# US 9,520,139 B2 · Post tone suppression for speech enhancement

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- Number: US 9,520,139 B2
- Document ID: US9520139B2
- Status: Granted
- Kind: B2
- Application number: 14/740,735
- Filed: 2015-06-16
- Published: 2015-12-24
- Granted: 2016-12-13
- Earliest claimed: 2014-06-19
- Applicants: Yang Gao
- Inventors: Yang Gao
- Web page: https://patentgrep.com/patent/US9520139B2

## CPC

- G10L 21/0208: Noise filtering
- H04M 9/082: {using echo cancellers (echo cancellers per se H04B3/23)}
- G10L 21/0232: Processing in the frequency domain
- G10L 21/0264: characterised by the type of parameter measurement, e.g. correlation techniques, zero crossing techniques or predictive techniques
- G10L 25/12: the extracted parameters being prediction coefficients
- G10L 25/90: Pitch determination of speech signals
- G10L 2021/02082: {the noise being echo, reverberation of the speech}
- G10L: SPEECH ANALYSIS TECHNIQUES OR SPEECH SYNTHESIS; SPEECH RECOGNITION; SPEECH OR VOICE PROCESSING TECHNIQUES; SPEECH OR AUDIO CODING OR DECODING
- H04M: TELEPHONIC COMMUNICATION (circuits for controlling other apparatus via a telephone cable and not involving telephone switching apparatus G08)

## 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

### Claim 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 wherein the parameters comprise at least one of 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.

### Claim 2 (depends on 1)

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

### Claim 3 (depends on 1)

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.

### Claim 4 (depends on 1)

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:
Sharp_ch
=
∑
n
=
0
L
-
1
⁢

S
ch
⁡
(
n
)

L
·
Max
⁢
{

S
ch
⁡
(
n
)

,
n
=
0
,
1
,
…
⁢
,
L
-
1
}
wherein Sch (n) is the component channel signal, and L is the fame length.

### Claim 5 (depends on 1)

The method of claim 1, wherein the normalized pitch correlation is calculated on the component channel signal in time domain:
Rp_ch
=
∑
n
⁢
S
ch
⁡
(
n
)
·
S
ch
⁡
(
n
-
Pitch
)
[
∑
n
⁢
S
ch
⁡
(
n
)
2
]
⁡
[
∑
n
⁢
S
ch
⁡
(
n
-
Pitch
)
2
]
wherein Pitch is estimated based on the full band signal (not the component channel signal).

### Claim 6 (depends on 1)

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.

### Claim 7 (depends on 1)

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

### Claim 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 wherein the parameters comprise at least one of 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.

### Claim 9 (depends on 8)

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

### Claim 10 (depends on 8)

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

### Claim 11 (depends on 8)

The speech signal processing apparatus 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:
Sharp_ch
=
∑
n
=
0
L
-
1
⁢

S
ch
⁡
(
n
)

L
·
Max
⁢
{

S
ch
⁡
(
n
)

,
n
=
0
,
1
,
…
⁢
,
L
-
1
}
wherein Sch (n) is the component channel signal, and L is the fame length.

### Claim 12 (depends on 8)

The speech signal processing apparatus of claim 8, wherein the normalized pitch correlation is calculated on the component channel signal in time domain:
Rp_ch
=
∑
n
⁢
S
ch
⁡
(
n
)
·
S
ch
⁡
(
n
-
Pitch
)
[
∑
n
⁢
S
ch
⁡
(
n
)
2
]
⁡
[
∑
n
⁢
S
ch
⁡
(
n
-
Pitch
)
2
]
wherein Pitch is estimated based on the full band signal (not the component channel signal).

### Claim 13 (depends on 8)

The speech signal processing apparatus 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.

### Claim 14 (depends on 8)

The speech signal processing apparatus of claim 8, wherein the reduction gain is larger than or equal to zero and smaller than or equal to one.