BBP Setup

[Anonymous]

First thing I wanted to say is that BBP is very impressive processing software, unbelievable that it can be achieved on a windows computer.
I’m using the plutonium preset with the punch and bass plugins.
Everyone who listened to the sound thought is was an Orban or Omnia but definitly not software 😉

But I was wondering how to setup BBP in our situation.

We are a local station and the studio is about one mile away from the transmitter site.
We transport the audio in PCM 24 bit with a 5Ghz wifi connection from studio to transmitter.

What I wanted to do is to place a PC with BBP in the studio and one at the transmitter to do the finishing touch.

Do I set BBP in the studio Pre-emphasis 15us and De-emphasis on ?
And at the transmitter use the Protection clip setting ? and ofcourse use the MPX output of BBP.

Greetz,
Otto

[sam]

Hi Otto,

If I was you i wouldn’t run any processing before your 96 KHz PCM link as that should already sound nice and clean, and just run BBP at your transmitter. There’s no point running 2 versions of BBP in series or any processor if you can avoid it. I would run BBP Live in the studio to feed your speakers etc.. it will offer better latency then listening off air. Can your STL operate at 192KHz? then you could run BBP in the studio and feed the MPX via your STL.

Sam

[JesseG]

[quote author=”Otto”]unbelievable that it can be achieved on a windows computer.[/quote]A modern computer could probably run 10 Omnia6’s on it. 🙂

[quote author=”Otto”]Everyone who listened to the sound thought is was an Orban or Omnia but definitly not software 😉[/quote]Even though an Orban or Omnia actually is software too? That’s a good sign that you have been a victim of marketing. 😉 In a way I know what you mean, but all digital audio processing is in fact software, and it goes to show how people’s thinking has been skewed by marketing… and Omnia and Orban are hardly the only companies responsible for that.

[quote author=”Otto”]What I wanted to do is to place a PC with BBP in the studio and one at the transmitter to do the finishing touch.[/quote]The argument against that is you’ll be limiting & clipping the audio twice in the final stage. That will cause distortion to happen, and you won’t get as loud or clean an output from the final BBP as it’s FM backend will compensate to try not re-distort those clipped waves as bad. Remember clipping IS distortion, BBP just does it with control never before heard.

I would take sam’s advice and think about using Breakaway Broadcast at the transmitter, and using Breakaway Live in your studio. Live will just process your studio monitors/cans sound, with MINIMAL latency (about 5ms total with a decent ASIO capable AD/DA converter). And BBP will just process the transmitter. The STL will be fed with the same audio being fed TO Live in the studio, not fed the output of Live (as that would double process, and you would get the same distortion product i mentioned above too except more IMD vs THD)

[Anonymous]

Hi Sam & Jesse

Yes I know that the other processors are in fact also software (algorhytms).
Instead of using Intel Pentium they are using Motorola DSP’s.

But the bigger stations here they have for example an Orban 8500 in the studio and a 2300 at the transmitter…
What is the reason behind that?
If I understand correctly, BBP clipping is audible on the normal output and not only on the MPX out?
My idea was just a littlebit of clipping to compensate any overshoot due to the STL.
But if it’s better to just send pure audio (unprocessed) to BBP at the transmitter than that’s an option for sure.
But how to control the outgoing level to make sure that it isn’t clipping at the input…

[JesseG]

[quote author=”Otto”]But the bigger stations here they have for example an Orban 8500 in the studio and a 2300 at the transmitter…
What is the reason behind that?[/quote]I would have to say that they don’t know what they are doing, and that their audio quality is without a doubt suffering because of it.

[quote author=”Otto”]If I understand correctly, BBP clipping is audible on the normal output and not only on the MPX out?[/quote]Any clipping at all is audible to the trained ear. The thing Breakaway avoids is audible distortion. Hearing clipping and hearing distortion artifacts of clipping are two different things. But yes, the FM backend processes in L/R, and that is available for the L/R outputs, then it goes on to the stereo encoder and is optionally mixed with RDS input, and that is available for the MPX output.

[quote author=”Otto”]My idea was just a littlebit of clipping to compensate any overshoot due to the STL.[/quote] Think about it this way… Breakaway (and any other audio device of any kind… digital or analog) only has so much headroom before it runs out.

In the digital world, 0db is the max, so there is a common "headroom" of "none" without compensating to adjust for that. Let’s say we’re compensating by considering 0dBu to be -12dBfs. So now we have 12dBfs of headroom above 0dBu. Whatever your broadcast engineering team has decided is best for your setup and the way you work should be what is stuck to, unless it’s not working of course. Either way, a standard for where your studio’s 0dBu is at in relation to dBfs should be defined, if you guy have not done so yet. I wouldn’t recommend going with less than 12db headroom. The "old" analog studio standard is +24dBu headroom, which is a LOT of headroom, but they also are not using such high gains and often better equipment (and noise reduction) than many broadcast studios.

Anyways… that being said…

If there is an "over" the headroom sent down the digital chain, does it matter if it hits Breakaway 1st or your STL 1st? Well it depends on how your STL handles clipping. You did mention that you were considering using a "Protection Limit" down the line to compensate for overshoots from the STL?? If you would have to do that, then your STL is just not lossless, it would HAVE to be doing something to the audio, and that is all the more reason to have Breakaway Broadcast at your transmitter.

[quote author=”Otto”]But if it’s better to just send pure audio (unprocessed) to BBP at the transmitter than that’s an option for sure. But how to control the outgoing level to make sure that it isn’t clipping at the input…[/quote] Imho it is better, you can always put a remote control application (i recommend RealVNC Enterprise) on the host computer and then it doesn’t matter where it is (or for that matter where you are) at to be adjusting your on-air sound.

As far as controlling outgoing level, that’s the mixer’s JOB to do. The only thing you can do is allow some headroom for mistakes before the audio processing. Putting something like a Compellor before your STL will NOT make any problems go away, in fact it can just make them worse, because a Compellor eventually runs out of headroom just like everything else AND it’s applying gain (which will change your sound, and it’s NOT remote controllable with VNC 😉 ) and possibly contributing to distortion. See what I’m getting at?
💡

[Anonymous]

[quote author=”JesseG”]I would have to say that they don’t know what they are doing, and that their audio quality is without a doubt suffering because of it.[/quote]
Ok that’s clear… It’s probably because they use a STL that is not lossless.

[quote author=”JesseG”]If there is an "over" the headroom sent down the digital chain, does it matter if it hits Breakaway 1st or your STL 1st? Well it depends on how your STL handles clipping. You did mention that you were considering using a "Protection Limit" down the line to compensate for overshoots from the STL?? If you would have to do that, then your STL is just not lossless, it would HAVE to be doing something to the audio, and that is all the more reason to have Breakaway Broadcast at your transmitter.[/quote]

This means that in our way, sending the audio in PCM, we don’t get any nasty artifacts…

[quote author=”JesseG”]Imho it is better, you can always put a remote control application (i recommend RealVNC Enterprise) on the host computer and then it doesn’t matter where it is (or for that matter where you are) at to be adjusting your on-air sound.

As far as controlling outgoing level, that’s the mixer’s JOB to do. The only thing you can do is allow some headroom for mistakes before the audio processing. Putting something like a Compellor before your STL will NOT make any problems go away, in fact it can just make them worse, because a Compellor eventually runs out of headroom just like everything else AND it’s applying gain (which will change your sound, and it’s NOT remote controllable with VNC 😉 ) and possibly contributing to distortion. See what I’m getting at?
💡[/quote]

😛 It’s funny, we talked about a Compellor to level the output.
But indeed, if we maintain enough headroom we don’t need anything in front of the STL
and there won’t be any problems…
And we allready work with VNC to operate remotely 😉

Well anyway thanks for your advice….

[Leif]

Hi Otto!

I agree with Sam and Jesse’s assessments.

A couple of questions though — you mentioned a 24-bit PCM link with a 5 GHz wifi connection.. Does this mean Two PCs with wifi cards talking to each other, or are you using some kind of hardware to accomplish this?

If it’s two PCs, then there is really no reason to run BBP at the studio, because you will still have a PC at the transmitter.

However: If your STL is hardware, AND your exciter has a good stereo generator built in, then you could actually run BBP at the studio, and completely avoid having a computer at the transmitter.

The peak control (clipping) in BBP is done entirely on the L/R channel — there is no composite processing. Thus, you can use an external stereo generator and get the exact same result, provided that the it’s a good, clean stereo generator which does not do any additional filtering.

The very easiest setup, though, is to have BBP in a stable computer at the transmitter. That way, you can use the MPX output and connect it directly to the exciter. In this case, there is no reason to do ANY processing in the studio — simply keep the level low enough to avoid clipping before the audio processor.

The reason why many networks are using an 8500 at the studio and 2300s at the transmitter is two-fold: price + lossy STLs.

With lossy STLs, you can’t maintain peak control over the STL. You could solve this by running unprocessed audio through the STL and having an 8500 at every transmitter, but the 8500 is a very expensive processor, and it (as evident) becomes too great of an expense even for major radio networks. Running an 8500 at the studio and 2300s at the transmitters is considered the "next best thing", but it’s quite significantly worse than having an 8500 at the transmitter instead.

BBP makes this a moot point. It’s cleaner than any other processor, and inexpensive enough to run on every transmitter site. Problem solved 😉.

Best,
///Leif

[sgeirk]

You just sold me, Leif! 🙂

In just about any situation, I’ve experienced superior results by having the SG and processor at the transmitter .

[Leif]

Thank you 🙂. I’d better stfu then so I don’t start preaching to the choir. One thing I’ve gotta say though:

I made some experiments with lossy codecs and MPX a while back — trying to find a way to reduce the bit rate of a composite baseband signal (for MPX-over-internet, for example).

It turns out that:

Using MPEG-1 Layer 2 at 256kbps 48 kHz Stereo, but actually running the codec at 4x speed (1024kbps, 192 kHz) and then feeding only the Left channel with audio, yields acceptable results for MPX: Only 1% added overshoot, and still 40dB pilot protection.

1024 kbps is a pretty high bitrate, but that’s actually 3:1 compression compared to the original bit rate of 192 kHz (3072kbps).

(Although, if we were to resample the original MPX to 128 kHz sampling rate — which would be lossless from the MPX’s point of view — we’d have 2048 kbps, so it’s really only 2:1 compression).

It was an interesting experiment, but one important fact makes it a moot point:

You would still need a computer at the transmitter site to receive and decode the signal.

So at that point — just load BBP into that computer instead, and be done with it already 😉. Then you can use any STL — analog, digital, bit rate reduced; all of a sudden it doesn’t matter that much anymore, because peak control is no longer an issue.

For a dedicated, solid stand-alone PC, Windows 2000 on a Solid State drive, or Windows XP embedded on a plain old USB flash drive, makes for a nice and robust appliance.

Best,
///Leif

[Anonymous]

Hello everybody,

Thanks for the explanations, it’s getting quite clear now.

We are going to use the Barix 1000 or Audio TX.
Only the Barix can deliver audio in 24/96 with Audio TX it´s 16/48 and we need a computer
at the transmitter.
I yet don’t know which we are going to use for sure.

For the wifi connection we allready use two Motorola Canopy units.
With a constant throughput of about 7Mbps, and have a Barix exstreamer 100
at the transmitter, using a 320kbps MP3 stream.

But if it’s correct that if we use two Barix units and therefore keeping BBP in the studio
so we don’t need a computer at the transmitter than I think that’s the way we should do it.

Thank you all for your advice..

Greetz,
Otto

(oops, I just saw that the Barix is also 16 bit, I think I didn’t read it right)

[sam]

Hey Leif, and running running BBP at your transmitter site, would be cheaper than a lossless hardware audio link. Especially if you have more then one transmitter location.. IE; 40 😉

Otto, man I have been down the same road and the difference is like chalk and cheese. Once you set up BBP at your transmitter and correct any tilt, tune your favourite pre-set, set your levels etc.. you will have the best sounding audio and even the loudest if you want it! 🙂 Well you will at least untill your competitors start running it 🙂 But then that’s good for the ears of the people and Leif’s wallet lol 😉 Have you seen Leif’s MPX tool software? If you haven’t you should take a look, Connected to the right receiver and set up correctly you will be able to align your levels and make the most of your FM bandwidth (don’t waste it) and to make sure your station is legal within it’s deviation limits. (if you didn’t already know 😉)

PS: Compellors lol I don’t use mine any more, Its still a classic but.

Sam

[Leif]

Otto, 320kbps MP3 codec will cause 10% overshoot on a peak-controlled signal, so it cannot be used.

Only 384kbps MP2 (in my test setup!), or LOSSLESS (PCM), has acceptable peak control. MP3 cannot be used at any bitrate.

That is, if you use MP3, it must be before the processor, not after. 🙂

///Leif

[sgeirk]

Actually, on our main FM airchain, we’re still running analog up to our digital STL on the roof, due to a wiring constraint in our building. To protect the input of the a/d converter, we’re using a compellor running between 6-8db of g/r….set to leveling only and slow…it’s very unobtrusive. I’m pretty picky.

It works well with the setting in my O6exi.

[JesseG]

[quote author=”sgeirk”]Actually, on our main FM airchain, we’re still running analog up to our digital STL on the roof, due to a wiring constraint in our building. To protect the input of the a/d converter, we’re using a compellor running between 6-8db of g/r….set to leveling only and slow…it’s very unobtrusive. I’m pretty picky.

It works well with the setting in my O6exi.[/quote]

I agree about the 6. The 6 is NOT happy when the input gets anywhere above like -9db with some presets, especially for voice. It should not be like that.

Breakaway doesn’t have this problem at all.

[knigget]

I would like to experiment with MP2 – where would I find the codec?

Any help is much appreciated

[Author]

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