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093824c4d2
This simplifies the following steps: * FrameCombiner infers the sample rate from channel size * Sends the inferred sample rate to FixedDigitalLevelEstimator and Limiter. * Those classes then convert the sample rate to channel size. Along the way perform checks that the derived channel size value is a legal value (which has already been done by FrameCombiner). To: * FrameCombiner sends channel size to FixedDigitalLevelEstimator and Limiter. Bug: chromium:335805780 Change-Id: I6d2953ba5ee99771f3ff5bf4f4a049a8a29b5577 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/352581 Reviewed-by: Per Åhgren <peah@webrtc.org> Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org> Cr-Commit-Position: refs/heads/main@{#42480}
75 lines
2.5 KiB
C++
75 lines
2.5 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_LIMITER_H_
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#define MODULES_AUDIO_PROCESSING_AGC2_LIMITER_H_
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#include <vector>
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#include "absl/strings/string_view.h"
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#include "api/audio/audio_frame.h"
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#include "modules/audio_processing/agc2/fixed_digital_level_estimator.h"
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#include "modules/audio_processing/agc2/interpolated_gain_curve.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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class Limiter {
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public:
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// See `SetSamplesPerChannel()` for valid values for `samples_per_channel`.
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Limiter(ApmDataDumper* apm_data_dumper,
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size_t samples_per_channel,
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absl::string_view histogram_name_prefix);
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[[deprecated("Use constructor that accepts samples_per_channel")]] Limiter(
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int sample_rate_hz,
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ApmDataDumper* apm_data_dumper,
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absl::string_view histogram_name_prefix);
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Limiter(const Limiter& limiter) = delete;
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Limiter& operator=(const Limiter& limiter) = delete;
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~Limiter();
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// Applies limiter and hard-clipping to `signal`.
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void Process(AudioFrameView<float> signal);
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InterpolatedGainCurve::Stats GetGainCurveStats() const;
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// Supported values must be
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// * Supported by FixedDigitalLevelEstimator
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// * Below or equal to kMaximalNumberOfSamplesPerChannel so that samples
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// fit in the per_sample_scaling_factors_ array.
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void SetSamplesPerChannel(size_t samples_per_channel);
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[[deprecated("Use SetSamplesPerChannel")]] void SetSampleRate(
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int sample_rate_hz) {
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SetSamplesPerChannel(SampleRateToDefaultChannelSize(sample_rate_hz));
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}
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// Resets the internal state.
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void Reset();
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float LastAudioLevel() const;
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private:
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const InterpolatedGainCurve interp_gain_curve_;
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FixedDigitalLevelEstimator level_estimator_;
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ApmDataDumper* const apm_data_dumper_ = nullptr;
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// Work array containing the sub-frame scaling factors to be interpolated.
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std::array<float, kSubFramesInFrame + 1> scaling_factors_ = {};
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std::array<float, kMaximalNumberOfSamplesPerChannel>
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per_sample_scaling_factors_ = {};
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float last_scaling_factor_ = 1.f;
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};
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} // namespace webrtc
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#endif // MODULES_AUDIO_PROCESSING_AGC2_LIMITER_H_
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