[PhaedraCorruption]

I'm currently struggling to understand the concept of high end Digital Components, for example, outputting TOSLINK from the motherboard of your computer VS outputting TOSLINK from a dedicated high end sound card. If my understanding of digital signals is correct, then there should be no difference, as it is just 0s and 1s, a distorted 1 and a clean 1 are both still 1s. Unless somehow 1s are being turned into 0s and vice versa with interference via poor shielding/pathways, etc etc, there should be no difference. With CD players, it could be because of the way the data is read from the disk that could be distorting them, but with other electronics, is there a difference? 
 
Thoughts? 

 

[henrylee]

bitrate and sample rates varies with each soundcard , some sound cards are only able to produce lower sample rate sounds .Sample rate is basically the frequency in which analogue signals are turned into digital or binaries .Well thats a level computing ,i dont know much about daps but i assume is something to do with like interferance and resistance of wires and capciters.

 

[AVU]

this is a common misconception, and there's plenty of information on it - just search for terms like "jitter."
 
Basically, you might understand it like this: digital data ordinarily isn't time-dependent. One component waits until all the data is there before proceeding. But music is streaming, and therefore time-dependent. You don't upload the song into your head as a set of data, you listen to it in time. And so any tiny variation in the timing of the data can cause some part to be misaligned. It's not actually misaligned, this is just a metaphor, but it's a simple way of understanding it. The differences it makes aren't usually that noticeable, but if you're a ridiculous hifi person, and you want everything to be absolutely precise, like many of us do, then you have to worry about this timing.  There are expensive devices called "clocks" specifically for the purposes of "re-clocking" the data signal.
 
Hope this helps.

 

[computerparts]

Google oscilloscope square wave and you will see many diferent wave patterns. It's not just about 1s and 0s, it's a bit more complicated. There are also many varaibles that set apart the high end from the low end. One often overlooked component is the power supply.

 

[Noumenon]

Could somone address that actual question of the OP, wrt difference between TOSLINK from a MB, and TOSLINK from a dedicated sound-card? The OP did not seem to be asking about DAC's, so none of the replies above seem relevent.
 
I don't think there would be a ratz-a$$ difference if you are just needing TOSLINK from the computer,.... i.e. wave-forms, sample-rates, jitter don't enter the equation at this stage.
 
Folks who buy "hi-end" sound-cards benefit because they are getting a better DAC, as compared to what they get on their mother-board,... but if you have an external DAC and your MB supports TOSLINK out, you don't benefit (or am I wrong?).
 
 

 

[tws1]

Correct, there is no difference. Sort of...
 
The problem with jitter originates at the MB, same as the clock. The MB pumping out the data isn't just streaming out the data as fast as it can, there is an imbedded clock in the digital signal that is adhered to during the DAC conversion. In other words, imagine a digital signal rather than a USB data connection. However, these things CAN'T be heard by human ears. Don't worry about it.
 
A FLAC file pushed through ANY digital connection to a half decent DAC (any external DAC is half decent), then amplified to your hearts content- WILL sound great.
 
It is true that some members will claim to hear differences in digital sources, which might be the case. However, these guys are spending thousands on cables and connectors let alone the DAC itself. The company that makes the DAC only has to make 3 or 4 to make amazing profit, kind of price.... It is more than likely they are justifying their purchases, or is 0.5% louder, therefore sounding 'better'....
 
The differences might be being able to output the signal 'bitperfect'. Resampling on a cheap MB chip might introduce some static hiss to the sound, or distort it somehow. Then there are effects like Dolby heaphone, artificial sound staging, etc, if you are into that sort of thing.

OP, no difference.

 

[DaveBSC]

AVU did a pretty good job of explaining why there's a difference, though toslink out if either a motherboard or a soundcard is hardly an ideal way to listen to music. The idea that "bits are bits" is a complete myth. Yes, when transferring data to say, a USB hard drive, bits are indeed just bits. The computer takes its time to make sure that the bits that left and the bits that arrived are identical. If they weren't, the file could be corrupted. None of that takes place in real time, the USB hard drive isn't reading the data as its receiving it. It's just sitting and waiting for the transfer to finish.
 
Digital audio doesn't have the luxury of getting everything ahead of time, and then checking to make sure it's exactly perfect from the source. It gets it now and it plays it now, and if there are errors they go straight on through. This is where a high-end digital source matters. A distorted 1 and a clean 1 are very much NOT the same thing.

 

[Noumenon]

Thank you for your reply.
 
Quote:

 
Digital audio doesn't have the luxury of getting everything ahead of time, and then checking to make sure it's exactly perfect from the source. It gets it now and it plays it now, and if there are errors they go straight on through. This is where a high-end digital source matters. A distorted 1 and a clean 1 are very much NOT the same thing.

We're talking about computer audio here. So you're saying that the computer does not buffer "audio" as it does every other computer data? Say I have a lossless audio track that is 50mb in size and Plays for 5 minutes,..... a typical computer HDD can load many times that amount of data in a fraction of that amount of time!! So i'm not buying the "It gets it now and it plays it now, [..] because doesn't have the luxury of getting everything ahead of time".
 
 
Quote:

 Originally Posted by DaveBSC
 
....though toslink out if either a motherboard or a soundcard is hardly an ideal way to listen to music.

 
Can you expand on this some. I use optical out from my MB to a DAC/AMP,... and if there is a better way (say a CDP) than I would be interested in looking into it.

 

[AVU]

Quote:

Could somone address that actual question of the OP, wrt difference between TOSLINK from a MB, and TOSLINK from a dedicated sound-card? The OP did not seem to be asking about DAC's, so none of the replies above seem relevent.

 
That's incorrect- the motherboard and a dedicated sound card are themselves just two different DACs. And yes, optical has similar problems to usb, because the optical connection is just a different way of transmitting the digital signal. Everything about timing errors still applies. In general toslink from a macbook pro, say, has an easier time getting this right that usb on most lower end dacs, which is why people who have the option (mac owners, or people with toslink capable soundcards on pcs) use that rather than usb on these dacs. But async usb dacs are supposed to handle all this better, it's just a relatively proprietary technology, so the units tend to be more expensive.

 

[Noumenon]

 
Quote:

 

Could somone address that actual question of the OP, wrt difference between TOSLINK from a MB, and TOSLINK from a dedicated sound-card? The OP did not seem to be asking about DAC's, so none of the replies above seem relevent.

 
That's incorrect- the motherboard and a dedicated sound card are themselves just two different DACs. And yes, optical has similar problems to usb, because the optical connection is just a different way of transmitting the digital signal. Everything about timing errors still applies. In general toslink from a macbook pro, say, has an easier time getting this right that usb on most lower end dacs, which is why people who have the option (mac owners, or people with toslink capable soundcards on pcs) use that rather than usb on these dacs. But async usb dacs are supposed to handle all this better, it's just a relatively proprietary technology, so the units tend to be more expensive.

 
Why are you speaking of DAC's (Digital-to-Analog-Converters)? Again the OP question was about the difference between a MB and a sound-card, in sending Digital data to an external DAC. I undertsnad that the MB and a sound-card has a DAC, but it apears that the OP is not outputing a analog signal, but instead is passing a digital signal. How would the DAC component of the MB or sound-card come into play in this case?

 

[DaveBSC]

Quote:

Thank you for your reply.
 

We're talking about computer audio here. So you're saying that the computer does not buffer "audio" as it does every other computer data? Say I have a lossless audio track that is 50mb in size and Plays for 5 minutes,..... a typical computer HDD can load many times that amount of data in a fraction of that amount of time!! So i'm not buying the "It gets it now and it plays it now, [..] because doesn't have the luxury of getting everything ahead of time".
 
Can you expand on this some. I use optical out from my MB to a DAC/AMP,... and if there is a better way (say a CDP) than I would be interested in looking into it.

No, it does not buffer the audio data in the same way as a file transfer. What the digital receiver in the DAC receives is what is being sent by the digital output of the sound card in real time. If the source clock data from the computer has timing errors, these will be passed on to the DAC, which must then attempt to deal with the timing errors in the clock stream. In general a higher end DAC will deal with jitter better, but you still would not want to use a $50 CD player with an ultra high end DAC. S/Pdif has no flow control or retransmission capabilities in the way that ordinary file or network transfer does. When errors in the data and clock occur, they are passed along and ultimately are audible not in terms of skips or drop outs, but just worse sonics.
 
As for why you would not use optical, it's because its the worst medium for transferring digital audio. While it does avoid RF and EMI, as well as the potential for ground loops, in all other aspects its worse than coaxial digital. Jitter is a massive problem with Toslink, and I don't think it's just the electrical > optical > electrical conversion that must take place because the old ST optical format doesn't seem to be nearly as bad in regards to timing errors as Toslink is.
 
The best format for going between source and DAC is I2S, but only a few components have that, and they are generally not compatible with each other. A close second is true 75 Ohm coaxial digital with BNC connections, and then maybe a tie between AES and coaxial digital via RCA jacks, and Toslink last.
 
In my experience, the best way to get digital audio from a computer, Mac or PC, is a USB > S/Pdif converter - provided it's a high quality asynchronous design that does not get its power from the USB bus. Examples are the ART Legato, Wavelength Wavelink HS, and the Empirical Off-Ramp 4. They are expensive, but they will sound better than any sound card you can name, and certainly FAR better than Toslink from a motherboard.
 

 

[Noumenon]

Thanks a bunch Dave for that excellent explanation!

 

[DaveBSC]

Quote:

Thanks a bunch Dave for that excellent explanation!

Sure. If you really want the dirty details on how all of this stuff works, contact Gordon at Wavelength or Steve at Empirical. They are on the absolute cutting edge of computer audio.
 

 

[sterling1]

My pre/pro  has both coax and optical S/PDIF inputs, but only one of these, an optical input, can receive 24/96. Sounds pretty good to me. I cannot distinguish between this input and others. What I'm saying is I don't believe optical is inferior.

 

[AVU]

 

Background information: Why drop-outs occur​

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Processing of streaming data in real-time is a very challenging task for Windows based applications and device drivers. This is because by design Windows is not a real-time operating system. There is no guarantee that certain (periodic) actions can be executed in a timely manner.​

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Audio or video data streams transferred from or to an external device are typically handled by a kernel-mode device driver. Data processing in such device drivers is interrupt-driven. Typically, the external hardware periodically issues interrupts to request the driver to transfer the next block of data. In Windows NT based systems (Windows 2000 and better) there is a specific interrupt handling mechanism. A device driver cannot process data immediately in its interrupt routine. It has to schedule a Deferred Procedure Call (DPC) which basically is a callback routine that will be called by the operating system as soon as possible. Any data transfer performed by the device driver takes place in the context of this callback routine, named DPC for short.​

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The operating system maintains DPCs scheduled by device drivers in a queue. There is one DPC queue per CPU available in the system. At certain points the kernel checks the DPC queue and if no interrupt is to be processed and no DPC is currently running the first DPC will be un-queued and executed. DPC queue processing happens before the dispatcher selects a thread and assigns the CPU to it. So, a Deferred Procedure Call has a higher priority than any thread in the system.​

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Note that the Deferred Procedure Call concept exists in kernel mode only. Any user-mode code (Windows applications) runs in the context of a thread. Threads are managed and scheduled for execution by the dispatcher.​

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While there is a pre-emptive multitasking for threads, DPCs are executed sequentially according to the first in, first out nature of a DPC queue. Thus, a sort of cooperative multitasking scheme exists for Deferred Procedure Calls. If any DPC runs for an excessive amount of time then other DPCs will be delayed by that amount of time. Consequently, the latency of a particular DPC is defined as the sum of the execution time of all DPCs queued in front of that DPC. In order to achieve reasonable DPC latencies, in the Windows Device Driver Kit (DDK) documentation Microsoft recommends to return from a DPC routine as quick as possible. Any lengthy operation and specifically loops that wait for a hardware state change (polling) are strongly discouraged.​

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Unfortunately, many existing device drivers do not conform to this advice. Such drivers spend an excessive amount of time in their DPC routines, causing an exceptional large latency for any other driver's DPCs. For a device driver that handles data streams in real-time it is crucial that a DPC scheduled from its interrupt routine is executed before the hardware issues the next interrupt. If the DPC is delayed and runs after the next interrupt occurred, typically a hardware buffer overrun occurs and the flow of data is interrupted. A drop-out occurs.​