Real-time Speaker State Analytics via Multi-feature Segmentation

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Solution Overview

Problem

Conventional speech-based emotion detection systems are inflexible, rely heavily on word-based information, and struggle with normalizing features for specific speakers and conditions, leading to poor performance in real-time applications and noisy environments, and are not capable of effectively analyzing short-time scales or changes in speaker state.

Innovation Solution

A flexible, adaptable real-time speech analytics system that extracts and analyzes various speech features from audio signals to detect and predict speaker states, including emotional, cognitive, and health-related states, using machine-learning techniques and automatic speech recognition, capable of operating in noisy conditions and providing real-time feedback.

Engineering Contradictions & Design Principles

VSEngineering Contradiction Analysis

1Measurement precision

If conventional speech-based emotion detection systems rely heavily on word-based information, then they can achieve basic emotion classification, but they fail to accurately detect speaker states in noisy environments and short time windows

Engineering Contradiction:
Improvespeaker state detection accuracyVSAvoidadaptability to noisy environments and short time scales
Core Design Contradiction:
Measurement precisionVSAdaptability or versatility

Solution Approach 1:

The system segments speech analysis into multiple feature categories (lexical, prosodic, acoustic, non-word features) and processes them through separate analysis channels before integration. This segmentation allows each feature type to be optimized independently for noisy environments and short time windows, improving overall detection accuracy without requiring long analysis periods

Inventive Principle:
Principle #1Segmentation

Solution Approach 2:

The system dynamically adjusts analysis parameters including time window duration, feature extraction thresholds, and model confidence levels based on environmental noise conditions and available speech duration. This enables accurate speaker state detection to adapt to varying noise levels and short time constraints, maintaining precision across different operational conditions

Inventive Principle:
Principle #35Parameter changes

2Adaptability or versatility

If conventional systems use fixed analysis methods, then they are simple to implement, but they cannot effectively normalize features for specific speakers and conditions

Engineering Contradiction:
Improvenormalization for specific speakers and conditionsVSAvoidsystem complexity
Core Design Contradiction:
Adaptability or versatilityVSDevice complexity

Solution Approach 1:

The system performs preliminary speaker-specific normalization by collecting baseline speech samples during initial interactions and pre-computing speaker-specific feature ranges and thresholds. This preliminary adaptation enables the system to quickly normalize subsequent speech analysis for that specific speaker without requiring complex real-time adjustments, reducing ongoing computational complexity

Inventive Principle:
Principle #10Preliminary action

Solution Approach 2:

The system continuously monitors detection results and environmental conditions, using this feedback to dynamically adjust feature normalization parameters and model weights. This feedback loop enables the system to adapt to specific speakers and conditions over time, improving accuracy while keeping the adaptation mechanism manageable through iterative learning rather than complex upfront configuration

Inventive Principle:
Principle #23Feedback

3Speed

If conventional systems analyze long time windows, then they can capture sufficient speech data, but they cannot provide real-time feedback and respond to changes in speaker state

Engineering Contradiction:
Improvereal-time response speedVSAvoidprediction reliability
Core Design Contradiction:
SpeedVSReliability

Solution Approach 1:

The system performs periodic speaker state analysis at optimized time intervals rather than continuously analyzing long speech segments. By analyzing speech in periodic, shorter windows with appropriate overlap, the system achieves real-time response capability while maintaining sufficient data for reliable detection through the periodic accumulation of feature statistics across multiple analysis cycles

Inventive Principle:
Principle #19Periodic action

Solution Approach 2:

The system compensates for shorter time windows by adding dimensional depth through multi-feature analysis across different speech characteristics (lexical content, prosody, acoustic properties, non-word features). This dimensional enrichment allows reliable speaker state prediction to be achieved in shorter time frames by gathering more comprehensive information across multiple feature dimensions rather than relying on long temporal windows

Inventive Principle:
Principle #17Another dimension (Dimensionality change)

Data Source

PatentUS10706873B2Real-time speaker state analytics platform
Publication Date: 2020.07.07 SRI INTERNATIONAL
  • US10706873B2 patent drawing
  • US10706873B2 patent drawing
  • US10706873B2 patent drawing

AI summary

Disclosed are machine learning-based technologies that analyze an audio input and provide speaker state predictions in response to the audio input. The speaker state predictions can be selected and customized for each of a variety of different applications.