Understanding Jitter in PC Audio


I have a fundamental doubt on the PC audio as a source. IN a traditional CDT/Dac combination we have a clock signal coming out in the SPDIF signal. Apologies if it sounds too silly but am planning of builing an HDD based transport as an alternative to my classe CDT1 :). was planning on a USB output from a dedicated PC and then use a good USB to SPDIF converter.

My Understanding is that in case of a HDD based transport, the File is converted into an Async format (Lossless) . This is then played via a PC/Mac and when given out as digital out, the clock that is synched to is the machines own clock (Am I right ?)

a) does this impact jitter of the Lossless file in anyway ? also what would the difference between an I2S and an USB interface be in this case as the clock is not really the original clock ?
b) Can the original information without any timing errors be reconstructed from this using an external reclocker like the empirical audio device OR Monarchy ?
c) If the clock is not present will an external DAC just assume the input to be as per its own clock. (If the rip were done by CDROM using the same clock freq as a DAc give any added benefit)
arj
"Shadorne
USB has no clock - it is an asynchronous protocol."

Actually, this depends on the USB protocol. All USB converters use Synchronous Adaptive mode, which uses the embedded clock in the data stream from the USB cable. It relies on the clock at the computer.

Steve N.
Empirical Audio
" if you use USB then it all comes down to the clock in the DAC."

Some DAC's have upsampling clocks in them, but many have no clock at all. NOS DAC's have no clock.

There are a handfull of very expensive DAC's that generate a local clock and allow you to use this, but only if the source has a word-clock input and can be synchronized to the local DAC clock using a word-clock cable.

Steve N.
Empirical Audio
Steve,
thanks a lot for that very informative post. Ill need to read it a couple of time more to get it all but I think the core of the issue has been very well explained.

It nice to know that the clock can be recovered and corrected as long as the information is still in the digital domain, irrespective of the format.

Thank you for that.
Arj,

referencing your original question about I2S, if you have the patience to wait a few months, I think we will start seeing more products trying to use a native I2S interface. PS Audio's new Transport and DAC will be making an I2S connection using HDMI plugs/cables. My hope is that others hop on the bandwagon and begin using HDMI to transmit in native I2S format. Since HDMI cables and plugs are readily available for other purposes, the market barriers to entry are significantly lower for this application.

This would represent a major technology upgrade to one of the weakest parts of PC and digital audio. I have a hard time imagining everyone NOT jumping on this once industry heavies like PS Audio take the lead. I'm also betting (hoping?) custom shops like Steve at Emperical will be able to modify a SqueezeBox or Sonos with an HDMI I2S connection while we wait for manufacturers to ante up.

If the industry could standardize on HDMI components to transmit native I2S, we could kill off the crappy optical and SPDIF connections we use now and virtually eliminate the need for expensive re-clockers (sorry about that part, Steve).

My $.02. In a horrible economy, people aren't going to plunk down big money for something unless it makes a big difference for them. I think this could be it.
I use Winamp (on WinXP, using ASIO4ALL) to stream from my Sony laptop's toslink output into a plastic toslink cable to a Behringer SRC2496 DAC which then outputs to my system's preamp. I set the Behringer to use its internal clock. Behringer claims the SRC2496 has a "High-precision quartz clock generator [that] removes jitter and corrects off-tune, incorrect sample rates".

My question to the DAC experts here: Is there anything glaringly wrong with how I've set this up? Are there any easy improvements to be made?