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4d01146f16
1. Adds support for Reset calls in AGC2. The AGC will be reset during analog gain changes. 2. Allows AdaptiveModeLevelEstimator to return estimates > 0. This can happen if the signal gain is too high. It's needed for letting the analog AGC know that the gain is too high. Bug: webrtc:7494 Change-Id: I38def17c21cc01c36aaea79a2401d8c2f289407b Reviewed-on: https://webrtc-review.googlesource.com/79360 Commit-Queue: Alex Loiko <aleloi@webrtc.org> Reviewed-by: Ivo Creusen <ivoc@webrtc.org> Cr-Commit-Position: refs/heads/master@{#23805}
77 lines
2.8 KiB
C++
77 lines
2.8 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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#include "modules/audio_processing/agc2/adaptive_mode_level_estimator.h"
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#include "modules/audio_processing/agc2/agc2_common.h"
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#include "modules/audio_processing/logging/apm_data_dumper.h"
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#include "rtc_base/numerics/safe_minmax.h"
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namespace webrtc {
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AdaptiveModeLevelEstimator::AdaptiveModeLevelEstimator(
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ApmDataDumper* apm_data_dumper)
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: saturation_protector_(apm_data_dumper),
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apm_data_dumper_(apm_data_dumper) {}
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void AdaptiveModeLevelEstimator::UpdateEstimation(
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const VadWithLevel::LevelAndProbability& vad_data) {
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RTC_DCHECK_GT(vad_data.speech_rms_dbfs, -150.f);
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RTC_DCHECK_LT(vad_data.speech_rms_dbfs, 50.f);
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RTC_DCHECK_GT(vad_data.speech_peak_dbfs, -150.f);
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RTC_DCHECK_LT(vad_data.speech_peak_dbfs, 50.f);
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RTC_DCHECK_GE(vad_data.speech_probability, 0.f);
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RTC_DCHECK_LE(vad_data.speech_probability, 1.f);
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if (vad_data.speech_probability < kVadConfidenceThreshold) {
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DebugDumpEstimate();
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return;
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}
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const bool buffer_is_full = buffer_size_ms_ >= kFullBufferSizeMs;
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if (!buffer_is_full) {
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buffer_size_ms_ += kFrameDurationMs;
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}
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const float leak_factor = buffer_is_full ? kFullBufferLeakFactor : 1.f;
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estimate_numerator_ = estimate_numerator_ * leak_factor +
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vad_data.speech_rms_dbfs * vad_data.speech_probability;
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estimate_denominator_ =
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estimate_denominator_ * leak_factor + vad_data.speech_probability;
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last_estimate_with_offset_dbfs_ = estimate_numerator_ / estimate_denominator_;
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saturation_protector_.UpdateMargin(vad_data, last_estimate_with_offset_dbfs_);
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DebugDumpEstimate();
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}
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float AdaptiveModeLevelEstimator::LatestLevelEstimate() const {
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return rtc::SafeClamp<float>(
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last_estimate_with_offset_dbfs_ + saturation_protector_.LastMargin(),
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-90.f, 30.f);
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}
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void AdaptiveModeLevelEstimator::Reset() {
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buffer_size_ms_ = 0;
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last_estimate_with_offset_dbfs_ = kInitialSpeechLevelEstimateDbfs;
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estimate_numerator_ = 0.f;
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estimate_denominator_ = 0.f;
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saturation_protector_.Reset();
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}
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void AdaptiveModeLevelEstimator::DebugDumpEstimate() {
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apm_data_dumper_->DumpRaw("agc2_adaptive_level_estimate_with_offset_dbfs",
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last_estimate_with_offset_dbfs_);
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apm_data_dumper_->DumpRaw("agc2_adaptive_level_estimate_dbfs",
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LatestLevelEstimate());
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saturation_protector_.DebugDumpEstimate();
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}
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} // namespace webrtc
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