bitflipper

SONAR will only recognize the device if Windows recognizes it. First step, then, is to verify that Windows knows it's there. Go to the Sound control panel and make sure the Behringer shows up in the input device dropdown list. If it is, then you know that Windows is aware of it and is able to talk to it, eliminating potential hardware issues such as a bad USB port or cable. Next, verify that SONAR has been told to use that device. Go to Preferences within SONAR and make sure the interface has been selected. If Windows is happy and SONAR is happy, switch to a native driver and see if that works. MME is the most reliable of the native drivers (but not recommended for recording) and tends to work when others do not. If MME works, try WASAPI. If that works, it'll actually do better performance-wise than Asio4All. Behringer does have a proper ASIO driver if you have your heart set on ASIO, and it'll be more reliable and more efficient than ASIO4All.

A journey down memory lane...yes, Charles did build an XP-compatible version, I tested it for him and became a fan in the process. I still like Imperial Delay, though, as well as Timeless 2 and even the good ol' Sonitus. But it'll always be the tritik delay that gets first crack at a track. Kudos to Clint for turning me onto it. Sorry to the OP for taking the thread off track. But there are parallels, I think. The main difference is that everybody needs a delay, everybody needs a conventional algorithmic reverb, but nobody needs an infinite-feedback reverb due to its limited applications.

This does the same thing as Shimmer, but takes the idea a bit further, e.g. the ability to shift pitch by a musical interval. If you already have Shimmer, this would be somewhat redundant given how seldom most people actually use such beautiful but potentially mix-destroying effects. But if you don't currently have Shimmer or similar effect in your kit, this one will do a fine job - and is currently the least-expensive in its class. Gonna pass because you already have plenty of versatile reverbs? I felt that way about delays, too, until I tried tritik's delay. Now it's my first-call delay plugin.

If the waveform is not visibly clipped but sounds distorted, then we can assume that the distortion isn't from clipping. Record a sine wave, then zoom in on the waveform display and see if it looks like a sine wave. For example, see if the peaks have an inverted section. Zoomed out, it'll look fine but sound nasty. That particular symptom indicates an overdriven ADC, which has nothing to do with recording levels in the DAW. Make sure you're going into the audio interface's LINE input and not its MIC input.

7 GB is indeed quite large, but not outside the realm of possibility. CbB's design philosophy of non-destructive editing means nothing ever gets deleted. Lots of comps means lots of unneeded bytes languishing on disk. I understand your reluctance to start deleting files. I'd feel the same way. But if you're short on drive space on your backup disk, you may have no choice but to do some purging. Fortunately, there is a safe way to do it. Use "Save As" to make a copy of the project. The new project will contain only those files that are actually referenced in the project, less the redundant files. You can then open and play the copy to reassure yourself that it really does contain everything you need. If it does, you can then safely delete the original project, or keep it on disk but back up the trimmed-down copy instead.

I used to be on insulin. It struck me as weird that the treatment for high blood sugar was more insulin, when the underlying problem was excessive insulin to begin with (due to insulin resistance). The biggest annoyance with the insulin wasn't the 4-times-a-day shots, but the hassle of having to do it before every meal, even on an airplane or in a restaurant. And making sure any hotel I stayed at had a fridge for my insulin. When they took me in, the paramedics in the ambulance said my blood sugar was something in the neighborhood of 500-600, and my blood pressure was pegging the meter. This was all happening during a heart attack, so it was a memorable day. That was 14 years ago. I am no longer diabetic and take no medication for it. The solution was weight loss and sugar avoidance (sugar, not carbs; I eat all the rice and potatoes I want). I relate this to you as a way to say the future need not be as bleak as it might seem at the moment.

Wikipedia sheds some light on the meaning of the group's name... I knew a bass player who had uncontrolled vocal cords, and now I know his condition has a name. No, it's not "tone deaf".

Great stuff. Wonder if they cover the topic of patients that never wake up. "Don't Fear the Reaper"? "Last Kiss"?

Welcome to the cult, Andrew. The learning curve will be worth it in the end, trust me. That echo you hear is the time it takes your computer to process the data, which is referred to as "latency". It's unavoidable, although you can, with a powerful-enough computer, reduce it. Maybe even reduce it enough to not be noticeable (as 2ms should be). But it'll always be there, which is why audio interfaces feature direct (sometimes boldly claimed to be "zero-latency") monitoring. Three main factors dictate overall latency. The latency introduced by larger buffers may not be the most significant of them, and the latency reported by your interface driver will never be accurate because there are factors it can't know about. First, there is some latency built in to the interface that you can't do anything about. High-end interfaces can have very low internal latencies (~0.5 ms or less) but those are very expensive. A mid-range interface like your Focusrite employs some engineering tradeoffs that sacrifice latency for cost, so its internal latency will be 1 or 2 milliseconds. But way cheaper than a high-end box. Solution: spend the money on a better interface. Didn't promise it would be a good solution. Second, there is the latency incurred by the computer system itself, transferring data from the interface into memory, juggling multiple buffers within memory, performing any needed conversions to the data, then experiencing the same delays when spitting the data back out (when echo is enabled). Assuming a faster CPU isn't in your budget, the best solution is to do one or both of the following: decrease buffer sizes or increase the sample rate. However, both will be limited by your hardware (e.g. disk drives' throughput may not be able to keep up with very high sample rates, or they'll fill up too fast). That's most likely why you hear nothing when your buffers get down into the 2ms range (try shooting for 5ms, which ought to be adequate unless you're Joe Satriani). Third, there is the overhead of additional processing within the DAW. This includes both the data housekeeping that is the DAW's primary duty, as well as FX plugins. The latter is the biggest variable, because depending on the type of effect the time it needs can be huge. Solution: don't use plugins while tracking, only adding them in during the mixing process when their latency is no longer a problem. Of course, you can avoid all (ok, most) of these complications by using your interface's direct-monitoring feature. The downside is that you'll not be able to use the computer as a stompbox, and no reverb on your vocal while you're recording. Some (again, more-expensive) interfaces feature internal effects that solve this issue. For the rest of us, we can do one of two things: use outboard effects while recording, or (better) learn to record dry and add effects later. When I say "learn", I mean that literally - it's an acquired skill that takes practice. We're not used to hearing guitars and voices completely dry and it's uncomfortable at first. The payoff is that you'll eventually accumulate a much wider array of virtual effects than you could ever afford in hardware form. Even better, you won't need to commit to an effect until you've heard it in the context of the finished song, leaving you with far more creative options.

The hat did look a lot better on her than on him. Last time I moved, I was leaving a house I'd lived in for 20 years. Strong motivator for cleaning, having to load all that stuff into a van, unload it in the new place and worst of all, figuring out where all that useless stuff would go in the new place. So I was brutal about it, just like the lady in the video. Wouldn't you know it, after the move my wife was frequently asking "where's the ...". I would have to slap my forehead and explain that it got thrown out. Whenever you toss something, it's a rule of the universe that you will later need that thing. That's why my garage contains a box of broken mic cables (might fix them someday), a box of 10" tape reels (with splice block and tape eraser), and a non-functioning audio interface (might need the knobs someday). Yeh, it won't be of any use when I get hit by a truck, but then it won't be my problem anymore.

First time I ever pasted a forum post verbatim into an email to send off to a friend. Good job, Steve.

Personally, I would never use any software tool that isn't crystal-clear as to exactly what it's doing and why, and (most important) logs what changes it has made and offers a convenient way to undo them. This has been my policy since the first time I used a so-called "registry cleaner" 30 years ago that hosed my system. Newer utilities have become more sophisticated, but I still won't relinquish my authority to make such decisions myself, based on research and understanding. There is no substitute for doing your homework.

I have never experienced hanging or crashing from mixing mono and stereo effects. I actually can't imagine any reason for that happening. If you can create a simple project that crashes (e.g. one track and a couple common plugins), send it to Noel so he can figure out why it's happening. Exactly how any specific plugin is going to react to an interleave mismatch is going to vary widely. In some cases, the developer has foreseen that situation and coded the plugin to transparently adapt. But trying to make software foolproof is hard, time-consuming and expensive, and not all developers take that much care. Sometimes, the plugin responds in unexpected ways. Rather than figure out which combinations don't work, it makes more sense as a matter of "best-practices" to simply avoid the situation. A compressor may not be the best class of effect to test because the problems can be quite subtle. The sidechain input to a compressor may be summed left-to-right internally so that each independent compressor in a stereo effect sees the same key. Alternatively, the left and right sidechain signals may be kept separate so that each compressor remains independent of one another. Many compressors give you a slider that determines how tightly coupled the sidechain sides are. When you insert a stereo compressor into a mono track, you are stuck with method #1 always. To answer why this could be important, ask yourself why there is more than one way to do it in the first place. Research that and you will understand a) how things can go wrong, and b) that the difference can be subtle or even unnoticeable. Also ask yourself why stereo compressors exist in the first place. If the signal is too hot, don't you want to turn the whole thing down? Why waste the CPU cycles turning down each side independently? Well, sometimes you do. Sometimes you don't. Understand how compressors can effect stereo imaging and you will gain insight as to why we have mono and stereo versions and why we have mono and stereo sidechains.

And here I thought I was an optimist. You're like the kid in a barn full of sh*t who's happy because he's sure with all that sh*t there's gotta be a pony in there somewhere.

Depends on what time I get up. My bedroom radio is permanently tuned to a jazz and blues station. If I'm up before 6:00 AM, that's what I wake up to. It makes me feel positive about the coming day, and musically inspired. But because it's an NPR affiliate, later in the morning they switch to news and features. I don't like waking up to the news, as it sets a dark mood for the day. Fortunately, I'm an early riser.

Most of the time, there is no penalty for setting a mono track's interleave to stereo. The exception is when you have a mono effect that must see a mono input in order to work correctly. Companies that provide both mono and stereo versions of their plugins do so for a reason, and you can get unexpected results by using the wrong version. But as long as you consistently use stereo plugins, your strategy should be OK. That said, I much prefer to keep mono tracks mono throughout, and avoid stereo tracks unless there is a compelling reason to use them. It's counter-intuitive, but too many stereo tracks will actually make your mix sound less wide.

I'd forgotten about the old picture cache issue. You naturally assume that a disk-write error would involve the saving of audio data, but that's not the only writing to disk that occurs. You're also saving pictures of the waveforms for your audio tracks. Over time, these can end up consuming a lot of disk space. That's why CW lets you specify where they're stored (the "picture cache") and to set a maximum amount of disk space for them. Corrupt files in the cache or insufficient disk space in the cache location can result in an error. That's why you'll often get the advice to delete everything in the cache folder as a possible resolution. You've already done that, so that's not your problem, but I just wanted to clear that up.

While some applications, e.g. disk diagnostics, are able to tell you exactly why a disk write failed, a speed-optimized program such as a DAW isn't going to perform a detailed analysis. It's only going to report the failure and assume the most common explanation, insufficient space. What's actually going on is that you may have a failing disk drive, one with unmapped dead spots, a flaky cable or even some bad RAM. Try copying the entire project and then renaming the original and target so that the files are physically moved to another location on the drive. If these messages persist, start shopping for a new drive.

I don't have a ***** account. Can you elaborate? Are these contacts via this forum?

Short answer to your not-silly question: yes, you can record any automatable parameter in real time. It's not an uncommon practice. It's called "automation recording". You'll find it much more ergonomic to use a hardware control surface, but that's optional. You can do it with just a mouse, it's just a little awkward. Here's one reference to get you started.

Jim's right - the version on baselines.com sounds much better. Like comparing flac to 128 kb/s MP3.

Power Pan Pro is probably the most sophisticated panner around. A bit pricey, though. Then again, if you're among the elite who can afford VSL in the first place, then I'm gonna guess you're probably not too concerned about that. No jealousy here. Much.

Yes, but pan laws only apply to mono panning. Unless, of course, you're using something like Boz's panner, which implements pan laws internally.

bvideo raises a good point, especially in the context of classical orchestration: there's more going on in there than just panning. Sometimes, you have to also take into account reverb. That might be natural room reverberation baked into the samples, or added via a reverb effect plugin. In the case of the former, that reverb is also going to shift along with the instrument, possibly sounding unnatural. In the latter case, you may want to use a "true stereo" reverb plugin and pan the sends. Or not. Sorry, I know any discussion of "true stereo reverb" could derail this thread even further. But it's been awhile since the topic was batted around here, so what the heck. After all, reverb (and delays in general) are very much a part of the whole stereophonic conversation.

Damn, this is so good, Jerry! Jeez, I wish I knew how to make brass that expressive.