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Tieline codec settings

We are considering hardware equipment to replace an ageing "Telelink" software/computer solution that used to feed our audio out to the transmitter, before it recently failed for the last time. What would be the best settings for and STL application for us via Bridge-IT units? Our T-1 is only a 512k pipe. The station also has a tendency to run mp3s on the air from time to time, so having something that not only utilizes all of the pipe we have available, but also cascades well is pretty important. Thanks for the advice guys....
 
OKCRadioGuy said:
We are considering hardware equipment to replace an ageing "Telelink" software/computer solution that used to feed our audio out to the transmitter, before it recently failed for the last time. What would be the best settings for and STL application for us via Bridge-IT units? Our T-1 is only a 512k pipe. The station also has a tendency to run mp3s on the air from time to time, so having something that not only utilizes all of the pipe we have available, but also cascades well is pretty important. Thanks for the advice guys....

I've had good success running MPEG2 J-Stereo at 192KBPS. This has about 1/4 second latency but sounds
really good on the marginal audio we sometimes get from network sources.
 
chriscollins said:
Goran Tomas said:
WNTIRadio said:
If you have 512kb/s then run the full 384k MPEG layer II.

+1


Regards,
Goran Tomas

+2

+3 If you're not running any other overhead traffic on the 512 K dedicated pipe- 384K makes best use of the available bandwidth.

But if you want to control an audio processor, offsite backup a server or use a GUI based remote control on the same T1, you need some headroom and QOS protection.
 
Thanks! So the consensus is to use MP2 then. Have any of you guys used the ACC stuff in this situation? Would it play worse with already digitally compressed audio?
 
In my opinion (and I am sensitive to cascading algorithms)... MPEG2 at 384K is more transparant than AAC, as it is a less aggressive algorithm than AAC. I don't hear the cascading as much with MPEG2. Now on a first gen encode, AAC at 256 sounds better, but doesn't cascade well.
 
There's also much lower delay with Layer II. It's also, at 384k, a more gentle codec than AAC. Less of your audio is getting mangled, and it does cascade better than AAC.

If you were using an unregulated IP link (such as the internet), I would go with AAC because it can handle a streaming type environment better than Layer II. Layer II likes dedicated bandwidth and a stable connection. That's why it works well for satellite downlinks, ISDN and over T1 lines.
 
Thanks. It sounds like the "Basic" Bridge-IT will do what we want without ACC. If L2 still is the better option for dueling algorithm issues, luckily that's included in the basic unit :)!
 
Amen (Layer 2 at 384), and use "stereo" not "joint stereo".

Or go all out and use enhanced APT 24 bit @512. This way you can have the processor at the TX site and not have to have protection for the STL when board ops don't do their job.

WNTIRadio said:
If you have 512kb/s then run the full 384k MPEG layer II.
 
Hi Rolf! When using enhanced APT 24 bit @512, would I not have overhead that would require me to use a lesser setting? We are only using this T1 for the audio. This may a pretty significant improvement of quality if we could really use a 512k audio stream :)!
 
OKCRadioGuy said:
Hi Rolf! When using enhanced APT 24 bit @512, would I not have overhead that would require me to use a lesser setting? We are only using this T1 for the audio. This may a pretty significant improvement of quality if we could really use a 512k audio stream :)!

Not sure i understand your question. APT-X is 4:1 which is what Layer 2 does well (it can be stretched further but I don't advise it). So yes, APT-X can do 384 KBPS as well. But if you have the bandwidth it can do 24 bit.

Did you have something else you wanted to use the T1 for? If so 24 bit probably isn't warranted. The nice thing about the 24 bit is you can locate the processing at the TX site and not have worry about protection processing for the link.

Maybe Tony Peterle can post the various bandwidth, sample rate, and bit rates for enhanced APT-X. I no longer recall the exact details but it is handy to know what the full range of options exists.

cheers

Rolf
 
Hi. The T1 is for audio only. There isn't anything else that will be riding down it other than the Bridge-IT units connecting to each other. Since the total bandwidth available is 512k, I guess what I'm asking is what format would be the best for repeating already cruded up compressed audio and what format would be best for utilizing the full 512k available. I think most fomrats have some extra data that is used for handshaking purposes. That's the only thing other than the audio data that would be on the circuit. With L2 I'm assuming I'd have to stay around 380 to accommodate the handshaking data. Would I get more actual throughput with the APTX?
 
L2tops out at 384k. You won't hear any perceptible difference between input and output at that rate.

24 bit APTX is excellent too at 512k.
 
Both of the above options are refering to synchronous codec streams. These would connect to a T1 via a V.35 or X.21 connection to a CSU/DSU that connects to the synchronous point-to-point T1. In this case you can use all 512 kbps.

If this is a T1 to a ISP then we are talking a completely different beast. In that case there will be an edge router and you would normally connect to that using Ethernet. In this case there is packetization overhead in addition to the raw codec bit rate.
 
Hmmm. This would be via IP. Interesting. Maybe we need to reconsider what we are using. Something with the correct interface would give us more throughput I guess. Thanks guys!
 
Is it an IP to an ISP or a point to point IP T1 line? I had to specify frame relay when ordering them so the phone company doesn't automatically put in an IP line.

They're used to tie two offices together on a private line instead of the public internet. In these parts they're referred to as "MPL" metro private line.

If there's no ISP and the public internet involved, it should work fine with little overhead.

Have you tried the 384kbps MP2 yet and given it a listen test?
 
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