Bump to WebRTC M120 release
Some API deprecation -- ExperimentalAgc and ExperimentalNs are gone. We're continuing to carry iSAC even though it's gone upstream, but maybe we'll want to drop that soon.
This commit is contained in:
Arun Raghavan
174
webrtc/modules/audio_processing/agc2/speech_level_estimator.cc
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174
webrtc/modules/audio_processing/agc2/speech_level_estimator.cc
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/*
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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/speech_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/checks.h"
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#include "rtc_base/logging.h"
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#include "rtc_base/numerics/safe_minmax.h"
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namespace webrtc {
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namespace {
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float ClampLevelEstimateDbfs(float level_estimate_dbfs) {
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return rtc::SafeClamp<float>(level_estimate_dbfs, -90.0f, 30.0f);
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}
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// Returns the initial speech level estimate needed to apply the initial gain.
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float GetInitialSpeechLevelEstimateDbfs(
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const AudioProcessing::Config::GainController2::AdaptiveDigital& config) {
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return ClampLevelEstimateDbfs(-kSaturationProtectorInitialHeadroomDb -
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config.initial_gain_db - config.headroom_db);
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}
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} // namespace
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bool SpeechLevelEstimator::LevelEstimatorState::operator==(
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const SpeechLevelEstimator::LevelEstimatorState& b) const {
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return time_to_confidence_ms == b.time_to_confidence_ms &&
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level_dbfs.numerator == b.level_dbfs.numerator &&
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level_dbfs.denominator == b.level_dbfs.denominator;
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}
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float SpeechLevelEstimator::LevelEstimatorState::Ratio::GetRatio() const {
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RTC_DCHECK_NE(denominator, 0.f);
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return numerator / denominator;
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}
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SpeechLevelEstimator::SpeechLevelEstimator(
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ApmDataDumper* apm_data_dumper,
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const AudioProcessing::Config::GainController2::AdaptiveDigital& config,
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int adjacent_speech_frames_threshold)
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: apm_data_dumper_(apm_data_dumper),
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initial_speech_level_dbfs_(GetInitialSpeechLevelEstimateDbfs(config)),
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adjacent_speech_frames_threshold_(adjacent_speech_frames_threshold),
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level_dbfs_(initial_speech_level_dbfs_),
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// TODO(bugs.webrtc.org/7494): Remove init below when AGC2 input volume
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// controller temporal dependency removed.
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is_confident_(false) {
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RTC_DCHECK(apm_data_dumper_);
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RTC_DCHECK_GE(adjacent_speech_frames_threshold_, 1);
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Reset();
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}
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void SpeechLevelEstimator::Update(float rms_dbfs,
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float peak_dbfs,
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float speech_probability) {
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RTC_DCHECK_GT(rms_dbfs, -150.0f);
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RTC_DCHECK_LT(rms_dbfs, 50.0f);
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RTC_DCHECK_GT(peak_dbfs, -150.0f);
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RTC_DCHECK_LT(peak_dbfs, 50.0f);
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RTC_DCHECK_GE(speech_probability, 0.0f);
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RTC_DCHECK_LE(speech_probability, 1.0f);
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if (speech_probability < kVadConfidenceThreshold) {
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// Not a speech frame.
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if (adjacent_speech_frames_threshold_ > 1) {
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// When two or more adjacent speech frames are required in order to update
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// the state, we need to decide whether to discard or confirm the updates
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// based on the speech sequence length.
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if (num_adjacent_speech_frames_ >= adjacent_speech_frames_threshold_) {
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// First non-speech frame after a long enough sequence of speech frames.
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// Update the reliable state.
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reliable_state_ = preliminary_state_;
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} else if (num_adjacent_speech_frames_ > 0) {
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// First non-speech frame after a too short sequence of speech frames.
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// Reset to the last reliable state.
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preliminary_state_ = reliable_state_;
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}
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}
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num_adjacent_speech_frames_ = 0;
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} else {
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// Speech frame observed.
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num_adjacent_speech_frames_++;
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// Update preliminary level estimate.
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RTC_DCHECK_GE(preliminary_state_.time_to_confidence_ms, 0);
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const bool buffer_is_full = preliminary_state_.time_to_confidence_ms == 0;
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if (!buffer_is_full) {
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preliminary_state_.time_to_confidence_ms -= kFrameDurationMs;
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}
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// Weighted average of levels with speech probability as weight.
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RTC_DCHECK_GT(speech_probability, 0.0f);
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const float leak_factor = buffer_is_full ? kLevelEstimatorLeakFactor : 1.0f;
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preliminary_state_.level_dbfs.numerator =
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preliminary_state_.level_dbfs.numerator * leak_factor +
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rms_dbfs * speech_probability;
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preliminary_state_.level_dbfs.denominator =
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preliminary_state_.level_dbfs.denominator * leak_factor +
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speech_probability;
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const float level_dbfs = preliminary_state_.level_dbfs.GetRatio();
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if (num_adjacent_speech_frames_ >= adjacent_speech_frames_threshold_) {
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// `preliminary_state_` is now reliable. Update the last level estimation.
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level_dbfs_ = ClampLevelEstimateDbfs(level_dbfs);
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}
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}
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UpdateIsConfident();
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DumpDebugData();
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}
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void SpeechLevelEstimator::UpdateIsConfident() {
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if (adjacent_speech_frames_threshold_ == 1) {
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// Ignore `reliable_state_` when a single frame is enough to update the
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// level estimate (because it is not used).
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is_confident_ = preliminary_state_.time_to_confidence_ms == 0;
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return;
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}
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// Once confident, it remains confident.
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RTC_DCHECK(reliable_state_.time_to_confidence_ms != 0 ||
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preliminary_state_.time_to_confidence_ms == 0);
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// During the first long enough speech sequence, `reliable_state_` must be
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// ignored since `preliminary_state_` is used.
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is_confident_ =
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reliable_state_.time_to_confidence_ms == 0 ||
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(num_adjacent_speech_frames_ >= adjacent_speech_frames_threshold_ &&
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preliminary_state_.time_to_confidence_ms == 0);
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}
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void SpeechLevelEstimator::Reset() {
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ResetLevelEstimatorState(preliminary_state_);
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ResetLevelEstimatorState(reliable_state_);
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level_dbfs_ = initial_speech_level_dbfs_;
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num_adjacent_speech_frames_ = 0;
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}
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void SpeechLevelEstimator::ResetLevelEstimatorState(
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LevelEstimatorState& state) const {
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state.time_to_confidence_ms = kLevelEstimatorTimeToConfidenceMs;
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state.level_dbfs.numerator = initial_speech_level_dbfs_;
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state.level_dbfs.denominator = 1.0f;
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}
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void SpeechLevelEstimator::DumpDebugData() const {
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if (!apm_data_dumper_)
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return;
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apm_data_dumper_->DumpRaw("agc2_speech_level_dbfs", level_dbfs_);
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apm_data_dumper_->DumpRaw("agc2_speech_level_is_confident", is_confident_);
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apm_data_dumper_->DumpRaw(
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"agc2_adaptive_level_estimator_num_adjacent_speech_frames",
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num_adjacent_speech_frames_);
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apm_data_dumper_->DumpRaw(
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"agc2_adaptive_level_estimator_preliminary_level_estimate_num",
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preliminary_state_.level_dbfs.numerator);
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apm_data_dumper_->DumpRaw(
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"agc2_adaptive_level_estimator_preliminary_level_estimate_den",
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preliminary_state_.level_dbfs.denominator);
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apm_data_dumper_->DumpRaw(
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"agc2_adaptive_level_estimator_preliminary_time_to_confidence_ms",
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preliminary_state_.time_to_confidence_ms);
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apm_data_dumper_->DumpRaw(
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"agc2_adaptive_level_estimator_reliable_time_to_confidence_ms",
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reliable_state_.time_to_confidence_ms);
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}
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} // namespace webrtc
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