From 26a029d407be480d791972afb5975cf62c9360a6 Mon Sep 17 00:00:00 2001 From: Daniel Baumann Date: Fri, 19 Apr 2024 02:47:55 +0200 Subject: Adding upstream version 124.0.1. Signed-off-by: Daniel Baumann --- .../agc2/fixed_digital_level_estimator.h | 66 ++++++++++++++++++++++ 1 file changed, 66 insertions(+) create mode 100644 third_party/libwebrtc/modules/audio_processing/agc2/fixed_digital_level_estimator.h (limited to 'third_party/libwebrtc/modules/audio_processing/agc2/fixed_digital_level_estimator.h') diff --git a/third_party/libwebrtc/modules/audio_processing/agc2/fixed_digital_level_estimator.h b/third_party/libwebrtc/modules/audio_processing/agc2/fixed_digital_level_estimator.h new file mode 100644 index 0000000000..d26b55950c --- /dev/null +++ b/third_party/libwebrtc/modules/audio_processing/agc2/fixed_digital_level_estimator.h @@ -0,0 +1,66 @@ +/* + * 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 +#include + +#include "modules/audio_processing/agc2/agc2_common.h" +#include "modules/audio_processing/include/audio_frame_view.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); + + FixedDigitalLevelEstimator(const FixedDigitalLevelEstimator&) = delete; + FixedDigitalLevelEstimator& operator=(const FixedDigitalLevelEstimator&) = + delete; + + // 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 ComputeLevel( + const AudioFrameView& 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_; +}; +} // namespace webrtc + +#endif // MODULES_AUDIO_PROCESSING_AGC2_FIXED_DIGITAL_LEVEL_ESTIMATOR_H_ -- cgit v1.2.3