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7f30dd11eb
follow up to https://webrtc-review.googlesource.com/c/src/+/352582 Bug: chromium:335805780 Change-Id: I47f2842da9e86b686e3a3c2f4f28fa03d1cd297d Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/356241 Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org> Reviewed-by: Sam Zackrisson <saza@webrtc.org> Cr-Commit-Position: refs/heads/main@{#42589}
70 lines
2.6 KiB
C++
70 lines
2.6 KiB
C++
/*
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* Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
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*
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* Use of this source code is governed by a BSD-style license
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* that can be found in the LICENSE file in the root of the source
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* tree. An additional intellectual property rights grant can be found
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* in the file PATENTS. All contributing project authors may
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* be found in the AUTHORS file in the root of the source tree.
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*/
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#ifndef MODULES_AUDIO_PROCESSING_AGC2_FIXED_DIGITAL_LEVEL_ESTIMATOR_H_
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#define MODULES_AUDIO_PROCESSING_AGC2_FIXED_DIGITAL_LEVEL_ESTIMATOR_H_
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#include <array>
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#include <vector>
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#include "modules/audio_processing/agc2/agc2_common.h"
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#include "modules/audio_processing/include/audio_frame_view.h"
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namespace webrtc {
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class ApmDataDumper;
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// Produces a smooth signal level estimate from an input audio
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// stream. The estimate smoothing is done through exponential
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// filtering.
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class FixedDigitalLevelEstimator {
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public:
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// `samples_per_channel` is expected to be derived from this formula:
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// sample_rate_hz * kFrameDurationMs / 1000
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// or, for a 10ms duration:
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// sample_rate_hz / 100
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// I.e. the number of samples for 10ms of the given sample rate. The
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// expectation is that samples per channel is divisible by
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// kSubFramesInSample. For kFrameDurationMs=10 and
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// kSubFramesInSample=20, this means that the original sample rate has to be
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// divisible by 2000 and therefore `samples_per_channel` by 20.
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FixedDigitalLevelEstimator(size_t samples_per_channel,
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ApmDataDumper* apm_data_dumper);
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FixedDigitalLevelEstimator(const FixedDigitalLevelEstimator&) = delete;
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FixedDigitalLevelEstimator& operator=(const FixedDigitalLevelEstimator&) =
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delete;
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// The input is assumed to be in FloatS16 format. Scaled input will
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// produce similarly scaled output. A frame of with kFrameDurationMs
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// ms of audio produces a level estimates in the same scale. The
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// level estimate contains kSubFramesInFrame values.
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std::array<float, kSubFramesInFrame> ComputeLevel(
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DeinterleavedView<const float> float_frame);
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// Rate may be changed at any time (but not concurrently) from the
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// value passed to the constructor. The class is not thread safe.
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void SetSamplesPerChannel(size_t samples_per_channel);
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// Resets the level estimator internal state.
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void Reset();
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float LastAudioLevel() const { return filter_state_level_; }
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private:
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void CheckParameterCombination();
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ApmDataDumper* const apm_data_dumper_ = nullptr;
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float filter_state_level_;
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int samples_in_frame_;
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int samples_in_sub_frame_;
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};
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} // namespace webrtc
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#endif // MODULES_AUDIO_PROCESSING_AGC2_FIXED_DIGITAL_LEVEL_ESTIMATOR_H_
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