| /* | |
| * Copyright (c) 2025 The WebRTC project authors. All Rights Reserved. | |
| * | |
| * Use of this source code is governed by a BSD-style license | |
| * that can be found in the LICENSE file in the root of the source | |
| * tree. An additional intellectual property rights grant can be found | |
| * in the file PATENTS. All contributing project authors may | |
| * be found in the AUTHORS file in the root of the source tree. | |
| */ | |
| #include "modules/audio_processing/agc2/speech_level_estimator_impl.h" | |
| #include "api/audio/audio_processing.h" | |
| #include "modules/audio_processing/agc2/agc2_common.h" | |
| #include "modules/audio_processing/logging/apm_data_dumper.h" | |
| #include "rtc_base/checks.h" | |
| #include "rtc_base/numerics/safe_minmax.h" | |
| namespace webrtc { | |
| namespace { | |
| float ClampLevelEstimateDbfs(float level_estimate_dbfs) { | |
| return SafeClamp<float>(level_estimate_dbfs, -90.0f, 30.0f); | |
| } | |
| // Returns the initial speech level estimate needed to apply the initial gain. | |
| float GetInitialSpeechLevelEstimateDbfs( | |
| const AudioProcessing::Config::GainController2::AdaptiveDigital& config) { | |
| return ClampLevelEstimateDbfs(-kSaturationProtectorInitialHeadroomDb - | |
| config.initial_gain_db - config.headroom_db); | |
| } | |
| } // namespace | |
| float SpeechLevelEstimatorImpl::LevelEstimatorState::Ratio::GetRatio() const { | |
| RTC_DCHECK_NE(denominator, 0.f); | |
| return numerator / denominator; | |
| } | |
| SpeechLevelEstimatorImpl::SpeechLevelEstimatorImpl( | |
| ApmDataDumper* apm_data_dumper, | |
| const AudioProcessing::Config::GainController2::AdaptiveDigital& config, | |
| int adjacent_speech_frames_threshold) | |
| : apm_data_dumper_(apm_data_dumper), | |
| initial_speech_level_dbfs_(GetInitialSpeechLevelEstimateDbfs(config)), | |
| adjacent_speech_frames_threshold_(adjacent_speech_frames_threshold), | |
| level_dbfs_(initial_speech_level_dbfs_), | |
| // TODO(bugs.webrtc.org/7494): Remove init below when AGC2 input volume | |
| // controller temporal dependency removed. | |
| is_confident_(false) { | |
| RTC_DCHECK(apm_data_dumper_); | |
| RTC_DCHECK_GE(adjacent_speech_frames_threshold_, 1); | |
| Reset(); | |
| } | |
| void SpeechLevelEstimatorImpl::Update(float rms_dbfs, | |
| float speech_probability) { | |
| RTC_DCHECK_GT(rms_dbfs, -150.0f); | |
| RTC_DCHECK_LT(rms_dbfs, 50.0f); | |
| RTC_DCHECK_GE(speech_probability, 0.0f); | |
| RTC_DCHECK_LE(speech_probability, 1.0f); | |
| if (speech_probability < kVadConfidenceThreshold) { | |
| // Not a speech frame. | |
| if (adjacent_speech_frames_threshold_ > 1) { | |
| // When two or more adjacent speech frames are required in order to update | |
| // the state, we need to decide whether to discard or confirm the updates | |
| // based on the speech sequence length. | |
| if (num_adjacent_speech_frames_ >= adjacent_speech_frames_threshold_) { | |
| // First non-speech frame after a long enough sequence of speech frames. | |
| // Update the reliable state. | |
| reliable_state_ = preliminary_state_; | |
| } else if (num_adjacent_speech_frames_ > 0) { | |
| // First non-speech frame after a too short sequence of speech frames. | |
| // Reset to the last reliable state. | |
| preliminary_state_ = reliable_state_; | |
| } | |
| } | |
| num_adjacent_speech_frames_ = 0; | |
| } else { | |
| // Speech frame observed. | |
| num_adjacent_speech_frames_++; | |
| // Update preliminary level estimate. | |
| RTC_DCHECK_GE(preliminary_state_.time_to_confidence_ms, 0); | |
| const bool buffer_is_full = preliminary_state_.time_to_confidence_ms == 0; | |
| if (!buffer_is_full) { | |
| preliminary_state_.time_to_confidence_ms -= kFrameDurationMs; | |
| } | |
| // Weighted average of levels with speech probability as weight. | |
| RTC_DCHECK_GT(speech_probability, 0.0f); | |
| const float leak_factor = buffer_is_full ? kLevelEstimatorLeakFactor : 1.0f; | |
| preliminary_state_.level_dbfs.numerator = | |
| preliminary_state_.level_dbfs.numerator * leak_factor + | |
| rms_dbfs * speech_probability; | |
| preliminary_state_.level_dbfs.denominator = | |
| preliminary_state_.level_dbfs.denominator * leak_factor + | |
| speech_probability; | |
| const float level_dbfs = preliminary_state_.level_dbfs.GetRatio(); | |
| if (num_adjacent_speech_frames_ >= adjacent_speech_frames_threshold_) { | |
| // `preliminary_state_` is now reliable. Update the last level estimation. | |
| level_dbfs_ = ClampLevelEstimateDbfs(level_dbfs); | |
| } | |
| } | |
| UpdateIsConfident(); | |
| DumpDebugData(); | |
| } | |
| void SpeechLevelEstimatorImpl::UpdateIsConfident() { | |
| if (adjacent_speech_frames_threshold_ == 1) { | |
| // Ignore `reliable_state_` when a single frame is enough to update the | |
| // level estimate (because it is not used). | |
| is_confident_ = preliminary_state_.time_to_confidence_ms == 0; | |
| return; | |
| } | |
| // Once confident, it remains confident. | |
| RTC_DCHECK(reliable_state_.time_to_confidence_ms != 0 || | |
| preliminary_state_.time_to_confidence_ms == 0); | |
| // During the first long enough speech sequence, `reliable_state_` must be | |
| // ignored since `preliminary_state_` is used. | |
| is_confident_ = | |
| reliable_state_.time_to_confidence_ms == 0 || | |
| (num_adjacent_speech_frames_ >= adjacent_speech_frames_threshold_ && | |
| preliminary_state_.time_to_confidence_ms == 0); | |
| } | |
| void SpeechLevelEstimatorImpl::Reset() { | |
| ResetLevelEstimatorState(preliminary_state_); | |
| ResetLevelEstimatorState(reliable_state_); | |
| level_dbfs_ = initial_speech_level_dbfs_; | |
| num_adjacent_speech_frames_ = 0; | |
| } | |
| void SpeechLevelEstimatorImpl::ResetLevelEstimatorState( | |
| LevelEstimatorState& state) const { | |
| state.time_to_confidence_ms = kLevelEstimatorTimeToConfidenceMs; | |
| state.level_dbfs.numerator = initial_speech_level_dbfs_; | |
| state.level_dbfs.denominator = 1.0f; | |
| } | |
| void SpeechLevelEstimatorImpl::DumpDebugData() const { | |
| if (!apm_data_dumper_) | |
| return; | |
| apm_data_dumper_->DumpRaw("agc2_speech_level_dbfs", level_dbfs_); | |
| apm_data_dumper_->DumpRaw("agc2_speech_level_is_confident", is_confident_); | |
| apm_data_dumper_->DumpRaw( | |
| "agc2_adaptive_level_estimator_num_adjacent_speech_frames", | |
| num_adjacent_speech_frames_); | |
| apm_data_dumper_->DumpRaw( | |
| "agc2_adaptive_level_estimator_preliminary_level_estimate_num", | |
| preliminary_state_.level_dbfs.numerator); | |
| apm_data_dumper_->DumpRaw( | |
| "agc2_adaptive_level_estimator_preliminary_level_estimate_den", | |
| preliminary_state_.level_dbfs.denominator); | |
| apm_data_dumper_->DumpRaw( | |
| "agc2_adaptive_level_estimator_preliminary_time_to_confidence_ms", | |
| preliminary_state_.time_to_confidence_ms); | |
| apm_data_dumper_->DumpRaw( | |
| "agc2_adaptive_level_estimator_reliable_time_to_confidence_ms", | |
| reliable_state_.time_to_confidence_ms); | |
| } | |
| } // namespace webrtc |