Leif

Yes, one. Breakaway Live sounds better. Night and day, actually.

At the end of the day, that’s what matters the most is it not?

///Leif

Exactly.. You’re playing dance music, and yet it doesn’t sound very good with the Dance preset. Hence, not a very good preset 😉.

To be honest, I just didn’t spend that much time on it. In the beginning, I had just finished the algorithms and the rest of the program (all the i/o, user interfaces etc), and needed to get something out there. Some of the presets were very good — others were quickly made, as it wouldn’t look very good to ship even a beta with 5 presets 😉. It has come a long way since then though. At this point, there are many excellent presets, and most of the ones that were thrown in there in the beginning are already gone.

It’s not a matter of keeping the program small (presets are tiny), but rather, not having a ridiculous amount of presets in the list.. Better to only keep the ones that are actually good, and thus maintain a higher average quality. 🙂

Definitely will not delete too much, but at this point there’s already 28 presets. I guess 30 would be a good place to draw the line, but I’d rather keep those two slots open for later.. Rustonium is going to use another slot too in the next release, so that’ll be 29 already.

Best,
///Leif

Hi Scott!

Whoa, 20-50ms??? That’s tiny! Nice, but how is it even possible?

For one, they’re not actually using aacPlus.

I just checked the aacPlus codec delay (by initializing an encoder library and asking it).. it replied 4280 samples.

That assumes that our audio is already in 2048-sample blocks, which is the aac frame size. If it’s not, we have to buffer to build up a whole frame of data to process, so that’s 2048 + 4280 = 6328 samples! That’s 130ms of *algorithmic* delay, at 48000hz sampling rate. It assumes that we’re encoding and decoding on an infinitely fast computer. If we have to go through sound cards, out on a network (even a local one), buffer to compensate for block timing jitter (which you have to do in windows) etc, delay just keeps going up and up.

20ms of latency is *extremely* impressive for taking digital audio from one place to another through a low bitrate codec. I looked up the specs — 43.2kbps in one direction and 28.8 in the other? I wonder what that sounds like for music, since the codec can’t be quite as efficient as aacPlus if the delay is this low.

Also, remember those codecs do not use the public internet — they use a cell network. Over the public internet, I believe it’s (currently) impossible to have reliable, stable hifi audio with less than a few hundred milliseconds of delay.. The internet just isn’t that reliable — it has no delivery guarantees.

I wish there was an easier way 🙂.

Regarding Buffer Size, I would just keep it at medium unless there are any glitches. There shouldn’t be any — BBP (in an over-the-air broadcast environment) is really meant to be run on a dedicated system. Computers are already so inexpensive compared to traditional audio processing hardware — why compromise performance by unnecessarily assigning more tasks to the same hardware?

In a web streaming environment, I would just use Huge, since delay is already several seconds.

Best,
///Leif

Hi Jeff!

I’m afraid it will be a while before I make those plugins, since I have two very important projects I have to do first.. Right now Breakaway Broadcast and Breakaway Live are in maintenance-mode while I work on those. You’ll like one of the projects I’m working on though 😉.

Vriendlijke groeten,
///Leif

I couldn’t agree more. I guarantee a properly configured BBP alone will sound the best.

However, being an honest engineer, as opposed to an ad-exec, I can’t quite leave it at that — you do speak about having a signature sound you want to keep.

The way I see it:

*Compressors* (that is, multiband, agc, levelers) are subjective. One can put labels one things (pumping/spectral balance shift/etc), but some people like some or all of those things, whereas others don’t. So, to each his own, and rightly so.

*Peak control* (that is, clippers, look-ahead limiters) is NOT subjective. Distortion is distortion — it’s an artifact, it’s highly objective, and it’s always bad. Anyone is free to claim otherwise — just like I am free to question their hearing. 🙂

My point with all of this is:

BBP does peak control more cleanly than any other processor.

So, if you wanted the "sound" of the Omnia One, without the clipping distortion, then you could run the Omnia One with HD software (not FM!), and Limiter / Clipper Drive set much lower than you normally would (although make sure the bass clipper is in line and doing its job).

Then, feed the Omnia One output to BBP, and use the Protection Clip preset, adjusting Final Drive until you have the desired loudness.

This way, BBP will not impart anything subjective onto your sound — only the near-distortion-free clipper will be used.

You could do the same with any front-end processor, even an Omnia 6 or an Optimod 8500. Kind of ironic to relegate a $10,000 box to function as a front-end for a $199 software processor, but there you go. 🙂

Front-ends are subjective, but back-ends are not. Race them any day — you’ll see what I mean. Make sure to include some saxophone, telephone and xylophone. 😉

Best,
///Leif

E-mail sent. 🙂

///Leif

Very strange indeed! I see three distinct machines, three different hardware ids, mac addresses etc.

Logging only happens on successful activation, and activation only happens after the application is started with a valid key. So, if the key isn’t accepted, I really don’t understand how the activation process would get started.

///Leif