bitflipper

My first amp was a Kustom in "Cascade", a sort of metallic turquoise pleated naugahyde. 100W head and 3 (!) 15" Jensens. No horn. Not a great amp for a Vox Continental, but everyone in the band had matching amps, including the PA, so at least it looked cool.

OK, got it. I'm thinking in terms of how I use tempo changes, which are never that severe. It's not just MIDI-driven synths that are affected by fast tempo changes; tempo-synced delays also have problems. So I generally avoid them. On the rare occasions that I do need a drastic tempo shift, I'll build the ritard into the performance and leave the tempo alone. Again, it's just my own way of doing things - all my MIDI parts are played, rarely hand-planted into the PRV. But regardless, the answer is still no, you cannot specify offsets in fixed time intervals. Seems that would introduce CPU overhead as the DAW continuously recalculated the number of ticks required to maintain a consistent absolute time. Tempo resolution can be as low as 3 ticks, so you could be recalculating offsets 320 times a second -assuming you haven't changed the number of ticks per second. I'd be looking at why MIDI timing offsets were needed to compensate for a sample library in the first place, and if there might be another way to approach it. Many synths let you change the starting point for sample playback, for example.

Wouldn't the time offset still be wrong after a tempo change whether you specified it in ticks or milliseconds? Assuming, of course, that the "negative delay" required by your sample library is dependent on tempo in the first place. Usually, when an offset is required, it's to compensate for a baked-in slow attack time in the samples, which would remain fairly consistent across a wide range of tempos. Just thinking out loud, as I've never had an issue with MIDI offsets. I usually set the time offset by ear anyway.

Well how 'bout that? I had no idea. Not sure why you'd want to do that, but good to know. Are you next gonna tell me I can load a ProChannel module into the fx bin?

I agree. When you're seriously into recording, you naturally become very nit-picky about even the most trivial of factors ("should I choose the linear-phase option on this tambourine EQ?"). Some of that attention to detail is bound to leak into other activities ("what, your bedsheets are only 400 thread count? Amateur!"). Truth is, in a live performance all you gotta do is make sure your beer doesn't rattle off the bass cab. (True story: a bartender once thought it was a good idea to put a Jack 'o Lantern atop my Leslie. I was too stupid to object, and midway through the second set the bottom rotor and amplifier were splattered with pumpkin guts, resulting in a repair bill that cost more than the gig paid.)

Thanks, CC. I wouldn't have even thought of Boss for amplifiers. Every Boss product I've ever owned ran on 9V batteries.

Oh, that wasn't my first computer. Just the first one I made music on. My first computer was a COSMAC VIP testbed. Came with 1KB of RAM; I had to etch my own circuit board to upgrade it to 4KB. It saved data to a cassette tape, sometimes. But it did have a nifty 9" monochrome monitor. And I walked 10 miles to school, uphill both ways.

Not in ProChannel. Of course, you can use anything you want in the fx bin. But ProChannel plugins are made specifically for ProChannel.

My first guess would be that you have a plugin running in demo mode. Nice production, btw.

My first music rig was over the top: 386 with 10 MB drive and 16MB RAM. In the mid-80's that was an enviable setup. The drive was $600 and half the RAM I stole out of a Sun workstation, as I deemed any computer that couldn't make music unworthy of a whole 16MB. Cakewalk 1.0 for DOS, five synths, a 2-track Pioneer and a 4-track TEAC 3340S. Altogether maybe $10-12k or so invested - when I was only making $30k a year. We have it so much better now!!!

Damn, if all tube amps had multicolored light shows inside, I might go back to valves myself! I did in fact have a look at the Hughes & Kettner amps during my research. I've always thought they looked awesome. But at $2500 I'll just buy some LED strip lights instead. Greg, is this your amp? It's considerably less-expensive than the Marshall, too.

Not to brag, but I was once the youngest person in the world. Granted, it was a title I did not hold for long. I like that you can approach CW old-school, as a replacement for a tape recorder if you want to. But if you're a beatz and loops kind of guy, CW will accommodate you just as comfortably. In fact, CW has encouraged me over the years to incorporate more modern production techniques. The result is a hybrid approach that combines old-school print-in-realtime audio, recording MIDI instead of audio, and hand-planting MIDI in the PRV. Even the occasional loop for tedious things like shakers and tambourines. The toolbox just continues to grow. The biggest breakthrough was discovering the hidden variable in aud.ini... DontSuck=1

Hmm. Now that's some food for thought, being as the Fender would be a whole lot cheaper than the Marshall, which will come to a grand with tax and slip cover. Only problem is he seems committed to tubes. Funny, he's 28 y.o. but often says he feels he was born in the wrong decade, being a fan of classic rock and classic tones. I guess that's why he's happy playing in a band with a bunch of geezers.

Good point. I've already warned him he'd better not become "that guy", you know, the one who decides he's the star of the show. Although in my experience it's usually been bass players who are more likely to have that problem. Anyway, his 40 Watts will still have to compete with my 4KW.

Put your glasses on, Ed.

Thanks for lending a few brain cycles to this, Shane. The amp had power and the light was on, but there was no sound at all out of the speaker. I spoke to the guitarist yesterday and suggested that he really needed a better amp anyway, as he has trouble getting enough volume on clean tones. Knowing he's broke, I volunteered to buy the amp myself and let him use it while he saved up enough money to buy it from me. Yeh, I know, that's almost as bad as loaning money to a friend, a practice I long ago vowed never to do again after bad experiences. But hey, if he bails on the deal at least I'll have a nice amp. This is the one we're considering. I watched some YT demos and it sounds quite nice, either dirty or clean. It's got an amp sim built in to the line out that I'll be able to run to the vocal monitors. Two channels, each with its own gain and master volume controls. Closed-back with a single 12" Celestion. Switchable between 20W and 40W.

That's how I do it. Always. That's really the key to recording most things: you can do anything with a clean recording, including changing your mind later. As to "real" vs. digital saturation - yes, it's an emulation and if you'd spent 40 years recording analog tape before moving to digital you'd be able to tell the difference. Nobody else will.

Channel Tools is an often-overlooked gem, probably because users don't initially get what it does or why they'd need it.

What you're dealing with is what's known as the "proximity effect". Basically, it just means that many microphones are overly sensitive to low frequencies when they're close to the source. Such mics typically have a high-pass filter built into them, with a switch. They're typically set to 80Hz, although there is no standard and some mics have two settings. Always use that filter if it's available. There are several ways to address the proximity effect: Sing further away from the microphone, not always practical if you don't have a well-treated room. Apply a high-pass filter in the DAW, not as effective as treating the problem at the source. Using a bipolar ribbon mic that doesn't have a proximity effect, which can cost some serious $$. Using a multi-pattern microphone such as a Shure KSM-44 or AKG C414. Again, serious $$. The most cost-effective solution is to create a dead space in which to record vocals, surrounding this "vocal booth" with rigid fiberglass panels, and maintain a distance of at least 10" when singing. This solution works best if you have a quiet room and a quality microphone. If you're trying to mitigate the problem in an already-recorded track, a dynamic EQ can do the job. Wait for a sale on Meldaproduction's MDynamicEQ, which was something like 50 bucks last time it went on sale.

So last night I plugged in the guitar amp to see what the problem was. I know the guitarist is extremely strapped for cash and wouldn't be able to take it to a shop. Frustratingly, it worked just fine. I've repaired a lot of amplifiers in my day, and Fenders are among the easiest to work on. But they've always had obvious, consistent problems (smoke being a reliable diagnostic indicator). Never had one that was dead one day and OK the next. I've reseated all the tubes, about all I can do for now. This is a model I've not seen before, much less opened up, called a Bassbreaker 15. In case anyone has any thoughts on what the problem might be.

Sounds like a great setup. Unfortunately, we're already a six-piece band and can't afford to bring in a second guitarist for operational redundancy. Then again, we have no fewer than three tambourine players. Just in case one of them goes out of tune.

Now, Ed, remember this is a family-friendly forum. There could be young children reading this who are also Joe Cocker fans. Or maybe not.

I've not tested this, but I think what happens is the aud.ini setting gets updated whenever you change the pan law, so that it always reflects your most-recent choice.

Could be related to the bug in the C++ redistributable installer that Noel talked about here. It was deleting a critical DLL.

There is a variable in aud.ini named PanLaw. This is what determines the default pan law in the absence of a project template.