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Let me try - too

Hi there.

Let me give you some hints:

First of all, just to make it clear. The PC on its own is regarded
and handled by todays "PC-audiophiles" more or less like a blackbox.
Most of them even accept software-players which are mainly
intended to play mp3. None of them was developed to playback audiophile audio.

The pro-scene is at least 5 years ahead of the consumer scene.

It started all with foobar, when audiophile-people realized there is a high audio potential on the PC. (However, try to talk about audiophile playback at Hydrogene, your thread won't survive for long.)

The PC world is just a different world.

Todays audiophile-PC-activities starts usually by tweaking the soundcard or DAC earliest. 99% of all people think it is all digtial! It doesn't matter what player I use they take it for granted that it can't be made better. They'd be right in case we'd have a perfect soundcard, which would be able to perfectly regenerate the data-stream. Unfortunately I never came across of such a card.

Just as an example: Perhaps you read above thread about the USB-audio-driver. This driver is available since a long time and ways better than anything else around. How can it be that it has been just ignored for such a long time!?!

That's also valid for playback from RAM.


Now some words regarding the problems I encountered by using a PC
as source:

IRQ handling is a big issue.
HW-Latencies are an issue
Noise and EMI are for sure an issue.
Power supply is an issue.
Clock is an issue.
Data conversions are issues.
Audio data-formats are an issue (44,1 just doesn't match the PC environm.)
Buffering is an issue
Blocksizes are an issue
Operating systems are a big issue.

If you put this all together you'll realize that the PC has never
been intended to be an audiphile device.

Above issues will severely impact the "timing" (Jitter,Latency Jitter) and induce nonlinearirties to the data-stream.

Even if people claim - my player plays bit-perfect - It doesn't say anything about audiophile sound quality.
Because above sound-deteriorating effects are just not considered. We're talking realtime data. That's something else than opening a word document, which is also a "bit-perfect" task!

FLAC conversion is just another task in a realtimeprocess-chain.
The PC processes all instructions sequentially and is giving them this
or that priority. The PC cannot finish ceratin tasks since higher
priotity tasks jump in, like graphics refresh or similar. This is causing nonlineraties. The buffers have to be quite big to cope
with these "interrupts". Buffer refresh and fill is potentially a hot topic here. The PC has to play and to convert at the same time. This
is not possible, even if the CPU would be loaded with 10% load only.

Since a while I am upsampling my 44,1 material to 48kHz using
Shibatch ssrc_hp. Because 48kHz is just a better choice for the PC
environment and my external DAC. If you compare realtime upsampling to
offline upsampling you'll find a big difference on sound quality, even
when using the same conversion algorithm. Doing it realtime will add
100-200ms or more of latency to the stream, which seems to be sufficiant to wipe out the advantages generated by the conversion.

By using a Realtime-Linux system I could manage to get around of many of the above mentioned issues, especially lowering the latencies lot, what led to the best sound I ever had on a PC.

Linux advantages:
Realtime operation
Higher precision timer
Higher Audio process priorities
Real Multitasking (there is still space for improvement)
Better process structure
asf.

Beside that playing back from RAM, > 500W powersuplies or battery
operation will improve the situation.

There are more things to do, that's for sure.

Just try it. And you'll see how it works.

Good luck.

Cheers
KLS



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