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It is generally accepted that any project left open for too long risks becoming stale, as it is constantly exposed to the drying effect of the air in your studio. You can slow this natural tendency by covering the monitor with a large freezer-style ZipLoc, being careful to tape the opening where the cords prevent the pinch seal from joining. That will at least keep the music from excessive drying, but the baggie is only a temporary solution. Inevitably the warm air trapped in the project will foster the development of mold. If this has not gone too far, it can sometimes be salvaged by cutting off the edges with careful application of equalization. There are also a variety of plugins designed for this purpose if you are willing to trust your work to algorithmic remediation. By far the safest method of handling half-finished projects is to put the entire computer in a refrigerator (some recommend the crisper drawer) as you would with any other perishable. Or you can just hibernate your computer.

Wear points on a PC are typically fans and mechanical hard drives. I expect most of the refurbing that actually gets done is replacement of the broken part that caused the previous owner to dump it. If it is part of a fleet of machines being rotated out to make room for new stuff, I doubt that they do much more than test it superficially to see if it seems to work and depend on the buyer to return it for a replacement if they miss something. Typically you will get a short warranty, but usually long enough to uncover a problem that was already present when you bought it. If the price is attractive and you have a cheap return option for DOA it is not a bad risk--most of the parts can be replaced with new or refurbs pretty inexpensively. It may be cheaper to buy from a commercial source than used from Craig's list or Ebay--previous owners remember how much they paid for the machine and almost always want too much.

I do use True Image for disk imaging (booting from a rescue CD), although I agree it is doing waaay too much most of the time if you just install it and let it run in the background. I use Synctoy to do file backups. It is reasonably fast, and once you understand the basics of it, you will be able to understand and control what it is doing. Copying every file manually is certainly something you can do, but if you have hundreds of files can be confusing and duplicative. Too many of these applications (and I include Trueimage here) are too much faith based--trust me I am automated and nothing can possibly go wrong.

Well, I assume that you at least are not getting breakage fees deducted.

Uninstalling just takes a few minutes, and does not leave scars on your computer. More to the point, the time it takes to figure out how to use new software is the major cost of anything new. If you spend a couple of dozen hours figuring out that the software is not really going to be useful, that is time you will never be able to buy back at any price.

I sprung for Waveform 8 Basic for $26.70 a couple of years ago. I really have had almost no use for it--so free might be a reasonable price for an upgrade--but not if this is an even more basic version than Basic. In what ways is this "free" version crippled? In fairness if you are a loop style composer this might be something for you. PS was anyone actually paying $386.00 for Waveform 10 Basic?

The first thing I would try when one program installation seems to break another program is to reinstall the broken program. If there is a "repair" option under "Apps & Features" under the application try that first--but that is uncommon. If a re-install fails check to see if you have everything you need to do a complete new installation and then un-install the broken program and try again. Pay attention to the locations where your installation is writing files. Often enough the problem is that a program overwrites an existing file with one that is incompatible with the older program.

Back in the early days of the Gibson>Bandlab conversion there was some discussion of what would become of the other Cakewalk (the company) products. I stumbled on this recently: https://www.musiciansfriend.com/pro-audio/xchange-producer-collection-with-presonus-cakewalk-ik-multimedia-image-line-loop-loft-ohm-force-and-sonnox Apparently CA-2A T-Type Leveling Amplifier has turned up in a discounted bundle. Maybe we will see the rest on one of those free free promotional DVD's from the music mags eventually.

This $4.95 deal is back--or maybe it never went away and I am just getting the email. It sounds like a no-brainer price for a big pile of weird sounds, but some of the videos they show keep flashing a red banner that says "this effect requires GPU acceleration," which gives me some pause. Also I cannot find a manual on their site. Has anone actually used this thing? ps the version on sale is 1.2 so maybe the original sale was for 1.1 and this is an intro price for the new version

I am afraid you missed my point. I am not saying that singers who employ a wide range of volumes in their performance are ignoramuses (ignorami?) or somehow doing something illegal--quite the contrary. My own personal taste in music is to have something when you are done that you cannot listen to in a moving car because the softest stuff falls below the wind and traffic noise threshold without the loudest stuff damaging your hearing--the kind of product turned out by every orchestra or serious non-pop singer in the history of music since the stone age. What I am saying is ignorant is to expect a real dynamic range to be reflected in a final recording that smashes the variance into a band of a few dB. If you want that kind of flattened volume range, then as a singer you should learn to produce it rather than depend on some feat of technology to achieve it from the original performance, and if you want to savor the beauty and expressiveness available in a wide dynamic range, then you need to use technology to capture it without exceeding the capacity of the technology. Look music-on-the-go and dodging-beer-bottle-venues have been major drivers of the loudness wars, and we are used to hearing stuff that is so compressed that it no longer qualifies as high fidelity. In fact radio stations for years have routinely compressed already squashed recordings and many streaming services and MP3 players routinely do the same. The days when people valued going to a sit down venue where alcohol was available only during intermissions and listening to a decent stereo in the dark filling the silence of their homes are probably long gone. Singers of pop music have in fact learned technique that makes that narrow-range product easy for engineers to produce. If the singer wants to make that flat-volume product, and still manages to clip or overload the equipment without falling below the noise floor he qualifies as out of control. Given the capability of decent equipment and the dynamic range of digital representation he also has a voice that is capable of more volume than a normal human. If he values the exploration of the range of loudness available to the human voice I applaud him. In any case, if you want to have the option to squash your performance in the box while maintaining the ability to let it stretch, then capturing the performance by recording less hot is the better way. You should realize that clipping in the digital realm is dependent on the total power at all frequencies (including those too low and high to hear), and that mixing all your tracks together is basically the process of addition. Perhaps the most effective way to avoid clipping in the final mix while getting something that is reasonably "loud" for today's listeners is to use is to use equalization in the box to filter out power at frequencies that are less important or are already occupied by other instruments so that they are not distinctly audible in the mixed track. That process subtracts dB from the final mix without requiring that loud become soft at least subjectively to the listener. You can then raise the volume of the entire mix without exceeding the clipping point. But early compression removes all the frequencies more or less indiscriminately and limits your options for that important technique in mixing.

You ha Marled, bingo! That is what I am battling with So you need to ask yourself why is a singer using a wide range of dynamics. If he is just an out of control ignoramus who expects an interventionist engineer to cobble together a "normalized" gain envelope on the digital track, then compression may save you some work. But what if he intends to go from a whisper to a scream in the same song as a matter of style or musical expression. Compression anywhere is going to thwart that intent by narrowing the dynamic range of the recording more than the performance that is its source. To put it in perspective, the digital representation of audio on a 16 bit CD is capable of representing a dynamic range of 96 dB--that is roughly the range in dBspl from a whisper at six feet in a quiet room to the threshold of pain. The usual loudness range for singers is about 30-80 dB with a smash the glass opera singer maybe getting to 100. Presumably your singers are not singing so loudly without amplification that they are hurting themselves or those near them as they perform, so there should be more than enough dynamic range in the digital realm. You might have a problem if your microphones etc. are not sensitive enough to deal with the quietest parts when the loudest parts are quiet enough to avoid clipping. If the singer (or his engineer) expects the volume in his headphone feed to be normalized regardless of the loudness at his mouth hole, then it might be better to put a compressor on an analog circuit from mic to headphone and record the full dynamic range, at a lower volume for later tweaking.

Just to be clear, the fact that you cannot use a software limiter on your input to avoid clipping is not a Cakewalk limitation--it derives from the mathematics of the process, and applies to all software. Software effects work only on digital data--stuff that is not audio at all but numbers that represent it in the computer. So if you send a signal from your microphone/preamp to the audio to digital (D/A) converter in your audio interface that produces a number bigger than the maximum bit depth your software uses to represent sound that is going to cause clipping (lost data) that is irretrievable in the box. Putting a limiter on the digital data stream after the clipping has occurred at the A/D will not bring the lost data back. A hardware limiter is what is needed if you want to avoid clipping on input, since it will act on the analog/electrical signal prior to it reaching the A/D. Technically you will also lose data using any kind of limiter at any point. The power/volume of the original signal is reduced and that is a characteristic of the original sound that the digital representation will not correctly encode--but reducing the analog power low enough to avoid digital overs at the A/D will prevent the nasty artifacts that digital overs produce. As others have noted the best practice is to record at lower input gain (record less "hot") and use a bit depth representation that gives you some extra zero bits at the top. You can then bring up the power of the digital data representation with software overall, or limit the variation between the softest and loudest parts of the recording using a software compressor/limiter.

76 USD is a really low price point for any contemporary audio interface with MIDI connections. Your best bet is to find a used interface at that price--even that is likely to be a stretch. The other option is to buy and pair a couple of the really cheapo audio and MIDI interfaces as separate units. If you have a firewire interface that you are not going to use, it is likely better than anything you can buy for your budget, and your best plan might be to find someone who will buy that or take it in trade to get you up to a reasonable budget. I would definitely not advise you to buy something that can be had for that cheap--try to see if you can get a job raking leaves for a few days and get into the $250 US or better range which will give you a bunch of quite usable USB connected units.

It sounds as if you are really having a problem composing, harmonizing or orchestrating rather than a MIDI problem. MIDI is just another way to record or notate music and if you already know how to edit MIDI, then you have everything you need to create a symphony from scratch if that is your goal. If you know the notes you want a MIDI controlled sound source to play, the simplest method for getting there is to use some kind of MIDI controller. If you lack the instrumental chops to play a controller in real time, then step record, enter notes in a piano roll editor, play on your instrument of choice or try to use pitch to MIDI conversion algorithms etc. You can even enter notes in an event list editor if you can convert the notes in your mind to note numbers, velocity values and duration. If you do not know the notes you want to play, then MIDI is not an answer to anything. Band in a Box and other algorithmic composition/harmonization tools may use MIDI inputs and outputs but they are not MIDI--they are computer assisted composition tools. I am not saying you should not use them, but by their nature they are designed to give you a solution to a problem or at best a choice of solutions that may or may not be what you want.

I remain confused about Seaweed's upgrade policy. It looks to me like you need to buy each new upgrade and that it will install only if you have an earlier (the most recent?) version installed. Can you explain how this works?

The statement is as opaque a collection of inane business jargon as I have read in a long time. Perhaps they are heading toward online subscriptions but if that is what "the development of a new, unified and fully integrated platform on which the company's entire portfolio of products and services will be available next year," means the Delphic oracle could have said it with more clarity. Not that they do not plan to do that--everyone else apparently does. Could just as well mean they are developing a DAW or have invented a direct link to the temporal lobe bypassing the ears entirely.

So it looks like you are replacing a PCI card with 4 channel duplex (using SPDIF and balanced pair of I/O) with 24/96 and MIDI in/out. If you need all that for under 500.00 you can probably find several USB connected solutions. You can come pretty close with a Focusrite Scarlett 6i6 for about half of your budget.

Excellent. Thank you, Simeon.

Is there a manual or how-to somewhere for this?

. . . and the guitarist can no doubt use his footwork skills as a member of the Riverdance troupe if there is ever a revival.

I expect there is some combination of gear snobbery and tax advantage involved here. Why anyone composing in the box would need 32 channels of conversion is something of a mystery.

I would think how you view this issue depends on how you are using your audio interface. For recording, the noise that comes into and is generated by the A/D converter is going to be additive ie 24 tracks of input will each have the noise of a single track in the digital domain, and 24 tracks will, when "mixed" digitally (added) will have 24 times as much noise as a single track alone. They will also have a higher "signal" in the digital domain, which can mitigate the audible noise relative to the signal so that it is not as noticeable as you might figure from simple addition of the noise. If you are going to be mixing dozens of recorded sound tracks, then minimizing noise prior to arrival at and generated within the A/D is likely to improve the final sound significantly. I suspect that even Realtek can still produce quality as good as the multiple magnetic tape bounces some of us grew up listening to on AM radios. But many of us use an audio interface only to listen to originally digital "audio" generated by softsynths etc. On output I would have thought that the "audio" of however many tracks you are dealing with remained as a digital internal representation until the mixed signal/data is sent to the D/A of the audio interface/card/chipset. In that case, any noise you encounter is generated from/applied to that (usually stereo) mixed digital output at the D/A conversion point and beyond. So that noise is going to be pretty much the same whether you are dealing with a single track or 24 tracks in the digital realm. In any event, that digital sound/data remains pristine regardless of how you hear the output from your sound card on playback/conversion, and it will sound less noisy on less noisy equipment if you play it there. So for that purpose, the noise floor of your audio interface is not all that critical.

Thans, Vernon. That is in fact mentioned on the page but I missed it.

i must say I am confused about what these things are. Are they full standalone instruments, VST's, soundsets, samples, presets for something else?

If you do not mind pedestrian grey color they can be had at Musician's Friend for $137.40. https://www.musiciansfriend.com/pro-audio/beyerdynamic-dt-990-pro-open-studio-headphones-250-ohms?rNtt=beyerdynamic studio headphones&index=4