dorkus:
hm.. i figured the power supply might need some upgrading. I suppose if worse comes to worse, it's not a huge deal to build a new linear power supply for it, but one would probably have to tear apart the stock PSU to get the power switch and LED assemblies. Also, the stock PSU will likely have some power switching features and control features which would need to be duplicated in the replacement...
I suspect that the DC offset's you've seen on the outputs are a result of the cheapo opamps. This, if nothing else, is reason enough for me to upgrade the opamps. At least I can get rid of the awful 'lytics in the signal path (yuk!). Also, it may be an option to use a nice film coupling cap at the input to the opamp if there is still some DC offset from prior circuitry. I'll also have to scope the input to the muting transistors to see if the player uses them during operation... I'm kinda hoping they're used in power up and power down only, which would make their removal irrelevant to me, as I follow a strict power-up / power-down procedure anyway (my preamp is also DC coupled). Whatever happens, I think these are the three key mods that really need to be done for an immediate improvement, which should suffice until I'm using an external DAC.
Regarding output in PCM format, one of my main reasons is that I want to be able to perform signal processing on the data, which is virtually impossible with DSD streams. In fact, I learned something interesting today while reading some online articles... it seems DSD studio equipment actually employs a quasi-PCM format for editing(!)... it would seem that all the marketing hype about maintaining the purity of the data streams by avoiding PCM conversions is just propoganda! It is well known that in order to do any useful studio editing, some form of PCM conversion is almost essential, and of course it turns out that this is pretty much exactly what happens during the process of producing a master. There are only a very small percentage of recordings (maybe 5%) which can be produced without any conversion into PCM. Of course, Sony and Philips would never want the consumers to find out that the recordings they're buying have actually gone through DSD-PCM conversion, and then back again! Personally, I understand the mathematical underpinnings of the conversion processes quite well, and I'm not the least bit afraid of them. A single conversion into PCM isn't likely to do much if any audible damage to the music.
I don't want to get into a DVD-A vs SACD war, but for the purposes of evaluating whether or not it is a worthwhile proposal to convert SACD streams into 24/96 or 24/192 PCM data for processing or the use of an external DAC, it might be useful to do a little channel capacity comparison. So, as a crude start, we can look at the raw bitrate per channel:
2.3 Mbps for 24/96
4.6 Mbps for 24/192
2.8 Mbps for SACD
Of course, this is just the raw bitrate, and doesn't account for whatever redundancies exist in the actual data streams. Both SACD and DVD-A use lossless compression schemes to remove as much redundant information as possible. Given the similarity and sophistication of the compression schemes (both are predictive encoders), I think it is safe to assume that the compressed bitstream should contain fairly close to 100% relevant data (perhaps within 10%?). Thus, it is perhaps more useful to look at the compressed data rates to see just how much actual information is really there. Conveniently, MLP (Meridian Lossless Packing for DVD-A) and SACD predictive run-length encoding both acheive about 2:1 compression, so we can just compare the raw bitrates anyway.
Based on this simplistic analysis, we could conclude that SACD should be capable of providing slightly better resolution than 24/96, but certainly less than 24/192PCM audio. There is, however, one thing which bugs me: the fact that 24/96 audio will only code signal components at or below 48kHz (most likely filtered to remove most stuff above 20 or 30kHz), while SACD's encoded spectral content is less clearly defined... how much of the non-redundant compressed data is useless stuff above 20kHz? SACD claims a bandwidth of 100kHz, and basic sigma-delta and noise-shaping theory tell us that there should be a tremendous amount of noise (and therefore uncompressible data) in the ultrasonic region of the spectrum. Now, here it would be helpful to know more about the SACD prediction algorithms, in order to determine more about how much of the compressed DSD data stream is wasted on irrelevant ultrasonic noise...
Anyway, I have basically concluded that there should be little difference between SACD and 24/96 in terms of absolute resolution capability, and my gut feeling gives the edge to 24/96. Whatever the reality is, I think that converting DSD into 24/96 should be an acceptable compromise in order to gain the benefits of all-digital processing for things like speaker crossovers and so on... the other benefit to me will be the ability to use a single high-quality DAC for all of the various audio sources I will be using (CDDA, MP3 CD-Rs, DTS audio discs, DTS and DD movies, SACD and of course, DVD-A). Oh yeah, and of course, if I am using a separate DAC, I won't need to worry about the noisy power supply
Heh, there goes yet another verbose post... oops. Oh well.
Jim:
I used a BUF-03FJ preamp for many years, and enjoyed it immensely. However, I have recently replaced the BUF-03s with a different circuit which is much better... the circuit I designed is actually based on an OPA134, and uses a high-current SE class-a mosfet output stage inside the opamp feedback loop. The improvement was startling, given how good the BUF-03 is... improved transparency and detail, as well as warmth and smoothness. The noise floor was also noticeably improved (perhaps responsible for some of the apparent increase in detail and resolution), since the BUF-03 is *very* noisy at something like 50nV/rt(Hz)! The replacement circuit is also much better at driving difficult loads - with the high current output stage, it will push 5Vp-p into 10ohms or less, or a 10uF cap without so much as a hiccup! Bandwidth is flat to 5MHz+ (couldn't measure past that at the time I built it). I don't think I'll ever go back to the BUF-03. It was a great device in it's day, but I think it has been surpassed by more modern devices.
hm.. i figured the power supply might need some upgrading. I suppose if worse comes to worse, it's not a huge deal to build a new linear power supply for it, but one would probably have to tear apart the stock PSU to get the power switch and LED assemblies. Also, the stock PSU will likely have some power switching features and control features which would need to be duplicated in the replacement...
I suspect that the DC offset's you've seen on the outputs are a result of the cheapo opamps. This, if nothing else, is reason enough for me to upgrade the opamps. At least I can get rid of the awful 'lytics in the signal path (yuk!). Also, it may be an option to use a nice film coupling cap at the input to the opamp if there is still some DC offset from prior circuitry. I'll also have to scope the input to the muting transistors to see if the player uses them during operation... I'm kinda hoping they're used in power up and power down only, which would make their removal irrelevant to me, as I follow a strict power-up / power-down procedure anyway (my preamp is also DC coupled). Whatever happens, I think these are the three key mods that really need to be done for an immediate improvement, which should suffice until I'm using an external DAC.
Regarding output in PCM format, one of my main reasons is that I want to be able to perform signal processing on the data, which is virtually impossible with DSD streams. In fact, I learned something interesting today while reading some online articles... it seems DSD studio equipment actually employs a quasi-PCM format for editing(!)... it would seem that all the marketing hype about maintaining the purity of the data streams by avoiding PCM conversions is just propoganda! It is well known that in order to do any useful studio editing, some form of PCM conversion is almost essential, and of course it turns out that this is pretty much exactly what happens during the process of producing a master. There are only a very small percentage of recordings (maybe 5%) which can be produced without any conversion into PCM. Of course, Sony and Philips would never want the consumers to find out that the recordings they're buying have actually gone through DSD-PCM conversion, and then back again! Personally, I understand the mathematical underpinnings of the conversion processes quite well, and I'm not the least bit afraid of them. A single conversion into PCM isn't likely to do much if any audible damage to the music.
I don't want to get into a DVD-A vs SACD war, but for the purposes of evaluating whether or not it is a worthwhile proposal to convert SACD streams into 24/96 or 24/192 PCM data for processing or the use of an external DAC, it might be useful to do a little channel capacity comparison. So, as a crude start, we can look at the raw bitrate per channel:
2.3 Mbps for 24/96
4.6 Mbps for 24/192
2.8 Mbps for SACD
Of course, this is just the raw bitrate, and doesn't account for whatever redundancies exist in the actual data streams. Both SACD and DVD-A use lossless compression schemes to remove as much redundant information as possible. Given the similarity and sophistication of the compression schemes (both are predictive encoders), I think it is safe to assume that the compressed bitstream should contain fairly close to 100% relevant data (perhaps within 10%?). Thus, it is perhaps more useful to look at the compressed data rates to see just how much actual information is really there. Conveniently, MLP (Meridian Lossless Packing for DVD-A) and SACD predictive run-length encoding both acheive about 2:1 compression, so we can just compare the raw bitrates anyway.
Based on this simplistic analysis, we could conclude that SACD should be capable of providing slightly better resolution than 24/96, but certainly less than 24/192PCM audio. There is, however, one thing which bugs me: the fact that 24/96 audio will only code signal components at or below 48kHz (most likely filtered to remove most stuff above 20 or 30kHz), while SACD's encoded spectral content is less clearly defined... how much of the non-redundant compressed data is useless stuff above 20kHz? SACD claims a bandwidth of 100kHz, and basic sigma-delta and noise-shaping theory tell us that there should be a tremendous amount of noise (and therefore uncompressible data) in the ultrasonic region of the spectrum. Now, here it would be helpful to know more about the SACD prediction algorithms, in order to determine more about how much of the compressed DSD data stream is wasted on irrelevant ultrasonic noise...
Anyway, I have basically concluded that there should be little difference between SACD and 24/96 in terms of absolute resolution capability, and my gut feeling gives the edge to 24/96. Whatever the reality is, I think that converting DSD into 24/96 should be an acceptable compromise in order to gain the benefits of all-digital processing for things like speaker crossovers and so on... the other benefit to me will be the ability to use a single high-quality DAC for all of the various audio sources I will be using (CDDA, MP3 CD-Rs, DTS audio discs, DTS and DD movies, SACD and of course, DVD-A). Oh yeah, and of course, if I am using a separate DAC, I won't need to worry about the noisy power supply
Heh, there goes yet another verbose post... oops. Oh well.
Jim:
I used a BUF-03FJ preamp for many years, and enjoyed it immensely. However, I have recently replaced the BUF-03s with a different circuit which is much better... the circuit I designed is actually based on an OPA134, and uses a high-current SE class-a mosfet output stage inside the opamp feedback loop. The improvement was startling, given how good the BUF-03 is... improved transparency and detail, as well as warmth and smoothness. The noise floor was also noticeably improved (perhaps responsible for some of the apparent increase in detail and resolution), since the BUF-03 is *very* noisy at something like 50nV/rt(Hz)! The replacement circuit is also much better at driving difficult loads - with the high current output stage, it will push 5Vp-p into 10ohms or less, or a 10uF cap without so much as a hiccup! Bandwidth is flat to 5MHz+ (couldn't measure past that at the time I built it). I don't think I'll ever go back to the BUF-03. It was a great device in it's day, but I think it has been surpassed by more modern devices.
Oh yeah, almost forgot... looks like we may be getting a top-o-the-line LeCroy jitter analyzer at work (woohoo!). The LeCroy sales guys came by and gave an interesting and insightful presentation on jitter. The LeCroy instrument looks like one *very* expensive and sophisticated tool! If I'm lucky, i may get to sneak in some private testing here and there...
yeah, DSD is often converted to PCM for processing, however you do not completely lose the benefits of DSD by doing so. i'm also hoping in a few years, if SACD becomes more standard, that DSD mastering gear will become more advanced and we'll see more pure DSD recordings. i'm not sure how current DSP theory would apply to a detal-sigma modulated signal (an AES paper i've glanced at seemed to imply it was currently impossible), but i'm sure it's something that could happen in a few years.
bitrate is a terrible gauge of resolution... well, ok, it will tell you how much data their is but that is about it. when it comes to different coding schemes though this becomes irrelevant. it is generally accepted that 24/192 resolution is the highest (nothing to do with bitrate though), and that SACD resolution is frequency dependent. to my ears, this means absolutely nothing. it's the fundamental nature of the encoding process that makes SACD superior to me. DSD does not have zero-crossing problems that PCM has, and the manner in which it captures a waveform is more akin to analog. ed meitner has said some interesting things on DSD vs. PCM, and obviously he is biased since he's in the SACD camp but i think his comments still have a lot of merit.
in the end though, the proof is in the listening, and at least for the music i listen to SACD is king. for me there is no mistaking the natural representation of tone and texture SACD provides. even 24/192 PCM can't beat it, and only the best analog can really compete. i think 24/192 does have a small edge in transient response (obvious technical advantage in that aspect), but it does not capture the musical whole with the same naturalness as SACD - it is still PCM and it still sounds digital to me. SACD is not perfect, but it allows me to relax and listen to the music more than anything since vinyl. and from a musician's standpoint (i played violin for many years), it captures the essense of an instrument's sound - timbre, articulation, etc. - better than anything i have heard. your mileage will vary though, and i can see some people preferring DVD-A with different kinds of music (e.g. non-acoustic). i'll gladly take either over 16/44 though.
something to remember when comparing DVD-A and SACD. the PCM format has benefited from almost a quarter decade of development, refinement, and implementation in the industry. DSD/SACD has had commercial use for only a few years. even before DVD-A, people were already moving to higher bitrates for PCM, so a lot was already in place for hi-res PCM in terms of recording and mastering equipment etc. DSD equipment is still very expensive and relatively hard to come by, and the consumer equipment is still in its first, maybe second generation. i don't see 24/192 PCM making huge technical advancements in the next 5 years, however DSD... we have not even seen the full potential of the format.
bitrate is a terrible gauge of resolution... well, ok, it will tell you how much data their is but that is about it. when it comes to different coding schemes though this becomes irrelevant. it is generally accepted that 24/192 resolution is the highest (nothing to do with bitrate though), and that SACD resolution is frequency dependent. to my ears, this means absolutely nothing. it's the fundamental nature of the encoding process that makes SACD superior to me. DSD does not have zero-crossing problems that PCM has, and the manner in which it captures a waveform is more akin to analog. ed meitner has said some interesting things on DSD vs. PCM, and obviously he is biased since he's in the SACD camp but i think his comments still have a lot of merit.
in the end though, the proof is in the listening, and at least for the music i listen to SACD is king. for me there is no mistaking the natural representation of tone and texture SACD provides. even 24/192 PCM can't beat it, and only the best analog can really compete. i think 24/192 does have a small edge in transient response (obvious technical advantage in that aspect), but it does not capture the musical whole with the same naturalness as SACD - it is still PCM and it still sounds digital to me. SACD is not perfect, but it allows me to relax and listen to the music more than anything since vinyl. and from a musician's standpoint (i played violin for many years), it captures the essense of an instrument's sound - timbre, articulation, etc. - better than anything i have heard. your mileage will vary though, and i can see some people preferring DVD-A with different kinds of music (e.g. non-acoustic). i'll gladly take either over 16/44 though.
something to remember when comparing DVD-A and SACD. the PCM format has benefited from almost a quarter decade of development, refinement, and implementation in the industry. DSD/SACD has had commercial use for only a few years. even before DVD-A, people were already moving to higher bitrates for PCM, so a lot was already in place for hi-res PCM in terms of recording and mastering equipment etc. DSD equipment is still very expensive and relatively hard to come by, and the consumer equipment is still in its first, maybe second generation. i don't see 24/192 PCM making huge technical advancements in the next 5 years, however DSD... we have not even seen the full potential of the format.
oh yeah, and you shoudl still worry about the noisy power supply if you are using it as a transport. i don't know why, but digital signals are still prone to modulation much like analog signals. it seems weird to me that jitter can be responsible for so much change in sound quality, but what else could it be other than error in the timing domain? as an EE i'd like to believe a good reclocking or buffering circuit would cure all this but apparently it does not. i did not want to believe changing digital cables could make a big difference but it does - as big as changing speaker cables or interconnects, even my friend who totally poo-pooed something so silly heard it and was left totally dumbfounded. ESPECIALLY if you are dealing with hi-bitrate DSD/PCM datastreams and you are doing conversion... you will need a super-clean power supply. that supply has got to go.
the DC performance of the opamp will determine offset to some extent, but the topology probably has more influence - the impedances at the different nodes etc. in this case, i'm not totally sure where the offset is coming from, it is not from the DAC i'm pretty sure, but you will definitely need to measure it after you make your changes as i'm sure it'll still be there to some extent. i was going to check if the muting circuit was only for power-up/down as well... maybe you are right. i guess most modern DAC ICs have internal muting anyway.
the DC performance of the opamp will determine offset to some extent, but the topology probably has more influence - the impedances at the different nodes etc. in this case, i'm not totally sure where the offset is coming from, it is not from the DAC i'm pretty sure, but you will definitely need to measure it after you make your changes as i'm sure it'll still be there to some extent. i was going to check if the muting circuit was only for power-up/down as well... maybe you are right. i guess most modern DAC ICs have internal muting anyway.
OPA134
Your right, the BUF-03 is a dated design , but they were free samples long ago. Also , the old PMI parts that were "AZ" came from the center of the best wafers, so their performance was a little better. I though about changing out the BUF-03 AZ's to the OPA134 a while back, however now I'm looking now to design a updated DAC. I designed this one about 93-94.
I've tried many different VI opamps too, and I'm still using the OP-27 AZ. These parts are old designs, what are you using for a VI amp. Having said that the DAC sounds good, but I'm sure it can sound better.
I use to have use of an Audio Precision at work to measure jitter. My favorite method is to use a precision stratum 1 clock source connected to HP synthesizer which I would use sync a fast scope. The jitter is easy to see with this setup. I have not see the LeCroy, do you have link for them.
Your right, the BUF-03 is a dated design , but they were free samples long ago. Also , the old PMI parts that were "AZ" came from the center of the best wafers, so their performance was a little better. I though about changing out the BUF-03 AZ's to the OPA134 a while back, however now I'm looking now to design a updated DAC. I designed this one about 93-94.
I've tried many different VI opamps too, and I'm still using the OP-27 AZ. These parts are old designs, what are you using for a VI amp. Having said that the DAC sounds good, but I'm sure it can sound better.
I use to have use of an Audio Precision at work to measure jitter. My favorite method is to use a precision stratum 1 clock source connected to HP synthesizer which I would use sync a fast scope. The jitter is easy to see with this setup. I have not see the LeCroy, do you have link for them.
hifiZen said:
i was just about to say, isn't the BUF03 noisy? then i realized chad already said so. duh.
i think if i were to build an IC-based output stage, i would use something like AD825 or AD8610... the BUF03 could still be useful to buffer the output of the chip (a la Walt Jung's topology) but there are other IC alternatives like BUF624 or something like that. this is only if i were too lazy to do a discrete borbely buffer though, which supposedly blows away any opamp when properly implemented...
chad, how are you implementing your MOSFET output buffer? how many active stages does it require? how did you deal with the DC offset of SE, just a blocking cap? a discrete SE class-A buffer after an opamp sounds like exactly what i was lookihng for!
i think if i were to build an IC-based output stage, i would use something like AD825 or AD8610... the BUF03 could still be useful to buffer the output of the chip (a la Walt Jung's topology) but there are other IC alternatives like BUF624 or something like that. this is only if i were too lazy to do a discrete borbely buffer though, which supposedly blows away any opamp when properly implemented...
chad, how are you implementing your MOSFET output buffer? how many active stages does it require? how did you deal with the DC offset of SE, just a blocking cap? a discrete SE class-A buffer after an opamp sounds like exactly what i was lookihng for!
Jim:
the Audio Precision test sets are probably still the best thing for audio use, and I can attest they they are very nice instruments... we have several of them at work... I only wish I had an excuse to make more use of them. But, the LeCroy instruments (www.lecroy.com) are far more sophisticated, being geared towards high-speed logic and telecommunications applications and so on. Some of the measurement capabilities are really interesting and innovative, and make jitter analysis significantly easier and better. I don't know if there are any technical papers on the LeCroy website, but if so, they're probably worth a read. Anyway, check it out... very nice stuff!
dorkus:
i agree, bitrate *is* a terrible way to estimate "resolution" (whatever that is...), since we're dealing with two fundamentally different encoding techniques. Unfortunately, I just don't know of any other way to get a handle on the numbers without approaching this from a purely information theory perspective. The real results will be in the listening, but much to my dismay, I have not yet had much chance to listen to either format.
The mosfet stage i use was something i cooked up in an afternoon as a quick'n dirty output buffer for a little project i was working on. It consists of a CFP or Sziklai Pair made from a small signal BJT and power mosfet. The other half is an active constant current source built with the complementary mosfet and another BJT. Quiescent current is something like 140mA. Part numbers were IRF510 and 9510, and 2n3904 if I recall correctly. The goal was to provide an easy load for the opamp, while providing the lowest possible output impedance to best stability into reactive loads. The whole thing is enclosed in the opamp feedback loop and operates at unity gain. There is no DC offset using this topology, so it can be direct-coupled, and even trimmed if the opamp has a trim feature. I have the schematic kickin around here somewhere, and I can send it to you if you like, but really I should toss it back on the bench and re-verify all the numbers, as well as do some optimization. At the time I didn't have time to properly optimize it, so I just threw it together and made sure it worked. I was genuinely surprised at how well it worked for a first-try! I do recall however, that the CFP stage can be a tad prone to oscillations, so that's one area where some additional care is necessary. One improvement I'm considering is to replace the active current source with just a simple resistor... should be good for stability and probably sound qaulity too. Anyway, let me know if you'd like the schematic, and I can dig it up for you...
btw - the ns500 is already starting to loosen up... i got a couple more SACDs yesterday, and as I listened last night, the detail started to slowly become more apparent, with greater texture and tonality becoming apparent... can't wait till this sucker gets fully broken in! I'm going to leave it running on disc repeat for a few days to try and accelerate the process...
the Audio Precision test sets are probably still the best thing for audio use, and I can attest they they are very nice instruments... we have several of them at work... I only wish I had an excuse to make more use of them.
But, the LeCroy instruments (www.lecroy.com) are far more sophisticated, being geared towards high-speed logic and telecommunications applications and so on. Some of the measurement capabilities are really interesting and innovative, and make jitter analysis significantly easier and better. I don't know if there are any technical papers on the LeCroy website, but if so, they're probably worth a read. Anyway, check it out... very nice stuff!dorkus:
i agree, bitrate *is* a terrible way to estimate "resolution" (whatever that is...), since we're dealing with two fundamentally different encoding techniques. Unfortunately, I just don't know of any other way to get a handle on the numbers without approaching this from a purely information theory perspective. The real results will be in the listening, but much to my dismay, I have not yet had much chance to listen to either format.
The mosfet stage i use was something i cooked up in an afternoon as a quick'n dirty output buffer for a little project i was working on. It consists of a CFP or Sziklai Pair made from a small signal BJT and power mosfet. The other half is an active constant current source built with the complementary mosfet and another BJT. Quiescent current is something like 140mA. Part numbers were IRF510 and 9510, and 2n3904 if I recall correctly. The goal was to provide an easy load for the opamp, while providing the lowest possible output impedance to best stability into reactive loads. The whole thing is enclosed in the opamp feedback loop and operates at unity gain. There is no DC offset using this topology, so it can be direct-coupled, and even trimmed if the opamp has a trim feature. I have the schematic kickin around here somewhere, and I can send it to you if you like, but really I should toss it back on the bench and re-verify all the numbers, as well as do some optimization. At the time I didn't have time to properly optimize it, so I just threw it together and made sure it worked. I was genuinely surprised at how well it worked for a first-try! I do recall however, that the CFP stage can be a tad prone to oscillations, so that's one area where some additional care is necessary. One improvement I'm considering is to replace the active current source with just a simple resistor... should be good for stability and probably sound qaulity too. Anyway, let me know if you'd like the schematic, and I can dig it up for you...
btw - the ns500 is already starting to loosen up... i got a couple more SACDs yesterday, and as I listened last night, the detail started to slowly become more apparent, with greater texture and tonality becoming apparent... can't wait till this sucker gets fully broken in! I'm going to leave it running on disc repeat for a few days to try and accelerate the process...
chad,
i would love a schematic, thanks. always looking for good circuit ideas. you can email me at marcyun@dorkus.org.
yeah, these Sony players take an eternity to break in for some reason. i'm not saying it'll sound like a XA777ES after break-in but it will certainly be a lot nicer to listen to in a few months. in particular, the inner detail, texturing, and imaging can be phenomenal, even with regular CDs, to the point where i can actually live with it as my only CD player and not miss my normal CD setup (Sony S7700 transport w/modified MSB Link DAC). i would try breaking it in at night while you sleep and letting it rest during the day though, or vice versa - the laser assembly gets quite hot with SACDs so playing 24-7 may not be the best thing for it. it isn't exactly the most durable piece of machinery ever made either.
i would love a schematic, thanks. always looking for good circuit ideas. you can email me at marcyun@dorkus.org.
yeah, these Sony players take an eternity to break in for some reason. i'm not saying it'll sound like a XA777ES after break-in but it will certainly be a lot nicer to listen to in a few months. in particular, the inner detail, texturing, and imaging can be phenomenal, even with regular CDs, to the point where i can actually live with it as my only CD player and not miss my normal CD setup (Sony S7700 transport w/modified MSB Link DAC). i would try breaking it in at night while you sleep and letting it rest during the day though, or vice versa - the laser assembly gets quite hot with SACDs so playing 24-7 may not be the best thing for it. it isn't exactly the most durable piece of machinery ever made either.
well chad...
you were right, i was wrong.
the output muting transistors were indeed only to prevent spikes on power on/off... so i was able to remove them all. as to be expected there is a turn-on transient now, but it is REALLY nasty - whether turning the unit on "hard" (with power switch) or "soft" (with remote control) you will get an absolutely huge spike at the outputs. soooo... you have to be very, VERY careful to completely mute your preamp before switching this thing on and off now.
the DC offset of the output stage was pretty much near zero even w/o the coupling caps. i'm not sure why i thought it was higher, but a few tenths of a millivolt seemed to be residual from the output muting circuit, which is now gone. i've bypassed the caps with shorting wires and there is zero DC offset.
on a side note, the output impedance of the player is probably around 500 ohms due to the 470 ohm output series resistor, placed after the coupling cap and before the muting circuit. in theory one could easily replace this resistor (a small SMT device) with a better metal film of lower value, say 200 ohms or so.
i took a closer look at the power supply on the output board, and it is some nasty stuff. the analog rails from the main switching supply seem to pass through some sort of pass transistor or regulator, but the device is tiny (3-terminal SMT). the dropout of this device is very small (around .6V) so i'm guessing it's a pass transistor and not a regulator IC. strangely, the negative rail also passes through a diode after the pass transistor, but the positive rail does not so the rails are assymmetric - i measured around +11.1V and -10.6V or something like that. then of course there is hardly any decoupling to speak of, and very very long traces to the multiple opamps. yuck. like i said, i'm not sure it's worthwhile just swapping opamp chips on this thing, it really needs a whole new analog section and power supply altogether. it wouldn't be hard at all to build except you'd need to break out into a bigger chassis. hmm...
in any case, your hunches about the output cap and muting circuit were totally right. i should do better homework before opening my mouth next time.
by the way, the player sounds GREAT with those two components out of the signal path - much more transparent, very clean and pure. at first i thought the sound was a tad thinner and bass-shy, but then i realized it was just rid of the bass bloat it had before probably due to the poor coupling caps. it sounds lovely, makes me wonder how great it could be with a full rebuild. hmmmm...
you were right, i was wrong.
the output muting transistors were indeed only to prevent spikes on power on/off... so i was able to remove them all. as to be expected there is a turn-on transient now, but it is REALLY nasty - whether turning the unit on "hard" (with power switch) or "soft" (with remote control) you will get an absolutely huge spike at the outputs. soooo... you have to be very, VERY careful to completely mute your preamp before switching this thing on and off now.
the DC offset of the output stage was pretty much near zero even w/o the coupling caps. i'm not sure why i thought it was higher, but a few tenths of a millivolt seemed to be residual from the output muting circuit, which is now gone. i've bypassed the caps with shorting wires and there is zero DC offset.
on a side note, the output impedance of the player is probably around 500 ohms due to the 470 ohm output series resistor, placed after the coupling cap and before the muting circuit. in theory one could easily replace this resistor (a small SMT device) with a better metal film of lower value, say 200 ohms or so.
i took a closer look at the power supply on the output board, and it is some nasty stuff. the analog rails from the main switching supply seem to pass through some sort of pass transistor or regulator, but the device is tiny (3-terminal SMT). the dropout of this device is very small (around .6V) so i'm guessing it's a pass transistor and not a regulator IC. strangely, the negative rail also passes through a diode after the pass transistor, but the positive rail does not so the rails are assymmetric - i measured around +11.1V and -10.6V or something like that. then of course there is hardly any decoupling to speak of, and very very long traces to the multiple opamps. yuck. like i said, i'm not sure it's worthwhile just swapping opamp chips on this thing, it really needs a whole new analog section and power supply altogether. it wouldn't be hard at all to build except you'd need to break out into a bigger chassis. hmm...
in any case, your hunches about the output cap and muting circuit were totally right. i should do better homework before opening my mouth next time.
by the way, the player sounds GREAT with those two components out of the signal path - much more transparent, very clean and pure. at first i thought the sound was a tad thinner and bass-shy, but then i realized it was just rid of the bass bloat it had before probably due to the poor coupling caps. it sounds lovely, makes me wonder how great it could be with a full rebuild. hmmmm...
wow, you beat me to the punch on all these items! I was just about to crank up the 'scope and poke around, but now i guess i won't have to. Thanks!
hmm... I wonder if those series output resistors should go too - tomorrow I can haul some misc. SMD parts home from work and try them out... I'm hoping the zero-ohm resistors will work. The only reason they might be there is to ensure stability of the cheapie opamps when driving capacitive cable loads (could this be another possible application for the high-powered preamp stage?)
btw - did you get the schematic? I just sent it out to you a few minutes ago...
Anyway, at some point, perhaps we can go in together on a replacement power supply PCB for the NS500? My thoughts turn to the ol' SMPS design from Audio Crafter's Guild as a good starting point for ideas...
Oh yeah, I'm about to install the loader-door sealing foam tape too. I'll let you know if it makes a difference.
hmm... I wonder if those series output resistors should go too - tomorrow I can haul some misc. SMD parts home from work and try them out... I'm hoping the zero-ohm resistors will work
. The only reason they might be there is to ensure stability of the cheapie opamps when driving capacitive cable loads (could this be another possible application for the high-powered preamp stage?)btw - did you get the schematic? I just sent it out to you a few minutes ago...
Anyway, at some point, perhaps we can go in together on a replacement power supply PCB for the NS500? My thoughts turn to the ol' SMPS design from Audio Crafter's Guild as a good starting point for ideas...
Oh yeah, I'm about to install the loader-door sealing foam tape too. I'll let you know if it makes a difference.
you will need an output resistor of some sort to keep the circuit stable. i don't think i've seen any application of an opamp that doesn't have the decoupling resistor in there - i've seen it as low as 10 or 20 ohms but sometimes it can oscillate with that low a value. i guess 100 ohms will leave a safe margin, maybe 50 if you use low capacitance cables?
p.s. i still think the sound of the player is just a tad thinner w/the mods (maybe a case of removing a complementary coloration or something) but it is definitely much clearer and the soundstage is HUGE - the airy, open, almost "floating" quality of SACD is further enhanced, and CDs sound even better as well. i really am tempted to rebuild the whole thing now!
p.s. i still think the sound of the player is just a tad thinner w/the mods (maybe a case of removing a complementary coloration or something) but it is definitely much clearer and the soundstage is HUGE - the airy, open, almost "floating" quality of SACD is further enhanced, and CDs sound even better as well. i really am tempted to rebuild the whole thing now!
Yep, after further listening last night, I'm starting to believe that this unit has a really nice DAC chip. With ordinary CDs, it's now sounding better than my old Rotel, albeit thinner and leaner, especially in the bass (a problem which could perhaps be addressed with some power supply upgrades). But, definitely better transparency and musicality overall. I think the ol Rotel is finally gonna get broken into and robbed of it's PMD-100 HDCD chip...
Good thing I had the sense to take lots of digital photos while the ns500 was all disassembled - now I can just look through the photos to get the DAC part number. It's a Sony proprietary part though, so it's unlikely I'll be able to find much data on it, but you never know...
Also, my first attempt at installing a foam loader door seal last night was unsatisfactory. It would have worked ok, except that the foam rubber was rather ugly, being visible in the little gap at the edge of the door (even though it's black). This time, i'm going to try and keep it concealed. It is also a bit of a pain to cut the foam to the correct thickness. The stuff I bought was just black foam rubber weatherstripping from Home Depot, 1/4" thick and 3/4" wide. After making some measurements with a caliper, i've determined the optimum thickness should actually be about 2.6mm, but it's tricky to slice the weatherstripping consistently to such a thin dimension. I may have to look around for a different foam to use...
Take 2 on the foam seal tonight, along with the first circuit mods.
Oh yes, another reason for those output resistors might just be for muting purposes. Obviously, you wouldn't want to dead short the opamp outputs when those muting transistors turn on. These resistors can have an effect on the bass too - I personally prefer the lowest possible output impedance when driving a cable. I think it's worth a try to see if they opamps will tolerate the absence of these resistors...
Good thing I had the sense to take lots of digital photos while the ns500 was all disassembled - now I can just look through the photos to get the DAC part number. It's a Sony proprietary part though, so it's unlikely I'll be able to find much data on it, but you never know...
Also, my first attempt at installing a foam loader door seal last night was unsatisfactory. It would have worked ok, except that the foam rubber was rather ugly, being visible in the little gap at the edge of the door (even though it's black). This time, i'm going to try and keep it concealed. It is also a bit of a pain to cut the foam to the correct thickness. The stuff I bought was just black foam rubber weatherstripping from Home Depot, 1/4" thick and 3/4" wide. After making some measurements with a caliper, i've determined the optimum thickness should actually be about 2.6mm, but it's tricky to slice the weatherstripping consistently to such a thin dimension. I may have to look around for a different foam to use...
Take 2 on the foam seal tonight, along with the first circuit mods.
Oh yes, another reason for those output resistors might just be for muting purposes. Obviously, you wouldn't want to dead short the opamp outputs when those muting transistors turn on. These resistors can have an effect on the bass too - I personally prefer the lowest possible output impedance when driving a cable. I think it's worth a try to see if they opamps will tolerate the absence of these resistors...
actually...
the 6-channel DAC is an AKM AK4357. sony just relabels it with their part number. i think there is also a separate AK4383 2-channel DAC chip for the stereo output but i am not sure.
you can see the (albeit skimpy) datasheets here:
http://www.asahi-kasei.co.jp/akm/usa/product/audio.html
everyone always talks about the Burr-Brown and AD chips, but this AKM part does some very special things. even though its design is very mass-market targetted (digital volume control taps, etc.) it does seem to reveal a remarkable amount of musical detail, despite the overal leanness of the player's sound as you mention. if it were practical, i think using 6 of the chips, each one with all 6 channels paralleled to form a single channel, could sound pretty killer.
let me know if the output stage does not oscillate with no series decoupling resistor. i suspect at the very least you will need 10 ohms but maybe not. you are right, the high value of the stock part was necessary due to the output muting switches.
the 6-channel DAC is an AKM AK4357.
sony just relabels it with their part number. i think there is also a separate AK4383 2-channel DAC chip for the stereo output but i am not sure.you can see the (albeit skimpy) datasheets here:
http://www.asahi-kasei.co.jp/akm/usa/product/audio.html
everyone always talks about the Burr-Brown and AD chips, but this AKM part does some very special things. even though its design is very mass-market targetted (digital volume control taps, etc.) it does seem to reveal a remarkable amount of musical detail, despite the overal leanness of the player's sound as you mention. if it were practical, i think using 6 of the chips, each one with all 6 channels paralleled to form a single channel, could sound pretty killer.
let me know if the output stage does not oscillate with no series decoupling resistor. i suspect at the very least you will need 10 ohms but maybe not. you are right, the high value of the stock part was necessary due to the output muting switches.
Hmm... very interesting little tidbit of info. Where did you find this out? I reviewed the photos last night, and sure enough, the CXD9675R and D9674TN located on the underside in section A1-A2 look to be pretty much identical to the aforementioned AKM parts. I did some comparison, and I think the two-channel outputs sound a little better than the 6-channel outputs. This pretty well conforms to the differences in the DAC specs. Too bad the AK4357 isn't pin compatible with the CS4362... that would be an interesting comparison / upgrade option. In any case, I'm very happy with these DACs so far.
I did the first of the mods last night... (a) foam seal around the loader door, which has been a bit disappointing in noise reduction. (b) shorted the output caps on the two-channel outputs.
I've decided to start modifying only the 2 channel outputs right now, since their sound is more promising to start with, and i don't have a multi-channel rig yet anyway. I'm going one step at a time to see the differences as I make each mod. After shorting the output caps, the transparency has opened up a bit, and the bass has become tighter and more extended. I could hear some near-subsonic rumbles which were absent before, but I don't think that the bass sounds any thinner, though. If anything it is a little more solid, and a little cleaner. Unfortunately, I have been doing all my listening on headphones, so I can't comment on the soundstage, but I imagine it is very good.
This weekend, I'll do muting transistors and series output resistors. I am a bit hesitant to take out the mute transistors, after you mentioned the large pop during soft power up/down. My concern is that the player could enter a soft power-down by itself after a period of inactivity while you're not monitoring it. Perhaps I'll devise some sort of reed relay replacement mechanism, or or other muting device which will be sonically benign. I'm starting to think about the power supply, too... in my eyes, it's starting to look ripe for a big change. Some of the connectors look like a bit of a pain though...
I did the first of the mods last night... (a) foam seal around the loader door, which has been a bit disappointing in noise reduction. (b) shorted the output caps on the two-channel outputs.
I've decided to start modifying only the 2 channel outputs right now, since their sound is more promising to start with, and i don't have a multi-channel rig yet anyway. I'm going one step at a time to see the differences as I make each mod. After shorting the output caps, the transparency has opened up a bit, and the bass has become tighter and more extended. I could hear some near-subsonic rumbles which were absent before, but I don't think that the bass sounds any thinner, though. If anything it is a little more solid, and a little cleaner. Unfortunately, I have been doing all my listening on headphones, so I can't comment on the soundstage, but I imagine it is very good.
This weekend, I'll do muting transistors and series output resistors. I am a bit hesitant to take out the mute transistors, after you mentioned the large pop during soft power up/down. My concern is that the player could enter a soft power-down by itself after a period of inactivity while you're not monitoring it. Perhaps I'll devise some sort of reed relay replacement mechanism, or or other muting device which will be sonically benign. I'm starting to think about the power supply, too... in my eyes, it's starting to look ripe for a big change. Some of the connectors look like a bit of a pain though...
This weekend, I'll do muting transistors and series output resistors. I am a bit hesitant to take out the mute transistors, after you mentioned the large pop during soft power up/down. My concern is that the player could enter a soft power-down by itself after a period of inactivity while you're not monitoring it. Perhaps I'll devise some sort of reed relay replacement mechanism, or or other muting device which will be sonically benign. I'm starting to think about the power supply, too... in my eyes, it's starting to look ripe for a big change. Some of the connectors look like a bit of a pain though... [/B][/QUOTE]
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If you do the PS, why not put in a relay in place of the transistors.
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If you do the PS, why not put in a relay in place of the transistors.
circuit analysis...
i've finished a very rudimentary analysis of the NS500V's circuits, primarly the analog circuit board.
the output stage uses the opamps in a differential amp topology. there are two RC filters in series going to the inputs of the opamp. both filters use 1k resistors, but the 2nd capacitor appears to be larger than the first (dunno the values). a 2.4k resistor is used for non-inverting input shunt and inverting input feedback. both opamp inputs also have a 680ohm series decoupling resistor. a feedback capacitor is employed to decrease gain to unity at high frequencies. the differential gain of the circuit is about 1.2. interestingly, Sony engineers actually did some optimization of the ".1" (subwoofer) output channel, where they use only a single RC filter, ostensibly with a much lower crossover frequency. a nice touch, i would have thought they would use 6 identical circuits for all outputs. the output decoupling resistor is 470 ohms, the high value necessitated by the output shunt muting circuit which i have removed. it's also followed by the 10uF coupling cap which i've shorted.
as i mentioned previously the opamps are powered by approx. +/- 11V. there is hardly any local decoupling to speak of except for the usual SMT ceramic caps by each opamp (probably .1uF). the +11V analog supply also feeds a 78M05 regulator near the analog stages, i believe this +5V output is used to power the analog section of the DAC chips. there is a 1000uF coupling cap after the regulator but the +5V must travel through the ribbon wire which connects the analog board to the main board. oh well.
there are a number of ancillary small transistors nearby the analog section. these are all for the muting circuit - there is a logical OR performed between the soft-mute signal from the digital section and the power supply mute signal (to prevent the thumping i'm getting now). i think the soft-mute signal is used to mute the extra channels when playing 2-channel material through the 5.1 outputs. these are all irrelevant now with the muting circuit gone anyway.
the video circuitry is very direct - the assorted video signals (component, composite, Y/C) are sent to a multi-channel buffer chip, followed by the usual impedance-matching resistor (68 ohms) and a small filter cap here and there. there is a little bit of additional circuitry, ostensibly for video muting. there is also a 79M05 regulator to provide the -5V to the buffer chip.
next up - plans of attack for improving the player.
i've finished a very rudimentary analysis of the NS500V's circuits, primarly the analog circuit board.
the output stage uses the opamps in a differential amp topology. there are two RC filters in series going to the inputs of the opamp. both filters use 1k resistors, but the 2nd capacitor appears to be larger than the first (dunno the values). a 2.4k resistor is used for non-inverting input shunt and inverting input feedback. both opamp inputs also have a 680ohm series decoupling resistor. a feedback capacitor is employed to decrease gain to unity at high frequencies. the differential gain of the circuit is about 1.2. interestingly, Sony engineers actually did some optimization of the ".1" (subwoofer) output channel, where they use only a single RC filter, ostensibly with a much lower crossover frequency. a nice touch, i would have thought they would use 6 identical circuits for all outputs. the output decoupling resistor is 470 ohms, the high value necessitated by the output shunt muting circuit which i have removed. it's also followed by the 10uF coupling cap which i've shorted.
as i mentioned previously the opamps are powered by approx. +/- 11V. there is hardly any local decoupling to speak of except for the usual SMT ceramic caps by each opamp (probably .1uF). the +11V analog supply also feeds a 78M05 regulator near the analog stages, i believe this +5V output is used to power the analog section of the DAC chips. there is a 1000uF coupling cap after the regulator but the +5V must travel through the ribbon wire which connects the analog board to the main board. oh well.
there are a number of ancillary small transistors nearby the analog section. these are all for the muting circuit - there is a logical OR performed between the soft-mute signal from the digital section and the power supply mute signal (to prevent the thumping i'm getting now). i think the soft-mute signal is used to mute the extra channels when playing 2-channel material through the 5.1 outputs. these are all irrelevant now with the muting circuit gone anyway.
the video circuitry is very direct - the assorted video signals (component, composite, Y/C) are sent to a multi-channel buffer chip, followed by the usual impedance-matching resistor (68 ohms) and a small filter cap here and there. there is a little bit of additional circuitry, ostensibly for video muting. there is also a 79M05 regulator to provide the -5V to the buffer chip.
next up - plans of attack for improving the player.
look what the FedEx man brought today...
i managed to pick up the now discontinued Sony C222ES SACD changer, last one in stock at Oade Brothers. only paid $325 too.
alas it does not use the same AKM DACs as the NS500V, but the Burr-Brown chip used is theoretically superior. we shall see.
don't worry... i will continue my work on the NS500V. but this new puppy will probably get priority pretty soon. oh well, the NS500V is destined for my brother in law school anyway.
i managed to pick up the now discontinued Sony C222ES SACD changer, last one in stock at Oade Brothers. only paid $325 too.
alas it does not use the same AKM DACs as the NS500V, but the Burr-Brown chip used is theoretically superior. we shall see.
don't worry... i will continue my work on the NS500V. but this new puppy will probably get priority pretty soon. oh well, the NS500V is destined for my brother in law school anyway.
Attachments
modding options...
ok, well i've already removed the muting transistors and bypassed the coupling caps, and that made a very worthwhile improvement. where do we go from here?
i could go all-out and do a rebuild of the unit, but then i realized i just bought a new player to replace this one (see above) and this unit will be going to my brother in law school. he needs an all-purpose unit (CD/DVD/SACD) that will be easy to use and reliable, so hacked-out tweaks are not an option - i need to stick as close to the original form as possible. my primary concern at the moment is with the big spikes when turning the unit on and off.
the weaknesses of this player, in approximate order of importance, are: 1.) the power supply (noisy and weak), 2.) the analog stage (mediocre all-around), 3.) everything else - steel chassis, mechanics, digital clock circuitry, etc. let's concentrate on points 1 and 2 for now.
replacing the power supply is really the only way to move this player up a notch - the stock circuit is just too crappy to be useful. this will require bigger and better transformers, so we will need an external chassis. no problem, i think my brother can deal with a small extra box for the power supply. internally, i can use the area where the current supply is for regulation and filtering banks, etc. a new power supply will also take care of the output turn-on thump problem since i will leave the output stages powered up at all times.
so my new power supply will use a small external chassis for 2 transformers (analog + digital), rectifiers, and smoothing caps. i'll probably use a multi-pin DIN connector to connect to the DVD.
now, what to do with the analog stages... hmm. my first impulse was to build an all-new PCB which could fit in place of the old one. i would retain the output jacks and propietary connectors to the DAC board but everything else would be all new. even the video circuit would be new - i would use high-bandwidth Analog Devices video buffers for all the video outputs. the analog stage would be based on the current one but with better opamps and none of the muting circuits. even better woudl be a Borbely discrete topology or the like but there is not enough room here and i want a quick and dirty solution.
after some further thought though, i figure it is too much work to go all new PCB, so i will have to deal with the old one - i am going to try some OPA2604 opamps tonight, maybe do some resistor substitutions (tantalum anyone?), get better supply decoupling caps. in combination with a new, better (linear not switching) power supply, i suspect this will smooth the sound out considerably.
ok, i am gonna try new opamps right now... will start thinking about the power supply next. i'll also try to sketch up some schematics of the sony's circuit. can someone recommend a good circuit diagramming program?
ok, well i've already removed the muting transistors and bypassed the coupling caps, and that made a very worthwhile improvement. where do we go from here?
i could go all-out and do a rebuild of the unit, but then i realized i just bought a new player to replace this one (see above) and this unit will be going to my brother in law school. he needs an all-purpose unit (CD/DVD/SACD) that will be easy to use and reliable, so hacked-out tweaks are not an option - i need to stick as close to the original form as possible. my primary concern at the moment is with the big spikes when turning the unit on and off.
the weaknesses of this player, in approximate order of importance, are: 1.) the power supply (noisy and weak), 2.) the analog stage (mediocre all-around), 3.) everything else - steel chassis, mechanics, digital clock circuitry, etc. let's concentrate on points 1 and 2 for now.
replacing the power supply is really the only way to move this player up a notch - the stock circuit is just too crappy to be useful. this will require bigger and better transformers, so we will need an external chassis. no problem, i think my brother can deal with a small extra box for the power supply. internally, i can use the area where the current supply is for regulation and filtering banks, etc. a new power supply will also take care of the output turn-on thump problem since i will leave the output stages powered up at all times.
so my new power supply will use a small external chassis for 2 transformers (analog + digital), rectifiers, and smoothing caps. i'll probably use a multi-pin DIN connector to connect to the DVD.
now, what to do with the analog stages... hmm. my first impulse was to build an all-new PCB which could fit in place of the old one. i would retain the output jacks and propietary connectors to the DAC board but everything else would be all new. even the video circuit would be new - i would use high-bandwidth Analog Devices video buffers for all the video outputs. the analog stage would be based on the current one but with better opamps and none of the muting circuits. even better woudl be a Borbely discrete topology or the like but there is not enough room here and i want a quick and dirty solution.
after some further thought though, i figure it is too much work to go all new PCB, so i will have to deal with the old one - i am going to try some OPA2604 opamps tonight, maybe do some resistor substitutions (tantalum anyone?), get better supply decoupling caps. in combination with a new, better (linear not switching) power supply, i suspect this will smooth the sound out considerably.
ok, i am gonna try new opamps right now... will start thinking about the power supply next. i'll also try to sketch up some schematics of the sony's circuit. can someone recommend a good circuit diagramming program?
I was browsing the net and found this page on modding the Sony NS500v:
http://home1.gte.net/res0f2t3/NS500V.htm
They also have C22ES
http://home1.gte.net/res0f2t3/C222ES.htm
All from:
http://home1.gte.net/res0f2t3/sacd.htm
They describe the mods that they do to the players.
I am probably going to buy the C222es and mod it myself.
--
Brian
http://home1.gte.net/res0f2t3/NS500V.htm
They also have C22ES
http://home1.gte.net/res0f2t3/C222ES.htm
All from:
http://home1.gte.net/res0f2t3/sacd.htm
They describe the mods that they do to the players.
I am probably going to buy the C222es and mod it myself.
--
Brian
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