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BARIX - Whats better, MPEG1 48khz, or MPEG2 24khz?

D

duckfan98

Guest
Trying to get the best sound out of our IP STL.....everything I've read is rather confusing. What is the best quality setting I should use on my Barix setup? Is it MPEG1 @ 48khz, or MPEG2 @ 24khz??

Our link wont allow PCM due to dropouts, but will handle any MPEG just fine. Thanks for your help!
 
Generally speaking, the higher the sample rate, the better the quality. This is MP3, so listen to the difference between files encoded at say 16 kHz, and one at 48 kHz. You'll choose the 48. If you're using it for AM, then 32 kHz is fine. If it's FM, use 44.1.
 
I concur... FM, use 44.1, you will hear 24Khz as as a shrill high end on FM. AM, use the 24 MPEG2. MPEG2 is far superior to MPEG1... I actually think MP2 at 384K is the most friendly codec you will find to transcoding.

Which Barix do you have? I'd be happy to look at the online manual and offer suggestions.
 
I believe the only thing the barix units will do at 44.1 or 48 is mp3. I've had good luck with running mp3 at 44.1 or 48 with the quality setting at the highest. 44.1 stereo at quality 7 will use around 192kbps on average, it is VBR so bitrate varies with complexity of program material.

44.1 PCM stereo needs around 1.6mbps to function well and that's with no other traffic on the link.

I just looked at the traffic stats on a router at Chris' site that has 2x PCM barix feeds going to it, 3.0-3.1mbps steady coming in. Just looked at one of mine thats 44.1 mp3, about 190kbps so I was right on..


Stephen
 
Our links work well linear, but it is ALL on Stephen's network, so it's not going through the entire Internet to reach it's spot. I just want to note that.
 
yes I should have clarified that. PCM over internet is okay at best. FLAC on comrex access works a little better, still a glitch from time to time. Best case over the public internet is something tcp based like shoutcast, if you can tolerate 30-45 secs of delay from studio to air. I will say that shoutcast is extremely robust and will get through glitch free as long as there is enough bandwidth to keep the buffer full.
 
Best case is something with intelligent way of overcoming variations inherent with open Internet connections. Take a look at Telos Z/IP One, it's designed for that and will dynamically adapt to network connections never to drop audio.


Regards,
Goran Tomas
 
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