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Time-domain blind separating method for in-line natural voice convolution mixing signal

A voice signal, mixed signal technology, applied in the field of signal processing

Inactive Publication Date: 2007-02-07
FUDAN UNIV
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  • Summary
  • Abstract
  • Description
  • Claims
  • Application Information

AI Technical Summary

Problems solved by technology

This method is applied in the actual speech processing, which can effectively separate the speech signals of different speakers, and solve the limitation of only a single speaker in the field of speech recognition. It is also applicable to speech communication and robot interaction. can achieve better results

Method used

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  • Time-domain blind separating method for in-line natural voice convolution mixing signal
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  • Time-domain blind separating method for in-line natural voice convolution mixing signal

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Embodiment Construction

[0053] The following uses the collected two actual voice signals to separate the signals using the traditional mutual information method and the new online convolution blind separation time-domain method proposed by the present invention and compare the results. In order to reflect the generality and robustness of the algorithm, the speaker in the processed speech signal has pauses and silences (unavoidable under actual conditions).

[0054] At the same time, in order to quantitatively compare the separation effect of the method proposed by the present invention and the existing method, we use the artificially constructed hybrid convolution matrix to simulate the actual speech environment, and the z-domain expression of the linear convolution hybrid matrix is:

[0055] H ( z ) = 0.65 + 0.27 z ...

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Abstract

The related unsupervised learning algorithm to on-line treat speech signal for speech interactive front-end process comprises: with sufficient respect to time-domain relevance and non-stationary of natural speech signal, constructing a group of adaptive feed-forward filter networks to separate speech signal from multiple sources. This invention enhances speech signal, reduces algorithm complexity for hardware implementation, and has wide application.

Description

Technical field [0001] The invention belongs to the technical field of signal processing, and is specifically an unsupervised learning algorithm that can process the problem of blind separation of voice signal convolution on-line. The algorithm fully considers the time-domain correlation and non-stationary characteristics of natural speech signals, can separate speech signals from multiple sources in real time, and overcomes the instability of the existing algorithm when the number of sources changes. Therefore, it has a wide range of application prospects in the field of voice signal processing such as teleconferencing, intelligent robots, and automobile communication. technical background [0002] Blind signal separation problem (BSS) refers to the process of signal propagation, when the source signal and propagation channel parameters are unknown, the source is recovered only by the observed mixed signal and some statistical characteristics of the source signal. The signal pro...

Claims

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Application Information

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Patent Type & Authority Applications(China)
IPC IPC(8): G10L15/28G10L15/22G10L15/00H03H17/02G10L19/26
Inventor 鲁晓丹张立明
Owner FUDAN UNIV
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