JoyWalk Music

The Compressor's Secrets

2009-05-02T21:48:00.000-07:00

JoyWalks knowledge, understanding and experience brings patients to your project, as well as a truly advanced use of professional digital equipment. Our commitment is to your music and individuality. It's about creating a comfortable working environment, developing your musical knowledge, and our ability to tailor a high level of service to any budget or timescale.

We are dedicated to provide you with the service you deserve. While Creating Sounds Unreal! Our popular recording service is grooving for the Memphis TN and surrounding areas. A unlimited track mobile recording service, easily capable of handling the majority of sound recording situations, from Schools, Community Groups, Poetry, Gospel, Pop, Rock, HipHop, Country to Jazz, Classical and R&B artist. A professional sounding recordings either live, in your home, in rehearsal spaces, studios, or at a location of your choice.

As an up-to-date business, we want to give you the opportunity to stay in touch with our company and our offers. We would be pleased to hear from you! Please let us know what your needs and questions are, we will be more than happy to help.You can reach us at: joyscj@gmail.com. We are looking forward to hearing from you.

The Compressor's Secrets

Less well known aspects of compression.

Every studio has one, every engineer uses one, and every popular music recording - almost - dating back to the 1950s and beyond has benefited from one. Of all the many and varied types of outboard in the processing and effects racks, the compressor is surely the one that is most often used, and one that repays its cost of ownership countless times over during its working life. So I don't need to tell you anything about compressors then? Maybe not - if there does happen to be anything that you don't know already then you can easily find it in textbooks and magazine articles that are often aimed more at the beginner than the seasoned pro. However, the compressor is a many-faceted instrument, and there are a number of tips, tricks and techniques that are not commonly covered in print. Are these the compressor's secrets, known to the few and hidden from the many? Like the Masked Magician, I intend to reveal these secrets to the world.

Merciful Release

A long time ago, when I was a fresh-faced student of sound engineering, I went to a trade show (in the days when you had to beg your way in, if you weren't in the business already) and alighted on the stand of a company who had a new and wonderful compressor to show off. 'Listen to this,' said the silver-tongued salesman. I listened as he demonstrated his amazing box. That's 30dB of compression. Does it sound compressed to you?' I looked at the gain reduction meter, I listened, I looked at the gain reduction meter, I listened. Sure enough, the meter was showing a full 30dB of gain reduction and the music I was listening to sounded as fresh as a live performance. I knew something about compressors, and I knew that 30dB of gain reduction ought to be the sonic equivalent of what an apple looks like after it has been through a cider press. It's a good thing I didn't have any money or I might have bought it on the spot. With the benefit of experience I know what happened.

I am sure that it was a reasonably good compressor, but not significantly better than any other. What the salesman had done was to turn the release control to maximum. Release, as you know, is the time it takes for gain reduction to return to zero after the signal has passed below the compression threshold. In this case, the signal never passed below the threshold long enough for the level to begin to return to normal, to any significant extent. The result was indeed 30dB of gain reduction, but not 30dB of compression. You don't need a compressor to get any amount of gain reduction - just lower the fader. Compression implies a constantly changing amount of gain reduction, and the gain reduction meter must be visibly dancing up and down. If it is not, you're wasting your time. How fast it dances up and down is up to you but, if you want value-for-money compression, a short release time will give you a more audible compression effect. A longer release will lessen the audibility of the compression, but you won't actually get as much real compression.

Over Compression

No one reads the manual for a compressor, and if you did you wouldn’t get any warning about the effects of over compression. I don't mean this in the sense of too much compression, your ears will tell you that, but in the sense of setting a lower threshold than you need to get the job done. This will always make the sound worse, with the sole exception of percussive sounds where it might sometimes be a useful effect. Let's assume a scenario where an instrument plays occasionally with silences in between. This is where over compression is most likely to happen. When setting the threshold, many users have an idea of how much gain reduction they want to hear and see on the meter. The amount of gain reduction is controlled by both the threshold and ratio controls. Suppose these controls are set so that the desired amount of gain reduction, let's say 12dB for example, is achieved. This should be fine shouldn't it? Look again at the gain reduction meter.

While the instrument is playing, does it ever go all the way down to zero? If it doesn't, if it only goes down to 3dB, then you haven't applied 12dB of gain reduction, you only have 9dB of compressive gain reduction. The other 3dB could have been achieved by simply lowering the fader. This, in itself, isn't necessarily a problem. The problem is that, when the instrument starts to play, the compressor has to go all the way from zero gain reduction to the full 12dB. The necessity of covering that additional 3dB will audibly distort the initial transient. Try it, and you will hear it for sure. This leads to rule number one of gain reduction - at some point in the course of the track while the instrument is playing, the gain reduction meter must indicate zero, otherwise the minimum reading obtained shows wasted gain reduction and over compression, leading to the distortion of transients that follow silences .

Compression by Stealth

One of the best-known uses of compression is to increase the apparent loudness of a mix, or an individual voice or instrument for that matter. Compression, as you know, works by reducing the high signal levels, bringing them closer to the low-level passages, and then applying make-up gain. Thus the low-level signals are brought up and the whole thing sounds louder. This is fine in theory; the trouble is that the effect of compressing the high-level signals is very audible, necessitating great care in the set-up of the compressor and judicious compromise between getting enough compression and not spoiling the overall sound. Ray Dolby told us this when, in the early A-type noise reduction system, he left high levels completely alone and modified the gain only of signals below -40dB. What we need is a compressor that only operates on low-level signals. Is there such a thing? Yes there is, and it's in your rack already. You just have to use it in a different way. Since in this situation the object is to bring up the lower levels of the track, what we need is a way of making the quiet sections louder without affecting the loud sections.

The answer is to mix the uncompressed signal with a compressed version of the same. At levels below the compressor's threshold the two signals will combine to produce a 6dB increase in level. Above the threshold the compressed signal will be progressively reduced and add hardly any additional level to the mix. The result is a form of compression where you can get more dynamic range reduction with fewer audible side-effects. I'm not going so far as to say that it is always the best way, but it's certainly worth a try. Maybe some enterprising company will bring out a gadget to do just this, in a convenient rack mounting package. By the way, if you try this with a digital compressor you will get a lesson in the delay involved in digital processing. You will get comb filtering and it will sound dreadful.

Compression vs. Clipping

While I'm on the subject of increasing apparent loudness, I don't know whether it is as widely appreciated as it should be that compression is only half the answer. Compression is a long-term type of gain reduction, working at the very least over periods of tens of milliseconds. If you try to achieve very fast acting compression by using very short attack and release times, you may well end up with distortion of low frequencies where the compressor actually changes the shape of the waveform. There comes a point in maximizing apparent loudness where the compressor has given all it has got to give. Clipping, on the other hand, works on a very short timescale. Transistorized circuitry reacts within microseconds to any level that is too great for the power supply to cope with and cuts it short, creating harsh harmonics, but at the same time extra loudness. The soft clipping of valve and valve-emulating designs rounds rather than clips the peaks but, once again, operates on a short time scale. The problem with soft clipping, if used alone, is that it only works on high-level signals. Clip-worthy peaks only occur in quantity in high-level signals and low-level signals, although they may indeed have the occasional clip able peak, are largely unaffected. The answer is to use a compressor and a soft clipper in series.

The compressor evens out the general level of the signal but, since it works over a comparatively long time scale, the peaks are not clipped but simply brought to a more uniform level. The clipper then has more material to work on. A useful alternative is to use a series-parallel configuration the compressor smoothes out the levels, the valve emulation device soft clips the peaks, and the result of that whole process is added to the uncompressed signal. The result is controllable enhancement over a wide range of levels. If you want to go further then you might add an equalizer after the compressor so that you can choose the frequency range that will be affected to add just the right hint of distortion without going over the top, particularly in the mid-range.

MS Compression

Here's an interesting curiosity. As you know, when compressing a stereo signal, a two-channel compressor must have its side chain linked; otherwise heavy compression in one channel will cause an image shift in the stereo sound stage. Both channels must, at all times, be compressed equally. Of course, this assumes that you are handling stereo as left and right channels - let's call this LR stereo. Not as popular but certainly very useful is mid-side or MS stereo, where the M channel is the mono sum of the whole sound stage and the S channel represents the difference between left and right. MS is a useful microphone technique and is sometimes used at other points in the signal chain for modifying the width of the stereo image. (It's a funny thing that proponents of MS often forget that you can do that to LR stereo signals with the pan controls.) But what about compressing a signal in MS format? Is it possible? Does it have anything new to offer? Yes, it is possible to compress MS signals without converting them to LR. Just pass the M signal through one channel of the compressor and the signal through the other. Once again, you will need to link the sidechains or funny things will happen, but it will all work perfectly.

Some might say that it works better than compressing LR stereo since, even when sidechains are linked, it is not guaranteed that analog compressors will handle both channels absolutely equally and some image shift may persist. But, if you compress in MS domain then any disparity between the channels will result not in an image shift, but in a variation in the width of the stereo image, which is arguably less obtrusive. But why not take this a stage further and do something really wacky like compressing the S signal only. What happens now? If you compress the S signal only, then anything panned center is unaffected and compression only affects signals panned left or right, or signals that are out of phase. Loud signals in these modes will cause a momentary reduction in level of the S channel resulting in a narrowing of image width. I can't say that I recognize any useful function for this myself, but in the hands of more creative people, who knows?

Serious Sidechain

Everyone knows how to direct a high-frequency boosted signal to the sidechain to perform a crude type of de-essing - now superseded by more sophisticated stand-alone de-essers such as the Drawmer MX50. But what about applying EQ to the sidechain in general, rather than this one specific application? If you have never done it, do it now. Parallel a signal so that it enters the normal input of the compressor, and at the same time is connected to the sidechain input via an equalizer. Now play some signals through this set-up. We all know that different compressors have different sounds, but this little trick allows the compressor that's in your rack right now to have an incredible range of sounds going far beyond the normal differences between models, when used in the standard configuration. You will find that the compressor becomes another type of EQ but, instead of simply cutting or boosting different frequencies, you allow different frequency bands to control the amount of compression applied. When you are in search of that elusive 'phat' sound and simple EQ and compression are not getting you there, EQ'ing the sidechain might just do it for you. In fact, I would go so far as to say that all serious compressors should have sidechain EQ built in. Once you have really tried it you won't want to do without it

The sidechain can do more. Everyone knows that different compressors sound different, and that soft-knee types are more subtle than hard-knee, which go immediately from uncompressed to compressed at the exact threshold level rather than the gentle blending of the soft-knee type. The precise knee curve of a compressor is an important factor in its sound, but few compressors allow you to modify the knee curve in any way. So can it be done? Well of course it can, otherwise I wouldn't have mentioned it. Here's the deal: set up a sidechain configuration as above, but this time, instead of an equalizer, insert a distortion processor. A guitar effects unit such as the SansAmp GT2 would be fine. Remember that you are not going to hear any signal coming out of the sidechain, unless there is some internal crosstalk within the compressor, so the output signal isn't going to be distorted. What the GT2, or similar device will do is apply soft or hard clipping, which will bend the shape of the knee curve of your compressor. What effect this has depends on the compressor itself, on such factors as whether peak or RMS detection is used for example. However, the result will be that you will feel as though you have a totally different compressor in your rack. In fact, when different settings are used on the distortion box you will feel as though you have installed a whole rack full of different compressors.

Another option for the sidechain is to insert an advanced version of the signal to control the level of the signal itself. One of the enduring problems of compressors, and gates for that matter, is that they can only react to whatever information they receive, they can never anticipate what is going to happen and prepare for it. Well now they can. Using a digital tape or hard-disk multitrack it is commonly possible to delay individual tracks with respect to the others. Even if it isn't possible to advance a single track, you can always delay the rest, and perhaps make a delayed copy of the track you want to process. Armed with this you can connect the advanced version of the track to the sidechain - just 50 to 100 milliseconds should do - and the delayed version to the normal input. Now you will find that the compressor anticipates the amount of gain reduction required and transients in particular are rendered very much more realistically than doing things the normal way. In fact, you can do it the other way around too - delay the sidechain so that the compressor takes a moment to react. 'Why would you want to do this?', you might ask. The answer is that percussive sounds often benefit from a relatively slow attack, allowing the initial transient to come through unaltered before the 'body' of the sound is compressed. This is just a different way to do it, but this time with a little more control.

Radical Ratios

When is a compression ratio not a ratio? I could give you a straight answer but instead I would like to ask another question. Whoever said that it should be a ratio? Some scientist I don't doubt. Virtually every compressor on the market offers logarithmic compression, such that, once the knee curve is passed then, for example, at a 2:1 compression ratio a 10dB increase in level at the input will result in a 5dB increase in level at the output. This is all very tidy, but I wonder whether this is always going to be the right approach? How about a compressor where, once the signal exceeds the threshold, it is subjected to a knee curve leading to logarithmic compression, as tradition dictates, but beyond that the compression is lessened and the curve reverts to a straight line, meaning no compression . Here, signals of a certain level are compressed, but louder transients are substantially unaffected.

With traditional compression, it is usually the transients that cause the problems, so once you have got the general run of signal sounding pleasant, along comes a transient and the whole thing goes crazy for a second. Why not just let the transient through so it can be on its way, and concentrate on the parts of the signal that will really make a difference. You can always limit the transient later if you need to. There is actually a range of compressors that do depart from the traditional logarithmic curve. I'll give you a clue - they are all bright green in color. But there's a whole world of options waiting to be explored, by users and by designers. Compression is boring in comparison with what it could be. Why not have a bit of fun and experiment? Most of the ideas I've outlined here won't cost you a penny, and you may never have to buy another compressor again because you'll be getting all the fun you need from the compressors you already own!

Achieving The Best Recording With A Soundcard

2009-03-05T07:03:00.000-08:00

Achieving The Best Recording With A Soundcard

Although Mac and PC soundcards have improved greatly over the years, this doesn't automatically guarantee better recordings. Sometimes it can be difficult to get your input signals to actually appear in the on-screen level meter, and even when they do, people still complain of unwanted distortion, high background noise, or break through of unwanted signals from elsewhere. If you're relying on your computer to monitor the input signals, rather than using an external hardware mixer, you've also got to set up some sort of internal monitoring, and if set up incorrectly this can cause digital feedback.

Here are answers to some of the most common questions about recording quality with soundcards.

1. Why do I get so much background noise on my soundcard recordings?

This is the most common question of all, and there are various possible reasons why the quality of soundcardrecordings can be marred by noise. Background noise is normally a sign that the input signal has beenrecorded at too low a level, which should be fairly easy to correct, but can also be due to other things. Forinstance, if your soundcard features a mic input, and your line-level recordings have a lot of background noise,check that the mic input fader is turned right down, and preferably muted altogether if possible.

If you really want to squeeze the quietest recording from a soundcard, and its mixer utility provides some form of input gain or level control, do a few checks at different settings to see whether the amount of background noise changes when you record 'silence' (with no input plugged in). If it does, and you have a hardware mixer, it may be better to leave the soundcard input level near its optimum setting, and adjust levels using your external mixer. Many soundcards provide up to 20dB additional gain beyond their nominal 0dB input setting, but you will get better recordings using the soundcard at settings around 0dB and using higher-level input signals.

If you are recording using a mic, bear in mind that the mic preamps on soundcards aren't likely to compare with those on even budget hardware mixers or stand-alone mic preamps. This is because thecomputer is a noisy electrical environment, and providing the necessary high levels of amplification on asoundcard can introduce all sorts of stray hum, buzzes, and hisses.

2. Why can't I hear anything when recording or playing back audio?

This is the second most common question, and locating the source of the problem can be an extremelyfrustrating experience. If soundcard signals are missing, it is generally because one of the software mixercontrols is accidentally turned right down. Confusion is often arises because there are several controls availablewith similar functions; if you get used to setting levels inside a MIDI + Audio sequencer application, you mayforget that there is an additional mixer utility elsewhere. Windows provides a general Volume Control utility thatcan be launched from the Taskbar; if you don't have this installed it can be made to appear by ticking theappropriate box of the Audio page found under Control Panel/Multimedia. Playback and Recording controls areshown in two different pages, and you can choose which is displayed by selecting Properties in its Optionsmenu. Often the cure for a missing signal is simply to launch the Volume Control utility and turn the offendingsignal level back up.

If your soundcard has a dedicated DSP mixer utility available there will probably be many more options forrouting, monitoring, and effects, and therefore more possibilities for missing signals. If your playbackdisappears, or your input signals can't be heard when recording, the prime candidate is a mis-routedmonitoring system. Many soundcards now provide 'zero'-latency input monitoring for recording purposes, but insome cases you will have to switch this off again manually after recording before you can hear your tracks playback normally. Do take the time to study the possibilities in the manual, since there may be several ways to setup monitoring for recording purposes. Sadly, some software is not exactly what you'd call intuitive, and the bestway to understand the possibilities is to spend an hour or so recording and playing back signals while trying outthe different options -- which leads us neatly to the next question.

3.Why do my recordings always have a little bit of previous tracks recorded along with the newinput signal?

I've had various emails from readers complaining of 'ghostly' versions of previous tracks being added to freshrecordings. This is nearly always due to incorrect setting up of the soundcard's monitoring facilities: forinstance, the SB Live! card has a 'What U Hear' function that allows you to record the entire output of itssoftware mixer. If this is selected when multitrack recording, existing tracks will get mixed in with fresh ones,which is not normally what you want. Most musicians will want to monitor their performance at the same time ashearing existing tracks being played back, to enable them to keep in time, but will want to record theperformance in isolation.

To check that your mixer (either an external hardware one or a software one) is correctly routed, temporarilyunplug your input signal, and try a dummy recording. If the input meters still move during the recording, and thetrack contains anything but a little background noise, then something's amiss, and you need to track downwhere the stray signal is coming from. You may have to resort to reading the manual again to find themanufacturer's suggested routing, but in most cases it should be possible to monitor inputs and outputssimultaneously without needing an external hardware mixer.

Another potential problem with incorrectly routed mixer utilities is subtler, but can degrade the sound quality of every one of your recordings if you don't spot it. If any signal gets accidentally routed back to itself, it may emit an obvious howl of feedback that no-one could possibly miss. However, if the routed level is low, it may pass unnoticed, but still add a flanging effect to your audio recordings. Once again, the best thing to do is explore the routing options of your mixer, and make sure that you know exactly what each control does: you will normally get the cleanest recordings if you mute every signal source but the one you are actually recording.

4. What's the best way to set up soundcard recording levels?

In general you just need to watch the level meters included in whatever software you are using for recording purposes, making sure that you get as high a level as you can manage without ever running into clipping. When you are recording 'live' inputs on acoustic instruments you will have to allow a reasonable margin for unexpected peaks (depending on the type of instrument this might be 10 or even 20dB). However, where signal levels are predictable, you can optimize recording levels far more to squeeze the last drop of dynamic range from your soundcard.

For instance, many people still predominantly use MIDI sounds, but then use the audio facilities of their sequencer to make the final master recording from the stereo output of their hardware mixer. You can take advantage of the fact that there will be no unexpected surges in level by playing back the entire song from start to finish while monitoring the soundcard input level: on most sequencer level meters this will then give you a readout of the highest peak level reached during the entire song. You can then adjust either your hardware mixer output level or soundcard input gain accordingly, to ensure that the track peaks at between 1 and 2dB below full digital level. For instance, if your peak meter reads –6dB by the end of the song, you can increase the input gain of your soundcard by 4dB (many are now calibrated in dBs), so that when you make the final recording it will peak at about -2dB. This will ensure that your recording makes the most of the dynamic range of your A-D converters.

If your soundcard has a dedicated DSP mixer of some description, there may be several gain controls at different points in the signal path. In this case you might start hearing distortion in your recording long before the signal level hits the top of the level meter. You may even find it impossible to get your input signal to reach the top of the meter display at all, since it is clipping earlier in the signal path. The solution to this is exactly the same as when using a hardware mixer -- you need to set the input gain control to a suitable position to match the level of your input signal. In the case of a hardware mixer the PFL (Pre-Fade Listen) control lets you check the setting of the input gain control in isolation, but this facility may not be available in your soundcard mixer. Check your soundcard manual (printed or electronic) to find the recommended way to line it up.

By the way, if your soundcard provides a dedicated mixer utility with its own level meters (such as Event's Echo Console), it may be preferable to monitor input signal levels with these rather than the meters inside your MIDI + Audio sequencer, as they are likely to suffer far less from latency, and thus be far more responsive and 'in time' with the input signal.

5. Can I ignore levels when recording digitally?

In general, yes, since when recording digital input signals the data should simply be copied bit-by-bit onto your hard drive. However, some soundcards provide a mixer level control for the digital input, so you should make sure that this is set to unity gain if you want to preserve the original signal. This often just means pushing the fader all the way up (to 0dB if there are any markings). The only reason you would want to alter this is in the case of cards like the Emu APS and SB Live!, which can mix all their input signals (including the digital ones) together. In this case a digital recording should be treated just like an analogue one, by watching the level meters.

Similarly, if a level control is available for a digital output, leave it full up unless you need to use it to set a monitoring level (for instance with external USB-connected digital speakers). Any position other than maximum will compromise signal quality by reducing its dynamic range, so try to avoid using such controls altogether if you can.

6. My soundcard can record at 24-bit resolution at 96kHz. Are there any special precautions I need to take?

It can be frightening to see how large your recorded files become if your soundcard has 20-bit or 24-bit converters -- both will occupy the same amount of space on your hard drive, and be 50 percent larger than 16-bit ones. If you move up to a 96kHz sample rate as well, the file sizes become three times the size of 44.1kHz/16-bit ones, and this huge increase in data flow will also mean that your computer will manage far fewer simultaneous tracks with the same hard drive. For this reason, some musicians with 24-bit cards but less powerful computers still carry on recording at 16-bit resolution to save space or achieve more audio tracks.

However, now that some audio software allows mixed file types (Cubase VST allows 16-bit and 24-bit files to be freely mixed, while Sonic Foundry's new Vegas Pro lets you mix files of different sample rates as well), you can mix and match a little more. Some sources, such as MIDI synths and samplers, may not warrant using more than 44.1kHz and 16 bits, because many only have 16-bit converters outputting 44.1kHz sampled sounds. However, if you are recording acoustic instruments or analogue synths, 24-bit recording may well give you better quality. You can save a lot of hard drive space and gain more simultaneous tracks by choosing a sensible resolution for each track individually.

Many of the latest 24-bit soundcards dither the 24-bit signal output from their A-D converters down to 16 bits, if you choose to record at that bit depth. If this is the case you are likely to get better 16-bit recordings than when only using 16-bit converters. However, if your 24-bit soundcard doesn't dither automatically, it may be preferable to initially record a 24-bit file, and then use a software dithering process to reduce this to 16 bits after the recording has finished.

7. How can I get the best performance from a soundcard with my chosen MIDI + Audio sequencer?

First of all, make sure you have the latest version of the soundcard drivers, since this will normally give you the best combination of features and performance. All manufacturers keep the latest updates on their web sites for free download, but even if you're not on the Internet, most UK distributors should be able to post you the latest drivers on a floppy disk if you ask them nicely. On rare occasions a new driver version may be released with a bug that wasn't in previous ones, but this will normally be speedily discovered by irate users. If you are concerned about this then wait for a week or two after a new release for the dust to settle before you install the new version -- by then the braver users (or the more foolhardy, depending on your point of view) will have published their findings on various relevant Internet forums.

Windows Cannot Find RECYCLER\S

2009-02-20T14:48:00.000-08:00

Windows Cannot Find RECYCLER\S

I had this problem on my PC and thought I would share it with you and what I did to fix the problem.

When I try to open (double click) local C drive, I got a warning that Windows cannot find “RECYCLER\S-8-9-94-100022539-100013076-100012326-629 6.com make sure you typed the name correctly, and then try again. To search for a file, click the start button, and then click search."

This warning appears only on drive C, when I try to open the drive it just won't open. However when I right click on the drive and select explore then the drive will open. I thought it was probably a Malware or something.

I had this virus or whatever it was for weeks. I did some research and found some automatic removal and protection tools that did the job, and worked like a charm.

Here is what I found:

First download and run Autorun Eater 2.3 this took care of the most of my problem. Autorun is directly related to the autorun.inf file on the disk while auto play automatically reacts to new devices or media, for example a music CD is automatically played in Windows when inserted.

Autorun Eater 2.3 scans and remove suspicious ´autorun.inf´ files found in the root directory of all drives, C-Z, in real-time. Download Autorun Eater 2.3 Here

Download and Run FlashDisinfector
You have a flash drive infection. These worms travel through your portable drives. If they have been connected to other machines, they may now be infected.

Please download Flash_Disinfector.exe by sUBs and save it to your desktop.
Double-click Flash_Disinfector.exe to run it and follow any prompts that may appear.
The utility may ask you to insert your flash drive and/or other removable drives including your mobile phone. Please do so and allow the utility to clean up those drives as well.
Wait until it has finished scanning and then exit the program.
Reboot your computer when done.

Note: Flash Disinfector will create a hidden folder named autorun.inf in each partition and every USB drive plugged in when you ran it. Don't delete this folder. It will help protect your drives from future infection.

Download and run MalwareBytes Anti-Malware
If you already have MBAM installed, simply update and run a quick scan.

Please download Malwarebytes Anti-Malware setup and store on desktop.
alternate download link 1
alternate download link 2

Refer to the steps given here on installing MalwareBytes, running the scan, and saving the log file (not on using File Assassin).

If you have trouble updating, try the other mirror download site.
Should the computer in question not be able update using the normal method download the update file from here, using another machine if needed. Simple double click the file to install the updates.
If MalwareBytes asks to reboot to remove certain items, do so right away.

I hope this was helpful. Please let me know if this worked for you.

Please be aware that JoyWalk accepts no responsibility for the software you are downloading. The software listed, can modify without notifying Joywalk. Even if we try to check the files for viruses ourselves, we cannot guarantee 100% that they are clean. For your own protection ALWAYS check downloaded files for viruses.

Windows XP DAW Optimization Guide

2009-02-07T22:02:00.000-08:00

Windows XP DAW Optimization Guide

Please read all of the following before attempting to change any settings in Windows XP. JoyWak Music is not responsible for any thing that may render your computer system unusable. We provide this information as suggestions to increase performance and get the best user experience possible.

Installing Windows XP

When you see the installation tell you to
press F6(Third Party SCSI or RAID Drivers)
press F5 instead. You will see Press F2 for Automated System Recovery(DONT press F2)
Right after that you will see a list
Press the UP arrow key to highlight Standard PC
Hit Enter..
Hit Enter to Continue
Hit F8 saying you agree.
Now, if this is an existing OS of say Win2k or a previous install of XP you will see options for
Esc=Don't Repair; R=Repair F3 to Quit
Well, we're doing a fresh install so we want to hit Esc for Dont Repair
Now, you'll see your drives and partitions here..
If you have Two "physical" drives it will show you these drives as C and D
Highlite C Drive
If you had an existing Install from any other OS press D for Delete Partition.
Press Enter to Continue
Press L for Delete

Now, we're back at the drive selection screen again and this time we see Unpartioned space..
We're ready to create a partition.

Typically you want to the OS drive to be as small as possible for drive reading purposes.
Reccomended OS and Application drive should be around 12GB tops.
If you have a 20GB drive partition it into two parts..
the first part being 10,000MB and the remaining to the
Second partition(comes to around 9500MB)...Which can be used for extra storage

Once the drive is partitioned and the main Parition is highlited press Enter to Install
Now we have to decide what File System do we use...NTFS or FAT32

FAT32 vs. NTFS

You have the option under W2k and XP to choose the file system that best suits your needs, FAT or NTFS. FAT (File Allocation Table) is the native file system based on the Windows 9x kernel (including 95, 95a, 95b, 95c, 98, 98se and ME). NTFS (New Technology File System) is the native file system for operating systems based on the Windows NT kernel (including NT4, 2000 and XP). During the installation, Win2K or XP will ask if you want to convert the installation partition to NTFS. If you need compatibility for Windows 98 – especially if you want to dual-boot – don't convert.

While NTFS offers a number of improvements over FAT32, most of these advantages are not all that applicable to audio, and you are unlikely to see a major performance difference between the two. Moreover, defragmenting your drives (something you should do on a regular basis anyway) is substantially quicker in FAT32. Be aware that Windows 9x cannot read data on NTFS-formatted drives. Generally speaking, you should use FAT32 if you are doing a parallel installation with Windows 9x/ME, or if you will need to work with files created with a FAT32 system (opening old song files, working with others using FAT32-based systems, etc.) leave everything with FAT32.

One exception would be if you’ll be working with video or other large files in excess of 4GB; or if you also use your PC to browse the internet or the computer is part of a LAN running XP Professional. Only NTFS can limit access rights to your files and therefore provide the security needed for a networked computer. In this case, select NTFS for all partitions except those which are to accommodate your audio data later.

One other interesting point: NTFS can read the "resource fork" of SDII files from MacOS, and can therefore recognize the timecode stamps used in these files; FAT(32) can’t do this. If you do a lot of work with ProTools or other MacOS-based DAW applications, you should consider at least one NTFS partition in your system. (Note that this does not imply that your NTFS-based system can mount/read from a MacOS drive; simply that individual SDII files imported to an NTFS drive will retain their time-stamp information.)

Windows will begin it's file copying, once it's done it will restart your machine.
It's a good idea from here to enter into the BIOS to stop the CDRom from being the
first boot device Set the Hard Drive as the main boot. Exit out and restart to begin the XP
installation Enter your Name and Orginzation(If Applicable..you can leave it blank)
Enter your Windows Key
Name your computer!
Hit Next
Continue Installation
If you have a NIC card it will ask you what type of Installation do you want to choose
Typical or Custom
Do Typical for faster install

Ok. so now we have a fresh install of Windows XP.

When XP starts up everytime you get that anoying Take the XP Tour pop up...
click on it to open the tour..once in the tour simply exit out and it wont open up anymore.

Turning of Windows Messenger from start up

Double click on the Messenger icon in the system tray to open it. Skip thru the internet and sign up stuff, just cancel it. When Messenger loads go to tools and Options then Preferences and uncheck 'Run this program when windows starts'

Switching to Classic Mode

Swithcing to Classic Mode is better for system performance because it uses as little colors or graphics as possible:

Right-click on your desktop, and then click Properties.
Click on Themes tab
Set Themes to Windows Classic
Click on the Screen Saver tab
Set Screensaver to None
Press the Power button near the bottom
Power Schemes..you can have the monitor turn off but set Turn Hard Drives off to NEVER Hibernate..If this is Enabled uncheck it. This is mainly for Laptops but uses a very large chunk of data. (I've seen this not show up on the first boot of XP..when you restart it will be there but it's disabled) APM..Enabling this will allow your computer to shut down properly when in Standard PC mode Hit OK
Click the Appearance tab.
On the Windows and Buttons menu, select Windows Classic
Press Effect button
Deselect all options.
Hit OK
Click the Settings tab
Set your bit depth to 16Bit. This is optimal for Audio machines due to less colors for video drawback which in turn gives you better audio performance

Optimizing the Start Menu

Right–click the Start button, and then click Properties.
Click Classic Start menu.
Click the Customize button to select items to display on the Start menu.
By default, selecting the Classic Start menu also adds the My Documents, My Computer, My Network Places, and Internet Explorer icons to your desktop.

Optimizing Computer properties

Right Click My Computer and select Properties:
System Restore tab:
Check the Turn System Restore on all drives.
Automatic Updates tab:
Turn Off Automatic Updates.
Remote tab:
Uncheck all options to turn off Remote Assistance.
Advanced tab:
Press Settings tab under Performance
Visual Effects tab:
set to Adjust for Best Performance.
Advanced tab:
Processor Scheduling:
Set this for Background Services
Memory Useage:
Set this for System Cache
Virtual Memory:
Press Change...
Depending on how much RAM you actually have is what you are going to enter here

If you have 256MB RAM set this to 512 for Min and Max
If you have 512MB RAM set this to 768 for Min and Max
If you have 1024MB RAM set this to 1536 for Min and Max
Once entered hit SET..Hit OK and then Hit OK..

Restart your machine at this point in time...
When you come back the first thing you should do is defrag the main drive even if it doesnt say it needs it. This way the swap file has been truly set and you're ready to continue.

Modifying the Windows XP Services

********FIRST THINGS FIRST!! MAKE A BACK UP OF YOU REGSITRY ***BEFORE ***DOING ANY REGEDIT OR SERVICES TWEAKS OF ANY KIND!!!!!!!!*********

Start Menu, Run...type in regedit and hit OK...
Hit the Drop menu for Registry and select Export Registry..
save this to another drive for safe keepings.

Now, let's stop that annoying balloon from popping up from our system tray
This is a Registry Tweak we have to do..

Start menu>Run...type in regedit
Hkey_Current_User\Software\Microsoft\Windows\Current Version\Explorer\Advanced
If enableballoontips is there set the value to 0
If it doesn't enter it in as a new DWORD and put the value to 0

What we want to do is turn off certain "services" that are running in the background that we dont need while recording or playing out Audio Software.

Control Panel>Administrative Tools:
Double Click on Services
Here is a list of what to disabled.
Keep in mind this list is for a computer that doesnt use the internet or a network in any way. If you have an Network Card or modem of any sort, pay attention to those services and what settings are selected.

Alerter Disabled
Application Layer Gateway Service Disabled
Application Management AppMgmt Manual
Automatic Updates Disabled
Background Intelligent Transfer Service Disabled
ClipBook Disabled
COM+ Event System EventSystem Disabled
COM+ System Application Disabled
Computer Browser Disabled
Cryptographic Services Disabled
DHCP Client Disabled (Set this to Manual for Internet)
Distributed Link Tracking Client Disabled
Distributed Transaction Coordinator Disabled
DNS Client Disabled (set this to Manual for Internet)
Error Reporting Service Disabled
Event Log Automatic
Fast User Switching Compatibility Disabled
Fax Service Disabled
Help and Support Disabled
Human Interface Device Access Disabled
IMAPI CD-Burning COM Service Manual
Indexing Service Disabled
Internet Connection Sharing Disabled
IPSEC Services PolicyAgent Disabled
Logical Disk Manager Manual
Logical Disk Manager Administrative Service Manual
Messenger Disabled
MS Software Shadow Copy Provider Disabled
Net Login Disabled
NetMeeting Remote Desktop Sharing Disabled
Network Connections Manual
Network DDE Disabled
Network DDE DSDM Disabled
Network Location Awareness (NLA) Disabled
NT LM Security Support Provider Disabled
Performance Logs and Alerts Disabled
Plug and Play PlugPlay Automatic
Portable Media Serial Number Disabled
Print Spooler Disabled
Protected Storage Disabled
QoS RSVP Disabled
Remote Access Auto Connection Manager Disabled
Remote Access Connection Manager Disabled
Remote Desktop Help Session Manager Disabled
Remote Procedure Call (RPC) Automatic
Remote Procedure Call (RPC) Locator Manual
Remote Registry Service Disabled
Removable Storage Disabled
Routing and Remote Access Disabled
Secondary Logon s Disabled
Security Accounts Manager Disabled
Server Disabled
Shell Hardware Detection Disabled
Smart Card Disabled
Smart Card Helper Disabled
SSDP Discovery Service Disabled
System Event Notification Disabled
System Restore Service Disabled
Task Scheduler Schedule Disabled
TCP/IP NetBIOS Helper Service Disabled (set this to Manual for Internet)
Telephony Disabled
Telnet Disabled
Terminal Services Disabled
Themes Disabled
Uninterruptible Power Supply Disabled
Universal Plug and Play Device Host Disabled
Upload Manager Disabled
Volume Shadow Copy Disabled
WebClient Disabled
Windows Audio Automatic
Windows Image Acquisition (WIA) Disabled
Windows Installer Manual
Windows Management Instrumentation Automatic
Windows Management Instrumentation Driver Manual
Windows Time Disabled
Wireless Zero Configuration Disabled
WMI Performance Adapter Disabled
Workstation Automatic

Once you have set all of these close out of the services and restart

Modifying the Registry

****Again...it may be a good idea to make a back up of the registry *****

Intel Chipsets need to have UDMA 66 enabled for Win2k and XP. This also enables UDMA100
HKEY_LOCAL_MACHINE\System\CurrentControlSet\Control\Class\
{4D36E96A-E325-11CE-BFC1-08002BE10318}\0000
You will need to add this value in
Right click and select New..DWORD
Type EnableUDMA66 and hit Enter..and Double click it and put the value to 1
It should the look like this:
EnableUDMA66=dword:00000001
Again, this is really only for intel chipsets.

Remove the Shared Documents folders from My Computer:
Windows XP user interface provides links to all of the Shared Documents folders on your system,
right at the top of the My Computer window.
HKEY_LOCAL_MACHINE\SOFTWARE\Microsoft\Windows\CurrentVersion\Explorer\MyComputer\
NameSpace\DelegateFolders
Delete this Key
{59031a47-3f72-44a7-89c5-5595fe6b30ee}

NTFS Disk Performance
The NTFS file system is the recommended file system because of its advantages in terms of
reliability and security and because it is required for large drive sizes. However, these advantages
come with some overhead. You can modify some functionality to improve NTFS performance as follows:

1. Disable creation of short names. By default,NTFS generates the style of file name for compatibility
with MS-DOS and Windows 3.x clients. If you are not supporting these types of clients, you can turn off
this setting by changing the default
HKEY_LOCAL_MACHINE\SYSTEM\CurrentControlSet\Control\Filesystem
NtfsDisable8dot3NameCreation set value to 1

2. Disable last access update. By default NTFS updates the date and time stamp of the last
access on directories this update process can slow performance. To disable:
HKEY_LOCAL_MACHINE\SYSTEM\CurrentContolSet\Control\Filesystem
You will need to enter this as a new Dword:
NtfsDisableLastAccessUpdate set value to 1

3. Reserve space for the master file table.
HKEY_LOCAL_MACHINE\SYSTEM\CurrentControlSet\Control\FileSystem.
You will need to enter this as a new Dword
NtfsMftZoneReservation set to 1

Reboot after making changes.

Speed up the Start Menu in Windows XP.
HKEY_CURRENT_USER\ControlPanel\Desktop
MenuShowDelay file set to 1
Click OK.
A restart is needed to see the results

Increase IRQ Priority of CMOS Real-time Clock
Improve overall system performance by increasing the IRQ priority of the CMOS real-time clock.
HKEY_LOCAL_MACHINE\System\CurrentControlSet\Control\PriorityControl
You will need to enter this as a new Dword:
"IRQ8Priority" set to 1

Windows Explorer caches DLLs(Dynamic-Link Libraries)in memory for a period of time after
the application using them has been closed. This can be an inefficient use of memory.
HKEY_LOCAL_MACHINE\SOFTWARE\Microsoft\Windows\CurrentVersion\Explorer
Create a new DWORD sub-key named
"AlwaysUnloadDLL" set to 1
Restart Windows for the change to take effect.

Speed up CD Copying to Hard Drives
HKEY_LOCAL_MACHINE\SYSTEM\CurrentControlSet\Control\FileSystem
You will need to create a Key here. Right Click on the FileSystem folder and select New>Key
Name it CDFS
CacheSize, this must be added as a binary value, then type in this value: ff ff 00 00
Prefetch, this key must be added as a DWORD value, then type in this value: 4000 hex
PrefetchTail, this key must be added as a DWORD value, then type in this value: 4000 hex

To Disable Dr. Watson
HKEY_LOCAL_MACHINE\Software\Microsoft\WindowsNT\CurrentVersion\AeDebug
Delete the AeDebug key

Other Windows XP Tweaks

Now is the time to Install your hardware drivers if you havent done it allready.
It makes no difference what order you install the drivers in.
After all the hardware drivers are done install your software.

Turn off CD Autoplay
Go to Start->Run->gpedit.msc
Computer Config -> Administrative Template -> System
Double click Turn off Autoplay
Enable it.

Control Panel>Double click Sounds and Audio Devices...Go to the Audio tab
Set the Playback and the Recording settings to either
Your built in soundcard or a soundblaster(If Applicable)
Check Only Use default devices down below
Go to the Sounds tab
Under Sound Scheme choose No Sounds
Press No to saving the previous Scheme
Hit Apply

Do this for all NTFS drives:
Open My Computer>Right Click on C:Drive and select Properies.
Deselect Allow Indexing Servicing....Hit Apply
Select Apply Changes to C:\ subfolders and files
You might get a message that says Access Denied...press Ignore All

By default, Windows 2000 logs the I/O traffic of your hard drive.
While this is a very useful setting for servers, for workstations it
doesn't do anything except use up system resources.
To disable it, go to the run menu and type
diskperf -n
Hit Enter to disable this logging.

Disable Error Reporting
Control Panel>Performance and Maintenance.
System>Advanced tab
Error-Reporting button
Select Disable Error Reporting.
Click OK
Click OK

Uninstall unwanted components (good for non networked computers)

Locate sysoc.inf (windows\inf\sysoc.inf) on the main drive and make a backup of it
Open the Sysoc.inf file. Each line of text in the file represents a component that
can be displayed in the Add/Remove Windows Components dialog. Delete the word HIDE for any component that you want to see in the dialog (do not erase the ommas). Save the Sysoc.inf file, then close it, and reboot your computer.

The Add/Remove Windows Components dialog will now display the items you want to remove.

20 Best Computer Recording Tips

2009-01-03T09:16:00.000-08:00

AUDIO RECORDING WITH COMPUTER SEQUENCERS

The recording studio inside the virtual world of the computer is real enough, but sometimes you have to treat it with care to get the best from it.

Computers offer us MIDI, audio recording, mixing, virtual effects, virtual synths and CD manufacturing facilities, but it doesn't pay to take them for granted. The following tips will help you get the best out of your system, whether it runs on a Mac or PC, and most assume that you already have a system that's up and running. If you want to buy a Mac system, either buy one of the newly obsolete grey Macs or wait until the peripherals and software copy protection needed to work with the new candy-colored Mac’s are ready. In either case, buy the fastest machine you can afford - even if you can't afford it!

1. Optimize your input signal level at source rather than relying on normalization to bring the level up: if your signals peak at only half the maximum level, you're effectively halving the signal-to-noise ratio of your recordings and wasting half of the theoretical resolution of your system. Digital processing such as EQ or reverb may also introduce far more noticeable rounding and quantizing errors in low-level recordings. Uses the level metering provided in the software and try to keep your peak levels just a few dBs below clipping.

2. Regardless of whether you have 16-, 20- or 24-bit recording, the real quality of your recording will be defined by the source. For vocals, consider buying a voice channel type of device that combines a good mic amp with EQ and compression. This may also be used when micing other instruments, and many feature an instrument DI input suitable for use with bass and clean electric/electro acoustic guitar.

3. The fact that computers and recording software are such good value for money can lead you into believing you can make do with equally cheap components in the rest of the studio. This simply isn't true. With good capacitor vocal mics, there's no excuse for using your old gigging dynamic microphone.

4. Use quality monitor loudspeakers and set them up so that you're at the apex of a roughly equilateral triangle with the monitors pointing directly at you. You don't need to monitor loudly, but you do need enough volume to overcome the physical noise your computer fans and drives make.

5. Use a separate hard drive for audio if at all possible as this will increase the number of tracks you can play back at the same time. This also allows you to defragment, or even reformat, the drive regularly without disturbing your program files. Most modern drives are suitable for audio use, but if in doubt, get a drive that is suitable for AV applications. The faster the drive you buy the more tracks you'll be able to play back. If you really can't afford a separate drive, at the very least create a separate partition on your main drive for audio use.

6. When choosing or upgrading a sound card, try to get one that can provide at least four outputs - and a digital S/PDIF out if you own a DAT machine or Minidisc recorder. This way you can use one pair of outputs for tracks that use software-based plug-in effects while the other output can carry tracks that you want to effect using external processors. If you can upgrade to a firewire sound card you will have even more flexibility.

7. Reverb is the most important effect in the studio, and good reverbs take up a lot of computing capacity. For this reason, it may be worth considering buying a soundcard with its own hardware reverb processing.

8. Unless you are using a fairly sophisticated soundcard with onboard DSP processing, you're likely to experience some latency or delay when monitoring the signal you're currently recording through the system. The ASIO II drivers will minimize this problem for compatible hardware, but it won't cure the problem in all soundcards. An alternative is to use a small mixer and arrange to monitor the computer's input rather than its output when overdubbing - a separate mixer will usually be needed to combine your audio and external synth/sampler signals anyway. Monitoring the input source will avoid latency problems, but will mean you have to monitor without plug-in software effects. However, a simple hardware reverb unit is generally all that's needed to put you in the mood for a good performance, and you can probably make use of this when mixing if your card has more than two outputs.

9. Use Antares' Autotune plug-in not only to clean up vocal pitching, but also to tighten up guitar solos. (As long as you set a slow enough tracking time, regular playing will be unaffected, but whenever you sustain a note, it will automatically settle on exactly the right pitch. This can be particularly useful for slow pieces that use a lot of string bends. You can also emulate that Cher 'Believe' vocal-type sound extremely convincingly by just setting the tracking speed to maximum and dialing in the correct key for the song rather than leaving Autotune on its Chromatic setting (although of course, Cher's producers claim Autotune was not used on that recording).

10. One problem that most guitarists come up against is that the computer's monitor interferes badly with the guitar pickups, resulting in a nasty buzz on the recording. Some humbucking pickups are reasonably good at rejecting this buzz providing you don't sit too close to the monitor while recording, but single-coil pickups tend to be very badly affected. One way around this problem is to switch off the monitor just before recording and use keyboard commands to start and stop the recording process. If you can't switch the monitor off for some reason, sit as far away from it as possible when recording and rotate your position to find the null point where the buzz is least obtrusive. You might also use a noise gate pedal to keep your guitar quiet between phrases. Flat-screen LCD monitors; they both save space and eliminate the electromagnetic interference generated by the scan coils of a typical monitor. If you record a lot of guitar, or are short on space, such a monitor could be a good investment.

11. Physical noise is also a problem when mic’ing instruments or voices in the same room as the computer. If possible, turn off unnecessary external drives, CD-ROM burners and so on, as these often make more noise than the main computer, Set up your mic (ideally a cardioid model) as far from the computer as possible and improvise an acoustic screen between the mic and the computer using a duvet or sleeping bag. Also make sure the surface the mic is pointing at is absorptive rather than reflective. Work as close to the mic as you can without compromising the sound (and always use a pop shield for vocals).

12. Virtually all sequencers capable of recording audio have a waveform edit page (though it isn't always called that) where it's possible to highlight and silence selected portions of audio. If background noise was a problem, you can sometimes improve matters by manually silencing all the gaps between words and phrases. This doesn't take as long as you think and can really improve the quality of a recording, especially where there are multiple audio tracks. It's a good idea to normalize your audio recordings before processing them so as to minimize rounding errors at the processing stage; though don't use this as a substitute for getting the record levels right in the first place. Normalizing can generally be done from within the waveform edit page.

13. You can also use the waveform edit page to clean up guitar solos. Often you may end up with an almost perfect take, but perhaps there's too much squeak or finger noise between notes, or maybe you caught the next string just after bending a note. You can use the silence function to surgically remove these little errors, though you may end up with a more natural sound if you leave them where they are but instead reduce them in level by between 6 and 20dB.

14. Try to record all parts dry - don't add reverb or delay unless you really have to. If you need to hear reverb to create a good performance, fakes it at the monitoring stage, but don't record it. This way, you'll be able to edit tracks without cutting holes in the echo or delay effects you've added, then when the editing is done, add the necessary delay or echo, which will help hide your edits, making the recording sound quite natural.

15. Plug-ins always take up a certain amount of your computing power, so if you want to add the same delay or reverb-based effects to several tracks, use a single plug-in configured as an aux send processor rather than using a separate Insert plug-in on every track. You can use the Aux Send controls in the same way as those on a regular mixer to add different amounts of the same effect to any tracks you like, all for the CPU overhead of a single plug-in. Note that under normal circumstances, you can't use the aux send with processes such as EQ, compression or gating - these have to be inserts.

16. Often, it's cheaper to buy a hardware reverb unit or signal processor than to buy a decent plug-in that does the same job and the chances are the hardware unit will still sound better. Don't try to force your software to do everything for you just because it can - very often you'll find you can get a better sound with discrete boxes, and of course they won't load your CPU. Even if you don't have a multi-output soundcard, you can still compress signals as you record them, ideally using a voice channel type of device as described earlier, and the same applies to EQ. Only the best digital EQs sound as natural as even the most basic analogue equalizers.

17. There are lots of tricks you can do using the audio manipulation facilities provided by your sequencer. These vary from model to model; pitch-changing and time-stretching, which are invaluable for massaging audio sample loops, are supported by most machines. You may also find other tools for level maximizing, denoising and so on. Many of these works off-line, so you can use them even on a slower machine - you just have to wait around a while for the results.

18. Consider using CD-R/DVD-R to backup your audio files along with your song files. Though you can't rewrite a CD-R/DVD-R, they're so cheap now that it doesn't really matter. If you create a 600Mb partition on one of your drives and store (or copy) you’re audio and song files there, you can back up the entire partition in one go. Of course the same CD-R/DVD-R machine can be used to burn audio CDs of your finished songs.

19. Most computer audio systems run best if you get rid of any superfluous software such as screensavers and games – and make sure you have no more drivers than you actually need (Extensions for Mac users). The cleaner your system, the less likely you are to run into problems. Also, check manufacturer’s web sites to make sure you have the latest drivers as improvements are being made all the time.

20. Do some tests to find out how many tracks and plug-ins your machine can run without falling over, and then try to work with no more than half to two-thirds this number. Most sequencers include some kind of CPU activity monitor to help you. The demands on your CPU aren't constant, and sometimes a lot of heavy processing loads can be imposed at the same time, which can cause a machine running close to its capacity to crash. Your disk drive will also slow down as it fragments, so try to allow for this - you can't be expected to defragment it after every track you record.

Recording Techniques

2008-12-12T15:23:00.000-08:00

GIVING YOUR RECORDINGS A 'PRODUCED' SOUND

Why is it that some perfectly well recorded songs sound like demos, while others sound like top commercial tracks?

One of the questions we hear most is "Why doesn't my music sound as 'produced' as the music I hear on commercial CDs?" I'm sure you won't be too surprised when I tell you that there isn't a single, simple answer. Some people assume that the superior equipment used in pro studios is the key, but although competent gear is required to do the job properly, you don't actually need anything esoteric. Even when it comes to recording vocals you don't have to use expensive high-end tube capacitor mics -- artists such as Phil Collins and Mick Jagger often use relatively inexpensive dynamic models because that's what works best for them. A few years ago, the drum sound was what gave away most demos, but now we have good drum machines, drum software, and drum samples or loops, as well as real drums, to choose from.

The secret of a produced sound starts with the source material. It doesn't matter what you do to your recording afterwards if this isn't up to scratch. It almost goes without saying that good timing and good tuning are essential, but the choice of sounds and the way in which acoustic instruments and voices are recorded has a huge bearing on the perceived quality of the end result.

Clever Vocals

If you record vocals in a small, untreated room, the chances are that the resulting sound will be boxy, so place your mic somewhere near the center (but not exactly in the center) of a larger room and put up improvised screens (sleeping bags, duvets, blankets and so on) where necessary to kill the reflections. Used in this way, virtually any respectable mic will give you good results providing you use a pop shield. You can also record acoustic guitars in the same environment.Vocals invariably need compression, but what kind and how much? Listen to what you've recorded and try to establish how much variation there is in the vocal level. If you hear a lot of fluctuation it might be better to use a model of compressor that can pin down the level without changing the sound too much. On the other hand, you may feel the vocals need thickening as well as leveling, in which case a compressor with a character of its own might be better suited to the job.

Tube and 'opto' compressors generally produce the fattest sounds, and of course there are software plug-ins that emulate just about anything you can buy in a rack mount box.The goal is to get the vocal sitting nicely with the backing track so that you don't feel the urge to turn it up or down in different parts of the song. Professional engineers may also spend some time fine-tuning vocal levels with their mixer automation systems, and if you use either a digital mixer or a computer-based recording system you can do the same.

Synth Key Facts

Synth sounds must be chosen with care, because a lot of factory patches are designed to sound big and impressive for the benefit of those who choose their new instruments on the strength of 'preset cruising'! What sounds wonderful on its own might take up too much space in a mix so, if you don't want to edit the patch, try using EQ to trim off excess bass or high end. The EQ'd patch might sound odd in isolation, but it may well fit the track better. Another tip for those reluctant to get into heavy editing is to layer patches to get the desired result. For example, a deep bass sound mixed with a more percussive patch might help you produce a bass that you can hear as well as feel.

It's important not to over-orchestrate your arrangements, especially when you have fat synth pads and overdriven guitars occurring at the same time. The same is true of some treated drum loops, which can actually take up a lot of space. If in doubt, listen to some commercial mixes in a similar style to the track you're working with. You may be surprised at how little there is going on at any one time.

It may help if you get your sounds as close as possible to correct at source so you don't need to use a lot of EQ. Few budget mixers have the kind of EQ that works well when called upon to make major tonal changes, and often you'll find that the more you EQ, the harsher, boomier or less focused your mix becomes.

Reduced Reverb

Once you've created space in your mix, don't give it all away by filling every available gap with heavy reverb. As it happens, reverb is one area where a decent-quality unit really helps, especially if you use a lot of small-room or ambient reverb. Bare in mind that heavy reverb tends to push a sound to the back of a mix, so if you want a vocal to appear up-front you should use a fairly bright reverb, with 80mS or so of pre-delay. Don't overdo the decay time, either, especially with up-tempo songs. Other effects should also be used carefully -- use an effect because the track needs it, not because you happen to have it! Dramatic effects can be made even more dramatic if you use them for short sections of a song rather than having them full-on all the way through, and delay effects often work best when the delay time is related to the tempo of the song.

Favorite Strings

Guitars and basses can be a dead giveaway that a recording is not a commercial one if they are poorly recorded. Sticking a mic in front of an amp is probably still the best way to get a live sounding recording of a performance, but if this is not feasible there are so many good recording preamps around now that there's little excuse for getting a thin or buzzy guitar sound. However, go easy on the overdrive, and consider using less overdrive but combining it with compression if you need sustain. Use a gate to keep your guitar tracks clear of unwanted noise, and also try to reduce clutter in the arrangement: where two guitars are playing essentially the same chords, for example, first decide whether both guitars are actually necessary. If they are, consider using different chord inversions for one of the parts, or even a capo. Incidentally, acoustic guitars almost always sound better miked than DI box. Basses can actually be more difficult to record than guitars, because although they may sound great in isolation with an active DI box and a compressor, they can still lack punch in the context of the overall mix. Again, consider miking the amp or using a guitar DI preamp so you can add just a little overdrive to warm up the sound. Compression will help keep the sound even and punchy. A good tip here is to make any necessary EQ adjustments when the rest of the track is playing, because then you'll be able to make the sound match the track. If you EQ the sound first it might sound great on its own, but could get completely lost when the other faders are brought up.

Equalizer

A good equalizer doesn't just change the spectral balance of a sound: it also seems to lift information out of a mix. One popular mastering technique is to apply an overall boost of just one or two dBs at around 15kHz with a wide bandwidth setting. This is what people mean when they talk about 'air EQ', 'sheen' or 'gloss'. With a nice equalizer this boost will lift out high-end detail while at the same time pulling the vocals forward, but it shouldn't make the sound harsh or topy. Similarly, adding a gentle dip at around 180-250Hz may help clarify a muddy lower mid-range, while a boost at 70-90Hz will firm up a weak bass end. It is vital to use a classy equalizer for this job, though -- a cheap one just won't deliver the necessary fairy dust!

Compression

A very gentle overall compression of around 1.1:1 with a threshold of -30 to -40dB will make a mix sound more even and more powerful. However, multi-band mastering processors add a lot of flexibility in the area of compression, because they give you the opportunity to perform operations such as applying more compression to the bass end than to the rest of the mix. This helps firm up the bass end only, and adjusting the levels of the various frequency bands at the compressor’s output can restore any spectral imbalance caused by the different compression ratios.

Master The Situation

What many people don't realize is just how great a difference is made to commercial records at the mastering stage. Prior to mastering, you might be surprised at just how ordinary some mixes sound. Mastering often involves nothing more than compression, limiting and equalization, (but with an infinitely high ratio) which control just the tips of loud peaks. Applying a little limiting will often make it possible to increase the average. "The secret of a produced sound starts with the source material. It doesn't matter what you do to your recording afterwards if this isn't up to scratch."Get the level of a mix by several dBs without any side effects becoming audible. If you're starting from a 20- or 24-bit master and you reduce to 16-bit right at the end of the process, this has the benefit of using the whole of the bit resolution of the CD format, which means less noise, less distortion and better low-level resolution. It also makes your CD sound as loud as the 'produced' commercial CDs in your collection. Use a limiter specifically designed for mastering (such as the Waves L1 plug-in or the limiter in your mastering processor) and don't over-limit, or you will start to hear the difference. Usually 4-5dB of limiting is all that's needed.A note on limiting: Any decision taken to limit or not to limit is a musical one. Some musical styles apply heavy limiting as part of the musical style’s “sound”, others don’t. Production requirements may suggest limiting is needed, for example preparing your music for broadcast might necessitate limiting in order to compensate for the radio’s smaller dynamic range. In our example we exaggerated in our limiting setting – The student should understand that limiting to produce 5.2dB attenuation is a bit heavy. Normally we should watch out for a maximum of 4dB attenuation.

Warm Tube

Processing via tube or simulated tube circuitry can also warm up a mix (which is why tube EQs and compressors are popular for mastering), but again you get even more flexibility if this tube processing comes as part of a multi-band package. For example, adding a little gentle tube saturation only to the low band will noticeably thicken the bass and kick drum without spilling over into the midrange and high end. Similarly, adding high-end saturation has an effect similar to an enhancer, enhancing detail and lending gloss. The secret with all these treatments is to use them sparingly and always compare the processed sound with the unprocessed to make sure you have not gone too far. A good processor will transform a recording with just a dB or two of adjustment where needed. If you find you're using a lot of processing, suspect your basic mix of being too wide of the mark.

Summing Up

As you can see, the magic of musical production isn't something you 'paint' on at some point in the recording process, but is rather the result of attention to detail at all points throughout the recording, starting with the musical arrangement and choice of sounds. Nevertheless, processing at the mastering stage (ie. after your mix) can make a huge difference. Professional mastering is expensive for a reason: pro mastering engineers have great equipment and a lot of experience in using it. If you're not confident you have the necessary equipment and expertise to do your mix justice, think about getting your work professionally mastered, especially if it's destined for commercial release. If you're going to do this, don't do any processing at all on your final mixes -- leave each track just as it is.

2008-11-18T10:26:00.000-08:00

Recording & Composing:

JoyWalks digital mobile recording service will allow us to come to any location you prefer. We can travel to any suitable venue, from rehearsal rooms, chapels, to schools and theaters. You can count on a real budgeted way to ensure you have the best sound recording possible.

For the person who has musical or lyrical ideals for there song, but have no equipment or knowledge on how to compose them. We offer a song writing and tracking service, a full suite of music production software and hardware. Here is just a sample of what we use.

Software:
Samplitude
Sonar
Reason, Waves

Mixers:
Mackie 24-8
Roland VM3100 Pro
Yamaha n12
V-Studio 100

Microphones:
Audio – Technica
AKG
Shure, CAD

We use several PC's loaded with P4 - 3.0 Ghz processors, 2GB of RAM, and 500GB of hard drive space.

But what's more important than all of this professional equipment is our experience, knowledge, and a producers ear to here the sound and music you are trying to create. Joywalk will develop your music, with pre-production, producing, tracking, song writing, mobile recording and CD mastering.

We are committed to provide the highest quality to these services.

We will help you produce a sound, style and message that meets your expectations. When your project is complete, we will deliver to you a CD that is ready for the radio air waves

Affordable Recording

No hourly rates, that's right NO HOURLY RATES, $30.00 pays for

recording of 1 song, or mixing of 1 song, or mastering of one song

Tracking & Composing of your music:
$350.00 (For tracking of one song)

If you have multiple songs we have affordable package deals.
Hourly rates are not charged.

Contact us for pricing on packages,
prices vary because of the amount of songs you want created.

The mobile recording services is only available for the Memphis TN and surrounding areas.

CD Mastering

JoyWalk Music (of Memphis TN) will add the ultimate spices to your already great mix. Taking your mix to the next level at a price you can afford. Restore and improves masters that have too much noise by removing hiss, distortions, clicks, and pops, using Waves™ Restoration tools. Your music deserves the finest in care and experience.

EQ-ING: One of the most important steps in mastering is applying just the right amount of equalization to the original mix and knowing what frequencies to alter and which ones to leave alone. Next to compression, this is probably the most crucial step. JoyWalk uses the newest generation equalizers that, unlike older analog gear, won’t alter the phase correlation of your mixes.

COMPRESSION: Compression can hugely alter the perception of a mix, but too much will result in a loud but lifeless and thus boring master. Too little won’t push the track with enough force to the listener, or, as in classical orchestral music, will be simply too much dynamic difference for normal listening situations. The character and sonic signature depends largely on what kind of compressor is being used and what specific settings are being applied. JoyWalk Music has a huge range of state of the art tools to accomplish this in a world class way.

If your music is your investment, you can't skip these crucial steps.

Sometimes, we use Our little trade secrets to create enhancements that vastly improve the original mix. An ambiance that wasn’t there in the original recording, a bass signal that couldn’t be heard even with eq-ing and compression, Stereo Imaging techniques to enhance and alter stereo separation, an effect that helps connect the listener to the next part of the song.

EDITING: Track assembly, sequencing, and fading between tracks are done using the editing tools of our Samplitude and Sonar digital stations. Our Memphis clients trust our musical expertise and editing skills to create for them a final product that is seamless and captures the musical sensibility of the recording artist.

BURNING: This is the only non-musical stage, the burning of the master CD, usually done at low speed or real-time for accuracy and bit preservation.

CD Mastering Prices....$ 30.00 per song

1 to 3 songs......$30.00 per song

4 songs............$120.00

5 to 7 songs.......Add $30.00 extra per song

8 songs............ $240.009

9 to 11 songs......Add $30.00 extra per song

12 songs........... $360.00

13 to 20 songs....Add $15.00 extra per song

Hourly rates are not charged

There's a $25.00 Charge for over night Express mailing of CD Master. Your mixes are transferred directly to a Red Book Audio mastered CD, ready for duplication. Also includes a Sample Rate Conversion form with dithering, track spacing, and fade information.

Helpful Sites:

ASCAP
Performance Right Society

Billboard
Music Trade Organization

BMI
Performance Right Society

College Music Journal
Music Trade Publication

The Industry Yellow Pages
Music Industry Contact Directory

Trademark Office
U.S. Patent and Trademark Office

Copyright Office
U.S. Copyright Office

The Harry Fox Agency
Song Licensing Organization

NARAS
Record Industry
Trade Organization

NMPA
Publishing Industry
Trade Organization

SESAC
Performance Right Society