charlesp210: I'm having very hard time finding out which converters used 1704's. Most I find mentioned actually don't. According to Stereophile the Mark Levinson 360 and 360S do:charlesp210
This will tell you (big download) don't all rush his server could blow.
CD-Player-DAC-Transport List
Cheers George
This will tell you (big download) don't all rush his server could blow.
CD-Player-DAC-Transport List
Cheers George
The pioneer DV-AX10 uses the sigma delta AD1955, probably the best of them.
The ML's do use PCM1704's. But don't buy a DAC just because it has the PCM1704. The digital from end is real crititcal with these you need something modern, proprietary and likely esxpensive unless you want want to go NOS and us the computer to upsample with USB-synchronous (buffer) jitter removal).
And then there is the output stage there are only two types that really workJocko style which has relatively high 2nd harm dist. or phono style (passive I/v+Jfet or tune gain) which also has relatively high 2nd harmonic distortion.
The only DAC you can buy like this is the commercial RD30 which is several thousands, the kits a great deal and sounds amazing but casing it is nearly impossible. The other option is Ians I2S converter which allows the computer to become the oversampler but you'll want to use it with his Fifi in order to isolate the DAC from the USB line (The RD30's achille's heal, big 10k spike on usb.)
How old is that manual, cause the one we opened were AD1955, thre PCM1704's are made by Ti. Does it show all 8 channels as TI PCM1704's?
Yada Yada, of course everything has to be good, or it's all bad, just as in marriage…
But are you saying ladder dacs like PCM1704 are more susceptible to jitter than newer sigma delta dacs? Therefore, they need BETTER digital front end?
Well, I don't know, but I pulled up a review of the Sonic Frontiers in Stereophile (also using PCM1704) and it had one of the jitter measurements I've ever seen published, 128pS. A much later Sony XA-777ES measured 183pS in rebook mode, the DSD "test" (a pure 10kHz signal prepared by Sony for testing purposes) measured 10dB higher noise across the entire spectrum. I would think that would be worse, though no spikes, but am suspicious about how good a test it was. But back to the era of the Sonic Frontier, I recall an early dCS stack measuring 280pS, costing $80k, and praised to the hilt. A year later Sonos measured better, about 220pS.
But I don't know, whatever the dither measurement, which could depend on a multitude of factors (and supposedly less than 10nS is not audible in JAES study) I think it fundamentally matters whether you have a ladder or delta sigma dac. They work differently. A ladder dac prepares an exacting sample every sampling interval. I think of that as honest points of data. A delta sigma sac creates a whole bunch of crude samples, and pads the vicinity of the nominal sample point with those crude samples, a compilation of crudity at slightly different moments of time.
Whatever it measures, I think that's a garbage approach, especially if done with one bit (as nearly everyone realizes now). But I think it's still a garbage approach if you do it with 3 or 4 bits.
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trick measures needed for 96kHz spdif and PCM 1704??
You here may be right that the digital interface is critical with these old PCM chips if you are going above 44.1kHz interface.
I had a PCM63 based Aragon D2A2, which I believe used fully synchronous connection (something I think is a good idea) and it had problems with my Dish network receiver over toslink. It frequently sounded underwatery, but the lock light never went out. A non-audiophile would notice. That would be linking at 48kHz, which was as high as it would go. I gave up and used the analog outputs of the satellite receiver, which sounded fine.
Was it the late 90's when the first ASRC (?) asynchronous interface chips appeared? These are used so ubiquitously now, I wonder how high you can easily go with SPDIF without them. I believe Lavry was able to avoid them, along with eschewing them. It doesn't make me feel good to know my digital interface is changing the data to match its own timing. I wonder how good that kind of interpolation really is. Once again, I'd rather have the real data, even if there is a tad of imperfect timing, a largely overrated concern IMO, because the approach of interpolating the data to fit the timing seems to me to embed the jitter in the data. On the other hand, if the servo locking is very unstable, interpolation would be better.
As far as I can tell from data sheets, the CS 8412, 8413, and 8414 are all the old fashioned synchronous receivers, with on chip buffers, doesn't look like any fancy DSP. The Burr Brown SRC4392 is the later asynchronous variety which relies on interpolation.
But fascinated by ladder DAC's, I just purchased up Onkyo RDV-1, which uses two 1704's according to this webpage.
The complete d/a DAC converter list - DutchAudioClassics.nl
I'm hoping to use it as a DAC up to 96kHz, which is what all my existing equipment runs at. The information I've been able to get so far doesn't say how high the SPDIF input runs. I might rather have gotten the Levinson 360s I mentioned before, which had about the same low measured jitter as the Sonic Frontiers I mentioned, around 120pS, but they go for about 4 times as much money as I paid for the Onkyo, and not much smaller either, and many of the units on the list that have 1704's are either even more pricey, or effectively unobtainable. If the Onkyo doesn't go allow 96kHz input, I might be looking for way to receiver with CS8412 or better. Speaking of clocks, the Onkyo blurb at the end of this blog makes a big deal about the clock the RDV-1 uses which they obtained from Apogee.
Integra RDV-1 CD/DVD Player | Steve Hoffman Music Forums
You here may be right that the digital interface is critical with these old PCM chips if you are going above 44.1kHz interface.
I had a PCM63 based Aragon D2A2, which I believe used fully synchronous connection (something I think is a good idea) and it had problems with my Dish network receiver over toslink. It frequently sounded underwatery, but the lock light never went out. A non-audiophile would notice. That would be linking at 48kHz, which was as high as it would go. I gave up and used the analog outputs of the satellite receiver, which sounded fine.
Was it the late 90's when the first ASRC (?) asynchronous interface chips appeared? These are used so ubiquitously now, I wonder how high you can easily go with SPDIF without them. I believe Lavry was able to avoid them, along with eschewing them. It doesn't make me feel good to know my digital interface is changing the data to match its own timing. I wonder how good that kind of interpolation really is. Once again, I'd rather have the real data, even if there is a tad of imperfect timing, a largely overrated concern IMO, because the approach of interpolating the data to fit the timing seems to me to embed the jitter in the data. On the other hand, if the servo locking is very unstable, interpolation would be better.
As far as I can tell from data sheets, the CS 8412, 8413, and 8414 are all the old fashioned synchronous receivers, with on chip buffers, doesn't look like any fancy DSP. The Burr Brown SRC4392 is the later asynchronous variety which relies on interpolation.
But fascinated by ladder DAC's, I just purchased up Onkyo RDV-1, which uses two 1704's according to this webpage.
The complete d/a DAC converter list - DutchAudioClassics.nl
I'm hoping to use it as a DAC up to 96kHz, which is what all my existing equipment runs at. The information I've been able to get so far doesn't say how high the SPDIF input runs. I might rather have gotten the Levinson 360s I mentioned before, which had about the same low measured jitter as the Sonic Frontiers I mentioned, around 120pS, but they go for about 4 times as much money as I paid for the Onkyo, and not much smaller either, and many of the units on the list that have 1704's are either even more pricey, or effectively unobtainable. If the Onkyo doesn't go allow 96kHz input, I might be looking for way to receiver with CS8412 or better. Speaking of clocks, the Onkyo blurb at the end of this blog makes a big deal about the clock the RDV-1 uses which they obtained from Apogee.
Integra RDV-1 CD/DVD Player | Steve Hoffman Music Forums
That would nearly all of them as they are usually preceded by an oversampling digital filter.
I can easily tell the difference between 1704 and delta-sigma / multibit / 1-bit converters. This particular current-out dac sounds more precise (less confused), and very true to the original recording (at least as true as I know the live performance / studio recording could sound through a digital reproduction system). There is also another positive side to 1704 with its low output; it demands a proper I/V converter to be used AND a very low ground plane noise , so the whole chain has to be of high quality. Furthermore, the use of valves in the output, or other quality methods to get analog out, could be implemented to provide final fine-tuning to meet individual tastes and/or blend with the rest of the system, knowing that digital section is of very good quality.
Nick
Totally agree, the PCM1704 reins supreme to me, and one of the simplest best I/V stages for it, is the (20ohm load) double stack no global feedback AD844. This combo is very hard to beat.
http://www.diyaudio.com/forums/digital-source/227677-using-ad844-i-v.html
Cheers George
Multiple PCM1704's with passive I/V and amplification (some of the best are high end photype preamp designs) is as good as it gets with the analog part of DAC design. The hard part is the digital filter and PLL/buffer for jitter control. The digital filter is the toughest for the hobbiest to obtain, the good SIW chips are 18 bits at most and I find the DF series terrible.
Building an NOS PCM1704 DAC with a FIFO using the computer to oversample competes with some real expensive stuff given good power supply regulation.
What ever happened to the Arta multibit chip that was supposed to take care of these hurdles?
With a single 1704 and only 1.2mA output noise is a big issue using passive I/V with very low resistances, as the following amplification gain has to be massive to get it back up to 2v out. This is where the zero global feedback AD844 shines, virtually no noise.
Cheers George
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