bitflipper

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.

Bottom line first: pick a pan law and stick with it. It almost doesn't matter which one you pick, because you'll quickly become accustomed to it and hopefully never think about it again. Stick with just the one pick, though, because the pan law is global, and changing it will mess with any previous projects you revisit. And of course, never change it mid-project unless you want to restart the mix from square one. As to which one is "best", most intuitive or most practical, that's been debated since the feature was first introduced in the 60's. Some prefer -3dB, some -6dB, with reasonable technical arguments for both. SSL introduced -4.5dB, not because it's better than those but merely as a compromise between them (conventional hardware consoles' pan laws are not selectable like in a DAW). The fact that it's not offered in CbB shouldn't trouble anyone in the slightest. Pan laws not only differ in whether they use 3, 4.5 or 6 decibels, but whether they achieve that compensation by lowering the center or by raising the sides. IOW, you can keep the volume steady by adding 3dB at the extreme left and right positions, or by lowering the center by 3dB. The latter helps avoid the effect described in the OP, wherein a track unexpectedly clips just from being panned. Unfortunately, Cakewalk's default option, "0dB, sin/cos taper, constant power" suffers from this potential problem because it raises the sides rather than lowering the center.

The problem, Eric, is that those of us who've used SONAR 6 are all old farts who have trouble remembering what we were doing last month, let alone in 2005. Trust me, the new DAW will be a huge step up from S6 (or even 8.5, which I agree with Mark was SONAR's pinnacle). Since you're getting back into it after a long absence, you're going to have to face a learning curve anyway. Might as well put that unavoidable effort into CbB. We'll be here for ya.

^^^Your point is well-taken: when there is a discrepancy, trust the DAW over the playback device/software. Rather than asking "what's wrong with my DAW?" instead ask "what's wrong with my player?". But to their credit neither the original poster nor the user who revived the thread reflexively blamed the DAW. In fact, Thierry's (correct) instinct was to verify the file's integrity. He just went about it wrong, making a reference mix within the project rather than exporting it and then importing it back into the DAW, as suggested by Noel. Craig's suggestion is probably the most likely explanation. Note to other folks experiencing this problem - if you think this is bad, wait until you play your masterpiece back in your car. Or on earbuds, or on your friend's hi-fi, or over a PA system. It'll sound different every stinkin' time.

In terms of audio quality, there is no practical difference between DX and VST. I am perfectly happy with DX plugins. And no, DX is not going away. At least, not as long as the XBox lives on.

USB ports can become unresponsive after a sleep, or just due to your power scheme. You can go into Device Manager and exclude them from whatever power-saving scheme you've specified. In Device Manager, locate your USB port (it'll be called "USB Root Hub", and there will likely be more than one so do this for all of them). Right-click on each USB device, select Properties and go to the Power Management tab. If the box labeled "allow the computer to turn off this device to save power" is checked, un-check it.

^^^ This is the answer. Assuming you're not going to send your mix to a third party for mastering, the last active component in your master bus fx bin should always be a limiter, followed only by metering plugins. You'll probably want a LUFS meter at the very end, which will do a pretty good job of telling you whether the master is going to be too quiet, too hot, or somewhere comfortably within the Goldilocks Zone. You'll probably want to import some music from your favorite commercial recordings, anything that sounds particularly good in the car, and use that as your LUFS target. Setting levels for your car is a tricky business, as the car's player probably has built-in compression and EQ that the owner's manual doesn't mention. Plus the acoustics inside a car are pretty awful. So don't be surprised if your carefully mastered songs sound good only in the car, and nowhere else.

Check to make sure it installed correctly and didn't fail the scan. Go to Preferences -> VST Settings -> Scan Options. Check the box labeled "Generate Scan Log", and then click the button labeled "Reset". Run the scan by clicking "Scan". This alone may do the trick, but if Synthmaster still isn't showing up, check the log. It'll be in %appdata%\Cakewalk\Logs. Open it in Notepad and search for SynthMaster.