Re: [LAD] FOSS Ethernet Soundcard

On Tue, Nov 24, 2009 at 09:48:53AM +0100, Karl Hammar wrote: Like always: Analog parts are hard to build. You need to calibrate your circuits, you have tolerances in almost every component, you need a lot of knowledge to manufacture a high-quality ADC/DAC. This is stuff you could easily avoid if you just outsource the whole ADC/DAC to a company which has the skills to do this. Luckily, there's ADAT: eight channels of audio over a single, optical link. So all you would need is one or more ADAT jacks somehow attached to your computer. Even better: studios usually have ADAT-ADC/DAC around, they're only looking for a way to get the data from/to the computer. If you want to win the whole pro-audio scene for this project, just provide ADAT plugs. The cheapest ADAT ADC/DAC I know is the Behringer ADA8000: http://www.thomann.de/gb/behringer_ultragain_pro8_digital_ada8000.htm There is better gear around, e.g., the Apogee Rosetta converters or the RME ADI line: http://www.thomann.de/gb/apogee_rosetta_800_24192khz.htm http://www.thomann.de/gb/rme_adi_8_qsm.htm Don't get me wrong, but I guess you cannot match their level of quality. And I don't see why one should fiddle around with soldering some crappy analog ports when he has decent studio gear around or can buy good-enough quality like the Behringer ADA8000. And while we are at it: RME and Apogee often also don't even dare to provide a preamp, because the studio guys have better stuff around: http://www.thomann.de/gb/avalon_ad2022_preamp.htm The preamp makes the music. If you feed your 4000EUR Neumann M149 microphone into an Avalon, all you need is a decent A/D conversion and you're halfway done with your radio production. ;) Sometimes, these preamps come with builtin A/D converters. That's what I use: http://www.thomann.de/gb/focusrite_liquid_4_pre.htm See it? It reads "ADAT I/O". If the FOSS soundcard provides an IP-to-ADAT converter with zero latency mixing facilities and builtin DSP effects, you'd be the man! You could instantly sell numbers of units, it would be the perfect onstage mix/split/monitoring solution. Come on, we're talking about a new development here. New development should never ever focus on IPv4, IPv6 has been there since 1998. Manually assigning an IPv4 address feels so 1990s and increases complexity. With IPv6, you just put your device on the net and it gets a link-local address, that is, fe80:MAC-Address. (it's not exactly the MAC-Address, but it's derived from it). No need to assign anything, you're instantly ready to go. And you have link-local multicast (read: broadcast in IPv4 terminology). Just use an address starting with ff02 and send your streams to it. Everyone subscribed to this address would receive it. I have a jack client streaming/receiving IPv6 multicast streams. It's so easy. No configuration at all, I just fire up the application and I'm ready to stream. I don't even care about the other addresses used on this LAN. Let's talk about discovery: how do you intend to find your devices on the net? With IPv6 multicast, all soundcards could statically listen on ff02::dead:beef. Then, the workstation sends its discovery packet to ff02::dead:beef, all cards receive it and respond (connect back) to the source IP (or an IP given in the packet, but that's not even necessary) In other words: A single address is sufficient to trigger the callbacks. The cards tell the workstation about their very own IPv6 multicast streaming address, and the workstation simply has to tune in. There you go. Streaming and discovery within approx. 200 lines of code. Perhaps it's even a good idea to provide a master clock on an IPv6 multicast address, but clocks are a separate issue (in a studio, the workstation would probably slave to some decent specialised clock, e.g., Apogee Big Ben) [RockNet, Roland Digital Snake, Ethersound] Approximately what you could expect from a 100MBit/s link. At least 32 channels, rocknet even more, but they use 400MBit/s.

Cheerio