Audio Processing in Multi-Speaker Environments

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Solution Overview

Problem

Existing speaker identification, verification, and diarization systems face challenges in efficiently processing audio data from variable numbers of microphones and unknown numbers of speakers and noise sources in complex acoustic environments.

Innovation Solution

The system employs techniques such as channel clustering, voice detection, channel mixing, and embedding clustering to preprocess and aggregate audio data from multiple channels, allowing for efficient speaker identification, verification, and diarization without prior knowledge of the number or arrangement of audio capture devices.

Engineering Contradictions & Design Principles

VSEngineering Contradiction Analysis

1Measurement precision

If traditional speaker identification systems process audio from multiple microphones, then speaker identification accuracy may improve, but computational complexity and processing time increase significantly

Engineering Contradiction:
Improvespeaker identification accuracyVSAvoidcomputational complexity
Core Design Contradiction:
Measurement precisionVSDevice complexity

Solution Approach 1:

The system segments the audio processing task by first performing voice activity detection to identify speech segments, then performing speaker identification only on those segments. This segmentation approach reduces the amount of audio data requiring complex processing while maintaining accurate speaker identification for actual speech content.

Inventive Principle:
Principle #1Segmentation

Solution Approach 2:

The system performs preliminary voice activity detection and audio preprocessing before speaker identification. By pre-identifying which portions of the audio contain speech and preparing the audio data in advance, the system reduces the computational burden on the speaker identification algorithm while maintaining accuracy.

Inventive Principle:
Principle #10Preliminary action

2Adaptability or versatility

If the system processes audio from a variable number of microphones and unknown numbers of speakers, then adaptability to different environments improves, but system complexity increases

Engineering Contradiction:
Improveenvironmental adaptabilityVSAvoidsystem complexity
Core Design Contradiction:
Adaptability or versatilityVSDevice complexity

Solution Approach 1:

The system employs universal audio processing components that can handle variable numbers of microphones and speakers without requiring configuration changes. The voice activity detector and speaker identification module are designed to process audio from any number of sources, making the system adaptable to different environments while maintaining a consistent, manageable architecture.

Inventive Principle:
Principle #6Universality (Multi-functionality)

Solution Approach 2:

The system dynamically adapts to the number of active speakers and microphones by continuously performing voice activity detection and adjusting processing accordingly. This dynamic approach allows the system to handle variable environments efficiently without requiring complex pre-configurations or static architectures.

Inventive Principle:
Principle #15Dynamics

3Measurement precision

If comprehensive audio processing is performed on all audio data, then transcription accuracy improves, but processing time and computational resources increase

Engineering Contradiction:
Improvetranscription accuracyVSAvoidprocessing time
Core Design Contradiction:
Measurement precisionVSLoss of time

Solution Approach 1:

The system extracts and processes only the relevant portions of audio data by performing voice activity detection to identify speech segments. By extracting and processing only these speech-containing segments rather than the entire audio stream, the system maintains transcription accuracy while significantly reducing processing time and computational resource requirements.

Inventive Principle:
Principle #2Taking out (Extraction)

Data Source

PatentUS20250029618A1Audio processing in multi-speaker multi-channel audio environments
Publication Date: 2025.01.23 NVIDIA CORP
  • US20250029618A1 patent drawing
  • US20250029618A1 patent drawing
  • US20250029618A1 patent drawing

AI summary

Disclosed are apparatuses, systems, and techniques that may use machine learning for implementing speaker recognition, verification, and/or diarization. The techniques include receiving a first set of audio data channels (ADCs) jointly capturing a speech produced by one or more speakers and obtaining, using the first set of ADCs, a second set of one or more ADCs. Individual ADCs of the second set of ADCs represent one or more channels of the first set of ADCs, and at least one channel of the second set of ADCs represents a cluster of two or more ADCs of the first set of ADCs, the two of more ADCs being selected based on similarity of audio data of the two or more ADCs. The techniques further include processing, using an audio processing neural network model, the second set of ADCs to obtain an association of the speech to the one or more speakers.