1. A method for processing an audio signal received by a hearing assistance device, the method comprising:
filtering the audio signal to generate a high frequency filtered signal, the filtering performed at a splitting frequency;
transposing at least a portion of an audio spectrum of the filtered signal to a lower frequency range by a transposition process to produce a transposed audio signal; and
summing the transposed audio signal with the audio signal to generate an output signal,
wherein the transposition process uses a prototype spectral peak modulated by a sinusoid having a time varying amplitude and frequency to control a center frequency and magnitude of a translated spectral feature.
2. The method of claim 1, wherein using the prototype spectral peak includes using a noise signal configured to be filtered to obtain a narrowband, low pass noise signal.
3. The method of claim 1, wherein using the prototype spectral peak includes using random samples generated at a decimated rate and up-sampled using interpolation or smoothing.
4. The method of claim 1, wherein using the prototype spectral peak includes using a single fixed-coefficient prototype low pass filter.
5. The method of claim 1, wherein modulating the prototype spectral peak includes using a look up table.
6. The method of claim 1, wherein the bandwidth and shape of the prototype spectral peak does not vary with the magnitude of the translated spectral feature.
7. The method of claim 1, wherein using the prototype spectral peak includes producing two or more peaks at different magnitudes without changing shape of either peak, to provide distinctive spectra for different translated sounds.
8. The method of claim 1, comprising using a weighted sum of narrowband spectra to compute a translated spectrum.
9. The method of claim 1, comprising multiplying by a sum of two or more sinusoids to render multiple translated peaks simultaneously.
10. The method of claim 1, comprising using a translated spectral envelope filter to increase low frequency energy suppressed by the splitting filter.
11. The method of claim 1, wherein the center frequency and magnitude of the translated spectral feature are configured to be updated for each block.
12. The method of claim 1, wherein the prototype spectral peak has a width of approximately 500 Hz, and is produced using a spectral envelope filter having a bandwidth of approximately 2 to 3 kHz.
13. A system for processing an audio signal received by a hearing assistance device, the system comprising:
a digital signal processor programmed to perform the steps of:
filtering the audio signal to generate a high frequency filtered signal, the filtering performed at a splitting frequency;
transposing at least a portion of an audio spectrum of the filtered signal to a lower frequency range by a transposition process to produce a transposed audio signal; and
summing the transposed audio signal with the audio signal to generate an output signal,
wherein the transposition process uses a prototype spectral peak modulated by a sinusoid having a time varying amplitude and frequency to control a center frequency and magnitude of a translated spectral feature.
14. The system of claim 13, wherein the hearing assistance device includes a hearing aid.
15. The system of claim 14, wherein the hearing aid includes an in-the-ear (ITE) hearing aid.
16. The system of claim 14, wherein the hearing aid includes a behind-the-ear (BTE) hearing aid.
17. The system of claim 14, wherein the hearing aid includes an in-the-canal (ITC) hearing aid.
18. The system of claim 14, wherein the hearing aid includes a receiver-in-canal (RIC) hearing aid.
19. The system of claim 14, wherein the hearing aid includes a completely-in-the-canal (CIC) hearing aid.
20. The system of claim 14, wherein the hearing aid includes a receiver-in-the-ear (RITE) hearing aid.
The claims below are in addition to those above.
All refrences to claim(s) which appear below refer to the numbering after this setence.
1. A communication system implementing voice over an Internet Protocol comprising:
a) an IP network,
b) a TDM source stream;
c) decoder to decode said TDM source stream;
d) a converter to convert and strip call progress tones into a separate data form;
e) an encrypterdecrypter to encrypt voice packets;
f) a compressor to compress remaining voice; and
g) a packetizer to form packets in output that is in an IP compatible format suitable for transfer over said IP Network.
2. The communication system of claim 1 further comprising means for silence suppression wherein said silence suppression is performed prior to voice stream compression.
3. The communication system of claim 1 further comprising a receiving section acquiring said packets from said packetizer and transferring said packets across said IP network.
4. The communication system of claim 1 further comprising out of band signaling of a maximum of seven commands and a command length less than ten bytes.
5. The communication system of claim 1 wherein said TDM source stream is an E1T1PRI TDM stream.
6. The communication system of claim 1 wherein said packetizer packets cells into UDP over IP frames.
7. The communication system of claim 1 wherein said TDM source stream originates at either a receiving station or a sending station.
8. The communication system of claim 3 wherein said receiving section further comprises:
a) a cell extractor to strip the cells from a UDP payload;
b) a reassembler to restructure stripped cells into their correct sequence;
c) a decompressor to decompress compressed voice to PCM;
d) a tone generator for the reinsertion of call progress tones into a decompressed voice; and
e) a framer and encoder.
9. The communication system of claim 3 wherein output from said framer and encoder is transmitted as PCM voice.
10. A communication system implementing voice over an Internet Protocol comprising:
a) an IP network;
b) a TDM source stream;
c) decoder to decode said TDM source stream;
d) a converter to convert and strip call progress tones into a separate data form;
e) an encrypterdecrypter to encrypt voice packets;
f) a compressor to compress remaining voice;
g) a packetizer to form packets in output that is in an IP compatible format suitable for transfer over said IP Network;
h) means for silence suppression wherein said silence suppression is performed prior to voice stream compression;
i) a receiving section acquiring said packets from said packetizer and transferring said packets across said IP network; and
j) out of band signaling of a maximum of seven commands and a command length less than ten bytes
11. The communication system of claim 10 wherein said TDM source stream is a E1T1PRI TDM stream.
12. The communication system of claim 10 wherein said packetizer packets cells into UDP over IP frames.
13. The communication system of claim 10 wherein said TDM source stream originates at either a receiving station or a sending station.
14. The communication system of claim 10 wherein said receiving section further comprises:
a) a cell extractor to strip the cells from a UDP payload;
b) a reassembler to restructure stripped cells into their correct sequence;
c) a decompressor to decompress compressed voice to PCM;
d) a tone generator for the reinsertion of call progress tones into a decompressed voice; and
e) a framer and encoder, wherein output from said framer and encoder is transmitted as PCM voice.
15. A communication system implementing voice over an Internet Protocol comprising:
a) an IP network;
b) a TDM source stream;
c) decoder to decode said TDM source stream;
d) a converter to convert and strip call progress tones into a separate data form;
e) an encrypterdecrypter to encrypt voice packets;
f) a compressor to compress remaining voice;
g) a packetizer to form packets in output that is in an IP compatible format suitable for transfer over said IP Network;
h) means for silence suppression wherein said silence suppression is performed prior to voice stream compression;
i) a receiving section acquiring said packets from said packetizer and transferring said packets across said IP network;
j) out of band signaling of a maximum of seven commands and a command length less than ten bytes;
k) said TDM source stream being a E1T1PRI TDM stream;
l) said packetizer packets cells into UDP over IP frames; and
m) said TDM source stream originates at either a receiving station or a sending station.
16. The communication system of claim 15 wherein said receiving section further comprises:
a) a cell extractor to strip the cells from a UDP payload;
b) a reassembler to restructure stripped cells into their correct sequence;
c) a decompressor to decompress compressed voice to PCM;
d) a tone generator for the reinsertion of call progress tones into a decompressed voice; and
e) a framer and encoder, wherein output from said framer and encoder is transmitted as PCM voice.
17. A communication system implementing voice over an Internet Protocol comprising:
a) an IP network;
b) a TDM source stream;
c) decoder to decode said TDM source stream;
d) a converter to convert and strip call progress tones into a separate data form;
e) an encrypterdecrypter to encrypt voice packets;
f) a compressor to compress remaining voice;
g) a packetizer to form packets in output that is in an IP compatible format suitable for transfer over said IP Network;
h) means for silence suppression wherein said silence suppression is performed prior to voice stream compression;
i) a receiving section acquiring said packets from said packetizer and transferring said packets across said IP network;
j) out of band signaling of a maximum of seven commands and a command length less than ten bytes;
k) said TDM source stream being a E1T1PRI TDM stream;
l) said packetizer packets cells into UDP over IP frames;
m) said TDM source stream originates at either a receiving station or a sending station; and
n) said receiving section having a cell extractor to strip the cells from a UDP payload; a reassembler to restructure stripped cells into their correct sequence; a decompressor to decompress compressed voice to PCM; a tone generator for the reinsertion of call progress tones into a decompressed voice; and a framer and encoder, wherein output from said framer and encoder is transmitted as PCM voice.