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I simply meant I was using a build of 0.90.94 that includes the speaker controller fixes. 😉 Not the final 0.90.94, because it’s not until it’s on the site. Please don’t make me regret that I’m open with some of the information I share. 8)
You can run the cores in speaker mode if you want, check that out in the main setup window. This will give you a bunch of speaker controls so you can filter (like a crossover), delay, invert, safety limit/clip, etc…. plus a 20-band parametric EQ… on every output. 🙂
Breakaway blows away the Solidyne, much clearer and undistorted on air. The only comparison I’ve heard though is versus an Omnia 6 EXi, and even there… the Solidyne demo gave itself an unfair advantage by having way less loudness for the Omnia. When that was compensated for, the Omnia won out – not by as large a margin as you would think… but the Omnia 6 is an aging box, and Omnia has a new box in the works which will probably be pretty stellar from what I’ve heard of it, and knowing who’s working on it.
Solidyne has MPX Tool licenses given to them, in the hopes they would do some fair comparisons by running the torture test and sending the MPX… They said they would, but so far – in some 2 years now i guess – that hasn’t been done. Draw your own conclusions as to why that is. 😉
Far as Embedded, that’s that way to go. The Breakaway hardware runs on XP Embedded and all of the "Leif-related" Linear Acoustic gear as well. Not sure on 2011, but I don’t see why not other than maybe ASIO drivers, if you are using Breakaway ASIO.
define "better" i guess then.
what decoder and DAC and everything else are you using to test this? what probability of randomness are you scoring in your blind ABX testing between the two, and from how many iterations?
Yeah, personally… if I was going to code anything to lossy, I wouldn’t use anything but the latest Lame, set to:
-V 0 -q 0
8)[quote author=”timmywa”]iTunes 9+ can stream AAC+ just fine. I was excited to hear they made that improvement!![/quote]
I think they meant stream… as in… the source side. 😉[quote author=”Q106″]However, although HE-AAC at 48 kbps sounds best of all other codecs, that does not mean it sounds excellent. Again, it’s the best codec for very low bitrates, but the sound quality is a compromise. It does have wide frequency response at the expense of metallic and gritty high-end, due to the artificial nature of SBR technology.[/quote]
You should check out the latest CT builds coming from Nullsoft then. Not only PS-AAC, and tons of recent improvements there lately, but the filter design has also been improved a lot lately and it sounds way less gritty/metallic since sometime last summer. Nullsoft releases a new build of their encoder DLL which has a new CT build every few months or so.Even 24kbps seems acceptable to me now, as a low bitrate solution like for mobile phones, or if you just need to do some cost/bandwidth savings.
[quote author=”iguman”]but with this resistans i use volume in my card is on maximum[/quote]
Then the exciter’s input is NOT a 50 ohm load. It’s probably more like 10,000 ohms… but this is something you still haven’t answered yet.Again, the outputs are designed to work with certain loads, but ARE NOT rated in resistance because they are not supposed to be loads. So please refer to them as "designed for a N ohm load" or something like that, where N = the load in ohms.
It’s not as annoying or stupid as people thinking that a polarity switch somehow "reverses" phase. I’ll give it that. 😉
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So, iguman, what adjustments do you have on your exciter, for audio gain? Try no additional load on the soundcard, soundcard cranked, and then adjust the exciter’s controls so it’s the modulation is correct?
Yes, two processes. One for the audio engine, and one for the GUI. That’s how the GUI is freezing, but the audio engine is running fine. The audio engine is running with realtime priority, and the GUI isn’t.
[quote author=”audiofreak”]* a 2 band AGC like orban 8300/8400/8500… for a constant fine bass[/quote]
actually… 2-band AGC isn’t as consistent overall. but something else to think about… the multiband itself has two bands in the bass where Orban has just one band. so it’s possible to get more consistency with Breakaway than with an Orban.[quote author=”audiofreak”]* a good wide band stereo anhancer like the orban 222a…I do not use the 4 band stereo anhancer in bbp..He does everything or nothing…mono / stereo ratio is poor ):[/quote]
there’s plugins that emulate that, i forget the one which but someone else remembers. search the forums, it’s already been posted here a few times. this is off topic. 😉Full control comes when the Breakaway hardware is released. The price tag is going to be into the thousands of dollars. How much I shouldn’t say, but there have been some changes that have allowed thousands to be taken off of the cost of MAKING the box itself, so… You’ll just have to wait and see if it comes this year or what.
I can tell you for sure though… You’ll want to save up for one. Breakaway already is the hottest cleanest sound on the dial, and the Breakaway box is even more hot & clean. Especially in 75uS markets.
Stationx,
Nobody said the Barix sounds rubbish. They simply stated that you *can* get better quality, in many cases, by eliminating the un-needed DAC/ADC process between the two, and to take it right into the computer with Winamp. Secondly… Winamp already does what you describe the Barix doing with switching between live feeds and local backup audio.
The problem Joop Krauthausen presented with the Barix was because of misuse. Did you notice Joop mentioned they were running the already pre-emphasized & clipped audio through it, with their broadcast processing on the encoder side? Thought you would pick that up from the fact they cared about peak control at all. That’s the problem there, and has nothing specifically to do with the Barix unit.
"Winamp will stop every now and then" is also unfounded. I’ve operated 200 machines running Winamp before, for 4.5 years at one particular NOC, with no issues whatsoever with Winamp. This is just my personal experience. Your personal experience in at least 1 case (you didn’t specify) Winamp stopped at least 1 time and you never figured out what caused it or if it was even Winamp that caused it. Are you aware that Windows NT itself (at least up to XP last i tested) has a bug in the audio subsystems where the machine will just NOT make any audio at all after 365 days of uptime, and you will have to reboot the machine?
You stated that RTP sounds "very close to WAVE quality", where RTP means "Real-time Transmission Protocol". In other words, it is not a codec, it doesn’t define what codecs should be used, nor what the format of audio or video used is at all. You could, for instance, use RTP to send 5mHz DSD or 8kbps Speex. Or send only video. Maybe check out whatever product/s you’re referring to, and what codecs they are using, and then make that part of your personal opinion.
Don’t mean to pick on you, but if you’re going to state something technical as an objective fact, not subjective personal opinion, then please back it up with some proof that stands up to applicable scientific protocols.
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Joop, you need to have a lossless unfiltered STL, or you MUST run your audio processing at the transmitter. Any lossy codec will cause the loss of peak control, it’s not anything’s "fault" including layer-3 or the Barix.
