bitflipper

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.

I used to run my keyboards stereo, first through a pair of Roland keyboard amps, later through a pair of QSC self-powered PA speakers (which sound great). But I eventually gave up trying to go stereo because the audience refused to all group in the sweet spot in front of the stage, equidistant from my speakers at 45 degrees. Leslie effects aren't quite as good, but mono improves overall clarity and tone. Plus the guitarist and bassist don't have to listen to just my left hand all night when I play piano. I still use two speakers, high up on stands on either side of the drums. For better or worse, everybody can always hear me clearly.

Oh, you fancy folks with your confusing knobs and stuff. All you need is Sausage Fattener. [EDIT] (for clarity)

That might just work better than a line out. And I've actually got both. But I don't carry them in my gig bag and we were playing on an island hours away from the nearest music store. I thought I was adequately prepared because I'd brought a hat.

And most of those are bapu's. Just sayin'.

YouTube is especially problematic, with levels all over the place. Although they do implement volume normalization, it doesn't work. Sorry, but I don't know of a solution for this, but I'd recommend using YouTube's volume slider rather than your audio interface's volume control. At least then you can keep your amp's volume consistent, which will improve your mixes. If you've never noticed it before, in the context menu on YouTube there is a selection labeled "Stats for Nerds". Check it out, it's illuminating. Along with video info such as dropped frames, it also shows the original audio level and how much reduction was applied by YouTube. I've observed some as low as -24 dB and as high as +8(!)dB. I guess it would be far worse if YT didn't do any automatic normalization at all. Here are examples of how the content level affects both the volume you hear and the overall quality. One comes in at +7.8dB, the second at -6.8dB. Same concert, different levels. One is noticeably distorted. See if you can figure out which is which without peeking (it's not hard).

Could have been a disaster. The guitar amp was DOA. He had a small second amp that's normally used as a satellite on my side of the stage so I can hear the guitar better. We had no choice but to mic that little amp and run it through the PA. With no extra mic stand, we had to dangle the mic down the front, so we're capturing it off-axis. An SM-58 is definitely not a side-address microphone, so as you can imagine it sounded pretty thin and nasty. Fortunately, we had a great crowd and the performance went over well. Everybody was firing on all cylinders, even me, being fueled by Starbucks iced mochas. With my high blood pressure I'm not supposed to do caffeine, and normally avoid it. Put a couple iced mochas in me and I'm like that squirrel in Open Season. One of the reasons the performance worked well was that for the first time I was getting guitar through my vocal monitor. Being able to hear him clearly meant that we were in better sync than ever and I was able to play off him in a complementary fashion. Sometimes adversity spawns epiphanies. So today I'm trying to figure out how I can monitor the guitar like that in future. I'm thinking a line out to the board and just routing it to the monitors but not the mains.

Sounds like what the OP is experiencing is the muting you get when you use the free player with a licensed library that requires the full version of Kontakt. GodinLG, which library are you using? We might be able to suggest an alternative that is compatible with the Player.

As a general rule, physical potentiometers should ideally be in about the 2:00-3:00 position, as noted by Alan above. Many mixers actually highlight this region with a bar printed on the silkscreened label around the pot (exception: trim controls, where the printed bar indicates the boundary between gain and attenuation). The input trim control is a pot at the very beginning of the signal chain, usually before any active components. This allows you set up each of the mixer's inputs so that they'll all be in the same general level, and the main faders can all start at "0dB" (which, if they were rotary pots instead of linear pots, would correspond to the 2:00-3:00 position). Anders, look into the topic of speaker calibration and the K-System. In a nutshell, this is the process of setting up your monitor for consistent levels at a given volume knob setting. It's based on a standard that was originally devised for motion picture exhibitors, meant to assure that watching the same movie in various venues will be played at a consistent volume. An excellent primer on the topic is Bob Katz's well-respected book Mastering Audio, a must-read for anyone trying to make sense of all this stuff.

What do you see as the input source for the MIDI track? Does it give the name of the USB driver there, or is it set to None? Is the driver listed in the dropdown list of input sources? How about the list of devices in Preferences?

I don't see how disabling a meter could affect clipping. Meters just show what's going on and don't affect signal levels. The only difference between gain and volume is where the adjustment happens in the signal chain: gain's at the front, volume's at the end. Metering happens at the end of the chain, so yes, it'll reflect the actual peak levels regardless of at what stage they were raised. And yes, raising either gain or volume can cause clipping. If it does, you'll definitely want to know about it, not hide it by disabling track volume metering. I have a feeling the original question was just not worded well, and Marcello's friend just didn't explain clearly what he's doing in Reaper or why.

That would be the logical conclusion. You'd have to load your default template, change the pan law and re-save it as a template. After that, new projects should inherit the new pan law.

You are correct. I mis-spoke when I called it "global". It isn't. A better word would be "persistent". Once you set it, that does become the default for subsequent projects. This is how you can inadvertently end up using a different pan law than you thought was in play.

It's a free synth; ergo, 99% on-topic. No need to move it unless you have a better place in mind. If you do, drop me a PM.

Anybody else see that pathname and think "what the heck is that?"? Maybe it's a sample file but doesn't have a .wav extension for purposes of obfuscation. Wouldn't surprise me, it's Waves. Audio files are not going to contain any reliable malware signatures, but without an audio-related extension the antivirus software wouldn't know that's what it was.

I'd suggest a more generalized solution: implement the ability to pull up any designated plugin's UI via a keypress. That would address your specific need, and also allow other handy shortcuts, such as quickly bringing up your mastering limiter, spectrum analyzer or a main multi-timbral synth. In the meantime, a possible solution would be to use screensets.

Question for the sax players: does this apply to altos specifically, or to all saxophones? I've long marveled at the way my band's sax player can quickly transpose in his head. I'm too lazy to even try. Fortunately, I play keyboards and there's a button for that. But if he's using different rules when switching between, say baritone and tenor (which he sometimes does mid-song) that would be an even more impressive skill.

Good to see you back, dubdisciple.

In order to accomplish this, you will need an audio interface with multiple outputs. Unfortunately, that also means investing in a higher-end (read: more expensive) interface. There are other reasons for acquiring such a device, though, such as the ability to have multiple headphone mixes and multiple speaker setups. So even if this scheme doesn't work out you won't be sorry you bought a full-featured audio interface. For simplicity, though, I'd echo Noel's suggestion and just run those backing tracks through a full-range system like a PA, or even headphones. Save the complexity for your music. That said, in a live performance situation I think separate amplification could be very cool. I once heard a solo guitar performance wherein each string on the guitar had its own output and its own amplifier, and the effect was awesome.