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Old 11th January 2005, 04:57   #1
Dtrump
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Quality sound cards?

I'm planning a Shoutcast stream for a radio station that is all balanced I/O. I would prefer to use a sound card that is also balanced.

I foolishly picked up an E-Mu 1212m tonight but the DSP encoder for WinAmp does not appear to interface with it. Audio is OK into the board and its mixer, but nothing gets to the Shoutcast Source Input. I had also figured it might be a decent board for my satellite client machines, but I'm planning on Linux clients and the E-Mu doesn't have Linux drivers. Back to CompUSA I go.

Can anybody recommend a quality board for these applications?
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Old 11th January 2005, 08:53   #2
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I stick with the standard SoundBlaster or SB compatible. They are one of the most widely accepted and used types of soundcards that I've seen.

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Old 11th January 2005, 11:33   #3
Dtrump
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Quote:
Originally posted by MegaRock
I stick with the standard SoundBlaster or SB compatible. They are one of the most widely accepted and used types of soundcards that I've seen.
That's specifically why I stayed with Creative Labs, hoping that anything high-end that they put out would work. Wrong assumption. There are no SB cards that I know of that that have any kind of professional I/O.

I've always considered SB as competent but falling short when I need something for critical applications. I own a recording studio and also engineer for this station and I've always used cards with balanced I/O. This is the first time I've been stymied.
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Old 11th January 2005, 17:43   #4
DJ AmPs
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The SB live uses the E-MU DSP and Crystal DAC/ADC's. Quality audio that is more than required for a 192k or less mp3 stream. You will have no complaints.
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Old 11th January 2005, 18:07   #5
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Quote:
Originally posted by DJ AmPs
The SB live uses the E-MU DSP and Crystal DAC/ADC's. Quality audio that is more than required for a 192k or less mp3 stream. You will have no complaints.
Thanks. I really don't worry much about the sound quality. This is spoken word only. But I am interfacing with equipment that is balanced, +4dbm which will require additional electronics if the card isn't capable of it directly.

I'm exploring some suggestions from a pro-audio listserv that I'm active on. Unfortunately, none of those folks use SHOUTcast, so they don't have the direct experience I was seeking.
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Old 11th January 2005, 18:22   #6
gaekwad2
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It certainly can't keep up with high quality cards:

Tests with Rightmark Audio Analyzer:

SB Live! (at 48kHz, at 44.1 you'd get additional IMD caused by its resampler)

M-Audio Delta Audiophile 2496

Edit: OK , for spoken word it won't make much difference.

Fuck this place.
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Old 11th January 2005, 18:31   #7
radioparadise
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Since you're only dealing with the input to the card, you can put together a resistive-pad adapter cable pretty cheaply. XLR connectors at one end into a resistive pad, then a mini-plug cable. Modern balanced outputs don't care a bit about being connected to unbalanced inputs - you just need to drop the level down to -10 or thereabouts using a couple of resistors.

The only situation you'd run into that could cause you grief is if the cables were very long (> 100') and/or you were operating in a high-RF environment (like at an AM transmitter site). In that case, spring $100 or so for a Henry matchbox.

Even then, the Matchbox + SB Live combo will come in cheaper than a pro-grade sound card - and you'll save yourself hours & hours of fuss & bother (especially if you're using Linux).

I've used SBLive cards for years at commercial FM stations. They're perfectly suitable.
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Old 11th January 2005, 19:24   #8
DJ AmPs
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I second this, if you plan on running Linux stick with the SB. Just use a level shifter.

And for comparing SB to other "pro" quality cards, once you compress (in other words, mangle) the audio into an mp3 or similar format all of those comparisons are gone out the window. SB will be fine for broadcast in any circumstance.
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Old 11th January 2005, 21:34   #9
Dtrump
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Thanks guys. Yep, I've built many resistive pads and used many different "match box" devices. And the DA driving the encoder/server will handle unbalanced just fine. I just dislike 1/8" connectors in this kind of environment. I'm more of an XLR or Phoenix compression type guy. Old school, I guess.

The satellite machines will be interfacing with SCA exciters which will want +4dbm so I'll need to have some kind of interface.

Anyway, thanks again for your help. I have returned the Creative E-Mu piece and picked up an SB Live to work with for now. Maybe it will be the final answer.
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Old 12th January 2005, 01:57   #10
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Quote:
Originally posted by Dtrump
I have returned the Creative E-Mu piece and picked up an SB Live to work with for now. Maybe it will be the final answer.
Not quite yet. The card that I picked up is an SB Live 24-Bit.

I'm currently running a test stream with it and all is fine. Considering that the source is the output of an SCA receiver, it really sounds pretty good. It sounds pretty much the same as our public Real Audio stream that is picked off ahead of the SCA feed.

But I also tried to bring it up as a client under Linux, and no dice. Apparently the 24-Bit isn't well supported yet. I found one person who had gotten it going but he reported that it sounded like crap. Very recent traffic.

I'll wait a little while before giving up on it. I suspect drivers will follow. BTW, it appears to NOT be based on the same chip set that other SB Live boards use.
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Old 12th January 2005, 15:59   #11
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Make sure the card you choose uses the EMU10k1 or 10k2 chip. This is what Linux supports very well.

I don't know if they're still around, but the SB Live 5.1 OEM for like $29-$39 at most stores is what you need.
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Old 12th January 2005, 18:12   #12
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Yeh, I should have checked. This was $29.95 and part of the Live line but it uses a different chip set. The store had some house brand (CompUSA) boards that may well have been a better choice.

I can use this one just fine for the encoder/server end and look for something else for the Linux clients.

I do have another question that should probably be a separate thread, but it has been discussed many times, yet one aspect hasn't been explained to my satisfaction.

I understand the need for "lag" or latency in the transmission for buffering of potential gaps in propagation. But I don't understand why it is all over the map. Just multiple restarts of server and client on my local network, I saw anywhere from 20 seconds to 90 seconds latency.

Whatever the needed latency for QoS, I would think it would be more predictable in a closed network situation. Do you think I should start a thread or would it be dismissed as "that's life"?

Thanks!
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Old 12th January 2005, 23:18   #13
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Definately needs a different thread, but I'd search before opening a new one... there's hundreds of them already.
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Old 12th January 2005, 23:46   #14
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Yes, I've read through quite a few of them, many of them quoting the "QoS" line. I'll probably just let it go.
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Old 17th January 2005, 08:51   #15
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What about SoundMAX Digital Audio? Does anyone know what kind of a sound card is this?
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