End-to-End Speech Recognition Neural Network for Far-Field Audio

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Solution Overview

Problem

Conventional speech recognition systems face challenges in far-field environments due to low signal-to-noise ratios and inconsistent optimization objectives between front-end speech enhancement and back-end recognition, leading to poor accuracy and reliability.

Innovation Solution

An end-to-end neural network-based model for integrated speech enhancement and recognition that extracts enhanced features from multi-channel voice signals, optimizing both speech enhancement and recognition objectives jointly, using a complex convolutional neural network to perform beamforming and feature extraction while preserving phase information, and simulating far-field voice signals in real-time for enriched training data.

Engineering Contradictions & Design Principles

VSEngineering Contradiction Analysis

1Ease of manufacture

If conventional speech enhancement and recognition are used separately, then each module can be optimized independently, but the overall recognition accuracy in far-field environments deteriorates due to inconsistent optimization objectives

Engineering Contradiction:
ImproveIndependent optimization of modulesVSAvoidSpeech recognition accuracy
Core Design Contradiction:
Ease of manufactureVSMeasurement precision

Solution Approach 1:

The patent merges the speech enhancement module and speech recognition module into a unified end-to-end neural network model. This integration allows the optimization objectives of both modules to be aligned, as they now share a common loss function and training process, thereby resolving the contradiction between independent optimization ease and overall recognition accuracy.

Inventive Principle:
Principle #5Merging (Combining)

2Adaptability or versatility

If far-field speech recognition is performed in noisy environments, then the system can operate in real-world scenarios, but the signal-to-noise ratio decreases leading to poor recognition performance

Engineering Contradiction:
ImproveReal-world scenario operationVSAvoidSpeech recognition performance
Core Design Contradiction:
Adaptability or versatilityVSReliability

Solution Approach 1:

The patent applies preliminary speech enhancement actions within the end-to-end model by incorporating convolutional neural network layers that perform beamforming and feature extraction before the main recognition process. This preliminary enhancement of speech signals in noisy environments improves the signal-to-noise ratio, thereby enhancing recognition reliability while maintaining adaptability to real-world scenarios.

Inventive Principle:
Principle #10Preliminary action

3Measurement precision

If deep learning technologies are applied to speech recognition, then recognition accuracy improves, but the computational complexity and resource requirements increase

Engineering Contradiction:
ImproveSpeech recognition accuracyVSAvoidComputational complexity
Core Design Contradiction:
Measurement precisionVSDevice complexity

Solution Approach 1:

The patent segments the speech recognition system into distinct functional layers within the neural network, including convolutional layers for feature extraction, beamforming operations for spatial filtering, and recognition layers for text output. This segmentation allows for optimized computation at each stage, reducing overall computational complexity while maintaining high recognition accuracy through specialized processing in each segment.

Inventive Principle:
Principle #1Segmentation

Data Source

PatentUS11620983B2Speech recognition method, device, and computer-readable storage medium
Publication Date: 2023.04.04 BAIDU ONLINE NETWORK TECH (BEIJIBG) CO LTD
  • US11620983B2 patent drawing
  • US11620983B2 patent drawing
  • US11620983B2 patent drawing

AI summary

The disclosure provides a speech recognition method, a device and a computer-readable storage medium. The method includes obtaining a first voice signal collected from a first microphone in a microphone array and a second voice signal collected from a second microphone in the microphone array, the microphone array including at least two microphones, such as two, three or six microphones. The method further includes extracting enhanced features associated with the first voice signal and the second voice signal through a neural network, and obtaining a speech recognition result based on the enhanced features extracted.