03ad9b892c
The FixedGainController is used in two places. One is the AudioMixer. There it's used to limit the audio level after adding streams. The other is GainController2, where it's placed after steps that could boost the audio level outside the allowed range. We log metrics from the FGC. To avoid confusion, this CL makes the two use cases log to different histograms. Chromium histogram CL is https://chromium-review.googlesource.com/c/chromium/src/+/1170833 Bug: webrtc:7494 Change-Id: I1abe60fd8e96556f144d2ee576254b15beca1174 Reviewed-on: https://webrtc-review.googlesource.com/93464 Commit-Queue: Alex Loiko <aleloi@webrtc.org> Reviewed-by: Ivo Creusen <ivoc@webrtc.org> Reviewed-by: Åsa Persson <asapersson@webrtc.org> Cr-Commit-Position: refs/heads/master@{#24284}
59 lines
2.0 KiB
C++
59 lines
2.0 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_GAIN_CURVE_APPLIER_H_
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#define MODULES_AUDIO_PROCESSING_AGC2_GAIN_CURVE_APPLIER_H_
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#include <vector>
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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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#include "rtc_base/constructormagic.h"
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namespace webrtc {
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class ApmDataDumper;
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class GainCurveApplier {
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public:
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GainCurveApplier(size_t sample_rate_hz,
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ApmDataDumper* apm_data_dumper,
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std::string histogram_name_prefix);
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~GainCurveApplier();
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void Process(AudioFrameView<float> signal);
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InterpolatedGainCurve::Stats GetGainCurveStats() const;
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// Supported rates must be
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// * supported by FixedDigitalLevelEstimator
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// * below kMaximalNumberOfSamplesPerChannel*1000/kFrameDurationMs
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// so that samples_per_channel fit in the
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// per_sample_scaling_factors_ array.
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void SetSampleRate(size_t sample_rate_hz);
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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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RTC_DISALLOW_COPY_AND_ASSIGN(GainCurveApplier);
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};
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} // namespace webrtc
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#endif // MODULES_AUDIO_PROCESSING_AGC2_GAIN_CURVE_APPLIER_H_
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