Originally Posted by Andrew_WOT
Does that mean that software oversampling at the player level is in general a bad idea?
I do not see the connection between jitter and software oversampling at the player. But since you brought oversampling up, I will say a couple of words about it, that may be of value.
In almost all cases, (my guess is over 99.9%), there is going to be oversampling (some may call it up-sampling) of the music data (samples) before it gets to the circuitry that actually does the conversion. The main question is (smaller issues aside) - how good is the oversampling. One can do a great over sampler, a real poor one, or anything in between.
The quality is what counts most. Where you do it is a secondary consideration. Almost all DA's do some oversampling, the exception is NOS DA's and they suffer seriously from the lack of oversampling. I posted about it under the NOS DA thread.
Say you feed a DA a 44.1KHz rate, but the device is going to over sample by X16 to 705.6Khz, or by X256 to 11.2896Mhz or what not. If you fed the DA a double rate which is 88.2KHz, the device will need to oversample by X8 to get to the same 705.6KHz, or by X128 for 11,2896MHz operation. You fed it twice the rate, and it is going to skip the first stage of X2 oversampling. If you feed it 176.4KHz, it will skip the first X2 and second X2 stages, because it was "already done".
So the issue gets back to the quality of the oversampling. One can do a nice job in a computer, or in a dedicated DSP, FPGA or even inside the DA. Or one can do a poor job of it anywhere... An external over sampler may be an improvement or degradation, and it depends on which over sampler is better. The concept of doing it outside in a separate box does not hold, unless it is done better in the external box.
There are 2 general uses of a DA (and AD):
1. There are devices that are designed with the goal of best sound.
2. There are also devices that are oriented towards real time monitoring. A guitar player or a drummer may want to hear what the play with a spot monitoring speaker or headphone. Or one may want to overdub tracks on top of each other - you listen to pre recorded music WHILE playing a new track.
In the first case, one can concentrate on making the best sound with no constraints. In the second case, it is important to keep the time delay through the AD and DA, and the computer (or workstation) short enough (a few milliseconds). If the delay is too long, the music one plays in real time will be heard too late, thus it will sound like an echo, and that is not good for listening in real time or over dobbing. So the second case calls for "LOW LATENCY converters (which means low time delay). In the first case, one does not care if the delay is many milliseconds. In the second case doing things in a hurry is important.
Not surprising is the fact that most often, when one has more time to do a job, it ends up yielding better results. One can do a lot more quality work in say 5msec, than in say 0.5msec. But the call for low latency is out there, especially when doing recording and over dobbing work in a audio work station, utilizing relatively long delay interface types such as fire wire. While such interfaces are capable of handling a lot of channels and that is a positive, they slow the delay time. So many DA and AD makers decided to push for low latency, in order to cover all bases.
And of course, some gear makers started advertizing low latency as some measure of quality, implying that low latency means better conversion (while the opposite is true). At least one DA IC maker got wise so they provide dual mode operation, low latency and high latency, where the sonic quality is better.
Latency is accumulated (AD, computer and interface and DA). Most of the latency is due to the interfaces such as fire-wire, but the AD and DA do add up some to the overall delay. Why am I talking about latency? You brought up oversampling, and that is where the major portion of the delay takes place. That is where the "corner cutting" takes place, lower latency, less computational hardware but at the expanse of quality.
In this group (head-fi) most people are interested in listening to already made music, not in real time over dobbing or spot monitoring. So it may be of value to realize that the notion that low latency being an indication of good quality is in fact upside down. There is no benefit to low latency, and most often it stands opposite to best sound quality.
The trade off between delay (latency) and quality takes place at the oversampling computational block. If someone tells you that they have a low latency converter, it does not mean better quality. Often the opposite is the case.