On-Device Language Model for Real-Time Transcription Error Correction

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Solution Overview

Problem

Existing large language models for speech recognition suffer from issues such as missing words, accuracy problems, and lapses, limiting their effectiveness for real-time transcription in applications like meeting notes or accessibility for hearing impairments.

Innovation Solution

A user electronic device is equipped with a language model that runs on-device, enabling near real-time error filtering, speaker identification, and summarization. This approach processes raw audio data to generate live transcriptions, corrects errors, and inserts speaker identifiers, all while maintaining user privacy and reducing latency.

Engineering Contradictions & Design Principles

VSEngineering Contradiction Analysis

1Speed

If a speech transcriber is used to generate live transcription in real-time, then transcription speed is improved, but transcription accuracy deteriorates due to missing words and errors

Engineering Contradiction:
Improvetranscription speedVSAvoidtranscription accuracy
Core Design Contradiction:
SpeedVSMeasurement precision

Solution Approach 1:

A language model acts as an intermediary between the speech transcriber and the final transcription output. The language model receives the live transcription tokens and processes them to correct errors, fill in missing words, and improve overall accuracy while maintaining real-time performance capabilities

Inventive Principle:
Principle #24Intermediary (Mediator)

2Measurement precision

If cloud-based language models are used for transcription correction, then transcription accuracy is improved, but latency increases and user privacy is compromised

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

Solution Approach 1:

The language model is deployed to run locally on the user's device, enabling it to service itself and perform transcription correction without needing to send data to external cloud servers. This self-contained approach eliminates network latency and keeps all processing local to the device

Inventive Principle:
Principle #25Self-service

3Measurement precision

If cloud-based language models are used for transcription correction, then transcription accuracy is improved, but user privacy and data security deteriorate due to data transmission requirements

Engineering Contradiction:
Improvetranscription accuracyVSAvoidprivacy risk
Core Design Contradiction:
Measurement precisionVSObject-affected harmful factors

Solution Approach 1:

The language model is extracted from cloud-based infrastructure and embedded directly into the local device. This extraction removes the need for data to leave the device, eliminating the privacy and security risks associated with transmitting sensitive audio and transcription data over networks

Inventive Principle:
Principle #2Taking out (Extraction)

4Productivity

If existing speech recognition models are used, then basic transcription functionality is provided, but transcription reliability deteriorates due to frequent errors and lapses

Engineering Contradiction:
Improvetranscription functionalityVSAvoidtranscription reliability
Core Design Contradiction:
ProductivityVSReliability

Solution Approach 1:

The system combines a speech transcriber model with a language model into a composite transcription system. The speech transcriber handles real-time audio-to-text conversion while the language model layer adds correction and refinement capabilities, creating a more reliable overall system that leverages the strengths of both model types

Inventive Principle:
Principle #40Composite materials

Data Source

PatentUS20250157473A1LLM as a transcription filter
Publication Date: 2025.05.15 GOOGLE LLC
  • US20250157473A1 patent drawing
  • US20250157473A1 patent drawing
  • US20250157473A1 patent drawing

AI summary

A user electronic device comprising: one or more microphones configured to capture raw audio data; and one or more processors and one or more storage devices storing instructions that when executed by the one or more computers cause the one or more computers to perform operations comprising: receiving the raw audio data captured by the one or more microphones; processing the raw audio data using a speech transcriber to generate a live transcription of the raw audio data that comprises a plurality of text tokens; processing the raw audio data to generate a speaker identification output that identifies, for each of the text tokens, a respective speaker for each of the text tokens in the live transcription; and processing a first input comprising (i) a first input prompt and (ii) an input text generated from the live transcription using a language model neural network to generate a modified transcription.