306 replies to this topic

[Audiobugged]

I2s is the only way to go.................

That's not quite true! What it should read is:
I2s the way its done by Mario is the only way to go! Everything else is crap... This is a proven and non-disputable fact - proven by years of experience and confirmed again and again by the many followers who worship at the temple of Kaj.

[regal]

Glad to see this discussion. The biggest problem is audio manufacturers in the business of integrating DAC chips are at the mercy of the big chip foundries. Of course they and the jounals have to defend the modulator technology, they have no choice as you can't develop a business plan based on an integration of an onsolete or "planned obsolete" chip. So I empathize with the attempts to change the subject and direct this read off tangent.

An important point was made, one can't design a voltage regulator that does anything with "noise" above 1Mhz. Today we have DSP's commonly running at 400 mhz, some 600 mhz. We have asynch usb sharing ground with a computer running at 2.4 GHZ. Then the cheap modulating chips themselves run well above any speed that can be dealt with. With DC, the ground is just as important as the regulator side. These high frequencies are inaudible in themselves but they 'cause instability in the wonderful analog circuitry we design, and that we hear.

I am by no means an expert but know that a PCM1704 with the BPO & servo defeated (big slow caps on the pins) with good discrete power regulation paired with a antique PMD100 and a modern <3ps spdif transport (Jocko Legato) is at a level where a S-D can't compete for my ears. Why? Maybe because a pulse transformer separates my analog from megahertz oscillation inducing garbage.

We all have different preferences and subjectivity plays a huge role with audio, but it is time for the DAC integrators to start pushing back to their suppliers and listening to their customers.

Tell TI " we don't want these modulators." Tell Analog Devices "your Sharc + AD1955 model isn't interesting our customers". Tell "ESS nice try but not what we want."

Instead of all the audio integrators trying to outsmart each other with jitter babble, maybe they should team up, listen to customers, and tell the big chip companies to wake the f up

Edited by regal, 13 April 2012 - 03:13 PM.

[techspurt]

The biggest problem is audio manufacturers in the business of integrating DAC chips are at the mercy of the big chip foundries.

As a DAC designer I see this as an opportunity, not a problem. I agree a lot of the designers think like this - but then there are
others who break the mould. Vincent Brien's 'TotalDAC' is one such. Cees at Metrum got away from using a DAC targeted at
audio - its still from a big foundry, but it demonstrates that designers aren't limited to what the big chip companies design
specifically for audio. A lot of the fun comes from re-purposing alternative and non-mainstream parts in my experience.

Of course they and the jounals have to defend the modulator technology, they have no choice as you can't develop a business plan based on an integration of an onsolete or "planned obsolete" chip. So I empathize with the attempts to change the subject and direct this read off tangent.

Businesses don't though have to depend on business plans (and VCs, etc. etc.). They
can grow organically. But sure there's a good helping of defensiveness around trying to
support established habits with all kinds of smokescreen BS. Rather than say 'yep, its
the only way we could raise finance by doing it this way.'.

With DC, the ground is just as important as the regulator side. These high frequencies are inaudible in themselves but they 'cause instability in the wonderful analog circuitry we design, and that we hear.

Yep - the myth of 'ground' is widespread.

I am by no means an expert but know that a PCM1704 with the BPO & servo defeated (big slow caps on the pins) with good discrete power regulation paired with a antique PMD100 and a modern <3ps spdif transport (Jocko Legato) is at a level where a S-D can't compete for my ears.

If you listen for yourself you're more of an expert than someone relying on 'established practice' and
measurements.

We all have different preferences and subjectivity plays a huge role with audio, but it is time for the DAC integrators to start pushing back to their suppliers and listening to their customers.

Tell TI " we don't want these modulators." Tell Analog Devices "your Sharc + AD1955 model isn't interesting our customers". Tell "ESS nice try but not what we want."

Its certainly one approach, but I doubt that they're in the mood for listening. They're driven by the majority
who don't much care about the sound quality (perhaps the 'iPod generation' ?). Big companies are compelled
by law to put shareholders before customers in the USA.

Instead of all the audio integrators trying to outsmart each other with jitter babble, maybe they should team up, listen to customers, and tell the big chip companies to wake the f up

I myself prefer the approach of disrupting their businesses by gradually stealing their customers. Then when
they wake up its because their company has lost its revenue stream...

[georgehifi]

If those who still think that newer (Delta Sigma) is better than R2R (multibit), and can't see that it's advertised to be better so the big companies can reap the larger profits, these three links may sink in and start some doubting the direction dac chip development has gone. Complete I'm sorry with more measurment graphs.

http://www.6moons.co...otaldac2/1.html

http://www.totaldac.com/ready_dac.htm

http://www.6moons.co...totaldac/1.html

Cheers George

Edited by georgehifi, 16 April 2012 - 07:51 AM.

[Arg]

Looks like a perfectly good DAC to me.

So, George, is your argument that the R2R measures better therefore it sounds better?

[georgehifi]

So, George, is your argument that the R2R measures better therefore it sounds better?

No as I said in my first post, properly implemented R2R (multibit) has always sounded better to me than properly implemented Delta Sigma (nothing mentioned about measurments).
But it's nice to know the the measurments made are backing up my listening preference of R2R Multibit sounding better to my ear than Delta Sigma.

As always no dac amp or any audio electronics worth it's salt has ever been developed without the use of measurments, it's nice to see them back each other up, and how it should be if the right measurments are made.

Cheers George

[techspurt]

http://www.totaldac.com/ready_dac.htm

Couldn't hold back a chuckle at the measurements for this one - even at a test tone of -80dBfs, noise modulation is clearly evident.
Also he's making the very popular error of equating an FFT bin contents with the noise floor. The claim is 'nose floor is -134dB' when
that's the (narrow-band) noise of one FFT bin.

[zenelectro]

If those who still think that newer (Delta Sigma) is better than R2R (multibit), and can't see that it's advertised to be better so the big companies can reap the larger profits, these three links may sink in and start some doubting the direction dac chip development has gone. Complete I'm sorry with more measurment graphs.

http://www.6moons.co...otaldac2/1.html

http://www.totaldac.com/ready_dac.htm

http://www.6moons.co...totaldac/1.html

Cheers George

George,

I don't want to buy into the DS versus multibit - but just want to correct a few things here wrt noise measurements.

These noise floor measurements are not really very well done and are ambiguous.
That noise floor measurement on the second link is NOT an RMS measurement and would not be close to -134dBFS RMS period.
The designer does not appear to correctly understand the nature of FFT measurements and how they work. This is a very common misconception.

When an FFT sweep is performed, the measuring device does a narrow bandwidth sweep, usually from 20Hz to 20kHz.
The narrower the bandwidth of the scan, the less effective noise power and so the lower the 'grass' on the FFT graph.
This is exactly how we can see a small level signal through much higher effective RMS noise.

Refer to the attached FFT below, which is of Anedio DAC that uses Sabre DAC chip. The quoted dynamic range is around 126dB (unweighted) but
the 'grass' in that FFT is below -150dB.
This is because the figure of -126dB is the RMS noise with a bandwidth of 20kHz. The bandwidth of the actual FFT sweep may have been only a few
Hz, as such it can 'see' the -140dB signal below the -126dB RMS noise.

The way those noise floor FFT's are presented, with grass around -130dB, I would assume a 20kHz RMS noise floor of around -100dB but it is
impossible to state without proper measurement conditions.

Z

Attached Files

•  anedio 1.jpg   114.46K   1 downloads

Edited by zenelectro, 18 April 2012 - 10:52 PM.

[georgehifi]

Same fudging of specs goes on with all of them, Abraxalito says they need to measure differently.
Too many well healed "Golden Ears" prefer old school R2R multibit for Red Book playback and if you have HDCD decoding as well with it's excellent filtering charactistics your almost some say better than sacd, even guys like Nelson Pass.

The other proof is in the secondhand market, try to pick up a s/h dac or cdp cheap that has TDA1541 crown, or PCM63k, or PCM1702k, PCM1704k with PMD100 or 200 these all fast becoming investment property rather than clean up day junk, where they should be at 20 and more years old, people have ears and they know what gives the best Red Book sound.

I'm not interested in sacd or high rez down loads all I what is my hundreds of Red Book cd's to sound at their best and Multibit (R2R) does this better than any of todays hyper Delta Sigma chips can which are part bitstream ( single bit) and we remember how bland they sounded when they tried to push them onto us.

http://www.diyaudio....tml#post2853112

Cheers George

[datafone]

Meh...

I have changed to a Marnatz CD65 and only changed the outout caps, it already has a bunch of Cerafine caps in it stock, and I must say that I really like the sound of this TDA1541 (non A) based player

Will do more mods as I go, Clock, Nos mod, seperated reg's...tube output.

[techspurt]

Definitely get rid of the SAA7220 - but that's not at all straight forward because its the clock master for the transport. A separate oscillator will be needed and you'll also lose a level of error correction, interpolation and soft muting. An easier alternative is to isolate its considerable power supply noise as far as possible and bypass the filter part to go NOS.

[datafone]

Yeah, I was going to connect one of the pins from SAA7220 to one on the controller chip (forget off hand the numbers) to correct the muting after by-passing it. I'm not sure if those other aspects are audible, those that have done the nos mod report improvements without audible drawbacks

Will be adding separate transformers for power to some of the chips, as well as running the new clock off a battery.

Edit: I read up on the nos mod on the Lampi site http://lampizator.eu...mpling/NOS.html but this player does use a different controller chip (m4803a I think), so I need to look at that a bit more before doing it.

Reading a thread elsewhere i think It's SAA7220 Pin 23 to pin 11 on the m4803a

Edited by datafone, 19 April 2012 - 11:56 PM.

[empirical]

I think you guys are missing the point of these older, simpler DACs. It has little to do with the S/N or the THD numbers. The lower these are, yes you will get a bit more detail in the playback. However the thing that makes these sound better has nothing to do with this. The natural sound of these has everything to do with the digital filtering or lack thereof IME. Take the digital filtering off any Sigma-Delta DAC (if you can) and find out for yourself. Really nice.

Steve N.

[jkeny]

Might it be that this "jump factor" is down to the ability of multibit NOS to track & reproduce music transients better than the averaging of sigma-delta?

[datafone]

Steve, while I see that you are saying... that if one was able to by-pass the filtering of a S-D chip there would be significant improvement?

But what are you implying regarding the relationship in the case of something like the TDA1541?

Sorry, and It might just be my lack of sleep, but I'm finding that part somewhat cryptic of sorts

Edit: Are you agreeing, or disagreeing that the TDA1541 benefits with by-passing the filter chip?

Edited by datafone, 20 April 2012 - 06:13 AM.

[empirical]

Might it be that this "jump factor" is down to the ability of multibit NOS to track &amp; reproduce music transients better than the averaging of sigma-delta?

I doubt it. I have not experienced this jump factor correlation. I get great jump factor out of all D/A chips. Jump factor has more to do with jitter and power subsystem of the DAC IME. Maybe it was just coincidence that these were related by the poster.

Steve N.

[empirical]

Steve, while I see that you are saying... that if one was able to by-pass the filtering of a S-D chip there would be significant improvement?

Exactly. Its possible on some chips. That's how I know this.

But what are you implying regarding the relationship in the case of something like the TDA1541?

TDA1541 can sound really really good. No digital filtering. It NOS. Part of the problem with this chip is current drive on the outputs. If you parallel several, problem solved.

Edit: Are you agreeing, or disagreeing that the TDA1541 benefits with by-passing the filter chip?

What filter chip?

The version that is most popular is the TDA1543.

Steve N.

[jkeny]

I doubt it. I have not experienced this jump factor correlation. I get great jump factor out of all D/A chips. Jump factor has more to do with jitter and power subsystem of the DAC IME. Maybe it was just coincidence that these were related by the poster.

Steve N.

Here's what I experience & surmise. SD DAc chips in the short experience I have had with them have definite strengths in the bass with more slam & definition but when it comes to top end resolution it can reveal some thinness & sibilance. NOS DACs have superiority in the HF rendition & clarity but are not somewhat noticeably bass shy. Jump factor is something that was introduced here but I've no direct experience of it.

My premise is that the averaging of DS DACs works well when there are a large number of sample per musical cycle over which to average (in the LF area) & produce an apparent higher bit depth but are less capable at HF where the number of samples per music cycle is much less leading to a less focused sound. The transient attack of the leading edge on cymbals, & other instruments would be less well catered for in this averaging at HF & therefore the jump factor would also be a more noticeable characteristic of NOS DACS where no averaging noise shaping is happening

Edited by jkeny, 20 April 2012 - 10:31 AM.

[zenelectro]

Here's what I experience & surmise. SD DAc chips in the short experience I have had with them have definite strengths in the bass with more slam & definition but when it comes to top end resolution it can reveal some thinness & sibilance. NOS DACs have superiority in the HF rendition & clarity but are not somewhat noticeably bass shy. Jump factor is something that was introduced here but I've no direct experience of it.

My premise is that the averaging of DS DACs works well when there are a large number of sample per musical cycle over which to average (in the LF area) & produce an apparent higher bit depth but are less capable at HF where the number of samples per music cycle is much less leading to a less focused sound. The transient attack of the leading edge on cymbals, & other instruments would be less well catered for in this averaging at HF & therefore the jump factor would also be a more noticeable characteristic of NOS DACS where no averaging noise shaping is happening

To be completely honest here, I think we are all wading around in the mud here trying to describe subjective artifacts that may or may not be produced by the DAC itself as opposed to the supporting circuitry.

For just about every argument here I can think of a technical counter argument. Just to use your above case here John (not picking on you), the HF performance of DS DACs is theoretically way superior to NOS multibit DAC's. If you
pass a 2 (or 3 or 4) tone 18k+19k tone through a DS DAC it will be almost perfect, do the same to a NOS 1541 and it will be obliterated. Measurement wise that is!

I think the main issue here generally is these DS DAC's are a complex amalgamation with quite a few different internal processes.

If we compare the 1541 to a Sabre, they are utterly diametrically opposed in just about every aspect.

- 1541 uses relatively slow / low noise current steering internal architecture // Sabre is a very high speed Cmos design
- 1541 has very easy PS requirements // Sabre will tax the PS, especially at high frequencies to the max
- 1541 is simple and devoid of all other internal functions such as digital filtering, ASRC etc.
- 1541's OP stage runs low sample rates and an easy +- 2mA // Sabre runs a high current, low impedance super fast switching OPS

The requirements on surrounding circuitry for Sabre are much more demanding - but as we know surrounding circuitry on 1541 is already -very- important.

There is also the issue that a not insignificant part of the typical 0xOS 1541 -> tube OP stage sound is -additive- ie; euphonic

A lot of factors to consider before we jump to conclusions about jump factors - there's a mouthful

[datafone]

What filter chip?

Steve N.

TDA1541 (non A), SAA7220P/A

Edit:Not SAA7220P/B, so corrected.

Edited by datafone, 21 April 2012 - 06:17 PM.

[pchan]

Hi guys

I have not found any doc to back these claims but does the Sabre 9018 sample regardless of bit rate and sampling and process the data to 32/192??
All I could find is that it accepts 32/192 in the s/pdif and limited to the USB converter used. Can you direct me to such documentation if they exist!

Regards

pchan

[techspurt]

Take the digital filtering off any Sigma-Delta DAC (if you can) and find out for yourself. Really nice.

I haven't seen one where the removal of all filtering is possible, unless you're saying that noise shaping does not really count as filtering. To my way of thinking, integrators
in the feedback loop of noise shapers are indeed doing filtering. So were you referring to the FIR filters that precede the nose shapers? If so, which chips have you
heard minus their on-board filters?

[techspurt]

If you pass a 2 (or 3 or 4) tone 18k+19k tone through a DS DAC it will be almost perfect, do the same to a NOS 1541 and it will be obliterated. Measurement wise that is!

I'm going to suggest its not the NOS DAC which is doing that but your measurement kit. There will of course be very close-in imaging products which
would need filtering out. Once that is done, the NOS DAC does just as well. But it might compromise the amplitude if its subject to the classic ZOH
droop.

The requirements on surrounding circuitry for Sabre are much more demanding - but as we know surrounding circuitry on 1541 is already -very- important.

This s a most excellent observation and one which plenty of designers of Sabre DACs probably do not take seriously enough. What if the audible
performance is limited by the physical impossibility of getting clean enough power to the die itself? Or perhaps no-one to-date has a good enough
(read linear enough up to very high RF frequencies) output stage? For myself I steer away from Sabre for those, and other reasons. Why make the
design of an already difficult to design power supply and output stage that much harder?

[georgehifi]

Take the digital filtering off any Sigma-Delta DAC (if you can) and find out for yourself. Really nice.

Steve N.

I haven't seen one where the removal of all filtering is possible, unless you're saying that noise shaping does not really count as filtering. To my way of thinking, integrators
in the feedback loop of noise shapers are indeed doing filtering. So were you referring to the FIR filters that precede the nose shapers? If so, which chips have you
heard minus their on-board filters?

I was wondering the same thing, correct me if I'm wrong, by removing the filtering (noise shaping) from Delta Sigma would'nt that then end up with more hf noise an anything you can imagine?

Cheers George

[pchan]

This s a most excellent observation and one which plenty of designers of Sabre DACs probably do not take seriously enough. What if the audible
performance is limited by the physical impossibility of getting clean enough power to the die itself? Or perhaps no-one to-date has a good enough
(read linear enough up to very high RF frequencies) output stage? For myself I steer away from Sabre for those, and other reasons. Why make the
design of an already difficult to design power supply and output stage that much harder?

How does this explained that Sabre 9018 dacs are more detailed in the HF than other dacs such as the WM and BB? This corresponds to all the sabre 9018 implementations that include WFS2, Weiss etc

[pchan]

I was wondering the same thing, correct me if I'm wrong, by removing the filtering (noise shaping) from Delta Sigma would'nt that then end up with more hf noise an anything you can imagine?

Cheers George

Its quite obvious that removing the filters will introduced distortion into the audio path, maybe that this distortion helps the SQ to sound better like valves do. But I actually disagree for technical reasons that this has a better product!!

[techspurt]

How does this explained that Sabre 9018 dacs are more detailed in the HF than other dacs such as the WM and BB?

As I've never actually listened to a 9018 I can only speculate. 'Detailed' means different things to different people -
to me it means having more audio MSG. I strive in my DAC designs to delete such 'detail' - IME it arises from RF
interactions with analog stages that can't cope with the phenomenal rise-times presented to them by today's digital
circuits in deep sub-micron CMOS. In other words 'detailed' to me is a bug, not a feature.

Now if you and I both agree on what 'detailed' means (perhaps we don't in fact, but let's assume we do mean the
same thing but disagree whether its a bug or a feature) then I have a hypothesis for why Sabre sounds more 'detailed'.
Its because the CMOS process used is finer geometry, hence faster rise times and more MSG. But as I say this is
just speculation, don't take it too seriously.

[pchan]

As I've never actually listened to a 9018 I can only speculate. 'Detailed' means different things to different people -
to me it means having more audio MSG. I strive in my DAC designs to delete such 'detail' - IME it arises from RF
interactions with analog stages that can't cope with the phenomenal rise-times presented to them by today's digital
circuits in deep sub-micron CMOS. In other words 'detailed' to me is a bug, not a feature.

Now if you and I both agree on what 'detailed' means (perhaps we don't in fact, but let's assume we do mean the
same thing but disagree whether its a bug or a feature) then I have a hypothesis for why Sabre sounds more 'detailed'.
Its because the CMOS process used is finer geometry, hence faster rise times and more MSG. But as I say this is
just speculation, don't take it too seriously.

I never take thing seriously Techspurt, its all cool bro,
Detail to me means things like room echos, feet taping, lips moving pages getting turned, this is one of the many things on how I gauge audio components.

Edited by pchan, 20 April 2012 - 02:38 PM.

[techspurt]

I never take thing seriously Techspurt, its all cool bro,

No fear of me taking things seriously - its all good fun!

Detail to me means things like room echos, feet taping, lips moving pages getting turned, this is one of the many things on how I gauge audio components.

Ah, ok that's a different meaning of 'detail' to the one that I commonly encounter. So
would require a different kind of speculation about differences - probably in part its
due to the reduced noise modulation of the Sabre compared to other parts. They're
the only guys who even admit its an issue so I'd expect them, amongst all the S-D
vendors, to make the best stab at fixing it up.

[datafone]

Yeah, I was going to connect one of the pins from SAA7220 to one on the controller chip (forget off hand the numbers) to correct the muting after by-passing it. I'm not sure if those other aspects are audible, those that have done the nos mod report improvements without audible drawbacks

Will be adding separate transformers for power to some of the chips, as well as running the new clock off a battery.

Edit: I read up on the nos mod on the Lampi site http://lampizator.eu...mpling/NOS.html but this player does use a different controller chip (m4803a I think), so I need to look at that a bit more before doing it.

Reading a thread elsewhere i think It's SAA7220 Pin 23 to pin 11 on the m4803a

Ah! I got it wrong, had a factory sticker over one chip, turns out to be the SAA7210p controller.

So easier now to follow what others have done

Edit:Ah bugger! the tracing on this PBC is very different in It's routing from chip to chip compared to other examples found on the net (other players), so as I have learnt the hard way in the past, if in doubt....don't bother, I'll have to just look at getting cleaner power to the SAA7220

Edited by datafone, 21 April 2012 - 08:32 PM.

[Nada]

How does this explained that Sabre 9018 dacs are more detailed in the HF than other dacs such as the WM and BB? This corresponds to all the sabre 9018 implementations that include WFS2, Weiss etc

Are you sure that DACs running the Sabre 9018 chip are more detailed in the high frequencies then the Burr Brown R2R PCM1704 implemented in a capable DAC like a Mark Levinson 360S?
I heard a WFS vs a PDX long ago but the high freqeuncy didnt seem different. Maybe cause Im to old to hear anything in the bat range?

[pchan]

Are you sure that DACs running the Sabre 9018 chip are more detailed in the high frequencies then the Burr Brown R2R PCM1704 implemented in a capable DAC like a Mark Levinson 360S?
I heard a WFS vs a PDX long ago but the high freqeuncy didnt seem different. Maybe cause Im to old to hear anything in the bat range?

BB as in the PCM 1798 not the 1704. Apologies for being miss leading!

[georgehifi]

Here is a list of dac chips used by Mark Levinson in their D/A Convertors, everyone is Multibit.



MARK LEVINSON No 30
2 x UA D20400 – SM5803AP – CS8412-CP
MARK LEVINSON No 30.5
2 x UA D20400A – PMD100 – CS8412-CP
MARK LEVINSON No 30.6
4 x PCM1704
MARK LEVINSON No 31
It’s a Transport
MARK LEVINSON No 31.5
It’s a Transport
MARK LEVINSON No 35
2 x UA D20400A – SM5803APT / PMD100
MARK LEVINSON No 36
4 x PCM1702-K – PMD100 – CS8412-CP
MARK LEVINSON No 37
It’s a Transport
MARK LEVINSON No 39
4 x PCM1702 (PCM1704) – PMD100
MARK LEVINSON No 360S
4 x PCM1704
MARK LEVINSON No 390S
2 x AD1853

Cheers George

[zenelectro]

'Detailed' means different things to different people -
to me it means having more audio MSG. I strive in my DAC designs to delete such 'detail' - IME it arises from RF
interactions with analog stages that can't cope with the phenomenal rise-times presented to them by today's digital
circuits in deep sub-micron CMOS. In other words 'detailed' to me is a bug, not a feature.

It is pretty easy to either filter the RF after the DAC or just design analog stages that are able to deal with it.

[techspurt]

It is pretty easy to either filter the RF after the DAC or just design analog stages that are able to deal with it.

Do you have any examples where this has been done to your satisfaction? I'd like to study methods of achieving
it. With my (not very fine CMOS geometry) TDA1545 prototype DAC, it did take me quite a few attempts at designing
a filter to effectively deal with it. So, IME I wouldn't describe it as 'pretty easy'.

[zenelectro]

Do you have any examples where this has been done to your satisfaction? I'd like to study methods of achieving
it. With my (not very fine CMOS geometry) TDA1545 prototype DAC, it did take me quite a few attempts at designing
a filter to effectively deal with it. So, IME I wouldn't describe it as 'pretty easy'.

Here are a few: 1) passive filtering straight after dac 2) transformer OP stage 3) open loop I-V circuitry, tube or SS - or combinations of them.

[techspurt]

Yeah I did the passive filtering bit quite early on using some common-mode chokes (the dac has diff-out). Then I moved on to a sequence of SMT inductors. Neither gave me complete satisfaction. Transformers for me are too bulky, heavy and expensive, same with tubes but I'm sure they would work well. I have a working solution now anyway with unorthodox filtering means followed by a dedicated RF amp IC (not an opamp).

[georgehifi]

I have a working solution now anyway with unorthodox filtering means followed by a dedicated RF amp IC (not an opamp).

This sounds interesting are you keeping the maker of the rf amp quiet, or can you tell us??? What is the settling time??

Sorry thought you were using the RF amp as an I/V converter straight after the dac output before filtering.

Cheers Geoge

Edited by georgehifi, 23 April 2012 - 09:16 AM.

[Arg]

Here is a list of dac chips used by Mark Levinson in their D/A Convertors, everyone is Multibit.



MARK LEVINSON No 30
2 x UA D20400 – SM5803AP – CS8412-CP
MARK LEVINSON No 30.5
2 x UA D20400A – PMD100 – CS8412-CP
MARK LEVINSON No 30.6
4 x PCM1704
MARK LEVINSON No 31
It’s a Transport
MARK LEVINSON No 31.5
It’s a Transport
MARK LEVINSON No 35
2 x UA D20400A – SM5803APT / PMD100
MARK LEVINSON No 36
4 x PCM1702-K – PMD100 – CS8412-CP
MARK LEVINSON No 37
It’s a Transport
MARK LEVINSON No 39
4 x PCM1702 (PCM1704) – PMD100
MARK LEVINSON No 360S
4 x PCM1704
MARK LEVINSON No 390S
2 x AD1853

Cheers George

All obsolete, I think. Their only current model sporting DA conversion is the 512 player which uses AD1955 Delta Sigma DACs.

[techspurt]

This sounds interesting are you keeping the maker of the rf amp quiet, or can you tell us??? What is the settling time??

Its not a secret - I've mentioned it over on DIYA and had all kinds of comments that's its not a very low distortion part
Its AD605 which includes an analog volume control. Its a bit noisy but good enough for 16bits.

Sorry thought you were using the RF amp as an I/V converter straight after the dac output before filtering.

I'm using passive I/V, then filtering then the AD605.

[Nada]

All obsolete, I think.

Its such a pity Mark Levinson stopped producing the flagship R2R PCM1704 based 360S.




Maybe they had little choice with TI ?



Not Recommended for New Design (NRND)

[georgehifi]

All obsolete, I think. Their only current model sporting DA conversion is the 512 player which uses AD1955 Delta Sigma DACs.

They used Multibits right up into the mid 2000's over Delta Sigma in those early 2000's years, then had to conceed and use it because of SACD and Hi-Rez, this is when they lost all hi-end Red Book following, because it didn't sound as good.
The Multibit dacs they made are so extremly sort after and are fetching outragous money from the Red Book crowd, because they sound better than their newer Delta Sigma ones they have been forced to use now for high rez.

Cheers George

Edited by georgehifi, 23 April 2012 - 12:35 PM.

[Arg]

I'm getting all teary eyed about early DACs now!

[techspurt]

The Multibit dacs they made are so extremly sort after and are fetching outragous money from the Red Book crowd, because they sound better than their newer Delta Sigma ones they have been forced to use now for high rez.

What forces audio companies to use S-D when PCM1704 is good for the bit depth and sample rates of hi-res?

[georgehifi]

What forces audio companies to use S-D when PCM1704 is good for the bit depth and sample rates of hi-res?

Availability, cheap, in big quantities.

Cheers George