Konnichiwa,
Maybe. I don't.
To quote Big "Abso-*******ing-lutly".
No, it actually sounds BETTER!
Nope, you need a filter that peaks around 3.3db @ 20KHz, like those I designed and published.
I have that. PDM-200 into parallel PCM1704 CD-Player with improved on-board clock etc. NON-OS still sounds better, HF correction or not. A lot of tweaking on the "big machine" has pushed it closer to NON-OS though (NON-OS DAC is Ack!dAck BTW).
I tried most.
Oversampling.
Crossgender sampling.
Lewgrade sampling.
Under & oversampling.
Bitstream.
Multibit.
Crossgenderbit.
Lesbitian.
The verdict:
Best DAC - Multibit
Second Best DAC - DIRECT BITSTREAM (eg. SM5872)
Thirdbest DAC - The rest
Best Digital Filter - No Filter, SINC Correction deployed
Secondbest Digital Filter - HDCD PDM100/200
Thirdbest Digital Filter - the rest
Oh yes, Upsampling definitly Lewinskies, even using dCS gear.
Oh Yes, DVD-Audio at 96/24 whips the butt of SACD like so much, if all recordings are the same and good.
Oh yes, ANALOG RULEZ KOOL, RIGHTOUS DUB, EARS GOOD.
TUBES RULEZ KOOL, TRANSISTORS RULES - TOO MUCH HARD GRAFT MON.
GOOD SPEAKERS ARE 95db+/1W/1m AND have even tonality and controlled dispersion.
Anyway, the above may put my comments into perspective.
Sayonara
Dave said:
Maybe. I don't.
Dave said:
To quote Big "Abso-*******ing-lutly".
Dave said:
No, it actually sounds BETTER!
Dave said:
Nope, you need a filter that peaks around 3.3db @ 20KHz, like those I designed and published.
Dave said:
I have that. PDM-200 into parallel PCM1704 CD-Player with improved on-board clock etc. NON-OS still sounds better, HF correction or not. A lot of tweaking on the "big machine" has pushed it closer to NON-OS though (NON-OS DAC is Ack!dAck BTW).
Dave said:
I tried most.
Oversampling.
Crossgender sampling.
Lewgrade sampling.
Under & oversampling.
Bitstream.
Multibit.
Crossgenderbit.
Lesbitian.
The verdict:
Best DAC - Multibit
Second Best DAC - DIRECT BITSTREAM (eg. SM5872)
Thirdbest DAC - The rest
Best Digital Filter - No Filter, SINC Correction deployed
Secondbest Digital Filter - HDCD PDM100/200
Thirdbest Digital Filter - the rest
Oh yes, Upsampling definitly Lewinskies, even using dCS gear.
Oh Yes, DVD-Audio at 96/24 whips the butt of SACD like so much, if all recordings are the same and good.
Oh yes, ANALOG RULEZ KOOL, RIGHTOUS DUB, EARS GOOD.
TUBES RULEZ KOOL, TRANSISTORS RULES - TOO MUCH HARD GRAFT MON.
GOOD SPEAKERS ARE 95db+/1W/1m AND have even tonality and controlled dispersion.
Anyway, the above may put my comments into perspective.
Sayonara
Elso, well, let us open the link, 
but this is second order ...
Elso wrote > In the schematic below R3= 3k65, c2 = 3n3 and ???= ground. The non-inverting input of the first opamp is not connected to ground but to Vref
Kuei Yang Wang wrote > ... like those I designed and published
Could you please mention source/link?
but this is second order ...Elso wrote > In the schematic below R3= 3k65, c2 = 3n3 and ???= ground. The non-inverting input of the first opamp is not connected to ground but to Vref
Kuei Yang Wang wrote > ... like those I designed and published
Could you please mention source/link?
Attachments
Filter
Hi Dimirtri,
The first pole is in the IV-converter. That is the 3n3 cap across the 3k65 resistor. In the Filterproprgram this first pole is actually a passive low-pass filter preceding the multiple feedback filter. Simulation with Microcap and scope pictures of squarewave from a test CD prove it works right. (No ringing or overshoot as we expect from a Bessel filter). Initially it was a balanced filter, hence the staggered schematic, but I abandoned the balanced concept for a number of reasons.
What source /link are you after? This was my own idea using as few opamps as possible.
dimitri said:Elso, well, let us open the link,
Elso wrote > In the schematic below R3= 3k65, c2 = 3n3 and ???= ground. The non-inverting input of the first opamp is not connected to ground but to Vref
Could you please mention source/link?
Hi Dimirtri,
The first pole is in the IV-converter. That is the 3n3 cap across the 3k65 resistor. In the Filterproprgram this first pole is actually a passive low-pass filter preceding the multiple feedback filter. Simulation with Microcap and scope pictures of squarewave from a test CD prove it works right. (No ringing or overshoot as we expect from a Bessel filter). Initially it was a balanced filter, hence the staggered schematic, but I abandoned the balanced concept for a number of reasons.
What source /link are you after? This was my own idea using as few opamps as possible.
Elso asked > What source /link are you after?
After Thorsten Loesch filter (that peaks around 3.3db @ 20KHz) which he designed and published:
http://groups.yahoo.com/group/Thund...rcuits/NOS-DAC-Analogue/TDA1543 I-V Conv.gif but I have no passwd
The previous filters are there:
http://www.audioasylum.com/scripts/t.pl?f=digital&m=75355
After Thorsten Loesch filter (that peaks around 3.3db @ 20KHz) which he designed and published:
http://groups.yahoo.com/group/Thund...rcuits/NOS-DAC-Analogue/TDA1543 I-V Conv.gif but I have no passwd
The previous filters are there:
http://www.audioasylum.com/scripts/t.pl?f=digital&m=75355
Given the spectral contents of typical music, given the Sin(f)/f response of a NONOS DAC's ZOH output stage, and given the expected (non)linearity of typical amplifiers and loudspeakers in the octave beyond 20kHz, some of you might want to reconsider the relevance or irrelevance of filtering out the first image.
Thundetone Technical Yahoo group forum
Register with Yahoo, then you can enter the forum but not post!
Oooops did not work, as I am not a member of Thunderstone technical. Hahahahahahaha. I will not apply for a membership!!
Anyway Thorstens filter has a lot of ringing and overshoot....Must have...
dimitri said:
Register with Yahoo, then you can enter the forum but not post!
Oooops did not work, as I am not a member of Thunderstone technical. Hahahahahahaha. I will not apply for a membership!!Anyway Thorstens filter has a lot of ringing and overshoot....Must have...
Hi Elso,
Maybe your ears cant hear very high frequencies so you dont lose to much sound ?
Anyone with some audio software should rip a cd and see how much sound is above the 10 k range - ie filter out everything below, its a tiny amount, very important yes but if you cannot hear much of it anyway you may as well filter it off to prevent some distortion I suppose ?
I have active crossover at 3.5k and when I forget to turn on the bass amp the treble is merely a whisper by itself so 10k must be very quiet indeed.
Maybe your ears cant hear very high frequencies so you dont lose to much sound ?
Anyone with some audio software should rip a cd and see how much sound is above the 10 k range - ie filter out everything below, its a tiny amount, very important yes but if you cannot hear much of it anyway you may as well filter it off to prevent some distortion I suppose ?
I have active crossover at 3.5k and when I forget to turn on the bass amp the treble is merely a whisper by itself so 10k must be very quiet indeed.
Hi Elso,
Im not saying you have bad hearing, probably not in fact.
I dont know what your system is ?
It could be that your system is a little top end bright ? And that with the filter it flattens out again ?
10K seems low but I just tried applying a filter starting at 10k and going to nothing at 20k, not much lost, just a little fizzz.
Why not try some test tones ? Have you got a cd burner to burn some ? You can then determine where your hearing ( or system goes to ). This would be of course on a cd player that doesnt have a 10k lowpass!
Im not saying you have bad hearing, probably not in fact.
I dont know what your system is ?
It could be that your system is a little top end bright ? And that with the filter it flattens out again ?
10K seems low but I just tried applying a filter starting at 10k and going to nothing at 20k, not much lost, just a little fizzz.
Why not try some test tones ? Have you got a cd burner to burn some ? You can then determine where your hearing ( or system goes to ). This would be of course on a cd player that doesnt have a 10k lowpass!
Low-pass filter
Hi Dimitri, Great, I designed it that way!
Group delay is constant till 10kHz as it should be for a Bessel filter.
It is one way or the other. Or you have a smooth 3150 sine wave and less extreme high's or you have a stepped wave (staircase) and reasonably flat frequency response.
Filterpro allows you to design the same filter with a -3dB crossover at 15kHz or 20kHz in just a few seconds, if that is more to your liking.
dimitri said:Dear Elso, your filter has -3dB@10kHz and -12.5dB@20kHz, which I can consider for the fun of the thing only.
Hi Dimitri, Great, I designed it that way!
Group delay is constant till 10kHz as it should be for a Bessel filter.
It is one way or the other. Or you have a smooth 3150 sine wave and less extreme high's or you have a stepped wave (staircase) and reasonably flat frequency response.
Filterpro allows you to design the same filter with a -3dB crossover at 15kHz or 20kHz in just a few seconds, if that is more to your liking.
all this filtering....yuck We need hi-rez!
There are some DVDs that have 24/192 on one side and 24/96 on the other (Classic Records, Hi-Rez Music).....24/192 is much better!.....so....imagine playing back 24/192 with a multi-bit DAC using parallel 24 bit devices (PCM1704 or discrete wild thangs) and running with no digital filger and NO analog filter (maybe one pole starting at 20K?). With the noise mostly at 192K the IM products will be far from 20K...This has got to sound incredible. Later this year I am going to start making live recordings on all hardwired fully tweaked 24/192 system. And for playback I will make one of the above....parallel 1704 with no digital filter and no analog filter and discrete class A all FET output stage....Oh Boy...Joy joy joy. My plan is to release the music in anyway someone can use it....the standard release might be CD on one side and 24/192 on the other....but if you send me hard drive or whatever, I will load the info directly bypassing the crappy laser reading jitter prone system....fun stuff ahead.
There are some DVDs that have 24/192 on one side and 24/96 on the other (Classic Records, Hi-Rez Music).....24/192 is much better!.....so....imagine playing back 24/192 with a multi-bit DAC using parallel 24 bit devices (PCM1704 or discrete wild thangs) and running with no digital filger and NO analog filter (maybe one pole starting at 20K?). With the noise mostly at 192K the IM products will be far from 20K...This has got to sound incredible. Later this year I am going to start making live recordings on all hardwired fully tweaked 24/192 system. And for playback I will make one of the above....parallel 1704 with no digital filter and no analog filter and discrete class A all FET output stage....Oh Boy...Joy joy joy. My plan is to release the music in anyway someone can use it....the standard release might be CD on one side and 24/192 on the other....but if you send me hard drive or whatever, I will load the info directly bypassing the crappy laser reading jitter prone system....fun stuff ahead.
ok, lets say the noise is at ~ 200kHz and we have an amplifier chain with loads of stages and gain and feedback however. There may be more than one stage doing some frequency multiplying / mixing. Lets say we have the 200khz and some music signal 1500Hz. We get as a result from the mixer 1500Hz, 200kHz, 198,5khz and 201,5khz. We also feed in 950Hz at same time and get 950Hz, 200kHz, 200,95Hz and 199,05 kHz. No problem so far. No we have one more stage doing some mixing - and get out of it 950Hz, 1500Hz, all the unimportant HF noise, and difference of 200,95and201,5 = 550Hz ?! in addition we have : 3kHz, 2kHz and 1kHz IMD products from signal with noise + the products of IMD from the audio signals.
I do not think its that easy - higher sampling rate no IMD. What about building only gainstages that do not tend to multiply frequencys instead and use the sampling rate we cango and buy music with on CD?
Does this thought make any sense or am i totaly wrong?
Hi Elso,
So are you saying we should abandon hi-rez because we need to get even better at clocking? That would be the analog equivalent of saying we should stick with cassettes because 30 inch half track reels are a pain in the butt. This is why God made you....to make us great clocks....he he.
I think that most people just don't get how long and complicated the digital path is. Every connection, every cable, every buffer, every coupling cap, every pulse transformer, every power supply, every power supply cap, all grounding, shielding, AC power supply filtering, clocks, etc. makes the sound different.
The basic nerd look at it is ones and Zeros....those of us that have seriously modded digital transports, digital cables, DACs know that it is a long very complicated signal path....this is why we need to get rid of as much of it as possible (and that is why I generally like players better than DACs....shorter signal path=purer sound). Look at an analog reel to reel deck. It has both input and output circuitry all in one box, it probably uses single ended class A circuits, it does not make RF or much at any rate and the signal path is incredibly short relative to the digital path. The best? A to D converters (Meitner, DCS, Prism, etc.) have op amps on the input, not great power supplies, no shielding, not the best parts and then they run the signal down some not so good cables to some recorder that might have to demultiplex the signal, etc. etc.....way too long a path.....and if you are going to do it that way then at least use the best wire, hardwire, use the worlds best clocks, get rid to those op amps (use super trannies on the analog input or a few resistors (like Guido says he is doing in his new converter), etc. etc....anotherwords tweak the crap out of it. This is what I intend to do in my recording chain. I am going to hardwire the mics directly to the mic preamp that will be directly hardwired to a tranny on the input of the A to D. The signal path between the A to D and the recorder (if not already one box) will be carefully examined and every connection will be hardwired if possible using the best wires, super line conditioning using ZSleeve technology and others, etc. etc. I want to see what is possible not merely do what the commercial complanies are doing. How come you don't see any tweakers doing serious digital? Why are all the SACD, DVD-A recordings done on stock commercial equipment? Remember the recorder that Reference recordings used and the modified Studer (via John Curl) that Dave Wilson used? We need this kind of serious tweaking in the digital domain....so please keep working on clocks, IM distortion removal, removal of digital filters, analog filters, power supplies, etc.
So are you saying we should abandon hi-rez because we need to get even better at clocking? That would be the analog equivalent of saying we should stick with cassettes because 30 inch half track reels are a pain in the butt. This is why God made you....to make us great clocks....he he.
I think that most people just don't get how long and complicated the digital path is. Every connection, every cable, every buffer, every coupling cap, every pulse transformer, every power supply, every power supply cap, all grounding, shielding, AC power supply filtering, clocks, etc. makes the sound different.
The basic nerd look at it is ones and Zeros....those of us that have seriously modded digital transports, digital cables, DACs know that it is a long very complicated signal path....this is why we need to get rid of as much of it as possible (and that is why I generally like players better than DACs....shorter signal path=purer sound). Look at an analog reel to reel deck. It has both input and output circuitry all in one box, it probably uses single ended class A circuits, it does not make RF or much at any rate and the signal path is incredibly short relative to the digital path. The best? A to D converters (Meitner, DCS, Prism, etc.) have op amps on the input, not great power supplies, no shielding, not the best parts and then they run the signal down some not so good cables to some recorder that might have to demultiplex the signal, etc. etc.....way too long a path.....and if you are going to do it that way then at least use the best wire, hardwire, use the worlds best clocks, get rid to those op amps (use super trannies on the analog input or a few resistors (like Guido says he is doing in his new converter), etc. etc....anotherwords tweak the crap out of it. This is what I intend to do in my recording chain. I am going to hardwire the mics directly to the mic preamp that will be directly hardwired to a tranny on the input of the A to D. The signal path between the A to D and the recorder (if not already one box) will be carefully examined and every connection will be hardwired if possible using the best wires, super line conditioning using ZSleeve technology and others, etc. etc. I want to see what is possible not merely do what the commercial complanies are doing. How come you don't see any tweakers doing serious digital? Why are all the SACD, DVD-A recordings done on stock commercial equipment? Remember the recorder that Reference recordings used and the modified Studer (via John Curl) that Dave Wilson used? We need this kind of serious tweaking in the digital domain....so please keep working on clocks, IM distortion removal, removal of digital filters, analog filters, power supplies, etc.
Actually I'd love to (have the time and money to) do a weird ADC, possibly a flash or pipelined multi-bit device running at 176kHz, with only a vestigial anti-aliasing filter etc. etc.
But I don't have this and that.
Audio Note are working on a newish-type ADC of which I know no details, except that it most probably does without a classical AA filter, and knowing PQ a bit, possibly uses just a transformer in that role.
I have a CDR with recordings off a turntable by this ADC prototype. Suffices to say that there are no strange things going on in the treble.
But I don't have this and that.
Audio Note are working on a newish-type ADC of which I know no details, except that it most probably does without a classical AA filter, and knowing PQ a bit, possibly uses just a transformer in that role.
I have a CDR with recordings off a turntable by this ADC prototype. Suffices to say that there are no strange things going on in the treble.
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