Apparatus and method for noise reduction and speech enhancement with microphones and loudspeakers

• US 20060206320A1
• Filed: 03/13/2006
• Published: 09/14/2006
• Est. Priority Date: 03/14/2005
• Status: Abandoned Application

• Current Assignee: Qi P. Li
• Original Assignee: Qi P. Li
• Inventors: Li, Qi P.
• Application Number:
  US11/374,511
• Publication Number:
  US 20060206320A1
• Time in Patent Office: Days
• US Class Current: 704/226
• CPC Class Codes: G10L 21/0208 Noise filtering
• CPC Class Codes: G10L 25/18 the extracted parameters being spectral information of each sub-band

Abstract

The present invention helps to reduce the noise level and to enhance the quality of speech signals, in communications, computers, entertainment and other applications, where microphones and loudspeakers are involved. Additionally, the invention includes a new noise reduction and speech enhancement algorithm which is created based on the principles of human hearing mechanism. Further, the algorithm uses a new set of speech recognition parameters instead of just signal-to-noise ratio (“SNR”) as used in the prior art.

Claims

1. A noise reduction and speech enhancement apparatus, comprising:

a computation unit including a programmable circuitry which implements a noise reduction and speech enhancement algorithm.

a sound receiving unit or generating unit.

2. The apparatus as claimed in claim 1, wherein said sound receiving unit can be one or more than one microphone component.

3. The apparatus as claimed in claim 1, wherein said sound generating unit can be one or more than one loudspeaker.

4. The apparatus as claimed in claim 1, wherein said computation unit can be within said sound receiving unit, or sound generating unit, or as a separate module at any stage within an application system.

5. The apparatus as claimed in claim 4, wherein said application system can be a wireless handset, conference phone, speaker phone, cordless phone, hearing aid, earphone, headset, telephone speech, wireless station, telephone switch, network router, or any device processing speech signals.

6. The apparatus as claimed in claim 1, wherein said programmable circuitry further comprises an analog-to-digit (A/D) converter, a digital signal processor (DSP), a memory including RAM or ROM, and a digit-to-analog (D/A) converter.

7. The apparatus as claimed in claim 6, wherein said noise reduction and speech enhancement algorithm and corresponding software implementation are pre-stored in said memory. All the functions are fabricated in one silicon die, and the die can be packaged as a chip when necessary. Alternatively, the die can also be packaged on a circuit board directly as system-on-board packaging.

8. The apparatus as claimed in claim 7, wherein said noise reduction and speech enhancement algorithm comprises a Cochlear Transform algorithm, which is implemented by said DSP.

9. The apparatus as claimed in claim 8, wherein said circuitry further comprises a bank of auditory-based filters or an array of auditory-based filters.

10. The apparatus as claimed in claim 9, wherein parameters of said auditory-based filters can be adjusted or adapted by a feedback method.

11. The apparatus as claimed in claim 10, wherein said feedback method is to use automatic speech recognition (ASR) error rates or other information related to the desired signal quality.

12. The apparatus as claimed in claim 11, wherein said ASR error rates are calculated by an ASR system and said other information are generated by a knowledge-based system.

13. The apparatus as claimed in claim 9, wherein said auditory-based filter banks are digital, analog, or mechanical. The filter bank has similar frequency response as the basilar membrane in the cochlear of hearing system. The filter bank decomposes received signal into different frequency bands for further processing.

14. The apparatus as claimed in claim 13, wherein output from each said auditory-based filter is then processed by a special nonlinear unit, which can be realized in forms of a hard-limit threshold, a log function, a nonlinear function, or an artificial neural network.

15. The apparatus as claimed in claim 14, wherein outputs of said nonlinear units after passing through a signal phase changer are added by an adder to re-synthesis the cleaned or processed speech signal.

16. The apparatus as claimed in claim 15, wherein said cleaned speech signal is then evaluated by an ASR system or a knowledge-based system. The evaluation results in terms of the quality of the processed speech are then fed back through a parameter optimizer or adaptor to adjust the parameters in the auditory filters and the nonlinear processor to further improve the quality of the processed sound.

18. The method as claimed in claim 16, wherein the parameters of said bank of auditory-based filters can be adjusted using the ASR error rates or other estimated information to further improve the quality of the processed signal.

17. A method for reducing noise in speech and enhancing speech quality, comprising the steps of:

receiving the speech signal;

sending received speech signal through a pre-amplifier;

converting the amplified signal into digital format using A/D converter;

transforming the digital signal to different frequency bands using the Cochlear Transform algorithm and the auditory-based filter bank;

estimating the background noise from filter bank output based on the pre-knowledge of speech and noise;

removing or reducing noise using a nonlinear function or unit;

re-synthesizing the processed, i.e. cleaned, signal through the Inverse Cochlear Transform;

converting the time-domain signal from digital format into analog signal through a digital-to-analog (“

D/A”

) converter if necessary;

outputting the analog or digital signal.