music production

Re: music production

now i have not so much time, but i will help to you with mastering, in mastering are some rules which you cant ignore
i use for mastering cubase... i think cubase have the best sound

you can listen my shit instrumentals there : www.myspace.com/trbeme

but mostly its old some new tracks i not put there

 

Re: music production

so first obout mastering.. when you master then do it in this sequence:
1. EQ
2. Reverb
3. Kompressor
4. Dynamic efects
5. Stereo
6. Maximizer/ Limiter
7. Spectrum

iam not profi but this should be ok

 

Re: music production

i kind of did it that way, but i mixed a few up lol i alway did put in a compressor before reverb and i totaly gave up on the spectrum thingy, cause thaatsss aloot of shitty work.
But i will try again tomorrow and see how it works.

I like your tracks btw al from scratch or do you use drum loops, or do you sample songs?

 

Re: music production

a have samples, never use loops...i make my loops and this i copy into song and then hypersonic or some plugins to make bass, brass, strings,pianos ......i have 10 giga samples btw

obout the mastering.... what soundkart do you have?
i can send to you some ebooks obout mastering

 

Re: music production

Wooow i aint got that much samples lol, and im very interested in the ebooks iy you could give me a link or something that would be great!

I have a shitty Realtek HD Audio 5.1 nothing special really, but i think its my speakers that really suck, see they (try) to pretty up the sound and that really sucks when atempt to master a song When i take my music downstairs to the stereo its almost sounds completely different from wath i just heared in my room and that its just fucking confusing lol XD

 

Re: music production

ouuuuuuu one from the most important parts is soundkart............ heh i have lot of samples because you cant use everytime the same obout the ebooks i havent got link, but we will do it

realtek make shit soundkarts..... mostly onboard its just for listening and manytimes just for "fun" i have two soundkarts x)

 

Re: music production

Ok, so im pretty much fucked When i buy a new PC i will pay xtra attention to the soundcard, Audigy is a great one right?

 

Re: music production

ehm, if you want profi soundkart....... you must be ready to pay 100 €, check this http://de.m-audio.com/index.php?do=prod ... interfaces

 

Re: music production

20 tips of mastering

There's a world of difference between what happens in a professional mastering suite and what the average project studio owner can do at home. But as more computer-based mastering tools become available it's quite possible to achieve very impressive results with relatively inexpensive equipment. Certainly there's a lot more to mastering than simply compressing everything, though compression can play an important role. The most crucial tool is the ear of the person doing the job, because successful mastering is all about treating every project individually. There's no standard blanket treatment that you can apply to everything to make it sound more 'produced'.

Every mastering engineer has preferences regarding the best tools for the job, but if you're just getting started I'd recommend a good parametric equaliser, a nice compressor/limiter, and perhaps an enhancer, such as an Aphex Exciter or an SPL Vitalizer. You also need an accurate monitoring environment with speakers that have a reasonable bass extension, and some form of computer editor that can handle stereo files. The latter should ideally have digital inputs and outputs, though if you're using an external analogue processor you'll probably be going into the computer via its analogue inputs, in which case these need to be of good quality too. A professional may want to start off with a 20- or 24-bit master tape or to work from a half-inch analogue master, but in the home studio most recording is done to 16-bit DAT. This shouldn't be a problem for most pop music, providing you proceed carefully.

Most mistakes are due to over-processing, and the old adage 'If it ain't broke, don't fix it' applies perfectly to mastering. Don't feel that you have to process a piece of music just because you can -- you might find that your master sounds worse than the original material. And now for the tips...

1. Where possible, handle fade-out endings in a computer editor, rather than using a master tape that was faded while mixing. Not only does the computer provide more control, it will also fade out any background noise along with the music, so that the songs end in perfect silence.

2. Editing on DAT is very imprecise, so when you beam the material into the computer (digitally, if at all possible) clean up the starts of songs using the Silence function. Use the waveform display to make sure you silence right up to the start of the song without clipping it. As a rule, endings should be faded out rather than silenced, as most instruments end with a natural decay. When the last note or beat has decayed to around 5% of its maximum level, start your fade and make it around a second long. You can also try this if the song already has a fade-out, though you may want a slightly longer fade time. Listen carefully to make sure you aren't shortening any long reverb tails or making an existing fade sound unnatural.

3. Once you've decided on a running order for the tracks on the album, you'll need to match their levels. This doesn't simply mean making everything the same level, as this will make any ballads seem inappropriately loud. The vocals often give you the best idea of how well matched levels are across songs, but ultimately your ears are the best judge. Use the computer's ability to access any part of the album at random to compare the subjective levels of different songs, and pay particular attention to the levels of the songs either side of the one you're working on. It's in the transition between one song and the next that bad level-matching shows up most.

"If an album's tracks were recorded at different times or in different studios, they may not sit well together without further processing."
4. If an album's tracks were recorded at different times or in different studios, they may not sit well together without further processing. The use of a good parametric equaliser (hardware or software) will often improve matters. Listen to the bass end of each song to see how that differs and use the EQ to try to even things out. For example, one song might have all the bass energy bunched up at around 80 or 90Hz while another might have an extended deep bass that goes right down to 40Hz or below. Rolling off the sub-bass and peaking up the 80Hz area slightly may bring the bass end back into focus. Similarly, the track with bunched-up bass could be treated with a gentle 40Hz boost and a little cut at around 120Hz. Every equaliser behaves differently, so there are no universal figures -- you'll need to experiment.

At the mid and high end, use gentle boost between 6 and 15kHz to add air and presence to a mix, while cutting at 1-3kHz to reduce harshness. Boxiness tends to occur between 150 and 400Hz. If you need to add top to a track that doesn't have any, try a harmonic enhancer such as an Aphex Exciter -- high-end EQ boost will simply increase the hiss.

5. To make a track sound louder when it's already peaking close to digital full scale, use a digital limiter such as the excellent Waves L1 plug-in or Logic Audio's built-in Energizer. In most cases you can increase the overall level by 6dB or more before your ears notice that the peaks have been processed. A nice feature of the L1 is that you can effectively limit and normalise in one operation. It's always good practice to normalise the loudest track on an album to peak at around -0.5dB and then balance the others to that track, but if you're using the L1 to do this, make normalising your last process, so that you can use the Waves proprietary noise-shaped dither system to give the best possible dynamic range. Normalising or other level-matching changes should always be the final procedure, as all EQ, dynamics and enhancement involves some degree of level change.
Proper re-dithering at the 16-bit level is also recommended if going direct via a digital output to the production master tape, as it preserves the best dynamic range. Analogue outputs will be
re-dithered by the A-D converter of the recorder.

6. If a mix sounds middly or lacking in definition, the SPL Vitalizer can be very useful (even the very inexpensive Stereo Jack version produces excellent results). This device combines EQ and enhancer principles in a single box, and one characteristic of the Vitalizer process is that the mid-range tends to get cleaned up at the same time as the high end is enhanced and deep bass is added. As with all enhancers, though, be very careful not to over-use it: keep switching the process in and out, to preserve your sense of perspective. The same applies to EQ and dynamics -- check regularly against the untreated version to ensure that you're not making things worse.

7. Have a CD player and reference material on hand to use as a comparison for your work. Not only does this act as a reference for your ears, it also helps to iron out any inaccuracies in your monitoring system.

8. Overall compression can add energy to a mix and even out a performance, but it isn't mandatory. Music needs some light and shade to provide dynamics. Often a compressor will change the apparent balance of a mix slightly, so you may need to use it in combination with EQ. Placing EQ before the compressor results in any boosted frequencies being compressed most, while placing it after the compressor allows you to equalise the compressed sound without affecting the compressor operation. Which is best depends on the material being treated, so try both.

9. A split-band compressor or dynamic equaliser gives more scope for changing the spectral balance of a mix, but these devices take a little practice before you feel you're controlling them and not vice versa!

10. One way to homogenise a mix that doesn't quite gel, or one that sounds too dry, is to add reverb to the entire mix. This has to be done very carefully, as excess reverb can create a washy or cluttered impression, but I find Lexicon's Ambience programs excellent for giving a mix a discreet sense of space and identity. If the reverb is cluttering up the bass sounds, try rolling off the bass from the reverb send.

If you want to add a stereo width enhancing effect to a finished mix, there are two main things to consider: the balance of the mix and the mono compatibility of the end result. Most width enhancers tend to increase the level of panned or stereo sounds while suppressing centre sounds slightly. Sometimes this can be compensated for by EQ, but being aware of what's happening is half the battle. Other than the simple phase-inversion width enhancement used in the SPL Vitalizer, which is completely mono-compatible, width enhancement tends to compromise the sound of the mono mix, so always check with the mono button in. While most serious listening equipment is stereo these days, many TVs and portable radios are not, so mono compatibility is important.

11. Listen to the finished master all the way through, preferably using headphones, as these have the ability to show up small glitches and noises that loudspeakers may mask. Digital clicks can occur in even the best systems, though using good quality digital interconnects that are no longer than necessary helps to reduce the risk.

12. Try to work from a 44.1kHz master tape if the end product will be a CD master. If you have to work from a 48kHz tape or one with different tracks recorded at different sample rates, a stand-alone sample-rate converter can be used during transfer of the material into a computer. If you don't have a sample-rate converter, most editing software will allow you to do a conversion inside the computer, though this takes processing time and the quality is not always as good as that from a good-quality dedicated unit.

When using a software sample-rate converter, ensure that the tracks are all recorded with the computer system set to the same sample rate as the source material. If you don't have a sample-rate converter at all, don't worry too much, as transferring in the analogue domain via decent external A-D and D-A converters may well produce better results than an indifferent sample rate converter (with free re-dithering thrown in too!). Alternatively, if your master is for commercial production rather than for making CD-Rs at home, leave your master at 48kHz and inform the mastering house so that they can handle the conversion for you.

13. When you're transferring digital material into a computer, ensure that the computer hardware is set to external digital sync during recording and internal sync during playback. Also double-check that your record sample rate matches the source sample rate -- people will often present you with DAT tapes at the wrong sample rate, or even with different tracks at different sample rates. All too often this is overlooked, until someone realises that one of the songs is playing back around 10 percent slow!

14. Don't expect digital de-noising programs to work miracles -- even the best systems produce side-effects if you push them too far. The simpler systems are effectively multi-band expanders, where the threshold of each band is set by first analysing a section of noise from between tracks. For this reason it's best not to try to clean up your original masters prior to editing, otherwise there may be no noise samples left to work from. With careful use you can achieve a few dB of noise reduction before the side-effects set in -- as low-level signals open and close the expanders in the various bands, the background noise is modulated in a way that can only be described as 'chirping'. The more noise reduction you try to achieve, the worse the chirping, so it's best to use as little as you can get away with.

15. When editing individual tracks -- for example, when compiling a version from the best sections of several recordings -- try to make butt joins just before or just after a drum beat, so that any discontinuities are masked by the beat. However, if you have to use a crossfade edit to smooth over a transition, try to avoid including a drum beat in the crossfade zone, or you may hear a phasing or flamming effect where the two beats overlap. As a rule, crossfades should be as short as you can get away with, to avoid a double-tracked effect during the fade zone. As little as 10-30ms is enough to avoid producing a click.

16. On important projects, make two copies of the final mastered DAT (one as a backup) and mark these as Production Master and Clone. Write the sample rate on the box, along with all other relevant data. If you include test tones, document their level and include a list of all the track start times and running lengths for the benefit of the CD manufacturer. As mentioned earlier, if, for any reason, you have produced a 48kHz sample rate master, mark this clearly on the Production DAT Master so that the CD manufacturer can sample-rate convert it for you.

It's always a good idea to avoid recording audio during the first minute or so of a new DAT tape, to avoid the large number of dropouts commonly caused by the leader clip in the tape-spool hub. You can, however, use this section to record test tones, which will also demonstrate to the person playing your tape that it isn't blank! If you put DAT start IDs on each track, check them carefully to make sure that there are no spurious ones, and don't use skip IDs.

17. When deciding on how much space to leave between tracks on an album, listen to how the first track ends and the second one starts. Gaps are rarely shorter than two seconds, but if the starts and ends are very abrupt you may need to leave up to four seconds between tracks. Use the pre-roll feature of your digital editor to listen to the transition, so that you can get a feel for when the next track should start.

18. When using a CD-R recorder to produce a master that will itself be used for commercial CD production, ensure that the disc can be written in disc-at-once mode rather than a track at a time, and that your software supports PQ coding to Red Book standard. Check with your CD manufacturer to confirm that they can work from CD-R as a master, and take note of any special requirements they may have. Be very careful when handling blank CD-Rs -- there are commercial CDs on the market with beautiful fingerprints embedded in the digital data!

"The old adage 'If it ain't broke, don't fix it' applies perfectly to mastering."
.19 Be aware that stand-alone audio CD recorders usually have an automatic shut-off function if gaps in the audio exceed a preset number of seconds, usually between six and 20. This may be a problem if you need large gaps between tracks. Occasionally, even very low-level passages in classical music can be interpreted as gaps. Also note that these recorders will continue recording for that same preset number of seconds after the last track, so you'll need to stop recording manually if you don't want a chunk of silence at the end of the album.

20. When making a digital transfer from a DAT recorder to a CD recorder that can read DAT IDs, it's best to manually edit the DAT IDs first, so that they occur around half a second before the start of the track. Then you don't risk missing part of the first note when the track is accessed on a regular CD player. Alternatively, there are commercial interface units (or CD-R recorders with the facility built in) that delay the audio stream in order to make coincident or slightly late DAT IDs appear before the audio on the CD-R

 

Re: music production

thanks rev reading it now

 

Re: music production

i hope it help to you,if not and you have some questions tell me and we will fix it

 

Re: music production

EQ

One of the big differences between home recording and studio recording is the number of tracks that are used. In home recording, you are often limited to 4 or 8 tracks where professional recording uses 24 to 32 tracks.

The lower number of tracks means that you will often have to pre-mix instruments into a final mix before all of the tracks are recorded. In the studio, we can spread the instruments over more tracks and save all of the final mixing until all of the instruments are recorded. We will shortly find out the reasons that this fact gives professional recording a big advantage in obtaining a professional sound, it all has to do with hearing.

Two Hearing/Mixing Limitations

There are two hearing limitations that interfere with mixing. By hearing limitation we mean that the ears don't accurately hear the sound that is there.

Fletcher Munson Effect

The first hearing limitation interferes with all types of mixing. It equally interferes in the professional studio and the home studio. This limitation is called the Fletcher Munson Effect. The Fletcher Munson Effect is simply stated a follows:

Humans do not hear the low frequencies (bass) and the extreme high frequencies (treble) as well at low volumes.

Humans can generally hear sounds that have a frequency between 20 Hz and 20 kHz (20,000 hertz). The lower-frequency sounds (250 Hz and below) are the Bass Frequencies. Instruments like the bass guitar and lower drums (foot and toms) put out the majority of their energy in this range. The higher-frequency sounds (6 kHz and above) are Treble Frequencies. Instruments such as cymbals put out a majority of their energy in this range. A lot of instruments such as vocals have some energy in this range. The Treble Frequencies, for instance, contain the breath sounds of the voice. The middle-frequency sounds (250 Hz to 6 kHz) are called the Midrange. Instruments such as vocals and guitars have a majority of their energy in this range although all instruments have some energy in this range.

In comparing the difference in hearing between conversation levels and loud music playback levels, human hearing finds it 64 times as hard to hear the bass frequencies at low levels. It is about 16 times as hard to hear the extreme highs. These are compared to the midrange.

Because of this, the level at which you listen to the mix makes a huge difference in how the mix sounds. It becomes very difficult to judge, for instance, how much bass energy is the correct amount of energy for the mix. The end listeners will listen to the final product at different volumes depending on where they are, what they are doing and how they feel.

The most effective way to judge any mix is to listen to it at different volumes. You should endeavor to obtain a mix that sounds the best loud, soft and in-between. Leaving the volume at one setting while working is a guaranteed way to mess up a mix and make something that sounds good one day and sounds bad the next.

You can experience the Fletcher Munson Effect by listening to any recording loud for a minute or so and then real softly. When the mix is played at low volumes, it will be very difficult to pick out the bass line.

Masking

The second hearing limitation affects everyone but effects the home recordist, with a limited number of tracks, much more than the studio engineer with 24 or more tracks. This hearing limitation is called masking. Simply stated, masking is:

When two sounds are at similar frequencies, the one that is slightly louder will make the softer sound unable to be heard.

Because of masking, every instrument in a mix sounds quite different when it soloed (listened to by itself) compared to how it sounds in the mix.

When instruments put out sounds, they put out the tuned frequency and they put out energy at multiples of the tuned frequency. Going to a piano and playing the A above middle-C will cause a sound of the tuned frequency at 440 Hz. But the piano will also put out energy at multiples of 880 Hz, 1320 Hz, 1760 Hz, 2200 Hz, 2640 Hz and so forth. The "multiples" are called harmonics. The energy level of the different harmonics determines the "tone" of the instrument and is the leading factor as to why a piano and a instrument playing the same note sound different.

When an instrument is played with other instruments, some of the harmonics are masked, making the instrument sound different "in the mix" than "by itself."

The studio engineer has a great advantage over a home recordist because he/she can hear all of the instruments when doing the mix. The instrument's tone can be adjusted so that it sounds the best, when playing with the other instruments. The home recordist has to commit to a mix before all of the instruments are recorded. Getting a mix that sounds good now may not sound as good when the other instruments are added. This is especially true for rock bands where more guitars are added after the original tracks are recorded. The mid-range instrument of the guitar can cause the first guitar recorded to sound dull or even unable to be heard. The new guitars can interfere with the bass line, vocals, synthesizers and almost all instruments that have been originally recorded. Since these original instruments have already been mixed together, adjusting them now is rather difficult if not impossible.

Equalization To The Rescue

An equalizer is a device that will increase, or decrease, the level of signals at a certain range of frequencies. The simplest equalizer has the treble and bass controls often found on stereos. The bass control will increase (or decrease) the energy of any signal present below 250 Hz. The treble control affects the energy above 6 kHz.

When you "adjust the tone" with the treble and bass knobs on your stereo, you are increasing the harmonics of certain instruments and perhaps the tuned frequencies of other instruments.

Equalizing the instrument's sound while mixing is one effective way to overcome the two hearing limitations, especially masking.

Professional and "Home" Equalizers

Professional boards in studios have a lot of different equalization settings. There are often 4 bands, meaning that 4 frequencies can be adjusted at the same time. Since the mid range is the most important range of frequencies, the professional board will often have two different bands to cover these frequencies and also a treble and bass band.

The least-expensive boards and board/recorder units only have a treble and bass control to adjust tone on the individual instruments. This type of unit does not really have enough features to obtain a professional sound at home.

There are many boards and board/recorders that are only slightly more expensive than the cheapest units and have a "sweepable" midrange equalization control on every channel. This control is getting very close to the professional controls found on professional boards. The sweepable midrange frequency control allows you to adjust the center frequency of the frequency range that will be boosted or cut with the other midrange knob. Now you have a control that can assist you obtain a professional-sounding result.

The Home Equalizer Controls

Midrange equalizer controls on professional consoles and better home-recording equipment have at least two knobs.

The first knob is the frequency knob. The frequency knob sets the frequency that will receive the maximum boost (or reduction) of energy and is technically referred to as the "center frequency." The typical home-recording equipment will vary the center frequency between 250 Hz and 5 kHz. The control often appears as seen in figure 1a.

Professional console frequency knobs are well labeled, allowing the studio engineer to exactly set the frequency of the equalization. Home recording equipment usually has little labeling, just showing what the maximum, minimum and middle settings that the knob will yield. Figure 1b shows what the knob would look like if 11 frequencies were labeled.

The second control is the "amount" knob. On professional equalizers the amount of equalization is marked with "dB" settings. One dB represents approximately the smallest change that can be heard and +6 dB represents the signal strength getting twice as high. A 12 dB increase would represent the signal strength at the center frequency getting 4 times as strong.

Home recording equipment usually does not label the amount knobs except with lines that allow the user to get the amount back to a setting used before, as shown in figure 1a. If the amount knob was labeled with "dB" markings, it would look like the control in figure 1b.

The "HF" control (High Frequency Control) on the equalizer allows the energy from 10 kHz and above to be boosted or reduced. The only knob available is the amount knob. The "LF" control (Low Frequency Control) allows the energy from 100 Hz and below to be boosted or reduced. Again the only knob available sets the amount of equalization.

The Home Recordist's Challenge

Because one instrument's sound will mask another instrument's sound, equalization is best applied when you can hear all instruments. Somehow the home recordist has to mix without hearing all of the instruments.

The home recordist needs to develop an "ear" for how instruments need to sound before they are combined with other instruments. You can't just adjust instruments to sound good by themselves and then combine them. Masking will give the mix a mushy, indistinct sound. Often you need to make instruments sound less-natural so that they will distinctly "cut-though" the other instruments.

This is especially true for the bass and for the vocals. Typically the bass guitar has to have accented midrange attack and string sounds to sound good in a mix. The tendency is to boost the fullness of the bass when you listen to it by itself. This will cause the bass to get "lost" in mix of instruments 90% of the time. The vocal usually has to have over-accented presence and be slightly "thin" sounding to sound good with other instruments playing.

When equalizing guitars, the 2.5 kHz range of frequencies are often boosted to give an increased "attack" to the guitar. Often, to sound good in a mix of instruments, the guitar has to have an unnaturally loud attack. If, however, there are several guitars, you should not boost all of them at the same attack frequency. If you boosted 3 guitars at 2.5 kHz, you will insure that one guitar will cover up the sound of another. The correct procedure in this case is to use slightly different frequencies on the different guitars, like 2.5 kHz, 4 kHz and 5 kHz.

Recommended Equalization Settings

Years of trial and error experience will allow the home recordist to know how to use the equalizer to get a clear sound. Most people don't want to wait that long to start getting professional sounding product.

In order to help, we have compiled the most-often used equalization settings used for different instruments in mixing. We when one step further by converting them to the settings that you would use with home recording gear. These settings appear in figure 2.

Using The Recommended EQ

The equalization settings are based on micing close with good-quality microphones. The equalization for bass was with the instrument recorded directly into the console and with the tone and volume controls on the bass in the fully-up position. The vocal equalization is for a male vocalist about 8 inches away from the microphone.

You can't just take these recommendations as a "bible" and always use the same settings. This is because different instruments and different microphones will sound different. You will find the most variance with guitars; this is because you have tone and volume controls on the guitars and on the amplifier. In addition, different guitars and guitar amplifiers sound quite different. The equalization for the bass will be the most consistent. In any case these settings give you a good "starting point" for using your equalizer.

Demonstration Tape

To further assist you we have made a demonstration tape available. On the demonstration tape you will hear instrument sounds without equalization and with equalization. You can compare the equalized instrument sounds with the sound you are obtaining in the session as a guide for setting your equalizer.

The demonstration tape takes the tune "All Behind" by the group "Traitor Gate" to give the equalization examples. The band is a rock band with 2 rhythm guitars and a lead guitar (as well as drums, bass, vocals and background vocals).

The demonstration tape, Professional Equalization Demonstration, is available from the Recording Institute of Detroit, 14611 9 Mile Road, Eastpointe, MI 48021, (810) 779-1388. The cost of the tape is $12 plus $3 shipping/handling.

Settings:

Drums:

+9 dB LF

-12 dB MF at 400

+6 dB HF

Rhythm Guitar 1 (Cleanest)

+3 dB LF

+6 dB MF at 2.5 kHz

Rhythm Guitar 2

+1.5 dB LF

+6 dB MF at 4.0 kHz

+1.5 dB HF

Lead Guitar

+6 dB MF at 5 kHz

+3 dB HF

Lead Vocal

+3 dB MF at 3.5 kHz

+3 dB HF

Background Vocals

+1.5 dB LF

-6 dB MF at 3 kHz

+3 dB HF

Bass Guitar

+2 dB LF

+4 dB MF at 400 Hz




EQUALIZATION BY THE OCTAVE

BY ROBERT DENNIS
Equalization is the most-used, most-mis-used, most-over-used and most-under-used signal processing device. It is also the most powerful. By definition an equalizer is a gain control that raises or lowers gain at a specific set of frequencies without affecting the gain at other frequency ranges.
I learned a lot about equalization by sitting down with a graphic equalizer and lots of records. After about 30 hours of listening to the effect of different bands of equalization on the records, I began to "hear" the effect of certain frequencies on overall mixes. Since I had a job in mastering, this was a direct application as to how I was going to use equalizers.
Doing an exercise like this is the very beginning of training in using equalization. There are commercial CD packages which attempt to duplicate this type of test. They are no real substitute for switching the equalization on and off as the CD (or music source) plays.
The next step in the training is to apply specific equalization to specific instruments. Recommended or "key" frequencies are used and the equalization is switched on and off while soloing the instrument and then while listening to the instrument in the mix.
The point of this "ear" training is to gain the ability to hear what frequencies would be needed to bring up or down in a specific mixdown. Until one obtains the ability to "hear" frequency, one has only limited ability to use equalization.

Frequency Ranges:
A key to understanding equalization is to gain an understanding of the effect of different frequency ranges on music and instrument sounds.

The "First Octave"
The first usable octave for most recording is the 40 - 80 Hz range, with equalization settings centered around 50 Hz. This range of frequencies is often referred to as "Low Bass"
There is sound between 20 Hz and 40 Hz but little or no sound from instruments. The lowest pipes of a pipe organ will get into this range but more "ordinary" instruments like Bass Guitar, Upright Bass and Foot Drums do not. The lowest pitch on a bass guitar or string bass is at 41 Hz. Thunder, earthquakes and rumble from the building shaking extend below 40 Hz. While mixing, watch out for objectionable sounds below 40 Hz caused by building shifts and mic stands moving with heavy footsteps. If there is objectionable sounds in this range, the range can usually be taken entirely out with a filter.
The first octave that we deal with (40 - 80Hz) gives more of a "feeling" and sense of "power" to the sound. This range is way down or non-existent in smaller stereo systems. This range is difficult to hear at all at medium and low volume levels because of the Fletcher Munson Effect.
To properly set the amount of low bass in your mix or in your instrument sound, you must listen both loud and soft. You also may want to listen to the mix or instrument on large and small speaker systems. Too much energy in this range will make the mix sound muddy on large speakers played loud and still sound good on small speakers played at a medium volume. You want the mix or instrument to sound larger and more powerful over large speakers without sounding muddy.
Rap, Hip Hop and "Dance" music (under various names) often have extra energy in the low-bass range. This is what causes cars equipped with sub-woofers to shake. Usually, however, it is not the entire mix that is boosted below 80 Hz, but just, for example, the foot drum. By boosting the energy on only one or two instruments, "clarity" can be achieved without "mud."

The Bass Range
Covering about 1.5 octaves, from 80 Hz to 250 Hz, this range of frequencies determines the "fatness" and "fullness" of the instrument's sound. Equalization is usually applied centered around two frequencies, 100 Hz and 200 Hz.
For guitars and bass, the 100 Hz range tends to add body and fullness. Excessive energy in this range tends to make these instruments sound "boomy.," This range of frequencies is still greatly affected by the Fletcher-Muson Effect; this means you will need to listen to the mix and instrument both loud and soft. Similar to how the 50 Hz range affects the bass and foot, the guitars should sound fatter when played loud, not boomy. Reducing the 100 Hz energy on the guitar will usually cause distinction between the bass and guitar parts. The lowest fundamental frequency on a guitar is around 80 Hz.
For vocals the 200 Hz range determines the fullness of the vocal. This range can often be reduced to increase distinction on the vocal. If, however, boosting in higher frequencies on the vocal makes the sound "thin" or "small" a boost of 200 Hz. will restore fullness.
When 100 Hz is reduced on a guitar or bass to reduce "boom," at small boost at 200 Hz can be helpful to keep the instrument from sounding "lumpy" (certain notes hard to hear and others standing out). The guitar and bass have almost equal energy at their fundamental and 2nd harmonic frequencies. Thus if a range of notes becomes hard to hear because of a at lot of 100 Hz, reducing energy at 100Hz and adding energy at 200 Hz will help the notes be heard again.

The Bass Presence / Lower Mid Range

Covering about one octaves from 250 Hz to 500 Hz, this range accents ambience of studio and adds clarity to the bass and lower-string instruments (Chello and Upright Bass). Too much boost can make higher-frequency instruments muffled sounding and low-frequency drums (foot and toms) have a cardboard box quality. Equalization in this range is applied at many frequencies but most often between 300 Hz and 400 Hz.

The lower part of this range (250 Hz to 350 Hz) is sometimes referred to as "Upper Bass" and is used to increase distinction and fullness on the vocal, especially on female singers.

The Lower Mid Range in general can be viewed as the "Bass Presence Range" Increasing this range gives clarity to the bass line and the lower-register of pianos and organs. Clarity and distinction can be obtained between the foot drum and bass guitar by both reducing the foot and increasing the bass guitar in this range, at the same frequency.

This range is often reduced for overhead drum and cymbal microphones to increase clarity and presence on these instruments' and reduced on lower drums (foot and toms) to reduce boxiness.

The Mid Range

The Mid Range band of frequencies covers two octaves from 500 Hz to 2 kHz. This range can give a horn-like quality to instruments (500 Hz to 1 kHz) and a "tinny" sound (1 kHz to 2 kHz) or a telephone-like quality (all of the range). Equalization usually centers around 800 Hz and 1,.5 kHz.

The mid-range also tends to accent the presence (800 Hz) and attack (1.5 kHz) of the bass guitar. The lower pitches of a rhythm guitar can be given more attack by a boost at 1.5 kHz.

For your Mid Range Instruments (vocals, guitars and piano) this range is most-often reduced rather than accented. Reducing 500 - 800 Hz on an acoustic guitar can remove the "cheep" sound and make it sound more "silvery." Reducing 800 Hz on a vocal makes it sound less nasal and have more body and presence. For snare drums, a reduction of 800 Hz can take the tinny, cheep sound out of the drum and make the snares have more sizzle rather than rattle.

The Upper Mid Range
Covering about one octave, this range of frequencies is responsible for the attack on percussive and rhythm instruments and the "projection" of mid range instruments. Equalization can be applied at any frequency in this range but still somewhat centers around 3 kHz.
On the foot drum, boosting 2.5 kHz or 4 kHz increases the attack. 2.5 kHz sounds more like a felt beater and 4 kHz sounds more like a hard-wood beater. These frequencies can also be used to increase the attack or "hit" sound on toms and snare drums.
Guitar lines often get more attack and distinction with equalization added at this range. A small boost (1-3 dB) for the vocal will increase projection. Adding too much energy, in this range, makes it hard to distinguish the syllables of the vocal and can cause listening fatigue. This range of frequencies is often reduced on background vocal to give them a more "airy" and "transparent" sound.

The Presence Range
Although this range covers a mere half-octave of 4 kHz to 6 kHz, it is an often-used band of frequencies. This range makes most vocals and melody instruments sound closer and more distinct. Over-boosting causes a irritating and harsh sound. Equalization centers around 5 kHz.

The Treble Range
Covering approximately that last two octaves of sound (6 kHz to 20 kHz), this band of frequencies is responsible for the brilliance and clarity on instruments. Equalization centers around 7 kHz, 10 kHz and 15 kHz.
The vocal "S" sounds are at about 7 kHz, making this a frequency that is avoided for vocals. Care must be exercised in reducing 7 kHz on vocals, however, because the vocal will sound dull very fast. The breath sound of the vocal is at 15 kHz and above, giving a breath quality without much accent on the "S": sound of the vocal.
The 7 kHz frequency is also the "metallic attack" frequency on drums The "sizzle" of cymbals is at 15 kHz.
When equalizing, 10 kHz and above is often used as a general "brilliance" frequency band.

next part obout EQ tomorow

 

Re: music production

LOOL i got lots of work to do XD im on it, right now!

 

Re: music production

yes and be good in mastering its take long time

 

Re: music production

EQ EXPLAINED


Equalisation is one of the most powerful tools in your sonic toolkit and can be your greatest enemy or your greatest ally in the battle for the perfect sound. DAVID MELLOR gives advice on how and when best to use it.



The next time you make a recording, as an experiment set all the EQ controls of your mixing console to their centre positions and leave them there until you have finished the final mix. Don't be satisfied with anything less than perfection, and don't give yourself the excuse that you can't get a good sound because you were not able to use the EQ.

EQ is a very powerful and effective item in your sonic toolkit, not unlike a circular saw in fact! But you wouldn't use your Bosch or Black & Decker for a fine carving, would you? No, you would use basic hand tools and, most importantly, your skill and judgement. As a recordist, it is your own abilities which are going to be most important to the degree of success of your recording, and you should always use the appropriate tool for the appropriate situation.

It is always best to ensure that you get as good a sound as possible from the microphone, synth or sampler coming into the mixing console. If you start off with good sounds, then a good result is almost inevitable. It is becoming increasingly popular to use microphones for recording, even when DI (direct injection) is possible, because of the wider variation of tonal qualities available. Even small variations in microphone position make vast differences to the sound picked up. It is a sign of an expert recording engineer that he or she will listen carefully to the sound from the mic and adjust its position and angle, and even try out several microphones, rather than pretend that it is always possible to get it right first time.


"Graphics are great for EQing an entire mix so that you can shape the sound as a whole, even after you have processed the individual elements."



Once you have built up your skills in this area, then you can think about using EQ. I could spout all sorts of proverbs about the things you can't make silk purses out of and the things you ought not try to polish, and these proverbs apply especially to EQ. You should always aim to use EQ to improve an already wonderful sound. If the sound isn't good without EQ, then you will never end up with anything but second best. The only time you should ever use EQ to 'save' a sound is when you have been given a tape to work on that was recorded by a lazy engineer.

Just as there is an art to creating a brilliant sound, there is an art to bringing that sound to perfection, and also blending several sounds together to make the perfect mix. Van Gogh didn't learn to paint overnight, and no-one is born with the inbuilt ability to EQ. It's a skill that is learned by experience and a good deal of careful listening.
WHAT IS EQ?

As a first step (although I know 99% of you have used EQ already!), let's see what EQ is and what it does. Then I'll move on to looking at the machinery and techniques.

Figure 1 shows one of the parameters you would expect any item of sound equipment to aspire to -- a flat frequency response. This, or at least a very close approximation, will be the frequency response of your mixing console with the EQ controls set to their centre positions, or with the EQ buttons switched off. Here, the balance of frequencies of the original signal is preserved in correct proportion at the output. In other words it is just as trebly, tinny, harsh, nasal, honky, bassy or boomy as it was when it left the microphone; or just as perfect perhaps.

Notice that the frequency response indicates what the EQ does to the sound. A cymbal will naturally have strong high frequencies, for example, and that emphasis towards HF will be preserved by a flat EQ setting. Likewise, a flat EQ will reproduce perfectly the boomy bottom end of an undamped bass drum.

If Figure 1 shows a flat response between about 20Hz and 20kHz, Figure 2 and Figure 3 show two of the curves you might expect to get from a mixing console EQ. Oddly enough, measuring the EQ and plotting the curve is something that only 0.001% of recording and sound engineers ever get around to doing at any stage in their creative careers, and only 0.0001% have their own equipment to do it to any reasonable accuracy. Even if it's hardly ever done, except on the test bench, it's a useful concept which you can carry around in your head without ever bringing to the forefront of your mind. So if a producer ever says to you, "Let's have a little more presence in the vocal", your subconscious mind will retrieve the bell-shaped curve of Figure 2 from your memory bank while your conscious mind adjusts the controls and judges the sound.

In Figure 2 we are adding an EQ boost, and there are three parameters that we would like to be able to control (if the EQ has knobs for all three). First and foremost is the frequency: this boost could be centred on any frequency according to the instrument and according to which characteristics you want to accentuate. Second is the gain, which is the degree of boost and can be measured in decibels (dB) at the centre frequency. Some mixing consoles even calibrate this control in dB, and a good thing too! You might like to have a range of up to 12 or 15dB as a maximum. Gain can also be negative, producing an EQ cut, which would be written as a gain of -6dB (or whatever) at the centre frequency, so the curve would dip downwards. EQ cut, by the way, is a vastly underused resource on many consoles, but more on this later...

The third parameter is Q, which is only occasionally offered on mixing console EQ. As well as being the star of the last ever episode of Star Trek: The Next Generation (or so my crystal ball informs me), Q is a measure of the width of the bell-shaped curve -- the bandwidth as some might say. A low Q -- 0.3 is low -- will allow the EQ to cover a wide range of frequencies, while a higher Q -- 5 is high -- will allow you to home in on a particular feature of the sound.

The bell-shaped curve of Figure 2 is often referred to as 'peaking' EQ, and applies to all mid frequency range EQ sections and a good proportion of high and low frequency EQ sections too. Figure 3 shows a 'shelving' EQ, where the boost (or cut) extends from the chosen EQ frequency all the way to the extreme end of the range. I have shown a low frequency shelving EQ in boost mode, but it could have been a high frequency cut with a similarly shaped but differently orientated curve. It isn't possible to say which type of curve is better, for it depends on what you want to achieve, but some consoles have a button to allow you to choose.
OUTBOARD EQ

Mixing console EQ is getting better and better, particularly in the low-to-mid price range. There was a stage where I was sure that the designers were inventing their EQs with the aid of a pointed finger and a pocket calculator rather than a keen pair of ears and advice from practising recording engineers, but this is no longer true of most console EQs. Nevertheless, no matter how good the EQ on your mixing console, there will come a time when you need to use an external or 'outboard' unit. This might be because you need a facility not available from your console EQ, or you might prefer to use an EQ unit for some subtle characteristic sheen it gives to the overall sound.


"You should always aim to use EQ to improve an already wonderful sound. If the sound isn't good without EQ, then you will never end up with anything but second best."



Outboard EQs come in two basic flavours: graphic and parametric. A good graphic equaliser typically has 30 or so slider controls for frequency bands nominally covering a third of an octave each. You would use two for stereo. The basic idea of a graphic is that as you set the slider controls to achieve the sound you want, the levels of the sliders 'draw' the EQ curve, as if you had measured and plotted it the long way. Unfortunately, graphic equalisers are somewhat economical with the truth and only give a rough idea of the actual curve. This is because each band does not cover only a third of an octave; its effects are felt most there but the slider will actually affect frequencies belonging to two or three bands either side of it to a distinctly noticeable extent.

Whatever deficiencies graphic equalisers may have under the Trades Descriptions Act, they are still very useful tools to have around. Mixing consoles can handle basic EQ tasks better and more quickly, but there are certain applications where graphic EQs have the edge. More on this shortly.

The alternative to a graphic outboard EQ unit is the parametric EQ. This is so called because it offers control over all three EQ parameters I mentioned earlier -- frequency, Q, and gain. A good parametric EQ unit may offer five bands, which cover the entire frequency range, or you might find three fully parametric bands with dedicated low and high frequency bands too.
USING EQ

Successful equalisation requires good equipment and a thoughtful approach from the engineer. Experienced engineers EQ by instinct and their fingers operate the controls as fluently as a jazz pianist tickles the ivories. But this fluency doesn't come automatically, it can only be won by experience. Anyone can grab the low frequency knob and wind up the bass to the maximum, but if you are serious about your recording then you will realise that it isn't just yourself you have to please; you have to consider what other listeners like and what systems they may be playing the recording on.

There is also a good technical reason why you should think before adding a lot of bass: for a given level of input, any small or medium size loudspeaker will produce much more sound at mid frequencies than at low, and if you boost the low frequencies too much then the overall level the speaker can achieve without significant distortion is less -- sometimes much less. It's a matter of compromise: the more bass you add, the lower the overall level can be. This also applies to other frequencies in the mixing console itself.

Adding EQ adds level, and it is very easy to boost the signal so much in the EQ section of the console that you run into clipping and distortion. Since the fader comes after the EQ, lowering the fader will do nothing to solve this. The answer is to reduce the gain, to allow the signal a little more headroom if necessary. One further technical point: changing the EQ of a signal nearly always changes the level, so each time you adjust the EQ you will have to consider moving the fader to compensate. It's something that will come automatically after a time, but newcomers to recording often concentrate more on the change in the sound itself and don't notice that it has suddenly become more or less prominent in the mix.

Enough of the technical stuff, recording is an artistic occupation so let's consider the subjective facets of EQ. If we consider individual sounds first, let's assume that the signal coming from the microphone is already as perfect as can be, being the result of careful positioning and angling. Each instrument has certain bands of frequencies that are strong and some that are weaker. The human voice, for example, is very strong around the 3 to 4kHz region, no matter whether male or female, or what note is being sung. When using EQ, you will be considering which characteristics of the sound you want to accentuate, or which you want to reduce. One way to consider this might be to imagine an instrument which was an 'average' of all real instruments, where the characteristics of normal instruments were smoothed out into something that had a neutral sound. When EQing a real instrument, you will either want to exaggerate its individual characteristics and make it more distinctive, or reduce its individuality and make it more like this hypothetical 'average' instrument.

This is quite simple to do, and we can make use of the standard sweep mid range control that is found on most mixing consoles, with controls for frequency and gain. A fully parametric equaliser with a Q control can offer even more precision.

First set the gain control to a medium amount of boost -- the three o'clock position of the knob is usually okay. Now sweep the frequency control up and down to the limits of its range and listen for the frequencies at which the effect is strongest. These are the frequencies in which the instrument is rich. Boosting the instrument's strong frequencies will enhance its individual characteristics and, for example, make a clarinet even more dissimilar to an oboe or any other instrument. In effect, you are making the clarinet even more clarinet-like.

When you have found the instrument's strongest frequency band, set the amount of boost according to taste and always compare what you are doing with the flat setting. If you have EQ sections to spare, you may be able to cut down on frequencies which don't enhance the sound of the instrument. Some instruments which are not known as bassy instruments nevertheless have a high low frequency content; cymbals for instance. On many occasions it will be well worth cutting down on frequencies which you don't consider to be any use to the instrument, freeing up a space in the frequency spectrum for another instrument to use.

Enhancing the sounds of individual instruments in this way is useful, but watch out when mixing that you are not boosting the same frequencies on each instrument. It is a trap for the unwary to boost every instrument at around 3kHz to help it cut through at a frequency where the ears are very sensitive. This will produce a mix that is very tiring to listen to.

The opposite of the enhancement technique is where you lessen the individuality of each instrument and make it more like our hypothetical 'average' instrument. To do this, find the instrument's strong frequencies with the mid EQ set to boost as before, but then cut these frequencies, by as much as you feel appropriate. This won't make the instrument sound better in isolation, but it will help it blend in with the other instruments in the mix.

Many aspiring engineers do not appreciate how useful EQ cut can be, but the expert will skilfully share the frequency spectrum among all the instruments so that each has its own space and doesn't have to fight with the others for attention. Using EQ in this way can result in a powerful and full sound from a small number of tracks.

Mixing consoles differ in the usefulness of their high and low frequency EQs, and it is often necessary to bring in an outboard EQ that can do the job better. I would say that it is the purpose of the low frequency control to add 'weight' to the sound without making it 'boomy'. These are subjective terms I know, but I think we can all appreciate the difference between a sound which is firm and solid in the bottom end, and one which has plenty of bass but gives the impression of being out of control. In the other direction, the low frequency control should cut low frequencies that are not contributing anything useful to the sound, while retaining the depth and body of the low mid. At the high frequency end, you should be able to cut any 'fizz' from the sound while still leaving it clear and incisive, and you should be able to make the sound brighter without the extreme top becoming aggressive. If you can't achieve all this with your console's EQ, you may have to spend a thousand pounds or more on an outboard unit that can.

When you have explored all the possibilities your console's EQ can afford and you have visited your local hire company for outboard units that perform the same function only better, you'll be keen to get your hands on a graphic equaliser. This is a rather different animal which appears at first to offer the ultimate in flexibility: just raise or lower the frequency bands you are interested in for quick and precise control. Unfortunately, you will find that precision is lacking because each individual band alters frequencies over quite a wide range on either side of its nominal centre frequency.

This doesn't mean that graphics are useless -- far from it. Graphics are great for EQing an entire mix so that you can shape the sound as a whole, even after you have processed the individual elements. If you know your way around, you can do this by taking a couple of outputs from the mixing console back into two channels and using the console's EQ again, but you'll only be applying more of the same, and doing it the graphic way really is much more satisfying. Graphics are also great for adding bite to a sound: just raise one or two sliders somewhere in the upper frequency region and you will make the sound more cutting without lifting the whole of the high frequency range. Experiment at your leisure.
PROBLEM SOLVING

If you are working on a tape made by another engineer who isn't quite as fastidious as you, then you may find yourself faced with problems that EQ can help rectify. Unwanted sounds have a knack of finding their way onto recordings, particularly live recordings. If you have a 50Hz mains hum, for example, then a graphic will be able to help at only a little loss to the musical sounds on the recording. You can also use a parametric equaliser set to a high Q to home in on the unwanted frequency. Some equalisers have special notch filters to cope with precisely these situations. 50Hz hum may be removed to a reasonable extent, but the buzz caused by lighting dimmers may be impossible to get rid of. If the buzz isn't too harsh then you can try cutting the 50Hz fundamental and its harmonics at 100Hz, 150Hz, 200Hz etc. I can't promise anything, but it may make the recording just listenable.

Apart from hum or dimmer noise, if a recording is too noisy then very often the noise is most noticeable at high frequencies. Here you can use your EQ to strike the best compromise between cutting as much of the offending component of the noise as possible while still retaining some brightness in the sound. You may be able to apply a little boost at high mid frequencies, although the result will remain a compromise.

Even if the recording has no hum, buzz or noise, it may previously have been over-EQ'd. It is quite difficult to ameliorate the results of over-zealous EQing, particularly if some frequencies have been cut to a large extent. Trying to boost these frequencies back up again may result in an unacceptable amount of noise becoming apparent. Once again, compromise is necessary, although if you were dealing with one instrument from a multitrack mix you may be able to patch in a noise gate to help in this instance.
THE FUTURE OF EQ

There is no doubt that the designers of EQ both in mixing consoles and outboard units are going to pay far more attention to the sound of the EQ rather than the technical specs. Some manufacturers have started to drop the conventional 'low', 'mid' and 'high' labels and describe their controls with names such as 'bottom', 'sheen' and 'glow'. I don't think this is a bad idea, since it will focus our energies less on the technicalities and more on the sound the EQ produces. I wouldn't be at all surprised to see EQ being combined -- not just in series within the same box, but fully integrated -- with compression or distortion far more often than it has been up to now.

Whatever the future may offer, EQ will always be one of the most powerful tools in your recording toolkit, so make the most of it.


EQ WITH BOLDNESS

When adjusting the amount of EQ to apply (ie. the EQ gain), it's tempting to adjust it very carefully and change the setting in small increments. The problems with this method are: (a) that if the EQ setting isn't right then it is wrong and thus needs total reconsideration; (b) that the ear quickly grows used to changes in the frequency balance of a sound.

It may not always be appropriate, but the next time you want to change the EQ level of a sound, grab the control firmly, twist it all the way up and all the way down and quickly settle on a new position which will hopefully be just right.

EQ TERMINOLOGY

CUTOFF FREQUENCY The frequency at which a high or low frequency EQ section starts to take effect. Also referred to as turnover frequency.

SLOPE The rate at which a high or low frequency EQ section reduces the level above or below the cutoff frequency. Usually 6, 12, 18 or 24dB/octave.

PASS BAND The frequency range that is allowed through.

STOP BAND The frequency range that is attenuated.

FILTER An EQ section of the following types:

HIGH PASS FILTER A filter section that reduces low frequencies.

LOW PASS FILTER A filter section that reduces high frequencies.

BAND PASS FILTER A filter section that reduces both high and low frequencies.

NOTCH FILTER A filter that cuts out a very narrow range of frequencies.

GAIN The amount of boost or cut applied by the equaliser.

Q How broad or narrow the range of frequencies that is affected.

SWEEP MID A middle frequency EQ section with controls for frequency and gain.

PARAMETRIC EQ An EQ section with controls for frequency, gain and Q.

GRAPHIC EQ An equaliser with a number of slider controls set on octave or third octave frequency centres.

BELL An EQ with a peak in its response.

SHELF A high or low frequency EQ where the response extends from the set or selected frequency to the highest or lowest frequency in the audio range.

HF High frequencies

LF Low frequencies

MID Midrange frequencies

TREBLE Hi-fi enthusiasts' word for HF.

EQ OFF BUTTON The sign of a good mixing console!

EQ HINTS & TIPS

• If your mix sounds 'muddy', boost the main frequency range of each of the principal instruments. Boost 'decorative' sounds even more and pull the faders right down.

• If you can't get your tracks to blend together in the mix, cut the main frequency range of the principal instruments.

• To make vocals stand out in the mix, boost at around 3kHz.

• For extra clarity, cut the bass element of instruments which are not meant to be bass instruments.

• Adding EQ boost often adds noise. Listen carefully to arrive at the best compromise.

• Changing the EQ changes the level. Always consider re-adjusting the level after you EQ.

• If you add a lot of EQ boost, you may run into clipping and distortion. Reduce the channel's gain to eliminate this.

• If you use EQ to reduce feedback in live work, take care not to take too much level out over too wide a range of important frequencies, particularly the vocal 'presence' range around 3kHz.

• If your mixing console has an EQ Off button, use it frequently to check that you really are improving the sound.

 

Re: music production

THE MYSTERIES OF METERING


All mechanical meters are VU meters, all bargraph meters read peak levels -- and both types will give the same reading if you feed in a test tone. Reasonable enough assumptions, but wrong on all counts, as PAUL WHITE explains.



The really wonderful thing about standards is that there are so many of them, and nowhere is this more evident than when you look at metering. This article examines the complicated issue of metering standards, but those unfamiliar with the general terminology of metering (eg. dBu, dbv, and the conventions of 'plus 4' and 'minus 10' operation) are advised to check out my article from SOS February 1994, 'dBs Explained', which should clarify many of the terms used here.

Tape machines have meters, mixers have meters and signal processors have meters, but what do they actually tell you? To take the last point first, most meters are designed to tell you when the piece of equipment to which they belong is being fed the correct signal level. Metering is vitally important, because all electronic devices have a lower signal limit (where the signal is so small it is overpowered by the circuit noise), and an upper limit where the signal reaches the unit's maximum level -- whereupon clipping occurs. By using a meter properly, you can choose a signal level which is as high as possible without clipping, which will produce the best possible signal-to-noise ratio.
THE NEEDLE & THE DAMAGE DONE

The first type of meter built specifically for audio use was the VU meter, VU standing for Volume Units. The idea was to build a meter that would produce a reading similar to the loudness or volume level perceived by the listener. The way the human ear hears sound is that very short-duration sounds appear quieter than longer bursts of sound at the same level. Moving coil meters can be built to simulate this characteristic pretty well, because the inertia of the mechanism limits the speed at which the meter can respond to transients. Put a drum beat into a VU meter, and the meter will barely have begun its climb than the beat will have ended, and the meters start back down again.

VU meters measure the RMS (root mean square) value of the input voltage: a sine wave that alternates between plus and minus 1 Volt, peak to peak, will actually produce a reading on a voltmeter of 0.707 volts, which is what you'd get if the voltage in the sine wave were averaged out into a steady DC voltage. Because the dBu scale used for audio is also an RMS-based scale, steady sine waves or test tones should result in complete agreement between the VU value and the dBu value. For example, a mixer designed to operate at a nominal +4dBu should be outputting exactly +4dBu when the VU meters read 0dB, providing the input is a steady sine wave tone. One exception to this is to be found on some of the newer Mackie mixers, where they have decided to make the VU meter read 0VU for 0dBu. This means that you can use the mixer at either +4 or -10, and the meters will always tell you the actual signal level at the output -- a practical and sensible option.

VU meters work fine with analogue tape, because analogue tape has quite a lot of headroom above its nominal operating level, during which the level of distortion increases progressively, unlike digital systems which merely clip. For this very reason, when used with digital systems, VU stands for Virtually Useless, because the peak levels produced by something like a drum kit could be driving the digital recorder into clipping while the VU meter is reading around-10dB or less.

One myth it is important to dispel is that only moving coil meters are VUs: you can also have bargraph VU meters, because the characteristics of a bargraph meter depend entirely on the circuitry driving them. A line of LEDs has no mechanical inertia -- so in theory, they can be made to respond as fast or as slow as you like.
PPM

Peak Programme or PPM meters are more in keeping with the digital age, because they are designed to respond fast enough to show any signal peaks that might cause distortion. Some also incorporate a peak hold facility, where the highest peak levels are displayed for several seconds to make sure you don't miss them. However, they still don't read absolute peak values, because clipped peaks shorter than a millisecond or so are generally inaudible. Unlike the VU meter which reads an RMS or average value, the PPM reads the voltage between the negative and positive signal peaks -- which explains why you don't see the same reading when a steady sine wave is fed into a VU meter and a PPM meter.

In fact, if you take an EBU standard PPM meter, it will read 8dB higher than a VU meter monitoring the same signal in the same system. This difference equates to the difference between a reading of 0.707 Volts and 2 Volts, the peak-to-peak reading you'd get from a +/- 1V sine wave. Of course, in real life, you can calibrate a meter to read anything you like, and the BBC have theirs calibrated so that a 0dBu sine wave reads +6dB on their PPMs. Some standards go one further, and rationalise that both types of meter should read the same, so some European and Scandinavian PPMs may read exactly the same as a VU meter for a steady sine wave input.

As if that wasn't confusing enough, you can also have moving-coil PPM meters. All you have to do is design the drive electronics to hold the peak level until the meter has had time to respond, and you've cracked it. The peaks may register a fraction late, but they'll still register.
DIGITAL MULTITRACK

A potentially confusing situation arises when using analogue mixers with digital multitrack machines, because the meters on the mixer and multitrack don't match up, not even when you put in a steady state test tone. Even if your mixer has true PPM meters, the chances are that the levels still won't match. Why?

I've spoken to several different people about this, and they all come up with slightly different answers, but as a rule, digital multitracks are calibrated so that a 0dB test tone (measured either VU or PPM) coming out of a mixer will read several dBs below 0dB (clipping) on the digital multitrack. This makes a lot of sense, because mixers are designed to be driven 'into the red', and if you have a model with VU meters, you could be a lot further into the red than you imagine.

Digital machines won't tolerate any overload unless the period of clipping is so brief that you can't hear it, so calibrating the input electronics in this way is one way of helping the user stay out of trouble. It also means that the mixer can be driven a little way into the red, as normal, to make the most of the available headroom. Similarly, when the signal comes back from a digital tape recorder, it's often hotter than you expect, for exactly the same reasons. That means when you're mixing from Alesis ADAT or Tascam DA88, you might find your mixer once again flicking into the red, even though the meter readings on the multitrack are below 0dB.

On my desk, which has moving coil meters and is calibrated to run at +4dBu, a 0VU test tone reads around -15dB on my ADAT. Even allowing for the difference of 8dB between PPM and VU metering, this still leaves around 7dB of artificially introduced headroom. To put it another way, your digital machine won't clip until the mixer output exceeds +7dB PPM.
USE YOUR HEAD (ROOM)

When recording to analogue, the meters can only give you a rough guide as to what the right recording level should be, and if they're VU meters, the readings will be different for percussive material and music with more sustained sounds. Add this to the fact that modern tape can often accept a lot more level before saturating than older types, and it soon becomes apparent that the only way you can really define the limits is to make a few test recordings at different levels, to find at what point distortion becomes audible. After a little experience, you get used to what to expect from a VU meter with different types of input material, but as far as I'm concerned, reading a VU meter is still as much an art as it is a science!

A CLIP ROUND THE EAR: SETTING LEVELS FOR DIGITAL RECORDING

The only golden rule when working with either DAT or digital multitrack is to use the meters on the tape machines themselves, and watch the peak levels, because in digital recording, it's the peak levels that count. The traditional notion of nominal operating levels isn't really relevant to digital, and whereas with analogue we aimed to get the meters bouncing around the 0dB point, with digital systems the only rule is: record the highest peak level you can, without allowing the machine to clip