample_rate_hz, int num_channels); 49 50 // Analyzes `info`, updates the digital gain and applies it to a 10 ms 51 // `frame`. Supports any sample rate supported by APM. 52 void Process(const FrameInfo& info, AudioFrameView<float> frame); 53 54 private: 55 ApmDataDumper* const apm_data_dumper_; 56 GainApplier gain_applier_; 57 58 const AudioProcessing::Config::GainController2::AdaptiveDigital config_; 59 const float max_gain_change_db_per_10ms_; 60 61 int calls_since_last_gain_log_; 62 int frames_to_gain_increase_allowed_; 63 float last_gain_db_; 64 65 std::vector<std::vector<float>> dry_run_frame_; 66 std::vector<float*> dry_run_channels_; 67 }; 68 69 } // namespace webrtc 70 71 #endif // MODULES_AUDIO_PROCESSING_AGC2_ADAPTIVE_DIGITAL_GAIN_APPLIER_H_ 72