| /* |
| * 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_FIXED_DIGITAL_LEVEL_ESTIMATOR_H_ |
| #define MODULES_AUDIO_PROCESSING_AGC2_FIXED_DIGITAL_LEVEL_ESTIMATOR_H_ |
| |
| #include <array> |
| #include <vector> |
| |
| #include "modules/audio_processing/agc2/agc2_common.h" |
| #include "modules/audio_processing/include/audio_frame_view.h" |
| #include "rtc_base/constructor_magic.h" |
| |
| namespace webrtc { |
| |
| class ApmDataDumper; |
| // Produces a smooth signal level estimate from an input audio |
| // stream. The estimate smoothing is done through exponential |
| // filtering. |
| class FixedDigitalLevelEstimator { |
| public: |
| // Sample rates are allowed if the number of samples in a frame |
| // (sample_rate_hz * kFrameDurationMs / 1000) is divisible by |
| // kSubFramesInSample. For kFrameDurationMs=10 and |
| // kSubFramesInSample=20, this means that sample_rate_hz has to be |
| // divisible by 2000. |
| FixedDigitalLevelEstimator(int sample_rate_hz, |
| ApmDataDumper* apm_data_dumper); |
| |
| // The input is assumed to be in FloatS16 format. Scaled input will |
| // produce similarly scaled output. A frame of with kFrameDurationMs |
| // ms of audio produces a level estimates in the same scale. The |
| // level estimate contains kSubFramesInFrame values. |
| std::array<float, kSubFramesInFrame> ComputeLevel( |
| const AudioFrameView<const float>& float_frame); |
| |
| // Rate may be changed at any time (but not concurrently) from the |
| // value passed to the constructor. The class is not thread safe. |
| void SetSampleRate(int sample_rate_hz); |
| |
| // Resets the level estimator internal state. |
| void Reset(); |
| |
| float LastAudioLevel() const { return filter_state_level_; } |
| |
| private: |
| void CheckParameterCombination(); |
| |
| ApmDataDumper* const apm_data_dumper_ = nullptr; |
| float filter_state_level_; |
| int samples_in_frame_; |
| int samples_in_sub_frame_; |
| |
| RTC_DISALLOW_COPY_AND_ASSIGN(FixedDigitalLevelEstimator); |
| }; |
| } // namespace webrtc |
| |
| #endif // MODULES_AUDIO_PROCESSING_AGC2_FIXED_DIGITAL_LEVEL_ESTIMATOR_H_ |