Noise adaptive post filtering
US-2015142425-A1 · May 21, 2015 · US
US10433076B2 · US · B2
| Field | Value |
|---|---|
| Publication number | US-10433076-B2 |
| Application number | US-201916291899-A |
| Country | US |
| Kind code | B2 |
| Filing date | Mar 4, 2019 |
| Priority date | May 30, 2016 |
| Publication date | Oct 1, 2019 |
| Grant date | Oct 1, 2019 |
A practical reading order for non-experts. Skip the full description unless you need deep technical detail.
What the patent document calls the invention.
A short plain-language summary of the technical disclosure.
Who owns or filed the patent and who is credited as inventor.
Filing, priority, publication, and grant dates set the timeline.
The legal scope of protection — read this for what is actually claimed.
Technology tags used to group this patent with similar filings.
Prior art links and similar publications in this corpus.
Official abstract text for this publication.
An audio processing device comprises a) at least one input unit for providing a time-frequency representation Y(k,n) of an electric input signal representing sound consisting of target speech and noise signal components, where k and n are frequency band and time frame indices, respectively, b) a noise reduction system configured to: determine a first signal to noise ratio estimate γ(k,n) of said electric input signal, and determine a second signal to noise signal ratio estimate ζ(k,n) of said electric input signal from said first signal to noise ratio estimate γ(k,n) based on a recursive algorithm providing non-linear smoothing, and wherein parameters of said smoothing are determined in dependence of the first and/or the second signal to noise ratio estimates corresponding to a multitude of frequency band indices. The invention may be used in hearing aids, headsets, ear phones, active ear protection systems, handsfree telephone systems, mobile telephones, etc.
Opening claim text (preview).
The invention claimed is: 1. An audio processing device, comprising at least one input unit for providing a time-frequency representation Y(k,n) of an electric input signal representing a time variant sound signal consisting of target speech signal components S(k,n) from a target sound source TS and noise signal components N(k,n) from other sources than the target sound source, where k and n are frequency band and time frame indices, respectively, a noise reduction system configured to determine a first signal to noise ratio estimate γ(k,n) of said electric input signal, determine second signal to noise ratio estimate ζ(k,n) of said electric input signal from said first signal to noise ratio estimate γ(k,n) based on a recursive algorithm comprising a recursive loop, and to determine said second signal to noise ratio estimate ζ(k,n) by non-linear smoothing of said first signal to noise ratio estimate γ(k,n), or a parameter derived therefrom, and wherein said non-linear smoothing is controlled by one or more bias and/or smoothing parameters; and a selector located in the recursive loop, wherein said selector is configured to select an input to determine said one or more bias and/or smoothing parameters based on a select control parameter; and wherein said select control parameter for a given frequency index k is determined in dependence of the first and/or the second signal to noise ratio estimates corresponding to a multitude of frequency indices. 2. An audio processing device according to claim 1 wherein said multitude of frequency indices includes one or more neighboring frequency indices. 3. An audio processing device according to claim 1 wherein said multitude of frequency indices comprises the immediately neighboring frequency indices (k−1, k, k+1). 4. An audio processing device according to claim 1 wherein said one or more neighboring frequency indices are determined according to a predefined or adaptive scheme. 5. An audio processing device according to claim 1 wherein said select control parameter for a given frequency index k is additionally determined in dependence of inputs from one or more detectors. 6. An audio processing device according to claim 5 wherein said one or more detectors comprise a general onset detector for detecting sudden changes in the time variant input sound, e.g. its modulation, a wind noise detector, a voice detector, e.g. an own voice detector, a head movement detector, a wireless transmission detector, voice detectors from microphones in other audio devices, and combinations thereof. 7. An audio processing device according to claim 5 wherein at least one of said one or more detectors is based on binaural detection. 8. An audio processing device according to claim 1 wherein said one or more bias and/or smoothing parameters at a given time instant n is determined in dependence of a first maximum likelihood estimate of the second signal to noise ratio estimate at that time instant n, and/or an estimate of the second signal to noise ratio estimate at the previous time instant n−1, selected by said select control parameter. 9. An audio processing device according to claim 1 configured to provide a noise reduction gain G NR in dependence of said second signal to noise ratio estimate ζ(k,n), and to apply said noise reduction gain G NR to said electric input signal or a signal derived therefrom. 10. An audio processing device according to claim 1 wherein said noise reduction system is configured to determine said second signal to noise ratio estimate ζ(k,n) for the n th time frame under the assumption that said first signal to noise ratio γ(k,n) is larger than or equal to 1. 11. An audio processing device according to claim 1 comprising a filter bank comprising an analysis filter bank for providing said time-frequency representation Y(k,n) of said electric input signal. 12. An audio processing device according to claim 11 configured to provide that said analysis filter bank is oversampled. 13. An audio processing device according to claim 1 comprising a hearing device, e.g. a hearing aid. 14. A method of estimating an a priori signal to noise ratio ζ(k,n) of a time-frequency representation Y(k,n) of an electric input signal representing a time variant sound signal consisting of target speech components and noise components, where k and n are frequency band and time frame indices, respectively, the method comprising determining a first signal to noise ratio estimate γ(k,n) of said electric input signal Y(k,n); determining a second signal to noise signal ratio estimate ζ(k,n) of said electric input signal from said first signal to noise ratio estimate γ(k,n) based on a recursive algorithm; determining said second signal to noise ratio estimate ζ(k,n) by non-linear smoothing of said first signal to noise ratio estimate γ(k,n), or a parameter derived therefrom, and wherein said non-linear smoothing is controlled by one or more bias and/or smoothing parameters; and selecting an input to determine said one or more bias and/or smoothing parameters based on a select control parameter; wherein said select control parameter for a given frequency index k is determined in dependence of the first and/or the second signal to noise ratio estimates corresponding to a multitude of frequency indices. 15. A method according to claim 14 comprising providing a noise reduction gain G NR in dependence of said second signal to noise ratio estimate ζ(k,n); and applying said noise reduction gain G NR to said electric input signal or a signal derived therefrom. 16. A data processing system comprising a processor and program code means for causing the processor to perform the method of claim 14 . 17. A non-transitory computer readable medium storing a computer program comprising instructions which, when the program is executed by a computer, cause the computer to carry out the method of claim 14 . 18. An audio processing device, comprising at least one input unit for providing a time-frequency representation Y(k,n) of an electric input signal representing a time variant sound signal consisting of target speech signal components S(k,n) from a target sound source TS and noise signal components N(k,n) from other sources than the target sound source, where k and n are frequency band and time frame indices, respectively, a noise reduction system configured to determine a first signal to noise ratio estimate γ(k,n) of said electric input signal, determine second target signal to noise ratio estimate ζ(k,n) of said electric input signal from said first signal to noise ratio estimate γ(k,n) based on a recursive algorithm comprising a recursive loop, determine said second signal to noise ratio estimate ζ(k,n) by non-linear smoothing of said first signal to noise ratio estimate γ(k,n), or a parameter derived therefrom, and wherein said non-linear smoothing is controlled by one or more bias and/or smoothing parameters; and a selector located in the recursive loop, wherein said selector is configured to select an input to determine said one or more bias and/or smoothing parameters based on said select control parameter; and one or more detectors; wherein said select control parameter for a given frequency index is determined in dependence of inputs from said one or more detectors. 19. An audio processing device according to claim 18 wherein said one or more detectors comprise a general onset detector for detecting sudden changes in the time variant input sound, e.g. its modulation, a wind noise detector, a voi
Synergistic effects of band splitting and sub-band processing · CPC title
Array processing for suppression of unwanted side-lobes in directivity characteristics, e.g. a blocking matrix · CPC title
for combining the signals of two or more microphones (specially adapted for hearing aids H04R25/407) · CPC title
Signal processing in hearing aids to enhance the speech intelligibility · CPC title
for correcting frequency response · CPC title
Related publications grouped by family.
Answers are generated from the same data shown on this page.