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#81
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DSD Recording Good. PCM recordings bad?
"Karl Uppiano" wrote in message ... "Randy Yates" wrote in message ... "Karl Uppiano" writes: "Arny Krueger" wrote in message ... "Karl Uppiano" wrote in message "François Yves Le Gal" wrote in message ... On Sun, 01 Feb 2004 12:53:18 -0800, chung wrote: One, DSD/SACD proponents claim the much wider bandwidth over CD's, and filtering will reduce significantly that claimed advantage. Having a gentle low pass filter at 60 or 100 KHz doesn't significantly reduce SACD's bandwith. You get more than PCM 96 or 192 in both cases! I would start a 3 dB/octave rolloff at 20kHz or so. Ironically, most if not all people can't hear the difference a brickwall filter at 16 KHz makes, if the filter is well-designed. Don't believe me? Listen for yourself at http://www.pcabx.com/technical/low_pass/index.htm . I believe you. I realize it isn't considered state of the art anymore, but I wonder how many people realize that FM stereo uses a 38 kHz sample rate? Actually, the L-R signal is AM-DSB-SC-modulated on a 38 kHz subcarrier and a 19 kHz pilot tone added. This, along with the L+R baseband signal, is then FM modulated onto the carrier. The balanced modulator is mathematically equivalent to a sampled system in which the left and right channels are alternately sampled at a 38 kHz rate and used to directly modulate the carrier. The 19 kHz pilot is derived from the 38 kHz sample frequency and injected onto the carrier at 9% modulation. Even without the math, it makes sense if you think about it: Alternately sampling a mono signal (L = R) will give you no subcarrier (L - R = 0), which is exactly what happens with AM-DSB-SC. The analysis is a little more complicated when a stereo signal is encoded, but the results are identical using either approach. The balanced modulator approach was used in FM stereo generators until the mid 1970's or so, when the alternately sampled approach became possible with the advent of TTL and FET analog switches. The switched stereo generators require less maintenance. The phase-lock-loop demodulators in most receivers is the same system in reverse. Nyquist's sampling theorem applies to FM stereo. Therefore, 19 kHz is the highest audio frequency theoretically possible, but you need a guard band for the pilot, so 15 kHz is typical, with some extremely good stereo generators yielding 16 to 18 kHz. By the way, the transition band for the analog anti-aliasing filters (required for any FM stereo generator) need to go from 0 dB at 15 kHz to -60 dB at 19 kHz. Stereo decoders in most receivers usually don't bother to control the high frequency energy very well, which is why many cassette decks with Dolby noise reduction have "MPX filters" to prevent this energy from messing up the Dolby encoding. |
#82
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DSD Recording Good. PCM recordings bad?
chung wrote in message news:ec705$401ebdd9$c247604
If you take into account tolerances of the 5 caps, you would have a noticeable ripple in the passband, as well as mismatches between L/R. Plus the cost of the 10 capacitors is not insignificant. And more if you have 5 channels. The much better way is simply apply digital filtering, or use better dithering schemes, but then we are back to LPCM, and not DSD . I agree. If you look at the data you will see that with perfect component values, the flatness is: -0.1/-0.2 which 0.1 dB p-p. Using real componet will increase the ripple. A pratical best flatness spec. (for real world components) would be +0.1/-0.3 dB. Using 1% tolerance components, I ran a Monte Carlo analysis (on 100,000 units). I got the following results: +0.20/-0.40 dB flatness 100% yield +0.15/-0.35 dB flatness 99.91% yield +0.10/-0.30 dB flatness 98.80% yield It would be fairly expensive to use 1% tolerance components. Probably add $3 or $4 to the cost of manufacture. For an SACD player that sells for $500 to $1500, that would be acceptable. However, I don't know if a flatness spec of +/- 0.2dB would be acceptable to high-end buyers? Bob Stanton |
#83
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DSD Recording Good. PCM recordings bad?
chung wrote in message news:ec705$401ebdd9$c247604
If you take into account tolerances of the 5 caps, you would have a noticeable ripple in the passband, as well as mismatches between L/R. Plus the cost of the 10 capacitors is not insignificant. And more if you have 5 channels. The much better way is simply apply digital filtering, or use better dithering schemes, but then we are back to LPCM, and not DSD . I agree. If you look at the data you will see that with perfect component values, the flatness is: -0.1/-0.2 which 0.1 dB p-p. Using real componet will increase the ripple. A pratical best flatness spec. (for real world components) would be +0.1/-0.3 dB. Using 1% tolerance components, I ran a Monte Carlo analysis (on 100,000 units). I got the following results: +0.20/-0.40 dB flatness 100% yield +0.15/-0.35 dB flatness 99.91% yield +0.10/-0.30 dB flatness 98.80% yield It would be fairly expensive to use 1% tolerance components. Probably add $3 or $4 to the cost of manufacture. For an SACD player that sells for $500 to $1500, that would be acceptable. However, I don't know if a flatness spec of +/- 0.2dB would be acceptable to high-end buyers? Bob Stanton |
#84
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DSD Recording Good. PCM recordings bad?
chung wrote in message news:ec705$401ebdd9$c247604
If you take into account tolerances of the 5 caps, you would have a noticeable ripple in the passband, as well as mismatches between L/R. Plus the cost of the 10 capacitors is not insignificant. And more if you have 5 channels. The much better way is simply apply digital filtering, or use better dithering schemes, but then we are back to LPCM, and not DSD . I agree. If you look at the data you will see that with perfect component values, the flatness is: -0.1/-0.2 which 0.1 dB p-p. Using real componet will increase the ripple. A pratical best flatness spec. (for real world components) would be +0.1/-0.3 dB. Using 1% tolerance components, I ran a Monte Carlo analysis (on 100,000 units). I got the following results: +0.20/-0.40 dB flatness 100% yield +0.15/-0.35 dB flatness 99.91% yield +0.10/-0.30 dB flatness 98.80% yield It would be fairly expensive to use 1% tolerance components. Probably add $3 or $4 to the cost of manufacture. For an SACD player that sells for $500 to $1500, that would be acceptable. However, I don't know if a flatness spec of +/- 0.2dB would be acceptable to high-end buyers? Bob Stanton |
#85
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DSD Recording Good. PCM recordings bad?
chung wrote in message news:ec705$401ebdd9$c247604
If you take into account tolerances of the 5 caps, you would have a noticeable ripple in the passband, as well as mismatches between L/R. Plus the cost of the 10 capacitors is not insignificant. And more if you have 5 channels. The much better way is simply apply digital filtering, or use better dithering schemes, but then we are back to LPCM, and not DSD . I agree. If you look at the data you will see that with perfect component values, the flatness is: -0.1/-0.2 which 0.1 dB p-p. Using real componet will increase the ripple. A pratical best flatness spec. (for real world components) would be +0.1/-0.3 dB. Using 1% tolerance components, I ran a Monte Carlo analysis (on 100,000 units). I got the following results: +0.20/-0.40 dB flatness 100% yield +0.15/-0.35 dB flatness 99.91% yield +0.10/-0.30 dB flatness 98.80% yield It would be fairly expensive to use 1% tolerance components. Probably add $3 or $4 to the cost of manufacture. For an SACD player that sells for $500 to $1500, that would be acceptable. However, I don't know if a flatness spec of +/- 0.2dB would be acceptable to high-end buyers? Bob Stanton |
#86
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DSD Recording Good. PCM recordings bad?
"François Yves Le Gal" wrote in message
On Tue, 03 Feb 2004 04:21:57 GMT, "Karl Uppiano" wrote: Nope, FM stereo is a sampled system. Well, it's not according to my book, as the 38 KHz carrier isn't a sampling frequency. Just another page in the big book of things you're ignorant about, François. The FM stereo system is generally demodulated these days by means of a circuit that could easily pass for an analog demultiplexer. It well known among technical folk who have studied it with sufficient depth, that there is an interesting duality about FM stereo. FM Stereo is as Randy says: "Actually, the L-R signal is AM-DSB-SC-modulated on a 38 kHz subcarrier and a 19 kHz pilot tone added. This, along with the L+R baseband signal, is then FM modulated onto the carrier." and it can be demodulated accordingly. But, FM Stereo can also be viewed as Karl says: "Left and right sampled alternately at 38 kHz is the same as L+R and L-R fed into a balanced modulator with a suppressed carrier. The balanced modulator approach was used in FM stereo generators until the mid 1970's or so, when the alternately sampled approach became possible with the advent of TTL and FET analog switches. Both systems also inject a 19 kHz pilot tone at a 9% modulation level to turn on the "stereo" light, and to give phase-lock-loop stereo decoders (also sampled systems) something to lock onto." Now I know from years of experience that in the posturing-ridden one-track mind of the anonymous internet troll that posts as "François Yves Le Gal" both of these statements can't be true at the same time. But, in the real world, they are. The reasons why can be instructive to those with the mental powers and sufficient training and experience to properly contemplate them. Sorry to leave you out, "François Yves Le Gal" or whoever you really are. |
#87
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DSD Recording Good. PCM recordings bad?
"François Yves Le Gal" wrote in message
On Tue, 03 Feb 2004 04:21:57 GMT, "Karl Uppiano" wrote: Nope, FM stereo is a sampled system. Well, it's not according to my book, as the 38 KHz carrier isn't a sampling frequency. Just another page in the big book of things you're ignorant about, François. The FM stereo system is generally demodulated these days by means of a circuit that could easily pass for an analog demultiplexer. It well known among technical folk who have studied it with sufficient depth, that there is an interesting duality about FM stereo. FM Stereo is as Randy says: "Actually, the L-R signal is AM-DSB-SC-modulated on a 38 kHz subcarrier and a 19 kHz pilot tone added. This, along with the L+R baseband signal, is then FM modulated onto the carrier." and it can be demodulated accordingly. But, FM Stereo can also be viewed as Karl says: "Left and right sampled alternately at 38 kHz is the same as L+R and L-R fed into a balanced modulator with a suppressed carrier. The balanced modulator approach was used in FM stereo generators until the mid 1970's or so, when the alternately sampled approach became possible with the advent of TTL and FET analog switches. Both systems also inject a 19 kHz pilot tone at a 9% modulation level to turn on the "stereo" light, and to give phase-lock-loop stereo decoders (also sampled systems) something to lock onto." Now I know from years of experience that in the posturing-ridden one-track mind of the anonymous internet troll that posts as "François Yves Le Gal" both of these statements can't be true at the same time. But, in the real world, they are. The reasons why can be instructive to those with the mental powers and sufficient training and experience to properly contemplate them. Sorry to leave you out, "François Yves Le Gal" or whoever you really are. |
#88
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DSD Recording Good. PCM recordings bad?
"François Yves Le Gal" wrote in message
On Tue, 03 Feb 2004 04:21:57 GMT, "Karl Uppiano" wrote: Nope, FM stereo is a sampled system. Well, it's not according to my book, as the 38 KHz carrier isn't a sampling frequency. Just another page in the big book of things you're ignorant about, François. The FM stereo system is generally demodulated these days by means of a circuit that could easily pass for an analog demultiplexer. It well known among technical folk who have studied it with sufficient depth, that there is an interesting duality about FM stereo. FM Stereo is as Randy says: "Actually, the L-R signal is AM-DSB-SC-modulated on a 38 kHz subcarrier and a 19 kHz pilot tone added. This, along with the L+R baseband signal, is then FM modulated onto the carrier." and it can be demodulated accordingly. But, FM Stereo can also be viewed as Karl says: "Left and right sampled alternately at 38 kHz is the same as L+R and L-R fed into a balanced modulator with a suppressed carrier. The balanced modulator approach was used in FM stereo generators until the mid 1970's or so, when the alternately sampled approach became possible with the advent of TTL and FET analog switches. Both systems also inject a 19 kHz pilot tone at a 9% modulation level to turn on the "stereo" light, and to give phase-lock-loop stereo decoders (also sampled systems) something to lock onto." Now I know from years of experience that in the posturing-ridden one-track mind of the anonymous internet troll that posts as "François Yves Le Gal" both of these statements can't be true at the same time. But, in the real world, they are. The reasons why can be instructive to those with the mental powers and sufficient training and experience to properly contemplate them. Sorry to leave you out, "François Yves Le Gal" or whoever you really are. |
#89
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DSD Recording Good. PCM recordings bad?
"François Yves Le Gal" wrote in message
On Tue, 03 Feb 2004 04:21:57 GMT, "Karl Uppiano" wrote: Nope, FM stereo is a sampled system. Well, it's not according to my book, as the 38 KHz carrier isn't a sampling frequency. Just another page in the big book of things you're ignorant about, François. The FM stereo system is generally demodulated these days by means of a circuit that could easily pass for an analog demultiplexer. It well known among technical folk who have studied it with sufficient depth, that there is an interesting duality about FM stereo. FM Stereo is as Randy says: "Actually, the L-R signal is AM-DSB-SC-modulated on a 38 kHz subcarrier and a 19 kHz pilot tone added. This, along with the L+R baseband signal, is then FM modulated onto the carrier." and it can be demodulated accordingly. But, FM Stereo can also be viewed as Karl says: "Left and right sampled alternately at 38 kHz is the same as L+R and L-R fed into a balanced modulator with a suppressed carrier. The balanced modulator approach was used in FM stereo generators until the mid 1970's or so, when the alternately sampled approach became possible with the advent of TTL and FET analog switches. Both systems also inject a 19 kHz pilot tone at a 9% modulation level to turn on the "stereo" light, and to give phase-lock-loop stereo decoders (also sampled systems) something to lock onto." Now I know from years of experience that in the posturing-ridden one-track mind of the anonymous internet troll that posts as "François Yves Le Gal" both of these statements can't be true at the same time. But, in the real world, they are. The reasons why can be instructive to those with the mental powers and sufficient training and experience to properly contemplate them. Sorry to leave you out, "François Yves Le Gal" or whoever you really are. |
#90
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DSD Recording Good. PCM recordings bad?
"chung" wrote in message
vers.com Robert Stanton wrote: "Arny Krueger" wrote in message ... "Robert Stanton" wrote in message m chung wrote in message ... Harry Lavo wrote: Isn't it interesting that we "subjectivists" here are always whipsawed by the "objectivists" for thinking that extended frequency response is a benefit "because it can't be heard". Then, in defense of DVD-A, the increased noise in the ultrasonic range is bandied about as making DSD/SACD "inferior". Actually it's interesting that subjectivists said that they need the ultrasonic bandwidth to fully perceive music, and yet they like SACD's despite the much higher (by orders of magnitude) ultrasonic noise inherent in the SACD format. I often see ultrasonic noise mentioned as a problem for SACD players. If ultrasonic noise were really a problem, it could be easily eliminated with a small, active lowpass filter. Why wouldn't the manufactures of "high end" SACD players, just filter it out? Wouldn't the filter affect the overtones of the music just as much as it affects the noise? Yes, it will chop off all overtones above 30kHz. But, we humans can't hear above 25kHz, so we won't hear the difference. You and I may agree on that, but one of the touted features of SACD is the much broader bandwidth, higher than 25 KHz. If you limit it to 25KHz, it could not compete against the other hi-rez formats, such as 24/96 or 24/192, which are flat up to close to half the sampling frequency. Agreed. |
#91
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DSD Recording Good. PCM recordings bad?
"chung" wrote in message
vers.com Robert Stanton wrote: "Arny Krueger" wrote in message ... "Robert Stanton" wrote in message m chung wrote in message ... Harry Lavo wrote: Isn't it interesting that we "subjectivists" here are always whipsawed by the "objectivists" for thinking that extended frequency response is a benefit "because it can't be heard". Then, in defense of DVD-A, the increased noise in the ultrasonic range is bandied about as making DSD/SACD "inferior". Actually it's interesting that subjectivists said that they need the ultrasonic bandwidth to fully perceive music, and yet they like SACD's despite the much higher (by orders of magnitude) ultrasonic noise inherent in the SACD format. I often see ultrasonic noise mentioned as a problem for SACD players. If ultrasonic noise were really a problem, it could be easily eliminated with a small, active lowpass filter. Why wouldn't the manufactures of "high end" SACD players, just filter it out? Wouldn't the filter affect the overtones of the music just as much as it affects the noise? Yes, it will chop off all overtones above 30kHz. But, we humans can't hear above 25kHz, so we won't hear the difference. You and I may agree on that, but one of the touted features of SACD is the much broader bandwidth, higher than 25 KHz. If you limit it to 25KHz, it could not compete against the other hi-rez formats, such as 24/96 or 24/192, which are flat up to close to half the sampling frequency. Agreed. |
#92
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DSD Recording Good. PCM recordings bad?
"chung" wrote in message
vers.com Robert Stanton wrote: "Arny Krueger" wrote in message ... "Robert Stanton" wrote in message m chung wrote in message ... Harry Lavo wrote: Isn't it interesting that we "subjectivists" here are always whipsawed by the "objectivists" for thinking that extended frequency response is a benefit "because it can't be heard". Then, in defense of DVD-A, the increased noise in the ultrasonic range is bandied about as making DSD/SACD "inferior". Actually it's interesting that subjectivists said that they need the ultrasonic bandwidth to fully perceive music, and yet they like SACD's despite the much higher (by orders of magnitude) ultrasonic noise inherent in the SACD format. I often see ultrasonic noise mentioned as a problem for SACD players. If ultrasonic noise were really a problem, it could be easily eliminated with a small, active lowpass filter. Why wouldn't the manufactures of "high end" SACD players, just filter it out? Wouldn't the filter affect the overtones of the music just as much as it affects the noise? Yes, it will chop off all overtones above 30kHz. But, we humans can't hear above 25kHz, so we won't hear the difference. You and I may agree on that, but one of the touted features of SACD is the much broader bandwidth, higher than 25 KHz. If you limit it to 25KHz, it could not compete against the other hi-rez formats, such as 24/96 or 24/192, which are flat up to close to half the sampling frequency. Agreed. |
#93
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DSD Recording Good. PCM recordings bad?
"chung" wrote in message
vers.com Robert Stanton wrote: "Arny Krueger" wrote in message ... "Robert Stanton" wrote in message m chung wrote in message ... Harry Lavo wrote: Isn't it interesting that we "subjectivists" here are always whipsawed by the "objectivists" for thinking that extended frequency response is a benefit "because it can't be heard". Then, in defense of DVD-A, the increased noise in the ultrasonic range is bandied about as making DSD/SACD "inferior". Actually it's interesting that subjectivists said that they need the ultrasonic bandwidth to fully perceive music, and yet they like SACD's despite the much higher (by orders of magnitude) ultrasonic noise inherent in the SACD format. I often see ultrasonic noise mentioned as a problem for SACD players. If ultrasonic noise were really a problem, it could be easily eliminated with a small, active lowpass filter. Why wouldn't the manufactures of "high end" SACD players, just filter it out? Wouldn't the filter affect the overtones of the music just as much as it affects the noise? Yes, it will chop off all overtones above 30kHz. But, we humans can't hear above 25kHz, so we won't hear the difference. You and I may agree on that, but one of the touted features of SACD is the much broader bandwidth, higher than 25 KHz. If you limit it to 25KHz, it could not compete against the other hi-rez formats, such as 24/96 or 24/192, which are flat up to close to half the sampling frequency. Agreed. |
#94
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DSD Recording Good. PCM recordings bad?
"François Yves Le Gal" wrote in message ... On Tue, 03 Feb 2004 04:21:57 GMT, "Karl Uppiano" wrote: Nope, FM stereo is a sampled system. Well, it's not according to my book, as the 38 Khz carrier isn't a sampling frequency. You're in denial, then. Switching generators are "the way FM stereo was done" right up until DSP processors came along that generate the signals numerically. All three systems are in daily use now. It doesn't matter anyway, 19 kHz is the Nyquist frequency regardless of the mechanism used to generate the signal. The spectra are identical. My original point was this: All stereo generators require brick wall anti-aliasing filters to keep the baseband and the subcarrier sidebands from overlapping (aliasing), in addition to a guard band to protect the 19 kHz pilot. So FM stereo is strictly limited in bandwidth to about 15 kHz to 18 kHz best case. FWIW, TV stereo uses a similar system, except it's tied to the horizontal sweep frequency, which is even lower (15.734 kHz doubled to a sampling rate of 31.468 kHz). Meaning that TV stereo is limited to about 14 kHz. Same type brick wall AA filters are required. |
#95
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DSD Recording Good. PCM recordings bad?
"François Yves Le Gal" wrote in message ... On Tue, 03 Feb 2004 04:21:57 GMT, "Karl Uppiano" wrote: Nope, FM stereo is a sampled system. Well, it's not according to my book, as the 38 Khz carrier isn't a sampling frequency. You're in denial, then. Switching generators are "the way FM stereo was done" right up until DSP processors came along that generate the signals numerically. All three systems are in daily use now. It doesn't matter anyway, 19 kHz is the Nyquist frequency regardless of the mechanism used to generate the signal. The spectra are identical. My original point was this: All stereo generators require brick wall anti-aliasing filters to keep the baseband and the subcarrier sidebands from overlapping (aliasing), in addition to a guard band to protect the 19 kHz pilot. So FM stereo is strictly limited in bandwidth to about 15 kHz to 18 kHz best case. FWIW, TV stereo uses a similar system, except it's tied to the horizontal sweep frequency, which is even lower (15.734 kHz doubled to a sampling rate of 31.468 kHz). Meaning that TV stereo is limited to about 14 kHz. Same type brick wall AA filters are required. |
#96
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DSD Recording Good. PCM recordings bad?
"François Yves Le Gal" wrote in message ... On Tue, 03 Feb 2004 04:21:57 GMT, "Karl Uppiano" wrote: Nope, FM stereo is a sampled system. Well, it's not according to my book, as the 38 Khz carrier isn't a sampling frequency. You're in denial, then. Switching generators are "the way FM stereo was done" right up until DSP processors came along that generate the signals numerically. All three systems are in daily use now. It doesn't matter anyway, 19 kHz is the Nyquist frequency regardless of the mechanism used to generate the signal. The spectra are identical. My original point was this: All stereo generators require brick wall anti-aliasing filters to keep the baseband and the subcarrier sidebands from overlapping (aliasing), in addition to a guard band to protect the 19 kHz pilot. So FM stereo is strictly limited in bandwidth to about 15 kHz to 18 kHz best case. FWIW, TV stereo uses a similar system, except it's tied to the horizontal sweep frequency, which is even lower (15.734 kHz doubled to a sampling rate of 31.468 kHz). Meaning that TV stereo is limited to about 14 kHz. Same type brick wall AA filters are required. |
#97
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DSD Recording Good. PCM recordings bad?
"François Yves Le Gal" wrote in message ... On Tue, 03 Feb 2004 04:21:57 GMT, "Karl Uppiano" wrote: Nope, FM stereo is a sampled system. Well, it's not according to my book, as the 38 Khz carrier isn't a sampling frequency. You're in denial, then. Switching generators are "the way FM stereo was done" right up until DSP processors came along that generate the signals numerically. All three systems are in daily use now. It doesn't matter anyway, 19 kHz is the Nyquist frequency regardless of the mechanism used to generate the signal. The spectra are identical. My original point was this: All stereo generators require brick wall anti-aliasing filters to keep the baseband and the subcarrier sidebands from overlapping (aliasing), in addition to a guard band to protect the 19 kHz pilot. So FM stereo is strictly limited in bandwidth to about 15 kHz to 18 kHz best case. FWIW, TV stereo uses a similar system, except it's tied to the horizontal sweep frequency, which is even lower (15.734 kHz doubled to a sampling rate of 31.468 kHz). Meaning that TV stereo is limited to about 14 kHz. Same type brick wall AA filters are required. |
#98
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DSD Recording Good. PCM recordings bad?
Robert Stanton wrote:
chung wrote in message news:ec705$401ebdd9$c247604 If you take into account tolerances of the 5 caps, you would have a noticeable ripple in the passband, as well as mismatches between L/R. Plus the cost of the 10 capacitors is not insignificant. And more if you have 5 channels. The much better way is simply apply digital filtering, or use better dithering schemes, but then we are back to LPCM, and not DSD . I agree. If you look at the data you will see that with perfect component values, the flatness is: -0.1/-0.2 which 0.1 dB p-p. Using real componet will increase the ripple. A pratical best flatness spec. (for real world components) would be +0.1/-0.3 dB. Using 1% tolerance components, I ran a Monte Carlo analysis (on 100,000 units). I got the following results: +0.20/-0.40 dB flatness 100% yield +0.15/-0.35 dB flatness 99.91% yield +0.10/-0.30 dB flatness 98.80% yield It would be fairly expensive to use 1% tolerance components. Probably add $3 or $4 to the cost of manufacture. For an SACD player that sells for $500 to $1500, that would be acceptable. 1% capacitors are very expensive. Maybe Sony can get them cheaper, but SACD is not that huge a market, yet. And a 98.8% yield for a filter is absolutely unacceptable for companies like Sony. (It's way too low.) I think you have to get at least a 6-sigma yield for a small circuit like a filter. That's why digital filtering is so wonderful. However, I don't know if a flatness spec of +/- 0.2dB would be acceptable to high-end buyers? It looks worse than good redbook CD players. The bigger problem, of course, is that you can't reduce the BW to 25KHz and still claim wide-band. |
#99
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DSD Recording Good. PCM recordings bad?
Robert Stanton wrote:
chung wrote in message news:ec705$401ebdd9$c247604 If you take into account tolerances of the 5 caps, you would have a noticeable ripple in the passband, as well as mismatches between L/R. Plus the cost of the 10 capacitors is not insignificant. And more if you have 5 channels. The much better way is simply apply digital filtering, or use better dithering schemes, but then we are back to LPCM, and not DSD . I agree. If you look at the data you will see that with perfect component values, the flatness is: -0.1/-0.2 which 0.1 dB p-p. Using real componet will increase the ripple. A pratical best flatness spec. (for real world components) would be +0.1/-0.3 dB. Using 1% tolerance components, I ran a Monte Carlo analysis (on 100,000 units). I got the following results: +0.20/-0.40 dB flatness 100% yield +0.15/-0.35 dB flatness 99.91% yield +0.10/-0.30 dB flatness 98.80% yield It would be fairly expensive to use 1% tolerance components. Probably add $3 or $4 to the cost of manufacture. For an SACD player that sells for $500 to $1500, that would be acceptable. 1% capacitors are very expensive. Maybe Sony can get them cheaper, but SACD is not that huge a market, yet. And a 98.8% yield for a filter is absolutely unacceptable for companies like Sony. (It's way too low.) I think you have to get at least a 6-sigma yield for a small circuit like a filter. That's why digital filtering is so wonderful. However, I don't know if a flatness spec of +/- 0.2dB would be acceptable to high-end buyers? It looks worse than good redbook CD players. The bigger problem, of course, is that you can't reduce the BW to 25KHz and still claim wide-band. |
#100
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DSD Recording Good. PCM recordings bad?
Robert Stanton wrote:
chung wrote in message news:ec705$401ebdd9$c247604 If you take into account tolerances of the 5 caps, you would have a noticeable ripple in the passband, as well as mismatches between L/R. Plus the cost of the 10 capacitors is not insignificant. And more if you have 5 channels. The much better way is simply apply digital filtering, or use better dithering schemes, but then we are back to LPCM, and not DSD . I agree. If you look at the data you will see that with perfect component values, the flatness is: -0.1/-0.2 which 0.1 dB p-p. Using real componet will increase the ripple. A pratical best flatness spec. (for real world components) would be +0.1/-0.3 dB. Using 1% tolerance components, I ran a Monte Carlo analysis (on 100,000 units). I got the following results: +0.20/-0.40 dB flatness 100% yield +0.15/-0.35 dB flatness 99.91% yield +0.10/-0.30 dB flatness 98.80% yield It would be fairly expensive to use 1% tolerance components. Probably add $3 or $4 to the cost of manufacture. For an SACD player that sells for $500 to $1500, that would be acceptable. 1% capacitors are very expensive. Maybe Sony can get them cheaper, but SACD is not that huge a market, yet. And a 98.8% yield for a filter is absolutely unacceptable for companies like Sony. (It's way too low.) I think you have to get at least a 6-sigma yield for a small circuit like a filter. That's why digital filtering is so wonderful. However, I don't know if a flatness spec of +/- 0.2dB would be acceptable to high-end buyers? It looks worse than good redbook CD players. The bigger problem, of course, is that you can't reduce the BW to 25KHz and still claim wide-band. |
#101
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DSD Recording Good. PCM recordings bad?
Robert Stanton wrote:
chung wrote in message news:ec705$401ebdd9$c247604 If you take into account tolerances of the 5 caps, you would have a noticeable ripple in the passband, as well as mismatches between L/R. Plus the cost of the 10 capacitors is not insignificant. And more if you have 5 channels. The much better way is simply apply digital filtering, or use better dithering schemes, but then we are back to LPCM, and not DSD . I agree. If you look at the data you will see that with perfect component values, the flatness is: -0.1/-0.2 which 0.1 dB p-p. Using real componet will increase the ripple. A pratical best flatness spec. (for real world components) would be +0.1/-0.3 dB. Using 1% tolerance components, I ran a Monte Carlo analysis (on 100,000 units). I got the following results: +0.20/-0.40 dB flatness 100% yield +0.15/-0.35 dB flatness 99.91% yield +0.10/-0.30 dB flatness 98.80% yield It would be fairly expensive to use 1% tolerance components. Probably add $3 or $4 to the cost of manufacture. For an SACD player that sells for $500 to $1500, that would be acceptable. 1% capacitors are very expensive. Maybe Sony can get them cheaper, but SACD is not that huge a market, yet. And a 98.8% yield for a filter is absolutely unacceptable for companies like Sony. (It's way too low.) I think you have to get at least a 6-sigma yield for a small circuit like a filter. That's why digital filtering is so wonderful. However, I don't know if a flatness spec of +/- 0.2dB would be acceptable to high-end buyers? It looks worse than good redbook CD players. The bigger problem, of course, is that you can't reduce the BW to 25KHz and still claim wide-band. |
#102
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DSD Recording Good. PCM recordings bad?
Fran=E7ois Yves Le Gal wrote:
On Tue, 03 Feb 2004 16:53:36 GMT, "Karl Uppiano" karl_uppiano@verizon.= net wrote: =20 You're in denial, then.=20 =20 I'm not, as FM is still transmitted in analog - not digital - form, eve= n if the wave is the result of digital manipulations. Sampling can be done in the analog as well as digital domains. Your=20 problem is that you cannot visualize analog sampling. You can't properly state that 38 KHz is the sampling frequency -=20 Why not? You are sequentially "looking at" left and right channels. and what about SCA side channels at 76 KHz, BTW ? =20 "Frequency modulation (FM): Modulation in which the instantaneous frequ= ency of a sine wave carrier is caused to depart from the center frequency by= an amount proportional to the instantaneous value of the modulating signal= =2E=20 Non-sequitors. Go get a textbook on communications, and look at the mathematical=20 descriptions of the FM multiplex signal. =20 Note 1: In FM, the carrier frequency is called the center frequency. " =20 The sampling frequency has nothing to do with the final carrier frequency= =2E Federal Standard 1037C =20 |
#103
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DSD Recording Good. PCM recordings bad?
Fran=E7ois Yves Le Gal wrote:
On Tue, 03 Feb 2004 16:53:36 GMT, "Karl Uppiano" karl_uppiano@verizon.= net wrote: =20 You're in denial, then.=20 =20 I'm not, as FM is still transmitted in analog - not digital - form, eve= n if the wave is the result of digital manipulations. Sampling can be done in the analog as well as digital domains. Your=20 problem is that you cannot visualize analog sampling. You can't properly state that 38 KHz is the sampling frequency -=20 Why not? You are sequentially "looking at" left and right channels. and what about SCA side channels at 76 KHz, BTW ? =20 "Frequency modulation (FM): Modulation in which the instantaneous frequ= ency of a sine wave carrier is caused to depart from the center frequency by= an amount proportional to the instantaneous value of the modulating signal= =2E=20 Non-sequitors. Go get a textbook on communications, and look at the mathematical=20 descriptions of the FM multiplex signal. =20 Note 1: In FM, the carrier frequency is called the center frequency. " =20 The sampling frequency has nothing to do with the final carrier frequency= =2E Federal Standard 1037C =20 |
#104
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DSD Recording Good. PCM recordings bad?
Fran=E7ois Yves Le Gal wrote:
On Tue, 03 Feb 2004 16:53:36 GMT, "Karl Uppiano" karl_uppiano@verizon.= net wrote: =20 You're in denial, then.=20 =20 I'm not, as FM is still transmitted in analog - not digital - form, eve= n if the wave is the result of digital manipulations. Sampling can be done in the analog as well as digital domains. Your=20 problem is that you cannot visualize analog sampling. You can't properly state that 38 KHz is the sampling frequency -=20 Why not? You are sequentially "looking at" left and right channels. and what about SCA side channels at 76 KHz, BTW ? =20 "Frequency modulation (FM): Modulation in which the instantaneous frequ= ency of a sine wave carrier is caused to depart from the center frequency by= an amount proportional to the instantaneous value of the modulating signal= =2E=20 Non-sequitors. Go get a textbook on communications, and look at the mathematical=20 descriptions of the FM multiplex signal. =20 Note 1: In FM, the carrier frequency is called the center frequency. " =20 The sampling frequency has nothing to do with the final carrier frequency= =2E Federal Standard 1037C =20 |
#105
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DSD Recording Good. PCM recordings bad?
Fran=E7ois Yves Le Gal wrote:
On Tue, 03 Feb 2004 16:53:36 GMT, "Karl Uppiano" karl_uppiano@verizon.= net wrote: =20 You're in denial, then.=20 =20 I'm not, as FM is still transmitted in analog - not digital - form, eve= n if the wave is the result of digital manipulations. Sampling can be done in the analog as well as digital domains. Your=20 problem is that you cannot visualize analog sampling. You can't properly state that 38 KHz is the sampling frequency -=20 Why not? You are sequentially "looking at" left and right channels. and what about SCA side channels at 76 KHz, BTW ? =20 "Frequency modulation (FM): Modulation in which the instantaneous frequ= ency of a sine wave carrier is caused to depart from the center frequency by= an amount proportional to the instantaneous value of the modulating signal= =2E=20 Non-sequitors. Go get a textbook on communications, and look at the mathematical=20 descriptions of the FM multiplex signal. =20 Note 1: In FM, the carrier frequency is called the center frequency. " =20 The sampling frequency has nothing to do with the final carrier frequency= =2E Federal Standard 1037C =20 |
#106
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DSD Recording Good. PCM recordings bad?
"François Yves Le Gal" wrote in message
On Tue, 03 Feb 2004 16:53:36 GMT, "Karl Uppiano" wrote: You're in denial, then. I'm not, as FM is still transmitted in analog - not digital - form, even if the wave is the result of digital manipulations. You can't properly state that 38 KHz is the sampling frequency Sure we can. - and what about SCA side channels at 76 KHz, BTW ? Irrelevant. "Frequency modulation (FM): Modulation in which the instantaneous frequency of a sine wave carrier is caused to depart from the center frequency by an amount proportional to the instantaneous value of the modulating signal. Note 1: In FM, the carrier frequency is called the center frequency. " Federal Standard 1037C So what? |
#107
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DSD Recording Good. PCM recordings bad?
"François Yves Le Gal" wrote in message
On Tue, 03 Feb 2004 16:53:36 GMT, "Karl Uppiano" wrote: You're in denial, then. I'm not, as FM is still transmitted in analog - not digital - form, even if the wave is the result of digital manipulations. You can't properly state that 38 KHz is the sampling frequency Sure we can. - and what about SCA side channels at 76 KHz, BTW ? Irrelevant. "Frequency modulation (FM): Modulation in which the instantaneous frequency of a sine wave carrier is caused to depart from the center frequency by an amount proportional to the instantaneous value of the modulating signal. Note 1: In FM, the carrier frequency is called the center frequency. " Federal Standard 1037C So what? |
#108
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DSD Recording Good. PCM recordings bad?
"François Yves Le Gal" wrote in message
On Tue, 03 Feb 2004 16:53:36 GMT, "Karl Uppiano" wrote: You're in denial, then. I'm not, as FM is still transmitted in analog - not digital - form, even if the wave is the result of digital manipulations. You can't properly state that 38 KHz is the sampling frequency Sure we can. - and what about SCA side channels at 76 KHz, BTW ? Irrelevant. "Frequency modulation (FM): Modulation in which the instantaneous frequency of a sine wave carrier is caused to depart from the center frequency by an amount proportional to the instantaneous value of the modulating signal. Note 1: In FM, the carrier frequency is called the center frequency. " Federal Standard 1037C So what? |
#109
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DSD Recording Good. PCM recordings bad?
"François Yves Le Gal" wrote in message
On Tue, 03 Feb 2004 16:53:36 GMT, "Karl Uppiano" wrote: You're in denial, then. I'm not, as FM is still transmitted in analog - not digital - form, even if the wave is the result of digital manipulations. You can't properly state that 38 KHz is the sampling frequency Sure we can. - and what about SCA side channels at 76 KHz, BTW ? Irrelevant. "Frequency modulation (FM): Modulation in which the instantaneous frequency of a sine wave carrier is caused to depart from the center frequency by an amount proportional to the instantaneous value of the modulating signal. Note 1: In FM, the carrier frequency is called the center frequency. " Federal Standard 1037C So what? |
#110
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DSD Recording Good. PCM recordings bad?
chung wrote in message news:936b4$401fe986$
1% capacitors are very expensive. Maybe Sony can get them cheaper, but SACD is not that huge a market, yet. And a 98.8% yield for a filter is absolutely unacceptable for companies like Sony. (It's way too low.) I think you have to get at least a 6-sigma yield for a small circuit like a filter. That's why digital filtering is so wonderful. That explains why SACD manufacturers allow ultrasonic noise. Analog active filters are a little too expensive. |
#111
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DSD Recording Good. PCM recordings bad?
chung wrote in message news:936b4$401fe986$
1% capacitors are very expensive. Maybe Sony can get them cheaper, but SACD is not that huge a market, yet. And a 98.8% yield for a filter is absolutely unacceptable for companies like Sony. (It's way too low.) I think you have to get at least a 6-sigma yield for a small circuit like a filter. That's why digital filtering is so wonderful. That explains why SACD manufacturers allow ultrasonic noise. Analog active filters are a little too expensive. |
#112
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DSD Recording Good. PCM recordings bad?
chung wrote in message news:936b4$401fe986$
1% capacitors are very expensive. Maybe Sony can get them cheaper, but SACD is not that huge a market, yet. And a 98.8% yield for a filter is absolutely unacceptable for companies like Sony. (It's way too low.) I think you have to get at least a 6-sigma yield for a small circuit like a filter. That's why digital filtering is so wonderful. That explains why SACD manufacturers allow ultrasonic noise. Analog active filters are a little too expensive. |
#113
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DSD Recording Good. PCM recordings bad?
Robert Stanton wrote:
chung wrote in message news:936b4$401fe986$ 1% capacitors are very expensive. Maybe Sony can get them cheaper, but SACD is not that huge a market, yet. And a 98.8% yield for a filter is absolutely unacceptable for companies like Sony. (It's way too low.) I think you have to get at least a 6-sigma yield for a small circuit like a filter. That's why digital filtering is so wonderful. That explains why SACD manufacturers allow ultrasonic noise. Analog active filters are a little too expensive. It's also non-trivial to design an active filter that preserves the 120dB linearity that is specified in these hi-rez formats. |
#114
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DSD Recording Good. PCM recordings bad?
Robert Stanton wrote:
chung wrote in message news:936b4$401fe986$ 1% capacitors are very expensive. Maybe Sony can get them cheaper, but SACD is not that huge a market, yet. And a 98.8% yield for a filter is absolutely unacceptable for companies like Sony. (It's way too low.) I think you have to get at least a 6-sigma yield for a small circuit like a filter. That's why digital filtering is so wonderful. That explains why SACD manufacturers allow ultrasonic noise. Analog active filters are a little too expensive. It's also non-trivial to design an active filter that preserves the 120dB linearity that is specified in these hi-rez formats. |
#115
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DSD Recording Good. PCM recordings bad?
Robert Stanton wrote:
chung wrote in message news:936b4$401fe986$ 1% capacitors are very expensive. Maybe Sony can get them cheaper, but SACD is not that huge a market, yet. And a 98.8% yield for a filter is absolutely unacceptable for companies like Sony. (It's way too low.) I think you have to get at least a 6-sigma yield for a small circuit like a filter. That's why digital filtering is so wonderful. That explains why SACD manufacturers allow ultrasonic noise. Analog active filters are a little too expensive. It's also non-trivial to design an active filter that preserves the 120dB linearity that is specified in these hi-rez formats. |
#116
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DSD Recording Good. PCM recordings bad?
"Karl Uppiano" writes:
"Randy Yates" wrote in message ... "Karl Uppiano" writes: "Arny Krueger" wrote in message ... "Karl Uppiano" wrote in message "François Yves Le Gal" wrote in message ... On Sun, 01 Feb 2004 12:53:18 -0800, chung wrote: One, DSD/SACD proponents claim the much wider bandwidth over CD's, and filtering will reduce significantly that claimed advantage. Having a gentle low pass filter at 60 or 100 KHz doesn't significantly reduce SACD's bandwith. You get more than PCM 96 or 192 in both cases! I would start a 3 dB/octave rolloff at 20kHz or so. Ironically, most if not all people can't hear the difference a brickwall filter at 16 KHz makes, if the filter is well-designed. Don't believe me? Listen for yourself at http://www.pcabx.com/technical/low_pass/index.htm . I believe you. I realize it isn't considered state of the art anymore, but I wonder how many people realize that FM stereo uses a 38 kHz sample rate? Actually, the L-R signal is AM-DSB-SC-modulated on a 38 kHz subcarrier and a 19 kHz pilot tone added. This, along with the L+R baseband signal, is then FM modulated onto the carrier. The balanced modulator is mathematically equivalent to a sampled system in which the left and right channels are alternately sampled at a 38 kHz rate and used to directly modulate the carrier. Yeah, I think I see that. I did a simple initial system analysis and it looks like it may indeed work out that way. There will be harmonics at n*38 kHz, too, though, but I guess you just filter those out? Thanks for the new perspective. -- % Randy Yates % "Rollin' and riding and slippin' and %% Fuquay-Varina, NC % sliding, it's magic." %%% 919-577-9882 % %%%% % 'Living' Thing', *A New World Record*, ELO http://home.earthlink.net/~yatescr |
#117
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DSD Recording Good. PCM recordings bad?
"Karl Uppiano" writes:
"Randy Yates" wrote in message ... "Karl Uppiano" writes: "Arny Krueger" wrote in message ... "Karl Uppiano" wrote in message "François Yves Le Gal" wrote in message ... On Sun, 01 Feb 2004 12:53:18 -0800, chung wrote: One, DSD/SACD proponents claim the much wider bandwidth over CD's, and filtering will reduce significantly that claimed advantage. Having a gentle low pass filter at 60 or 100 KHz doesn't significantly reduce SACD's bandwith. You get more than PCM 96 or 192 in both cases! I would start a 3 dB/octave rolloff at 20kHz or so. Ironically, most if not all people can't hear the difference a brickwall filter at 16 KHz makes, if the filter is well-designed. Don't believe me? Listen for yourself at http://www.pcabx.com/technical/low_pass/index.htm . I believe you. I realize it isn't considered state of the art anymore, but I wonder how many people realize that FM stereo uses a 38 kHz sample rate? Actually, the L-R signal is AM-DSB-SC-modulated on a 38 kHz subcarrier and a 19 kHz pilot tone added. This, along with the L+R baseband signal, is then FM modulated onto the carrier. The balanced modulator is mathematically equivalent to a sampled system in which the left and right channels are alternately sampled at a 38 kHz rate and used to directly modulate the carrier. Yeah, I think I see that. I did a simple initial system analysis and it looks like it may indeed work out that way. There will be harmonics at n*38 kHz, too, though, but I guess you just filter those out? Thanks for the new perspective. -- % Randy Yates % "Rollin' and riding and slippin' and %% Fuquay-Varina, NC % sliding, it's magic." %%% 919-577-9882 % %%%% % 'Living' Thing', *A New World Record*, ELO http://home.earthlink.net/~yatescr |
#118
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DSD Recording Good. PCM recordings bad?
"Karl Uppiano" writes:
"Randy Yates" wrote in message ... "Karl Uppiano" writes: "Arny Krueger" wrote in message ... "Karl Uppiano" wrote in message "François Yves Le Gal" wrote in message ... On Sun, 01 Feb 2004 12:53:18 -0800, chung wrote: One, DSD/SACD proponents claim the much wider bandwidth over CD's, and filtering will reduce significantly that claimed advantage. Having a gentle low pass filter at 60 or 100 KHz doesn't significantly reduce SACD's bandwith. You get more than PCM 96 or 192 in both cases! I would start a 3 dB/octave rolloff at 20kHz or so. Ironically, most if not all people can't hear the difference a brickwall filter at 16 KHz makes, if the filter is well-designed. Don't believe me? Listen for yourself at http://www.pcabx.com/technical/low_pass/index.htm . I believe you. I realize it isn't considered state of the art anymore, but I wonder how many people realize that FM stereo uses a 38 kHz sample rate? Actually, the L-R signal is AM-DSB-SC-modulated on a 38 kHz subcarrier and a 19 kHz pilot tone added. This, along with the L+R baseband signal, is then FM modulated onto the carrier. The balanced modulator is mathematically equivalent to a sampled system in which the left and right channels are alternately sampled at a 38 kHz rate and used to directly modulate the carrier. Yeah, I think I see that. I did a simple initial system analysis and it looks like it may indeed work out that way. There will be harmonics at n*38 kHz, too, though, but I guess you just filter those out? Thanks for the new perspective. -- % Randy Yates % "Rollin' and riding and slippin' and %% Fuquay-Varina, NC % sliding, it's magic." %%% 919-577-9882 % %%%% % 'Living' Thing', *A New World Record*, ELO http://home.earthlink.net/~yatescr |
#119
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DSD Recording Good. PCM recordings bad?
"François Yves Le Gal" wrote in message ... On Tue, 03 Feb 2004 16:53:36 GMT, "Karl Uppiano" wrote: You're in denial, then. I'm not, as FM is still transmitted in analog - not digital - form, even if the wave is the result of digital manipulations. You can't properly state that 38 KHz is the sampling frequency - and what about SCA side channels at 76 KHz, BTW ? The application of sampling theorems is not limited to digital applications. There are lots of ways to multiplex analog signals, and periodic sampling is one of them. It's called time-division multiplexing. 38 kHz is the sampling frequency for the FM stereo service. SCA is a different service completely unrelated to FM stereo. Please refer to http://transmitters.tripod.com/stereo.htm for more information. It isn't the best writeup I've seen, but it's the only one I could find on short notice. "Frequency modulation (FM): Modulation in which the instantaneous frequency of a sine wave carrier is caused to depart from the center frequency by an amount proportional to the instantaneous value of the modulating signal. The modulating signal... which is provided by a stereo generator that samples the left and right audio channels at 38 kHz, summed with a 19 kHz pilot, and optionally, other subcarrier services. Note 1: In FM, the carrier frequency is called the center frequency. " Federal Standard 1037C |
#120
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DSD Recording Good. PCM recordings bad?
"François Yves Le Gal" wrote in message ... On Tue, 03 Feb 2004 16:53:36 GMT, "Karl Uppiano" wrote: You're in denial, then. I'm not, as FM is still transmitted in analog - not digital - form, even if the wave is the result of digital manipulations. You can't properly state that 38 KHz is the sampling frequency - and what about SCA side channels at 76 KHz, BTW ? The application of sampling theorems is not limited to digital applications. There are lots of ways to multiplex analog signals, and periodic sampling is one of them. It's called time-division multiplexing. 38 kHz is the sampling frequency for the FM stereo service. SCA is a different service completely unrelated to FM stereo. Please refer to http://transmitters.tripod.com/stereo.htm for more information. It isn't the best writeup I've seen, but it's the only one I could find on short notice. "Frequency modulation (FM): Modulation in which the instantaneous frequency of a sine wave carrier is caused to depart from the center frequency by an amount proportional to the instantaneous value of the modulating signal. The modulating signal... which is provided by a stereo generator that samples the left and right audio channels at 38 kHz, summed with a 19 kHz pilot, and optionally, other subcarrier services. Note 1: In FM, the carrier frequency is called the center frequency. " Federal Standard 1037C |
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