[digitalradio] New release (4.5) of MULTIPSK (DTMF - ALE400)
New release (4.5) of MULTIPSK RX/TX: PSK10/BPSK31-63-125/QPSK31-63-125/CHIP (64/128)/PSKFEC31/PSKAM10-31-50/PSK63F - PSK220F + DIGISSTV Run/DTMF/CW/CCW/CCW-FSK/THROB/THROBX/DTMF/MFSK8/ MFSK16 (+ SSTV)/MIL-STD-188-141A (+ARQ FAE)/ALE400/OLIVIA/CONTESTIA/RTTYM/ VOICE/DominoF DF/DominoEX/MT63/RTTY 45/75/ RTTY 50+SYNOP+SHIP/ASCII/AMTOR FEC/ PACKET 110-300-1200 + APRS+ DIGISSTV Run/PACTOR 1-FEC/PAX+PAX2 + APRS/ FELD HELL/PSK HELL/FM HELL (105-245)/HELL 80/HF-FAX/SSTV/ RX only: AMTOR ARQ/NAVTEX/RTTY 100/1382/GMDSS DSC DSP: Filters + CW binaural reception PSK Panoramic (BPSK31/BPSK63/PSKFEC31): RX 23 channels simultaneously CW Panoramic: RX 8 or 23 channels simultaneously RTTY Panoramic: RX 8 RTTY QSO decoded simultaneously on 22 channels Programmation of Multipsk reception TCP/IP digital modem CLOCK 1.7.6 (FRANCE-INTER, DCF77, HBG, RUGBY, WWVB, WWV, WWVH, CHU, GPS, JJY) MULTIDEM 2.1.1 Modulator/demodulator for DSB and SdR transceiver Pour les francophones: la version française de ce message se trouve sur mon site (http://f6cte.free.fr). Il suffit de cliquer sur le lien Principales modifications (courriel avertissant de la sortie de la nouvelle version). Hello to all Ham and SWL, The new release of MULTIPSK (4.5), CLOCK (1.7.6) and MULTIDEM (2.1.1) are in my Web site (http://f6cte.free.fr). The main mirror site is Earl's, N8KBR: http://multipsk.eqth.info/index.html (click on United States). Another mirror site isTerry's: http://www.hamshack.co.uk/ Multispk associated to Clock are freeware programs but with functions submitted to a licence (by user key). CLOCK 1.7.6 has now a possibility to directly interface a SdR receiver through the sound card. MULTIDEM 2.1.1 fixes bugs. The main modifications of MULTIPSK 4.5 are the following: 1) Decoding/coding of the DTMF (Dual-tone multi-frequency) mode. This mode is used for telephone (to dial the phone number) but also in VHF and UHF for different uses as, for example, activation of repeaters by radio. It could be used in HF, for radio control of Ham equipments. Fonctions of DTMF handling on reception are available for licencied copies, only. See specifications further on. 2) New ALE400 mode (ALE in a 400 Hz bandwidth) This ALE system has exactly the same functions as the ones of the 141A of Multipsk except that: * the bandwidth is 400 Hz instead of 2000 Hz as in standard ALE (so ALE400 can be transmitted anywhere where 500 Hz digital modes are authorized), * the modulation speed is 50 bauds instead of 125 bauds and consequently the text throughput are 2.5 slower, * no fix frequency (as in MFSK16...), the automatic tuning being able to be done thanks to the RS/ID transmission, * the signal to noise ratio is 5 dB better: - 9 dB for sounding, AMD messages and Unproto mode, - 11.5 dB (- 13.5 dB with many repetitions) for ARQ FAE. For ARQ FAE, it has been added a compression system using a modified IZ8BLY (Nino) MFSK Varicode. So the ALE400 text throughput is typically 60 wpm (up to 107 wpm in bilateral and 63 characters frames). ARQ FAE covers all ASCII and ANSI characters (8 bits) There is a Word document which goal is to show from Multipsk snapshots how to do the basic operations in ALE and ALE400. This document (1.1 Mo) is available from my site site http://f6cte.free.fr/ALE_and_ALE400_easy_with_Multipsk.doc; (copy and paste this adress in Internet Explorer (or equivalent) Net adress field). Look also at http://hflink.com/ale400/; which is a specific page for ALE400, with a lot of information. ALE400 frequencies: 1837.0, 3589.0, 7037.5, 10141.5, 14074.0, 14094.0, 18104.5, 21094.0, 24926.0, 28146.0, 50162.5, 144162.5 (AF at 1625 Hz). 3) Direct SdR interface through the sound card It is the best way to do as there is no additional transmission delay. It allows the working in all digital modes. The modulation and demodulation operations of the I/Q signals coming from the sound card are directly done by Multipsk (which plays the role of a SdR program). The bandwidth considered is the USB side, in base band. So there is neither frequency shift nor consideration of the LSB side. After selection, the working is transparent for the user. 4) Rewind function This function makes you able to decode the signal from a point in time located before you click on the waterfall. It is permitted to select a rewind duration from 5 seconds to 3 minutes (from 20 seconds to 3 minutes for only the licencied copies). Powerful computers can take profit of this function, decoding of the rewind period being quick. For information, for all the Multipsk exotic modes (PSKFEC31, PSK10, PSKAM, PSK63F, PSK220F (+DIGISSTV), CCW-FSK, MFSK8, THROBX, DominoF, DominoEX, PAX, CHIP, Voice, Packet 110 bauds...), I propose the QRP frequency: 14075 Khz USB (AF around 1000 Hz), at 17h00 UTC. 73 Patrick DTMF Description : Duration of the symbol: variable, in general between 40 to 100 ms for the carrier and 20 to 60 ms for the silence
[digitalradio] Welcome to ve3lvv
Welcome to ve3lvv, DRCC # 1611. Andy K3UK.
[digitalradio] I Apologize
To the great group of Digital Pic Guys that we had on 7.178 on Sat. I apologize for not being able to have a net this Sunday morning. I apologize the the arrogant and rude hams that do contesting and don't listen to a freq before transmitting, and do splits without listening and move within 1 kc with 1500 watts. I apologize for CQ mag. for having such a contest and making any type of communications (other than thier contest) impossible. (and for making it three days long!) I apologize to hams in other countries for trashing ALL the US freqs with CQ Contest (etc) for 3 days. I apologize for the FCC for allowing this deliberate type of interference to go on and continue on ALL SSB freqs. (give them 100kc on each band and let them have at it) I hope to see you all on next Saturday 8am on 7.178 for more pictures and conversation. (and Sunday) Garrett/ AA0OI
Re: [digitalradio] Re: digital voice within 100 Hz bandwidth
Hi Vojtech, Thank you for your reply to my papers. I will do more work on the phonemes. The project I want to do uses new computers that were no available 10 years ago. Every 10 mS a decision is made to send a one or a zero. To make that decision I have 68 parallel FFT's running in the background. I believe the brain could handle mispronounce words better than you think. Mike On Nov 17, 2007 3:55 PM, r_lwesterfield [EMAIL PROTECTED] wrote: I have a few radios (ARC-210-1851, PSC-5D, PRC-117F) at work that operate in MELP for a vocoder – Mixed Excitation Linear Prediction. We have found MELP to be superior (more human-like voice qualities – less Charlie Brown's teacher) to LPC-10 but we use far larger bandwidths than 100 khz. I do not know how well any of this will play out at such a narrow bandwidth. Listening to Charlie Brown's teacher will send you running away quickly and you should think of your listeners . . . they will tire very quickly. Just because voice can be sent at such narrower bandwidths does not necessarily mean that people will like to listen to it. Rick – KH2DF -- *From:* digitalradio@yahoogroups.com [mailto:[EMAIL PROTECTED] *On Behalf Of *Vojtech Bubník *Sent:* Saturday, November 17, 2007 9:11 AM *To:* [EMAIL PROTECTED]; digitalradio@yahoogroups.com *Subject:* [digitalradio] Re: digital voice within 100 Hz bandwidth Hi Mike. I studied some aspects of voice recognition about 10 years ago when I thought of joining a research group at Czech Technical University in Prague. I have a 260 pages text book on my book shelf on voice recognition. Voice signal has high redundancy if compared to a text transcription. But there is additional information stored in the voice signal like pitch, intonation, speed. One could estimate for example mood of the speaker from the utterance. Voice tract could be described by a generator (tone for vowels, hiss for consonants) and filter. Translating voice into generator and filter coefficients greatly decreases voice data redundancy. This is roughly the technique that the common voice codecs do. GSM voice compression is a kind of Algebraic Code Excited Linear Prediction. Another interesting codec is AMBE (Advanced Multi-Band Excitation) used by DSTAR system. GSM half-rate codec squeezes voice to 5.6kbit/sec, AMBE to 3.6 kbps. Both systems use excitation tables, but AMBE is more efficient and closed source. I think the clue to the efficiency is in size and quality of the excitation tables. To create such an algorithm requires considerable amount of research and data analysis. The intelligibility of GSM or AMBE codecs is very good. You could buy the intelectual property of the AMBE codec by buying the chip. There are couple of projects running trying to built DSTAR into legacy transceivers. About 10 years ago we at OK1KPI club experimented with an echolink like system. We modified speakfreely software to control FM transceiver and we added web interface to control tuning and subtone of the transceiver. It was a lot of fun and a very unique system at that time. http://www.speakfreely.org/ The best compression factor offers LPC-10 codec (3460kbps), but the sound is very robot-like and quite hard to understand. At the end we reverted to GSM. I think IVOX is a variant of the LPC system that we tried. Your proposal is to increase compression rate by transmitting phonemes. I once had the same idea, but I quickly rejected it. Although it may be a nice exercise, I find it not very useless until good continuous speech multi-speaker multi-language recognition systems are available. I will try to explain my reasoning behind that statement. Let's classify voice recognition systems by the implementation complexity: 1) Single-speaker, limited set of utterances recognized (control your desktop by voice) 2) Multiple-speaker, limited set of utterances recognized (automated phone system) 3) dictating system 4) continuous speech transcription 5) speech recognition and understanding Your proposal will need implement most of the code from 4) or 5) to be really usable and it has to be reliable. State of the art voice recognition systems use hidden Markov models to detect phonemes. Phoneme is searched by traversing state diagram by evaluating multiple recorded spectra. The phoneme is soft-decoded. Output of the classifier is a list of phonemes with their probabilities of detection assigned. To cope with phoneme smearing on their boundaries, either sub-phonemes or phoneme pairs need to be detected. After the phonemes are classified, they are chained into words. Depending on the dictionary, most probable words are picked. You suppose that your system will not need it. But the trouble are consonants. They carry much less energy than vowels and are much easier to be confused. Dictionary is used to pick some second highest probability detected consonants in the word.
[digitalradio] Slow MFSK?
OK, I give up. What is the slow MFSK I am seeing on 20 meters lately. It idles at the lowest frequency an looks like there may be around frequencies used? Gary N0GW
[digitalradio] Slow MFSK?
OK, I give up. What is the slow MFSK I am seeing on 20 meters lately. It idles at the lowest frequency an looks like there may be around 16 frequencies used? Gary N0GW
[digitalradio] Re: Slow MFSK?
Gary If its around 14.076 then its JT65A. You will find stations TX/RX in even/odd minute segments. Google JT65A for the software package called WSJT by K1JT. 73 Paul --- In digitalradio@yahoogroups.com, grwescom [EMAIL PROTECTED] wrote: OK, I give up. What is the slow MFSK I am seeing on 20 meters lately. It idles at the lowest frequency an looks like there may be around 16 frequencies used? Gary N0GW
Re: [digitalradio] Slow MFSK?
This is likely a mode developed for moonbounce/meteor scatter that some have been using on HF to see another hams callsigns and signal report with weak signals. JT-65A perhaps? 73, Rick, KV9U grwescom wrote: OK, I give up. What is the slow MFSK I am seeing on 20 meters lately. It idles at the lowest frequency an looks like there may be around 16 frequencies used? Gary N0GW
[digitalradio] Re: Slow MFSK?
Cool! Thanks Paul. Gary --- In digitalradio@yahoogroups.com, paul181696 [EMAIL PROTECTED] wrote: Gary If its around 14.076 then its JT65A. You will find stations TX/RX in even/odd minute segments. Google JT65A for the software package called WSJT by K1JT. 73 Paul
RE: [digitalradio] I Apologize
Which CQ Magazine contest are you referring to that runs for 3 days and was running yesterday morning? There was no contest shown on their website and usually their contests are 48 hours not 72. Just curious. Barry VE3CDX/W7 _ From: digitalradio@yahoogroups.com [mailto:[EMAIL PROTECTED] On Behalf Of aa0oi Sent: Sunday, November 18, 2007 6:34 AM To: digitalradio@yahoogroups.com Subject: [digitalradio] I Apologize To the great group of Digital Pic Guys that we had on 7.178 on Sat. I apologize for not being able to have a net this Sunday morning. I apologize the the arrogant and rude hams that do contesting and don't listen to a freq before transmitting, and do splits without listening and move within 1 kc with 1500 watts. I apologize for CQ mag. for having such a contest and making any type of communications (other than thier contest) impossible. (and for making it three days long!) I apologize to hams in other countries for trashing ALL the US freqs with CQ Contest (etc) for 3 days. I apologize for the FCC for allowing this deliberate type of interference to go on and continue on ALL SSB freqs. (give them 100kc on each band and let them have at it) I hope to see you all on next Saturday 8am on 7.178 for more pictures and conversation. (and Sunday) Garrett/ AA0OI
Re: [digitalradio] Re: digital voice within 100 Hz bandwidth
There are several (military/gov) standard intelligibility tests that do a pretty good job of scoring what most humans can and can not reliably understand. You might try taking a look at them to get some ideas of which voice characteristics make the most difference to intelligibility. There is actually a surprising amount of data out there, especially if you include the data peripheral to the various computerized speech translator research projects. It's not *exactly* signal processing... but understanding what parts of the signal matter the most can be surprisingly helpful. This may be unusually productive, because as of yet there hasn't been a huge amount of cross-discipline work between the codec researchers and the speech-to-meaning researchers. While there's a lot of duplicate research in there, it tends to be from slightly different perspectives, and the stereo view can sometimes help. On Nov 18, 2007 9:12 AM, Mike Lebo [EMAIL PROTECTED] wrote: Hi Vojtech, Thank you for your reply to my papers. I will do more work on the phonemes. The project I want to do uses new computers that were no available 10 years ago. Every 10 mS a decision is made to send a one or a zero. To make that decision I have 68 parallel FFT's running in the background. I believe the brain could handle mispronounce words better than you think. Mike On Nov 17, 2007 3:55 PM, r_lwesterfield [EMAIL PROTECTED] wrote: I have a few radios (ARC-210-1851, PSC-5D, PRC-117F) at work that operate in MELP for a vocoder – Mixed Excitation Linear Prediction. We have found MELP to be superior (more human-like voice qualities – less Charlie Brown's teacher) to LPC-10 but we use far larger bandwidths than 100 khz. I do not know how well any of this will play out at such a narrow bandwidth. Listening to Charlie Brown's teacher will send you running away quickly and you should think of your listeners . . . they will tire very quickly. Just because voice can be sent at such narrower bandwidths does not necessarily mean that people will like to listen to it. Rick – KH2DF From: digitalradio@yahoogroups.com [mailto:[EMAIL PROTECTED] On Behalf Of Vojtech Bubník Sent: Saturday, November 17, 2007 9:11 AM To: [EMAIL PROTECTED]; digitalradio@yahoogroups.com Subject: [digitalradio] Re: digital voice within 100 Hz bandwidth Hi Mike. I studied some aspects of voice recognition about 10 years ago when I thought of joining a research group at Czech Technical University in Prague. I have a 260 pages text book on my book shelf on voice recognition. Voice signal has high redundancy if compared to a text transcription. But there is additional information stored in the voice signal like pitch, intonation, speed. One could estimate for example mood of the speaker from the utterance. Voice tract could be described by a generator (tone for vowels, hiss for consonants) and filter. Translating voice into generator and filter coefficients greatly decreases voice data redundancy. This is roughly the technique that the common voice codecs do. GSM voice compression is a kind of Algebraic Code Excited Linear Prediction. Another interesting codec is AMBE (Advanced Multi-Band Excitation) used by DSTAR system. GSM half-rate codec squeezes voice to 5.6kbit/sec, AMBE to 3.6 kbps. Both systems use excitation tables, but AMBE is more efficient and closed source. I think the clue to the efficiency is in size and quality of the excitation tables. To create such an algorithm requires considerable amount of research and data analysis. The intelligibility of GSM or AMBE codecs is very good. You could buy the intelectual property of the AMBE codec by buying the chip. There are couple of projects running trying to built DSTAR into legacy transceivers. About 10 years ago we at OK1KPI club experimented with an echolink like system. We modified speakfreely software to control FM transceiver and we added web interface to control tuning and subtone of the transceiver. It was a lot of fun and a very unique system at that time. http://www.speakfreely.org/ The best compression factor offers LPC-10 codec (3460kbps), but the sound is very robot-like and quite hard to understand. At the end we reverted to GSM. I think IVOX is a variant of the LPC system that we tried. Your proposal is to increase compression rate by transmitting phonemes. I once had the same idea, but I quickly rejected it. Although it may be a nice exercise, I find it not very useless until good continuous speech multi-speaker multi-language recognition systems are available. I will try to explain my reasoning behind that statement. Let's classify voice recognition systems by the implementation complexity: 1) Single-speaker, limited set of utterances recognized (control your desktop by voice) 2) Multiple-speaker, limited set of utterances recognized
Re: [digitalradio] Re: digital voice within 100 Hz bandwidth
Robert, I agree. The thing that is different is that speech recognition is not real time. Voice over the radio is real time. Mike n6ief On Nov 18, 2007 10:46 AM, Robert Thompson [EMAIL PROTECTED] wrote: There are several (military/gov) standard intelligibility tests that do a pretty good job of scoring what most humans can and can not reliably understand. You might try taking a look at them to get some ideas of which voice characteristics make the most difference to intelligibility. There is actually a surprising amount of data out there, especially if you include the data peripheral to the various computerized speech translator research projects. It's not *exactly* signal processing... but understanding what parts of the signal matter the most can be surprisingly helpful. This may be unusually productive, because as of yet there hasn't been a huge amount of cross-discipline work between the codec researchers and the speech-to-meaning researchers. While there's a lot of duplicate research in there, it tends to be from slightly different perspectives, and the stereo view can sometimes help. On Nov 18, 2007 9:12 AM, Mike Lebo [EMAIL PROTECTED]mike-lebo%40ieee.org wrote: Hi Vojtech, Thank you for your reply to my papers. I will do more work on the phonemes. The project I want to do uses new computers that were no available 10 years ago. Every 10 mS a decision is made to send a one or a zero. To make that decision I have 68 parallel FFT's running in the background. I believe the brain could handle mispronounce words better than you think. Mike On Nov 17, 2007 3:55 PM, r_lwesterfield [EMAIL PROTECTED]r_lwesterfield%40bellsouth.net wrote: I have a few radios (ARC-210-1851, PSC-5D, PRC-117F) at work that operate in MELP for a vocoder – Mixed Excitation Linear Prediction. We have found MELP to be superior (more human-like voice qualities – less Charlie Brown's teacher) to LPC-10 but we use far larger bandwidths than 100 khz. I do not know how well any of this will play out at such a narrow bandwidth. Listening to Charlie Brown's teacher will send you running away quickly and you should think of your listeners . . . they will tire very quickly. Just because voice can be sent at such narrower bandwidths does not necessarily mean that people will like to listen to it. Rick – KH2DF From: digitalradio@yahoogroups.com digitalradio%40yahoogroups.com[mailto: digitalradio@yahoogroups.com digitalradio%40yahoogroups.com] On Behalf Of Vojtech Bubník Sent: Saturday, November 17, 2007 9:11 AM To: [EMAIL PROTECTED] mike-lebo%40ieee.org; digitalradio@yahoogroups.com digitalradio%40yahoogroups.com Subject: [digitalradio] Re: digital voice within 100 Hz bandwidth Hi Mike. I studied some aspects of voice recognition about 10 years ago when I thought of joining a research group at Czech Technical University in Prague. I have a 260 pages text book on my book shelf on voice recognition. Voice signal has high redundancy if compared to a text transcription. But there is additional information stored in the voice signal like pitch, intonation, speed. One could estimate for example mood of the speaker from the utterance. Voice tract could be described by a generator (tone for vowels, hiss for consonants) and filter. Translating voice into generator and filter coefficients greatly decreases voice data redundancy. This is roughly the technique that the common voice codecs do. GSM voice compression is a kind of Algebraic Code Excited Linear Prediction. Another interesting codec is AMBE (Advanced Multi-Band Excitation) used by DSTAR system. GSM half-rate codec squeezes voice to 5.6kbit/sec, AMBE to 3.6 kbps. Both systems use excitation tables, but AMBE is more efficient and closed source. I think the clue to the efficiency is in size and quality of the excitation tables. To create such an algorithm requires considerable amount of research and data analysis. The intelligibility of GSM or AMBE codecs is very good. You could buy the intelectual property of the AMBE codec by buying the chip. There are couple of projects running trying to built DSTAR into legacy transceivers. About 10 years ago we at OK1KPI club experimented with an echolink like system. We modified speakfreely software to control FM transceiver and we added web interface to control tuning and subtone of the transceiver. It was a lot of fun and a very unique system at that time. http://www.speakfreely.org/ The best compression factor offers LPC-10 codec (3460kbps), but the sound is very robot-like and quite hard to understand. At the end we reverted to GSM. I think IVOX is a variant of the LPC system that we tried. Your proposal is to increase compression rate by
RE: [digitalradio] I Apologize
I suppose if the pictures had to get through because of an emergency, the VFO could have been used as per Hollingsworth's comments @ Dayton. Maybe a secondary frequency should be selected for the net or a VFO procedure since none of us own a frequency no matter how long we may have been using it. Communications can always go on, if we want to! William A. Collister N7MOG _ From: digitalradio@yahoogroups.com [mailto:[EMAIL PROTECTED] On Behalf Of aa0oi Sent: Sunday, November 18, 2007 7:34 AM To: digitalradio@yahoogroups.com Subject: [digitalradio] I Apologize To the great group of Digital Pic Guys that we had on 7.178 on Sat. I apologize for not being able to have a net this Sunday morning. I apologize the the arrogant and rude hams that do contesting and don't listen to a freq before transmitting, and do splits without listening and move within 1 kc with 1500 watts. I apologize for CQ mag. for having such a contest and making any type of communications (other than thier contest) impossible. (and for making it three days long!) I apologize to hams in other countries for trashing ALL the US freqs with CQ Contest (etc) for 3 days. I apologize for the FCC for allowing this deliberate type of interference to go on and continue on ALL SSB freqs. (give them 100kc on each band and let them have at it) I hope to see you all on next Saturday 8am on 7.178 for more pictures and conversation. (and Sunday) Garrett/ AA0OI
Re: [digitalradio] Re: digital voice within 100 Hz bandwidth
That is not entirely true. Besides, I wasn't focusing so much on their real research as the voice characterization research that they had to do before they could usefully work on recognition. It turns out that the very areas that are most necessary for digital voice recognition are the ones most necessary for human brains to recognize and interpret. Voice is a mixed-information-density signal, and if you simplify the signal by filtering out and discarding the less necessary elements, you have significantly reduced the effort the next stage has to do, whether it's digital encoding or speech recognition. On Nov 18, 2007 1:31 PM, Mike Lebo [EMAIL PROTECTED] wrote: Robert, I agree. The thing that is different is that speech recognition is not real time. Voice over the radio is real time. Mike n6ief On Nov 18, 2007 10:46 AM, Robert Thompson [EMAIL PROTECTED] wrote: There are several (military/gov) standard intelligibility tests that do a pretty good job of scoring what most humans can and can not reliably understand. You might try taking a look at them to get some ideas of which voice characteristics make the most difference to intelligibility. There is actually a surprising amount of data out there, especially if you include the data peripheral to the various computerized speech translator research projects. It's not *exactly* signal processing... but understanding what parts of the signal matter the most can be surprisingly helpful. This may be unusually productive, because as of yet there hasn't been a huge amount of cross-discipline work between the codec researchers and the speech-to-meaning researchers. While there's a lot of duplicate research in there, it tends to be from slightly different perspectives, and the stereo view can sometimes help. On Nov 18, 2007 9:12 AM, Mike Lebo [EMAIL PROTECTED] wrote: Hi Vojtech, Thank you for your reply to my papers. I will do more work on the phonemes. The project I want to do uses new computers that were no available 10 years ago. Every 10 mS a decision is made to send a one or a zero. To make that decision I have 68 parallel FFT's running in the background. I believe the brain could handle mispronounce words better than you think. Mike On Nov 17, 2007 3:55 PM, r_lwesterfield [EMAIL PROTECTED] wrote: I have a few radios (ARC-210-1851, PSC-5D, PRC-117F) at work that operate in MELP for a vocoder – Mixed Excitation Linear Prediction. We have found MELP to be superior (more human-like voice qualities – less Charlie Brown's teacher) to LPC-10 but we use far larger bandwidths than 100 khz. I do not know how well any of this will play out at such a narrow bandwidth. Listening to Charlie Brown's teacher will send you running away quickly and you should think of your listeners . . . they will tire very quickly. Just because voice can be sent at such narrower bandwidths does not necessarily mean that people will like to listen to it. Rick – KH2DF From: digitalradio@yahoogroups.com [mailto:[EMAIL PROTECTED] On Behalf Of Vojtech Bubník Sent: Saturday, November 17, 2007 9:11 AM To: [EMAIL PROTECTED]; digitalradio@yahoogroups.com Subject: [digitalradio] Re: digital voice within 100 Hz bandwidth Hi Mike. I studied some aspects of voice recognition about 10 years ago when I thought of joining a research group at Czech Technical University in Prague. I have a 260 pages text book on my book shelf on voice recognition. Voice signal has high redundancy if compared to a text transcription. But there is additional information stored in the voice signal like pitch, intonation, speed. One could estimate for example mood of the speaker from the utterance. Voice tract could be described by a generator (tone for vowels, hiss for consonants) and filter. Translating voice into generator and filter coefficients greatly decreases voice data redundancy. This is roughly the technique that the common voice codecs do. GSM voice compression is a kind of Algebraic Code Excited Linear Prediction. Another interesting codec is AMBE (Advanced Multi-Band Excitation) used by DSTAR system. GSM half-rate codec squeezes voice to 5.6kbit/sec, AMBE to 3.6 kbps. Both systems use excitation tables, but AMBE is more efficient and closed source. I think the clue to the efficiency is in size and quality of the excitation tables. To create such an algorithm requires considerable amount of research and data analysis. The intelligibility of GSM or AMBE codecs is very good. You could buy the intelectual property of the AMBE codec by buying the chip. There are couple of projects running trying to built DSTAR into legacy
Re: [digitalradio] Re: digital voice within 100 Hz bandwidth
Oops, sent too quickly. What I meant was: That ( speech recognition not being real time) is not entirely true. There are many commercial packages that do minimal-lag realtime speech recognition. One example would be the voice command features built into Apple's OSX. Another would be any one of a number of speech-to-text transcription packages. I apologize if my unsupported and abrupt original phrasing appeared to be inflammatory. Such was not intended. On Nov 18, 2007 2:11 PM, Robert Thompson [EMAIL PROTECTED] wrote: That is not entirely true. Besides, I wasn't focusing so much on their real research as the voice characterization research that they had to do before they could usefully work on recognition. It turns out that the very areas that are most necessary for digital voice recognition are the ones most necessary for human brains to recognize and interpret. Voice is a mixed-information-density signal, and if you simplify the signal by filtering out and discarding the less necessary elements, you have significantly reduced the effort the next stage has to do, whether it's digital encoding or speech recognition. On Nov 18, 2007 1:31 PM, Mike Lebo [EMAIL PROTECTED] wrote: Robert, I agree. The thing that is different is that speech recognition is not real time. Voice over the radio is real time. Mike n6ief On Nov 18, 2007 10:46 AM, Robert Thompson [EMAIL PROTECTED] wrote: There are several (military/gov) standard intelligibility tests that do a pretty good job of scoring what most humans can and can not reliably understand. You might try taking a look at them to get some ideas of which voice characteristics make the most difference to intelligibility. There is actually a surprising amount of data out there, especially if you include the data peripheral to the various computerized speech translator research projects. It's not *exactly* signal processing... but understanding what parts of the signal matter the most can be surprisingly helpful. This may be unusually productive, because as of yet there hasn't been a huge amount of cross-discipline work between the codec researchers and the speech-to-meaning researchers. While there's a lot of duplicate research in there, it tends to be from slightly different perspectives, and the stereo view can sometimes help. On Nov 18, 2007 9:12 AM, Mike Lebo [EMAIL PROTECTED] wrote: Hi Vojtech, Thank you for your reply to my papers. I will do more work on the phonemes. The project I want to do uses new computers that were no available 10 years ago. Every 10 mS a decision is made to send a one or a zero. To make that decision I have 68 parallel FFT's running in the background. I believe the brain could handle mispronounce words better than you think. Mike On Nov 17, 2007 3:55 PM, r_lwesterfield [EMAIL PROTECTED] wrote: I have a few radios (ARC-210-1851, PSC-5D, PRC-117F) at work that operate in MELP for a vocoder – Mixed Excitation Linear Prediction. We have found MELP to be superior (more human-like voice qualities – less Charlie Brown's teacher) to LPC-10 but we use far larger bandwidths than 100 khz. I do not know how well any of this will play out at such a narrow bandwidth. Listening to Charlie Brown's teacher will send you running away quickly and you should think of your listeners . . . they will tire very quickly. Just because voice can be sent at such narrower bandwidths does not necessarily mean that people will like to listen to it. Rick – KH2DF From: digitalradio@yahoogroups.com [mailto:[EMAIL PROTECTED] On Behalf Of Vojtech Bubník Sent: Saturday, November 17, 2007 9:11 AM To: [EMAIL PROTECTED]; digitalradio@yahoogroups.com Subject: [digitalradio] Re: digital voice within 100 Hz bandwidth Hi Mike. I studied some aspects of voice recognition about 10 years ago when I thought of joining a research group at Czech Technical University in Prague. I have a 260 pages text book on my book shelf on voice recognition. Voice signal has high redundancy if compared to a text transcription. But there is additional information stored in the voice signal like pitch, intonation, speed. One could estimate for example mood of the speaker from the utterance. Voice tract could be described by a generator (tone for vowels, hiss for consonants) and filter. Translating voice into generator and filter coefficients greatly decreases voice data redundancy. This is roughly the technique that the common voice codecs do. GSM voice compression is a kind of Algebraic Code Excited
[digitalradio] Signalink USB ext. sound
I have a problem, just looking for confirmation. I think I have lost the Audio out of the Data Port on my TS480. Just to confirm. I Plugged earphones into the spkr jack and the mon jack on the back of the USB, No Sound, so I am pretty sure it has a problem. the xmt circuit is working ok. Joe WB6AGR ** See what's new at http://www.aol.com
Re: [digitalradio] Re: RTTY contester's survey
Well, he has a very good point that perhaps you should consider. Using more bandwidth than you need to communicate is worse than using too much power. 73, Charles Brabham, N5PVL - Original Message - From: Roger J. Buffington [EMAIL PROTECTED] To: digitalradio@yahoogroups.com Sent: Saturday, November 17, 2007 5:01 PM Subject: Re: [digitalradio] Re: RTTY contester's survey Brian A wrote: Roger, What about shared resoures don't you understand? I don't particularly care for the tone of your post. Thanks for the lecture. Conversation ended. SK de Roger W6VZV No virus found in this incoming message. Checked by AVG Free Edition. Version: 7.5.503 / Virus Database: 269.16.0/1136 - Release Date: 11/17/2007 2:55 PM
Re: [digitalradio] PSK63 activity!
At 06:10 PM 11/18/2007, you wrote: N3WT United States 14,073.1 PSK63 2007-18-11 18:19K3UK PSK63 36 K6MKF United States 14,073.7 PSK63 2007-18-11 18:26K3UK PSK63 34 K7RE United States 14,072.6 PSK63 2007-18-11 18:27K3UK PSK63 44 K0SZ United States 14,075.3 PSK63 2007-18-11 17:59K3UK PSK63 50 OK for all us none sound card operators, what's the difference between all the stuff in blue. Or dose one need to know the secret hand shake and password first?
Re: [digitalradio] PSK63 activity!
Mode, and signal strength. On Nov 18, 2007 7:30 PM, John Becker, WØJAB [EMAIL PROTECTED] wrote: At 06:10 PM 11/18/2007, you wrote: N3WT United States 14,073.1 PSK63 2007-18-11 18:19K3UK PSK63 36 K6MKF United States 14,073.7 PSK63 2007-18-11 18:26K3UK PSK63 34 K7RE United States 14,072.6 PSK63 2007-18-11 18:27K3UK PSK63 44 K0SZ United States 14,075.3 PSK63 2007-18-11 17:59K3UK PSK63 50 OK for all us none sound card operators, what's the difference between all the stuff in blue. Or dose one need to know the secret hand shake and password first? -- Andy K3UK www.obriensweb.com (QSL via N2RJ)
Re: [digitalradio] PSK63 activity!
Yes, it is very gratifying to see it finally take off a little. Now, if we can only convince the RTTY contest sponsers to specifically include and mention PSK63, or hopefully even give it a multiplier to encourge folks to try it... What I noticed is that the turnover speed rivaled RTTY, with exchanges so fast that simultaneous multichannel decoding and display was almost essential to see who to call. Of course, it is the narrow bandwidth of PSK63 that makes that possible. Maybe a similar American-sponsored PSK63 QSO Party would bring out more stateside stations. Many thanks to Andy for helping launch PSK63! 73, Skip KH6TY - Original Message - From: Andrew O'Brien [EMAIL PROTECTED] To: DIGITALRADIO digitalradio@yahoogroups.com Sent: Sunday, November 18, 2007 7:10 PM Subject: [digitalradio] PSK63 activity! I assume that Skip will be happy. His PSK63 efforts appear to be paying off, the activity in this year's EPSK PSK63 QSO Party was quite high. At one time, I counted 15 simeukatenous QSO's in my 20M waterfall. Again, European activity seemed quite high compared to North American. I saw no Asian or South Pacific stations but did see reports of some ANZAC activity. FYI, here are a few of the stations my antenna captured...(not worked) N3WT United States 14,073.1 PSK63 2007-18-11 18:19K3UK PSK63 36 K6MKF United States 14,073.7 PSK63 2007-18-11 18:26K3UK PSK63 34 K7RE United States 14,072.6 PSK63 2007-18-11 18:27K3UK PSK63 44 K0SZ United States 14,075.3 PSK63 2007-18-11 17:59K3UK PSK63 50 CT3EE Madeira Island 14,074.1 PSK63 2007-18-11 17:28K3UK PSK63 50 N5ARA United States 14,072.4 PSK63 2007-18-11 18:58K3UK PSK63 39 AC5ZS United States 14,073.3 PSK63 2007-18-11 19:06K3UK PSK63 12 KF2GQ United States 14,073.6 PSK63 2007-18-11 19:13K3UK PSK63 46 W6LED United States 14,075.1 PSK63 2007-18-11 18:30K3UK PSK63 24 NC5O/QPR/5WUnited States 14,073.6 PSK63 2007-18-11 19:19K3UK PSK63 36 VA7KOJ Canada 14,075.5 PSK63 2007-18-11 18:16K3UK PSK63 0 J39BS Grenada14,073.6 PSK63 2007-18-11 19:12K3UK PSK63 38 N5PU United States 14,075.1 PSK63 2007-18-11 18:46K3UK PSK63 51 VE9DX Canada 7,038.8 PSK63 2007-18-11 19:26K3UK PSK63 56 SP7IIT Poland 7,037.7 PSK63 2007-18-11 19:27K3UK PSK63 7 KF3AA United States 7,037.5 PSK63 2007-18-11 19:31K3UK PSK63 44 S51MA Slovenia7,037.3 PSK63 2007-18-11 19:47K3UK PSK63 6 CT4DK Portugal7,038.4 PSK63 2007-18-11 19:47K3UK PSK63 38 AO1OS Spain 7,039.2 PSK63 2007-18-11 19:53K3UK PSK63 37 OK1VSL Czech Republic 7,038.8 PSK63 2007-18-11 19:47K3UK PSK63 42 ON8UM Belgium 7,037.7 PSK63 2007-18-11 20:00K3UK PSK63 47 CT3Madeira Island 7,038.8 PSK63 2007-18-11 20:09K3UK PSK63 38 CN8YZ Morocco 7,038.3 PSK63 2007-18-11 19:59K3UK PSK63 43 DK8VQ Germany 7,037.9 PSK63 2007-18-11 20:17K3UK PSK63 30 CT3BD Madeira Island 7,038.8 PSK63 2007-18-11 19:45K3UK PSK63 38 G4KMH England 7,038.5 PSK63 2007-18-11 20:19K3UK PSK63 40 CN8KD Morocco 7,038.1 PSK63 2007-18-11 19:49K3UK PSK63 0 OP3A Belgium 7,039.2 PSK63 2007-18-11 20:19K3UK PSK63 16 WP3UX Puerto Rico 7,036.6 PSK63 2007-18-11 19:58K3UK PSK63 37 RU3QR European Russia 7,038.5 PSK63 2007-18-11 20:25K3UK PSK63 44 N9FTC/4United States 14,074.7 PSK63 2007-18-11 20:26K3UK PSK63 40 W5VGR United States 14,074.4 PSK63 2007-18-11 20:28K3UK PSK63 24 W1MNY United States 14,074.7 PSK63 2007-18-11 20:28K3UK PSK63 37 CQ7EPC
Re: [digitalradio] PSK63 activity!
At 07:10 PM 11/18/2007, you wrote: Yes, it is very gratifying to see it finally take off a little. Now, if we can only convince the RTTY contest sponsers to specifically include and mention PSK63, Skip with all due respect. why ? It's not RTTY. Would this not be like adding CW to a side band contest? Or vice verse. John, W0JAB
Re: [digitalradio] PSK63 activity!
Because it does a better job than rtty (less fills) in less space. If everyone used PSK63 instead of RTTY, there would not be so many complaints by non-contesters about having so little space to use during contests. A PSK63 stations signal, operated linearly, takes up only 1/5 the space of a RTTY signal. Isn't accomplishing the same job in less bandwidth what we should all be trying to do in an ever more crowded world? It is not like adding CW to a phone contest because both RTTY and PSK63 are keyboard modes. Phone and CW are not. 73, Skip KH6TY - Original Message - From: John Becker, WØJAB [EMAIL PROTECTED] To: digitalradio@yahoogroups.com Sent: Sunday, November 18, 2007 8:21 PM Subject: Re: [digitalradio] PSK63 activity! At 07:10 PM 11/18/2007, you wrote: Yes, it is very gratifying to see it finally take off a little. Now, if we can only convince the RTTY contest sponsers to specifically include and mention PSK63, Skip with all due respect. why ? It's not RTTY. Would this not be like adding CW to a side band contest? Or vice verse. John, W0JAB No virus found in this incoming message. Checked by AVG Free Edition. Version: 7.5.503 / Virus Database: 269.16.0/1135 - Release Date: 11/16/2007 10:58 PM
Re: [digitalradio] PSK63 activity!
At 08:00 PM 11/18/2007, you wrote: It is not like adding CW to a phone contest because both RTTY and PSK63 are keyboard modes. Phone and CW are not. Well just add the rest of the keyboard modes while your at it... And please make sure you do add both the keyboard mode of Amtor and Pactor. I still fail to see why psk should be added to a RTTY contest.
Re: [digitalradio] PSK63 activity!
John Becker, WØJAB wrote: Well just add the rest of the keyboard modes while your at it... Great idea! With mode multipliers. And please make sure you do add both the keyboard mode of Amtor and Pactor. Ten extra points for using a time machine, because that is what you'll need to work anyone on these modes. de Roger W6VZV
Re: [digitalradio] PSK63 activity!
Roger regardless of what you think about Amtor and Pactor - both are still doing very well. Other then a hand full of CW and SSB QSO's the log book is full of both Amtor and Pactor 1, 2 and 3. John, W0JAB At 08:16 PM 11/18/2007, you wrote: John Becker, WØJAB wrote: Ten extra points for using a time machine, because that is what you'll need to work anyone on these modes.
RE: [digitalradio] PSK63 activity!
-Original Message- From: digitalradio@yahoogroups.com [mailto:[EMAIL PROTECTED] On Behalf Of John Becker, WØJAB Sent: Sunday, November 18, 2007 9:11 PM To: digitalradio@yahoogroups.com Subject: Re: [digitalradio] PSK63 activity! At 08:00 PM 11/18/2007, you wrote: It is not like adding CW to a phone contest because both RTTY and PSK63 are keyboard modes. Phone and CW are not. Well just add the rest of the keyboard modes while your at it... And please make sure you do add both the keyboard mode of Amtor and Pactor. I still fail to see why psk should be added to a RTTY contest. Possibly for the same reason that they started allowing horseless carriages on the same streets as horses.
RE: [digitalradio] PSK63 activity!
More akin to an AM contest of the 60s including SSB to encourage it. RTTY is an older digital mode. It _should_ be replaced by the newer narrow band mode just as SSB replaced AM, and for the same reasons. Equivalent performance with improved RF usage, mainly bandwidth. Rud Merriam K5RUD ARES AEC Montgomery County, TX http://TheHamNetwork.net -Original Message- From: digitalradio@yahoogroups.com [mailto:[EMAIL PROTECTED] On Behalf Of John Becker, WØJAB Sent: Sunday, November 18, 2007 7:21 PM To: digitalradio@yahoogroups.com Subject: Re: [digitalradio] PSK63 activity! At 07:10 PM 11/18/2007, you wrote: Yes, it is very gratifying to see it finally take off a little. Now, if we can only convince the RTTY contest sponsers to specifically include and mention PSK63, Skip with all due respect. why ? It's not RTTY. Would this not be like adding CW to a side band contest? Or vice verse. John, W0JAB Announce your digital presence via our Interactive Sked Page at http://www.obriensweb.com/drsked/drsked.php Yahoo! Groups Links
[digitalradio] Re: PSK63 activity!
--- In digitalradio@yahoogroups.com, [EMAIL PROTECTED] wrote: -Original Message- From: digitalradio@yahoogroups.com [mailto:[EMAIL PROTECTED] On Behalf Of John Becker, WØJAB Sent: Sunday, November 18, 2007 9:11 PM To: digitalradio@yahoogroups.com Subject: Re: [digitalradio] PSK63 activity! At 08:00 PM 11/18/2007, you wrote: It is not like adding CW to a phone contest because both RTTY and PSK63 are keyboard modes. Phone and CW are not. Well just add the rest of the keyboard modes while your at it... And please make sure you do add both the keyboard mode of Amtor and Pactor. I still fail to see why psk should be added to a RTTY contest. Possibly for the same reason that they started allowing horseless carriages on the same streets as horses. - Yes, of course the older technology was displaced by the horseless carriage. However, when it comes to contesting, the horse tracks continue to support a sizeable following and they don't mix the two technologies during the races.
