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A novel way to massively improve the SQ of computer audio streaming


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Most important: please realize this thread is about bleeding edge experimentation and discovery. No one has The Answer™. If you are not into tweaking, just know that you can have a musically satisfying system without doing any of the nutty things we do here.

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  • 4 months later...
21 hours ago, cat6man said:

 

@rob

@ray-dude

@hols

@austinpop

 

have you experimented with the hqp buffer time?

i have found that increasing it always improves the SQ.

 

the maximum permitted through the hqp web interface is 250ms (range checking on input**)

with an opticalRendu, i found better SQ with 500ms, which could be set by editing the settings.xml file

also, with a NUC (audio-linux for example, haven't tried w/ euphony) which has a lot more RAM available,

you can increase the buffer time even much more significantly (just note that the buffer is in units of time, not bytes,

so that a larger value is possible with red book than with hi-rez sources)

 

i'd be interested in any conclusions you have with both the extreme and diy servers w.r.t. buffer time settings.

 

**interesting story--------i had a vp who was getting a demo of a new modeling program for laying out the location and design

parameters for mobile cell sites, incorporating terrain, propagation, etc. 

he sat down and immediately entered a value for a cell tower that was 2000 miles tall, the program crashed, and he walked away saying their program design had flaws.  in my case, i set the buffer value to use the maximum amount of ram available, calculated using red book cd parameters.  when i went to play a 24/96 track, i crashed the NUC.  however, the maximum ram buffer did improve SQ.

 

 

I've spent several months messing with these settings on the Extreme and agree with Ray Dude that they are basically tone controls when using 10ms - 250ms,  until I tried "Default" on both settings (buffer time and bits).  To me, "Default" ended up being the best parts of all the other settings with none of the weaknesses.  

 

When using any setting between 10-250ms, I would prefer one over the other depending on the album I was listening to or because of some recent system upgrade or change.  "Default" has sounded good across all my music and through all the recent changes and upgrades.  It was a very obvious improvement, not something I had to focus on to hear.  

 

When ever I would increase the buffer, let's say from 20ms to 50ms, the top end sounded rolled off, there was a loss in dynamics, loss in details and speed, but sound was fuller and smoother.  With Default - I am hearing all the details, speed, fullness, dynamics, etc with none of the drawbacks.  

 

Wanted to see if anyone else is hearing the same thing with "default" that I am so I can finally put this dead horse to rest.  

 

I'm running Extreme to Dave to AFC-10 to Susvara.  

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