Post Tone Suppression for Speech Enhancement

• US 20150371659A1
• Filed: 06/16/2015
• Published: 12/24/2015
• Est. Priority Date: 06/19/2014
• Status: Active Grant

• Current Assignee: Yang Gao
• Original Assignee: Yang Gao
• Inventors: Gao, Yang
• Granted Patent: US 9,520,139 B2
• Time in Patent Office: Days
• US Class Current: 1/1
• CPC Class Codes: G10L 2021/02082 the noise being echo, reverberation of the speech
• CPC Class Codes: G10L 21/0208 Noise filtering
• CPC Class Codes: G10L 21/0232 Processing in the frequency domain
• CPC Class Codes: G10L 21/0264 characterised by the type of parameter measurement, e.g. correlation techniques, zero crossing techniques or predictive techniques
• CPC Class Codes: G10L 25/12 the extracted parameters being prediction coefficients
• CPC Class Codes: G10L 25/90 Pitch determination of speech signals
• CPC Class Codes: H04M 9/082 using echo cancellers echo cancellers per se H04B3/23

Abstract

A method for reducing disturbing tone signals from acoustic echoes or background noises in speech/audio signal enhancement processing comprises decomposing a full-band input microphone speech signal into plurality of sub-band component channel signals by using a filter-bank; disturbing tone signal in each sub-band component channel signal is detected by using parameters such as a second reflection coefficient from LPC analysis, a time domain sharpness parameter, a normalized pitch correlation, and/or a SNR parameter; the energy of the current sub-band component channel signal is reduced by multiplying a reduction gain if the disturbing tone signal is detected in the current sub-band component channel signal; all the component channel signals are summed back to output a tone-reduced full band speech signal.

Claims

1. A method for reducing disturbing tone signals from acoustic echoes or background noises in speech/audio signal enhancement processing, the method comprising:

decomposing a full-band input microphone speech signal into plurality of sub-band component channel signals by using a filter-bank;

detecting disturbing tone signal in each sub-band component channel signal by using parameters such as a second reflection coefficient from LPC analysis, a time domain sharpness parameter, a normalized pitch correlation, and/or a SNR parameter;

reducing the energy of the current sub-band component channel signal by multiplying a reduction gain if the disturbing tone signal is detected in the current sub-band component channel signal;

summing back all the component channel signals to output a tone-reduced full band speech signal.

2. The method of claim 1, wherein reducing disturbing tone signals happens after acoustic echo cancellation and/or microphone array beamforming processing.

3. The method of claim 1, wherein the second reflection coefficient obtained from LPC analysis is calculated on the component channel signal with a Levinson-Durbin approach.

4. The method of claim 1, wherein the time domain sharpness parameter is defined as an average value divided by a maximum value wherein the average value and the maximum value are calculated on a frame of the component channel signal in time domain:

5. The method of claim 1, wherein the normalized pitch correlation is calculated on the component channel signal in time domain:

6. The method of claim 1, wherein the SNR is calculated in dB on a frame of the full band speech signal and/or the sub-band component channel signal.

7. The method of claim 1, wherein the reduction gain is larger than or equal to zero and smaller than or equal to one.

8. A speech signal processing apparatus comprising:

a processor; and

a computer readable storage medium storing programming for execution by the processor, the programming including instructions to;

decompose a full-band input microphone speech signal into plurality of sub-band component channel signals by using a filter-bank;

detect disturbing tone signal in each sub-band component channel signal by using parameters such as a second reflection coefficient from LPC analysis, a time domain sharpness parameter, a normalized pitch correlation, and/or a SNR parameter;

reduce the energy of the current sub-band component channel signal by multiplying a reduction gain if the disturbing tone signal is detected in the current sub-band component channel signal;

sum back all the component channel signals to output a tone-reduced full band speech signal.

9. The method of claim 8, wherein reducing disturbing tone signals happens after acoustic echo cancellation and/or microphone array beamforming processing.

10. The method of claim 8, wherein the second reflection coefficient obtained from LPC analysis is calculated on the component channel signal with a Levinson-Durbin approach.

11. The method of claim 8, wherein the time domain sharpness parameter is defined as an average value divided by a maximum value wherein the average value and the maximum value are calculated on a frame of the component channel signal in time domain:

12. The method of claim 8, wherein the normalized pitch correlation is calculated on the component channel signal in time domain:

13. The method of claim 8, wherein the SNR is calculated in dB on a frame of the full band speech signal and/or the sub-band component channel signal.

14. The method of claim 8, wherein the reduction gain is larger than or equal to zero and smaller than or equal to one.