|
What is the point of digital voice?
On 3/8/2015 7:57 PM, Jerry Stuckle wrote:
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. Ok, so I have answered your questions. Prove me full of crap by showing us a reference for the ADC you are describing. -- Rick |
What is the point of digital voice?
On 3/8/2015 8:08 PM, rickman wrote:
On 3/8/2015 7:57 PM, Jerry Stuckle wrote: 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. Ok, so I have answered your questions. Prove me full of crap by showing us a reference for the ADC you are describing. I never asked any questions, Ricky. -- ================== Remove the "x" from my email address Jerry Stuckle ================== |
What is the point of digital voice?
On 3/8/2015 8:15 PM, Jerry Stuckle wrote:
On 3/8/2015 8:08 PM, rickman wrote: On 3/8/2015 7:57 PM, Jerry Stuckle wrote: 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. Ok, so I have answered your questions. Prove me full of crap by showing us a reference for the ADC you are describing. I never asked any questions, Ricky. So you have given up trying to explain yourself and we should consider your previous posts to be things you misremembered and are unwilling to retract? I expect you have confused the functioning of an ADC with that of a compression method like DPCM, possibly because they were in the same device although totally separate functions. That's fine. As long as we are clear. I'm not the only person who seems to think you are wrong on this. -- Rick |
What is the point of digital voice?
On 3/8/2015 9:34 PM, rickman wrote:
On 3/8/2015 8:15 PM, Jerry Stuckle wrote: On 3/8/2015 8:08 PM, rickman wrote: On 3/8/2015 7:57 PM, Jerry Stuckle wrote: 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. Ok, so I have answered your questions. Prove me full of crap by showing us a reference for the ADC you are describing. I never asked any questions, Ricky. So you have given up trying to explain yourself and we should consider your previous posts to be things you misremembered and are unwilling to retract? I expect you have confused the functioning of an ADC with that of a compression method like DPCM, possibly because they were in the same device although totally separate functions. That's fine. As long as we are clear. I'm not the only person who seems to think you are wrong on this. Nope. IF you ever got the MS you claim (which I doubt), you sure didn't learn anything, as has been indicated by multiple posts by you in this newsgroup. But a REAL grad of UMD would know a lot more than you do. But you ARE the only one who thinks I'm wrong. But then you've repeatedly shown you have no understanding of anything but the most basic electronic circuits. And this has NOTHING to do with DPCM - but you're hung up on that, also, because you don't understand the difference. So I suggest you go back to school and actually learn something. And FYI - they were teaching ADC's at Iowa State in 1972-3. And I know they're teaching ADC's at UMD. So your claim you didn't study them is as full of crap as you have repeatedly shown you are. Go back to school. Find some other sucker who's willing to teach the pig to sing. -- ================== Remove the "x" from my email address Jerry, AI0K ================== |
What is the point of digital voice?
In rec.radio.amateur.equipment Jerry Stuckle wrote:
snip piles of crap But you ARE the only one who thinks I'm wrong. Nope. -- Jim Pennino |
What is the point of digital voice?
BORING ...I KNOW MORE THAN YOU ....NA NA NA NA NA ....bloody
professionals...they should get to f**k out of amateur radio and take up gardening or some hobby non-related to their profession ..... |
What is the point of digital voice?
On 08/03/2015 23:42, rickman wrote:
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. 1/2 lsb is resolution of the ADC, any reading can never be certain to be any closer than this. You don't really want your sample changing by more than this during the conversion process. If you don't believe me, I suggest you look at some application notes on SAR ADCs, this is standard stuff. |
What is the point of digital voice?
On 3/9/2015 9:29 AM, Brian Reay wrote:
On 08/03/2015 23:42, rickman wrote: 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. 1/2 lsb is resolution of the ADC, any reading can never be certain to be any closer than this. You don't really want your sample changing by more than this during the conversion process. If you don't believe me, I suggest you look at some application notes on SAR ADCs, this is standard stuff. That's not an explanation. But whatever. Thanks -- Rick |
What is the point of digital voice?
On 3/8/2015 10:19 PM, Jerry Stuckle wrote:
On 3/8/2015 9:34 PM, rickman wrote: That's fine. As long as we are clear. I'm not the only person who seems to think you are wrong on this. Nope. IF you ever got the MS you claim (which I doubt), you sure didn't learn anything, as has been indicated by multiple posts by you in this newsgroup. But a REAL grad of UMD would know a lot more than you do. But you ARE the only one who thinks I'm wrong. But then you've repeatedly shown you have no understanding of anything but the most basic electronic circuits. And this has NOTHING to do with DPCM - but you're hung up on that, also, because you don't understand the difference. So I suggest you go back to school and actually learn something. And FYI - they were teaching ADC's at Iowa State in 1972-3. And I know they're teaching ADC's at UMD. So your claim you didn't study them is as full of crap as you have repeatedly shown you are. Go back to school. Find some other sucker who's willing to teach the pig to sing. Ok, so now you have heard from Jim who also knows you are wrong. I think the fact that you do keep responding, but can only summon ad hominem attacks shows that you realize you are wrong. Rather than face that fact you throw up as much dirt as possible. I guess we are done. -- Rick |
What is the point of digital voice?
On 3/9/2015 11:58 AM, rickman wrote:
On 3/8/2015 10:19 PM, Jerry Stuckle wrote: On 3/8/2015 9:34 PM, rickman wrote: That's fine. As long as we are clear. I'm not the only person who seems to think you are wrong on this. Nope. IF you ever got the MS you claim (which I doubt), you sure didn't learn anything, as has been indicated by multiple posts by you in this newsgroup. But a REAL grad of UMD would know a lot more than you do. But you ARE the only one who thinks I'm wrong. But then you've repeatedly shown you have no understanding of anything but the most basic electronic circuits. And this has NOTHING to do with DPCM - but you're hung up on that, also, because you don't understand the difference. So I suggest you go back to school and actually learn something. And FYI - they were teaching ADC's at Iowa State in 1972-3. And I know they're teaching ADC's at UMD. So your claim you didn't study them is as full of crap as you have repeatedly shown you are. Go back to school. Find some other sucker who's willing to teach the pig to sing. Ok, so now you have heard from Jim who also knows you are wrong. I think the fact that you do keep responding, but can only summon ad hominem attacks shows that you realize you are wrong. Rather than face that fact you throw up as much dirt as possible. I guess we are done. Oh, you mean the argumentative troll who has his head so far up his arse he can see his tonsils? I killfiled him weeks ago. He's even worse than you. -- ================== Remove the "x" from my email address Jerry, AI0K ================== |
What is the point of digital voice?
In rec.radio.amateur.equipment Jerry Stuckle wrote:
snip Oh, you mean the argumentative troll who has his head so far up his arse he can see his tonsils? Yeah sure, everyone that disagrees with you is a troll. Yeah sure, your credentials are better than anyone that disagrees with you. Yeah sure, your references are better than those of anyone that disagrees with you. I killfiled him weeks ago. He's even worse than you. Yeah sure. -- Jim Pennino |
What is the point of digital voice?
|
What is the point of digital voice?
On 3/6/2015 1:12 PM, 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. No... I'm not.... I do know the difference. Had a Super Regen for VHF as well. -- 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 3/6/2015 3:03 PM, Michael Black wrote:
On Fri, 6 Mar 2015, gareth wrote: 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. You sir... Are correct. -- 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 3/6/2015 1:06 PM, John Davis wrote:
On 2/26/2015 3:55 AM, AndyW wrote: 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). I just got my Digital equipment back in operating condition (Had several issues, New computer lacked an AUDIO IN jack (Now have a dongle) Dongle needed audio on the RING, not the TIP, I had used a mono plug (no ring) Bad transformer (replaced with a better one) When I say JUST... I mean yesterday But back before the old computer died this is what I observed with digital text 1: Signals I Could not hear (and I have good hearing) It could often decode. 2: Most signals either decoded...Or not... But sometimes.... IT was broken. From other expierence, both with digital video and audio As the signal degrades with analog it is 10,9,8,7 and so on With Digital it is 10,10,10,10,10,10,10,5,0 (NOTE: Number of 10's is not to scale) -- 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 3/6/2015 3:11 PM, Jerry Stuckle wrote:
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. You are partially right.. CD uses a specific sample rate and bit size, And I believe you are correct as to what they are MP3 can use a very wide range of sample rates, and different bit sizes as well.... Crank the bit rate and sample size up enough and yes, I am not going to be able to tell the difference. (A golden ear I'm not). But the bandwith needed goes way up. Some "Cred"info..My daughter, in whom I am well pleased, IS, among other things, a Classical Musician and music teacher...In the past I have been her "Recording Engineer" I also do recording in other venues as well... Usualy ATRAC, but sometimes CD quality. I have played a lot with Bit Rates, sample sizes and sample rates.using Total Recorder PRO and the LAME codec. Fun Fact: Remember Spaceship One, the one that won the X-Prize? Well, Dr Space (David Livingston) complained during the first flight about having to change cassettes in his audio recorder... I tossed Total Recorder on the job and within minutes of the program end he had an MP3 in his mail box. Next flight he mentioned his new log recorder (Total Recorder PRO) and credited me with the suggestion. I have hundreds (IF not thousands) of hours of live recordings lying about here. -- 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 3/6/2015 1:12 PM, 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. No... I'm not.... I do know the difference. Had a Super Regen for VHF as well. So, if the Knight Kit could also go into super-regeneration, how was the quenching achieved? |
What is the point of digital voice?
On Sun, 15 Mar 2015, gareth wrote:
"John Davis" wrote in message ... On 3/6/2015 1:12 PM, 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. No... I'm not.... I do know the difference. Had a Super Regen for VHF as well. So, if the Knight Kit could also go into super-regeneration, how was the quenching achieved? By the same means it happened by accident when Armstrong was playing with regen receivers. Most discussion of superregens has the same tube doing the receiving and quenching. That tends to confuse the operation of the scheme, and like I said, eventually the ARRL Handbook description of the superregen devolved to explaining how the tube could quench. The right components and that regen receiver can be a superregen receiver. And like I said, it can be both, just picking components so the regen control can get some regeration (for improved selectivity), turn into an oscillator (for beating against incoming signals) and then a bit more and it goes into superregeneration. Like I said, superregeneration is just an addition to regeneration, a useful feature since it makes operation much less fussy (no need to constantly ride the regen control), but nevertheless just an extension of the basic concept of regeneration. Michael |
What is the point of digital voice?
"Michael Black" wrote in message
news:alpine.LNX.2.02.1503161526020.18553@darkstar. example.org... Like I said, superregeneration is just an addition to regeneration, a useful feature since it makes operation much less fussy (no need to constantly ride the regen control), but nevertheless just an extension of the basic concept of regeneration. So, let's say that you are listening to a weak CW signal, you turn up the regen control until you get super-regen, at which point the CW signal disappears because the CIO effect has now been shifted out to the sidebands created by the quenching? |
What is the point of digital voice?
In rec.radio.amateur.equipment gareth wrote:
"Michael Black" wrote in message news:alpine.LNX.2.02.1503161526020.18553@darkstar. example.org... Like I said, superregeneration is just an addition to regeneration, a useful feature since it makes operation much less fussy (no need to constantly ride the regen control), but nevertheless just an extension of the basic concept of regeneration. So, let's say that you are listening to a weak CW signal, you turn up the regen control until you get super-regen, at which point the CW signal disappears because the CIO effect has now been shifted out to the sidebands created by the quenching? Nope. http://en.wikipedia.org/wiki/Regener...ative_receiver -- Jim Pennino |
| All times are GMT +1. The time now is 11:45 PM. |
Powered by vBulletin® Copyright ©2000 - 2025, Jelsoft Enterprises Ltd.
RadioBanter.com