 Ethernet vs. Digital Audio Out
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So I'm curious if anyone can tell me if there is going to be an audible difference between using the digital out of a soundcard (say a decent $100 card from M-Audio or SoundBlaster) to the digital in of a receiver vs. using a receiver with an Ethernet port and adding that receiver as a client (such as with http://www.yamaha.com/yec/products/productdetail.html?CNTID=451033&CTID= ) The soundcard itself will be the biggest wildcard, I suppose, which is a point in the favor of using a network connection...but I don't understand the fine points of information transfer over coaxial/optical audio lines vs. raw data transfer over CAT5 copper. Or whatever. I'm seeing more and more of these LAN connections on receivers popping up on cheaper and cheaper models, and I'm assuming in a few years time it will be a standard 'input'. So at some point it will no longer be a matter of whether or not I want to spend the money for the feature--since *both* will be standard--rather a question of which is better. Right?
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Re: Ethernet vs. Digital Audio Out
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Optical carries a digital signal (zeros and ones). Coax can carry digital or analog signals. Cat5 can also carry both and some people use it for more than computer networks (digital). It has been used as speaker cable connections (analog) as well.
Digital signals will carry the most true set of information as it is hard to degrade except over long distances. Analog signals are necessary for many applications (e.g. speakers) and will be perfectly fine as long as there are no significant sources of outside influence (e.g. do not place strong magnets or power boxes near analog wires in use).
"Those who preach the myths of audio are ignorant of truth."
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Re: Ethernet vs. Digital Audio Out
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I believe the ethernet jack on newer receivers are for the purpose of making the unit part of a "digital home entertainment" setup. In other words, you can play mp3 files stored on a computer or play streaming internet radio directly through your receiver. In other words, straight data.
Coax is used to send either digital (data) or analog signals between components. It acts as a point-to-point interconnect. Fiber does the same, but is digital only. I don't see Cat-5/5e becoming an alernate standard for interconnects per se, but I do see them becoming a standard feature on more and more devices.
Let's say as time goes on more devices are embedded with chips that can decode digital audio signals. Say your CD player has ethernet capability. You set it to bypass its DAC and send a compressed digital stream onto your network. Now any device connected to your network can pick up the stream, decode it, and play it. So you could have little "mini-receivers" all over the house, in the back yard, etc. Or even more likely, powered speakers that take the signals and play them back directly. My guess is you will start to see wireless ethernet antennas sticking out of receivers in the near future as well.
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Re: Ethernet vs. Digital Audio Out
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I looked at both building a MCPC VS purchasing various "MediaPlayer" (very tempted by www.z500series.com). I went for the MCPC in the end (Win XP with Nero Home) using S/PDIF(will go for HDMI later) for sound to a Yami Amp (V2600) due to the flexibility you get from software. I can play almost anything on the MCPC and away it goes. If you decide to use the RX-N600 as a combined MediaPlayer / Amp you may be disappointed in the future as its firmware only supports playing MP3, WMA and WAV files from a server over E'Net. So you WILL NOT be able to stream other files types including MPEG's etc over ethernet. You will also still need a PC to stream the data from. You should not notice a quality difference as both Ethernet and S/PDIF are just a digital transport. You will use either: 1) A PC Software player's CODEC is used to covert MP3's / WMA's to PCM or extract the DD / DTS and transmit these data streams over S/PDIF where the Yami will decode and amplifiy it. 2) The Yami acts as a fancy MP3 player by reading (streaming) the MP3's / WMA's from a network attached drive where it's CODEC does the conversion then amplificaiton. The sound quality is determined by the "quality"/brand of CODEC used but I doubt you will be also to tell the difference. I've found that compressed music (eg MP3's) sound poor in comparison to the original CD's on the Axioms. I am now using WMA Lossless as my CODEC and can not tell the difference! Thanks Nathan
Last edited by jmone; 08/15/06 07:07 AM.
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Re: Ethernet vs. Digital Audio Out
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Yes, they are both a way of transporting data. The difference is that ethernet is a networking standard that uses a specific scheme of timer cycles and carrier detect sensing to send packets of data across a wire while not interrupting transmissions from other nodes on the network. The listed 2-meter minimum/100 meter maximum distances for copper-based ethernet node distances are a careful balance of distance needed for 1 timer cycle (1 meter), capabilities of the copper transport, and distance covered before timeout (100 meters). Not the best way to do high-bitrate point to point data transfers. SPDIF is a high-bitrate digital point to point connection. It doesn't have or need the overhead necessary to run on a network. With SPDIF you don't have frames, headers, carrier sensing, collisions detection, etc. Just the 1s and 0s.
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Re: Ethernet vs. Digital Audio Out
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But do those added features necessary for networking really degrade a digital signal? I think if it is digital and the transport process doesn't compress the information it will sound the exact same as it would through any other type of cable. That is one of the major benefits to a digital signal and to take full advantage of it you should convert to analog at the last possible point in the system, thus minimizing chances of noise getting into the signal.
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Re: Ethernet vs. Digital Audio Out
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I imagine that it kind of depends on how the data is encapsulated. Is it using TCP? UDP?
I mean, if they can do phone calls over TCP/IP (using UDP), they may well be able to do audio just fine.
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Re: Ethernet vs. Digital Audio Out
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Using UDP, yes, but with the QoS bit flipped. This gives Voip priority on the wire. Which in turn could squash an audio stream. But I guess most home networks can handle uni- or multicast streams ok. But you're still dealing with compressed data instead of a full-width digital data stream. Unless you mean sending the data uncompressed? Not sure too sure about that. Remember the ethernet nodes listen to the wire for traffic before sending packets out. You might get some gaps. Plus you would be flooding the wire to begin with. To reiterate, data sent over ethernet is sent in discrete packets with all the associated overhead. Digital interconnects send raw data, just a raw stream of 1's and 0's with no excess information. I just don't know that ethernet is the most efficient way to do this uncompressed.
Last edited by LightninJoe; 08/26/06 04:16 AM.
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Re: Ethernet vs. Digital Audio Out
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You sound like a man who knows that "all people seem to need delicious pizza". Or perhaps that "people don't need to see Prince's ass".
(somebody besides Ken will get this, I'm sure...)
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Re: Ethernet vs. Digital Audio Out
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While it's true that all people seem to need delicious pizza and doubly true that people don't need to see Prince's ass I'm not sure what you're getting at.
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Re: Ethernet vs. Digital Audio Out
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Obviously someone hasn't cold cocked Neil Armstrong.
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Re: Ethernet vs. Digital Audio Out
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It's a mnenonic for something to do with network layering.
Like Every Good Boy Deserves Fudge for the 5 staff lines of the treble clef or Bad Boys Rape Our Young Girls But Violet Goes Willingly to remember resistor colour band values (yes, I was actually taught a more racist version of that in grade 10 electronics).
Bren R.
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Re: Ethernet vs. Digital Audio Out
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Apparently then, Violet was a favorite of the good boys(even better than fudge).
-----------------------------------
Enjoy the music, not the equipment.
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Re: Ethernet vs. Digital Audio Out
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Mmmmm you may want to read http://www.epanorama.net/documents/audio/spdif.htmlFor those interested in the data structure used for S/PDIF, it can transmite either audio sample (PCM) or data bits (DD / DTS) in an encapsulated 32-bit word (subframe) - as follows: bits meaning ------------------------------------------------- 0-3 Preamble (see above; special structure) 4-7 Auxillary-audio-databits 8-27 Sample (A 24-bit sample can be used (using bits 4-27). A CD-player uses only 16 bits, so only bits 13 (LSB) to 27 (MSB) are used. Bits 4-12 are set to 0). 28 Validity (When this bit is set, the sample should not be used by the receiver. A CD-player uses the 'error-flag' to set this bit). 29 Subcode-data 30 Channel-status-information 31 Parity (bit 0-3 are not included Seriously though, the real question is where you want to decode the data, and what devices you have that can do the required tasks – the transport layer is almost irrelevant. Both Ethernet and S/PDIF have the bandwidth required to transport DATA STREAMS required between devices - it is much more important to understnad what device in the chain has the CODEC required to convert it to an Analogue signal. Typically a main steam Receiver (to date) handles the decoding and amplification of PCM, DD, DTS, and MPEG-Audio data streams transmitted over S/PDIF. Support is now appearing in these moderate priced Receivers that can decode additional formats (eg MP3/WMA/SACD/DVD-A) as well as accept additional transport layers which these formats can be transmitted over (eg Digital Radio, Ethernet both in wired and unwired formats,USB etc). Nathan
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Re: Ethernet vs. Digital Audio Out
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Re: Ethernet vs. Digital Audio Out
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Re: Ethernet vs. Digital Audio Out
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Bren was correct. My two mnemonics were for remembering the parts of the 7 layer OSI model, which has to do with computer networking:
Physical, Datalink, Network, Transport, Session, Presentation, Application
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Re: Ethernet vs. Digital Audio Out
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Quote:
Physical, Datalink, Network, Transport, Session, Presentation, Application
Which are also the 7 layers of flirting for me.
Bren R.
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Re: Ethernet vs. Digital Audio Out
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Quote:
It's a mnenonic for something to do with network layering.
Like Every Good Boy Deserves Fudge for the 5 staff lines of the treble clef or Bad Boys Rape Our Young Girls But Violet Goes Willingly to remember resistor colour band values (yes, I was actually taught a more racist version of that in grade 10 electronics).
Bren R.
Shizzle. All Cars Eat Gas. Funny thing is the pizza one started making neurons fizzle a little in the deep dark recesses.
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Re: Ethernet vs. Digital Audio Out
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-----------------------------------
Enjoy the music, not the equipment.
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Re: Ethernet vs. Digital Audio Out
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The use of mnemonics has never been a big part of my studying process. I guess my "damn the torpedoes, full speed ahead" thought matrix has precluded this. But I do recollect the previously mentioned pizza line from early studies of the OSI model, now that the fellas brought it to light again.
EDIT: Criminy, gotta supress the subconscious I guess. I didn't mean to use the words "Mnemonic" and "Matrix" in one sentence. You know what I'm referring to.
Last edited by LightninJoe; 08/28/06 03:14 AM.
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Re: Ethernet vs. Digital Audio Out
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Quote:
I didn't mean to use the words "Mnemonic" and "Matrix" in one sentence. You know what I'm referring to.
And that's totally righteous, dude! *wheedle wheedle wheedledeeee*
Bren R.
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Re: Ethernet vs. Digital Audio Out
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Re: Ethernet vs. Digital Audio Out
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Hey I just wanted to thank you all for your responses. This was just the type of discussion and answers I was hoping for. I had a feeling that there'd be little if any difference in sound but figured I'd throw it at the experts and see for sure.
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Re: Ethernet vs. Digital Audio Out
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I'm in the same boat. Here are my assumptions.
If you have a receiver with an ethernet port, then you can send the signal direct from the PC to the receiver. Easy enough. I would however need a router so that I can also connect my cable modem to the PC.
If your receiver does not have an ethernet port, you can use a sound card with a digital output. The quality of the sound to the receiver depends on the quality of the sound card. All sound cards are known to make a popping sound and I can't imagine that to be pleasing to hear nor good for the speakers. Basically, it's not a direct route because the data either passes or gets filtered through the sound card. I'm not sure if the data gets passed or filtered, but if it gets filtered, I assume there might be some loss in quality. I'm testing this option right now and the sound is almost as good as a cd/dvd player. I'm trying different sound cards to see if I can get it to sound as good as my dvd player.
I've been told another option would be to get an ethernet to optical cable adapter. I'm not sure if there's a loss in quality, but I can't imagine it would be much. I plan on testing this route too. It's hard to find such an adapter, but I was told the Apple Airport Express does the job. I guess I would also need a wireless router with this route.
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Re: Ethernet vs. Digital Audio Out
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Hi,
I've got a HTPC with a sound card & a V2700 that also has a Ethernet port. I currently have 3 connection options:
1) S/PDIF: This is my prefred option (no popping). The HTPC can pass via the S/PDIF connection to the Reciever all the current major audio specs including PCM, DTS, DD in perfect digital quality which are then decoded by the Receiver.
2) Analoge Audio - I "could" use the cheapo audio card to decode the audio streams and output each channel to the Receiver but I would only ever look at this option if my Receiver could no longer decode the digital audio stream from the S/PDIF connection.
3) Ethernet on the Receiver: The limitation is that the receiver has to provide the Interface as well as the decoding. While I "could" user this "feature" I'd would then be limited to what could be decoded (eg MP3/WPA) and would still need my current conneciton over S/PDIF to play DVD's and anything with DD/DTS tracks. For me why bother - I'll push it all over the S/PDIF IF.
4) In the future I'll probably replace my Video card with an HDMI one and route Audio over this as well......
Nathan
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Re: Ethernet vs. Digital Audio Out
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local
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hello
you guys mention " a good soundcard".
I've been wondering, if you are using spdif optical connection, is the quality of the soundcard that important ??
granted, I am sure there are soundcards that cost several hundred dollars which would have better chips, but would there be any sound quality difference between say, an onboard soundcard with spdif (nvidia 3/4 chipset) compared to a soundblaster audigy flavor or the month ??
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Re: Ethernet vs. Digital Audio Out
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Great Q Randy....I've used a couple of cards (one old, one new, both at the cheaper end) and while I did not hear any difference between them I have no imperical evidence that there is one. I am "guessing" that while you could get dodgy cards most should use little reprossing for the transmition of the digital signal of S/PDIF so should be the same. The only way would be to compare each of the S/PDIF streams to check for differences....way beyond me!
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Re: Ethernet vs. Digital Audio Out
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This actually opens up a whole new can of worms. Not all soundcards are equal, not all s/pdif is equal. The sound cards most definitely color the sound and it's not as simple as taking a wav file and just sending the same (amongst different sound cards) digital data over the S/PDIF. Analog output from a sound card is usually really bad since PC's in general have very bad shielding. The next issue is the DAC. This is what converts the digital audio into analog (drivers are analog so there has to be a DAC somewhere in the pathway). Not all DAC's are equal. If you look at the different squeezebox offerings, their highest end box (Transporter $1999) claims to have a high quality DAC (AKM AK4396 Multi-bit delta-sigma digital to analog converter). Can you really hear a difference? I don't know. My guess is the differences are subtle and it will depend on the type of music you are listening to. If you pop over to www.head-fi.org you will find people that claim to hear differences with cotton shielded silver wire. . . and further differences after the wire has been broken in. The next question is what quality audio files are you listening to? If you are going to be playing mp3's that are encoded with anything less than lame -V 0 (256 vbr) then S/PDIF is probably good enough. With audiophile grade speakers however, you really should be using FLAC,APE or WAV. The Yamaha only supports mp3 and WAV, however the squeezebox does support FLAC. I'm pretty much in the same boat as you. I have most of my collection ripped to FLAC and APE. So do I go for the Yamaha RX-V2700 and play mp3's or WAV, the $249 Squeezebox or perhaps get the $1999 transporter.
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Re: Ethernet vs. Digital Audio Out
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Hello thanks for the reply Yea I agree the analog output varies widely. Besides the quality, the popping sound when you turn them on sometimes is a real pain. about the DAC. I have my linux apps setup to pass the audio out the spdif. In some apps it is called ac3passthru and in some I just call out the hardware location of the spdif. I am using the Alsa sound subsystem. I just assumed that a DAC was not used in this case. I cant imagine why one would be utilized. By the same token I was thinking the dac on the squeezebox would only be utilized if you were using the analog output of the squeezebox. If you have some quick links to clarify this, I'd love to see them. this is something i havent researched, just assumed.  Now I looked at a couple of reference docs and maybe even on spdif output the mixing is done in analog ? I am hoping I am mistaken. Seems sorta silly, but my knoweldge is limited. My nvidia motherboard has a realtec ALC850 chip in it for audio. link to ALC850. Most of this soars over my head. Cant really comment on the cotton shielded silver wire, I just go middle of the road on that kind of stuff. Just ordered some HDMI cable from monoprice.com, and felt guilty they were so cheap. I have a large mp3 collection that unfortunately is at 128kb. back in the day i just had small computer speakers, and couldnt hear a difference and stored all my stuf that way. Now I am using FLAC for all new stuff, and actually using lala.com to retrieve my old favorites to facilitate reripping in FLAC. Using the spdif out is new to me, so thanks for discussing it with me. Randy ps. my daughters love the little smiley faces on here. the 4 year old calls em smiley stickers Quote:
This actually opens up a whole new can of worms.
The next issue is the DAC. This is what converts the digital audio into analog (drivers are analog so there has to be a DAC somewhere in the pathway).
If you pop over to www.head-fi.org you will find people that claim to hear differences with cotton shielded silver wire. . . and further differences after the wire has been broken in.
The next question is what quality audio files are you listening to?
I'm pretty much in the same boat as you. I have most of my collection ripped to FLAC and APE. So do I go for the Yamaha RX-V2700 and play mp3's or WAV, the $249 Squeezebox or perhaps get the $1999 transporter.
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Re: Ethernet vs. Digital Audio Out
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You guys sound like this is so complex. I have been using my soundcard to play my FLAC music for a couple of years and have had no problems or sound issues, popping, etc. at all. I am using a chaintek 710 card and using the digital optical out to my receiver and it sounds just like my cd player (which is now is the bedroom). I am also using the ASIO plugin which I'm told bypasses all the stuff that can degrade the sound? I don't know how it works, but it does sound great. What it does do is it plays music in 44.1 instead of 48k which is like the difference between cd sound quality and the sound quality of the music channels on cable. I'm using winamp as my player and I have all of the commands (play, pause, next, random, playlist toggle, etc.) programmed into my main system remote. One remote (to rule them all - sorry  ) controls everything in my system; I couldn't be happier.
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Re: Ethernet vs. Digital Audio Out
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>>What it does do is it plays music in 44.1 instead of 48k
That seems like an important point. I remember shopping for sound cards a few years ago and finding (IIRC) that many sound blasters could not run at 44.1 KHz and so all of the audio had to be resampled to 48 KHz sampling rate, presumably in software. This seemed like something to avoid if possible, unless everyone feels that the resampling algorithms are so well understood these days that the loss of quality would be un-noticeable -- or if the s/pdif protocol allowed 44.1 KHz samples to be bursted out at 48 KHz with periodic gaps and then automatically retimed to 44.1 KHz on the receiving end.
Any idea which chipset integrated audio subsystems can output s/pdif at 44.1 KHz, or if this is even something to worry about ?
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Re: Ethernet vs. Digital Audio Out
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I could tell the difference between 48 and 44.1. It wasn't huge, but it was enough that I would still be using my cd player if I couldn't get 44.1. I don't know what all soundcards can get 44.1, but my card (the chaintek) was only 25 bucks or something close to that.
I will say that it was a pain to get it to switch to 44.1. I had to closely follow a thread over at avsforum that explained it step by step using the ASIO plugin. I didn't work a few times, but eventually I got it and it's been good to go ever since.
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Re: Ethernet vs. Digital Audio Out
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Hello
the Popping sound i mentioned is only on analog output.
anyway
I just looked it up and the spdif on my mobo is 48khz. Never thought about that before. I'll have to see what the reciever says it is getting.
I guess what I would really like to know is:
If I am listening to a Flac file outputed through the spdif,
does the stream stay digital in the computer,
or is it converted to analog for the mixer function then reconverted back to digital to be outputed through the spdif.
I guess this would determine does the quality of the dac matter to me in my application ?
concerning the resampling from 44.1 to 48khz, I am hearing you say the quality was worse ? Makes me wonder if I should rip my Flac files at 48khz ?
Randy
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Re: Ethernet vs. Digital Audio Out
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I'm fairly certain that most audio cards will convert, mix and then output in digital on the spdif.
You might want to try to abx a few FLACS with the CD. I use a creative labs X-FI and even with all the effects/equalization etc disabled it doesn't sound as good as through my stereo (listening to both via ES500 IEM's.)
As far as the DAC is concerned, I think if you used the digital out on a sqeezebox, it would be bypassing the DAC in the sqeezebox and relying on the one in your receiver. I suppose that might warrant an ABX test to see which DAC you like better (assuming there is an audible difference).
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Re: Ethernet vs. Digital Audio Out
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Here's a link to the squeezebox forums which basically has a few replies regarding pc sound cards compared to a squeezebox. http://forums.slimdevices.com/showthread.php?t=30504
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Re: Ethernet vs. Digital Audio Out
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I did some more research into the transporter/squeezebox/spidf issues. The following page has some pretty good info as well as additional links to pages that discuss jitter. In short, each time you pass on the digital stream (via spdif or digital coax), you will be introducing jitter. Thus, even if you were dealing with FLAC and a soundcard that didn't mix and/or resample that sends the data to the receiver via spdif it wouldn't sound the same as if you played it directly on the receiver (networked receiver with decoder and built in DAC). Just to be clear, the digital data being received can be identical, yet due to jitter (syncronization and word clock issues), it won't be played the same. So in answer to your original question, you are better off with the networked receiver that will decode the FLAC and use it's own DAC. Another solution which might be better, depending on budget and how descriminating your ears are, would be either a sqeezebox ($299), modified squeezebox (an analog modded SB with the Sonicap Platinum upgrade but without the Bybees and with an upgraded linear power supply - $600), or Transporter ($1999). From what I've read so far, if you do go with one of the squeezebox options you would be using analog out, otherwise you would still have to deal with jitter and it would still rely on the DAC in your receiver (thus why bother with the squeezebox if you have a networked receiver.) http://www.lessloss.com/about.htmlhttp://www.stereophile.com/features/368/http://forums.slimdevices.com/showthread...red+transporter
Last edited by packetlosss; 01/01/07 03:57 PM.
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Re: Ethernet vs. Digital Audio Out
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Very interesting about jitter.
I had only heard of it regarding the ripping of Cd's, and that was a long time ago.
seems like the spdif way of doing things is flawed. I read so much my head hurts, but i got the impression that the recieving end extracts the clock signal from the data, which is a poor substitute for having its own copy of the source clock. There also appears to be some subjectiveness about how much this can be heard, which is to be expected.
It does seem that a squeezebox would have less digital connections in which to introduce this problem. Unfortunatly I am going to stick with my computer, cause I can watch movies, play music, and do some limited web surfing via my wireless keyboard. spdif is a huge improvement over analog out.
i did like one solution I read which involved clock injection, but I imagine eventually manufacturers will replace spdif with something else. maybe it will be hdmi ?
thanks
randy
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Re: Ethernet vs. Digital Audio Out
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Quote:
seems like the spdif way of doing things is flawed.
But how flawed? I put up some audio files here a couple of years ago the last time all the hubbub about jitter came up... I introduced jitter by hand into some WAV files (at a much higher rate than would normally appear) and let everyone give a listen to see if they could hear it. Consensus was that no one could, not even after burning to a CD then bringing it into their home theatre. Even with 100 instances of jitter per second, the audio stream is still 99.77% accurate.
I'll reproduce the files again if anyone cares to give them a listen to put their own minds at ease.
The audio may be flawed by jitter, but it's not a flaw like your son being born with 13 fingers and webbed toes, it's a small issue on par with, say, light dust on a CD.
Bren R.
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Re: Ethernet vs. Digital Audio Out
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thanks for the reply. to clarify what I meant by flawed: What I had read is spdif is technically flawed because the clock signal is not transmitted independantly, but it is extracted from the data signal. Of course I could have easily misunderstood what I was reading. As far as the sound quality, I had concluded that the ability to hear the effects of jitter is more aptly defined as subjective, as oppossed to obvious. I think its great that your tests conclude you can not hear the difference, and thanks for mentioning it. Although I havent done any A/B tests, my gut feeling is that I wont hear any differences either, so I guess I am biased already I think I will do a test of playing the same compact disk from 2 sources. One being computer-->spdif--reciever and the other test dvdPlayer-->hdmi-->reciever and see if i notice a difference. probably will have to do a diff of the cd's to be sure they are bit identical  I have to wait for my Toshiba upconverting dvdplayer to get returned from warranty repair before I can do the test. The player had an interesting problem. After trying to play the latest crippled/bad sector/pseudo_enhanced_encryption/non_standard dvd's from blockbuster, the unit couldnt read *any* disks. Unplugging unit for 30+ seconds and restarting would fix problem, until similar disk was played again. The labor portion of my warranty had expired but Toshiba gracefully is going to repair/replace it at no charge. I think they were aware of the issue, because the nailed it immediately. my original concern was that the audio chip on my computer kept the data in the digital format, so that the dac was not utilized. I was concerned because it appeared the mixing functions were done in analag. I have since noticed that the software mixer volume control has no affect on my spdif output, so I am assuming there is no dac conversion happening within the computer. Thanks again for the post Randy
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Re: Ethernet vs. Digital Audio Out
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Quote:
I introduced jitter by hand into some WAV files (at a much higher rate than would normally appear) and let everyone give a listen to see if they could hear it. Consensus was that no one could, not even after burning to a CD then bringing it into their home theatre.
There's a lot of dissinformation regarding audio in general floating around and I'm more than willing to believe jitter is one of them. My issues with PC sound cards has more to do with their mixing or resampling etc than jitter.
That being said, I do have to ask how exactly you inserted jitter into a wav file. From what I've read, jitter is a syncronization issue on the digital to analog conversion end. If you take the digital stream and convert it directly to analog you get sound. If however, some of that data that is being converted is not received and converted with the intended timing, you get still get analog sound, but it will be off from what is is supposed to be. This should have nothing to do with the wav file file.
If you take a VOIP call, jitter there gets introduced because not all packets necessarilly get received in correctly timed intervals and you end up with a robotic sounding connection.
With an spdif connection the timing issues should be small, perhaps undetectable, but it would be in the timing of the conversion of the data of the wav file, not the data in the wave file.
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Re: Ethernet vs. Digital Audio Out
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Quote:
I do have to ask how exactly you inserted jitter into a wav file.
Simply by changing at random, single samples. Selecting a single sample and entering RND (65536) into the numeric equation for each channel. That's how jitter will be received by the other end. Unlike SMPTE audio in broadcast or audio-video interleaving, you've got nothing else to go out of sync with with straight audio. You'll lose two binary words (16 bits per channel) and at the very worst at the other end, it'll completely randomly make up the sample. Best case scenario (and even the cheapest CD players do this), it will interpolate the signal if it can't read it.
It's not like you'll lose one bit in the stream and every future bit will be ROLed (rotated left) to fill the "hole"... if that happened, the rest of the track would play as static.
Bren R.
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Re: Ethernet vs. Digital Audio Out
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Quote:
If you take a VOIP call, jitter there gets introduced because not all packets necessarilly get received in correctly timed intervals and you end up with a robotic sounding connection.
VOIP works via packets over UDP, and a traceroute of even a call from me to my next door neighbour will show the data packets hopping around the province a few times before ending up at their end. It's a much different animal than digital audio within a building along a wire or fibreoptic cable.
If I was sending my audio stream of Fresh Fruit for Rotting Vegetables over "teh Intarweb" to Toronto and back before it made it to my speakers, I'd probably get some of the same issues.
Bren R.
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Re: Ethernet vs. Digital Audio Out
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Quote:
Simply by changing at random, single samples. Selecting a single sample and entering RND (65536) into the numeric equation for each channel. That's how jitter will be received by the other end. Bren R.
Randomly changing parts of the data stream isn't the same thing as jitter (Although if you damaged every single sample and no one could tell then they aren't going to notice jitter either). That would be simulating read errors and lost data. Jitter is defined as a time based instability. This instability is going to occur on every word from every single sample and the shift is going to be random (sometimes faster sometimes slower.) Your not losing and interpolating the data, your just not reconstructing the sound waves with the same timing (which will equate to shape) they were encoded with. An extreme example of this affect would be to take a turntable that allows fine tuning of the rotation speed and to keep making random small adjustments to the speed.
This is the reason all DAC's require an oscillator crystal. The crystal establishes the time sync with the audio samples for reproduction of the analog sound wave from the digital sample. When transfering the digital samples over spdif, the clock sync is not transmitted and that is where the jitter is introduced.
What really matters of course is if you can you hear the difference. These fluctuations are probably so small that you wouldn't notice, not to mention since the timing instability is random it's not like you would hear the same imperfections (assuming you heared any) if you played the music over and over.
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Re: Ethernet vs. Digital Audio Out
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Hello
I agree with 2 points recently made.
VIOP can not be compared to jitter. Its my understanding UDP is a stateless protocol and there is no packet delivery confirmation transmitted back to the originator. damaged packets are rerequested, hence sometimes not arriving in time and utilized out of order. the issue mentioned regarding VOIP could be due to the compression used, or other reasons. Thats not to say that jitter couldnt occur in the dac process of encoding the audio data, but the fidelity is so low, i cant see it mattering. I believe phones lines used to have a tap/load_coil/I_forgot_the_technical_term on them to keep the bandwidth to 3khz, and I am guessing a phone line sounds better than VOIP.
Jitter is not randomly changing bits. What I read agrees with what was posted about how it is related to the timing of the bits, not the data within the bits. I dont have a better understanding of the transmission of digital audio though. I also agree that you cant distribute wav files to test jitter. each individial setup that plays the wav file would be effected by jitter differently. hadnt really thought about the randomness of the same machine playing the same track over and over.
with that all said, I am learning as I go and find this an interesting topic.
Randy
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Re: Ethernet vs. Digital Audio Out
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Quote:
Jitter is defined as a time based instability. This instability is going to occur on every word from every single sample and the shift is going to be random (sometimes faster sometimes slower.)
Please, explain further... you're suggesting that it would make it halfway between beats of a crystal oscillator, and that would cause the bits in the stream to shift left or right?
I'm kind of confused on what exactly you believe happens... besides a vague "time instability".
Bren R.
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Re: Ethernet vs. Digital Audio Out
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jitter
A flicker or fluctuation in a transmission signal or display image. The term is used in several ways, but it always refers to some offset of time and space from the norm. For example, in a network transmission, jitter would be a bit arriving either ahead or behind a standard clock cycle or, more generally, the variable arrival of packets. In computer graphics, to "jitter a pixel" means to place it off side of its normal placement by some random amount in order to achieve a more natural antialiasing effect.
For clock jitter, there are three commonly used metrics: absolute jitter, period jitter, and cycle to cycle jitter.
Absolute jitter is the absolute difference in the position of a clock's edge from where it would ideally be if the clock's frequency was perfectly constant. The absolute jitter metric is important in systems where a large number of clock sources are trying to pass data to one another (eg. SONET).
Period jitter (aka cycle jitter) is the difference between any one clock period and the ideal clock period. Accordingly, it can be thought of as the discrete-time derivative of absolute jitter. Period jitter tends to be important in synchronous circuitry like digital state machines where the error-free operation of the circuitry is limitted by the shortest possible clock period, and the performance of the circuitry is limitted by the average clock period. Hence, synchronous circuitry benefits from minimizing period jitter, so that the shortest clock period approaches the average clock period.
To really explain what is happening, we need to take a look at what digital audio really is. Analog sound is a wave. I'll assume everyone has seen graphs of sound waves and knows that what one looks like. Eseentially a sound wave can be thought of as a signal which OVER TIME can vary continuously in amplitude.
To get a digital representation of this signal samples are done at different intervals along the wave. For CD audio this sampling is done 44,100 samples a second with 16 bits of resolution. DVD audio is between 96,000 and 192,000 with up to 24 bits of resolution. What this means is we are taking timed samplings of this original analog sound wave and making a digital approximation of it. There is of course much discussion with regards to how much sampling is needed to make the interpolation of the original analog sound indestiguishable from the original but that has nothing to do with jitter.
What a DAC does is it takes this digital rendering of a sound wave and it reassembles the analog sound wave. To do this properly it needs to have a clock sync so it can establish the correct interval of the samples.
Think of it like a plotter. The timing is the rate at which the paper is moving under the pen. If the paper isn't moving at a constant and determined speed, you are not going to reproduce a proper graph. It will either end up drawing the wave over a longer duration of time than the orignal or a shorter duration.
This is the time instability which causes jitter. Without a proper clock that sound wave can't be reproduced with accuracy. This is why all DACs and CD players have an oscillator crystal.
My VOIP example was a rather extreme example of the timing issue. Jitter for VOIP represents the same problem, it just happens for a different reason and on a different scale (large chunks of the wave are transmitted - so it's not individual samples that are out of time sync). Some packets make it there fast, some slow, some don't make it and have to be resent, but essentially the sound can't be reproduced with proper timed accuracy.
Last edited by packetloss; 01/05/07 03:22 PM.
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Re: Ethernet vs. Digital Audio Out
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I had to go back and reread what the original question surrounding this was. And to be clear, jitter in regards to networking - I'm not really clear on, besides a basic understanding of transport protocols, once the IT bar gets raised beyond basic networks, I spread Chee-zee Poofs around my desk and club the IT guy that gets closest and force him to do my bidding. Digital audio I've been working with for a while. From the stone age to now. Quote:
What a DAC does is it takes this digital rendering of a sound wave and it reassembles the analog sound wave. To do this properly it needs to have a clock sync so it can establish the correct interval of the samples.
Absolutely correct.
Quote:
Think of it like a plotter. The timing is the rate at which the paper is moving under the pen. If the paper isn't moving at a constant and determined speed, you are not going to reproduce a proper graph. It will either end up drawing the wave over a longer duration of time than the orignal or a shorter duration.
Which is why digital audio is reframed every frame to the internal oscillator. Again, the part I'm unclear on is how the jitter is supposed to let the stream run long or short with regards to time. If you could shed some light on that - the actual practicality of what occurs to disrupt the waveform - do you suggest that some bits pass other bits on the wire (or in the fibreoptic bundle) to arrive at the receiver before others? That some travel slower than others and cause a traffic jam on the wire? That the sending unit somehow loses time and starts firing the info at a non-regular rhythm? Or that both are like drummers and each ones' oscillator is off-beat with the other, and one is receiving on the upbeat while the other is sending on the downbeat?
In home audio, there are sometimes issues with digital audio, my DVD player transport is sometimes slow in switching over chapter lines, and the data becomes corrupted. My receiver will switch from Dolby Digital or DTS ES 6.1 or whatever it's decoding and revert to the standard DSP for the DVD connection (Neo 6 Cinema in my case) when it loses any part of the data. This happens maybe once a movie and only happens with this one DVD player. It's a bit of a PoS (curse you Toshiba!)
And on the other side of things, something I go back to a lot. What do the pros use? Whatever the guy actually cutting the movie, TV show or sporting event is using is probably more than good enough for the home user. And what is that standard? AES/EBU, SPDIF's big brother and nearly identical except for voltage (AES3 is 3-10v, I think SPDIF is <1V), connector (balanced or unbalanced XLR or BNC vs "RCA"/whatever that little toslink connector is called) and some of the data transfer.
It's really a great system, well designed, flexible and well thought out by the Audio Engineering Society and European Broadcasting Union.
Bren R.
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Re: Ethernet vs. Digital Audio Out
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Quote:
Digital audio I've been working with for a while. From the Stone age to now
circa 1992 my stone age
and another shot showing CS800 amp at bottom, used to get drunk and crank it, was on middle floor of apartment building, neighbors never smiled at me
I could do a now photo , but it would be so boring.
sorry to drift off topic, but when you mentioned stone(d) age, somehow it rang the little bell in my head
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Re: Ethernet vs. Digital Audio Out
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That a Roland stage piano?
Bren R.
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Quote:
Which is why digital audio is reframed every frame to the internal oscillator. Again, the part I'm unclear on is how the jitter is supposed to let the stream run long or short with regards to time. If you could shed some light on that - the actual practicality of what occurs to disrupt the waveform - do you suggest that some bits pass other bits on the wire (or in the fibreoptic bundle) to arrive at the receiver before others.
Bren R.
I had to do a little more research to get a better explanation.
Here is a good link which describes the problem and presents the solutions. Professional equipment uses a double phase lock solution, which requires a seperate clock sync line.
http://www.tnt-audio.com/clinica/diginterf2_e.html
As it turns out clock information is in fact transported with the data along an spdif connection. The problem is that the way it was designed, the clock sync information is flawed. What happens is the data is assembled and passed to the spdif transmitter which then grabs the clock info and inserts it into the data stream. He says that due to a structural problem with spdif (hence the stigma of spdif being associated with jitter), the data and the clock sync get messed up.
The first attempts at solving this problem, but also moving into very expensive equipment, the DAC needs to provide it's own internal sync which it then syncs up with the clock from the data (inserted directly into the stream rather than pulled directly by the transmitter). Alternatively, some designs allow for a seperate clock line. This of course means you need a transport that supports clock out and a DAC that supports the input.
In short, spdif by itself is not providing a proper clock. If you have an expensive DAC, it can resync the data with it's own clock. He also claims that cheap timing crystals have been proven to not be accurate enough for high end audio.
He also has a page describing jitter specifically (what I tried to do in my previous post). He has some pretty good pictures showing the overtiming and undertimings. It's worth reading.
http://www.tnt-audio.com/clinica/jitter1_e.html
Last edited by packetlosss; 01/06/07 12:39 AM.
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No it was an ensoniq vxd-sd ? workstation. I bought it in the summer of 1990. first real purchase I made after getting a good job. I can play and read a little, but my timing is poor. I guess you could say my 'jitter' is noticable. I sold it a long time ago at the pawnshop. Long sad story I will spare us all the details. Now I have a cheap usb powered M-audio unit. I really havent played with it much with the move and everything else this year. If I was to buy a 'real' music keyboard, I would get a roland now. I think they have the best sound. although I really dont keep up with what is happening. I didnt take many pictures back then, but I am glad I have a picture of that rack. I spent hours tweaking and reconfiguring that stuff. Lots of good times, I was the soundman for a band. I'd be running sound and an attractive chick would come up. I'd think , finally, I was about to be recoginized me for my studleeness. And then she would say, " Is the drummer married"  Randy
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Re: Ethernet vs. Digital Audio Out
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Quote:
He says that due to a structural problem with spdif (hence the stigma of spdif being associated with jitter), the data and the clock sync get messed up.
He elaborates more about this on the second page I linked. Basically the way the data is encoded it relys on certain voltages to represent those states. What happens is there are some instances where if too many 0's follow each other there is a delay before a 1 can be represented and this messes up the clock information. It's quite a bit more complicated than that and my backround is in CS not electrical engineering.
In any case after reading through his pages, there are lots of places where jitter gets introduced. The bottom line is, it's impossible to get 100% time synced data (since no timing crystal exists with that kind of accuracy). From the original recording, all the way through to playback you will be picking up jitter.
So back to the all important question. Does this really make a difference? Here is what he has to say about that.
Quote:
In my personal experience, and I would dare say in common understanding, there is a huge difference between the sound of low and high jitter systems. When the jitter amount is very high, as in very low cost CD players (2ns), the result is somewhat similar to wow and flutter, the well known problem that affected typically compact cassettes (and in a far less evident way turntables) and was caused by the non perfectly constant speed of the tape: the effect is similar, but here the variations have a far higher frequency and for this reasons are less easy to perceive but equally annoying. Very often in these cases the rhythmic message, the pace of the most complicated musical plots is partially or completely lost, music is dull, scarcely involving and apparently meaningless, it does not make any sense. Apart for harshness, the typical "digital" sound, in a word.
In lower amounts, the effect above is difficult to perceive, but jitter is still able to cause problems: reduction of the soundstage width and/or depth, lack of focus, sometimes a veil on the music. These effects are however far more difficult to trace back to jitter, as can be caused by many other factors.
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I respectfully disagree with you about the Rolands, at least if you're talking piano sound. I find the Yamahas, Kawais (just a bit of bias on this one. ), and Viscounts to be far superior. I don't know about effects sounds, though.
I am the Doctor, and THIS... is my SPOON!
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I respectfully disagree with you about the Rolands,
thanks for the reply, I dont feel slighted.
what prompted me to say that is 17 yrs ago i got that ensoniq. it has a built in sequencer and aftertouch and all kinds of goodies i have long forgotten, but it didnt sound lush. Another guy got a roland, and another got a yamaha , and both sounded much better. I'm sure I'm outa touch
Randy
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Well, a few issues immediately spring to life with this. First is it comes from TNT-Audio, a "tweakers" site, while not as huile de la serpente as some (ooh, cable risers!), the guy's drunk the Kool-Aid and tied on his sneakers.
By the third diagram, he's already off the mark - AES/EBU does not rely on a separate connection for clock, the "pros" don't send the clock on a separate wire - just like SPDIF, it's embedded in the single data stream. I'm sure someone probably has a proprietary way of sending the clock discretely, but I've never seen it done.
If we jump past his half dozen ways to clean his clock (pun intended) he finally gives hard information about how awful this poor mistimed signal has become... even at the worse he claims (30ns - 15 times more than the worst DIYAudio measures) that equals a mistiming of 0.00000003 seconds... a single sample of CD program audio only lasts 0.000023 seconds, so my hand-creation of jitter was actually 7.5 times worse than a theoretical "worst case" scenario, and was still undetectable.
Again, correct me if I'm wrong, I sometimes misunderstand the intent of some of these articles, but the complaint is that the signal can get as sloppy as 3.0 x 10^(-8) seconds one way or the other? I sincerely hope I've misread that... if I haven't, perhaps someone could suggest to him that a tenth of a point of relative humidity or degree of ambient temperature in his room would stiffen or loosen the grease on the CD/DVD drive spindle creating a much larger problem than that.
Bren R.
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Quote:
Again, correct me if I'm wrong, I sometimes misunderstand the intent of some of these articles,
well I read some of the links posted earlier, and this guy
http://www.lessloss.com/about.html
has several products for sale
spdif cable , without ends ? for 225 euro x 1.3 = $291
a dac for 2004 euro x 1.3 = $2605
too rich for my blood
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Joined: Dec 2003
Posts: 3,602
connoisseur
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connoisseur
Joined: Dec 2003
Posts: 3,602 |
Quote:
In my personal experience, and I would dare say in common understanding, there is a huge difference between the sound of low and high jitter systems. When the jitter amount is very high, as in very low cost CD players (2ns), the result is somewhat similar to wow and flutter, the well known problem that affected typically compact cassettes (and in a far less evident way turntables) and was caused by the non perfectly constant speed of the tape: the effect is similar, but here the variations have a far higher frequency and for this reasons are less easy to perceive but equally annoying.
Wow and flutter on cassette tapes I will give him. He says turntables are far less susceptable to W&F than cassettes (and usually 0.15% is good for a turntable, belt or direct drive) meanwhile my CBC Radio Reference Disc suggests that a "pass" for a CD player is less than 0.002%... doing that math, a "passable" CD player exhibits 75 times less W&F than an excellent turntable. This is equally annoying? To listen to, or when you have to admit that your audio system isn't nanosecond perfect?!?
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Very often in these cases the rhythmic message, the pace of the most complicated musical plots is partially or completely lost, music is dull, scarcely involving and apparently meaningless, it does not make any sense. Apart for harshness, the typical "digital" sound, in a word.
Oh, god... melodramatic much? The music is dull and lifeless because of a 0.00000003 second "swing" in sample timing? Gracious, this guy must be a hit at the symphony... "that bassoon player ruined it for me, I could tell he drank mineral and not bottled water before the concert, his reed was vibrating 2 nanoseconds behind the alto sax!"
Bren R.
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Re: Ethernet vs. Digital Audio Out
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Joined: Dec 2006
Posts: 24
hobbyist
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hobbyist
Joined: Dec 2006
Posts: 24 |
I was more interested in finding out what was supposed to be wrong with spdif. If we start with the assumption that a proper clock sync is needed and you don't get it all the time (supposedly it's data dependent, therefore sometimes it's correct and sometimes it's not), then there has to be jitter.
I didn't get the impression he was trying to sell me anything. The lessloss people were, but I'm not spending $2000 to reduce jitter, nor would I suggest anyone else do that.
What I understood the 2nd guy as saying, was that if you go out and buy a $20 CD player it will be using a cheap and unreliable crystal. Avoiding the bottom end players doesn't sound like bad advice to me. As far as DACs go, there are some relatively inexpensive ones (I don't have models or pricing) that do their own syncing.
In any case we went down this road because I contend that making random data changes to a wav file is not the same as jitter. I will agree that if you made changes to almost every sample (not just say 10 changes) and no one heard a difference, then they probably wouldn't notice jitter either. Still, these are 2 different things being tested.
I don't know if anyone can really notice or hear jitter. This can all be factual, yet at the same time it could also be meaningless if even with crappy crystals, it's undetectable by the human ear.
The proper way to test this would be to set up a DAC that allows you set parameters for it's timing. This way you could force it to deviate by some fixed amount from the clock. Once you have this, you would run ABX tests and see how much deviation from the clock is needed for differences to be heard. If it turns out to be a value that is much greater than even the cheapest equipment puts out, then we know this whole issue is BS.
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Re: Ethernet vs. Digital Audio Out
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Joined: Dec 2003
Posts: 3,602
connoisseur
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connoisseur
Joined: Dec 2003
Posts: 3,602 |
Quote:
I was more interested in finding out what was supposed to be wrong with spdif. If we start with the assumption that a proper clock sync is needed and you don't get it all the time (supposedly it's data dependent, therefore sometimes it's correct and sometimes it's not), then there has to be jitter.
I always assumed the jitter complaint was that a SPDIF frame was actually bolloxed by jitter, his article shows the complaint is actually much smaller than that - the information is right, it's just "marred" by timing issues. I'm not sure that's an issue we'll ever get away from. As long as there are moving parts in a player's transport (CD/DVD drive spindles, turntable platter motors, hard drive platter motors) there will be some wow and flutter. As long as data is synced to an oscillator, there will be timing errors. I think, like with THD levels in new solid-state amplifier circuits, we're at a subsonic level with the current technology... the weak link's proving to be our own hearing.
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What I understood the 2nd guy as saying, was that if you go out and buy a $20 CD player it will be using a cheap and unreliable crystal. Avoiding the bottom end players doesn't sound like bad advice to me.
Another frame of reference for CD audio playback requirements is that a $30 computer CD-ROM can pass data reliably at up to 48x what is required for CD audio. For data. Which has to be bit-perfect. If you want to skip to a certain comfort level on the price tree, that's understandable, but you can stop at any level that has the features you want.
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In any case we went down this road because I contend that making random data changes to a wav file is not the same as jitter. I will agree that if you made changes to almost every sample (not just say 10 changes) and no one heard a difference, then they probably wouldn't notice jitter either. Still, these are 2 different things being tested.
At this point, knowing the complaint about it, I wouldn't have to damage any data. I was always under the assumption that there was an expected loss/destruction of a single 44.1/16/Stereo frame of audio. They don't even contend that much, just that the timing isn't perfect to the nanosecond... that they measured timing differences of 2 billionths of a second one way or the other during playback. I'd point to my own testing here that if no one could hear noise inserted lasting roughly 2 one-hundred thousandths of a second, that this is probably inaudible.
For more perspective - it takes an electrical (audio) signal about 6.6 nanoseconds to travel through a 6 ft cable or a flash to travel through a TOSlink cable the same length.
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The proper way to test this would be to set up a DAC that allows you set parameters for it's timing. This way you could force it to deviate by some fixed amount from the clock. Once you have this, you would run ABX tests and see how much deviation from the clock is needed for differences to be heard. If it turns out to be a value that is much greater than even the cheapest equipment puts out, then we know this whole issue is BS.
I'd be interested in hearing results from that.
Bren R.
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Re: Ethernet vs. Digital Audio Out
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Joined: Dec 2006
Posts: 24
hobbyist
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hobbyist
Joined: Dec 2006
Posts: 24 |
Quote:
Another frame of reference for CD audio playback requirements is that a $30 computer CD-ROM can pass data reliably at up to 48x what is required for CD audio. For data. Which has to be bit-perfect.
Bren R.
Actually it wasn't the speed or accuracy of the drive. In all cases we are really assuming a 100% correct data stream that is buffered and more than fast enough to keep up with the needs of the DAC. It was that the cheaper units used a cheaper and more inacurate timing crystal. Kind of like those $5 watches that are off by a minute a day. The price point to get a more accurate crystal isn't much higher. Kind of like those $15 watches that are only off by a few seconds a day.
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Re: Ethernet vs. Digital Audio Out
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Joined: Dec 2006
Posts: 24
hobbyist
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hobbyist
Joined: Dec 2006
Posts: 24 |
Just a little update on this topic. I found this thread on the avs forum discussing the yamaha RX-1700 and 2700 and a few posters were saying that they prefered the sound of analog inputs from their DVD player to digital inputs. http://www.avsforum.com/avs-vb/showthread.php?t=713397&page=30&pp=30Since the digital input source didn't matter(HDMI/spdif etc), the issue really came down to which DAC they liked better. So basically I'd say DAC choice is the real issue, more so than spdif/jitter etc. The bottom line is DAC choice is subjective and different DAC's have comeletely different sound signatures. So test out all your components and decide which has the better sounding DAC and then take analog outs from there.
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Re: Ethernet vs. Digital Audio Out
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Joined: Aug 2006
Posts: 213
local
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local
Joined: Aug 2006
Posts: 213 |
Quote:
So basically I'd say DAC choice is the real issue, more so than spdif/jitter etc.
I can believe this.
I have 3 digital audio sources:
dvd,cable_box,PC.
none of them offer analog multichannel out, so I will stick to using the dac in the reciever.
My denon has something called AL24 Processing plus, So I guess I am in decent shape.
from the Denon site
" The acclaimed Advanced AL24 DSP processing improves the fidelity of high resolution stereo PCM sources such as CD and DVD (up to 192kHz sampling frequencies). Through sophisticated DSP processing algorithms, it improves low level detail and enhances fidelity by up-sampling and adaptive filtering techniques. Advanced AL24 also provides increased dynamic range and spatial information, bringing out all the nuances with optimum clarity and natural fidelity. For additional information, visit the Denon Tech Showcase."
see ya
RAndy
Axiom M80, Ep600, Qs8, VP150, Crown XLI 2500 , DBX Driverack PA2, Focusrite Scarlet 2i4
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