# US 10,245,430 B2 · Signal processing method for cochlear implant

> Agents: this data is available as JSON from the patentgrep API and MCP server — see https://patentgrep.com/llms.txt

- Number: US 10,245,430 B2
- Document ID: US10245430B2
- Status: Granted
- Kind: B2
- Application number: 15/693,351
- Filed: 2017-08-31
- Published: 2017-12-21
- Granted: 2019-04-02
- Earliest claimed: 2015-08-27
- Assignees: iMEDI PLUS Inc.
- Applicants: iMEDI PLUS Inc.
- Inventors: Kun-Hsi Tsai
- Web page: https://patentgrep.com/patent/US10245430B2

## CPC

- A61N 1/36038: {Cochlear stimulation}
- G10L 19/00: Speech or audio signals analysis-synthesis techniques for redundancy reduction, e.g. in vocoders; Coding or decoding of speech or audio signals, using source filter models or psychoacoustic analysis (in musical instruments G10H)
- G10L 21/0224: Processing in the time domain
- G10L 21/0208: Noise filtering
- G10L 21/0324: Details of processing therefor
- H04R 25/505: {using digital signal processing}
- H04R 2225/43: Signal processing in hearing aids to enhance the speech intelligibility
- H04R 2225/67: Implantable hearing aids or parts thereof not covered by H04R25/606
- H04R 2410/00: Microphones
- A61N: ELECTROTHERAPY; MAGNETOTHERAPY; RADIATION THERAPY; ULTRASOUND THERAPY (measurement of bioelectric currents A61B; surgical instruments, devices or methods for transferring non-mechanical forms of energy to or from the body A61B18/00; anaesthetic apparatus in general A61M; incandescent lamps H01K; infrared radiators for heating H05B)
- G10L: SPEECH ANALYSIS TECHNIQUES OR SPEECH SYNTHESIS; SPEECH RECOGNITION; SPEECH OR VOICE PROCESSING TECHNIQUES; SPEECH OR AUDIO CODING OR DECODING
- H04R: LOUDSPEAKERS, MICROPHONES, GRAMOPHONE PICK-UPS OR LIKE ACOUSTIC ELECTROMECHANICAL TRANSDUCERS; ELECTRIC HEARING AIDS; PUBLIC ADDRESS SYSTEMS (producing sounds with frequency not determined by supply frequency G10K)

## Abstract

A signal processing method for cochlear implant is performed by a speech processor and comprises a noise reduction stage and a signal compression stage. The noise reduction stage can efficiently reduce noise in a electrical speech signal of a normal speech. The signal compression stage can perform good signal compression to enhance signals to stimulate cochlear nerves of a patient with hearing loss. The patient who uses a cochlear implant performing the signal processing method of the present disclosure can accurately hear normal speech.

## Claims

### Claim 1

A signal processing method for a cochlear implant, the cochlear implant comprising a microphone and a speech processor, the signal processing method being executed by the speech processor comprising:
receiving an electrical speech signal from the microphone;
segmenting the electrical speech signal to a plurality of time-sequenced noisy frames;
reducing noise in each of the plurality of time-sequenced signal frames to obtain a plurality of clean signal frames, the plurality of clean signal frames comprising a (t-1)-th clean frame xt-1 and a t-th clean frame xt;
obtaining a (t-1)-th compression factor αt-1 according to the (t-1)-th clean frame xt-1;
obtaining a t-th compression factor αt for the t-th clean frame xt according to the compression factor αt-1 and the t-th clean frame xt;
obtaining a t-th output frame zt based on the t-th compression factor αt; and
outputting the t-th output frame zt.

### Claim 2 (depends on 1)

The signal processing method of claim 1, further comprising:
obtaining a (t-1)-th amplitude envelope of the (t-1)-th clean frame xt-1 and calculating a (t-1)-th upper boundary and a(t-1)-th lower boundary of the (t-1)-th amplitude envelope;
wherein the (t-1)-th compression factor αt-1 for the (t-1)-th clean frame xt-1 is obtained based on the (t-1)-th upper boundary and the (t-1)-th lower boundary.

### Claim 3 (depends on 2)

The signal processing method of claim 2, further comprising:
obtaining a t-th amplitude envelope of the t-th clean frame xt and calculating a t-th upper boundary and a t-th lower boundary of the t-th amplitude envelope;
wherein the t-th compression factor αt for the t-th clean frame xt is obtained based on the compression factor αt-1, the t-th upper boundary and the t-th lower boundary.

### Claim 4 (depends on 3)

The signal processing method of claim 3, wherein when the t-th output frame zt falls within a range between a the t-th upper boundary and the t-th lower boundary, the t-th compression factor αt is calculated by: αt=αt-1+Δα1, and Δα1 is a positive value.

### Claim 5 (depends on 3)

The signal processing method of claim 3, wherein when the t-th output frame zt falls beyond a range between a the t-th upper boundary and the t-th lower boundary, the t-th compression factor αt is calculated by: αt=αt-1+Δα2, and Δα2 is a negative value.

### Claim 6 (depends on 1)

The signal processing method of claim 1, wherein the t-th output frame zt, is obtained by: zt=αt×(xt−xt)+xt, and xt is a mean of the t-th amplitude envelope of the t-th clean frame xt.

### Claim 7 (depends on 1)

The signal processing method of claim 1, wherein the t-th clean frame xt is calculated by:
xt=InvF{(W2′h′(F{yt})+b2′)}
wherein F{} is a Fourier transform function to transfer the t-th noisy frame yt from time domain to frequency domain;
h′( )is a function including W1′ and b1′;
W1′ and W2′ are default connection weights in frequency domain;
b1′ and b2′ are default vectors of biases of hidden layers of a DDAE-based NR structure in the frequency domain; and
InvF { } is an inverse Fourier transform function.

### Claim 8 (depends on 7)

The signal processing method of claim 7, wherein the h′(F{yt}) is calculated by:
h
′
⁡
(
F
⁢
{
y
t
}
)
=
1
1
+
exp
⁡
[
-
(
W
1
′
⁢
F
⁢
{
y
t
}
+
b
1
′
)
]
.

### Claim 9 (depends on 1)

The signal processing method of claim 1, wherein the t-th upper boundary (UB) is calculated by UB=xt+α0×(max(xt)−xt), the t-th lower boundary (LB) is calculated by LB=xt+α0×(max(xt)−xt), and xt is a mean of the t-th amplitude envelope of the t-th clean frame xt.