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_FIXED_DIGITAL_LEVEL_ESTIMATOR_H_ 12 #define MODULES_AUDIO_PROCESSING_AGC2_FIXED_DIGITAL_LEVEL_ESTIMATOR_H_ 13 14 #include <array> 15 #include <vector> 16 17 #include "modules/audio_processing/agc2/agc2_common.h" 18 #include "modules/audio_processing/include/audio_frame_view.h" 19 #include "rtc_base/constructor_magic.h" 20 21 namespace webrtc { 22 23 class ApmDataDumper; 24 // Produces a smooth signal level estimate from an input audio 25 // stream. The estimate smoothing is done through exponential 26 // filtering. 27 class FixedDigitalLevelEstimator { 28 public: 29 // Sample rates are allowed if the number of samples in a frame 30 // (sample_rate_hz * kFrameDurationMs / 1000) is divisible by 31 // kSubFramesInSample. For kFrameDurationMs=10 and 32 // kSubFramesInSample=20, this means that sample_rate_hz has to be 33 // divisible by 2000. 34 FixedDigitalLevelEstimator(size_t sample_rate_hz, 35 ApmDataDumper* apm_data_dumper); 36 37 // The input is assumed to be in FloatS16 format. Scaled input will 38 // produce similarly scaled output. A frame of with kFrameDurationMs 39 // ms of audio produces a level estimates in the same scale. The 40 // level estimate contains kSubFramesInFrame values. 41 std::array<float, kSubFramesInFrame> ComputeLevel( 42 const AudioFrameView<const float>& float_frame); 43 44 // Rate may be changed at any time (but not concurrently) from the 45 // value passed to the constructor. The class is not thread safe. 46 void SetSampleRate(size_t sample_rate_hz); 47 48 // Resets the level estimator internal state. 49 void Reset(); 50 LastAudioLevel()51 float LastAudioLevel() const { return filter_state_level_; } 52 53 private: 54 void CheckParameterCombination(); 55 56 ApmDataDumper* const apm_data_dumper_ = nullptr; 57 float filter_state_level_; 58 size_t samples_in_frame_; 59 size_t samples_in_sub_frame_; 60 61 RTC_DISALLOW_COPY_AND_ASSIGN(FixedDigitalLevelEstimator); 62 }; 63 } // namespace webrtc 64 65 #endif // MODULES_AUDIO_PROCESSING_AGC2_FIXED_DIGITAL_LEVEL_ESTIMATOR_H_ 66