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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.