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#41
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DSD Recording Good. PCM recordings bad?
"Karl Uppiano" writes:
[...] A well-designed PCM system (e.g., CD-Audio) uses phase-linear, oversampling digital FIR filters to allow the use of very gentle, phase-linear analog filtering. I think the original Sony/Philips engineers that knew what they were doing when they designed the CD-Audio spec must have retired without passing their expertise on to the next generation. Right-on, brother. I've been watching (sometimes not so passively) this "DVD-A/SACD" blindside on the consumers for a couple of years now and my conclusion, as one whose career is in digital signal processing, is that it's a complete and utter farce. In the end, consumers will have emptied out their pocketbooks for new players and, more importantly, new media which are indistinguishable in sound quality from CD audio. The only possible rational justification for a new format is the inclusion of multiple ( 2) tracks. The sound quality arguments are empty. -- % Randy Yates % "Though you ride on the wheels of tomorrow, %% Fuquay-Varina, NC % you still wander the fields of your %%% 919-577-9882 % sorrow." %%%% % '21st Century Man', *Time*, ELO http://home.earthlink.net/~yatescr |
#42
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DSD Recording Good. PCM recordings bad?
"Randy Yates" wrote in message
"Karl Uppiano" writes: [...] A well-designed PCM system (e.g., CD-Audio) uses phase-linear, oversampling digital FIR filters to allow the use of very gentle, phase-linear analog filtering. I think the original Sony/Philips engineers that knew what they were doing when they designed the CD-Audio spec must have retired without passing their expertise on to the next generation. Right-on, brother. I've been watching (sometimes not so passively) this "DVD-A/SACD" blindside on the consumers for a couple of years now and my conclusion, as one whose career is in digital signal processing, is that it's a complete and utter farce. In the end, consumers will have emptied out their pocketbooks for new players and, more importantly, new media which are indistinguishable in sound quality from CD audio. The only possible rational justification for a new format is the inclusion of multiple ( 2) tracks. The sound quality arguments are empty. Looks to me like a failed marketing effort. The drive behind attempts to obsolete the CD Audio format have to be money. Once the Chinese started producing CD players with what most consumers found to be acceptable sound quality, the Japanese were largely cut out of the picture unless they had some unique technology to sell. SACD and DVD-A were that technology, but insufficient product was sold before the new-technology players were commoditized. With something like 600,000 DVD-A discs and perhaps as many as a few times that in SACD discs sold last year, it's quite clear that there just aren't a lot of players out there that consumers are trying to "feed". With SACD-DAV-A players selling for under $300, this is hardly a true high end, nice market play. It's a failed attempt to sell consumers a line of 'bigger numbers always sound better" BS. Now, we're faced with retro-tech flacks like François who push vinyl out of one side of their mouths, and high sample rates out of the other. Anybody who has a memory long enough to remember the last time they talked out of both sides of their mouth will not grant them sufficient credibility to actually sell product. |
#43
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DSD Recording Good. PCM recordings bad?
"Randy Yates" wrote in message
"Karl Uppiano" writes: [...] A well-designed PCM system (e.g., CD-Audio) uses phase-linear, oversampling digital FIR filters to allow the use of very gentle, phase-linear analog filtering. I think the original Sony/Philips engineers that knew what they were doing when they designed the CD-Audio spec must have retired without passing their expertise on to the next generation. Right-on, brother. I've been watching (sometimes not so passively) this "DVD-A/SACD" blindside on the consumers for a couple of years now and my conclusion, as one whose career is in digital signal processing, is that it's a complete and utter farce. In the end, consumers will have emptied out their pocketbooks for new players and, more importantly, new media which are indistinguishable in sound quality from CD audio. The only possible rational justification for a new format is the inclusion of multiple ( 2) tracks. The sound quality arguments are empty. Looks to me like a failed marketing effort. The drive behind attempts to obsolete the CD Audio format have to be money. Once the Chinese started producing CD players with what most consumers found to be acceptable sound quality, the Japanese were largely cut out of the picture unless they had some unique technology to sell. SACD and DVD-A were that technology, but insufficient product was sold before the new-technology players were commoditized. With something like 600,000 DVD-A discs and perhaps as many as a few times that in SACD discs sold last year, it's quite clear that there just aren't a lot of players out there that consumers are trying to "feed". With SACD-DAV-A players selling for under $300, this is hardly a true high end, nice market play. It's a failed attempt to sell consumers a line of 'bigger numbers always sound better" BS. Now, we're faced with retro-tech flacks like François who push vinyl out of one side of their mouths, and high sample rates out of the other. Anybody who has a memory long enough to remember the last time they talked out of both sides of their mouth will not grant them sufficient credibility to actually sell product. |
#44
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DSD Recording Good. PCM recordings bad?
"Randy Yates" wrote in message
"Karl Uppiano" writes: [...] A well-designed PCM system (e.g., CD-Audio) uses phase-linear, oversampling digital FIR filters to allow the use of very gentle, phase-linear analog filtering. I think the original Sony/Philips engineers that knew what they were doing when they designed the CD-Audio spec must have retired without passing their expertise on to the next generation. Right-on, brother. I've been watching (sometimes not so passively) this "DVD-A/SACD" blindside on the consumers for a couple of years now and my conclusion, as one whose career is in digital signal processing, is that it's a complete and utter farce. In the end, consumers will have emptied out their pocketbooks for new players and, more importantly, new media which are indistinguishable in sound quality from CD audio. The only possible rational justification for a new format is the inclusion of multiple ( 2) tracks. The sound quality arguments are empty. Looks to me like a failed marketing effort. The drive behind attempts to obsolete the CD Audio format have to be money. Once the Chinese started producing CD players with what most consumers found to be acceptable sound quality, the Japanese were largely cut out of the picture unless they had some unique technology to sell. SACD and DVD-A were that technology, but insufficient product was sold before the new-technology players were commoditized. With something like 600,000 DVD-A discs and perhaps as many as a few times that in SACD discs sold last year, it's quite clear that there just aren't a lot of players out there that consumers are trying to "feed". With SACD-DAV-A players selling for under $300, this is hardly a true high end, nice market play. It's a failed attempt to sell consumers a line of 'bigger numbers always sound better" BS. Now, we're faced with retro-tech flacks like François who push vinyl out of one side of their mouths, and high sample rates out of the other. Anybody who has a memory long enough to remember the last time they talked out of both sides of their mouth will not grant them sufficient credibility to actually sell product. |
#45
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DSD Recording Good. PCM recordings bad?
"Randy Yates" wrote in message
"Karl Uppiano" writes: [...] A well-designed PCM system (e.g., CD-Audio) uses phase-linear, oversampling digital FIR filters to allow the use of very gentle, phase-linear analog filtering. I think the original Sony/Philips engineers that knew what they were doing when they designed the CD-Audio spec must have retired without passing their expertise on to the next generation. Right-on, brother. I've been watching (sometimes not so passively) this "DVD-A/SACD" blindside on the consumers for a couple of years now and my conclusion, as one whose career is in digital signal processing, is that it's a complete and utter farce. In the end, consumers will have emptied out their pocketbooks for new players and, more importantly, new media which are indistinguishable in sound quality from CD audio. The only possible rational justification for a new format is the inclusion of multiple ( 2) tracks. The sound quality arguments are empty. Looks to me like a failed marketing effort. The drive behind attempts to obsolete the CD Audio format have to be money. Once the Chinese started producing CD players with what most consumers found to be acceptable sound quality, the Japanese were largely cut out of the picture unless they had some unique technology to sell. SACD and DVD-A were that technology, but insufficient product was sold before the new-technology players were commoditized. With something like 600,000 DVD-A discs and perhaps as many as a few times that in SACD discs sold last year, it's quite clear that there just aren't a lot of players out there that consumers are trying to "feed". With SACD-DAV-A players selling for under $300, this is hardly a true high end, nice market play. It's a failed attempt to sell consumers a line of 'bigger numbers always sound better" BS. Now, we're faced with retro-tech flacks like François who push vinyl out of one side of their mouths, and high sample rates out of the other. Anybody who has a memory long enough to remember the last time they talked out of both sides of their mouth will not grant them sufficient credibility to actually sell product. |
#46
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DSD Recording Good. PCM recordings bad?
"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? |
#47
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DSD Recording Good. PCM recordings bad?
"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? |
#48
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DSD Recording Good. PCM recordings bad?
"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? |
#49
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DSD Recording Good. PCM recordings bad?
"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? |
#50
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DSD Recording Good. PCM recordings bad?
"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 manfactures 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. I took a standard Chebychev 0.1 dB ripple, 5th order, active filter from a text book, and scaled it to 20kHz, 2000 Ohms, using two SE3353 op-amps. Here are the simulation results: Freq. Gain Group Delay Hz dB usec 1000 -0.09 20.9 5000 -0.15 20.7 10000 -0.18 21.3 20000 -0.21 25.6 30000 -3.01 43.5 40000 -19.7 10.4 50000 -31.7 4.7 60000 -40.7 2.8 70000 -48.1 80000 -54.4 90000 -60.0 The filter is flat from 20 to 20kHz, and the group delay is almost flat from 20 to 20kHz. Bob Stanton |
#51
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DSD Recording Good. PCM recordings bad?
"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 manfactures 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. I took a standard Chebychev 0.1 dB ripple, 5th order, active filter from a text book, and scaled it to 20kHz, 2000 Ohms, using two SE3353 op-amps. Here are the simulation results: Freq. Gain Group Delay Hz dB usec 1000 -0.09 20.9 5000 -0.15 20.7 10000 -0.18 21.3 20000 -0.21 25.6 30000 -3.01 43.5 40000 -19.7 10.4 50000 -31.7 4.7 60000 -40.7 2.8 70000 -48.1 80000 -54.4 90000 -60.0 The filter is flat from 20 to 20kHz, and the group delay is almost flat from 20 to 20kHz. Bob Stanton |
#52
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DSD Recording Good. PCM recordings bad?
"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 manfactures 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. I took a standard Chebychev 0.1 dB ripple, 5th order, active filter from a text book, and scaled it to 20kHz, 2000 Ohms, using two SE3353 op-amps. Here are the simulation results: Freq. Gain Group Delay Hz dB usec 1000 -0.09 20.9 5000 -0.15 20.7 10000 -0.18 21.3 20000 -0.21 25.6 30000 -3.01 43.5 40000 -19.7 10.4 50000 -31.7 4.7 60000 -40.7 2.8 70000 -48.1 80000 -54.4 90000 -60.0 The filter is flat from 20 to 20kHz, and the group delay is almost flat from 20 to 20kHz. Bob Stanton |
#53
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DSD Recording Good. PCM recordings bad?
"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 manfactures 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. I took a standard Chebychev 0.1 dB ripple, 5th order, active filter from a text book, and scaled it to 20kHz, 2000 Ohms, using two SE3353 op-amps. Here are the simulation results: Freq. Gain Group Delay Hz dB usec 1000 -0.09 20.9 5000 -0.15 20.7 10000 -0.18 21.3 20000 -0.21 25.6 30000 -3.01 43.5 40000 -19.7 10.4 50000 -31.7 4.7 60000 -40.7 2.8 70000 -48.1 80000 -54.4 90000 -60.0 The filter is flat from 20 to 20kHz, and the group delay is almost flat from 20 to 20kHz. Bob Stanton |
#54
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DSD Recording Good. PCM recordings bad?
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 manfactures 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. I took a standard Chebychev 0.1 dB ripple, 5th order, active filter from a text book, and scaled it to 20kHz, 2000 Ohms, using two SE3353 op-amps. Here are the simulation results: Freq. Gain Group Delay Hz dB usec 1000 -0.09 20.9 5000 -0.15 20.7 10000 -0.18 21.3 20000 -0.21 25.6 30000 -3.01 43.5 40000 -19.7 10.4 50000 -31.7 4.7 60000 -40.7 2.8 70000 -48.1 80000 -54.4 90000 -60.0 The filter is flat from 20 to 20kHz, and the group delay is almost flat from 20 to 20kHz. 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 . Bob Stanton |
#55
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DSD Recording Good. PCM recordings bad?
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 manfactures 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. I took a standard Chebychev 0.1 dB ripple, 5th order, active filter from a text book, and scaled it to 20kHz, 2000 Ohms, using two SE3353 op-amps. Here are the simulation results: Freq. Gain Group Delay Hz dB usec 1000 -0.09 20.9 5000 -0.15 20.7 10000 -0.18 21.3 20000 -0.21 25.6 30000 -3.01 43.5 40000 -19.7 10.4 50000 -31.7 4.7 60000 -40.7 2.8 70000 -48.1 80000 -54.4 90000 -60.0 The filter is flat from 20 to 20kHz, and the group delay is almost flat from 20 to 20kHz. 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 . Bob Stanton |
#56
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DSD Recording Good. PCM recordings bad?
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 manfactures 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. I took a standard Chebychev 0.1 dB ripple, 5th order, active filter from a text book, and scaled it to 20kHz, 2000 Ohms, using two SE3353 op-amps. Here are the simulation results: Freq. Gain Group Delay Hz dB usec 1000 -0.09 20.9 5000 -0.15 20.7 10000 -0.18 21.3 20000 -0.21 25.6 30000 -3.01 43.5 40000 -19.7 10.4 50000 -31.7 4.7 60000 -40.7 2.8 70000 -48.1 80000 -54.4 90000 -60.0 The filter is flat from 20 to 20kHz, and the group delay is almost flat from 20 to 20kHz. 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 . Bob Stanton |
#57
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DSD Recording Good. PCM recordings bad?
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 manfactures 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. I took a standard Chebychev 0.1 dB ripple, 5th order, active filter from a text book, and scaled it to 20kHz, 2000 Ohms, using two SE3353 op-amps. Here are the simulation results: Freq. Gain Group Delay Hz dB usec 1000 -0.09 20.9 5000 -0.15 20.7 10000 -0.18 21.3 20000 -0.21 25.6 30000 -3.01 43.5 40000 -19.7 10.4 50000 -31.7 4.7 60000 -40.7 2.8 70000 -48.1 80000 -54.4 90000 -60.0 The filter is flat from 20 to 20kHz, and the group delay is almost flat from 20 to 20kHz. 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 . Bob Stanton |
#58
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DSD Recording Good. PCM recordings bad?
"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. -- % Randy Yates % "And all that I can do %% Fuquay-Varina, NC % is say I'm sorry, %%% 919-577-9882 % that's the way it goes..." %%%% % Getting To The Point', *Balance of Power*, ELO http://home.earthlink.net/~yatescr |
#59
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DSD Recording Good. PCM recordings bad?
"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. -- % Randy Yates % "And all that I can do %% Fuquay-Varina, NC % is say I'm sorry, %%% 919-577-9882 % that's the way it goes..." %%%% % Getting To The Point', *Balance of Power*, ELO http://home.earthlink.net/~yatescr |
#60
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DSD Recording Good. PCM recordings bad?
"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. -- % Randy Yates % "And all that I can do %% Fuquay-Varina, NC % is say I'm sorry, %%% 919-577-9882 % that's the way it goes..." %%%% % Getting To The Point', *Balance of Power*, ELO http://home.earthlink.net/~yatescr |
#61
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DSD Recording Good. PCM recordings bad?
"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. -- % Randy Yates % "And all that I can do %% Fuquay-Varina, NC % is say I'm sorry, %%% 919-577-9882 % that's the way it goes..." %%%% % Getting To The Point', *Balance of Power*, ELO http://home.earthlink.net/~yatescr |
#62
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DSD Recording Good. PCM recordings bad?
chung writes:
François Yves Le Gal wrote: On Sun, 01 Feb 2004 14:42:29 -0800, chung wrote: That does very little to the noise between 20KHz and 50KHz. The noise is at much lower levels in these frequency bands. The noise starts rising from 10KHz, and is significant between 20KHz and 50KHz. Check out some player measurements: http://www.stereophile.com/digitalso...06/index5.html Noise is 30 dB higher than 24 bit LPCM at 20 KHz. Oh piddle - that means the noise is only 114 dB below full-scale at 20 kHz. You know, we just might hear that... -- % Randy Yates % "Remember the good old 1980's, when %% Fuquay-Varina, NC % things were so uncomplicated?" %%% 919-577-9882 % 'Ticket To The Moon' %%%% % *Time*, Electric Light Orchestra http://home.earthlink.net/~yatescr |
#63
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DSD Recording Good. PCM recordings bad?
chung writes:
François Yves Le Gal wrote: On Sun, 01 Feb 2004 14:42:29 -0800, chung wrote: That does very little to the noise between 20KHz and 50KHz. The noise is at much lower levels in these frequency bands. The noise starts rising from 10KHz, and is significant between 20KHz and 50KHz. Check out some player measurements: http://www.stereophile.com/digitalso...06/index5.html Noise is 30 dB higher than 24 bit LPCM at 20 KHz. Oh piddle - that means the noise is only 114 dB below full-scale at 20 kHz. You know, we just might hear that... -- % Randy Yates % "Remember the good old 1980's, when %% Fuquay-Varina, NC % things were so uncomplicated?" %%% 919-577-9882 % 'Ticket To The Moon' %%%% % *Time*, Electric Light Orchestra http://home.earthlink.net/~yatescr |
#64
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DSD Recording Good. PCM recordings bad?
chung writes:
François Yves Le Gal wrote: On Sun, 01 Feb 2004 14:42:29 -0800, chung wrote: That does very little to the noise between 20KHz and 50KHz. The noise is at much lower levels in these frequency bands. The noise starts rising from 10KHz, and is significant between 20KHz and 50KHz. Check out some player measurements: http://www.stereophile.com/digitalso...06/index5.html Noise is 30 dB higher than 24 bit LPCM at 20 KHz. Oh piddle - that means the noise is only 114 dB below full-scale at 20 kHz. You know, we just might hear that... -- % Randy Yates % "Remember the good old 1980's, when %% Fuquay-Varina, NC % things were so uncomplicated?" %%% 919-577-9882 % 'Ticket To The Moon' %%%% % *Time*, Electric Light Orchestra http://home.earthlink.net/~yatescr |
#65
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DSD Recording Good. PCM recordings bad?
chung writes:
François Yves Le Gal wrote: On Sun, 01 Feb 2004 14:42:29 -0800, chung wrote: That does very little to the noise between 20KHz and 50KHz. The noise is at much lower levels in these frequency bands. The noise starts rising from 10KHz, and is significant between 20KHz and 50KHz. Check out some player measurements: http://www.stereophile.com/digitalso...06/index5.html Noise is 30 dB higher than 24 bit LPCM at 20 KHz. Oh piddle - that means the noise is only 114 dB below full-scale at 20 kHz. You know, we just might hear that... -- % Randy Yates % "Remember the good old 1980's, when %% Fuquay-Varina, NC % things were so uncomplicated?" %%% 919-577-9882 % 'Ticket To The Moon' %%%% % *Time*, Electric Light Orchestra http://home.earthlink.net/~yatescr |
#66
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DSD Recording Good. PCM recordings bad?
Randy Yates wrote:
chung writes: =20 Fran=E7ois Yves Le Gal wrote: On Sun, 01 Feb 2004 14:42:29 -0800, chung wrote:= That does very little to the noise between 20KHz and 50KHz. The noise is at much lower levels in these frequency bands. The noise starts rising from 10KHz, and is significant between 20KHz and 50KHz. Check out some player measurements: http://www.stereophile.com/digitalso...06/index5.html Noise is 30 dB higher than 24 bit LPCM at 20 KHz. =20 Oh piddle - that means the noise is only 114 dB below full-scale at 20 kHz. You know, we just might hear that... Actually the noise in a 1/3 octave BW centered at 20KHz is about 87 dB=20 below FS, according to that plot. Not that it is necessarily audible,=20 but the possibility exists that the sharply rising noise could cause=20 problems with some amps. |
#67
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DSD Recording Good. PCM recordings bad?
Randy Yates wrote:
chung writes: =20 Fran=E7ois Yves Le Gal wrote: On Sun, 01 Feb 2004 14:42:29 -0800, chung wrote:= That does very little to the noise between 20KHz and 50KHz. The noise is at much lower levels in these frequency bands. The noise starts rising from 10KHz, and is significant between 20KHz and 50KHz. Check out some player measurements: http://www.stereophile.com/digitalso...06/index5.html Noise is 30 dB higher than 24 bit LPCM at 20 KHz. =20 Oh piddle - that means the noise is only 114 dB below full-scale at 20 kHz. You know, we just might hear that... Actually the noise in a 1/3 octave BW centered at 20KHz is about 87 dB=20 below FS, according to that plot. Not that it is necessarily audible,=20 but the possibility exists that the sharply rising noise could cause=20 problems with some amps. |
#68
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DSD Recording Good. PCM recordings bad?
Randy Yates wrote:
chung writes: =20 Fran=E7ois Yves Le Gal wrote: On Sun, 01 Feb 2004 14:42:29 -0800, chung wrote:= That does very little to the noise between 20KHz and 50KHz. The noise is at much lower levels in these frequency bands. The noise starts rising from 10KHz, and is significant between 20KHz and 50KHz. Check out some player measurements: http://www.stereophile.com/digitalso...06/index5.html Noise is 30 dB higher than 24 bit LPCM at 20 KHz. =20 Oh piddle - that means the noise is only 114 dB below full-scale at 20 kHz. You know, we just might hear that... Actually the noise in a 1/3 octave BW centered at 20KHz is about 87 dB=20 below FS, according to that plot. Not that it is necessarily audible,=20 but the possibility exists that the sharply rising noise could cause=20 problems with some amps. |
#69
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DSD Recording Good. PCM recordings bad?
Randy Yates wrote:
chung writes: =20 Fran=E7ois Yves Le Gal wrote: On Sun, 01 Feb 2004 14:42:29 -0800, chung wrote:= That does very little to the noise between 20KHz and 50KHz. The noise is at much lower levels in these frequency bands. The noise starts rising from 10KHz, and is significant between 20KHz and 50KHz. Check out some player measurements: http://www.stereophile.com/digitalso...06/index5.html Noise is 30 dB higher than 24 bit LPCM at 20 KHz. =20 Oh piddle - that means the noise is only 114 dB below full-scale at 20 kHz. You know, we just might hear that... Actually the noise in a 1/3 octave BW centered at 20KHz is about 87 dB=20 below FS, according to that plot. Not that it is necessarily audible,=20 but the possibility exists that the sharply rising noise could cause=20 problems with some amps. |
#70
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DSD Recording Good. PCM recordings bad?
"François Yves Le Gal" wrote in message ... On Mon, 02 Feb 2004 16:52:57 GMT, "Karl Uppiano" wrote: 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? Sample rate? FM is 100% analog, no smpale rate, but a carrier frequency modulated in frequency. Nope, FM stereo is a sampled system. It's analog, but it's definitely sampled. 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. |
#71
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DSD Recording Good. PCM recordings bad?
"François Yves Le Gal" wrote in message ... On Mon, 02 Feb 2004 16:52:57 GMT, "Karl Uppiano" wrote: 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? Sample rate? FM is 100% analog, no smpale rate, but a carrier frequency modulated in frequency. Nope, FM stereo is a sampled system. It's analog, but it's definitely sampled. 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. |
#72
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DSD Recording Good. PCM recordings bad?
"François Yves Le Gal" wrote in message ... On Mon, 02 Feb 2004 16:52:57 GMT, "Karl Uppiano" wrote: 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? Sample rate? FM is 100% analog, no smpale rate, but a carrier frequency modulated in frequency. Nope, FM stereo is a sampled system. It's analog, but it's definitely sampled. 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. |
#73
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DSD Recording Good. PCM recordings bad?
"François Yves Le Gal" wrote in message ... On Mon, 02 Feb 2004 16:52:57 GMT, "Karl Uppiano" wrote: 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? Sample rate? FM is 100% analog, no smpale rate, but a carrier frequency modulated in frequency. Nope, FM stereo is a sampled system. It's analog, but it's definitely sampled. 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. |
#74
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DSD Recording Good. PCM recordings bad?
"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. |
#75
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DSD Recording Good. PCM recordings bad?
"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. |
#76
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DSD Recording Good. PCM recordings bad?
"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. |
#77
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DSD Recording Good. PCM recordings bad?
"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. |
#78
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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. |
#79
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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. |
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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. |
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