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question regarding spdif output on motherboard audio

post #1 of 4
Thread Starter 
Hopefully I'm not reposting a previous question, but i did search quite a bit.

It is my understanding that pcm audio coming from the optical audio output of my computer is never processed through a DAC until it hits the receiver. Am I correct in this assumption?
I currently enjoy most of my music collection and all movies via this method.

Movie audio encoded in their respective formats are recognised by the receiver (dts/dd).
Playing mp3/flac formats results in a pcm logo on the receiver to be decoded by the DAC there and I assume it is not being converted to analog in the computer audio hardware just to be encoded as a pcm stream before output. This would seem redundant and affect quality. The reason I ask is there are a few threads regarding a seperate unit (external DAC) via usb to convert digital audio to optical signal or analog signal.

So I guess I am wondering if there would be a benefit to an external decoder to convert first (once) to analog and input to the receiver that way or if the optical output on my pc to the optical input on the receiver is achieving the exact same signal flow.

Also, is it better to have the computer output 24bit 192khz signal hoping that it is oversampling or leave the output at the native mp3/ bitrate of 16bit /44khz and have the receiver do oversampling. (does the receiver do oversampling of pcm signal?)

The setup sounds quite fantastic the way I use it, but if I am missing something, it is always nice to upgrade .
post #2 of 4
Not sure who Mike is, sorry he's dead though....

You can't oversample digital data. You can't create a more accurate signal than you have available in a digital format. The only way to "create" a better PCM representation is to have a higher bitrate on the source material. That said, I set my audio on 24-bit 96kHz as my default although all it's doing is sampling analog "system sound" audio that was created from a digital file... of course, you make a good point: maybe the original file is simply streamed via PCM... hmmm.....

If what you're hearing sounds right then why worry about the details - especially as a means to spend money?
post #3 of 4
Thread Starter 
I was referring to interpolation of the signal as a cd player would oversample the digital signal to approximate the missing analog information between sample slices. As in, a 44.1khz sine wave over one second of time would have 44,100 steps to approximate an analog wave. Upsampling would effectively smooth out the steps and fill in the "in between" information. This has been done on cd players since the early 90's no? Would there be no benefit to apply this to any pcm signal at some point along the way? Or is it implied/expected?

Andrew, you're saying the optical output of my pc is sampling the analog recreation of the digital file (wav/flac/mp3/cd-audio) i guess? If that is the case, and since I wouldn't consider the hardware on a computer motherboard to be "high end", maybe there would be some benefit to test driving an external audio processor via USB.
post #4 of 4
Thread Starter 
Haven't seen much reply to my thread. I did some looking around the interwebs since my first post while "working". Figured I would see if anyone has any comments on my finding.

The schematic for my receiver shows the dsp doing an upsample algorithm to any digital signal regardless of source. Sending it a 24/192 is not necessary if the source material isn't of that bitrate. It is better to send it only what the source file is encoded at, regardless of the system setting. The winamp plugin Maiko WASAPI does this if you allow it to take exclusive control of the audio device.
There is also a WASAPI output plugin for Foobar, but keeps the same bitrate for any file played.

All these digital beeps and boops are sent to the digital output as is. No different than if i were to be using an external DAC..... Except for jitter. The external units claim better jitter management. I guess they need something as a selling point.
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