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Hypothetical scenario: short weak speech at start of call, then high noise. The digital adaptive AGC2 would pick a high gain, and then continue to apply it on the noise. Unless the noise is detected by the noise estimator, the gain would never be reduced. This CL addresses the issue by sending limiter gain info to the adaptive digital AGC2. Bug: webrtc:7494 Change-Id: Idf5c2686af0f5e5bad981d39a95b8efc9ffb9d64 Reviewed-on: https://webrtc-review.googlesource.com/102641 Reviewed-by: Sam Zackrisson <saza@webrtc.org> Commit-Queue: Alex Loiko <aleloi@webrtc.org> Cr-Commit-Position: refs/heads/master@{#24922}
63 lines
2 KiB
C++
63 lines
2 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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// 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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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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