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Diffstat (limited to '')
-rw-r--r-- | third_party/libwebrtc/modules/audio_processing/agc2/speech_level_estimator.cc | 174 |
1 files changed, 174 insertions, 0 deletions
diff --git a/third_party/libwebrtc/modules/audio_processing/agc2/speech_level_estimator.cc b/third_party/libwebrtc/modules/audio_processing/agc2/speech_level_estimator.cc new file mode 100644 index 0000000000..7bf3252116 --- /dev/null +++ b/third_party/libwebrtc/modules/audio_processing/agc2/speech_level_estimator.cc @@ -0,0 +1,174 @@ +/* + * Copyright (c) 2018 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.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/logging.h" +#include "rtc_base/numerics/safe_minmax.h" + +namespace webrtc { +namespace { + +float ClampLevelEstimateDbfs(float level_estimate_dbfs) { + return rtc::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 + +bool SpeechLevelEstimator::LevelEstimatorState::operator==( + const SpeechLevelEstimator::LevelEstimatorState& b) const { + return time_to_confidence_ms == b.time_to_confidence_ms && + level_dbfs.numerator == b.level_dbfs.numerator && + level_dbfs.denominator == b.level_dbfs.denominator; +} + +float SpeechLevelEstimator::LevelEstimatorState::Ratio::GetRatio() const { + RTC_DCHECK_NE(denominator, 0.f); + return numerator / denominator; +} + +SpeechLevelEstimator::SpeechLevelEstimator( + 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 SpeechLevelEstimator::Update(float rms_dbfs, + float peak_dbfs, + float speech_probability) { + RTC_DCHECK_GT(rms_dbfs, -150.0f); + RTC_DCHECK_LT(rms_dbfs, 50.0f); + RTC_DCHECK_GT(peak_dbfs, -150.0f); + RTC_DCHECK_LT(peak_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 SpeechLevelEstimator::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 SpeechLevelEstimator::Reset() { + ResetLevelEstimatorState(preliminary_state_); + ResetLevelEstimatorState(reliable_state_); + level_dbfs_ = initial_speech_level_dbfs_; + num_adjacent_speech_frames_ = 0; +} + +void SpeechLevelEstimator::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 SpeechLevelEstimator::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 |