|
What is the point of digital voice?
On 2/26/2015 5:04 PM, Jerry Stuckle wrote:
On 2/26/2015 3:28 PM, rickman wrote: On 2/26/2015 10:09 AM, Jerry Stuckle wrote: Yes, the TV only has a certain amount of time to decode the signal. But in the U.S., the method used is proprietary to one company. The chipsets required to decode the signal are all produced by this company, so all TV's have similar decoding. I think you are confusing all chip makers using the same algorithm with all TV makers buying their chips from the same chip maker. http://www.toshiba.com/taec/componen...GProdBrief.pdf http://www.broadcom.com/products/Cab...utions/BCM3560 http://www.fujitsu.com/cn/fsp/home-e...t/MB86H01.html Are you suggesting that all of these chip makers are reselling one company's products? If you would bother to understand what you referenced, NONE of these chipsets are hi-def (1080). And yes, H.264 is a proprietary algorithm, with only one company providing the chipsets. The decoding is very much *not* proprietary to one company. There is a consortium of companies who own patents for the MPEG-2 decoder alone... http://www.mpegla.com/main/programs/...ts/m2-att1.pdf Once again you show you don't understand the technology, but have to argue anyway. MPEG-2 is NOT H.264. "The BCM3560 combines a cable/terrestrial 4/1024 QAM and 8/16-VSB receiver, an out-of-band QPSK receiver, NTSC demodulator, DVI/HDMI receiver, a transport processor, a digital audio processor, a high-definition (HD) MPEG video decoder, 2D graphics processing, digital processing of analog video and audio, analog video digitizer and DAC functions, stereo high-fidelity audio DACs, a 250-MHz MIPS processor, and a peripheral control unit providing a variety of television control functions." I am happy to admit I don't know everything about digital TV. But I do know a ridiculous statement when I see it. "But in the U.S., the method used is proprietary to one company. The chipsets required to decode the signal are all produced by this company, so all TV's have similar decoding." qualifies as a ridiculous statement. No one in the industry would have allowed the FCC to entrench one company as the sole manufacturer of decoder chips for digital TV. BTW, you are right that MPEG-2 is not H.264. It's just not relevant. They are both used for digital TV. -- Rick |
What is the point of digital voice?
On 2/26/2015 8:41 PM, rickman wrote:
On 2/26/2015 5:04 PM, Jerry Stuckle wrote: On 2/26/2015 3:28 PM, rickman wrote: On 2/26/2015 10:09 AM, Jerry Stuckle wrote: Yes, the TV only has a certain amount of time to decode the signal. But in the U.S., the method used is proprietary to one company. The chipsets required to decode the signal are all produced by this company, so all TV's have similar decoding. I think you are confusing all chip makers using the same algorithm with all TV makers buying their chips from the same chip maker. http://www.toshiba.com/taec/componen...GProdBrief.pdf http://www.broadcom.com/products/Cab...utions/BCM3560 http://www.fujitsu.com/cn/fsp/home-e...t/MB86H01.html Are you suggesting that all of these chip makers are reselling one company's products? If you would bother to understand what you referenced, NONE of these chipsets are hi-def (1080). And yes, H.264 is a proprietary algorithm, with only one company providing the chipsets. The decoding is very much *not* proprietary to one company. There is a consortium of companies who own patents for the MPEG-2 decoder alone... http://www.mpegla.com/main/programs/...ts/m2-att1.pdf Once again you show you don't understand the technology, but have to argue anyway. MPEG-2 is NOT H.264. "The BCM3560 combines a cable/terrestrial 4/1024 QAM and 8/16-VSB receiver, an out-of-band QPSK receiver, NTSC demodulator, DVI/HDMI receiver, a transport processor, a digital audio processor, a high-definition (HD) MPEG video decoder, 2D graphics processing, digital processing of analog video and audio, analog video digitizer and DAC functions, stereo high-fidelity audio DACs, a 250-MHz MIPS processor, and a peripheral control unit providing a variety of television control functions." I am happy to admit I don't know everything about digital TV. But I do know a ridiculous statement when I see it. "But in the U.S., the method used is proprietary to one company. The chipsets required to decode the signal are all produced by this company, so all TV's have similar decoding." qualifies as a ridiculous statement. No one in the industry would have allowed the FCC to entrench one company as the sole manufacturer of decoder chips for digital TV. BTW, you are right that MPEG-2 is not H.264. It's just not relevant. They are both used for digital TV. No, you don't know a "ridiculous statement when you see it". You have proven multiple times you don't even know your arse from a hole in the ground. You really should stick with things you know something about. Maybe eventually you can figure out what those things are. -- ================== Remove the "x" from my email address Jerry, AI0K ================== |
What is the point of digital voice?
On 2/26/2015 8:55 PM, Jerry Stuckle wrote:
On 2/26/2015 8:41 PM, rickman wrote: On 2/26/2015 5:04 PM, Jerry Stuckle wrote: On 2/26/2015 3:28 PM, rickman wrote: On 2/26/2015 10:09 AM, Jerry Stuckle wrote: Yes, the TV only has a certain amount of time to decode the signal. But in the U.S., the method used is proprietary to one company. The chipsets required to decode the signal are all produced by this company, so all TV's have similar decoding. I think you are confusing all chip makers using the same algorithm with all TV makers buying their chips from the same chip maker. http://www.toshiba.com/taec/componen...GProdBrief.pdf http://www.broadcom.com/products/Cab...utions/BCM3560 http://www.fujitsu.com/cn/fsp/home-e...t/MB86H01.html Are you suggesting that all of these chip makers are reselling one company's products? If you would bother to understand what you referenced, NONE of these chipsets are hi-def (1080). And yes, H.264 is a proprietary algorithm, with only one company providing the chipsets. The decoding is very much *not* proprietary to one company. There is a consortium of companies who own patents for the MPEG-2 decoder alone... http://www.mpegla.com/main/programs/...ts/m2-att1.pdf Once again you show you don't understand the technology, but have to argue anyway. MPEG-2 is NOT H.264. "The BCM3560 combines a cable/terrestrial 4/1024 QAM and 8/16-VSB receiver, an out-of-band QPSK receiver, NTSC demodulator, DVI/HDMI receiver, a transport processor, a digital audio processor, a high-definition (HD) MPEG video decoder, 2D graphics processing, digital processing of analog video and audio, analog video digitizer and DAC functions, stereo high-fidelity audio DACs, a 250-MHz MIPS processor, and a peripheral control unit providing a variety of television control functions." I am happy to admit I don't know everything about digital TV. But I do know a ridiculous statement when I see it. "But in the U.S., the method used is proprietary to one company. The chipsets required to decode the signal are all produced by this company, so all TV's have similar decoding." qualifies as a ridiculous statement. No one in the industry would have allowed the FCC to entrench one company as the sole manufacturer of decoder chips for digital TV. BTW, you are right that MPEG-2 is not H.264. It's just not relevant. They are both used for digital TV. No, you don't know a "ridiculous statement when you see it". You have proven multiple times you don't even know your arse from a hole in the ground. You really should stick with things you know something about. Maybe eventually you can figure out what those things are. This is why it is so much fun discussing things with you, your professional demeanor, your courteous style and you all around good nature. Thanks for helping me learn. :) -- Rick |
What is the point of digital voice?
On 2/26/2015 9:42 PM, rickman wrote:
On 2/26/2015 8:55 PM, Jerry Stuckle wrote: On 2/26/2015 8:41 PM, rickman wrote: On 2/26/2015 5:04 PM, Jerry Stuckle wrote: On 2/26/2015 3:28 PM, rickman wrote: On 2/26/2015 10:09 AM, Jerry Stuckle wrote: Yes, the TV only has a certain amount of time to decode the signal. But in the U.S., the method used is proprietary to one company. The chipsets required to decode the signal are all produced by this company, so all TV's have similar decoding. I think you are confusing all chip makers using the same algorithm with all TV makers buying their chips from the same chip maker. http://www.toshiba.com/taec/componen...GProdBrief.pdf http://www.broadcom.com/products/Cab...utions/BCM3560 http://www.fujitsu.com/cn/fsp/home-e...t/MB86H01.html Are you suggesting that all of these chip makers are reselling one company's products? If you would bother to understand what you referenced, NONE of these chipsets are hi-def (1080). And yes, H.264 is a proprietary algorithm, with only one company providing the chipsets. The decoding is very much *not* proprietary to one company. There is a consortium of companies who own patents for the MPEG-2 decoder alone... http://www.mpegla.com/main/programs/...ts/m2-att1.pdf Once again you show you don't understand the technology, but have to argue anyway. MPEG-2 is NOT H.264. "The BCM3560 combines a cable/terrestrial 4/1024 QAM and 8/16-VSB receiver, an out-of-band QPSK receiver, NTSC demodulator, DVI/HDMI receiver, a transport processor, a digital audio processor, a high-definition (HD) MPEG video decoder, 2D graphics processing, digital processing of analog video and audio, analog video digitizer and DAC functions, stereo high-fidelity audio DACs, a 250-MHz MIPS processor, and a peripheral control unit providing a variety of television control functions." I am happy to admit I don't know everything about digital TV. But I do know a ridiculous statement when I see it. "But in the U.S., the method used is proprietary to one company. The chipsets required to decode the signal are all produced by this company, so all TV's have similar decoding." qualifies as a ridiculous statement. No one in the industry would have allowed the FCC to entrench one company as the sole manufacturer of decoder chips for digital TV. BTW, you are right that MPEG-2 is not H.264. It's just not relevant. They are both used for digital TV. No, you don't know a "ridiculous statement when you see it". You have proven multiple times you don't even know your arse from a hole in the ground. You really should stick with things you know something about. Maybe eventually you can figure out what those things are. This is why it is so much fun discussing things with you, your professional demeanor, your courteous style and you all around good nature. Thanks for helping me learn. :) No, you repeatedly argue about things you know nothing about. Your claims that mp3 is not a lossy format and white noise exists in this thread are perfect examples. And you never admit you were wrong. Trying to educate you is like trying to teach a pig to sing. And I'm not wasting more of my time on you. And BTW - "pi" is not a compression. It is a representation used by agreement. Someone who does not know the meaning of "pi" cannot discern the number. OTOH, the person need know nothing about a compressed file or signal other than the means required to expand it to recover the contents. -- ================== Remove the "x" from my email address Jerry, AI0K ================== |
What is the point of digital voice?
On 2/27/2015 8:26 AM, Jerry Stuckle wrote:
On 2/26/2015 9:42 PM, rickman wrote: On 2/26/2015 8:55 PM, Jerry Stuckle wrote: On 2/26/2015 8:41 PM, rickman wrote: On 2/26/2015 5:04 PM, Jerry Stuckle wrote: On 2/26/2015 3:28 PM, rickman wrote: On 2/26/2015 10:09 AM, Jerry Stuckle wrote: Yes, the TV only has a certain amount of time to decode the signal. But in the U.S., the method used is proprietary to one company. The chipsets required to decode the signal are all produced by this company, so all TV's have similar decoding. I think you are confusing all chip makers using the same algorithm with all TV makers buying their chips from the same chip maker. http://www.toshiba.com/taec/componen...GProdBrief.pdf http://www.broadcom.com/products/Cab...utions/BCM3560 http://www.fujitsu.com/cn/fsp/home-e...t/MB86H01.html Are you suggesting that all of these chip makers are reselling one company's products? If you would bother to understand what you referenced, NONE of these chipsets are hi-def (1080). And yes, H.264 is a proprietary algorithm, with only one company providing the chipsets. The decoding is very much *not* proprietary to one company. There is a consortium of companies who own patents for the MPEG-2 decoder alone... http://www.mpegla.com/main/programs/...ts/m2-att1.pdf Once again you show you don't understand the technology, but have to argue anyway. MPEG-2 is NOT H.264. "The BCM3560 combines a cable/terrestrial 4/1024 QAM and 8/16-VSB receiver, an out-of-band QPSK receiver, NTSC demodulator, DVI/HDMI receiver, a transport processor, a digital audio processor, a high-definition (HD) MPEG video decoder, 2D graphics processing, digital processing of analog video and audio, analog video digitizer and DAC functions, stereo high-fidelity audio DACs, a 250-MHz MIPS processor, and a peripheral control unit providing a variety of television control functions." I am happy to admit I don't know everything about digital TV. But I do know a ridiculous statement when I see it. "But in the U.S., the method used is proprietary to one company. The chipsets required to decode the signal are all produced by this company, so all TV's have similar decoding." qualifies as a ridiculous statement. No one in the industry would have allowed the FCC to entrench one company as the sole manufacturer of decoder chips for digital TV. BTW, you are right that MPEG-2 is not H.264. It's just not relevant. They are both used for digital TV. No, you don't know a "ridiculous statement when you see it". You have proven multiple times you don't even know your arse from a hole in the ground. You really should stick with things you know something about. Maybe eventually you can figure out what those things are. This is why it is so much fun discussing things with you, your professional demeanor, your courteous style and you all around good nature. Thanks for helping me learn. :) No, you repeatedly argue about things you know nothing about. Your claims that mp3 is not a lossy format and white noise exists in this thread are perfect examples. And you never admit you were wrong. Trying to educate you is like trying to teach a pig to sing. And I'm not wasting more of my time on you. And BTW - "pi" is not a compression. It is a representation used by agreement. Someone who does not know the meaning of "pi" cannot discern the number. OTOH, the person need know nothing about a compressed file or signal other than the means required to expand it to recover the contents. I never said MP3 is not lossy. I can't be wrong about something I didn't say. Actually, pi is the word for a number which has unique properties which define its value. You only need to convey the concept using a finite amount of data and it can produce an infinite string of digits (or bits) that have no repeating pattern and have the properties of randomness. So sure, "pi" is not compression, but the algorithm for producing the digits is. One sure sign that you are having trouble with these concepts is the way you attack me on a personal level. You can say my ideas are wrong, or even silly, but you insist in being rude. I would be only too happy if you didn't respond to any of my posts... but you do. -- Rick |
What is the point of digital voice?
On 2/27/2015 3:35 PM, rickman wrote:
On 2/27/2015 8:26 AM, Jerry Stuckle wrote: On 2/26/2015 9:42 PM, rickman wrote: On 2/26/2015 8:55 PM, Jerry Stuckle wrote: On 2/26/2015 8:41 PM, rickman wrote: On 2/26/2015 5:04 PM, Jerry Stuckle wrote: On 2/26/2015 3:28 PM, rickman wrote: On 2/26/2015 10:09 AM, Jerry Stuckle wrote: Yes, the TV only has a certain amount of time to decode the signal. But in the U.S., the method used is proprietary to one company. The chipsets required to decode the signal are all produced by this company, so all TV's have similar decoding. I think you are confusing all chip makers using the same algorithm with all TV makers buying their chips from the same chip maker. http://www.toshiba.com/taec/componen...GProdBrief.pdf http://www.broadcom.com/products/Cab...utions/BCM3560 http://www.fujitsu.com/cn/fsp/home-e...t/MB86H01.html Are you suggesting that all of these chip makers are reselling one company's products? If you would bother to understand what you referenced, NONE of these chipsets are hi-def (1080). And yes, H.264 is a proprietary algorithm, with only one company providing the chipsets. The decoding is very much *not* proprietary to one company. There is a consortium of companies who own patents for the MPEG-2 decoder alone... http://www.mpegla.com/main/programs/...ts/m2-att1.pdf Once again you show you don't understand the technology, but have to argue anyway. MPEG-2 is NOT H.264. "The BCM3560 combines a cable/terrestrial 4/1024 QAM and 8/16-VSB receiver, an out-of-band QPSK receiver, NTSC demodulator, DVI/HDMI receiver, a transport processor, a digital audio processor, a high-definition (HD) MPEG video decoder, 2D graphics processing, digital processing of analog video and audio, analog video digitizer and DAC functions, stereo high-fidelity audio DACs, a 250-MHz MIPS processor, and a peripheral control unit providing a variety of television control functions." I am happy to admit I don't know everything about digital TV. But I do know a ridiculous statement when I see it. "But in the U.S., the method used is proprietary to one company. The chipsets required to decode the signal are all produced by this company, so all TV's have similar decoding." qualifies as a ridiculous statement. No one in the industry would have allowed the FCC to entrench one company as the sole manufacturer of decoder chips for digital TV. BTW, you are right that MPEG-2 is not H.264. It's just not relevant. They are both used for digital TV. No, you don't know a "ridiculous statement when you see it". You have proven multiple times you don't even know your arse from a hole in the ground. You really should stick with things you know something about. Maybe eventually you can figure out what those things are. This is why it is so much fun discussing things with you, your professional demeanor, your courteous style and you all around good nature. Thanks for helping me learn. :) No, you repeatedly argue about things you know nothing about. Your claims that mp3 is not a lossy format and white noise exists in this thread are perfect examples. And you never admit you were wrong. Trying to educate you is like trying to teach a pig to sing. And I'm not wasting more of my time on you. And BTW - "pi" is not a compression. It is a representation used by agreement. Someone who does not know the meaning of "pi" cannot discern the number. OTOH, the person need know nothing about a compressed file or signal other than the means required to expand it to recover the contents. I never said MP3 is not lossy. I can't be wrong about something I didn't say. Actually, pi is the word for a number which has unique properties which define its value. You only need to convey the concept using a finite amount of data and it can produce an infinite string of digits (or bits) that have no repeating pattern and have the properties of randomness. So sure, "pi" is not compression, but the algorithm for producing the digits is. One sure sign that you are having trouble with these concepts is the way you attack me on a personal level. You can say my ideas are wrong, or even silly, but you insist in being rude. I would be only too happy if you didn't respond to any of my posts... but you do. I'm just correcting you where you're wrong. It's not for your benefit - it's so the rest of the people in the newsgroup don't get the wrong ideas. Whether YOU accept them or not is of no matter to me. But I have to once again correct you on what you said. Me: Some compression algorithms (i.e. mp3) remove what they consider is "unimportant". However, the result after decompressing is a poor recreation of the original signal. You: That is a value judgement which most would disagree with not to mention that your example is not valid. MP3 does not *remove* anything from the signal. It is a form of compression that simply can't reproduce the signal exactly. The use of the term "poor" is your value judgement. Most people would say an MP3 audio sounds very much like the original. The compression removes data from the signal during the compression. That is why the signal cannot be recreated exactly. And the term "poor" is used by all experts in the field. Did you even bother to read the reference where no less than Neil Young and (the late) Steve Jobs talked about how bad it is? But no - you won't admit you're wrong here, either. I'm not having any problems with any of the concepts. But you sure do. And you refuse to admit you're wrong. As for the "personal attacks" - just calling a spade a spade. Nothing more, nothing less. And I really don't care if the truth hurts you or not. -- ================== Remove the "x" from my email address Jerry, AI0K ================== |
What is the point of digital voice?
On 2/27/2015 4:11 PM, Jerry Stuckle wrote:
On 2/27/2015 3:35 PM, rickman wrote: On 2/27/2015 8:26 AM, Jerry Stuckle wrote: On 2/26/2015 9:42 PM, rickman wrote: On 2/26/2015 8:55 PM, Jerry Stuckle wrote: On 2/26/2015 8:41 PM, rickman wrote: On 2/26/2015 5:04 PM, Jerry Stuckle wrote: On 2/26/2015 3:28 PM, rickman wrote: On 2/26/2015 10:09 AM, Jerry Stuckle wrote: Yes, the TV only has a certain amount of time to decode the signal. But in the U.S., the method used is proprietary to one company. The chipsets required to decode the signal are all produced by this company, so all TV's have similar decoding. I think you are confusing all chip makers using the same algorithm with all TV makers buying their chips from the same chip maker. http://www.toshiba.com/taec/componen...GProdBrief.pdf http://www.broadcom.com/products/Cab...utions/BCM3560 http://www.fujitsu.com/cn/fsp/home-e...t/MB86H01.html Are you suggesting that all of these chip makers are reselling one company's products? If you would bother to understand what you referenced, NONE of these chipsets are hi-def (1080). And yes, H.264 is a proprietary algorithm, with only one company providing the chipsets. The decoding is very much *not* proprietary to one company. There is a consortium of companies who own patents for the MPEG-2 decoder alone... http://www.mpegla.com/main/programs/...ts/m2-att1.pdf Once again you show you don't understand the technology, but have to argue anyway. MPEG-2 is NOT H.264. "The BCM3560 combines a cable/terrestrial 4/1024 QAM and 8/16-VSB receiver, an out-of-band QPSK receiver, NTSC demodulator, DVI/HDMI receiver, a transport processor, a digital audio processor, a high-definition (HD) MPEG video decoder, 2D graphics processing, digital processing of analog video and audio, analog video digitizer and DAC functions, stereo high-fidelity audio DACs, a 250-MHz MIPS processor, and a peripheral control unit providing a variety of television control functions." I am happy to admit I don't know everything about digital TV. But I do know a ridiculous statement when I see it. "But in the U.S., the method used is proprietary to one company. The chipsets required to decode the signal are all produced by this company, so all TV's have similar decoding." qualifies as a ridiculous statement. No one in the industry would have allowed the FCC to entrench one company as the sole manufacturer of decoder chips for digital TV. BTW, you are right that MPEG-2 is not H.264. It's just not relevant. They are both used for digital TV. No, you don't know a "ridiculous statement when you see it". You have proven multiple times you don't even know your arse from a hole in the ground. You really should stick with things you know something about. Maybe eventually you can figure out what those things are. This is why it is so much fun discussing things with you, your professional demeanor, your courteous style and you all around good nature. Thanks for helping me learn. :) No, you repeatedly argue about things you know nothing about. Your claims that mp3 is not a lossy format and white noise exists in this thread are perfect examples. And you never admit you were wrong. Trying to educate you is like trying to teach a pig to sing. And I'm not wasting more of my time on you. And BTW - "pi" is not a compression. It is a representation used by agreement. Someone who does not know the meaning of "pi" cannot discern the number. OTOH, the person need know nothing about a compressed file or signal other than the means required to expand it to recover the contents. I never said MP3 is not lossy. I can't be wrong about something I didn't say. Actually, pi is the word for a number which has unique properties which define its value. You only need to convey the concept using a finite amount of data and it can produce an infinite string of digits (or bits) that have no repeating pattern and have the properties of randomness. So sure, "pi" is not compression, but the algorithm for producing the digits is. One sure sign that you are having trouble with these concepts is the way you attack me on a personal level. You can say my ideas are wrong, or even silly, but you insist in being rude. I would be only too happy if you didn't respond to any of my posts... but you do. I'm just correcting you where you're wrong. It's not for your benefit - it's so the rest of the people in the newsgroup don't get the wrong ideas. Whether YOU accept them or not is of no matter to me. But I have to once again correct you on what you said. Me: Some compression algorithms (i.e. mp3) remove what they consider is "unimportant". However, the result after decompressing is a poor recreation of the original signal. You: That is a value judgement which most would disagree with not to mention that your example is not valid. MP3 does not *remove* anything from the signal. It is a form of compression that simply can't reproduce the signal exactly. The use of the term "poor" is your value judgement. Most people would say an MP3 audio sounds very much like the original. The compression removes data from the signal during the compression. That is why the signal cannot be recreated exactly. And the term "poor" is used by all experts in the field. Did you even bother to read the reference where no less than Neil Young and (the late) Steve Jobs talked about how bad it is? But no - you won't admit you're wrong here, either. I'm not having any problems with any of the concepts. But you sure do. And you refuse to admit you're wrong. As for the "personal attacks" - just calling a spade a spade. Nothing more, nothing less. And I really don't care if the truth hurts you or not. Indeed. I find you amusing most of the time, especially your inability to resist the urge to continue this discussion. You clearly hate hearing anything from me. So why continue to post? Ok, which data is "removed" from the signal in MP3 compression? -- Rick |
What is the point of digital voice?
"Brian Reay" wrote in message
... PI is irrational number ie it cannot be expressed as one integer divided by another and give either a terminating or recurring decimal. It's major property is that it is transcendental |
What is the point of digital voice?
On 26/02/2015 16:14, Custos Custodum wrote:
AndyW wrote in news:54eee0a5$0$17091$862e30e2 @ngroups.net: On 25/02/2015 19:08, FranK Turner-Smith G3VKI wrote: But ... if EVERYONE else was wrong that included the author of the booK he was quoting from. Time for a drinK Thanks very much. Kind of you to offer. Mine's a nice bitter, maybe a Harviestoun Bitter and Twisted if they have it. Good call! Deuchars IPA is also very popular "apud Custodum". Good call. I'll email you one over, just pop a pint glass under your USB (Universal Shipping for Beer) port while opening the email. Andy |
What is the point of digital voice?
"AndyW" wrote in message
... On 26/02/2015 16:14, Custos Custodum wrote: AndyW wrote in : On 25/02/2015 19:08, FranK Turner-Smith G3VKI wrote: But ... if EVERYONE else was wrong that included the author of the booK he was quoting from. Time for a drinK Thanks very much. Kind of you to offer. Mine's a nice bitter, maybe a Harviestoun Bitter and Twisted if they have it. Good call! Deuchars IPA is also very popular "apud Custodum". Good call. I'll email you one over, just pop a pint glass under your USB (Universal Shipping for Beer) port while opening the email. So THAT'S where the pool of beer on the floor under my PC is coming from! No wonder my dog is always ****ed! -- ;-) .. 73 de Frank Turner-Smith G3VKI - mine's a pint. .. http://turner-smith.co.uk |
What is the point of digital voice?
On 2/25/2015 5:37 PM, gareth wrote:
Here is your big chance to prove your superiority of knowledge about the super-regrenerative method, but you've gone strangely silent, which is a bit bizarre when you consider how many times you have oft repeated your childish sneer? Perhaps you will listen to the voice of expierence. My first receiver was a Knight Kit Star Roamer.. now this is a superhet, true, but as it turns out it had a REGEN control in one stage, that stage could be made super regenerative,, You used this to receive CW or SSB,, i used that radio for many years. But the fact is.. It worked,, NOT as well as a modern well filtered Superhet,, But that has a lot to do with the Filters more than the receiver's other parts. I would not mind getting another of those.. Nostalga value and all that. -- Home, is where I park it. --- This email has been checked for viruses by Avast antivirus software. http://www.avast.com |
What is the point of digital voice?
On 2/26/2015 3:55 AM, AndyW wrote:
MP3 is lossy, it cannot be used to reproduce the original but it does not 'remove' signal, they get lost. IIRC some sound encoding deliberately removes some frequencies if the are low amplitude and are close to a higher amplitude frequency. Loses is passive, the data just gets lost. Remove implies some active removal of data. All of what you type is true yet MP3 is good enough for most music lovers (The true "Golden Ears" do not like it but not many are that good) I can occasionaly hear the difference but not always. The major advantage of digital over analog modulation is that the computer's "ears" (The de-mod unit) are way more discreaning than my ears. First. Under noisy low signal conditions,,, Most of the noise is lost simply because it is not present at the proper time,, With analog none of it is lost you need to spend heavy duty effort to filter it out.. But with DSP you look for 1 or zero at the right time, noise that happens when you are not looking... is ignored.. And with protocol some errors caused by noise get corrected, others can not be but in some cases a re-peat of the packet is requested and delivered. Far less power is needed to make the trip,, Digital signals can travel farther on less power all because of the above. It truly is an amazing way to chat,, I have used both digital and analog or many years, and where as with analog, as the sigal goes down the amount of operator skill to hear the voice goes up, way up, and more and more folks start wonering what it is I am hearing, cause they sure can not hear it, but I seem to be writing down good inormation. With digital you are there, or you are not, and "There" means it sounds like you are sitting beside me. (Perhaps that is why I operate SSB, I like to keep the skills honed a bit). -- Home, is where I park it. --- This email has been checked for viruses by Avast antivirus software. http://www.avast.com |
What is the point of digital voice?
"John Davis" wrote in message
... On 2/25/2015 5:37 PM, gareth wrote: Here is your big chance to prove your superiority of knowledge about the super-regrenerative method, but you've gone strangely silent, which is a bit bizarre when you consider how many times you have oft repeated your childish sneer? Perhaps you will listen to the voice of expierence. My first receiver was a Knight Kit Star Roamer.. now this is a superhet, true, but as it turns out it had a REGEN control in one stage, that stage could be made super regenerative,, You used this to receive CW or SSB,, i used that radio for many years. I fear that you will be incorrect and confusing regeneration and super-regeneration. |
What is the point of digital voice?
On Fri, 6 Mar 2015, gareth wrote:
"John Davis" wrote in message ... On 2/25/2015 5:37 PM, gareth wrote: Here is your big chance to prove your superiority of knowledge about the super-regrenerative method, but you've gone strangely silent, which is a bit bizarre when you consider how many times you have oft repeated your childish sneer? Perhaps you will listen to the voice of expierence. My first receiver was a Knight Kit Star Roamer.. now this is a superhet, true, but as it turns out it had a REGEN control in one stage, that stage could be made super regenerative,, You used this to receive CW or SSB,, i used that radio for many years. I fear that you will be incorrect and confusing regeneration and super-regeneration. I almost missed it. No, he's talking about a superhet with standard 455Khz IF, where some feedback was added around an IF stage (usually a "gimmick" capacitor so one can adjust it), and with control of the cathode, one could increase selectivity and put it into oscillation so there was something to beat against the incoming signals to demodulate CW and SSB. But that's really just a more complicated method of regeneration and superregeneration. One of the problems with superregenerative receivers is that they were long treated as a black box. ONce they fell out of leading edge circuity (where they helped to homestead the higher bands), people forgot how they worked and the book descriptions were pretty uninformative. I remember one ARRL Handbook going into how the same active device could be the receiver and the quenching oscillator, without explaining what the quenching oscillator did. That said, a superregenerative receiver is just a superset of a regenerative receiver. Armstrong came up with the latter early on, patented in 1914. It showed not only how to make a better receiver, but how to make a tube oscillate, real cutting edge. Then later, when he was on the eve of a court case over that regen patent, he went back to the regen to remind himself about its operation, and came across a phenomena that he'd noticed almost a decade earlier, but hadn't pursued. This was superregeneration, and it happened with a regular regen receiver. It's just kicking things further along. I'm sure some circuits are better to get the quenching, but if you view the superregen as a regen receiver with exteral quenching oscillator, it's all so much easier to visualize. The quenching modulates the regen. If it's one device, the one device does both, it's just a matter of getting the quenching going. So the same receiver can be both. Indeed, in the late fifties or early sixties, the ARRL had a popular VHF station construction series, using a 14MHz regen and converters. And they even say by adjuting regen, you can use the receiver as a superregen. You can't use superregeneration for receiving SSB and CW, but you can use the same circuit, so long as it can be adjusted through regeneration to actual feedback and beyond. Michael |
What is the point of digital voice?
On 3/6/2015 1:06 PM, John Davis wrote:
On 2/26/2015 3:55 AM, AndyW wrote: MP3 is lossy, it cannot be used to reproduce the original but it does not 'remove' signal, they get lost. IIRC some sound encoding deliberately removes some frequencies if the are low amplitude and are close to a higher amplitude frequency. Loses is passive, the data just gets lost. Remove implies some active removal of data. All of what you type is true yet MP3 is good enough for most music lovers (The true "Golden Ears" do not like it but not many are that good) I can occasionaly hear the difference but not always. Both MP3 and CD use 16/44 (16 bits, 44kHz sample rate) formats. The difference is that the CD will have the entire signal stored, while MP3 will remove some of the signal which is not as important as others. If you play an MP3 and a CD on any decent (not even audiophile) equipment, the difference is noticeable, even to a non-audiophile. And the difference between MP3 and high resolution 24/192 is even greater if you're playing music with wide frequency and volume ranges, such as much classical music. But you won't hear that much of a difference between MP3 and 24/192 on a many rock songs :) The major advantage of digital over analog modulation is that the computer's "ears" (The de-mod unit) are way more discreaning than my ears. Computers are lousy playback mechanisms. The frequency response of the amplifier is nowhere near flat, and the speakers generally stink. It would be better if you hooked up a decent set of stereo speakers - but even then a cheap amplifier will outperform virtually any computer. First. Under noisy low signal conditions,,, Most of the noise is lost simply because it is not present at the proper time,, With analog none of it is lost you need to spend heavy duty effort to filter it out.. But with DSP you look for 1 or zero at the right time, noise that happens when you are not looking... is ignored.. And with protocol some errors caused by noise get corrected, others can not be but in some cases a re-peat of the packet is requested and delivered. Noise is like any other part of the signal. If you have a 1kHz noise spike, it will be present for approximately 1ms. That is plenty long for any ADC to detect it. And if the noise pulse is shorter than the sampling time, it would be of too high of a frequency to hear, anyway. Plus, DSPs do not look at amplitude. They measure the instantaneous slope of the signal and store it as a digital value depending on the number of bits, i.e. 16 bit samples would have 2^15 negative slope values and 2^15-1 positive slope values (plus zero slope). By recreating the instantaneous slope that is stored digitally, the DAC converts the digital signal back to an analog signal. And digital error-correction protocols have nothing to do with the digital signal itself - only the transmission of it from one system to another. But that is an entirely different subject. Far less power is needed to make the trip,, Digital signals can travel farther on less power all because of the above. It truly is an amazing way to chat,, I have used both digital and analog or many years, and where as with analog, as the sigal goes down the amount of operator skill to hear the voice goes up, way up, and more and more folks start wonering what it is I am hearing, cause they sure can not hear it, but I seem to be writing down good inormation. Yes, I understand that. I was working RTTY back in the 60's, and it was amazing how you could get good copy on a signal you couldn't even hear in the noise. Of course, the narrow filters used on the audio signal made a big difference - just like a narrow filter helps pull a CW signal out of the mud. With digital you are there, or you are not, and "There" means it sounds like you are sitting beside me. (Perhaps that is why I operate SSB, I like to keep the skills honed a bit). Yes and no. Digital does for the most part work or not work. However, when you get into marginal conditions, it can get iffy, with some packets lost and not recoverable. Probably the easiest way to see this is watching a digital TV signal. When the signal becomes marginal, the picture will start to display junk in random small spots on the screen, similar to snow (known as pixelation). Satellite TV users have seen it during heavy rain, and even cable TV users can see it when a network's satellite link suffers from a marginal signal. -- ================== Remove the "x" from my email address Jerry, AI0K ================== |
What is the point of digital voice?
On 3/6/2015 3:11 PM, Jerry Stuckle wrote:
Plus, DSPs do not look at amplitude. They measure the instantaneous slope of the signal and store it as a digital value depending on the number of bits, i.e. 16 bit samples would have 2^15 negative slope values and 2^15-1 positive slope values (plus zero slope). By recreating the instantaneous slope that is stored digitally, the DAC converts the digital signal back to an analog signal. This is just plain wrong. I'm not sure why you make a distinction between DSP's [sic] and any other digital device since a DSP is not needed at all to digitize or compress a signal, but the sample produced by an ADC *is* the instantaneous value of the signal and not the slope. If you were to compare adjacent ADC samples and calculate the slope that would be a form of ADPCM. The DAC in turn converts this instantaneous value back into analog followed by filtering to remove the higher frequency images if important. -- Rick |
What is the point of digital voice?
"Michael Black" wrote in message
news:alpine.LNX.2.02.1503061451360.32579@darkstar. example.org... On Fri, 6 Mar 2015, gareth wrote: I fear that you will be incorrect and confusing regeneration and super-regeneration. I almost missed it. No, he's talking about a superhet with standard 455Khz IF, where some feedback was added around an IF stage (usually a "gimmick" capacitor so one can adjust it), and with control of the cathode, one could increase selectivity and put it into oscillation so there was something to beat against the incoming signals to demodulate CW and SSB. But that's really just a more complicated method of regeneration and superregeneration. He is discussing a regenerative IF detector, but not a superregenerative one where the feedback is increased well past the point of oscillation to give very high gain. There would not have been a quenching oscillator in what he described. The quencher acts like a balanced modulator onto the oscillatory stage to remove the presence of the on-channel carrier out to two sidebands distanced away by the quench frequency, which is why the super-regenerative technique does not resolve SSB and CW. |
What is the point of digital voice?
On 3/6/2015 3:48 PM, rickman wrote:
On 3/6/2015 3:11 PM, Jerry Stuckle wrote: Plus, DSPs do not look at amplitude. They measure the instantaneous slope of the signal and store it as a digital value depending on the number of bits, i.e. 16 bit samples would have 2^15 negative slope values and 2^15-1 positive slope values (plus zero slope). By recreating the instantaneous slope that is stored digitally, the DAC converts the digital signal back to an analog signal. This is just plain wrong. I'm not sure why you make a distinction between DSP's [sic] and any other digital device since a DSP is not needed at all to digitize or compress a signal, but the sample produced by an ADC *is* the instantaneous value of the signal and not the slope. If you were to compare adjacent ADC samples and calculate the slope that would be a form of ADPCM. The DAC in turn converts this instantaneous value back into analog followed by filtering to remove the higher frequency images if important. Once again you are wrong, Rick. Integrating ADCs have been used at least since the 70's and are much more accurate and noise immune than a simple level ADC. ADPCM isn't even closely related. And I mentioned DSPs because that is what John was discussing. -- ================== Remove the "x" from my email address Jerry, AI0K ================== |
What is the point of digital voice?
On 3/6/2015 5:13 PM, Jerry Stuckle wrote:
On 3/6/2015 3:48 PM, rickman wrote: On 3/6/2015 3:11 PM, Jerry Stuckle wrote: Plus, DSPs do not look at amplitude. They measure the instantaneous slope of the signal and store it as a digital value depending on the number of bits, i.e. 16 bit samples would have 2^15 negative slope values and 2^15-1 positive slope values (plus zero slope). By recreating the instantaneous slope that is stored digitally, the DAC converts the digital signal back to an analog signal. This is just plain wrong. I'm not sure why you make a distinction between DSP's [sic] and any other digital device since a DSP is not needed at all to digitize or compress a signal, but the sample produced by an ADC *is* the instantaneous value of the signal and not the slope. If you were to compare adjacent ADC samples and calculate the slope that would be a form of ADPCM. The DAC in turn converts this instantaneous value back into analog followed by filtering to remove the higher frequency images if important. Once again you are wrong, Rick. Integrating ADCs have been used at least since the 70's and are much more accurate and noise immune than a simple level ADC. ADPCM isn't even closely related. I'm only going to point out your error and then I won't argue with you further. No one is talking about integrating ADCs. You said, "They measure the instantaneous slope of the signal and store it as a digital value". That is not what an integrating ADC does, nor does any other ADC. The integrating ADC uses the input to charge up a capacitance (the integrator) for some period of time, then a reference is used to discharge the "integrated" voltage and the time this takes is measured. This is *not* measuring the "instantaneous slope" of the input signal. In fact "integrating" and "instantaneous" are contradictory since "integrating" takes time and "instantaneous" is... well, instantaneous. Also I will mention that although integrating ADCs are good for noise rejection, they are *very* slow and only used in such low sample rate apps as volt meters and the like. More accurate systems like weight scales typically use sigma-delta converters for low noise, low power and high resolution or in the case of and high end audio sigma-delta converters offer high linearity and low distortion. I think one reason integrating converters are used in volt meters is that they can be designed to always display 0 for a 0 input voltage which is important to consumer confidence. ADPCM is a form of compression comparing adjacent ADC samples to calculate the differential of the signal which is the closest thing to what you are describing by the "instantaneous slope". -- Rick |
What is the point of digital voice?
On 3/7/2015 2:55 AM, rickman wrote:
On 3/6/2015 5:13 PM, Jerry Stuckle wrote: On 3/6/2015 3:48 PM, rickman wrote: On 3/6/2015 3:11 PM, Jerry Stuckle wrote: Plus, DSPs do not look at amplitude. They measure the instantaneous slope of the signal and store it as a digital value depending on the number of bits, i.e. 16 bit samples would have 2^15 negative slope values and 2^15-1 positive slope values (plus zero slope). By recreating the instantaneous slope that is stored digitally, the DAC converts the digital signal back to an analog signal. This is just plain wrong. I'm not sure why you make a distinction between DSP's [sic] and any other digital device since a DSP is not needed at all to digitize or compress a signal, but the sample produced by an ADC *is* the instantaneous value of the signal and not the slope. If you were to compare adjacent ADC samples and calculate the slope that would be a form of ADPCM. The DAC in turn converts this instantaneous value back into analog followed by filtering to remove the higher frequency images if important. Once again you are wrong, Rick. Integrating ADCs have been used at least since the 70's and are much more accurate and noise immune than a simple level ADC. ADPCM isn't even closely related. I'm only going to point out your error and then I won't argue with you further. No one is talking about integrating ADCs. You said, "They measure the instantaneous slope of the signal and store it as a digital value". That is not what an integrating ADC does, nor does any other ADC. The integrating ADC uses the input to charge up a capacitance (the integrator) for some period of time, then a reference is used to discharge the "integrated" voltage and the time this takes is measured. This is *not* measuring the "instantaneous slope" of the input signal. In fact "integrating" and "instantaneous" are contradictory since "integrating" takes time and "instantaneous" is... well, instantaneous. Also I will mention that although integrating ADCs are good for noise rejection, they are *very* slow and only used in such low sample rate apps as volt meters and the like. More accurate systems like weight scales typically use sigma-delta converters for low noise, low power and high resolution or in the case of and high end audio sigma-delta converters offer high linearity and low distortion. I think one reason integrating converters are used in volt meters is that they can be designed to always display 0 for a 0 input voltage which is important to consumer confidence. ADPCM is a form of compression comparing adjacent ADC samples to calculate the differential of the signal which is the closest thing to what you are describing by the "instantaneous slope". Sorry - I used the wrong term. The integration is done by the DAC, to invert the actions of the ADC. But no, if you understood ANY calculus, you would understand that "integrating" and "instantaneous" are not contradictory. But then "instantaneous" is only a theoretical concept, not possible in the real world. But the word is still in common usage. I wonder why that is? ADPCM (Adaptive Differential Pulse Code Modulation) is something completely different. Slope ADCs are used because they can more accurately recreate the waveform. To make it simple - let's see the ADC is sampling at twice the frequency being sampled, i.e. 10kHz signal and 20kHz sampling rate. If the sample happens to be at the zero crossing point, your ADC will show zero volts - IOW, no signal. But a slope detecting ADC will show a fairly high positive slope on one sample and an equally negative slope on the next sample. By integrating these, the DAC can closely recreate the signal because it can estimate the maximum amplitude by the slopes of the samples. No, it won't be perfect - but it will be a lot closer than your simple ADC. Now I know you're going to find all kinds of problems with this example - but I made the example simple so that even you might be able to understand it. As you increase the sample rate relative to the frequency of the signal being sampled, the difference becomes less. But the slope detecting ADC will always provide a more accurate signal (until you get to an infinitely small sample anyway). The math is somewhat complex, and I'm sure beyond anything you could possibly understand. But it can be proven. As for them not existing. I guess the whole quarter we spent on ADCs in my EE classes were wrong then. Of course, this was over 40 years ago. But I doubt physics has changed in that time. -- ================== Remove the "x" from my email address Jerry, AI0K ================== |
What is the point of digital voice?
On 3/7/2015 11:35 AM, Jerry Stuckle wrote:
Slope ADCs are used because they can more accurately recreate the waveform. To make it simple - let's see the ADC is sampling at twice the frequency being sampled, i.e. 10kHz signal and 20kHz sampling rate. If the sample happens to be at the zero crossing point, your ADC will show zero volts - IOW, no signal. But a slope detecting ADC will show a fairly high positive slope on one sample and an equally negative slope on the next sample. By integrating these, the DAC can closely recreate the signal because it can estimate the maximum amplitude by the slopes of the samples. No, it won't be perfect - but it will be a lot closer than your simple ADC. I don't enjoy discussing things with you because you have to make everything personal. But I will explain the fallacy of your argument on the Nyquist sampling rate concept. You pick a sampling point for the dual slope, integrating converter that happens to give valid results. But if you shift the phase by 90° so that this converter sees positive values half the integrating period and negative values for the other half, it produces all zero samples as well. So there is really no difference in the two converters regarding Nyquist rate sampling. It merely depends on the phasing of the sample clock to the input signal. It also depends on how you define the "sample point" of an integrating converter, the start, the end or the middle of the integration period. I will finally point out that your use of the term "slope detecting ADC" is invalid. Google returns exactly 4 hits when this term is entered with quotes. The name of this converter may have slope in it, but that is because the circuit generates a slope, not because it is detecting a slope. Please look up the circuit and use a proper name for it such as integrating ADC or dual slope ADC. The integrating converter is not at all sensitive to the slope of the input signal, otherwise it would not be able to measure a DC signal which has a slope of zero. I'm only replying so that others are not confused by your misstatements. -- Rick |
What is the point of digital voice?
On 3/7/2015 1:33 PM, rickman wrote:
On 3/7/2015 11:35 AM, Jerry Stuckle wrote: Slope ADCs are used because they can more accurately recreate the waveform. To make it simple - let's see the ADC is sampling at twice the frequency being sampled, i.e. 10kHz signal and 20kHz sampling rate. If the sample happens to be at the zero crossing point, your ADC will show zero volts - IOW, no signal. But a slope detecting ADC will show a fairly high positive slope on one sample and an equally negative slope on the next sample. By integrating these, the DAC can closely recreate the signal because it can estimate the maximum amplitude by the slopes of the samples. No, it won't be perfect - but it will be a lot closer than your simple ADC. I don't enjoy discussing things with you because you have to make everything personal. But I will explain the fallacy of your argument on the Nyquist sampling rate concept. You pick a sampling point for the dual slope, integrating converter that happens to give valid results. But if you shift the phase by 90° so that this converter sees positive values half the integrating period and negative values for the other half, it produces all zero samples as well. So there is really no difference in the two converters regarding Nyquist rate sampling. It merely depends on the phasing of the sample clock to the input signal. It also depends on how you define the "sample point" of an integrating converter, the start, the end or the middle of the integration period. I will finally point out that your use of the term "slope detecting ADC" is invalid. Google returns exactly 4 hits when this term is entered with quotes. The name of this converter may have slope in it, but that is because the circuit generates a slope, not because it is detecting a slope. Please look up the circuit and use a proper name for it such as integrating ADC or dual slope ADC. The integrating converter is not at all sensitive to the slope of the input signal, otherwise it would not be able to measure a DC signal which has a slope of zero. I'm only replying so that others are not confused by your misstatements. As I said - I was using this as an example that even your simple mind might understand. And I knew you would find some fault with it. But that's why I tried to make it simple. In real life you use at least three times the frequency; at that rate you would have sample 120 degrees apart - which always provides more accuracy than your simple detector. And you think Google is a valid reference? Try EE texts. Of course, you'll have to learn a few things to understand them. But I know you'll just dismiss my updates because you refuse to learn. You can have the last word. I'm not trying to teach the pig to sing any more. -- ================== Remove the "x" from my email address Jerry, AI0K ================== |
What is the point of digital voice?
On 3/7/2015 4:44 PM, Jerry Stuckle wrote:
On 3/7/2015 1:33 PM, rickman wrote: On 3/7/2015 11:35 AM, Jerry Stuckle wrote: Slope ADCs are used because they can more accurately recreate the waveform. To make it simple - let's see the ADC is sampling at twice the frequency being sampled, i.e. 10kHz signal and 20kHz sampling rate. If the sample happens to be at the zero crossing point, your ADC will show zero volts - IOW, no signal. But a slope detecting ADC will show a fairly high positive slope on one sample and an equally negative slope on the next sample. By integrating these, the DAC can closely recreate the signal because it can estimate the maximum amplitude by the slopes of the samples. No, it won't be perfect - but it will be a lot closer than your simple ADC. I don't enjoy discussing things with you because you have to make everything personal. But I will explain the fallacy of your argument on the Nyquist sampling rate concept. You pick a sampling point for the dual slope, integrating converter that happens to give valid results. But if you shift the phase by 90° so that this converter sees positive values half the integrating period and negative values for the other half, it produces all zero samples as well. So there is really no difference in the two converters regarding Nyquist rate sampling. It merely depends on the phasing of the sample clock to the input signal. It also depends on how you define the "sample point" of an integrating converter, the start, the end or the middle of the integration period. I will finally point out that your use of the term "slope detecting ADC" is invalid. Google returns exactly 4 hits when this term is entered with quotes. The name of this converter may have slope in it, but that is because the circuit generates a slope, not because it is detecting a slope. Please look up the circuit and use a proper name for it such as integrating ADC or dual slope ADC. The integrating converter is not at all sensitive to the slope of the input signal, otherwise it would not be able to measure a DC signal which has a slope of zero. I'm only replying so that others are not confused by your misstatements. As I said - I was using this as an example that even your simple mind might understand. And I knew you would find some fault with it. But that's why I tried to make it simple. In real life you use at least three times the frequency; at that rate you would have sample 120 degrees apart - which always provides more accuracy than your simple detector. And you think Google is a valid reference? Try EE texts. Of course, you'll have to learn a few things to understand them. But I know you'll just dismiss my updates because you refuse to learn. You can have the last word. The last word on what exactly? You have made several statements that were wrong. When you try to justify your misstatements you make more misstatements. There is nothing wrong with your example. Your conclusion is wrong. I'm glad that we can put this to bed. -- Rick |
What is the point of digital voice?
I will finally point out that your use of the term "slope detecting ADC" is invalid. Google returns exactly 4 hits when this term is entered with quotes. The name of this converter may have slope in it, but that is because the circuit generates a slope, not because it is detecting a slope. Please look up the circuit and use a proper name for it such as integrating ADC or dual slope ADC. The integrating converter is not at all sensitive to the slope of the input signal, otherwise it would not be able to measure a DC signal which has a slope of zero. I'm only replying so that others are not confused by your misstatements. He is probably referring to a CVSD, otherwise known as a Delta Modulator. Jeff |
What is the point of digital voice?
On 3/8/2015 7:35 AM, Brian Reay wrote:
Jeff wrote: I will finally point out that your use of the term "slope detecting ADC" is invalid. Google returns exactly 4 hits when this term is entered with quotes. The name of this converter may have slope in it, but that is because the circuit generates a slope, not because it is detecting a slope. Please look up the circuit and use a proper name for it such as integrating ADC or dual slope ADC. The integrating converter is not at all sensitive to the slope of the input signal, otherwise it would not be able to measure a DC signal which has a slope of zero. I'm only replying so that others are not confused by your misstatements. He is probably referring to a CVSD, otherwise known as a Delta Modulator. Jeff I don't think so. In fact, I have to say Jerry seems a bit confused in this particular area, perhaps I have missed something. ADC tend to have a sample and hold prior to the actual ADC convertor, thus the value converted is that at the beginning of the sample period OR if another approach to conversion is used, you get some kind of average over the conversion period. (There are other techniques but those are the main ones.) If you think about, a S/H is required if the rate of change of the input signal means it can change by 1/2 lsb during the conversion time for a SAR ADC. This limits the overall BW of the ADC process. (I recall spending some time convincing a 'seat of the pants engineer' of this when his design wouldn't work. Even when he adopted the suggested changes he insisted his design would have worked if the ADC was more accurate. In fact, it would have made it worse.) No, Brian, I am not confused. It is a form of delta modulation, but is used in an ADC. Two samples are taken, 2 or more times the sample rate (i.e. if the sample rate were 20us, the first sample would be taken every 20us, with the second sample following by 10us or less). The difference is converted to a digital value for transmission. On the other end, the reverse happens. Yes, the signal can change by 1/2 lsb - but that's true of any ADC. For any sufficiently high sample rate (i.e. 3x input signal or more), this method is never less accurate than a simple voltage detecting ADC, and in almost every case is more accurate. However, it is a more complex circuit (on both ends), samples a much smaller analog value and requires more exacting components and a higher cost (which is typically the case for any circuit improvements). As I said - we studied them in one of my EE coursed back in the 70's. I played with them for a while back then, but at the time the ICs were pretty expensive for a college student. -- ================== Remove the "x" from my email address Jerry, AI0K ================== |
What is the point of digital voice?
No, Brian, I am not confused. you tell the big know all .........tee hee |
What is the point of digital voice?
Jerry Stuckle wrote:
On 3/8/2015 7:35 AM, Brian Reay wrote: Jeff wrote: I will finally point out that your use of the term "slope detecting ADC" is invalid. Google returns exactly 4 hits when this term is entered with quotes. The name of this converter may have slope in it, but that is because the circuit generates a slope, not because it is detecting a slope. Please look up the circuit and use a proper name for it such as integrating ADC or dual slope ADC. The integrating converter is not at all sensitive to the slope of the input signal, otherwise it would not be able to measure a DC signal which has a slope of zero. I'm only replying so that others are not confused by your misstatements. He is probably referring to a CVSD, otherwise known as a Delta Modulator. Jeff I don't think so. In fact, I have to say Jerry seems a bit confused in this particular area, perhaps I have missed something. ADC tend to have a sample and hold prior to the actual ADC convertor, thus the value converted is that at the beginning of the sample period OR if another approach to conversion is used, you get some kind of average over the conversion period. (There are other techniques but those are the main ones.) If you think about, a S/H is required if the rate of change of the input signal means it can change by 1/2 lsb during the conversion time for a SAR ADC. This limits the overall BW of the ADC process. (I recall spending some time convincing a 'seat of the pants engineer' of this when his design wouldn't work. Even when he adopted the suggested changes he insisted his design would have worked if the ADC was more accurate. In fact, it would have made it worse.) No, Brian, I am not confused. It is a form of delta modulation, but is used in an ADC. Two samples are taken, 2 or more times the sample rate (i.e. if the sample rate were 20us, the first sample would be taken every 20us, with the second sample following by 10us or less). The difference is converted to a digital value for transmission. On the other end, the reverse happens. Yes, the signal can change by 1/2 lsb - but that's true of any ADC. For any sufficiently high sample rate (i.e. 3x input signal or more), this method is never less accurate than a simple voltage detecting ADC, and in almost every case is more accurate. However, it is a more complex circuit (on both ends), samples a much smaller analog value and requires more exacting components and a higher cost (which is typically the case for any circuit improvements). As I said - we studied them in one of my EE coursed back in the 70's. I played with them for a while back then, but at the time the ICs were pretty expensive for a college student. Ok Jerry. You can, of course, find the rate of change (slope) by that method if you know ( or assume) the signal is either only increasing or decreasing between the samples. (A Nyquist matter). However, the 1/2 lsb matter I mentioned is more for during the conversion, rather that for different samples. It is particularly important for slower ADC types, such as SAR implementations. It may well be that we are talking at crossed purposes. I'm not making an issue of it. |
What is the point of digital voice?
On 3/8/2015 9:03 AM, Jerry Stuckle wrote:
On 3/8/2015 7:35 AM, Brian Reay wrote: Jeff wrote: I will finally point out that your use of the term "slope detecting ADC" is invalid. Google returns exactly 4 hits when this term is entered with quotes. The name of this converter may have slope in it, but that is because the circuit generates a slope, not because it is detecting a slope. Please look up the circuit and use a proper name for it such as integrating ADC or dual slope ADC. The integrating converter is not at all sensitive to the slope of the input signal, otherwise it would not be able to measure a DC signal which has a slope of zero. I'm only replying so that others are not confused by your misstatements. He is probably referring to a CVSD, otherwise known as a Delta Modulator. Jeff I don't think so. In fact, I have to say Jerry seems a bit confused in this particular area, perhaps I have missed something. ADC tend to have a sample and hold prior to the actual ADC convertor, thus the value converted is that at the beginning of the sample period OR if another approach to conversion is used, you get some kind of average over the conversion period. (There are other techniques but those are the main ones.) If you think about, a S/H is required if the rate of change of the input signal means it can change by 1/2 lsb during the conversion time for a SAR ADC. This limits the overall BW of the ADC process. (I recall spending some time convincing a 'seat of the pants engineer' of this when his design wouldn't work. Even when he adopted the suggested changes he insisted his design would have worked if the ADC was more accurate. In fact, it would have made it worse.) No, Brian, I am not confused. It is a form of delta modulation, but is used in an ADC. Two samples are taken, 2 or more times the sample rate (i.e. if the sample rate were 20us, the first sample would be taken every 20us, with the second sample following by 10us or less). The difference is converted to a digital value for transmission. On the other end, the reverse happens. That is not what you have been describing. Now you are saying that the ADC samples the amplitude of the signal just as I have been saying, but now you are adding a step in which the delta is calculated which is what I was describing with ADPCM (although I should have used the simpler and more like your approach DPCM). I have never heard of using it in the way you are describing though. Even in DPCM the samples are taken at a fixed interval and the delta is calculated on *every* pair of adjacent samples, not just every other. So a sample stream of x0, x1, x2, x3, etc would produce delta values of d0, d1, d2,... not just d0, d1... You describe two samples being taken for each data sample transmitted, ignoring the change in signal between x1 and x2. The signal could not be reconstructed with this data missing. Yes, the signal can change by 1/2 lsb - but that's true of any ADC. The sample and hold issue is a red herring and in fact, is counter productive in a dual slope converter whose point is to average (integrate) the signal over a period of time filtering higher frequency content. For any sufficiently high sample rate (i.e. 3x input signal or more), this method is never less accurate than a simple voltage detecting ADC, and in almost every case is more accurate. However, it is a more complex circuit (on both ends), samples a much smaller analog value and requires more exacting components and a higher cost (which is typically the case for any circuit improvements). The sampling method you describe is *not* different from a voltage detecting ADC and therefore can't be better. All you are doing that is different is the analog circuitry is obtaining the slope of the signal over a short interval and is losing the slope of the signal between the samples being ignored. Can you explain how it could be *more* accurate? I suspect you are confusing the efficiency of the data rate with accuracy. DPCM does provide some compression of the data rate when the signal is over sampled as you seem to be describing. But it does nothing to make the samples more accurate. As I said - we studied them in one of my EE coursed back in the 70's. I played with them for a while back then, but at the time the ICs were pretty expensive for a college student. Does this technique have a name? Any references? -- Rick |
What is the point of digital voice?
On 3/8/2015 9:53 AM, Brian Reay wrote:
Jerry Stuckle wrote: On 3/8/2015 7:35 AM, Brian Reay wrote: Jeff wrote: I will finally point out that your use of the term "slope detecting ADC" is invalid. Google returns exactly 4 hits when this term is entered with quotes. The name of this converter may have slope in it, but that is because the circuit generates a slope, not because it is detecting a slope. Please look up the circuit and use a proper name for it such as integrating ADC or dual slope ADC. The integrating converter is not at all sensitive to the slope of the input signal, otherwise it would not be able to measure a DC signal which has a slope of zero. I'm only replying so that others are not confused by your misstatements. He is probably referring to a CVSD, otherwise known as a Delta Modulator. Jeff I don't think so. In fact, I have to say Jerry seems a bit confused in this particular area, perhaps I have missed something. ADC tend to have a sample and hold prior to the actual ADC convertor, thus the value converted is that at the beginning of the sample period OR if another approach to conversion is used, you get some kind of average over the conversion period. (There are other techniques but those are the main ones.) If you think about, a S/H is required if the rate of change of the input signal means it can change by 1/2 lsb during the conversion time for a SAR ADC. This limits the overall BW of the ADC process. (I recall spending some time convincing a 'seat of the pants engineer' of this when his design wouldn't work. Even when he adopted the suggested changes he insisted his design would have worked if the ADC was more accurate. In fact, it would have made it worse.) No, Brian, I am not confused. It is a form of delta modulation, but is used in an ADC. Two samples are taken, 2 or more times the sample rate (i.e. if the sample rate were 20us, the first sample would be taken every 20us, with the second sample following by 10us or less). The difference is converted to a digital value for transmission. On the other end, the reverse happens. Yes, the signal can change by 1/2 lsb - but that's true of any ADC. For any sufficiently high sample rate (i.e. 3x input signal or more), this method is never less accurate than a simple voltage detecting ADC, and in almost every case is more accurate. However, it is a more complex circuit (on both ends), samples a much smaller analog value and requires more exacting components and a higher cost (which is typically the case for any circuit improvements). As I said - we studied them in one of my EE coursed back in the 70's. I played with them for a while back then, but at the time the ICs were pretty expensive for a college student. Ok Jerry. You can, of course, find the rate of change (slope) by that method if you know ( or assume) the signal is either only increasing or decreasing between the samples. (A Nyquist matter). However, the 1/2 lsb matter I mentioned is more for during the conversion, rather that for different samples. It is particularly important for slower ADC types, such as SAR implementations. Can you explain your 1/2 lsb effect? What type of ADC are you referring to? Different ADC types do require a S/H on the input for signals that are not *highly* oversampled. For example a flash converter can mess up and be quite a bit off if the signal is slewing during conversion. Same with SAR converters. But I don't know of any effect where 1/2 lsb is a threshold. Some converters will be negatively affected by a S/H on the input. An integrating converter can reduce the affect of higher frequency noise by averaging the signal over a period of time reducing the requirement for input filtering. Adding a S/H circuit will eliminate that benefit. -- Rick |
What is the point of digital voice?
On 08/03/15 18:46, rickman wrote:
On 3/8/2015 9:53 AM, Brian Reay wrote: Jerry Stuckle wrote: On 3/8/2015 7:35 AM, Brian Reay wrote: Jeff wrote: I will finally point out that your use of the term "slope detecting ADC" is invalid. Google returns exactly 4 hits when this term is entered with quotes. The name of this converter may have slope in it, but that is because the circuit generates a slope, not because it is detecting a slope. Please look up the circuit and use a proper name for it such as integrating ADC or dual slope ADC. The integrating converter is not at all sensitive to the slope of the input signal, otherwise it would not be able to measure a DC signal which has a slope of zero. I'm only replying so that others are not confused by your misstatements. He is probably referring to a CVSD, otherwise known as a Delta Modulator. Jeff I don't think so. In fact, I have to say Jerry seems a bit confused in this particular area, perhaps I have missed something. ADC tend to have a sample and hold prior to the actual ADC convertor, thus the value converted is that at the beginning of the sample period OR if another approach to conversion is used, you get some kind of average over the conversion period. (There are other techniques but those are the main ones.) If you think about, a S/H is required if the rate of change of the input signal means it can change by 1/2 lsb during the conversion time for a SAR ADC. This limits the overall BW of the ADC process. (I recall spending some time convincing a 'seat of the pants engineer' of this when his design wouldn't work. Even when he adopted the suggested changes he insisted his design would have worked if the ADC was more accurate. In fact, it would have made it worse.) No, Brian, I am not confused. It is a form of delta modulation, but is used in an ADC. Two samples are taken, 2 or more times the sample rate (i.e. if the sample rate were 20us, the first sample would be taken every 20us, with the second sample following by 10us or less). The difference is converted to a digital value for transmission. On the other end, the reverse happens. Yes, the signal can change by 1/2 lsb - but that's true of any ADC. For any sufficiently high sample rate (i.e. 3x input signal or more), this method is never less accurate than a simple voltage detecting ADC, and in almost every case is more accurate. However, it is a more complex circuit (on both ends), samples a much smaller analog value and requires more exacting components and a higher cost (which is typically the case for any circuit improvements). As I said - we studied them in one of my EE coursed back in the 70's. I played with them for a while back then, but at the time the ICs were pretty expensive for a college student. Ok Jerry. You can, of course, find the rate of change (slope) by that method if you know ( or assume) the signal is either only increasing or decreasing between the samples. (A Nyquist matter). However, the 1/2 lsb matter I mentioned is more for during the conversion, rather that for different samples. It is particularly important for slower ADC types, such as SAR implementations. Can you explain your 1/2 lsb effect? What type of ADC are you referring to? Different ADC types do require a S/H on the input for signals that are not *highly* oversampled. For example a flash converter can mess up and be quite a bit off if the signal is slewing during conversion. Same with SAR converters. But I don't know of any effect where 1/2 lsb is a threshold. What threshold would you expect? As I recall, 1/2 lsb is the limit to ensure that the conversion would be the 'same' over the conversion time. |
What is the point of digital voice?
On 3/8/2015 3:31 PM, Brian Reay wrote:
On 08/03/15 18:46, rickman wrote: On 3/8/2015 9:53 AM, Brian Reay wrote: Jerry Stuckle wrote: On 3/8/2015 7:35 AM, Brian Reay wrote: Jeff wrote: I will finally point out that your use of the term "slope detecting ADC" is invalid. Google returns exactly 4 hits when this term is entered with quotes. The name of this converter may have slope in it, but that is because the circuit generates a slope, not because it is detecting a slope. Please look up the circuit and use a proper name for it such as integrating ADC or dual slope ADC. The integrating converter is not at all sensitive to the slope of the input signal, otherwise it would not be able to measure a DC signal which has a slope of zero. I'm only replying so that others are not confused by your misstatements. He is probably referring to a CVSD, otherwise known as a Delta Modulator. Jeff I don't think so. In fact, I have to say Jerry seems a bit confused in this particular area, perhaps I have missed something. ADC tend to have a sample and hold prior to the actual ADC convertor, thus the value converted is that at the beginning of the sample period OR if another approach to conversion is used, you get some kind of average over the conversion period. (There are other techniques but those are the main ones.) If you think about, a S/H is required if the rate of change of the input signal means it can change by 1/2 lsb during the conversion time for a SAR ADC. This limits the overall BW of the ADC process. (I recall spending some time convincing a 'seat of the pants engineer' of this when his design wouldn't work. Even when he adopted the suggested changes he insisted his design would have worked if the ADC was more accurate. In fact, it would have made it worse.) No, Brian, I am not confused. It is a form of delta modulation, but is used in an ADC. Two samples are taken, 2 or more times the sample rate (i.e. if the sample rate were 20us, the first sample would be taken every 20us, with the second sample following by 10us or less). The difference is converted to a digital value for transmission. On the other end, the reverse happens. Yes, the signal can change by 1/2 lsb - but that's true of any ADC. For any sufficiently high sample rate (i.e. 3x input signal or more), this method is never less accurate than a simple voltage detecting ADC, and in almost every case is more accurate. However, it is a more complex circuit (on both ends), samples a much smaller analog value and requires more exacting components and a higher cost (which is typically the case for any circuit improvements). As I said - we studied them in one of my EE coursed back in the 70's. I played with them for a while back then, but at the time the ICs were pretty expensive for a college student. Ok Jerry. You can, of course, find the rate of change (slope) by that method if you know ( or assume) the signal is either only increasing or decreasing between the samples. (A Nyquist matter). However, the 1/2 lsb matter I mentioned is more for during the conversion, rather that for different samples. It is particularly important for slower ADC types, such as SAR implementations. Can you explain your 1/2 lsb effect? What type of ADC are you referring to? Different ADC types do require a S/H on the input for signals that are not *highly* oversampled. For example a flash converter can mess up and be quite a bit off if the signal is slewing during conversion. Same with SAR converters. But I don't know of any effect where 1/2 lsb is a threshold. What threshold would you expect? As I recall, 1/2 lsb is the limit to ensure that the conversion would be the 'same' over the conversion time. I'm not sure what you mean by "the conversion would be the 'same' over the conversion time", but I don't see how 1/2 lsb is any magic threshold. If you are working with a flash converter, there are a number of comparators each with a different threshold. The input signal could be right at the edge of one of these thresholds so that a very tiny change in the input signal will cause that threshold to be crossed during the conversion. Maybe I'm not understanding your point. -- Rick |
What is the point of digital voice?
On 3/8/2015 9:53 AM, Brian Reay wrote:
Jerry Stuckle wrote: On 3/8/2015 7:35 AM, Brian Reay wrote: Jeff wrote: I will finally point out that your use of the term "slope detecting ADC" is invalid. Google returns exactly 4 hits when this term is entered with quotes. The name of this converter may have slope in it, but that is because the circuit generates a slope, not because it is detecting a slope. Please look up the circuit and use a proper name for it such as integrating ADC or dual slope ADC. The integrating converter is not at all sensitive to the slope of the input signal, otherwise it would not be able to measure a DC signal which has a slope of zero. I'm only replying so that others are not confused by your misstatements. He is probably referring to a CVSD, otherwise known as a Delta Modulator. Jeff I don't think so. In fact, I have to say Jerry seems a bit confused in this particular area, perhaps I have missed something. ADC tend to have a sample and hold prior to the actual ADC convertor, thus the value converted is that at the beginning of the sample period OR if another approach to conversion is used, you get some kind of average over the conversion period. (There are other techniques but those are the main ones.) If you think about, a S/H is required if the rate of change of the input signal means it can change by 1/2 lsb during the conversion time for a SAR ADC. This limits the overall BW of the ADC process. (I recall spending some time convincing a 'seat of the pants engineer' of this when his design wouldn't work. Even when he adopted the suggested changes he insisted his design would have worked if the ADC was more accurate. In fact, it would have made it worse.) No, Brian, I am not confused. It is a form of delta modulation, but is used in an ADC. Two samples are taken, 2 or more times the sample rate (i.e. if the sample rate were 20us, the first sample would be taken every 20us, with the second sample following by 10us or less). The difference is converted to a digital value for transmission. On the other end, the reverse happens. Yes, the signal can change by 1/2 lsb - but that's true of any ADC. For any sufficiently high sample rate (i.e. 3x input signal or more), this method is never less accurate than a simple voltage detecting ADC, and in almost every case is more accurate. However, it is a more complex circuit (on both ends), samples a much smaller analog value and requires more exacting components and a higher cost (which is typically the case for any circuit improvements). As I said - we studied them in one of my EE coursed back in the 70's. I played with them for a while back then, but at the time the ICs were pretty expensive for a college student. Ok Jerry. You can, of course, find the rate of change (slope) by that method if you know ( or assume) the signal is either only increasing or decreasing between the samples. (A Nyquist matter). Even if the slope is neither increasing nor decreasing, it still has a slope. That slope happens to be zero. And with a sufficiently small time between samples, you will be very close, even if the amplitude is not just increasing or decreasing. But then sampling just the voltage assumes the voltage increases or decreases linearly between samples. Again, the shorter the time between samples (successive samples in this case), the closer that will be to the actual signal. However, the 1/2 lsb matter I mentioned is more for during the conversion, rather that for different samples. It is particularly important for slower ADC types, such as SAR implementations. It's a problem with any ADC converter, and one to which there is no answer. To perfectly recreate an analog signal you would have to have an infinite number of bits (actually, some current physics theories suggest everything can be broken into discreet pieces - even time, but that's beyond this discussion). Anything short of an infinite number of bits would always suffer from 1/2 lsb error. It may well be that we are talking at crossed purposes. I'm not making an issue of it. Not really; you are correct with the 1/2 lsb, as I indicated. -- ================== Remove the "x" from my email address Jerry, AI0K ================== |
What is the point of digital voice?
On 3/8/2015 2:39 PM, rickman wrote:
On 3/8/2015 9:03 AM, Jerry Stuckle wrote: On 3/8/2015 7:35 AM, Brian Reay wrote: Jeff wrote: I will finally point out that your use of the term "slope detecting ADC" is invalid. Google returns exactly 4 hits when this term is entered with quotes. The name of this converter may have slope in it, but that is because the circuit generates a slope, not because it is detecting a slope. Please look up the circuit and use a proper name for it such as integrating ADC or dual slope ADC. The integrating converter is not at all sensitive to the slope of the input signal, otherwise it would not be able to measure a DC signal which has a slope of zero. I'm only replying so that others are not confused by your misstatements. He is probably referring to a CVSD, otherwise known as a Delta Modulator. Jeff I don't think so. In fact, I have to say Jerry seems a bit confused in this particular area, perhaps I have missed something. ADC tend to have a sample and hold prior to the actual ADC convertor, thus the value converted is that at the beginning of the sample period OR if another approach to conversion is used, you get some kind of average over the conversion period. (There are other techniques but those are the main ones.) If you think about, a S/H is required if the rate of change of the input signal means it can change by 1/2 lsb during the conversion time for a SAR ADC. This limits the overall BW of the ADC process. (I recall spending some time convincing a 'seat of the pants engineer' of this when his design wouldn't work. Even when he adopted the suggested changes he insisted his design would have worked if the ADC was more accurate. In fact, it would have made it worse.) No, Brian, I am not confused. It is a form of delta modulation, but is used in an ADC. Two samples are taken, 2 or more times the sample rate (i.e. if the sample rate were 20us, the first sample would be taken every 20us, with the second sample following by 10us or less). The difference is converted to a digital value for transmission. On the other end, the reverse happens. That is not what you have been describing. Now you are saying that the ADC samples the amplitude of the signal just as I have been saying, but now you are adding a step in which the delta is calculated which is what I was describing with ADPCM (although I should have used the simpler and more like your approach DPCM). It is EXACTLY what I've been describing, but you're too stoopid to understand it. But as usual, rather than trying to learn, you argue and prove your stoopidity. I have never heard of using it in the way you are describing though. Even in DPCM the samples are taken at a fixed interval and the delta is calculated on *every* pair of adjacent samples, not just every other. So a sample stream of x0, x1, x2, x3, etc would produce delta values of d0, d1, d2,... not just d0, d1... That's OK. Those types of ADC's haven't heard of you, either, so I guess you don't exist. You describe two samples being taken for each data sample transmitted, ignoring the change in signal between x1 and x2. The signal could not be reconstructed with this data missing. Once again you are proving you have no idea. Yes, the signal can change by 1/2 lsb - but that's true of any ADC. The sample and hold issue is a red herring and in fact, is counter productive in a dual slope converter whose point is to average (integrate) the signal over a period of time filtering higher frequency content. Which has nothing to do with what I'm discussing. But you have to argue, anyway. For any sufficiently high sample rate (i.e. 3x input signal or more), this method is never less accurate than a simple voltage detecting ADC, and in almost every case is more accurate. However, it is a more complex circuit (on both ends), samples a much smaller analog value and requires more exacting components and a higher cost (which is typically the case for any circuit improvements). The sampling method you describe is *not* different from a voltage detecting ADC and therefore can't be better. All you are doing that is different is the analog circuitry is obtaining the slope of the signal over a short interval and is losing the slope of the signal between the samples being ignored. Can you explain how it could be *more* accurate? Once again you show you have no idea what I'm talking about, yet you have to prove your stoopidity by arguing, anyway. I suspect you are confusing the efficiency of the data rate with accuracy. DPCM does provide some compression of the data rate when the signal is over sampled as you seem to be describing. But it does nothing to make the samples more accurate. Once again you show you have no idea what I'm talking about, yet you have to prove your stoopidity by arguing, anyway. As I said - we studied them in one of my EE coursed back in the 70's. I played with them for a while back then, but at the time the ICs were pretty expensive for a college student. Does this technique have a name? Any references? Go to school, get an EE degree, then maybe we can talk about it intelligently. I'm not wasting my time trying to teach the pig to sing. Maybe - IF you were ever more interested in learning than arguing, I would be more interested in discussing it with you. But you have repeatedly proven that is not the case, so I'm not. -- ================== Remove the "x" from my email address Jerry, AI0K ================== |
What is the point of digital voice?
On 3/8/2015 4:37 PM, Jerry Stuckle wrote:
On 3/8/2015 2:39 PM, rickman wrote: On 3/8/2015 9:03 AM, Jerry Stuckle wrote: On 3/8/2015 7:35 AM, Brian Reay wrote: Jeff wrote: I will finally point out that your use of the term "slope detecting ADC" is invalid. Google returns exactly 4 hits when this term is entered with quotes. The name of this converter may have slope in it, but that is because the circuit generates a slope, not because it is detecting a slope. Please look up the circuit and use a proper name for it such as integrating ADC or dual slope ADC. The integrating converter is not at all sensitive to the slope of the input signal, otherwise it would not be able to measure a DC signal which has a slope of zero. I'm only replying so that others are not confused by your misstatements. He is probably referring to a CVSD, otherwise known as a Delta Modulator. Jeff I don't think so. In fact, I have to say Jerry seems a bit confused in this particular area, perhaps I have missed something. ADC tend to have a sample and hold prior to the actual ADC convertor, thus the value converted is that at the beginning of the sample period OR if another approach to conversion is used, you get some kind of average over the conversion period. (There are other techniques but those are the main ones.) If you think about, a S/H is required if the rate of change of the input signal means it can change by 1/2 lsb during the conversion time for a SAR ADC. This limits the overall BW of the ADC process. (I recall spending some time convincing a 'seat of the pants engineer' of this when his design wouldn't work. Even when he adopted the suggested changes he insisted his design would have worked if the ADC was more accurate. In fact, it would have made it worse.) No, Brian, I am not confused. It is a form of delta modulation, but is used in an ADC. Two samples are taken, 2 or more times the sample rate (i.e. if the sample rate were 20us, the first sample would be taken every 20us, with the second sample following by 10us or less). The difference is converted to a digital value for transmission. On the other end, the reverse happens. That is not what you have been describing. Now you are saying that the ADC samples the amplitude of the signal just as I have been saying, but now you are adding a step in which the delta is calculated which is what I was describing with ADPCM (although I should have used the simpler and more like your approach DPCM). It is EXACTLY what I've been describing, but you're too stoopid to understand it. But as usual, rather than trying to learn, you argue and prove your stoopidity. I have never heard of using it in the way you are describing though. Even in DPCM the samples are taken at a fixed interval and the delta is calculated on *every* pair of adjacent samples, not just every other. So a sample stream of x0, x1, x2, x3, etc would produce delta values of d0, d1, d2,... not just d0, d1... That's OK. Those types of ADC's haven't heard of you, either, so I guess you don't exist. You describe two samples being taken for each data sample transmitted, ignoring the change in signal between x1 and x2. The signal could not be reconstructed with this data missing. Once again you are proving you have no idea. Yes, the signal can change by 1/2 lsb - but that's true of any ADC. The sample and hold issue is a red herring and in fact, is counter productive in a dual slope converter whose point is to average (integrate) the signal over a period of time filtering higher frequency content. Which has nothing to do with what I'm discussing. But you have to argue, anyway. For any sufficiently high sample rate (i.e. 3x input signal or more), this method is never less accurate than a simple voltage detecting ADC, and in almost every case is more accurate. However, it is a more complex circuit (on both ends), samples a much smaller analog value and requires more exacting components and a higher cost (which is typically the case for any circuit improvements). The sampling method you describe is *not* different from a voltage detecting ADC and therefore can't be better. All you are doing that is different is the analog circuitry is obtaining the slope of the signal over a short interval and is losing the slope of the signal between the samples being ignored. Can you explain how it could be *more* accurate? Once again you show you have no idea what I'm talking about, yet you have to prove your stoopidity by arguing, anyway. I suspect you are confusing the efficiency of the data rate with accuracy. DPCM does provide some compression of the data rate when the signal is over sampled as you seem to be describing. But it does nothing to make the samples more accurate. Once again you show you have no idea what I'm talking about, yet you have to prove your stoopidity by arguing, anyway. As I said - we studied them in one of my EE coursed back in the 70's. I played with them for a while back then, but at the time the ICs were pretty expensive for a college student. Does this technique have a name? Any references? Go to school, get an EE degree, then maybe we can talk about it intelligently. I'm not wasting my time trying to teach the pig to sing. Maybe - IF you were ever more interested in learning than arguing, I would be more interested in discussing it with you. But you have repeatedly proven that is not the case, so I'm not. Ok Jerry. I'm not going to argue with you. I asked you for the name of this ADC technique and you can't come up with one. In this post *every* single one of your replies is ad hominem rather than discussing the issue. Clearly you have no basis for what you are saying. So there is no point in trying to get you to explain any further. -- Rick |
What is the point of digital voice?
In rec.radio.amateur.equipment rickman wrote:
snip Ok Jerry. I'm not going to argue with you. I asked you for the name of this ADC technique and you can't come up with one. In this post *every* single one of your replies is ad hominem rather than discussing the issue. Clearly you have no basis for what you are saying. So there is no point in trying to get you to explain any further. As will always happen if you dare to contradict the all-knowing and mighty Stuckle. -- Jim Pennino |
What is the point of digital voice?
On 3/8/2015 5:20 PM, rickman wrote:
On 3/8/2015 4:37 PM, Jerry Stuckle wrote: On 3/8/2015 2:39 PM, rickman wrote: On 3/8/2015 9:03 AM, Jerry Stuckle wrote: On 3/8/2015 7:35 AM, Brian Reay wrote: Jeff wrote: I will finally point out that your use of the term "slope detecting ADC" is invalid. Google returns exactly 4 hits when this term is entered with quotes. The name of this converter may have slope in it, but that is because the circuit generates a slope, not because it is detecting a slope. Please look up the circuit and use a proper name for it such as integrating ADC or dual slope ADC. The integrating converter is not at all sensitive to the slope of the input signal, otherwise it would not be able to measure a DC signal which has a slope of zero. I'm only replying so that others are not confused by your misstatements. He is probably referring to a CVSD, otherwise known as a Delta Modulator. Jeff I don't think so. In fact, I have to say Jerry seems a bit confused in this particular area, perhaps I have missed something. ADC tend to have a sample and hold prior to the actual ADC convertor, thus the value converted is that at the beginning of the sample period OR if another approach to conversion is used, you get some kind of average over the conversion period. (There are other techniques but those are the main ones.) If you think about, a S/H is required if the rate of change of the input signal means it can change by 1/2 lsb during the conversion time for a SAR ADC. This limits the overall BW of the ADC process. (I recall spending some time convincing a 'seat of the pants engineer' of this when his design wouldn't work. Even when he adopted the suggested changes he insisted his design would have worked if the ADC was more accurate. In fact, it would have made it worse.) No, Brian, I am not confused. It is a form of delta modulation, but is used in an ADC. Two samples are taken, 2 or more times the sample rate (i.e. if the sample rate were 20us, the first sample would be taken every 20us, with the second sample following by 10us or less). The difference is converted to a digital value for transmission. On the other end, the reverse happens. That is not what you have been describing. Now you are saying that the ADC samples the amplitude of the signal just as I have been saying, but now you are adding a step in which the delta is calculated which is what I was describing with ADPCM (although I should have used the simpler and more like your approach DPCM). It is EXACTLY what I've been describing, but you're too stoopid to understand it. But as usual, rather than trying to learn, you argue and prove your stoopidity. I have never heard of using it in the way you are describing though. Even in DPCM the samples are taken at a fixed interval and the delta is calculated on *every* pair of adjacent samples, not just every other. So a sample stream of x0, x1, x2, x3, etc would produce delta values of d0, d1, d2,... not just d0, d1... That's OK. Those types of ADC's haven't heard of you, either, so I guess you don't exist. You describe two samples being taken for each data sample transmitted, ignoring the change in signal between x1 and x2. The signal could not be reconstructed with this data missing. Once again you are proving you have no idea. Yes, the signal can change by 1/2 lsb - but that's true of any ADC. The sample and hold issue is a red herring and in fact, is counter productive in a dual slope converter whose point is to average (integrate) the signal over a period of time filtering higher frequency content. Which has nothing to do with what I'm discussing. But you have to argue, anyway. For any sufficiently high sample rate (i.e. 3x input signal or more), this method is never less accurate than a simple voltage detecting ADC, and in almost every case is more accurate. However, it is a more complex circuit (on both ends), samples a much smaller analog value and requires more exacting components and a higher cost (which is typically the case for any circuit improvements). The sampling method you describe is *not* different from a voltage detecting ADC and therefore can't be better. All you are doing that is different is the analog circuitry is obtaining the slope of the signal over a short interval and is losing the slope of the signal between the samples being ignored. Can you explain how it could be *more* accurate? Once again you show you have no idea what I'm talking about, yet you have to prove your stoopidity by arguing, anyway. I suspect you are confusing the efficiency of the data rate with accuracy. DPCM does provide some compression of the data rate when the signal is over sampled as you seem to be describing. But it does nothing to make the samples more accurate. Once again you show you have no idea what I'm talking about, yet you have to prove your stoopidity by arguing, anyway. As I said - we studied them in one of my EE coursed back in the 70's. I played with them for a while back then, but at the time the ICs were pretty expensive for a college student. Does this technique have a name? Any references? Go to school, get an EE degree, then maybe we can talk about it intelligently. I'm not wasting my time trying to teach the pig to sing. Maybe - IF you were ever more interested in learning than arguing, I would be more interested in discussing it with you. But you have repeatedly proven that is not the case, so I'm not. Ok Jerry. I'm not going to argue with you. I asked you for the name of this ADC technique and you can't come up with one. In this post *every* single one of your replies is ad hominem rather than discussing the issue. Clearly you have no basis for what you are saying. So there is no point in trying to get you to explain any further. You're right - I'm not answering your questions, because you have proven yourself to be incapable of understanding even the simplest explanation. The fact I WON'T answer you questions only means I refuse to try to keep teaching the pig to sing - not that I don't know what I'm talking about. If you want to discuss this, get yourself an EE degree. Then just maybe we can discuss technical topics intelligently. Until then, you can continue to suck your pacifier. -- ================== Remove the "x" from my email address Jerry Stuckle ================== |
What is the point of digital voice?
On 08/03/15 19:58, rickman wrote:
On 3/8/2015 3:31 PM, Brian Reay wrote: On 08/03/15 18:46, rickman wrote: On 3/8/2015 9:53 AM, Brian Reay wrote: Jerry Stuckle wrote: On 3/8/2015 7:35 AM, Brian Reay wrote: Jeff wrote: I will finally point out that your use of the term "slope detecting ADC" is invalid. Google returns exactly 4 hits when this term is entered with quotes. The name of this converter may have slope in it, but that is because the circuit generates a slope, not because it is detecting a slope. Please look up the circuit and use a proper name for it such as integrating ADC or dual slope ADC. The integrating converter is not at all sensitive to the slope of the input signal, otherwise it would not be able to measure a DC signal which has a slope of zero. I'm only replying so that others are not confused by your misstatements. He is probably referring to a CVSD, otherwise known as a Delta Modulator. Jeff I don't think so. In fact, I have to say Jerry seems a bit confused in this particular area, perhaps I have missed something. ADC tend to have a sample and hold prior to the actual ADC convertor, thus the value converted is that at the beginning of the sample period OR if another approach to conversion is used, you get some kind of average over the conversion period. (There are other techniques but those are the main ones.) If you think about, a S/H is required if the rate of change of the input signal means it can change by 1/2 lsb during the conversion time for a SAR ADC. This limits the overall BW of the ADC process. (I recall spending some time convincing a 'seat of the pants engineer' of this when his design wouldn't work. Even when he adopted the suggested changes he insisted his design would have worked if the ADC was more accurate. In fact, it would have made it worse.) No, Brian, I am not confused. It is a form of delta modulation, but is used in an ADC. Two samples are taken, 2 or more times the sample rate (i.e. if the sample rate were 20us, the first sample would be taken every 20us, with the second sample following by 10us or less). The difference is converted to a digital value for transmission. On the other end, the reverse happens. Yes, the signal can change by 1/2 lsb - but that's true of any ADC. For any sufficiently high sample rate (i.e. 3x input signal or more), this method is never less accurate than a simple voltage detecting ADC, and in almost every case is more accurate. However, it is a more complex circuit (on both ends), samples a much smaller analog value and requires more exacting components and a higher cost (which is typically the case for any circuit improvements). As I said - we studied them in one of my EE coursed back in the 70's. I played with them for a while back then, but at the time the ICs were pretty expensive for a college student. Ok Jerry. You can, of course, find the rate of change (slope) by that method if you know ( or assume) the signal is either only increasing or decreasing between the samples. (A Nyquist matter). However, the 1/2 lsb matter I mentioned is more for during the conversion, rather that for different samples. It is particularly important for slower ADC types, such as SAR implementations. Can you explain your 1/2 lsb effect? What type of ADC are you referring to? Different ADC types do require a S/H on the input for signals that are not *highly* oversampled. For example a flash converter can mess up and be quite a bit off if the signal is slewing during conversion. Same with SAR converters. But I don't know of any effect where 1/2 lsb is a threshold. What threshold would you expect? As I recall, 1/2 lsb is the limit to ensure that the conversion would be the 'same' over the conversion time. I'm not sure what you mean by "the conversion would be the 'same' over the conversion time", but I don't see how 1/2 lsb is any magic threshold. If you are working with a flash converter, there are a number of comparators each with a different threshold. The input signal could be right at the edge of one of these thresholds so that a very tiny change in the input signal will cause that threshold to be crossed during the conversion. Maybe I'm not understanding your point. Sorry, I was referring to SAR converters. I should have been more precise. With an SAR converter, if the signal changes during the conversion period, then the converter will fail (at best)*, if the change is more than 1/2 lsb. Therefore, the signal must remain constant (within 1/2 lsb) for the period of the conversion. If the maximum rate of change of signal is known to be such that this will be the case, all is well, if not, you need a sample and hold. You sample the signal, convert the sample, and repeat the process for the next sample. The S/H is designed to minimise the sampling time while ensuring the required hold time is maintained- ie the sample stays within the required 1/2 lsb for the conversion period. Of course, some SAR ADCs have the S/H incorporated within the device, others require either an external one or have provision for the C to be external to permit design flexibility. *by fail, rather depends on the converter. You will at least get an false reading. I recall using one ADC which set a bit indicating a failure to 'find' a 'match'. I recall the details of the parameters of the S/H design being in the application notes of the various ADCs I used over the years, I expect if you look at some you can see for yourself. By their nature (and application) flash converters don't require an S/H but lack the resolution of SAR ADCs. They have other limitations of course. If memory serves, one being that they are not monotonic which was a requirement in the application I tended to apply ADCs (control circuits, feedback loops don't like non-monotonic converters). |
What is the point of digital voice?
On 3/8/2015 6:06 PM, Brian Reay wrote:
On 08/03/15 19:58, rickman wrote: On 3/8/2015 3:31 PM, Brian Reay wrote: On 08/03/15 18:46, rickman wrote: On 3/8/2015 9:53 AM, Brian Reay wrote: Jerry Stuckle wrote: On 3/8/2015 7:35 AM, Brian Reay wrote: Jeff wrote: I will finally point out that your use of the term "slope detecting ADC" is invalid. Google returns exactly 4 hits when this term is entered with quotes. The name of this converter may have slope in it, but that is because the circuit generates a slope, not because it is detecting a slope. Please look up the circuit and use a proper name for it such as integrating ADC or dual slope ADC. The integrating converter is not at all sensitive to the slope of the input signal, otherwise it would not be able to measure a DC signal which has a slope of zero. I'm only replying so that others are not confused by your misstatements. He is probably referring to a CVSD, otherwise known as a Delta Modulator. Jeff I don't think so. In fact, I have to say Jerry seems a bit confused in this particular area, perhaps I have missed something. ADC tend to have a sample and hold prior to the actual ADC convertor, thus the value converted is that at the beginning of the sample period OR if another approach to conversion is used, you get some kind of average over the conversion period. (There are other techniques but those are the main ones.) If you think about, a S/H is required if the rate of change of the input signal means it can change by 1/2 lsb during the conversion time for a SAR ADC. This limits the overall BW of the ADC process. (I recall spending some time convincing a 'seat of the pants engineer' of this when his design wouldn't work. Even when he adopted the suggested changes he insisted his design would have worked if the ADC was more accurate. In fact, it would have made it worse.) No, Brian, I am not confused. It is a form of delta modulation, but is used in an ADC. Two samples are taken, 2 or more times the sample rate (i.e. if the sample rate were 20us, the first sample would be taken every 20us, with the second sample following by 10us or less). The difference is converted to a digital value for transmission. On the other end, the reverse happens. Yes, the signal can change by 1/2 lsb - but that's true of any ADC. For any sufficiently high sample rate (i.e. 3x input signal or more), this method is never less accurate than a simple voltage detecting ADC, and in almost every case is more accurate. However, it is a more complex circuit (on both ends), samples a much smaller analog value and requires more exacting components and a higher cost (which is typically the case for any circuit improvements). As I said - we studied them in one of my EE coursed back in the 70's. I played with them for a while back then, but at the time the ICs were pretty expensive for a college student. Ok Jerry. You can, of course, find the rate of change (slope) by that method if you know ( or assume) the signal is either only increasing or decreasing between the samples. (A Nyquist matter). However, the 1/2 lsb matter I mentioned is more for during the conversion, rather that for different samples. It is particularly important for slower ADC types, such as SAR implementations. Can you explain your 1/2 lsb effect? What type of ADC are you referring to? Different ADC types do require a S/H on the input for signals that are not *highly* oversampled. For example a flash converter can mess up and be quite a bit off if the signal is slewing during conversion. Same with SAR converters. But I don't know of any effect where 1/2 lsb is a threshold. What threshold would you expect? As I recall, 1/2 lsb is the limit to ensure that the conversion would be the 'same' over the conversion time. I'm not sure what you mean by "the conversion would be the 'same' over the conversion time", but I don't see how 1/2 lsb is any magic threshold. If you are working with a flash converter, there are a number of comparators each with a different threshold. The input signal could be right at the edge of one of these thresholds so that a very tiny change in the input signal will cause that threshold to be crossed during the conversion. Maybe I'm not understanding your point. Sorry, I was referring to SAR converters. I should have been more precise. With an SAR converter, if the signal changes during the conversion period, then the converter will fail (at best)*, if the change is more than 1/2 lsb. Therefore, the signal must remain constant (within 1/2 lsb) for the period of the conversion. If the maximum rate of change of signal is known to be such that this will be the case, all is well, if not, you need a sample and hold. You sample the signal, convert the sample, and repeat the process for the next sample. The S/H is designed to minimise the sampling time while ensuring the required hold time is maintained- ie the sample stays within the required 1/2 lsb for the conversion period. Of course, some SAR ADCs have the S/H incorporated within the device, others require either an external one or have provision for the C to be external to permit design flexibility. I understand what you are describing, but you still have not explained the basis of the 1/2 lsb threshold. In an SAR converter the thresholds are still fixed. So the amount of room for noise depends on the value of the signal. If the signal is 1/4 of an lsb from the next conversion threshold then 1/4 lsb of noise will cause a wrong reading. If the signal is within 0.001 lsb of the threshold then 0.001 lsb of change in the signal will cause an error. *by fail, rather depends on the converter. You will at least get an false reading. I recall using one ADC which set a bit indicating a failure to 'find' a 'match'. I recall the details of the parameters of the S/H design being in the application notes of the various ADCs I used over the years, I expect if you look at some you can see for yourself. By their nature (and application) flash converters don't require an S/H but lack the resolution of SAR ADCs. They have other limitations of course. If memory serves, one being that they are not monotonic which was a requirement in the application I tended to apply ADCs (control circuits, feedback loops don't like non-monotonic converters). Actually even flash converters work better with S/H in front of them. A S/H circuit can have a very small aperture window while the converter itself often has a much larger window. Remember that all of these comparators work in parallel with different delays. Even if those delays are small, these devices are designed to sample the fastest signals possible and the variations can only be minimized, not eliminated. So a slewing signal will not convert as accurately and can cause the sort of error where the thermometer code output from the comparators is not self consistent having more than one 0/1 transition in the code. Some flash devices have circuitry to prevent this from causing an output error, but it can add inaccuracy to the result. -- Rick |
What is the point of digital voice?
On 3/8/2015 5:51 PM, Jerry Stuckle wrote:
On 3/8/2015 5:20 PM, rickman wrote: On 3/8/2015 4:37 PM, Jerry Stuckle wrote: On 3/8/2015 2:39 PM, rickman wrote: On 3/8/2015 9:03 AM, Jerry Stuckle wrote: On 3/8/2015 7:35 AM, Brian Reay wrote: Jeff wrote: I will finally point out that your use of the term "slope detecting ADC" is invalid. Google returns exactly 4 hits when this term is entered with quotes. The name of this converter may have slope in it, but that is because the circuit generates a slope, not because it is detecting a slope. Please look up the circuit and use a proper name for it such as integrating ADC or dual slope ADC. The integrating converter is not at all sensitive to the slope of the input signal, otherwise it would not be able to measure a DC signal which has a slope of zero. I'm only replying so that others are not confused by your misstatements. He is probably referring to a CVSD, otherwise known as a Delta Modulator. Jeff I don't think so. In fact, I have to say Jerry seems a bit confused in this particular area, perhaps I have missed something. ADC tend to have a sample and hold prior to the actual ADC convertor, thus the value converted is that at the beginning of the sample period OR if another approach to conversion is used, you get some kind of average over the conversion period. (There are other techniques but those are the main ones.) If you think about, a S/H is required if the rate of change of the input signal means it can change by 1/2 lsb during the conversion time for a SAR ADC. This limits the overall BW of the ADC process. (I recall spending some time convincing a 'seat of the pants engineer' of this when his design wouldn't work. Even when he adopted the suggested changes he insisted his design would have worked if the ADC was more accurate. In fact, it would have made it worse.) No, Brian, I am not confused. It is a form of delta modulation, but is used in an ADC. Two samples are taken, 2 or more times the sample rate (i.e. if the sample rate were 20us, the first sample would be taken every 20us, with the second sample following by 10us or less). The difference is converted to a digital value for transmission. On the other end, the reverse happens. That is not what you have been describing. Now you are saying that the ADC samples the amplitude of the signal just as I have been saying, but now you are adding a step in which the delta is calculated which is what I was describing with ADPCM (although I should have used the simpler and more like your approach DPCM). It is EXACTLY what I've been describing, but you're too stoopid to understand it. But as usual, rather than trying to learn, you argue and prove your stoopidity. I have never heard of using it in the way you are describing though. Even in DPCM the samples are taken at a fixed interval and the delta is calculated on *every* pair of adjacent samples, not just every other. So a sample stream of x0, x1, x2, x3, etc would produce delta values of d0, d1, d2,... not just d0, d1... That's OK. Those types of ADC's haven't heard of you, either, so I guess you don't exist. You describe two samples being taken for each data sample transmitted, ignoring the change in signal between x1 and x2. The signal could not be reconstructed with this data missing. Once again you are proving you have no idea. Yes, the signal can change by 1/2 lsb - but that's true of any ADC. The sample and hold issue is a red herring and in fact, is counter productive in a dual slope converter whose point is to average (integrate) the signal over a period of time filtering higher frequency content. Which has nothing to do with what I'm discussing. But you have to argue, anyway. For any sufficiently high sample rate (i.e. 3x input signal or more), this method is never less accurate than a simple voltage detecting ADC, and in almost every case is more accurate. However, it is a more complex circuit (on both ends), samples a much smaller analog value and requires more exacting components and a higher cost (which is typically the case for any circuit improvements). The sampling method you describe is *not* different from a voltage detecting ADC and therefore can't be better. All you are doing that is different is the analog circuitry is obtaining the slope of the signal over a short interval and is losing the slope of the signal between the samples being ignored. Can you explain how it could be *more* accurate? Once again you show you have no idea what I'm talking about, yet you have to prove your stoopidity by arguing, anyway. I suspect you are confusing the efficiency of the data rate with accuracy. DPCM does provide some compression of the data rate when the signal is over sampled as you seem to be describing. But it does nothing to make the samples more accurate. Once again you show you have no idea what I'm talking about, yet you have to prove your stoopidity by arguing, anyway. As I said - we studied them in one of my EE coursed back in the 70's. I played with them for a while back then, but at the time the ICs were pretty expensive for a college student. Does this technique have a name? Any references? Go to school, get an EE degree, then maybe we can talk about it intelligently. I'm not wasting my time trying to teach the pig to sing. Maybe - IF you were ever more interested in learning than arguing, I would be more interested in discussing it with you. But you have repeatedly proven that is not the case, so I'm not. Ok Jerry. I'm not going to argue with you. I asked you for the name of this ADC technique and you can't come up with one. In this post *every* single one of your replies is ad hominem rather than discussing the issue. Clearly you have no basis for what you are saying. So there is no point in trying to get you to explain any further. You're right - I'm not answering your questions, because you have proven yourself to be incapable of understanding even the simplest explanation. The fact I WON'T answer you questions only means I refuse to try to keep teaching the pig to sing - not that I don't know what I'm talking about. If you want to discuss this, get yourself an EE degree. Then just maybe we can discuss technical topics intelligently. Until then, you can continue to suck your pacifier. My degree is from University of Maryland, an MSEE, 1981. But that is irrelevant. My degree didn't teach me about how ADCs work. I learned that from using them and reading every data book and app note I could find over the years. I'm still waiting for you to show me some sort of evidence that any ADC converters work the way you describe. -- Rick |
What is the point of digital voice?
On 3/8/2015 7:46 PM, rickman wrote:
On 3/8/2015 5:51 PM, Jerry Stuckle wrote: On 3/8/2015 5:20 PM, rickman wrote: On 3/8/2015 4:37 PM, Jerry Stuckle wrote: On 3/8/2015 2:39 PM, rickman wrote: On 3/8/2015 9:03 AM, Jerry Stuckle wrote: On 3/8/2015 7:35 AM, Brian Reay wrote: Jeff wrote: I will finally point out that your use of the term "slope detecting ADC" is invalid. Google returns exactly 4 hits when this term is entered with quotes. The name of this converter may have slope in it, but that is because the circuit generates a slope, not because it is detecting a slope. Please look up the circuit and use a proper name for it such as integrating ADC or dual slope ADC. The integrating converter is not at all sensitive to the slope of the input signal, otherwise it would not be able to measure a DC signal which has a slope of zero. I'm only replying so that others are not confused by your misstatements. He is probably referring to a CVSD, otherwise known as a Delta Modulator. Jeff I don't think so. In fact, I have to say Jerry seems a bit confused in this particular area, perhaps I have missed something. ADC tend to have a sample and hold prior to the actual ADC convertor, thus the value converted is that at the beginning of the sample period OR if another approach to conversion is used, you get some kind of average over the conversion period. (There are other techniques but those are the main ones.) If you think about, a S/H is required if the rate of change of the input signal means it can change by 1/2 lsb during the conversion time for a SAR ADC. This limits the overall BW of the ADC process. (I recall spending some time convincing a 'seat of the pants engineer' of this when his design wouldn't work. Even when he adopted the suggested changes he insisted his design would have worked if the ADC was more accurate. In fact, it would have made it worse.) No, Brian, I am not confused. It is a form of delta modulation, but is used in an ADC. Two samples are taken, 2 or more times the sample rate (i.e. if the sample rate were 20us, the first sample would be taken every 20us, with the second sample following by 10us or less). The difference is converted to a digital value for transmission. On the other end, the reverse happens. That is not what you have been describing. Now you are saying that the ADC samples the amplitude of the signal just as I have been saying, but now you are adding a step in which the delta is calculated which is what I was describing with ADPCM (although I should have used the simpler and more like your approach DPCM). It is EXACTLY what I've been describing, but you're too stoopid to understand it. But as usual, rather than trying to learn, you argue and prove your stoopidity. I have never heard of using it in the way you are describing though. Even in DPCM the samples are taken at a fixed interval and the delta is calculated on *every* pair of adjacent samples, not just every other. So a sample stream of x0, x1, x2, x3, etc would produce delta values of d0, d1, d2,... not just d0, d1... That's OK. Those types of ADC's haven't heard of you, either, so I guess you don't exist. You describe two samples being taken for each data sample transmitted, ignoring the change in signal between x1 and x2. The signal could not be reconstructed with this data missing. Once again you are proving you have no idea. Yes, the signal can change by 1/2 lsb - but that's true of any ADC. The sample and hold issue is a red herring and in fact, is counter productive in a dual slope converter whose point is to average (integrate) the signal over a period of time filtering higher frequency content. Which has nothing to do with what I'm discussing. But you have to argue, anyway. For any sufficiently high sample rate (i.e. 3x input signal or more), this method is never less accurate than a simple voltage detecting ADC, and in almost every case is more accurate. However, it is a more complex circuit (on both ends), samples a much smaller analog value and requires more exacting components and a higher cost (which is typically the case for any circuit improvements). The sampling method you describe is *not* different from a voltage detecting ADC and therefore can't be better. All you are doing that is different is the analog circuitry is obtaining the slope of the signal over a short interval and is losing the slope of the signal between the samples being ignored. Can you explain how it could be *more* accurate? Once again you show you have no idea what I'm talking about, yet you have to prove your stoopidity by arguing, anyway. I suspect you are confusing the efficiency of the data rate with accuracy. DPCM does provide some compression of the data rate when the signal is over sampled as you seem to be describing. But it does nothing to make the samples more accurate. Once again you show you have no idea what I'm talking about, yet you have to prove your stoopidity by arguing, anyway. As I said - we studied them in one of my EE coursed back in the 70's. I played with them for a while back then, but at the time the ICs were pretty expensive for a college student. Does this technique have a name? Any references? Go to school, get an EE degree, then maybe we can talk about it intelligently. I'm not wasting my time trying to teach the pig to sing. Maybe - IF you were ever more interested in learning than arguing, I would be more interested in discussing it with you. But you have repeatedly proven that is not the case, so I'm not. Ok Jerry. I'm not going to argue with you. I asked you for the name of this ADC technique and you can't come up with one. In this post *every* single one of your replies is ad hominem rather than discussing the issue. Clearly you have no basis for what you are saying. So there is no point in trying to get you to explain any further. You're right - I'm not answering your questions, because you have proven yourself to be incapable of understanding even the simplest explanation. The fact I WON'T answer you questions only means I refuse to try to keep teaching the pig to sing - not that I don't know what I'm talking about. If you want to discuss this, get yourself an EE degree. Then just maybe we can discuss technical topics intelligently. Until then, you can continue to suck your pacifier. My degree is from University of Maryland, an MSEE, 1981. But that is irrelevant. My degree didn't teach me about how ADCs work. I learned that from using them and reading every data book and app note I could find over the years. I'm still waiting for you to show me some sort of evidence that any ADC converters work the way you describe. MSEE from University of Maryland? ROFLMAO! I happen to live just a few miles from UMD. I know several graduates of there, some of them EE's. And they know a lot more about EE than you have shown. Including ADC's. I have much more respect for UMD and its grads than that. -- ================== Remove the "x" from my email address Jerry, AI0K ================== |
| All times are GMT +1. The time now is 11:41 PM. |
Powered by vBulletin® Copyright ©2000 - 2025, Jelsoft Enterprises Ltd.
RadioBanter.com