Dual Microphone System and Method for Enhancing Voice Quality

Information

  • Patent Application
  • 20070165879
  • Publication Number
    20070165879
  • Date Filed
    January 13, 2007
    18 years ago
  • Date Published
    July 19, 2007
    17 years ago
Abstract
Techniques to enhance voice signals in a dual microphone system are disclosed. According to one aspect of the present invention, there are at least two microphones that are positioned in a pre-configured array. Two audio signals x1(k) and x2(k) are received and coupled to an adjusting module that is provided to control the gain of each of the audio signals x1(k) and x2(k) to minimize signal differences between the two signals. A separation module is provided to receive matched audio signals x′1(k) and x′2(k) from the adjusting module. The separation module separates the audio signals x′1(k) and x′2(k) to obtain a first audio signal s(k) containing mainly the voice and a second audio signal n(k) containing mainly the noise. An adaptive filtering module is provided to eliminate the noise component in the audio signal s(k) to obtain an estimated voice signal e_s(k) with a higher S/N ratio. Furthermore, the adaptive filtering module can be also configured to suppress echo in the audio signal s(k) at same time. The voice signal e_s(k) may be further coupled to a single-channel voice enhancement module that is configured to eliminate any residual of the noise component in the voice signal e_s(k) according to the differences between the voice signal and the noise signal in time domain and frequency domain, whereby, the S/N ratio is further enhanced.
Description

BRIEF DESCRIPTION OF THE DRAWINGS

These and other features, aspects, and advantages of the present invention will become better understood with regard to the following description, appended claims, and accompanying drawings where:



FIG. 1 is a functional block diagram of processing signals from a dual microphone system according to one embodiment of the present invention;



FIG. 2A is a functional block diagram showing how to train an adaptive filter into a compensation filter;



FIG. 2B shows an exemplary adjusting process that may be used in the functional block diagram of FIG. 2A;



FIG. 3 shows that two signals from two microphones MIC A and MIC B are coupled to an average energy comparator that calculates respective average energy of the two signals in a short time frame;



FIG. 4 shows a functional block diagram of determining an estimated audio signal and a noise signal from two processed signals from two microphones;



FIG. 5 shows modules configured to realize an MT/N fractional delay; and



FIG. 6 shows a linear latter filtering module that may be used in the functional block diagram of FIG. 1.


Claims
  • 1. A method for voice enhancement, the method comprising: obtaining two audio signals from two microphones;adjusting the two audio signals so that characteristics of the two audio signals are substantially similar;producing from the two audio signals a first audio signal mainly containing a voice signal and a second audio signal mainly containing a noise signal according to differences between a voice source and a noise source in a space domain;eliminating the noise signal mixed in the first audio signal to produce a voice signal with a S/N ratio; andenhancing the voice signal in a single-channel voice enhancement module so that the S/N ratio in the voice signal is further enhanced.
  • 2. The method as claimed in claim 1, wherein the two microphones are in a communication device, one of the two microphones is primarily for receiving the voice signal and the other one of the two microphones is primarily for receiving the noise signal.
  • 3. The method as claimed in claim 1, wherein said adjusting the two audio signals comprises adjusting respective gains of the two audio signals.
  • 4. The method as claimed in claim 1, further comprising eliminating the noise signal in the voice signal according to differences between the voice signal and the noise signal in either one or both of a time domain and a frequency domain.
  • 5. The method as claimed in claim 1, wherein the two audio signals are labeled, respectively, as x1(k) and x2(k), and the two corresponding adjusted audio signals are labeled respectively, as x′1(k) and x′2(k), said producing from the two audio signals a first audio signal and a second audio signal is performed in accordance with equations as follows:
  • 6. The method as claimed in claim 5, further comprising: adding N−1 zeros between any two points in N times upper sampling the signal x(k); andgetting N times upper sampling the signal x′(k).
  • 7. The method as claimed in claim 6, further comprising: using a low pass filter H2(k) to filter a mirror frequency component brought in from said upper sampling,limiting a signal bandwidth to f0/2; andoutputting a signal w1(k).
  • 8. The method as claimed in claim 7, still further comprising: delaying the signal w1(k) by M points to obtain a signal w2(k);doing N times abstraction to w2(k) through an N times down sampling device;getting a first output signal;getting a second output signal in the same way as getting the first output; andcomparing and balancing respective energies of both first and second signals.
  • 9. The method as claimed in claim 5, further comprising: comparing respective energy values of the signal s(k) and the signal n(k) to generate an adaptive filter H3(k) enable control signal Adapt_en, wherein the control signal Adapt_en is used to control whether an adaptive filter coefficient shall be updated;delaying the signal s(k) to get a delayed signal s′(k);adaptively filtering the signal n(k) to get a signal n′(k); andadding the signal s′(k) and the signal n′(k) to get an estimated signal e_s(k).
  • 10. The method as claimed in claim 9, wherein the signal Adapt_en is used to assure that the adaptive filter coefficient adjusted is not aimed at the voice signal but the noise signal.
  • 11. A device for voice enhancement, the device comprising: a separation module for separating two input audio signals x′1(k) and x2′(k) to produce a first audio signal s(k) mainly containing voice and a second audio signal n(k) mainly containing noise according to differences between a voice source and a noise source in an air domain; andan adaptive filtering module for eliminating the noise mixed in the first audio signal s(k) according to relativity of the noise contained in the first audio signal s(k), to produce a voice signal e_s(k).
  • 12. The device as claimed in claim 11, further comprising: an adjusting module for adjusting a gain value of either one or both of the two audio signals according to differences between the two audio signal; anda voice enhancement module for eliminating the noise in the voice signal e_s(k) according to differences between voice signal and noise signal in time domain and frequency domain.
Priority Claims (1)
Number Date Country Kind
200610001158.6 Jan 2006 CN national