That was my thinking - as interesting as higher sample and bitrates are, for the time being the majority of demand in digital audio distribution appears to be for CD quality and lower. This might change quickly if DVD-Audio becomes popular though. Just trying to keep things as simple as possible! As far as cost goes, you can get a good quality external DAC on eBay for about the price of an M-Audio Audiophile PCI. You're probably right, as I don't intend to try mastering from analogue sources. I'm undecided as to whether I also want to digitize vinyl on this particular machine, which would need an analogue input of course. Project files copied over the LAN from the studio machine, probably. Cheers Daniel

Steve Harris wrote: Yeah. See part of my thought is that I *want* to master at 24-bit, get things right with EQ and compression, then add dithering and make sure it doesn't mess things up. To do that last step I want to go back and forth between 24-bit and 16-bit dithered, most especially on fade outs and in quiet passages, etc. I don't see how Daniel could do this with the setup he's proposing. What he could do is choose different dithering algorhythms and compare how each sounded, which is important, but if they *all* mess things up he won't know that. Dithering is a really interesting issue. One a few things I've done it's been important. On many type of source material it seems to make little or no difference. Much to learn. Bob Katz's book has some really interesting sections on this subject. If you don't do everything right in front of the dither, then dithering will make no difference at all. Anyway, it's just an observation about the system he's proposing to build. - Mark

Steve Harris wrote: I agree. I've never thought the dither option in Jack made sense to me. Jack should be a patch bay, not a processor. Too many options for mistakes. Since I still do all my final mixes in Pro Tools, I'll note that the Waves L2 UltraMaximixer offers dithering to different bit lengths (24, 20, 18, 16 and 8 I think) as well a few options for changing the dither type and noise shaping. Jamin should consider this sort of support when you feel it appropriate. You *might* consider having 2 outputs from Jamin - dithered and undithered - so that possibly you could A/B the two. Please time align the two if you ever do add an option like this. - Mark

Steve Harris wrote: OK, good points one and all. I hadn't considered it in quite that way before. To me (and I could be wrong) what you talk about above sounds more like format conversion. I'm not a programmer so I could have this messed up from a programmer's POV, but converting from a 24-bit ADC on receive data to some internal float format and then later back to a 24-bit DAC on transmit is not what I think of as 'dithering'. I grant you that there might be some form of dithering, in the sense of removing low-order bits, taking place. Some ADC's insert noise in their conversion at record time. However, I wouldn't have thought that either apps or Jack is inserting noise into the process at all those steps are they? My POV in the previous statement was more about the form of dithering we use when outputting specifically to a CD, for instance. The lost bits at the bottom problem and the fact that people can hear below the noise floor of 16-bit audio. I don't think people can hear below the noise floor of 24-bit audio, or at least my ears certainly can't. Great! If that can be done in a smooth way, without clicks and pops, it would be really useful for blind testing of dithering agorhythms. Take care, Mark

On Thu, 2004-03-18 at 16:49, Mark Knecht wrote: Clock is important, depending on D/A converter. Converter can also reclock the signal. My converter has jumper selectable reclocking. Other important thing on S/PDIF interfaces is that some soundcards can only output at some samplerate or do other nasty things before outputting the signal in digital format. Some cards use fixed internal samplerate of 48 kHz and then do samplerate conversion for other rates. Those may not be able to do direct 44.1 -> 44.1 output, but may rather do something like 44.1 -> 48 -> 44.1. Some examples are Aureal Vortex2 and some Sound Blaster cards. -- Jussi Laako <jussi.laako(a)pp.inet.fi>

On Thu, Mar 18, 2004 at 07:54:46AM -0800, Mark Knecht wrote: I think it would depend on how the card hardled 24bit data, but running jack at 16 might be safer, as JACK has a known float -> 16 bit algorithm. [ actually the current, undithered float -> 16bit algorithm is broken, it produces clicks with certain input, I fixed it in CVS, but theres not been a release for a while ] - Steve

Daniel, I didn't catch your initial message. Why do you want to do D/A. I would maintain digital all the way unless I've got some external analog gear that is very high quality. --- Anahata <anahata(a)treewind.co.uk> wrote: Do you know what type of power is used in digital mixers? Is it variable filtered or constant? Or maybe it's a mixture with constant to the analog circuits and filtered for the digital. I've never noticed any electrical related noise in my computer but then I don't have any analog in that system so there probably wouldn't be any. Is this assumption consistent with your understanding of the problem? I don't know much about these issue but I'm curious because my system will need analog i/o added. In fact, I don't know if my terminology of filtered and constant power is correct. Probably not. ron __________________________________ Do you Yahoo!? Yahoo! Mail - More reliable, more storage, less spam http://mail.yahoo.com

On Thursday 18 March 2004 18.43, Steve Harris wrote: Unless you got a firewire connector behind your ear... coming up on a gadget show in the near future.... :) /R

--- Daniel James <daniel(a)mondodesigno.com> wrote: My method seems to change everytime I turn around but I've always got a consol in the chain. My thing is to trust the JAMin meters, burn to file, and then to CD. I have been taking the stereo out from the Tascam DM-24 and routing it to an external CD-R. I do not trust the metering on the DM-24. For monitoring purposes a mixer is fine but it's not ideal for sending the final signal. In fact, it simply creates alot of doubts for me. I'm not familiar with those monitors. Someone I know that has been mastering for years uses Tannoy. I forget the model. These are much better than most Is the studio in the house? If yes, maybe you could run cable from the mixing and printing setup to the mastering room. That could eliminate the need for any new audio cards. I'm not familiar with the 1010. Does it have A/D with a pair of A and D for I/O? If yes, I wouldn't worry to much about upgrading it. Everything in the recording chain matters. To add budget mastering into your chain is a huge payback. could pale in comparison with your budget mastering setup. I'm not saying A/D doesn't matter, it does. I am suggesting that you look for an investment that will give the equivalent bang for the buck that the mastering has. A good preamp is a fine example of bang for the buck. If you record with overdubs then it can be used on the kick drum, bass guitar, guitar, vocals, etc. It's probably equal in value to having mastering capabilities. BTW, good preamps are expensive. It costs ardound $250.00 to test componentry and assemble these types of devices. You can purchase DIY preamp kits. I've seen what appear to be good kits at $500 per channel. If you do any accoustical design on the room, definitely post a message. I can probably help you understand the challange--obviously many people on this list can. Hidden away in my empire of crap, I have spreadsheets with formulas for simple cube shaped rooms. You can specify the desired reverberation time and calculate frequency absorption based on sabin value from the total square footage of existing materials. In my simple requirements, the objective is to achieve flat frequency response from 20Hz to 4kHz. I'm sure the million dollar rooms require much more intesive calculations than the rooms I've been involved with. If you build the treatment as panels that hang or stand on walls, they can be moved. Owens Corning accoustical insulation, the primary material, is expensive. The Makes sense to me. ron __________________________________ Do you Yahoo!? Yahoo! Finance Tax Center - File online. File on time. http://taxes.yahoo.com/filing.html

On Tue, Mar 23, 2004 at 10:27:02 +0000, Daniel James wrote: I think coax spdif is limited to something on the order of 10m, Toslink might be further, but if you convert to 110ohm AES/EBU over 3core XLR you can get a few hundred meters and still be in spec: http://www.ebu.ch/trev_t3250.pdf - Steve

On Tue, Mar 23, 2004 at 11:39:00 +0000, Anahata wrote: I'd be interested if you find anything - I've seen a few balanced analogue <-> CAT5 boxes (I guess doing AD/DA?) but they are very expensive. I suspect that 110ohm AES/EBU cable is a similar price to CAT5, and its easier to run than coax IIRC. FWIW you can get optical boosters that will send Toslink signals over 30m or so, but optical cable is no easier than S/PDIF coax to run. - Steve

Anahata wrote: Of course! But that wasn't my point. My point was he should really be using a 24-bit external DAC. Of course, it's more money... Don't fool yourself. Internal and external DACs can have plenty of jitter. External DACs that derive their clock directly from a USB source are pretty suspect in my mind, but they could have substantial PLL technology embedded in them that could help this. External DACs that use an external low-jitter clock source, and then a PC sound card that will take it's clock from that same sound source, is likely to be a better system, all other things kept equal. Whether we *hear* the jitter is a different (and interesting!) discussion. I don't know what the CD/MD/whatever means, but yes, extenal units are often purchased as 'upgrades'. Most systems offer good quality internal DACs. My Pro Tools system is like that. None the less people report better sound using (for instance, not a recommendation) the Cranesong HEDD 192 in bypass mode and separate word clock generator. Two problems: 1) You'll drop about $5K for the opportunity to try it out. 2) Who expects anyone who just dropped $5K to tell you it *didn't* make a difference!! One big problem with pushing toward high-end audio is that if you have a reasonably good system then you *cannot* hear the benifits of a great system by just trying it out for a few minutes. It's very interesting that when you sit in your studio using a great system and you substitute a reasonable component, like a mid-priced DAC, into it you often hear the differences right away. In short term listening tests human ears seem to be more able to pick out problems than hear improvements. My (small) concern about Daniel's proposed system is that once mastered he'll send his stuff to the people and maybe not get the response that he was hoping for. But again, there is no reason that he cannot start this way and improve the system over time. We all do it. What he's talking of doing is very sensible given a small budget. We're just pointing out some of the issue that might arise. Damn straight! ;-)

On Thu, 2004-03-18 at 10:49, Anahata wrote: A 24-bit converter will accept the 16-bit dithered output inside of a 24-bit stream. The bottom 8 bits are zeros. The upper 16 bits are the same dithered bits that will get written to the CD. What's the problem? A 16-bit converter will NEVER let him hear the difference between the dithered stream and the original 24-bit audio. Even if you are only going to a CD, which today is increasingly unlikely, there is still a good reason to have more bits to listen to. Certainly.

On Thu, 2004-03-18 at 19:41, Mark Knecht wrote: Having 24-bit DAC for just the sake of having 24-bit DAC isn't good reason. I have a high quality external DAC connected to S/PDIF of my M-Audio Delta1010 (which has 24-bit DACs in external box). And I really think that my external S/PDIF connected DAC sounds better, although the data stream is only 16-bits. It has high quality digital interpolation filter (8x / 16 -> 20 bits) and high quality true multibit DAC connected to it. Also the analog reconstruction filter is made of top quality components. Single bit (Sigma-Delta and the like, PWM) DACs tend to have IMD. I have designed two external audio DACs, and I still like the multibit one most. Of course, precision multibit DACs cost significantly more compared to "single bit" ones. With 24-bit DACs you are anyway wasting multiple bits, because you practically can't have ~140 dB S/N ratio for audio bandwidths. You are most likely to get little over 100 dB. So having 16-bit interpolated to 18 or 20 bits is usually enough (unless you have really quiet studio). Conclusion? High quality S/PDIF connected converter can be a good option. Just requires that computer's S/PDIF output is properly implemented. -- Jussi Laako &lt;jussi.laako(a)pp.inet.fi&gt;