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GuardianOne
10-03-2006, 01:02 AM
Had a conversation with Ancient Arts yesterday and i just thought of creating a thread like this (not sure if it should be here or in Know The Ledge)

Anyway its hands free, anyone can post on this page...

PS: Sometimes creating music can get boring so to say, but music isn't just about creating instrumentals (there's writing lyrics, improving skills) an such so....

GuardianOne
10-03-2006, 01:06 AM
Till this day i have not worked with .wav formats...

The short answer is "no." You cannot improve a sound's quality by raising its sample rate or bit depth. For example, if you have 11Khz WAV files, raising the sample rate for these files will only allow you to apply audio processes (effects) to the files at that level of quality. The audio data that you're working with is already of a low quality, and this can't be "turned back" by changing the file's attributes. Once the file has been recorded at or converted to a lower rate or bit depth, it will possess that lower quality from then on.

GuardianOne
10-03-2006, 01:07 AM
Just thought of sharing the info usually get when buying software, there is always these feedbacks an stuff....

GuardianOne
10-03-2006, 01:10 AM
Setting up a home studio can be expensive and difficult. However, to many musicians, it is well worth the expense. There are plenty of books available with advice on buying and wiring in equipment. The focus of this article is about constructing, soundproofing, and tuning your home studio.
A major concept in studio acoustics is to eliminate parallel hard surfaces that reflect sound waves. It's also important to stop sound from entering or exiting the room through holes. Also, forcing sound to pass through multiple surfaces that are insulated from one another will dramatically decrease the sound that escapes the room. For example, two walls with an air gap or sound insulation between them will stop more sound than one thick wall.

When selecting a room in which to build your studio, think about how any sound leakage will affect the rooms near it. Putting your studio next to the baby's room (or any bedroom) is probably not a good idea. A loft in a detached garage would be perfect. A room in the corner of the basement would work well also. The best suited rooms have non-parallel walls (so the sound won't reverberate between them). If you can't use non-parallel walls, consider placing bookcases diagonally across the corners of the room to break up the wall surface.

Soundproofing takes a bit of work, and for the best results you can expect to spend a fair amount for this step. The best way is to build a room with double drywall walls and insulation in between, then leave 3 to 4 inches and build another double drywall wall that is fully insulated as well. The double wall serves as a bass trap and will absorb the majority the sound before it leaves your room. Do the same with the ceiling and the floor if you can.

The easiest solution for the floor is to put the studio in the basement, but if you can't do that then you'll have to insulate the floor. A double layer of sub-flooring with sound insulation in between can be covered with a carpet. Make sure to stagger the joins on the sub-floor so that there is no complete gap for the sound to penetrate. Don't lay the sub-floor all the way to the walls because then the walls and floor will bridge and conduct sound between them. Leave about ¼ to ½ an inch around the sub-floor and fill it in with caulk.

In addition to all the above sound proofing, consider installing baffles in your heating and air conditioning ducts to prevent sound from exploiting these handy connections to other rooms.

Once your studio is built, you will need to tune the room to improve sound quality inside it. Blankets and quilts hung on the walls can be a cheap way of solving the sound reflection problems. Another inexpensive solution is to glue styrofoam panels to the wall and then glue styrofoam egg containers to that. There are also plenty of internet stores that sell sound-absorbing or diffusing materials. Look for panels that have adhesive backing to simplify the job.

The final step is to find your new studio's "sweet spot" or optimal listening point. The speakers should be located equal distances from your ears, forming an isosceles triangle with the two speakers and your head. Don't put the speakers up against the wall or you'll get a bassier sound. Move the speakers around and try a bunch of different configurations.
Now that you've built and tuned your home studio, you should get great sound within it and hardly any sound outside of it.

GuardianOne
10-03-2006, 01:16 AM
What is a sound wave?

When you hear a sound, you are really sensing changes in the air pressure around your ear drum. (You can think of air pressure as how densely packed together air molecules are.) However, we do not hear air pressure changes caused by the weather. Instead, we hear air pressure differences which vary rapidly over time.

What is timbre?

The characteristic sound of a waveform is called its timbre. Timbre, also referred to as tone color, is said to be rich or full when there are many different frequencies in a sound. Most people would consider a sound from a sine wave to be dull since it only has one frequency.

A sound's different frequencies, each with varying amplitudes, are also referred to as the spectral content of a waveform. The spectral content, which you might say is the more scientific term for timbre, usually varies over time. Otherwise, the sound remains static and is again considered dull. The spectral characteristic of a waveform over time is the signature of a tone which allows you to refer to it as string-like or horn-like.

What is sampling?

When you put a microphone up in the air and scream, the microphone converts the changes in sound pressure into changes in electrical voltage. If you were to graph the changing voltage inside a microphone cord, it would look exactly like the graph of the air pressure going up and down. This is what is called an analog signal.

Up until not too long ago, sound was always recorded as an analog signal in magnetic tape or vinyl grooves. One of the problems with storing a signal in this form is that it is hard to accurately record the signal without adding noise, and when you make copies of your recording, you again have to convert to an electrical analog signal and re-record it. Listen to a third generation cassette recording, and you'll know what we're talking about. Also, editing with tape is a not an easy task since you must always be fast forwarding or rewinding to a section, splicing, etc. This is called linear editing.

With recent advances in computer technology, it has become efficient and economical to record sound waves by digital sampling. In digital sampling, the voltage analog of the sound wave is divided and stored as numbers representing the amplitude of the wave over very small segments of time.

What is a sampling rate?

The number of times the waveform is sampled per second is the sampling rate. As you can imagine, with higher sampling rates you store more information about the sound's changing amplitude. This gives you more fidelity. As a matter of fact, it is theoretically impossible to accurately record frequencies above one-half of the sampling rate. This threshold frequency is called the Nyquist frequency, and it should be considered when selecting a sampling rate. Frequencies higher than the Nyquist frequency show up as alias noise. The downside to very high sampling rates is that since each sample takes up space in memory, higher sampling rates will fill up your hard drive faster than lower sampling rates.

What is quantization noise?

Another factor determining the recording fidelity is how you decide to represent each sample. For example, if you were to represent the amplitude of each sample as a number from one to four, you would have to be rounding all the time to the closest value. This rounding error is called quantization noise. With 8-bit binary numbers, up to 256 different sample values are available. With 16-bit numbers, 65,536 different values are possible. Even though 16-bit samples take up twice as much space as 8-bit samples, it is recommended that when at all possible you use 16-bit samples.

Graveyard shift
10-09-2006, 04:10 PM
Educate these motherfuckers

GuardianOne
04-17-2007, 07:20 AM
There is no right or wrong answer to this, just differences of approach. If the vocal is all-important in a song, then you could start with just the lead vocal in your speakers. Get this sounding exactly as you want it with EQ, compression and reverberation. Then start building the instruments around it. When the vocal is adequately supported, then your mix is done. You will probably end up with the vocal very prominent, with the instruments in the background.
Alternatively you can start with the instruments and get a good overall sound, then add the vocal. At first it won't fit in well, because your instrumental mix is taking up all the 'space'. But if you adjust the EQ so that the instruments are lowered in level at around the frequencies where the vocal is strong, you will be able to get them to fit together well.
In this case, the vocal will end up on a similar level to the main instruments. This is sometimes called 'treating the vocal like an instrument', instead of putting it right out front.
The third possibility is that you mix the vocal and main instruments at the same time, and when you have a good mix of the important features, you start adding the rest of the instruments.

ClymaX
04-17-2007, 07:26 AM
God bless u guardian!

GuardianOne
04-17-2007, 07:33 AM
If you record a track and mix it on a digital audio workstation, you shouldn't expect your work to sound the same when you play it back on another similar system. What can you do to ensure your mix stays the same?


It has been a known problem for some time that a multitrack recording and mix created on one digital audio workstation may not be playable on another - even of the same make and type. Or it might be playable to an extent, but the mix doesn't sound the same, and the timing is suspect.

This was proven recently by a simple eight track recording made on a Pro Tools HD system. The engineer had used a variety of plug-ins, nothing exotic, and in one instance had found the need to apply a time correction to account for the processing delay of a chain of two plug-ins in a stereo aux.
Since the recording had turned out particularly well, I took a copy to listen to on my own Pro Tools system. My system isn't the more recent HD system but its predecessor, the Pro Tools 24 MIXplus system. Despite the fact that manufacturers like to put forward the idea that their previous models are dinosaurs compared to their new range, Pro Tools 24 MIXplus is an excellent system and will reliably turn in great recordings.
Since I had all the same plug-ins, I didn't anticipate any problems in opening the recording. However, when I played it, it didn't sound at all the way I remembered it on the HD system. Never one to trust my ears, I closed the session and went back to the HD system where I bounced the tracks into a stereo mix.
Now when I compared the stereo mix with the multitrack version on my MIXplus system, I could clearly hear the difference. In particular some of the plug-in effects had changed massively. Plainly the parameter values had not translated across correctly.
Now the simple answer to this might be to buy a Pro Tools HD system. But that's just brushing the problem under the carpet. Because from this experience I can extrapolate that whatever I create on an HD system runs the risk of not playing correctly on a future Pro Tools system. And this uncertainty factor applies to any DAW.
Best practice therefore is to print every track including its effects into a single file. You can keep all of the original data of course, but this way you have all of the tracks of your multitrack recording in permanent form that should play correctly on any system. In fact it's probably a good idea to print copies without effects too. Each track should be continuous and not divided into regions. This will remove any potential errors in timing and you will have the freedom to apply any new options in plug-in technology at any time in the future.
My feeling is that this is a serious problem that most people do, as I said, brush under the carpet. Eventually however it is highly likely to sneak out and bite you on the ankle.
You could of course take the Abbey Road approach and keep every multitrack recorder you have ever used, so as long as you can handle the maintenance, you will be guaranteed to be able to play your tracks back at any time. Apart from that, printing your tracks into continuous files is definitely the best option.

GuardianOne
04-17-2007, 07:38 AM
there are some things you will get from a commercial studio that you won't get at home, no matter how good your set up. In no particular order...

Being in a professional environment, mixing with people in the business. This cannot be underestimated. The atmosphere of professionalism will raise your game by an order of magnitude.
Related to the above, rubbing shoulders with other creative people. The route to success is often through collaboration - where better to meet people to collaborate with?
Acoustics. The acoustics of the control room are significant in producing a mix that sounds good, and sounds good no matter what it is played on.
You don't always have to record in the same studio. I used to have a studio at home. The day after day sameness gave me a bad case of cabin fever. In the days before home studios were commonplace, it was a known fact that when anyone achieved enough success to build a studio at home, that was the end of their success.
Hire a good engineer. Will a good engineer want to come to your home studio? No. Does a good engineer make a difference? Yes - would you have a trained guitar tech fix your broken guitar, or would you do it yourself?
You can go home. If you record at home there is nowhere to get away from your work. Believe me, you need to.
Work with 'dodgy' people. Many of the great talents of rock 'n' roll, hip hop and every other musical style have been known to have certain 'habits'. You want these people on your recording, but do you want these people in your home?
Paying $3000 a day certainly concentrates your mind!

Hellspawn
04-17-2007, 08:39 AM
there are some things you will get from a commercial studio that you won't get at home, no matter how good your set up. In no particular order...

Being in a professional environment, mixing with people in the business. This cannot be underestimated. The atmosphere of professionalism will raise your game by an order of magnitude.
Related to the above, rubbing shoulders with other creative people. The route to success is often through collaboration - where better to meet people to collaborate with?
Acoustics. The acoustics of the control room are significant in producing a mix that sounds good, and sounds good no matter what it is played on.
You don't always have to record in the same studio. I used to have a studio at home. The day after day sameness gave me a bad case of cabin fever. In the days before home studios were commonplace, it was a known fact that when anyone achieved enough success to build a studio at home, that was the end of their success.
Hire a good engineer. Will a good engineer want to come to your home studio? No. Does a good engineer make a difference? Yes - would you have a trained guitar tech fix your broken guitar, or would you do it yourself?
You can go home. If you record at home there is nowhere to get away from your work. Believe me, you need to.
Work with 'dodgy' people. Many of the great talents of rock 'n' roll, hip hop and every other musical style have been known to have certain 'habits'. You want these people on your recording, but do you want these people in your home?
Paying $3000 a day certainly concentrates your mind!

yeah fo sho :)

anyway, how can you improve your record when you get a beat from someone via the net so all the instruments are in the same track and you cant work with it?

and...is it better if you feel like your voice is louder than the beat or does the beat should be a little louder?

thanx

MaXiMus Da MaNtis
04-17-2007, 06:18 PM
to anserw that hellspawn ....they have to send it in layered tracks...in a high bitrates .....the real best way is to get it via mail in a cd in broken down tracks layered

GuardianOne
04-24-2007, 09:22 AM
As much as you might like one, you may not actually need a Pro Tools HD digital audio workstation in your studio...

Question: "A friend of mine called me the other day and started to tell me about a new system for audio recording called High Definition recording (HD Recording). He then continued to tell me that if we all don't up grade to a High Def. system we will no longer be able to compete with major studios. He also says this upgrade cost in the upwards of $15,000. Is this true? Or is this just a bunch of B.S.?"
This is an interesting question. If you produce your own music in your own studio then you can use any equipment you like. Having a Digidesign Pro Tools HD system (which I presume you mean) would be nice. But you could record on a 30 year old analog machine if you wanted to.
But if you have a commercial facility that people are going to hire, then you need to have the kind of equipment they expect. Pro Tools HD is pretty much standard in the industry, so yes you do need it.
This is the same kind of logic that made SSL so popular in the 1980s and 1990s. Once a few people had recorded hit records using SSL consoles, other studios started buying them. Then at some point there was a 'critical mass' where to work as an engineer, you had to have SSL experience. And then since all the successful engineers had SSL experience, more studios had to become SSL-equipped.
But if you are producing, say, music to picture in your own home recording studio, you can use any equipment you like as you have no-one to please but yourself.

GuardianOne
04-24-2007, 09:25 AM
As much as you might like one, you may not actually need a Pro Tools HD digital audio workstation in your studio...

Question: "A friend of mine called me the other day and started to tell me about a new system for audio recording called High Definition recording (HD Recording). He then continued to tell me that if we all don't up grade to a High Def. system we will no longer be able to compete with major studios. He also says this upgrade cost in the upwards of $15,000. Is this true? Or is this just a bunch of B.S.?"
This is an interesting question. If you produce your own music in your own studio then you can use any equipment you like. Having a Digidesign Pro Tools HD system (which I presume you mean) would be nice. But you could record on a 30 year old analog machine if you wanted to.
But if you have a commercial facility that people are going to hire, then you need to have the kind of equipment they expect. Pro Tools HD is pretty much standard in the industry, so yes you do need it.
This is the same kind of logic that made SSL so popular in the 1980s and 1990s. Once a few people had recorded hit records using SSL consoles, other studios started buying them. Then at some point there was a 'critical mass' where to work as an engineer, you had to have SSL experience. And then since all the successful engineers had SSL experience, more studios had to become SSL-equipped.
But if you are producing, say, music to picture in your own home recording studio, you can use any equipment you like as you have no-one to please but yourself.

"As to sound quality, 24-bit is a clear winner over 16-bit. You always want to work at a better resolution than the end product, which for most of us is still CD. Sampling at 88.2 kHz is a subtle improvement over 44.1 kHz. In my opinion, sampling at 176.4 kHz is theoretically better than 88.2, but there's no evidence that people can hear the difference, so why bother? Avoid sampling at 48.0, 96.0 or 192.0 kHz if your product will end up on CD (44.1 kHz) because you CAN hear the conversion."

This of course, explains why CD's are 16 bit, 44.1k. All you do when you mix down is dither away 8 bits, so what's the point? If 16 bit is good enough for your final mix (you know, the one everyone actually LISTENS to), it's good enough for tracking. When they start releasing 24 bit CD's, I'll worry about the upgrades. The attitude that you can't have a commercial facility without Pro Tools HD is pure B.S. snobbery. Not everyone that wants to record their music has $90-$150 plus per hour to spend! While it can be argued that it's just a matter of taste, I simply don't like the sound of Pro Tools. It's brittle and lifeless. Its the same as saying if I don't have $20,000 to spend on a Pro Tools system, I'm not allowed to have a studio! I'm sorry, but that is not based in reality.

GuardianOne
04-24-2007, 09:30 AM
Hm... a mixer that makes good sound automatically. That's what we are all looking for isn't it..?

“What are motorized faders for in a mixer? Do they mix and make good sound automatically?”
Well yes, there is such a thing as a mixer that makes good sound automatically. That would be one of the very few 'first-call' mix engineers who are most in demand and turn out hit records day after day. Just send them your multitrack and get back the perfect mix. Oh, don't forget the $2000 a track or more they can charge.
For the rest of us however, the idea of a piece of hardware or software that could analyze a track and come up with the perfect mix is intriguing.
If such a device or software existed, you would probably be able to select 'In the style of...' from a simple drop-down menu.
Sadly, the day when that will be possible is far off. And if everyone had it, then the competition would be so much tougher.
No, the purpose of motorized faders is as part of the mix automation system. The engineer decides how the faders should be set, which should be moved, at what times in the song and by how much. The automation system remembers every move so that the engineer is free to refine and add as much as he or she likes. In the old days before automation the engineer had to remember every move!
Some systems play back the mix using 'invisible fingers' on motorized faders. Some don't, and on some you can switch the motors off. Whether you prefer to see the faders move or not depends on the individual engineer.
Fully automated mixing though... A fascinating thought.

GuardianOne
04-24-2007, 09:34 AM
Is it true that studio microphones are no good for live sound, and vice versa? Maybe the manufacturers want us to buy two sets of mics...


“What makes some microphones good for studio, and others good for live work?”
Well of course the first issue would be durability. It applies primarily to mics that are handled by live performers. Mics on stands shouldn't be subject to any more stress than they would be in the studio. But if a mic stops working in the studio, the engineer can easily replace it with another. In the middle of a concert the problem is rather more significant.
Particularly for vocal mics, there is a huge difference in what is typically used in the studio and what is typically used live.
In the studio, an engineer might choose a large-diaphragm capacitor (condenser) microphone, preferably a vacuum tube model. A vintage Neumann U47 would be perfect. And since it is a vintage model, it is fragile and valuable.
So there are already several reasons why you wouldn't want to use a mic like this in a live show.
Firstly, although modern microphones are generally pretty robust, you wouldn't expect an old microphone to stand up to rough and tumble. And you wouldn't want a valuable vintage microphone to 'disappear', as mics occasionally do on the road.
Secondly, although the sound quality of a large-diaphragm tube mic is excellent for vocals, you would generally use it at a distance of at least a few centimeters, perhaps with a pop shield.
You can't use a pop shield in live sound – the audience wants to see the singer's face! And keeping the mic at a suitable distance throws up other problems. For one, it's difficult to keep the distance consistent. For another, it is important in live sound for the singer to be as close to the mic as possible to prevent feedback.
So a vocal mic for live work is designed to be used very close up, without a pop shield. This inevitably compromises the sound quality compared to the studio mic, but overall this compromise is necessary.
In summary, a vocal mic for the studio should have a great sound. Nothing else matters. A vocal mic for live sound should be robust and capable of being used very close to the mouth without a pop shield.

GuardianOne
04-24-2007, 09:36 AM
Sometimes the snare is loud, sometimes it's not so loud. How can you use a compressor to even out the level?

“If I have a track with snare recorded on it and it's sitting comfortably say around -6 dB but with stray peaks every now and again of around -2 dB. Where would be a good place to set my threshold to attempt to contain them. Starting with say a 4:1 ratio. This would be easier for me if my compressor had a gain reduction meter on it for more visual aid. I just can't seem to get this question straight in my mind.”
One of the skills of a good drummer is consistency. But you don't always get to work with a good drummer. It has been known for bands to hire drummers on the basis that they have a van! Drumming skills are sometimes secondary.
So suppose you are recording a band where the drummer isn't consistent. How can you use a compressor to even things out?
Well one answer is that you don't use a compressor!
Think about it. The problem is that the level is inconsistent. Yes, a compressor can reduce the difference between louds and quiets, but a compressor also changes the sound quality. What comes out doesn't sound exactly like what went in. And if only the loud hits are compressed, then all you have done is created another inconsistency.
No, you have to go the extra mile (1.6 kilometers) to solve this. And it shouldn't be too much trouble if you want the best mix possible.
The way to do it is to use the power and convenience of the digital audio workstation to adjust the level of every strike individually, by hand. The thought of adjusting a hundred drum hits over the course of a song might sound daunting. But I personally wouldn't hesitate if I thought the song would benefit overall. And anyway, the level of most of them might be perfectly OK.
Clearly there will still be some difference in sound quality between the hits that were originally loud and those that were originally quiet, due to the way the drum responds. But the difference will be less than if a compressor had been used.
Now let's start on the kick drum...

GuardianOne
05-03-2007, 09:37 AM
There are two alternatives here. One is to use headphones, as the questioner asks. The other is to listen to the PA system and hear what the audience hears.
The answer to this question might seem like a no-brainer. Clearly the engineer has to optimize the sound for the audience. And he is best placed to do that if he listens to the PA.
One the other hand, headphones can be a useful additional tool in several ways.
Sometimes in a live PA or theater setting, the sound doesn't seem quite right somehow, but you can't really put your finger on why. Listening on headphones can give the extra clarity that is necessary to pinpoint the problem and set about solving it.
Another use of headphones is to PFL (pre-fade listen) individual instruments. You might do this if you suspected their was a problem with a certain instrument - a blown speaker in a guitar cabinet for example. Or you might do this as an aid in optimizing the sounds of individual instruments.
The third use of headphones is to detect feedback. If a channel is on the point of feedback, it is far easier to hear this on headphones (using PFL). You can hear the 'ringing' that precedes feedback before the audience is aware of it.
One important point is that your headphones should be very well sealed against outside sound, otherwise the PA will either drown out what you are trying to hear, or you will be tempted to turn up the headphone volume to dangerous levels.

GuardianOne
05-03-2007, 09:42 AM
Who would have thought that the mistress of King Louis the 14th of France could have had such an impact on sound engineering?

"I just wanna ask how can I minimize the popping sound on a microphone? I'm using a lavalier mic."



Yes, apparently the Duchesse de la Vallière, mistress of King Louis the 14th of France, had a liking for wearing a ruby dangling from her necklace. This form of jewelry became known as a lavaliere necklace, and in turn a microphone mounted in the same way became a lavalier microphone. Somewhere in history the final 'e' was dropped.
Microphones are no longer dangled from cords around the neck. Instead, miniature microphones are clipped to the clothing in a convenient place.
Microphones clipped on in this position are not in the direct line of fire of the breath, so in theory they should be resistant to popping.
In practice however, the occasional pop still does happen. So what is the solution?
One would be to fit the tiny pop shield that often comes among the accessories supplied with these microphones. But the solution is even easier than that...
Clip the microphone to the clothing upside down, so that the diaphragm points away from the mouth.
This might seem counter-intuitive, but most miniature microphones are omnidirectional and it doesn't matter which way you point them.
Another side benefit is that the cable often lies more neatly too.
A simple answer to a simple problem, and it works.
Oh, and thanks Duchess!

tekunique
05-03-2007, 08:16 PM
ill thread.... dont sleep yall....

Sun Tzu, Tha Soul Controllah
05-03-2007, 08:52 PM
http://www.wutang-corp.com/forum/showthread.php?t=31581

tekunique
05-03-2007, 09:02 PM
http://www.wutang-corp.com/forum/showthread.php?t=31581

tru..mods should join it to that then

GuardianOne
05-04-2007, 01:34 AM
So you have a cheap mic with a jack connector. If you replace the jack with an XLR, do you have an expensive mic?


"There are lots of cheap microphones with 1/4" plugs. Is it possible to replace the plug with an XLR and connect it to a snake cable? Or can I just connect these cheap microphones to a direct box and so convert the output to XLR?"
The (imaginary) scenario is this... a band is building up their own PA system from scratch and there's not a lot of money to go around. They have an XLR stage box and multicore cable that takes the signals from the stage to the mixing console, but they have spent most of their money and need to economize on microphones.
I would like to respond to this by saying that you can buy pro mics at amazingly cheap prices, so why bother with microphones that are obviously not professional?
But everyone has to start somewhere and sometimes the budget is severely limited. So let's have a look at this problem.
The microphone may have a fixed cable terminating in a quarter inch jack. If this is the case, then the microphone is almost certainly unbalanced, meaning that it only has one signal-carrying conductor in the cable, plus earth.
In this case, the jack can be replaced with an XLR by connecting the inner conductor to Pin 2 and the earth to both Pin 1 and Pin 3.
That will work, although the mic will be more susceptible to interference than a balanced mic.
If the microphone has a detachable cable, then the mic itself almost certainly has an XLR output. In this case, the jack on the detachable cable can be replaced with an XLR, or you can buy an XLR to XLR cable.
If you replace the jack, then take a look inside the female XLR at the other end to see where the wires go. Wire the male XLR to the other end keeping the pin numbers the same, with respect to the colors of the inner conductors. The earth always goes to Pin 1.
Of course, changing the connector will not improve the sound of the microphone in any way.
Connecting the mic to a DI box probably won't work too well. If it is a transformer DI box, then it might work. But replacing the connector is the better option.
Once you have got your mics working with their new XLR connectors, it's time to start saving for better mics!

One thing not mentioned is that many cheap mics with 1/4 plugs are hi-Z, and all xlr input jacks are lo-Z. If all you do is wire a xlr plug on to the mic cable and plug it in to a xlr input, you will most likely have miss-matched impedances. The loss in signal power due to the miss-match will force you to turn up the input gain on your mixing console, which will bring up the noise floor and degrade the s/n ratio. There is a simple device called a matching transformer that converts 1/4 to xlr and has a built in transformer to convert hi-Z to lo-Z. This will solve the problem without loss of signal power or need to re-wire.

GuardianOne
05-04-2007, 01:38 AM
Recordings that feature expensive pro studio reverbs are sometimes enviably better than can be achieved in the home studio. Are the manufacturers holding something back from us?

Not meant to travel down the left fork of the reverbing road here, but maybe we're all comparing apples to oranges (so to speak)? There exists an old adage that reverb is what seperates the mixing men from the boys, but talking about acheiving a Celine Dion level of reverb perfection without prefacing that conversation with questions about room, mic, mic placement, compression, EQ, etc etc, is like conversing on a caboose pulling a train. It just doesn't necessarily work because it's a backwards beginning conversation.

Consider that an artist of that stature records in the very best studios on the very best equipment with some of the most highly sought after pros in the industry. Perhaps aspiring to that goal is more than can be reached right now?

Still, some things you can do to help move in the direction of acheiving that mix goal is to find out what type of room Celine Dion records her vocals in (size, shape, materials type (oak, mahogany, granite / baffles, etc)), what type of mic she sings to, what is the board / signal path her producer uses, how is her vox signal compressed and chained, and how is she EQ'd. Then also find out the specific reverb settings used on a particular vocal and on what machine. Get that info in play as much as can be allowed, fake the rest as best you can, apply it to your artist and make adjustments accordingly. Your ears and intuition will kick in and do the rest.

Consider also that this approach is much less time consuming than fumbling around twisting knobs forever and onward in frustration. Mostly, A-listers are pretty relaxed, helpful folks. They don't have to be stressed or even arrogant because they've already achieved what most others struggle towards. Chances are, Celine Dion's producer or engineer might very well offer some helpful tips with a polite, coaxing email or two.


There's no 'cool' button on the pro units, and there's no 'cool' preset.

GuardianOne
05-15-2007, 09:35 AM
Recording background vocals to the standard required in today's music is difficult and time-consuming. But there is a way the process can be made more efficient...




Listen to modern successful chart recordings, particularly pop and R&B. Notice the complexity of the background vocals (backing vocals) - often twenty or more layers, sometimes with intricate counterpoint.
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The main feature about background vocals is that they should not attract attention, but support the lead vocal. So quite often you can listen to a track and not be aware of them. But they are in fact a key component of modern production style.
But laying down twenty tracks of background vocals is a significant piece of work. It can take literally days. So how can it be made easier, both in terms of arrangement and actual recording?
The trick is not to record background vocals for the entire duration of the song, but to break the song down into musical phrases or lines. Not even verses or choruses - just take four or eight bars at a time.
Now, with the aid of your DAW and versatile recording software, highlight a musical phrase. It helps if it is a precise number of bars, and even better if it is four bars or eight bars because that is how most music is constructed.
Now set this so that it loops, and you can record as you loop. Sometimes the boundaries of the selection will not be at the beginnings of bars, but as long as the selection is four bars or eight bars, the loop will sound smooth and not jerky or discontinuous.
Now you can rehearse the first line of background over and over until your singer finds something that fits in. And you can record it over and over - Pro Tools will record as many takes as you like without stopping. Then when you have a good take, simply hit stop. You can use that take, or select an earlier take that was better.
Now switch to a new track and use the same loop to rehearse and record the second line of background. Repeat this process until the backgrounds are as rich and complex as you like. (Hint - richness and complexity are better achieved if you have more than one singer).
Once you have that, you can move on to the next line.
A great advantage of working in this way is in the timing. Getting multitracked background vocals to line up precisely was always a major problem in the past. But the rehearsal period working on a very short section of the song allows the singer to home in very quickly on the correct timing, almost without thinking.
It will still take you a day to record backgrounds for an R&B track, but it will be a day well spent, and with a good singer the result can be very professional indeed.

GuardianOne
05-15-2007, 09:42 AM
The panpot of a mixing console has to blend the signal from left to right in the correct proportions. But do panpots always obey the letter of the law?


In early stereo mixing consoles, the choice of placement of a mono signal in the stereo 'sound stage' was simple - left, center or right. There were no other options. But then some clever person invented the panoramic potentiometer, which today we know as the panpot or pan control. The panpot is a continuously operating device that can position a mono signal anywhere in the stereo sound stage.

But let's look at what happens to the levels as a signal is panned.
Start with the signal panned left. The sound comes only from the left speaker, and nothing comes from the right. Now move the pan towards the center. Since the right speaker is now contributing more and more to the level in the room, the signal sent to the left speaker has to be lowered. In fact, when a signal is panned center, it needs to be 6 dB lower in level than when it is panned hard left or hard right. If the panpot is designed so this happens, then the sum of the levels in the left and right channels will remain constant whether it is panned left, center or right..
In the early days of stereo, when mono compatibility was important, then panpots did indeed have a 6 dB drop in level at the center, which meant that a signal could be panned all the way from left to right, and someone listening in mono would hear no change in level.
But that doesn't account for stereo listening. It should work in theory - sound pressure adds in exactly the same way as voltage. So the pan control with a 6 dB drop in the center ought to work fine. You should be able to pan all the way from left to right, and the signal level should be subjectively the same all the way.
But that doesn't account for the fact that most listeners enjoy their sound from a point in the diffuse field, where the sound has had chance to bounce around the walls of the room many times before reaching the listener's ears. This is totally different to the near-field monitoring technique that is currently popular.
If measurements are taken from a point in the diffuse field, then it will become apparent that it is better to have only a 3 dB drop in the center of the panpot's travel. Many panpots in real world designs compromise on a 4.5 dB center drop.
Added to this is the question about what happens in positions other than hard left, center and hard right. This will be determined by the 'law' of the resistive tracks of the panpot (or the digital algorithm, where the same factors apply), which has a closer relationship to the laws of physics than the legal system. It has been known science since the 1930's that the law should follow the sine and cosine functions for the two channels respectively. However, many analog consoles 'bodge' this instead of using the more expensive component necessary.
To be honest, most of this doesn't matter that much in music production. But where it does matter is in 5.1 post-production for film and home cinema. Designing a 5.1 channel panpot is a far more complex affair than stereo, and similar considerations regarding levels apply. However, a stereo panpot simply has to go from left to right. A 5.1 panpot might have to pan from left to left surround, center to right surround, perhaps even into the low frequency effects channel. And all the channels will have to contribute to the acoustic signal in the correct proportion.
A fascinating field of research, and real heavy-duty sound engineering.

GuardianOne
11-01-2007, 04:07 AM
Back to the basics...

GuardianOne
11-01-2007, 05:01 AM
Firstly, there's no such thing as a stupid question. You have to be careful about the answers sometimes though ;-)

This question is an oddly common one, perhaps because equipment manufacturers assume that everyone knows the answer already, and therefore knows what equipment to buy.
The answer here is in your choice of audio interface.
Many audio interfaces, such as the Digidesign Mbox 2 (http://www.zzounds.com/item--DGDMB2/a--3820), have two analog inputs. That means you can record up to two signals at the same time, from two microphones for instance.

[Actually you can use the digital inputs of the Mbox 2 in addition, but let's not over-complicate things.]

You could record an entire band, if you had a mixing console.
You could connect every microphone and direct inject to the mixing console, providing it has enough channels, and mix the band into stereo as they perform. You can record this to your computer through the two inputs of your audio interface.
This will work. However...

The band has to be able to perform very well. You can retake sections that didn't go too well, and then edit. But the band will have to be pretty good for this to work.
Your mixing skills should be very good. It's not impossible - live sound engineers mix bands like this all the time. But unless you are really good at mixing, the end result may sound 'rough around the edges'.To achieve what is expected of a modern recording, you need to be able to put each instrument on its own track. The advantages are...

You can fix any mistakes individually. For example if the drummer played a couple of beats out of time.
You can take your time over mixing.
You can make overdubs - for instance the singer can also do background harmonies. And to be able to do this you need...

As many microphone preamplifiers as the number of microphones you will be using.
An audio interface with multiple inputs - as many inputs as the number of tracks you want to record at the same time, including both microphones and line input sources. It is probably more convenient if the microphone preamplifiers and audio interface are combined into the same unit, as in the Mark of the Unicorn (MOTU) 8pre (http://www.zzounds.com/item--MTU8PRE/a--3820), which has eight inputs. You can use two to give you sixteen inputs, which is just enough for a typical band. Make sure that your recording software is compatible though. That is always an important check to make.

In summary therefore...

To record a band playing all at the same time, you need a minimum typically of sixteen microphone preamplifiers and sixteen channels of audio interface. You can achieve this with, for example, the MOTU 8pre (http://www.zzounds.com/item--MTU8PRE/a--3820). Other interfaces are also available.

GuardianOne
11-01-2007, 05:07 AM
The key issue here is would a singer sound the same performing right in front of you as they do on their recordings?
For most popular music recordings, the answer is a clear and resounding NO!
If you wanted to capture the natural sound of the voice, firstly you would use a singer who didn't need studio tricks. There are some around, you know.

Next, good acoustics. Generally a large room that has had acoustic treatment to control the reverberation would be a good place to start. It's very difficult to get a good sound in the average living room, so I don't count using good natural acoustics as cheating.

Next, an accurate microphone. Now this is a bit of a problem because many mics are sold on the 'character' of their sound. And if a mic's sound has character, then it isn't accurate.
Having said that, a small-diaphragm capacitor from a manufacturer such as Sennheiser, Schoeps or DPA would be accurate enough not to be considered cheating. Don't go any closer than about a meter or so, or the accuracy will be degraded. (The room has to be quite large to allow placement at this distance without too much reverberation.)
So go ahead and record - you will capture a sound that is as close to the original as you are likely to get. No enhancements, just reality.
But as we know, 'enhancements' are popular in many of life's activities. And so it is in recording.

So let's go for a mild one - use a large-diaphragm capacitor mic, and put it close to the mouth.
Suddenly the sound is much bigger, and more 'present'. It might be different from real life, but we often prefer it.
Still, I wouldn't say this is cheating. It's enhancement. If it sounds better, then it is simply better. Why not?
You can go further and use a vacuum tube microphone, and perhaps a tube preamp. The sound will get thicker and stronger. More pleasing to the ear. Still not cheating, really.

What about a bit of artificial reverberation? Not the type that's realistic, the type that just sounds good - like the old analog echo plates and Lexicon digital reverbs.
Rich and lush. Much better than the natural sound. Still not exactly cheating. Enhancement to the max, I would say.
But what if the tuning of the singer is a bit doubtful at times?
Well, do as many retakes as necessary so he or she can hit the right notes. If they could do it once, then it's not cheating. Just a careful selection from the available material.
Or, you could automate the tuning correction process with software such as Auto-Tune.
Oops... we just seem to have crossed a line there.
Perhaps the artist can't sing in tune. But with Auto-Tune, they can.

That's definitely cheating!

Or maybe they could have sung in tune, given enough takes, and this just makes the process faster. If you think about it enough, you can probably justify it to yourself.
There are further steps that can be taken towards vocal 'improvements', but I think that's enough for now.

GuardianOne
11-01-2007, 05:13 AM
Are you worried about your preamp? How can you know for sure whether it's giving you the sound quality you need?


First a simple truth - as long as your preamp is of professional quality and working properly, there is nothing about it that will prevent you making a good recording.

Preamps don't matter anywhere near as much as the currently popular myth suggests. Your musicians are important, their instruments are important, the acoustics of the room are important, your microphone positioning and microphone selection are important, your skills and artistry are important. The microphone preamp comes after all of that in order of importance.
But still, you might have concerns about your preamp, and one common concern is noise.
The correct way to set preamp gain is to increase it to whatever value that is necessary to achieve a good strong reading on the meters of your recording system, without clipping of course.
But you may find that when you do this, there is a lot of background noise.
This may be the acoustic background noise that is present in the room. Raising the gain of the preamp does raise this, but only in proportion to the signal you want to record. So the signal-to-noise ratio from this source of noise stays constant.
But it might also be that the noise generated by the preamp itself increases. This should not happen. In fact, many preamp manufacturers quote their noise levels measured at maximum gain, because this is where the signal-to-noise ratio is greatest.
So here is a simple test...

Set up a microphone up in a quiet room and set the gain on your preamp to maximum. Make a recording (A) of the background noise of the room.
Lower the gain of the preamp by 20 dB. Make another recording (B).
Normalize both recordings in your digital audio workstation so that they are the same level.Now, there are three possibilities...

The noise levels are about the same. In this case you can stop worrying and get back to recording!
Recording B is noisier than Recording A. This is normal. As above, stop worrying.
Recording A is noisier than Recording B. This is a distinct warning sign that something is wrong with your preamp. It simply should not happen. Time to get a better one!Of course, we have only discussed noise at high gain settings. But at least in this significant parameter it is possible to see clearly whether your preamp is up to the job.

GuardianOne
11-01-2007, 05:16 AM
There are few 'rules' in recording. But sometimes you just have to know where to set the pan control.


This might seem like an obvious question, but I'm a firm believer in having the obvious well under control before progressing to more advanced issues.
So if you pan to the left, the sound should come out of your left speaker. Pan to the right and it comes out of the right speaker. If it doesn't, swap your speaker cables round.
OK that's basic. But there are other basics too. One common error in recording is swapping the channels unintentionally.
The effects of this could be...

In an orchestral recording, the violins are on the right rather than the left. (If the conductor has chosen to put the second violin section on the right, then the first violin section, which generally plays all the tunes, should be on the left.)
In a film or TV soundtrack, the character on the left of the screen speaks and their voice comes out of the speaker on the right.Clearly either of these would be major errors, and you probably wouldn't work again in this town! It's an easy mistake to make though, so you have to be sure.
Sometimes where you pan is open to interpretation.
Take the example of a recording of a grand piano...
You might say that the low notes should come mostly from the left speaker, and the high notes mostly from the right. That's how the piano keyboard is laid out.
But that's not the way the audience hears it...
The grand piano is always positioned on stage with the keyboard on the left. So the low-pitched strings, which are longer, extend further to the right than the high pitched strings.
So by this logic, the high notes should mostly come from the left and the low notes mostly from the right.
It's up to you! There's no right or wrong. It's either the pianist's perspective or the audience's perspective. Either is equally valid.

If anyone has any other interesting examples of right/wrong/up-to-you panning, we would love to hear.