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The AdaptiveAgc often boosts the signal outside of Float S16 range. It is expected, which is why we have a limiter after it in the process chain. But it turns out that this happens regularly even for simple input examples. The output signal peaks can be as high as +4 dBFs for a single speaker example (which should be easy). It leads to excessive gain modulation by the limiter. This CL is a new tuning designed to produce a safer gain. After this, we shouldn't hit the saturation region of the limiter as often. But we will still maintain a high gain. We have a 'configurable kill-switch': the settings can be changed via field trials WebRTC-Audio-Agc2Force(Initial|Extra)SaturationMargin. Bug: webrtc:7494, chromium:892043 Change-Id: I5014377050c74c32ae8998282991141eae31cf58 Reviewed-on: https://webrtc-review.googlesource.com/c/102922 Commit-Queue: Alex Loiko <aleloi@webrtc.org> Reviewed-by: Sam Zackrisson <saza@webrtc.org> Cr-Commit-Position: refs/heads/master@{#25006}
68 lines
1.9 KiB
C++
68 lines
1.9 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_SATURATION_PROTECTOR_H_
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#define MODULES_AUDIO_PROCESSING_AGC2_SATURATION_PROTECTOR_H_
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#include <array>
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#include "modules/audio_processing/agc2/agc2_common.h"
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#include "modules/audio_processing/agc2/vad_with_level.h"
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namespace webrtc {
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class ApmDataDumper;
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class SaturationProtector {
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public:
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explicit SaturationProtector(ApmDataDumper* apm_data_dumper);
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// Update and return margin estimate. This method should be called
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// whenever a frame is reliably classified as 'speech'.
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//
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// Returned value is in DB scale.
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void UpdateMargin(const VadWithLevel::LevelAndProbability& vad_data,
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float last_speech_level_estimate_dbfs);
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// Returns latest computed margin. Used in cases when speech is not
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// detected.
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float LastMargin() const;
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// Resets the internal memory.
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void Reset();
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void DebugDumpEstimate() const;
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private:
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// Computes a delayed envelope of peaks.
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class PeakEnveloper {
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public:
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PeakEnveloper();
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void Process(float frame_peak_dbfs);
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float Query() const;
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private:
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size_t speech_time_in_estimate_ms_ = 0;
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float current_superframe_peak_dbfs_ = -90.f;
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size_t elements_in_buffer_ = 0;
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std::array<float, kPeakEnveloperBufferSize> peak_delay_buffer_ = {};
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};
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ApmDataDumper* apm_data_dumper_;
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float last_margin_;
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PeakEnveloper peak_enveloper_;
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float extra_saturation_margin_db_;
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};
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} // namespace webrtc
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#endif // MODULES_AUDIO_PROCESSING_AGC2_SATURATION_PROTECTOR_H_
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