Is All SPDIF Audio Output the Same?

CyberSpaZtiK asks: "I am going to build a Linux audio appliance to hold my music collection in various formats and for output to my stereo system. Because of a probable lack of Linux availability or support for audio cards with high quality D/A converters and low-noise electronics (or am I mistaken?), I want to keep the output path completelely digital by using a card with SPDIF output. However, it occurs to me that I actually know very little about SPDIF - are all SPDIF outputs made equal? Can I expect every SPDIF interface to emit the exact PCM data of the source audio, or are there over/under-sampling/aliasing, etc. issues that you sometimes get with digital signal processing? What do I need to understand about SPDIF and/or other digital output interfaces to make an informed decision?"

Comments

Use TOSLINK instead

[RzUpAnmsCwrds]

SPDIF outputs are usually pretty consistent at passing the PCM data or the DD/DTS sountrack if you have them configured right.

Some cards, however, such as Creative's Audigy series, are notorious for resampling inputs/outputs, so you might want to check.

Even a cheap card, like the $15 cards on Newegg, should provide a clean output. Don't buy the garbage about "jitter" that I'm sure someone will bring up - so long as your card and cabling are operating within the specification, you won't have any problems.

Do consider TOSLINK instead, however. TOSLINK uses fiber-optics, so your audio equipment and PC are electrically isolated. This reduces the chance of creating a ground loop or introducing RF noise into your reciever/amp. Moreover, it protects your equipment in the event of an electrical mishap.

Re:Use TOSLINK instead

[sffubs]

Well, my Audigy is probably doing all kinds of crap to SPDIF audio signal, but to tell you the truth I can't find fault with it by listening. (My hearing isn't below average, before you ask :) ).

I would just like to confirm what you say about TOSLINK. My PC is currently too far from my receiver for my optical cables to stretch, so I have to use the SPDIF connection. Every time there is an electrical event in my house (heating, fridge, freezer, kettle switching) the audio cuts out for a second or so.

Re:Use TOSLINK instead

[FlexAgain]

RzUpAnmsCwrds (262647) wrote
Do consider TOSLINK instead, however. TOSLINK uses fiber-optics, so your audio equipment and PC are electrically isolated. This reduces the chance of creating a ground loop or introducing RF noise into your reciever/amp. Moreover, it protects your equipment in the event of an electrical mishap.

One slight clarification, TOSLINK normally does carry SPDIF. TOSLINK is primarily just detailing the physical medium, the data is still encoded as SPDIF (which can also be carried on wire). The original author didn't specify how he was intending to use SPDIF, it may have been over either medium.

Re:Use TOSLINK instead

[littlerubberfeet]

++ TOSLINK. I use both S/PDIF and ADAT Optical. I have had far less problems with optical connections then S/PDIF over coax/RCA cables. I do have to worry about jitter, but if you are operating a home theater/computer with only 4 or 5 digital input sources, all within the same room, you have nothing to worry about. Just use decent cables. Some problems can arise from varying sample rates and bit depth, but that is pretty easy to deal with.

Re:Use TOSLINK instead

[Devil's Avocado]

"""
One slight clarification, TOSLINK normally does carry SPDIF. TOSLINK is primarily just detailing the physical medium, the data is still encoded as SPDIF (which can also be carried on wire). The original author didn't specify how he was intending to use SPDIF, it may have been over either medium.
"""

Um, I'm pretty sure S/PDIF is also the name for the physical connector, namely the RCA plug type of digital audio connector. The data is encoded as Dolby Digital, PCM, or whatever.

Of course, I'm no stereo nut so I could be wrong.

Re:Use TOSLINK instead

[Sad Loser]

If you are using electrical connection, I read somewhere that you need to use 75 ohm cable for these digital connections - at least this is what I use and it seems to be fine.

I use video leads which are 75 ohm and cheaper than the specialist 'digital ' links they sell to hi-fi suckers, and I reckon it doesn't re-arrange too many of those pesky '0's and '1's.

I think that jitter probably is crap. It is a digital signal. Having a good quality stable clock is probably important, buy if my computer can keep all its bits in order 3,000,000,000 times a second, then it can probably do so at 44,000 times a second.

Having said that, some digital outputs definitely sound better than others. It is possible to buy secondhand DACs that the sad hifi tinfoil hats have finished with, which are very cheap and very good. I got a Meridian one for a tenth of the new price, and it is very good, driven by a standard (ASUS onboard) spdif output - ultimate cheap hifi hack.

To come back to the topic, then I have heard elsewhere that either all spdif outputs resample, or that only the expensive ones do, but I am not sure of evidence to back this up.

Re:Use TOSLINK instead

[Anonymous Coward]

...of course if you are trying to avoid wires (almost) entirely, you can try beaming your SPDIF outputs over a home 2.4 Ghz video sender pair (e.g. as sold by Radio Shack). Plug the coax SPDIF into the video input of the sender and, on the other side, pull the SPDIF off the video receiver into your stereo's amplifier. It sound's crazy, but the video sender is the only non-network wireless device with the bandwidth to pull off this trick. This is good to the limit of the video sender (less than 100 ft) and noise free...so long as you avoid 802.11b/g frequencies. It seems to be microwave immune, but will be interrupted if people walk across the line of sight. In short, you can get digital wireless sound for the cost of an analog video sender. Apparently I'm not alone: http://www.andrewkilpatrick.org/mind/spdif/

Re:Use TOSLINK instead

[gregmac]

I would just like to confirm what you say about TOSLINK. My PC is currently too far from my receiver for my optical cables to stretch, so I have to use the SPDIF connection. Every time there is an electrical event in my house (heating, fridge, freezer, kettle switching) the audio cuts out for a second or so.

This is most likely due to faulty wiring and/or a ground loop. The linked page provides a very good description of the problem.. Unfortunately, it's usually quite hard to locate a ground loop, and they are fairly common in older homes.

Re:Use TOSLINK instead

[Anonymous Coward]

Ugh, Audigy. For games, feel free to use the Audigy, but that thing WILL resample everything, which is probably not noticeable if you're going straight to speakers after, but you start getting lovely second-and-third-generation artifacts if you pass it to something ELSE that resamples.

Most games use DirectSound3D or OpenAL these days, so you shouldn't even notice if you don't have the Audigy, since they do it in software mode that's quite indistinguishable from the hardware -- audio's simply less demanding than video that way. The soundcard built into your motherboard is probably better than an audigy because of its resampling shenanigans alone.

Re:Use TOSLINK instead

[pyite]

Um, I'm pretty sure S/PDIF is also the name for the physical connector, namely the RCA plug type of digital audio connector. The data is encoded as Dolby Digital, PCM, or whatever.

Of course, I'm no stereo nut so I could be wrong.

Yes, you are wrong. S/PDIF is effectively the data format. It can run over TOSLINK or 75 ohm (read: normal coaxial video) cabling.

And of course the ubiquitous wikipedia link.

Re:Use TOSLINK instead

[alienw]

SPDIF is designed for 75 ohm video cables, so your cable should work perfectly. Jitter isn't "crap", it's there. If you don't care whether your THD is 0.01% or 0.1%, you don't need to worry about it -- and most people don't need to worry about such differences. If you want to get the best possible audio quality, you do need to consider jitter. It's not just some mythical audiophile invention, almost any datasheet for a DAC chip will have a jitter/THD graph.

In any case, I would be more worried about the soundcard doing nasty crap to the signal. Many cards are known for resampling everything to 48KHz, thus lowering the quality by an order of magnitude. They most likely to this because they can get away with a standard crystal and don't need a PLL.

Re:Use TOSLINK instead

[farnz]

Jitter may well be there, but with a digital transmission, either the data doesn't get through (in which case the sound doesn't come out), or it does. There's no inbetween state, so unless there are errors, jitter is not worth caring about.

In any case, most (if not all) equipment with SPDIF inputs reclocks the signal before doing anything (even loopthrough is reclocked). Again, so long as no link is reporting errors, you're A-OK, no matter what the level of jitter is.

Re:Use TOSLINK instead

[petermgreen]

no talk page on that wikipedia article which tells me its not to be too highly trusted

i'm sure that what i heared was that S/PDIF was the original coaxial based system and that TOSLINK was a fibre cabling system based on it that worked using the same protocol.

Re:Use TOSLINK instead

[ngrier]

Correct: the S/PDIF connection can be optical or coaxial. You connect the equipment with either TOSLINK (fiber) or "coaxial digital" cables. Really any 75 ohm cable will work. (and in reality most any cable will work if your equipment is flexible enough).

The one thing to know, though, is that the PCM data can be sent at any number of frequencies. While most amps can read at 44.1kHz and 48 (and usually lower frequencies as well), not all can. Additionally, some sound cards (particularly the more generic or on-board ones, including the nForce) will resample everything to a fixed frequency, usually 48kHz. This can be problematic if the two don't speak the same frequency. You also have to be careful about coax to optical converters (most sound cards nowadays have coax S/PDIF outputs, but many amps have no or very few coax inputs). Many can only translate sources at 44.1 kHz so you're SOL if your sound card outputs at 48kHz (yes, I'm speaking from bitter experience).

Re:Use TOSLINK instead

[justforaday]

Um, I'm pretty sure S/PDIF is also the name for the physical connector...

Admittedly, I don't know squat about this, but I'm pretty sure S/PDIF stands for something along the lines of Sony/Philips Digital Interface (my old roomies were audio engineers). It specifies how the signal is transimitted over the wire. The form the connector takes may vary.

Re:Use TOSLINK instead

[UberLame]

A proper S/PDIF over coax implementation should use isolation transformers to electrically isolate the signal, preventing groundloops. RF is still a posibility, but using proper coax cable instead of any old RCA cable will help greatly prevent problems there as well.

A card like the M-Audio Audiophile 2496 would be a respectable choice for under $100. I don't have my card handy to look at, but I think it uses a proper transformer for isolation. I don't think the creative labs cards do though. There may be some adequate USB interfaces, but I have no hands on experience with them.

Re:Use TOSLINK instead

[RedWizzard]

I've had the opposite experience: no problems with S/PDIF over coax, but trouble with S/PDIF over Toslink. Optical cables are also more expensive than coax, so I'd recommend trying coax first.

Re:Use TOSLINK instead

[pete-classic]

Since we're being pedantic, most "coaxial" A/V cables ("RCA" baseband) aren't coaxial. The connectors are, but the cables aren't.

Take a look. The wires are side-by-side.

-Peter

Re:Use TOSLINK instead

[PedanticSpellingTrol]

What the fuck are you talking about?

Re:Use TOSLINK instead

[alienw]

Equipment cannot completely reclock the signal. It can run it through a PLL, but that still lets through quite a bit of jitter (especially low-frequency jitter). You aren't going to get errors in the data regardless of jitter level because it uses Manchester coding. However, the quality of the clock you extract from the signal is quite important if you are running it to a DAC. That's where jitter starts to be important.

Re:Use TOSLINK instead

[farnz]

Bullshit. Complete and utter bullshit. In decent equipment, when you reclock, you use a PLL to lock to the existing signal, and create a binary bitstream in a buffer; you then use your own clock to pump bits out of the buffer. So long as the jitter isn't too bad, there will be bits going in often enough that the buffer never drains. All that then matters is the quality of your clock.

Jitter is an issue for equipment designers; it is not an issue for equipment users. With the aid of a decent lab, you can verify this in double-blind tests; jitter either makes no difference at all, or causes the sound to glitch.

Re:Use TOSLINK instead

[50000BTU_barbecue]

Then stop buying your cables at Dollarama. Seriously. If that's what your cables look like, they are shit. Period. The only "side-by-side" you should be seeing is two or more coaxial cables running side by side, under the same sleeve.

Look, I went the Dollarama way once, when I wanted to hook up my Commodore 64 to a LCD monitor via a scan doubler. The picture had horrible dot crawl. The cheap-ass telephone cables with s-video connectors wee the culprit. A 7$ cable with real coax fixed that problem. I'd ahte to see your TV picture if your video cables are telephone cables...

Re:Use TOSLINK instead

[Anonymous Coward]

It always amuses me that the same people who swear left and right that a mature codec like ATRAC3plus sounds like shit, will swear left and right that their POS Audigy is golden...

Re:Use TOSLINK instead

[jovlinger]

So spdif is a container, or codec?

Ie, mpeg / avi (containers) or divx / mpeg-4 / mp3 (codecs) ?

Re:Use TOSLINK instead

[pete-classic]

I wasn't even drinking when I posted this.

Weird.

Anyway, they're coaxial. I'm dumb.

-Peter

Re:Use TOSLINK instead

[Tiroth]

I can tell that you haven't read the AES whitepaper on jitter.

Re:Use TOSLINK instead

[Anonymous Coward]

And I can tell that you haven't understood it. Key point to take away: any jitter induced noise is outside the resolution of the digital stream in the first place. While in the digital domain, you can interpolate up to an arbitrary sample frequency and bits/sample to push the HF component well away from the signal, and the LF component well away too. You can then apply a band-pass filter to only let through signal, and not noise.

Jitter reduction by cabling changes the pattern of the noise, but doesn't remove it. Band-pass filtering removes it.

Re:Use TOSLINK instead

[goosman]

Which paper? Link?

Re:Use TOSLINK instead

[mink]

I think people are trying to justify Monster brand cable purchases for 3 foot optical connections. Since the main selling point and main excuse I hear from the devout followers of Monster Cable is that their plastic optic cable reduces jitter that would otherwise rape your pets if left unchecked.

Way to go

[1967mustangman]

I can't answer your question, but I will tell you one thing for sure. The lack of anything that effectively manages a music collection in Linux has long been a beef of mine. I hate to say it, but music "jukeboxes" are one area where Widows has a definate advantage.

Re:Way to go

[Anonymous Coward]

;) i will trump that with an Itunes which works far better on Mac than windows , Plus xmms has plenty of jukebox plugins and if you look there are a fair few jukebox players for linux .

Re:Way to go

[sneakers563]

Really? I've had exactly the opposite experience. I like using a web frontend to mserv because I like the random "jukebox" features, but I've seen tons and tons of other jukeboxes out there for Linux. There's jukeboxes out there whether you want a commandline, gui or web frontend, whether you want your collection indexed on the fly or stored in a database, whether you want to play the music locally or stream it, whether you want to play wav, mp3, ogg, whatever.

What are the Linux jukeboxes missing that the Windows ones have?

Re:Way to go

[1967mustangman]

The ones I have found have not seemed very good to me. Which one have you used? Because I am very open to recomendations.

Re:Way to go

[Vilim]

I am a huge fan of rhythmbox. I prefer it to iTunes any day. I use rhythmbox at home however at work I have to use iTunes (I take my ipod back and forth and plug it in at work). iTunes is alot slower, and has useless eye candy (so many animations ... so useless)

Re:Way to go

[sneakers563]

Well, I should say that I wanted to build a living room "jukebox" and DVR for parties, so my requirements might be a bit different from yours. I've used Mserv because I wanted a kiosk-type jukebox that would act like a real jukebox. That is, if no songs were selected, it would start picking songs based on ratings and how long it had been since they had last been played. I don't know of any other jukeboxes, Windows or Mac (perhaps someone can enlighten me) that will weight it's random selections like that. I wrote my own kiosk-style frontend using Python, but it appears that someone else has done the same thing with Shrill, complete with album art. I have a friend who's doing something similar with MPD, but I haven't used it myself. I've also played around with MythTV, which was nice because of the DVR features, but it didn't have the random feature that I wanted.

Re:Way to go

[1967mustangman]

That is still really cool because that is another thing I have been lloking into for a while now. Thanks for all the valuable tips.

Re:Way to go

[Anonymous Coward]

If you're a KDE user, you could do worse than checking out Amarok. It seems to be pretty well regarded by most people (I tried it out, but I'm an XMMS user by nature).

Re:Way to go

[Zeal17]

iTunes' party shuffle feature will do this. You can set it to play higher rated songs more frequently, or just use a dynamic playlist that has the features you want.

Re:Way to go

[DJProtoss]

hmmm. how about: * replaygain * proper gapless * script language based system for * tag setting & renaming * osd / title bar * main screen * support for cue files embedded in audio files [ 1 file / album, but appears as a normal set of tracks in the player ] all in the same player? (for those who don't know, i'm specifically referring to foobar2000 under windows, the one non-game program that makes me want to keep windows on my system. Amarok is getting close (I even heard rumours of someone working on replaygain) - I love the auto generating playlists w/ audioscrobbler support. But I digress :)

Re:Way to go

[sneakers563]

I can't even understand what you've written, so I'll bow to your expertise!

Re:Way to go

[Kesha]

Unless you've tried both JuK and amaroK and found them both lacking, you can not claim that there are no music "jukeboxes" on Linux.

Paul.

Re:Way to go

[bergeron76]

Dude, I think that's a perl script or something. I think you run it on the command line to get the actual output. Something like:
perl -e "[that post here]"

At least, I think that's how to turn it into English.

Re:Way to go

[DJProtoss]

damn that lack of autoformatting! too much time using phpBB forums I geuss :S. Heres it properly:
Features I would like in an audio player that (afaik) are not currently availiable under linux (or at least, are not availiable together to any degree:

1. replaygain
2. proper gapless playback support
3. A scripted language based system to for determining how the player outputs/reads:
   1. tags
   2. filenames
   3. on-screen displays / title bars / etc
   4. the actual main player window
4. support for embedding cue files in id tags (rip album as a single mp3/ogg/whatever, but it appears as a set of tracks in the player)
5. playing albums from within rar/zip files (another way of doing the above, some benefits/drawbacks by comparision)

all in the same player?
Amarok is getting close (I even heard rumours of someone working on replaygain) - I love the auto generating playlists w/ audioscrobbler support. But I digress :)

Just google for jitter.

[Kickasso]

It will occupy you for years to come.

Re:Answer

[Anonymous Coward]

1. Slashdot actually is a news site, not a support number.

Then why do they have an "Ask Slashdot" section?

Jitter is a bunch of crap

[Hidyman]

We are talking digital signals here.
Any self respecting DAC circuit will not be affected by jitter.
I use toslink all the time and there is no problem with "jitter".

Jitter is marketing hype.

Re:Jitter is a bunch of crap

[Anonymous Coward]

Since we pronounce it digital, signal fronts are ideal delta functions, the frequency is perfectly stable, and conditions for the Nyquist theorem automatically hold. Right?

Re:Jitter is a bunch of crap

[stienman]

Jitter is a problem for electrical engineers and programmers. By the time we're done with the system, you won't be able to tell whether there is any jitter or not, nevermind how much. Regardless, there WILL be jitter.

Unless, of course, all your units have synchronized clocks, or each have their own atomic clock.

Unlikely, to say the least.

Jitter is not a problem the average prosumer really needs to worry about, nevermind the average consumer.

The audiophiles who care about it care the same way about their tubes, oxygen-free cables, and green highlighters. Whatever gives you a warm fuzzy feeling, man.

But, technically, it does exist, and it is a problem that results in either doubling up on samples, skipping samples, or some sort of macabre clock synchronization scheme that only ends in tears.

Only, technically, that's not jitter either.

-Adam

Re:Jitter is a bunch of crap

[Andy_R]

Actually, Jitter is a real, audible and nasty issue in certain specific applications... but not this one.

It's not a problem you are ever likely to come across outside a big recording studio where several devices are talking to each other digitally with DAC clocks drifiting compared to each other (oh, and it's easliy solved, the solution is to slave everything to a master clock).

The problem of sending a 44.1kHz signal from one end of your house to another is trivial compared to feeding a broadband signal from your ISP to your computer, and you never have jitter issues there.

Re:Jitter is a bunch of crap

[chriso11]

I spend a lot of time dealing with jitter in my job. For example, if you want to capture a TV signal, then jitter is quite important (even more if it is HDTV). But for generating audio signals over a serial link, then jitter should not be a problem, provided the digital data is properly latched, and the clocking method is not absolute crap (jitter of 1ns on the clock would not be an issue, and that is a crappy level of jitter - it gives a SNR of ~93dB at 20KHz - jitter requirements for WLAN is more like 50ps, and you see how cheap the cards are).

What is amazing is the "high quality" TOSLINK and SPDIF cables! It is digital data folks - as long as it is not a crappy cable, then you will be fine!

Re:Jitter is a bunch of crap

[Hidyman]

Right, I meant jitter in this application.
I know jitter is real, but it's not really a factor in this instance.
I work for a WISP so I know about jitter in a real environment.

Re:Jitter is a bunch of crap

[BRonsk]

You might be interested in knowing that the SPDIF signal is carried over a signal oscillating at the sample rate. So as far as SPDIF is concerned, both devices have a synchronized clock, through the carrier of the SPDIF stream.

What is it you were saying about jitter again?

Re:Jitter is a bunch of crap

[blacksmith]

SPDIF does indeed exist on such a signal. However, the very fact you're sending information is going to mean you don't get a simple clock signal, and you've got to somehow get that clock back with a PLL or somesuch. If your PLL isn't perfect, you're going to get drifts in your recovered clock, hence jitter. Whether you can hear it is another issue....

$25 TOSLINK card

[hab136]

I've been using a Chaintech AV-710 with my linux home theater PC for a long time now (a year?), outputs to my surround sound receiver. Fully supported under ALSA. mplayer, xine, and ogle all pass through the AC3 5.1 sound for my receiver to decode. I went for fiber optic, mainly because I didn't want to worry about grounding effects.

Chaintech's product page

Re:$25 TOSLINK card

[pete-classic]

Does it work with DTS?

I'm trying to find an inexpensive solution for S/PDIF over TOSLINK for Linux that supports AC3 and DTS, but my messages don't make it to the ALSA list for some reason.

Maybe it's because too many words are all caps!

-Peter

Re:$25 TOSLINK card

[hab136]

Does it work with DTS?

In my setup (playing DVDs), the AC3 audio is sent straight to the receiver. The receiver does the Dobly/DTS/THX/whatever decoding. So yes, it works with DTS, since Linux is just shoveling data off the disk and then onto the wire, no matter the encoding.

Here's my /etc/asound.conf where everything goes out the optical out:

pcm.!default {
type plug
slave.pcm "cards.pcm.iec958"
}

pcm.!spdif {
type plug
slave.pcm "cards.pcm.iec958"
}

pcm.!iec958 {
type plug
slave {
pcm "hw:0,1"
format S32_LE
}
}

(slashdot ate my indenting, but it'll still work)

There are Consumer and Pro modes

[Marillion]

The S/PDIF protocol has a consumer mode and a professional mode. I do some professional audio work and my DiO-2496 will emit both. My MD player will only accept the consumer mode which includes Serial Copy Management System (SCMS) flags which indicates if the source is first generation (allowed) or second generation (not allowed). The other nice thing about this card, it is completely ignores inbound SCMS and can re-code a second generation stream as a first generation consumer stream or a professional stream. Haven't needed it, but cool. I've connected it to professional DAT units, consumer MD units and DVD players.

Re:There are Consumer and Pro modes

[Anonymous Coward]

The DiO-2496 is discontinued ... do later models also ignore SCMS?

Re:There are Consumer and Pro modes

[Marillion]

The DiO-2496 is discontinued ... do later models also ignore SCMS?

The Delta-66 is based upon the same chipset and the website claims SCMS control.

It doesn't have optical S/PDIF, only coaxial, which for me is a royal !#$%. There are converters. A friend of mine has this card and says the analogue ports are very clean and free from a lot of noise.

Re:There are Consumer and Pro modes

[isorox]

You mean AES/EBU? Yes, the 2 bit copyprotect flag isn't in that spec, although as you say you can get a device to change a normal SPDIF stream.

IIRC there are 4 settings for copy-protect: prohibit, don't prohibit, and one generation (the other setting is unused). It repeats every frame.

However that's only a minor part of the difference. The big difference between professional and consumer kit is pro kit is ballanced, additionally AES/EBU can support 3 frquencies (44.1, 48 and 96), I believe SPDIFF is limited to 44.1. Oh, and AES/EBU kit can have an external reference.

MAudio Delta 44

[medgooroo]

Your mistaken. the MAudio cards all work beautifully... and sound it too... *currently listening to music in superb definition on said card* together with jack and assorted other toys, audio on linux is not something that is lagging.

Re:MAudio Delta 44

[a whoabot]

This man speaks the truth.

Most M-Audio cards work with Linux ALSA and JACK. If you just want some decent audio output you can buy the Audiophile 24/96 for less than $100 at the store. It has SPDIF out as well.

The Mia card by Echo works as well.

RME has soundcards that work well with Linux too. They will get you some higher quality at a price.

Re:MAudio Delta 44

[sneakers563]

I have a MAudio Dio 2496 that I have a problem with: the digital outs (both coax and optical) cut off the first half-second or so of any sounds, including songs. The analog outputs, however, work beautifully. I've had this card in 2 or 3 different Linux installations and everytime it's the same thing. I don't think it's my receiver because my dvd/cd player is also hooked up to a digital input on the receiver and it doesn't show the same behavior.

Consider Griffin iMic

[HeelToe]

If digital inputs are at a premium, consider a Griffin iMic - works perfectly with the ALSA USB Audio driver.

In my testing, it's the cleanest sound I've heard from a computer with the exception of optical toslink.

Re:Consider Griffin iMic

[YrWrstNtmr]

I have one of these, and I love it. Currently, I use it mostly for recording my vinyl LP collection, but also just audio playback. The stereo stack sees the laptop as just another tape deck (Play/Record).

Re:Answer

[IKillYou]

Hey shithead - Slashdot discussions answer plenty of questions like this. YOUR answer is entirely useless, and poorly worded to boot.

"read errors and the like" - Brilliant, thanks. Why don't you post under your member name so I can come kick your ass.

One thing to be aware of!

[amliebsch]

I didn't realize this before going to SPDIF: as far as I know, there are no sound cards and only one chipset that will output more than 2 channels through the digital link. Even if the card supports 5.1 surround by analog jacks, e.g., the SB Audigy, it will not encode your digital signal in anything other than 2-channel PCM; except when you are directly passing it raw AC3 or DTS digital data (say, from a DVD or an AC3 encoded file.) You will not be able to get, for example, surround sound over SPDIF from games that support multi-channel surround sound.

If anybody know of sound cards available for purchase that actually support this, (the feature is called DICE), let me know.

Re:One thing to be aware of!

[strstrep]

You could try Xitel's Pro Hi-Fi Link. It's a bit pricey, but it connects via the standard USB Audio protocol to your computer. It also includes all the audio cables you need. I believe it supports passing through stuff like AC3.

Re:One thing to be aware of!

[amliebsch]

I don't think that will do it. It only says it supports AC3 and DTS pass-through. I'm looking for an audio device that will encode multi-channel sound input in AC3 or DTS. Old nForce2 boards had a technology called "soundstorm" which did just that - but that technology appears to be dead and buried. Without it, it seems impossible to use the SPDIF link for things like multichannel games.

Re:One thing to be aware of!

[bartle]

Even if the card supports 5.1 surround by analog jacks, e.g., the SB Audigy, it will not encode your digital signal in anything other than 2-channel PCM; except when you are directly passing it raw AC3 or DTS digital data (say, from a DVD or an AC3 encoded file.)

Annoying ain't it? I know more than one person who would love to be able to buy a multichannel soundcard that did realtime AC3 encoding. I believe the now defunct nVidia motherboard was the only way you could do get this type output in a PC and I suspect that only came into existance because they needed the technology for the XBox.

Realtime AC3/DTS encoding will probably be in all three of the next generation consoles but there's nothing on the horizon that I've heard of for PCs. I'm starting to wonder if it's a conspiracy, if the powers that be have decided that no one should want to hook their PC directly up to their entertainment system

Re:One thing to be aware of!

[Anonymous Coward]

It's worse than that I have had a Cambridge Soundworks DTT2500 5.1 surround for my computer for several years. IT has never worked right for games. I have been pulling my hair out for years regarding this. THe speakers are great movies sound awesome but the stinking SDIF does not work for games. AAAAAAHHHHH!!!! I would buy a card that did this as soon as it hit the shelves.

Re:One thing to be aware of!

[David Horn]

I disagree. My Audigy will output a 5.1 signal via SPDIF to my surround sound amp for decoding. Works fine. Alternatively, the card will just send a Dolby Digital stream to the amp which it also handles fine.

Re:One thing to be aware of!

[amliebsch]

are you talking about any sound other than during DVD playback? How did you get this to work?

Re:One thing to be aware of!

[David Horn]

Erm - in the Creative settings, I just turned on digital output and told it I had a 5.1 setup. The amp (a Creative something-or-other) is set to default mode and I know it works, because I get surround sound in games.

I wish I could give you more detail but the computer in question is 400 miles away.

Re:One thing to be aware of!

[HalfFlat]

It's not hardware, but: real-time AC3 encoding for JACK might do the trick!

Of course, not so useful for playing games under Windows.

Re:One thing to be aware of!

[DeadMeat (TM)]

Ask and ye shall receive.

Re:One thing to be aware of!

[amliebsch]

If this works, it must be by some proprietary Creative flimflammery. I believe some of their speaker sets use a multi-conductor DIN which actually transmits multiple two-channel PCM signals simultaneously. Obviously, this is not "real" multichannel SPDIF and will not work with your average digital receiver.

I have tried everything I can think of to get 5.1 AC3 or DTS over regualr RCA coaxial SPDIF, to no avail. Everything I've found leads me to believe it is impossible.

Re:One thing to be aware of!

[jameslore]

Sorry, but the grand-parent is correct. The Audigy will not mix DD5.1 on the fly - only nVidia's nForce did this (and it was fantastic).

The Creative cards will, however, use CMSS (another Creative invention) to upmix a 2 channel source to 5.1 when used with certain Creative amps. Alternately, there's always Dolby Pro Logic.

Re:One thing to be aware of!

[amliebsch]

Super awesome terrific! Thank you!

Re:One thing to be aware of!

[aderusha]

Unfortunately this is incorrect. It may sound like surround sound, it may even be Pro Logic (which is an analog hack to get surround sound out of a 2 channel source), but nVidia holds the only license Dolby ever sold for realtime Dolby Digital 5.1 encoding hardware. If you don't have an nForce-based motherboard, you aren't doing real Dolby Digital 5.1 through SPDIF.

Re:One thing to be aware of!

[BRonsk]

If you are running on Windows, Just try to use ac3filter (sourceforge project). It does encode AC3 realtime (v1+ at least)

Re:One thing to be aware of!

[user32.ExitWindowsEx]

www.bluegears.com -- the only DD Live on a PCI card...imported from Korea.

SP/DIF chips

[leighklotz]

I'm looking for a low-cost DAC/ADC chip for SP/DIF, something that takes audio and produces SP/DIF, and vice versa. If it can use fixed modes and doesn't require a uC that would be great.

The Phillips UDA1355H looks like what I want, but Phillips doesn't even list availability information, and DigiKey and Mouser say either nothing or non-stock, which leads me to think that the chip doesn't exist.

Does anybody have anything like this?

I already know about PCM2902 USB DAC project, and while that's useful (similar to the Griffin iMic) it's the opposite of what I want.

Re:SP/DIF chips

[Anonymous Coward]

Intriguing... the status implies that only samples are available, typical if hasn't been released yet (which is quite possible).

The tinfoil hat crowd may wish to speculate on whether they are holding back production until they decide whether it should have "analog hole" security.

Echo card

[ikekrull]

My (old but still good) Echo Gina works with Linux, but needs some work to get drivers functioning, and doesnt work with the standard ALSA mixer API (has separate mixer app).

It does 20bit 2-in/8-out recording/playback with a very low noise floor, and also offers stereo digital in/out with SP/DIF consumer/pro modes on Linux.

My friend gave me his since Echo no longer do Windows drivers for their old cards.

Re:One thing to be aware of! (Ditto)

[Anonymous Coward]

Ditto on this comment. I bought an external USB device for the digital output to hook up to an old laptop I thought to use for stereo input.

Surprise to me: no 5.1 output on the digital out. Cheap bastards -- I guess that would involve licensing Dolby encoding from Dolby labs.

Interestingly enough, a newer laptop (Dell Inspiron 8000) has a digital output for use when watching DVD's. However, it only works with DVDs, not wave or CD audio output. Was hoping for CD->SPDIF->Stereo PCM, but nada....only analog (with a known problem of audio noise on the analog outputs). I don't know about their newer laptops, but Dell Support basically waived their hands and, basically, said too bad so sad -- must be in your cables (of multiple users complaining in their online forum). The digital output signal doesn't suffer from the same problem.

Would be nice to know which cards can encode Surround, 5.1 or DTS onto a digital out.

I want multi-channel PCM

[Anonymous Coward]

Is there a (cheap) card which will send 6 channels of linear PCM out of the TOSLINK/coax? I want to use the DACs on my receiver, rather than the ones on the sound card. I *don't* want real-time encoding to a lossy compressed format like DD or DTS.

Well, actually I want to build an active speaker system, and those 6 channels are actually brutefir-generated for each individual driver from a standard 2-channel stereo track.

Any ideas?

SPDIF -- not all are equal

[crazy_monkey]

Can I expect every SPDIF interface to emit the exact PCM data of the source audio, or are there over/under-sampling/aliasing, etc. issues that you sometimes get with digital signal processing?

At least according to this site, no. See " 44 KHz Digital Data To Digital Output" sections such as Turtle Beach Santa Cruz. A full list of tested cards is here Here

Good luck, or buy a squeezebox from slimp3

[tetrode]

If this makes you tick, build it, but there are easier options, such as the squeezebox from slimp3

Mark

No, but Yes if you have the good DAC

[claudebbg]

Of course there are differences between good and bad SPDIF outputs (good and bad systems with SPDIF outputs to be precise). The impedance, the connectors, the regularity of the data output, the jitter... suposing the system won't resample the datas.
Concerning your computer, I don't think it would have any problem in forwarding data at the right rate.
Avoid too much cpu-intensive tasks when listening your music.

People talk about jitter and it's interesting because it mainly affects only the end segment, the DAC, because before it you can do all the crap you want.
If you remember to have a good DAC that will just put the data back to normal (I mean jitter-less).

An amazing analysis can be found here (not so far from your setup) Apple AirPort Express Wi-Fi Hub-D/A processor.
This guy uses a "bad" setup ending in 0% (yes, 0) data lost. And with a reclocking DAC with less jitter than a multi-thousands $ system.

I may recommend you the Mini-Dac if it's in your budget (not so expensive for high-end systems, but expensive compared to $99 stuff).

Re:Answer

[Slime-dogg]

Yes.

Slashdot: News for nerds, stuff that matters.

That second part encompasses any discussion relevant to "mattering." This includes functioning as a support number.

Jitter

[Anonymous Coward]

Jitter exists but it unimportant when you use a digital buffer. That is, if you are reading more than 1 bit, and you are buffering at least 1 bit the jitter suffered in the signal will not be noticable.

Also be aware that audiophiles never use actual measurements to describe how something sounds, they have subjective and emotional vocabularies to describe the experience. This is because audiophilia is much like a religon, it cares little for evidence.

on spdif jitter

[TheGratefulNet]

way back when I was on the 'dat-heads' mailing list (back when DAT taping at live shows was still new and not at all common), there was quite a bit of talk about jitter and its real world effects.

without getting deep in math and tech, the short answer that everyone seemed to agree upon was:

- when sending the signal from a source device to PLAYBACK device, if the target device is a DAC, then jitter does matter.

- when COPYING the data from a device to a storage device (DAT, computer, etc), then jitter does not matter ON THAT LINK. it only matters when the last link goes from player->DAC.

and this is to be tempered by the quality of the DAC. early DACs didn't extract clocking info (timing) from the signal as well. pro audio uses a separate clock line from data line (smarter that way). consumer i/o does not, it has to extract clock from the data stream itself.

any modern DAC should not have jitter problems due to minor diffs in cable quality. I know a lot of people who swore that ATT glass fiber was AUDIBLY better than cheap toslink plastic fiber. bollocks, is all I can say, to that one.

resampling by internal architecture

[TheGratefulNet]

one thing to watch out for, in spdif output cards, is that MANY internally resample ALL data at 48k. this will alter the bitstream of a true 44.1 file if you try to diff it of what you see on the wire vs what is on disk.

sound blasters and their ilk are famous for this.

the envy24 chip is known to be bit-accurate. what you send is what you get. m-audio has these cards.

another good 'musician quality' (bit accurate) card is one that uses the 8738 chip. its cheap and very common.

I think it was M$ (I may be wrong) who first defined the internal audio architecture to be 48k sample rate. not sure if this was to foil those who cared about bit-accurate 44.1 content. it seems suspicious to me. ie, "lets not let consumers get direct bit-for-bit access to 44.1 streams. let them get digital, but a RESAMPLED digital, so our copyright folks 'feel' better". I may be totally off, here, though.

at any rate, assume your cheapie card WILL be a 48k style card. unless you know for sure its bit-accurate.

one place to check is an old time vendor of digital audio and taping supplies: http://www.core-sound.com/

they have 'pda audio' stuff that is supposed to be bit-accurate.

Benchmark DAC-1

[TibbonZero]

I'd pick up a Benchmark DAC-1 instead. The output is much tighter than the MiniDAC from Apoggee. That is unless you are running with a Big Ben, but even then, the clocking on the DAC-1 is SUPER tight.