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[quote author=”lpy”]I guess it’s just what I’m used to, although one difference I can think of is, when Winamp is running with a good crossfade plugin (the sqrsoft version for example), the plugin can "hear" the processing of the audio and works accordingly, whereas in its current form, it doesn’t affect any operation of Winamp’s plug-ins.[/quote]
Yeah, I see what you mean there. Although I prefer to have the cross-fade points relate to the dynamics of the music THEN get processed, as… to me… the fades seem to make a lot more sense, and are smoother and often longer.
If you load a DSP directly within SqrSoft it will process the audio AFTER cross-fading (and also use way less cpu), so i’m guessing that you’re running VL as a dsp in Winamp’s normal DSP section (or in a DSP chainer in that).
Oh and by the way…. 😉
At the discretion of Leif, and I’m pretty sure he will do it…. is he’s able to be including some new presets I’ve designed. 🙂 Namely Zenith Light. And a new preset series of mine called Helix. There will also be a Light version of that.
Helix is designed to have decent amounts of short term dynamics, but over the long-term average it’s VERY aggressive. Even more aggressive than "Magnifying Glass" with 75% power!! Except it sounds a lot better than doing that. And it’s also a 7-bander.
I’m also working on another new series of presets which I’m going to keep under wraps for now, due to the experimental nature of the idea I’m pursuing… it actually kind of "abuses" the Breakaway engine in some regard. 😉 So I want to make sure the outcome is solid before I say that anyone is ever going to hear it. But if it works, it should make our "Hi-Fi" listeners pretty happy.
Anyways, just thought I would let you know.
XP can be extremely stable though. When I worked for "the #1 net-radio station and network" I managed over 200 production machines running XP SP1, and except for the "bug" with the Windows Audio Service, which made it so the machine just would NOT make sound at all after exactly 365 days uptime 😆 I would have had all 200+ machines uptime OVER 5 years without rebooting, before they switched to a new datacenter and new machines.
What’s the last linux distribution you heard of with even 365 days uptime, for over 5 cycles, without any problems whatsoever (on 200+ machines)? The only problem at all we ever had was 2 hard drives failing, which is perfectly acceptable with over 200 1U servers running non-stop for over 5 years.
At any rate, I’m sure there’s a few linux distros that could pull it off, but… XP is hangin’ tough right there with em. Windows2000 also isn’t half bad, although I’ve never tried to run one that long, I bet it could. But it wouldn’t be a candidate for running HDFM because it doesn’t support 192kHz sampling-rate. 😉
Leif just got back from the states. 🙂 The Linear Acoustic product will most likely stay under the Linear Acoustic brand name – but I can’t speak for Telos, that’s their business.
As far as AEROMAX-HDFM becoming an Omnia product, that’s something that hasn’t happened yet (and may not happen). That’s why Leif is considering releasing it himself as Breakaway-HDFM (exact naming semantics unknown to me, that’s what i’ll word it for now, so don’t quite me on the exact naming convention)
Honestly, again… I can’t speak for Omnia either, but it’s probably something which wouldn’t be smart for them to take on right now, even though they probably could get it to run on the OmniaONE platform, because the quality of the processing is MUCH BETTER than the Omnia6EXi – and undercutting that market for them is probably not an option. 😉
If Leif releases Breakaway-HDFM himself, they don’t really have anything to do about that, other than try to compete. But they would be foolish to do it to themselves.
😉fragmented and noisy… sounds like a buffer issue, not a "not working issue" in which case you wouldn’t have any audio at all. (and windows would say that the driver is not functioning properly)
As far as getting the Vista64 drivers signed, this has already been answered a few times in this forum (did we use the search function before posting? 😉 )
[quote author=”knightrider”]I took a look at the description of the MP3Gain system – it got me wondering – does that system just flag the file in the tag information – giving an instruction to turn the volume up? Or – does it actually edit the file itself? I suspect that it flags via the tag. That might be the problem – if something in your audio chain fails to read that instruction – the cut will playback at the original low level…..????[/quote]
mp3Gain will do a few things. First off, it’s using the ReplayGain algorithm to read the track, to decide it’s actual "loudness".
Then it will figure out the difference in relation to the reference db level you’re asking mp3Gain to make all of the mp3s you’re going to be processing.
After having that all figured out, when you click the button to apply the gain settings… It’s doing three things.
1. it changes a scale value built into all mp3s, which does NOT re-encode the file, but only changes how loud it will decode. and this is the function that works in ALL mp3 players of any kind. this is really where the magic happens. 🙂
2. it’s setting/changing ReplayGain values in an APE tag in the mp3 files, which is in place partly so that ReplayGain-capable players don’t try to figure out the gain and then change the gain, and also so they don’t bother to write the ReplayGain values, since they are already there. This also allows the player to use it’s own reference playback loudness, without any analysis.
3. it’s setting the mp3Gain correction values in an APE tag in the mp3 files. This will allow the mp3Gain modifications to be undon in the actual mp3 file’s scaling as well as reversing the changes in the ReplayGain APE tags, so that ReplayGain-capable players will continue to properly adjust the gain to whatever the reference loudness is set in that player.
Pretty neat eh? Actually, mp3gain will sound more transparent than normal ReplayGain, the more extreme the gain changes in ReplayGain are… because ReplayGain-capable players are altering the gain after the mp3 has been decoded back into PCM, but mp3gain is altering the gain *BEFORE* decoding into PCM, so you don’t lose a single bit of accuracy. 🙂
[quote author=”jimwms”]You can’t forget the CBS Volumax and Audimax.[/quote]
and the Conax, and 100 other great pieces of gear. It’s really quite staggering how much innovation has happened in such a short time from the days of Tesla bringing electricity to the world. Not even 100 years ago.
[quote author=”Olli”]Therefore i thought, it could be a problem with the bass. When the speakers can’t handle frequencies below, let’s say, 80hz and the overall loudness is always the same, there should be a difference in effective loudness between songs without much bass and others with a lot of sub bass, shouldn’t it?[/quote]
By "can’t handle" I think you mean to say "can’t reproduce". And then yes, you’re exactly right. When an audio signal’s dynamics have a higher average to peak ratio, the timbre of the sounds becomes a lot more important. Actually, any well produced, well mixed, and well mastered recording will take timbre into account for the vast majority of it’s emotive rise & fall, and compensating for certain elements they want to continue to "stick out" in the mix when the averages do rise up.
That being said, when the reproduction system can’t reproduce signals that fall within the frequency ranges that are important to accurately convey the recording’s emotions & whatnot… then it will seem to have less impact, or more of an average impact than it would have if the system was able to fully reproduce the recording.
This holds especially true for classical music which relies HEAVILY on timbre to convey it’s emotions, it’s ups & downs. Because of this, and the very high quality of some classical recordings, classical music is often used to audition and test the highest-end playback systems. To a very skilled listener, they can use the balance (or lack thereof) to judge the accuracy of the system quite easily. Pipe organ music too, especially for bass. 😉
man, Gates Sta-Level user in the house… step aside people… VIP comin through. 😉 lol, seriously though, that’s some old and revolutionary at the time hardware. 🙂 1956 wasn’t it? With the Level Devil in 1959?
Anyways… glad to have some people around who not only know… but have experienced some real history. 😀
the audio processing engine itself already supports it for years now. it’s just a matter of Leif getting enough time to finish an end-user version of it, and also the timing of releasing it has to match what-ever the Leif’s plans are. 8) i could be wrong, but technically the engine should only be limited by CPU power, and actual soundcard I/O (if needed)
🙂
to change volume on mp3s reliably, and losslessly (including ability to undo the change):
http://mp3gain.sourceforge.net/as far as the proper way to use breakaway with winamp, or any other software that provides a volume control… the software’s volume control should ALWAYS be set to maximum output.
you must use breakaway’s volume setting to control the output volume of the soundcard, after your soundcard’s physical master output (not wave out) has been properly matched with whatever system it’s running in to… be it an amp, or small crappy speakers, or headphones.
this way Breakaway’s volume knob at 100% should be as loud as you will ever want to turn up the sound, and then turn down Breakaway’s volume setting to where-ever you happen to want to listen at the moment.
Try using "Safe Mode" (for the method it uses to access your converter/soundcard) instead. I have no idea if that will work, but from what I’m thinking of it makes the most sense for the next thing to try.
It could be a number of things. Ots is historically not so stable on some machines (and more-so OS configurations), sometimes having nothing to do with the soundcard. I would suggest trying to have Ots support you so you can at least get Ots stable on your machine, without routing through Breakaway.
When you get that taken care of, it should be no problem to use Breakaway. Clearly the skipping problems are from Ots, and not Breakaway.
A few things to try in Ots to possibly stop that from happening… Set your mic input to the same soundcard, and disable the "open soundcard" option, so that it doesn’t open the input even if you’re not using it.
Also try switching from the DirectX output, to the "Compat" output (aka MME).
If your converter (aka soundcard) has the option of setting the sync to be external or from a digital input OR using the internal clock… try using the internal clock — unless you’re running digital audio into the converter of course, and in that case you would want to sync to the input card, and try different sync methods on that card’s master clock. Trying internal first.
Also try running Ots in "Highest" priority (*not* realtime) and see if it has any more stutters.
It appears that Dolby Volume is going to be (it’s not out yet) only a 3-band AGC, and it appears to have basic gating (perhaps windowed/targeted) and doesn’t use digital sample peak levels for it’s control signal (probably weighted RMS, with some kind of transient-induced attack/release modifier)
But yeah, it’s still nothing new, and if you understand enough to "peer into" their marketing fluff (which is the only information available right now)… Then you can understand how Breakaway is superior in every regard, and much much more.
In my opinion nothing is going to beat Breakaway at the "game" of emulating analog-styled processing. The next processor that sounds better is going to have to leave all of that behind.
Even processors like IDT’s "DVP" platform are still half analog-emulation machines at heart, as revolutionary as the processor was at the time and still is. And it still has a hard time competing with Breakaway even for DAB, but thanks to Leif’s AWESOME FM limiting & clipping, *NOTHING* can touch Breakaway on the FM dial. Not even close.
But only time will tell. I think eventually the newer methods of audio processing, the ones that don’t emulate analog but enter into the realms of math, are going to eventually blow away the abilities of the processors we have now. But that will be at least another 10 years.
