There are hundreds of ways to record acoustic guitar and I have tried most of them over the past 25 years. This is my favorite and it gets me the best acoustic guitar recording.
Its a 3 microphone technique:
- One mic right by my ear with the diaphragm facing towards the acoustic guitar hole. This captures the sound exactly how i hear it. I like to use a LDC mic for this (LDC = large diaphragm condenser)
- The 2nd microphone will be placed by the 12th fret, about 2 to 6 inches away from it. I like to use a dynamic mic, like Sure SM57. Everyone has one of these mic's laying around. Not only do they make good hammers to nail your set list down to the stage floor, but they record great vocals, guitar and snare tracks as well.
- The 3rd Microphone will be placed by the sound hole, off axis about 6 to 12 inches away. I like to use an LDC mic for this as well.
1.) Play around with room position and make sure you find the best position in your room to get you the best sound. You do not want to be too close to any walls or you may get some unintended room reflections in your recording.
2.) You also need to check each microphone for phase issues before recording the acoustic guitar.
3.) Record each mic on its own mono track. Do not combine all the mics onto one track.
4.) When you are done recording, you will mix all 3 mono tracks until you get your desired acoustic guitar sound.
In alphabetical order:
Big - A huge and large sound that contains a wide range of frequencies withe good clarity. It contains sparkling highs and punchy lows. It can also contain large roomed reverbs and nice reverbing effects.
Cool - This changes with musical style. Its left up for interpretation. Music that has style and sophistication.
Dry - Without any reverbs or effects.
Edge - Upper frequencies that are abrasive when not used in moderation. These frequencies are from 3- 8kHz.
Lush - This is widely used as a reference for strings. It sounds very smooth and has a pleasing texture to it.
Moo - Rich sounding smooth and creamy lows.
Open - No compression. This sound has a wide dynamic range. It sounds natural and can be heard through and seen through.
Raunchy - A sound that does not include the very high and the very low frequencies. It can be described as a soulfully, gut wrenching performance.
Shimmer - Contains high frequency reverbs and decays.
Sizzle/Sparkle - These are the upper frequency sounds you hear form cymbals and bells. There are from 8 to 20kHz.
Squawk - Accentuation of the mid range frequencies. It can sound like a very small and cheap transistor radio.
Squashed - Compressed very heavily. This sound has a very low dynamic range.
Sweet- Lush and smooth sounding. Very pleasing to your ears. It can include some reverb, but not too much.
Syrupy - A very sweet sound. It will have a lot of reverberation and the music can be very predictable.
Thump - Low Frequencies that can be felt and heard. Thee between 80 to 150kHz.
Transparent - A broad range of frequencies, but the sound isn't capable of covering the sound around it. Silence can be heard through this sound.
Washy - A lot of reverb. It goes from one note to the other. Strings use this allot.
Wet - A sound that is close to 100% reverberation. It has none of the original sound. This term is used in a lot of other effects as well.
Between 75 and 85dB SPL. At this level your ears will not fatigue as fast, but you still need to take frequent brakes to refresh your ears. FYI: The best time to mix is in the morning, right after you wake up. Your ears will never be as fresh as they are when you first wake up.
What are the frequency range groups for audio ?
The Basic Categories include:
Highs - above 3.5kHz
Mids - between 250Hz and 3.5kHz
Lows - below 250Hz
More Specific Categories Include:
Brilliance - above 6kHz
Presence - 3.5 to 6kHz
Upper Mid range - 1.5 to 3.5kHz
Lower Mid range - 250Hz to 1.5kHz
Bass - 60 to 250Hz
Sub Bass - below 60Hz
A compressor is an automated volume control that turns loud parts of a music signal down when the VCA see's the signal exceeding a certain level that was set by the user. This enables you to bring the overall volume of the song up and in return, the softer sounds get louder. This enables you to print the entire track with a louder signal level. It's a fantastic tool for recording and mixing instruments with a wide dynamic range.
A limiter is a compressor that has its ratio settings greater or equal to 10:1. At a 10:1 ratio, that's when a compressor stops compressing and starts limiting. So anything below a ratio setting of 10:1 is compressing and anything above that ratio is limiting. That's why when you see many compressors, it also says limiter. The only difference is the ratio setting and when limiting, your ratio may be higher and your attack may be faster.
We need compressors for a wide variety of things. It protects against overly loud peaks that can clip the audio signal. It also evens out the volume changes in an instrument created by the artist playing it or singing it. There are also compressors that can control each frequency band individually. There called multi-band compressors. Multi-band compressors are used mainly for mastering, because of the control you get withe each band.
The 8 Parameters That A Compressor Can Have:
1. Threshold is the point where the compressor starts recognizing the signal's amplitude. When the amplitude rises above a certain point, it will start to act in a way defined by the attack time, release time and ratio settings. There are 2 ways that the threshold works. It can boost the signal up into the threshold or it can be moved down into the signal. In both the ways, the only part of the audio that gets processed is the part that goes above the threshold.. After the signal goes above the threshold, the VCA turns down the part that is above the threshold, leaving the rest of the signal unaffected.
2. Attack Time controls the amount of time it takes the compressor to turn down the signal after it passes the threshold. The attack time needs to be adjusted just right. If it is set to fast, then the compressor can turn down the transients and that can cause the instrument or song to lose its life. It can also effect vocals if its set to fast by making the 't" and 's' sounds disappear. The opposite can happen if the attack time is set to slow. It will exaggerate the 't' and 's' sounds, because it will pass through uncompressed because the attack time was too slow. So you have to be careful when setting up the compressor for vocals.
3. Release Time is the time it takes the compressor to let go or turn the affected signal back up when It gets below the threshold. Fast release times work very well with the fast attack times and slow release times work very well with slow attack times. Release settings are crucial, because if its set to fast, it can boost noise that is between the notes and if its set to low, it can compress a quieter note that follows the note the was above the threshold.
4. Output Level makes up for reduction of gain that the VCA causes. If the signal was reduced by 5dB, the output level can boost the signal back to its original level.
5. The Ratio determines how much compression is applied to a signal after it goes over the threshold setting. The ratio will tell how much gain reduction is going to be applied when the compressor starts to work.
An example: If you set a compressor's threshold at -10dB and a 3:1 ratio. If you have a some what constant audio signal the signal at -1dB it will become compressed so it ca only reach -7dB.
Ratio determines how extreme the VCA action will be. The ratio is a comparison between what goes through the threshold and the output of the VCA. The first number of the ratio will indicate the increase of how much dB will result in 1dB of increased output. The higher the ratio is, the more compression there is. If you adjust the threshold so the loudest note of the song passes the threshold by 3dB and the ratio is set to 3:1, the 3dB peak is reduced to a 1dB peak and the gain is then reduced by 2dB.
6. Hard Knee Compression and Soft Knee Compression:
The Knee setting tells the compressor how to react to a signal after it passes the threshold. The knee settings determine how fast and how severe the compressor reacts to the signal that crosses the threshold.
- With Soft Knee compression, the audio signal is gradually decreased throughout the first 4 to 6dB (give or take) of gain reduction when it passes the threshold.
- With Hard Knee compression the audio signal is reduced rapidly and very severely in amplitude.
- Both knee settings are dependent on the attack, release, ratio, and threshold settings.
- Soft knee compression/limiting is better suited for ratio's at a high setting.
- Hard knee compression/limiting is really good for when the audio has a lot of transient peaks. Hard knee is for the extreme and immediate response.
- Hard knee limiting is great when you need 100% absolute control of your peaks, like peak limiting
- Soft knee is more forgiving than hard knee compression/limiting
7. Peak and RMS Compressor Settings:
- Peak is more for limiting. RMS is more gentle than peak
- The RMS and Peak setting on a compressor/limiter will determine if it responds to the peak or RMS changes in amplitude.
- Peak stands for immediate amplitude of an audio signal
- RMS stands for Root Mean Square. It's the average amplitude of an audio signal
8. Gain Reduction:
Gain reduction is the amount that was turned down by the VCA after it crossed the threshold.
Gain reduction can usually be seen and measured with a VU meter or LEDs inside the compressor/limiter.
For a VU meter, 0 VU indicates no gain reduction.
For LED, there is usually a scale from right to left. Each LED can represent two or more dB of the audio signals gain reduction.
Effects From Using A Compressor:
When using a compressor during tracking (Use hardware compressors only, if you're recording into a program like pro tools or sonar) it will allow the engineer to record the track at a higher level than normal. Like if the compressor decreases your signal level by 4dB on the hottest parts, the entire tack can be recorded 4dB higher to make up the reduced gain. This will make the soft passages of the track hotter as the loud passages will be least effected. While the compressors really controls the loud passages, the final result is an increase in the soft passage levels.
A lead vocal track that is compressed will sound more up front in the mix, only when compression is used correctly.
Bass guitars are most always compressed. Because of the low frequencies it possesses. If not tamed, the low frequencies will saturate the overall mix level and this can make your song level artificially hot. When the bass guitar is compressed correctly, the low will be tamed and your mix can be mixed and mastered at a proper level.
The pumping effect is an effect of a compressor when the level control of reducing the gain as the amplitude passes the threshold. Then it turns it back up when the signal falls below the threshold.
The breathing effect is the same as the pumping effect, but the breathing effect is heard with high frequency airy sounds only.
- They divide each audio signal into multiple frequency ranges. Each frequency range (band) is processed separately.
- The bands that define each cross over point of each frequency range are most of the times adjustable.
- Each band is controlled separately by the user. Meaning you can have different threshold, ratio, attack, release, and gain settings for each band.
- There are usually 5 bands in a multi-band compressor. There are some with 4 bands as well.
- Multi-band compressors are mostly used in the mastering process, to fix the mixes problem areas.
EQ is used for 2 different things:
1. To boost (enhance) part of a tone we want
2. To cut part of a tone we do not want
There are 3 types of EQ
1. Semi-Parametric EQ, also called sweepable EQ:
- Each sweepable band has 2 controls. A frequency selector and a cut and boost.
- With this EQ, you can zero in on the exact frequency that needs to be cut or boosted.
- This EQ is good for finding the right frequency that brings life into your instrument, cause you can set a cut or boost and then sweep (dial in) the frequency that makes that track shine.
- The sweepable EQ in mixers will usually have 3 bands on each channel. They are lows, highs, and minds.
2. Parametric EQ:
- These are the most popular of the EQ's because of its flexibility.
- It works just like the semi-parametric EQ, but it has one more control.
- The Q stands for width and it controls the bandwidth of the cut and boost.
- With the addition of the Q, this EQ is the most precise of them all.
- The Q set at 1.0 means the cut or boost will affect 1 1/3 octaves from the centered band. If the Q is set at 3, it will boost or cut 3 octaves from the centered band.
3. Graphic EQ:
- This is the most visual of them all, hence the name.
- These EQ's comprise of sliders that represent the frequency spectrum (20Hz to 20kHz and sometimes they go beyond).
- These EQ's are more commonly used in live performances and for known tuning problems with your mixing environment.
- These EQ's can be used to cut and boost specific frequencies.
- They have a predetermined Q setting
- A 10 band graphic EQ has a Q (bandwidth) of one octave.
- A 31 band graphic EQ has a Q of 1/3 of an octave.
Different Kinds of Filters:
1. High Pass Filter:
- A high pass filter cuts the lows frequencies, as it lets the high frequencies pass through unaffected.
- You can specify the frequency at which the cut begins.
- The cutoff frequency is the frequency that you specified the cut to begin at.
- The rate of cut is called the slope and the slope is calibrated in dB per octave.
- Normal cuts are at a rate between 6 and 12dB per octave
- This filter is great for getting rid of the 60-cycle hum and the high pass filer is great for reducing background rumbles in your environment, like street noise, a/c that may bleed into a vocal mic or any other mic.
- Most high pass filters have a sweepable frequency selector and is superb on getting rid of any unwanted or unused low frequencies.
2. Low Pass Filter:
- The low pass filter cuts the high frequencies, as the low frequencies pass unaffected.
- Uses for this filter can be for getting rid of a high buzzing guitar amp, getting rid of string noise on a bass guitar and to help minimize leakage onto drum toms and cymbal tracks.
- They also use a sweepable frequency selector to define its cuts.
3. Bandpass Filter:
The bandpass filter lets the desired frequency range pass though unaffected. For example, all the frequencies below and above the desired frequency range will be filtered out and everything between will pass unaffected.
The bandpass filter is really the high pass and the low pass filter combined together as one.
4. Notch filter:
This filter is used to find and then get rid of any problem frequencies.
The notch filters have a narrow bandwidth and most of the times they are sweepable.
An example of use of this filter is to set up a peak or boost of a set frequency and then sweep the boosted or cut frequency until the problem sound goes away.
5. Peaking filter:
The peaking filter cuts or boost a band (frequency) in the shape of a bell curve. The peak that is made is the center defining frequency.
These filters are the most widely used by far.
6. Shelving EQ:
The shelving EQ will leave all the frequencies flat and then it will turn all frequencies above or below that point.
These filters are used for adding air to a mix by sweeping the cutoff frequency (from 10KHz and above) and then raise the shelf to your desired effect. A little goes a long way with this filter.
Q Setting Chart:
Q Setting Bandwidth Octave Calculations
0.7 = 2 Octaves
1.0 = 1 1/3 Octaves
1.4 = 1 Octave
2.8 = 1/2 Octave
Equalization is fundamental for getting your tracks to blend together.
Evaluate each and every frequency range in every single track and then make adjustments that builds a smooth and cohesive sound that blends well together.
Avoid cutting and boosting all your tracks at the same frequency range. You need to create EQ settings that work well together. If every track is boosted and cut at the same frequency range, your song will most likely sound very harsh and your tracks will be competing for the same frequency ranges. This will end up in instruments masking other instruments.
You need to make strategic cuts and boost on your tracks so instruments in the same frequency range do not mask each other. For example the Kick drum and the bass guitar both sit in the low end of the frequency spectrum. If you boost the kick drum at 65kHz, you should not boost your bass guitar at 65kHz. You should actually cut your bass guitar at 65kHz and boost it someplace else, like 250Hz. If you boost your bass guitar at 250Hz, then you need to cut your kick drum at 250Hz. Are you following the trend here? Practicing this technique will improve your mixes drastically.Complimentary EQ
Linear Phase EQ:
EQ's are made from filters that change the frequency of an audio signal. When an EQ filters frequencies within the range, the signal can be delayed a very small amount. These delays can cause phase issues with the audio signal. A linear phase EQ fixes that issue.
That's why linear phase EQ's are smooth and transparent EQ's.
For example: Chorus's and flanges work with changing the phase and they make beautiful effects. Those effects alter the phase (timing) of the path of an audio signal.
Wouldn't it suck if you or anyone else belted out that awesome vocal track, only to discover upon playback that there was some issues that screwed it up. Never let that happen!
The lead vocal track needs special attention so it can maintain its visibility and impact throughout the entire song. Because the vocal, in most cases, is the primary focal point of the song. The vocal needs to have and maintain constant space in the mix.
You need to determine what the best sounding signal path is for that specific vocalist:
This is the most time consuming overly repetitious task. But in the end, when you find the right gear and positioning that fits your vocalist. Your pay off will be priceless!
Mic placement is important in getting that great vocal sound:
There are 2 issues to consider.
1. The position of the mic in the room.
You need to find the best part of the room to record in. Not all corners, spaces and walls sound alike. This needs to be done way before the vocalist enters the room/studio by trial and error.
2. The other issue is the position of the mic, to the singer.
There are 2 factors to consider that effect the sound. One is the angle of the mic to the singer and the other is the space between the mic and the singer. Both are very important factors.
The Proximity Effect - The closer the singer is to the mic, the more bass frequencies are enhanced. This can be used as a tool, by having the singer move closer or farther away from the mic, depending on the mood of the vocal passage. The omnidirectional mic is the only mic not to have the proximity effect.
The mic of choice for most singer are a cardioid condenser mic and a good starting point for this mic is about six to eight inches away form the mic capsule. If the voice sounds to thin, then you move the singer up a bit to use the proximity effect. But be very careful. Moving only an inch or so will increase the bass and fullness allot. If the sound is too big, then move the singer back a bit. It's a balancing act.
If you're using an omnidirectional mic or an omnidirectional pattern setting, there will be no proximity effect. Moving the singer back and forth will only create distance and the bass frequencies will not be enhanced as the singer moves toward the mic.
The omni pattern is a good mic to use if the singer cannot stay still and/or is inexperienced in vocal recording. But this mic has its fall backs, since it picks up all directions equally. You need a very quiet room to use this mic and a room that is acoustically treated.
The effects of a condenser mic on axis and off axis with the singers mouth are very important. When a condenser mic is on axis to the singers mouth, the sound is harsher and brighter. When the mic is off axis to the singers mouth, the sound gets a bit warmer and darker. This is due to the sound hitting the mic capsule. The mic capsule captures the singers chest resonance and by changing the axis of the capsule, you change the sound that mic records. An off axis tilt towards the ceiling can help prevent popping and sibilant.
Now that you are aware that the slightest movements and the slightest change of positions can alter and change the sound dramatically. You should make notes on the distance and mic position relative to the singer, in case you need to Punch In.
Be aware of plosives. Plosives are loud and exaggerated sounds that occur with the letters P 's, B's T's and S's that are pronounced. Plosives are caused by a sudden rush of air from the singers mouth that hits the microphone capsule. A pop filter, along with mic techniques helps prevent the occurrence of plosives.
Have you ever wondered why you see Mic's hanging upside down? They do this so they can make room for lyric sheets and music stands.
Things that can ruin a vocal take:
- Jewelry (necklaces, bracelets) can make a lot of noise. If a singer cannot take them off due to some unknown reason, you can wrap a towel around it and put some tape on it. Just as long as it doesn't move.
- Early reflections from a music stand that is to close to the mic. Try to avoid the metal music stands, as they can cause early reflections more than a fold-able music stand. Yes! Cheaper is better when it comes to music stands. Save when you can and this is where you can save some money.
- Avoid wearing shirts with buttons and other things that could be noisy. A nice plain t-shirt is good.
- Always have water close and available for the singer. A dry mouth can cause lip smacking and other noises.
Record in 24bit. This goes for vocals and everything else:
- When recording in 24bit, there is no need to record hot. Recording hot could get you in trouble. One small clip can ruin your vocal take.
Record your vocals between -20dB and -6dB. Those levels are fine for 24bit
- With 16bit, you have 65,536 possible levels
- With 24bit, you have 16,777,216 levels
- So in 24bit, your audio has more room in the digital realm
- You do not have to record as hot in 24bit as you do in 16bit because of the noise floor. In 24bit you can record at a lower level while staying above the noise floor. Meaning you can record at lower levels because of the more headroom 24bit gives you.
Double tracking vocals:
- Over Pronounce your vowels when recording your vocals. This will give your performance more emotion. The consonants should take a non predominant role and let the vowels give shape to your words as you sing.This will lead to a great flow with the music your vocals are mixed in with.
- It will make the vocal part sound fuller and more powerful. This greatly depends on the singers skill on reproducing the exact vocal take that he/she performed before.
- During the 2nd take, you can change the singers distance from the mic. For example, if the singer was 7 inches away form the mic on its first take, then record the second take 14 inches away from the mic.
- You can even try a 3rd pass at it.
Tips for mixing vocal tracks:
- Exciter is a great tool that adds clarity to your vocal.
- EQ - If you used proper mic choice and technique, your vocals may not need any EQ. Except for maybe a high pass filter to cut the lows. Vocals, normally do not use anything below 60Hz to 100Hz. When using EQ on vocal tracks, try not to cut and boost dramatically, A little goes along way, especially if you want it to translate onto different sound systems. To add a bit of clarity to your vocals, try boosting between 4 - 5kHz
- Delay - A simple slap back delay can do wonders to your vocal track when set up in time with 8th note triplets.
Compression tips for vocals:
Inconsistent vocal levels - The settings for compression depends on how consistent the vocal track is. If the vocal track is inconsistent, you will need a fast attack time with a medium release time and a ratio setting of 6:1 to 10:1.
Your threshold is adjusted for gain reduction on the loudest parts only. So most parts will go threw the compressor unaffected. You only want to even out the volume level of the entire vocal track without doing any extreme compression.
Breathy vocal effect - This creates a whispery and highly present vocal. Set the attack time very fast, the release time should be moderate, the ratio should be between 5:1 - 10:1 and the threshold level should be between 7 to 21dB below the peak level. You'll also need to add a bit of reverb to achieve this effect. Note that you will definitely need to use a pop filter with these vocals, as the intense compression will overly exaggerate lip smack, breath sounds and other artifacts.
Smooth vocal effect - This one is easy. Set your ratio between 2:1 and 4:1 with a moderately fast attack time, a slow release time and the threshold set from 2 to 6dB below the highest peak level (like everything, adjust to taste). Since this is a very low compression you may have some high peaks that cannot be tamed. To solve this, run it into a limiter after the compressor with a fast attack and fast release time and set the threshold to limit only those pesky peaks.
A de-esser is just a fancy name for a frequency specific EQ/compression that reacts very quickly to audio signals with strong high frequencies. The frequencies include the letters "s", "t", and "k"'s.
- The de-esser is used to get rid of overly exaggerated transients that are caused by overly compressing or poor mic techniques.
- The de-esser is good for getting rid of these transients, since it reacts very fast to them.
- A de-esser works in the frequency range between 3kHz to 6 kHz. Some can be set to work below and above those ranges.
- Compressors can be set up and used as a de-esser. Simply set the attack to a fast msec time and then patch the side chain as the trigger for the processor. Then you adjust the threshold so the gain reduction starts when there is a transient problem.
There both in the same family as a compressor and they have the same controls as a compressor (threshold, ratio, attack, release, and output.)
- A gate opens and closes when the signal passes across the threshold.
- The VCA in a Gate/Expander will turn everything down below the threshold and the VCA in a compressor will turn everything down above the threshold.
- When the gate closes behind the sound, the gate doesn't open back up until the audio signal is above the threshold.
- Gates are good for getting rid of ambient room noise. For example, a noisy electric guitar.
- Gates can be used as effects. They are commonly used on drum tracks to give it that 80's Phil Collins snare drum sound (Just a reverb and a gate.)
- An expander, expands the dynamic range. It makes a bigger difference between the softer and louder parts by turning the softer parts down.
- The range can be adjusted so the VCA will only turn the signal down part of the way when it gets below the threshold
- An expander will turn the noise down, rather than turning it off, like a gate does.
- Expanders are smoother in their level changes.
There are 2 kinds of expanders, upward and downward expanders
1. Upward expanders are not common and they tend to be too noisy.
2. Downward expanders are the most commonly used.
Expanders are good for restoring dynamic range to a signal that has been overly compressed too much
To sum this up, a gate and an expander are mostly the same tool, but the gate turns the soft parts off and the expander turns the soft parts down.
There are 3 terms when talking about balanced and unbalanced:
- Lead: Its just another term for wire.
- Hot Lead: It's the wire that carries the sound from the magnetic pickup to the amplifiers input.
- Braided shield: This surrounds the wire and shields it from electrostatic noises and other interference's by diffusing, absorbing or rejecting it.
In General, most guitars are unbalanced and most low impedance mics are balanced.
Unbalanced cables (often called line cables) like for guitars and keyboards, contain one hot lead to carry its instrument signal and that hot lead is surrounded by the braided wire shield if the cable is shorter than 20 feet. If the cable is longer, it can act like an antenna and can and will pick up transmissions. Those transmissions will be carried by the hot lead.
Balanced cables can be longer than 1,000 feet without picking up electrostatic interference or the addition of noise. 3 point connectors are used for balanced cables. You have a neutral, hot and ground (shield) pin.
- Most balanced cables have 2 separate leads twisted together though out the cable. Both of the leads carry the audio signal and connect to the ground and the braided shield connector.
- XLR and the 1/4 tip ring sleeve plug are your most common balanced cable connectors
Reverb (digital) is the simulation of sound in an acoustical space (environment). For example, halls, rooms, bathrooms, blues club, arena and all other acoustical environments have a different sound to them. You cannot find 2 rooms that sound alike.
Reflections are the sound that bounces back from all the surfaces in a specific room and then goes to the microphone or to the listeners' ear. Those bounced sounds are called reflections.
The combination of the direct sound and the reflections in a room creates the distinct tonal character for every acoustical space.
Each reflection acts like a single delay. When you take may reflections from the same room, it creates the reverberation for that specific room. So a simple delay that is set to regenerate many times over can act like a reverb.
The reverb takes onto its own when you have thousands of reflections bouncing off every surface in a room and then coming back to you or the mic. With so many reflections, your unable to distinguish between an individual reflection and all your hear is the specific room reflection.
Different Sounds In Reverbs:
Room Reverb settings mimic the many types of rooms that are smaller than chamber and hall sounds.
Plate Reverb imitates an actual plate reverb. Plate verbs are the brightest sounding of all reverbs .A real plate reverb is made form an actual sheet of metal that is suspended in a box. Then you attach a speaker to the plate and this makes the plate vibrate and it gives the plate reverb effect. Its easy and fun to make your own plate reverb. You should try it some day.
Hall Reverb are sounds from a concert hall. They tend to be the richest and smoothest sounding reverbs. They consist of long delay times that blend together for that smooth decay. Hall reverbs usually have a decay of over 2 seconds.
Chamber Reverb imitates an echo chamber (acoustic reverberation chamber). These chambers consist of large rooms with hard surfaces. The chamber sound is made when you play music into the room with some hi-fidelity speakers. Then you place a mic in that room and the mic is then patched into the mixer's effects return. The sound of the chamber reverb is like the hall reverb, but the chamber has more mid and high frequency sounds.
Reverse Reverb is just a backwards reverb. It turns around the reverb when the sound stops swelling
Gated Reverb makes a sound for a period of time that is defined by the user and then it stops very quickly. This creates a very large sound that doesn't override the mix. Are you thinking of Phil Collins right now? He was known to use a gated reverb back in his time.
Spring Reverb is a combination of electrical and mechanical devices that use the sound properties of a metal spring that imitates reverberation.
Parameters of the Reverb:
- Pre Delay is the delay in time that happens before you hear the reverb. Its the sound we hear dry (without reverb) for a period of time and then the reverb starts to come along after the defined period of time. This can make the sound more up front, while adding richness and filling in the holes. Pre Delay setting can be from a few milliseconds to one or 2 seconds long
- Diffusion controls the space between the reflections.
- Decay Time is the time it takes for the reverb to fade away. Normal decay time can be from 1/10 of a second to upwards of 99 seconds.
- Density controls the initial short delay times. Low density settings are good for strings. Anything that needs to sound smooth. High density settings work great on drums and percussion sounds.
- Wet/Dry percentage is exactly what it says. This control the amount of the processed (wet) signal and the amount of the unaffected (dry) signal. If you set the reverb on a bus, then in most cases you will have it set to 100% wet, because you control the amount of the processed signal with the send level.
Impulse response reverbs:
They accurately simulate the sampled acoustics of real spaces. For example, halls, rooms, chambers and just about any other room you can think of.
It models an acoustic environment in the digital domain and this modeled sound is called an impulse response.
Its made by firing a starter pistol or by playing a sine wave from a speaker into the room its simulating. The decay from the reverberation is then recorded into a digital audio file. This can then be used to re-create the acoustics of any actual space. That's cool how they make it!
The distance between each speakers should be the same distance as your ears are to them. You and your speakers should form an equal distance, that forms a triangle. For close field monitoring, the speakers should be about 3 feet, give or take for speaker size and what is more doable ergonomically.
The center of the triangle should be of equal distance from each wall.
At times, this may still not be optimal for your room. Room acoustics play a big role in creating and reducing problems like the sound being to boomy, bass, or muddy. Your high or low end may be too loud or to soft. So you may need to move your speaker placement around a bit, before you settle on the right location.
These are suggestions that can yield you a huge fat guitar sound:
You first need to start with a decent sounding guitar tone. If the distortion sounds thin and buzzy, then you need to fix that first and foremost. It's much easier to get a big guitar sound form a sound source that sounds good to begin with. Crap in equals crap out. No matter what you do with it.
The use of short delays is good for widening up the guitar sound across the stereo fields. This technique is very effective.
Your tone needs to be very broad with extended high and low frequencies. This is a must for a huge sound.
Your depth is very important. Long delays and reverb can make the guitar sound like its being listened to in a large room. A slap back delay defines the size of the room that the guitar is placed in. For example, a delay of 500ms will create the illusion that the guitar is being listened to in a space that is 500 feet long. This is because sound travels at a speed of one foot per millisecond.
Compression is very important. I don't have a setting, because you need to use your ears for this. Each guitar/instrument/song will need different settings. But remember that compression helps keep the guitar consistent in the mix space. So use compression for this.
Try low tuning your guitar. Record one take with standard tuning and one take with low tuning and combine them both for one huge guitar sound.
Double, triple, quadruple track guitar takes. Pan them far left and far right. This is by far the best way to achieve a huge guitar sound.
You can also clone/copy the guitar track onto a new track and then transpose then entire track down an octave and combine both tracks for a huge sound.
This is my favorite miking technique to get that big, thick, and chunky guitar sound: (Note: You need 2 Mic's for this technique). This technique will put the "power" in your power chords.
- Place one mic close to your amp's speaker and compress that signal with an 8:1 ratio settings, a fast attack, a semi fast release and a threshold of 6 to 20dB below the highest peak of the audio level. This high compression will cause your guitar sound to pump.
- Place the 2nd mic and place it 5 to 9 feet away (room mic) from the amp's speaker.
- Compress this signal with a ratio of 5:1, a medium attack, a slow release, and the threshold is the same as the other one, between 6 to 20dB below the highest peak.
- Combine both sounds together and using the room mic just enough to give it that thick and chunky sound.
Bass guitars either have an active pick up or a passive pick up. If your bass has active pick ups, then you can usually plug directly into the input of your sound card/interface. If your bass has passive pick ups (the most common), you need to have some sort of DI box or an external amp simulator, like a bass pod. These DI boxes take the low level signal of your bass and raise it to a line level. If your sound card/interface has mic preamps, you can use that as your DI box.
If you record direct, without the use of an external amp simulator, you will need to edit the sound with a bass amp simulator, compression, EQ, and maybe a bass chorus, to make it sound warm, full and alive.
The best and most consistent results come from close miking a bass amp cabinet that is just off center a tad bit. You can and should also add a 2nd mic and set it about 4 feet back. Good Mic's to use are the AKG 414 and a sennheiser 421.
Compression is needed for bass guitars because each string produces different dynamics and the dynamic range can get pretty big. Compression is used to smooth out that dynamic range so the bass track has that sonic backbone most songs desire.
To tighten up the low-end, set the ratio to 2:1 to 4:1, with an attack between 5ms to 20ms and a release between 120 and 250ms and a threshold between -5 and -10dB. Set the output to make up for the gain that was reduced.
Valve amplifiers are known for some of the best bass sounds and these can get expensive for a home studio budget. So adding a Tape simulator or some slight distortion from an amp simulator is a great idea. There are also valve DI boxes and using one of those is a great tool for beefing up your bass sound without totally distorting it.
Combining DI and Mic Recording:
This is by far the best way, cause you have the option to use blend both signals into one huge one. The only worry is that the phase may be off between the DI and the mic'd bass. So you may need to reverse the phase on one of the sound sources.
The fundamental bass frequencies are between 125 to 400Hz and boosting these can bring out more of the bass lines in the mix.
The harmonics for the bass are from 1.5 to 3kHz. Boosting these frequencies will increase the clarity and pluck.
Boosting between 5 to 7kHz will increase the finger sound.
Cutting between 40 and 50Hz will reduce the boom.
Playing with a pick can add harmonics up to 4kHz and will make the bass sound brighter. Playing with your fingers will produce a more mellow sound
Remember to never boost or cut the same frequencies for the bass guitar and kick drum. If you boost the bass guitar at 100Hz, 250Hz and 3kHz, do not boost the kick drum in those same frequency ranges. If anything, you should cut those same frequency ranges.
What is Latency:
- It's the time difference between input and output of any digital audio workstation. It is cause by mathematical/algorithmic issues and by mechanical/physical procedures that occur mostly in software A/D and D/A converters, and when hard drives are used.
- Latency literally means the build up of delays in an audio signal as it passes through the audio interface.
- Its measured in milliseconds.
- There is input latency, output latency and round trip latency.
How do you experience Latency:
- You get latency when you monitor an audio signal through a computers signal chain. If you ever heard a delay sound when triggering a synth with a midi controller, you actually experienced latency.
- You can also get latency form using effects (VST / DX's) with hidden buffers. These effects are CPU intensive and usually meant to be used in mastering stage of a project.
- You experience latency if your ASIO buffers are set to high or your WDM latency is set to high.
How to solve Latency:
- Your round trip latency should be less than 11 milliseconds, if you do not want to experience latency.
- In ASIO driver mode, make sure your ASIO buffers are at its lowest settings. A setting of 32, 64, 128, or 192 should be acceptable. The Lower the better for this setting.
- In WDM driver mode, make sure you slide the latency slider all the Way to the left. A millisecond settings of 5ms or less should be acceptable.
- Make sure you go to your audio interface/sound card manufactures website and download & install the latest drivers for your operating system.
- Try both driver modes to see what works best for you and your PC.
Zero Latency Monitoring:
These days, many recording platforms and audio interfaces offer zero latency monitoring. This means that your audio signal is sent to your main outs and/or headphone outs during recording is split from the input audio signal, before the digital conversion takes place. Basically, before it enters the computer.
This means that you will not hear any software effects when you're recording using this method.
Also, some recording systems will allow for both latent and non latent signals to be heard at the same time. You need to make sure that this doesn't happen.
Any of these manufactures are good. Just get one that best fits your needs and is within your budget. Also make sure they have drivers for your operating system.
These are my favorites: RME, MOTU, Lynx, Focusrite, Apogee. Antelope, Audient
- MOTU 16A 32x32 Thunderbolt / USB 2.0 Audio Interface with AVB
- RME Fireface UFX+ USB 3.0/Thunderbolt Audio Interface
- Apogee Ensemble Thunderbolt Audio Interface with Remote
- Lynx Aurora (n) 16-TB
- Focusrite Clarett 8Pre X 26x28 Thunderbolt Audio Interface
- Antelope Audio Discrete 8 Microphone Preamp and Thunderbolt/USB Interface
- Audient iD44
MIDI stands for Musical Instrument Digital Interface. MIDI is a protocol that enables electronic instruments to communicate, control and sync to one another. MIDI's first claim to fame was that it allowed you to play multiple synths using just one keyboard controller. It became very popular and became the industry standard. The perfectly timed robotic sequences that midi creates and can sync to drum machines helped create the sound of the 80's (god help us).
The midi signal doesn't carry any audio data. MIDI carries specific details of events that relate to notes. The information that is carried can control the type of instrument, pitch, duration, volume, attack, decay, etc. that is specified in the midi. Each midi channel corresponds to a different instrument or voice.
Midi has a defined list of sounds/patches. Its called General MIDI (GM). GM has a standard set of 128 sounds. General MIDI doesn't define the way the sound will be reproduced. It only names that sound. Meaning that each manufacturer can provide their sounds that is an acceptable representation of the data written for general MIDI. MIDI contains 16 channels and of those 16 channels, only channel 10 is reserved for percussion or drum sounds.
What Is An IRQ?
- IRQ Stands For Interrupt Request
- The IRQ have channels that are numbered and the devices use these channels to get the processors attention
Symptoms of IRQ Conflicts:
- Crackling or other artifacts when playing back a project
- Your PC wont boot up
- Your PC will lock Up
- Corrupt files when transferring
- Not being able to browse your network
Causes of IRQ Conflicts
They happen when more than one device shares IRQs
IRQ conflicts can happen when you install new hardware or reconfigure hardware
How To Detect and Fix IRQ Conflicts:
- In Windows Click Start — Control Panel — System
- Click Hardware — Device Manager
- Click View — Resource from drop down menu
- Click the expansion box next to the IRQ icon. This will display a list of IRQ numbers assigned to them and a list of system devices.
- Right-click each device that has a conflict and select properties. When your in the properties window, click the resource tab to see if it has conflicting device has a reserved IRQ. If the option is grayed out or unavailable, then you cannot change the IRQ and reassign it to another one. If its available, then you can re-assign it to a new available IRQ, If there is one.
Resolving IRQ conflicts with PCI cards or ISA cards:
- Manually move them to other available slots in your motherboard.
- By changing the slot, you change the IRQ channel.
Recording The Snare Drum:
- First, make sure your snare drums tuning pegs are tuned correctly. It's usually the drummers call. He will know when it sounds and feels right. Your snare drum also has different sounds to it, depending on the location being hit by the drum stick. If your drummer is sloppy, take it into consideration and hit the snare head in all the different locations to check its sound.
- As far as Mic's go, there are many to chose from, but for this discussion, I'm suggesting the good ole SM-57 for the top of the snare head. Place it form a few centimeters to an inch above the edge of the snare head. You can get away with using just one mic, but why settle for good sound when you can have great sound! So the bottom of the snare head must be Mic'd. Chose a mic that is good at picking up the mid-high to high frequencies, like an AKG 451B small diaphragm mic. Due to the small size of this mic, its a great fit under your snare drum.
- When using 2 mics for the snare drum or any other instrument, you need to check the phase. If the Mic's are out of phase, you can try moving the position of one of the microphones to get both microphones in phase with each other. You may need to reverse the phase of one of the Mic's if you cannot get both Mic's in phase with each other.
These two Mic's together are a great match because the SM-57 is great for the low-mids to mid range and the AKG is great for picking up the mid-high to high frequencies. Its a match made in "snare drum heaven".
EQ For The Snare Drum: (Note: These are just suggestions and guidelines, as nothing is written in stone. You must use your ears, as each song will need different EQ settings)
- Try using a high-pass filter set at 120Hz and under. 120Hz is a great starting point and then just slide the filter downward for desired cut
- Boost between 150 - 300Hz. This will fatten the snare drum up for you
- Try cutting around 400 - 900Hz to eliminate some boxiness in the low end
- Boost between 5 - 7kHz for a crispness
- A boost between 9 - 15kHz will add some nice brightness to the snare. Just make sure it doesn't interfere with the vocals in that range
Peak is the highest dB point of a continuous audio signal and RMS meaning root mean square, is the average dB level of an continuous audio signal. The Peak is usually twice the amount of the RMS.
To get a solid kick sound, you need to use a large diaphragm mic. An example of a good kick drum mics is the AKG D112, the Sennheiser MD 421 and an Electro-Voice RE20. There are many microphones made for recording the kick drum and each mic will have its own flavor and will favor different frequencies over others. For example, if you want that "tick" sound of the kick drum petal beater hitting the head, you would choose a mic that favors the lower mid range and the upper frequencies as well.
- For a Tight Drum: (NOTE: There are countless ways to mic a kick drum)
This technique does not use 2 heads. Remove the outer head and pace the mic inside the kick drum and position it at the batter head. The distance from the batter head depends on the sound you are going for. The closer you are to the batter head, the more impact and less resonance sound you get form that mic. Inserting a blanket or a foam pad inside the bottom of the kick drum will help dampen unwanted reflections. A good starting point for mic placement is about 5 inches inside the kick drum, slightly tilted towards the floor tom.
- For a Large and Live Kick Drum:
To get large and live kick drum sounds, you should use 2 kick drum heads. The sound will sound more resonant with rich overtones. You may want to put a strip of cloth across both heads. This helps dampen overtones without destroying the boominess of the kick drum. Try placing your mic about 2 feet in front of the kick drum and have it point to the center of the head. You'll need to turn the pad setting on for the mic, or you may overload your signal.
Some drummers have a front head with a hole that they cut out or bought. If this is the case, you can position the mic slightly inside the hole or aimed into the hole. Note, that having a hole in the outer head can cause an annoying ring to it. To fix this annoying issue, you can lay a foam pad or blanket against a section of the front head to remove the annoying ring without effecting the overall sound.
Signal Processing Tips:
You can compress the kick drum during the recording phase, but you do not have too. You can do all this after its recorded into your DAW program of choice. When I process the signal during recording, I will generally use a compressor, EQ, and a noise gate. I'll compress with a threshold set to around -10dB below the highest peak with a moderate to fast attack and moderate release with a ratio set to 2:1. Then I'll boost around 100Hz to taste and then I'll run it through a noise gate, with the gate set up to close its gate after a few milliseconds after the kick sound. This will make the kick drum very defined and very stimulating. As I said before, you can do all that after its recorded using the program of your choice.
Some General EQ Tips:
- Less Boominess - Cut around 80Hz
- More Boominess - Boost around 80Hz
- More Thud - Boost around 1kHz
- More Click -Boost around 12kHz
- More Attack - Boost 5kHz
- Less Wallop - cut around 220Hz
- More Wallop - Boost around 220Hz
Kick Drum For Rock Music
- Use a double headed kick drum
- If the kick is too boomy, try cutting 2 to 6dB at 80Hz and then another cut with 5 to 12dB at 200Hz. Use your ears to determine the amount of Db to cut
- After these cuts are made, you should not hear any boominess anymore. In fact you should be hearing allot more of the head and way less resonance
- To add more "flap" boost between 2 to 10dB at 500Hz
- You can also boost a bit at 12kHz with a peak or better yet a shelving filter
A Long Tone kick Drum:
This is for simulating the sound form the famed Roland TR-808 Long Kick. I strongly suggest you turn your monitors down a bit for these tweaks :)
- Boost between 8 to 10dB at 80Hz
- Cut between 8 to 10 dB at 300Hz
- Cut between 8 to 10dB at 1kHz
- Boost at 12kHz until it sounds good to your ears
- Add a hall reverb with medium pre-delay and a long decay for sustain. Perfect space IR reverb has a kick and snare preset. I suggest trying one of the kick presets and tweaking it to taste. If you don't have Perfect Space, any reverb that contains hall sounds is good
Funky Funk Kick Drum:
- Cut between 5 to 7dB at 80Hz to get rid of the boominess. The more boominess you have, the more dB you should cut at 80Hz and the less boominess you have, the less dB you should cut at 80Hz.
- Boost between 5 to 7dB at 350Hz
- Boost between 3 to 6dB at 3kHz
- Boost a bit at 12kHz using a peak filter or shelving filter. The dB boosted is just to add a bit of presence to it, so not too much boosting
- You are going to need to compress with a medium attack, slow release, a threshold set to taste and the ratio set to around 4:1
Aux buses and sends include a switch or button that decides if the audio signal goes to the bus and then the track (Pre) or if the signal goes to the bus, after the track (post). So the pre setting is before the fader and the post setting is after the fader.
- Pre fader lets you set a mix up that's totally separate from the input faders and the effects on the bus. Some people use pre for headphones mixes because you can get 2 separate mixes form it. Pre send is not good for effect buses. If pre is used for effect bus's, the track fader can be completely off, but the send level is still on and therefore the effect can still be heard.
- Post fader is great for effect sends, because the effect decreases as the track fader is turned down. As the fader is decreased, the send level is also decreased. Therefore the effects on the bus maintain a constant balance between the dry and wet effected audio signal.
Gain Staging (structure) refers to the signal level as it moves form its source to its final destination. Along this path, you can have points where signal level changes can be made. Monitoring the strength in each point is a must! You do not want to clip or over saturate a gain stage.
An example of a gain stage:
Voice to Microphone to Preamp to Compressor to Sound Card. With this normal vocal recording chain, you have 5 possible ways to change and alter the signal strength. The strength of your voice and the positioning of the mic to your mouth is a gain stage. The mic is also a gain stage, because it can have pad settings on it. All these components affect the signal strength.
You have to experiment with many approaches to find what works best for you and your set-up. Note, that not every approach will work for every situation. You need to trust your ears. If you're getting a great sound and your settings don't look right, that's OK, as there are no rules to gain staging. Over time you will build confidence when setting up gain structures. Most important is to use your ears and do not clip your audio signal.
Things To Watch Out For:
- If your input level is too high, The track fader may have to be set very low. This can make it too low to have control in the upper part. When your track fader is low, its very difficult to adjust and fine tune the audio levels.
- If your preamp level is too high, the signal can overdrive the sound card's input or the next gain stage in the signal path. Preamp settings are the upmost important. A bad preamp setting will result in failure
- If your preamp level is too low, your track fader will have to be too high and you can get a bad signal to noise ratio.
As you may already know, each microphone has its own distinct characteristics. How do you decide what mic to use for a particular sound source? It's a choice you have to make and there is no wrong and right way to pick one. The general rule of thumb is, if it sounds good, use it. With that in mind, every microphone has its mechanical limits. For example, the maximum volume it can handle before it starts distorting or even worse, it gets damaged.
SPL (sound pressure level) indicates that maximum volume. It can be found in the microphones spec sheet. You can ruin an expensive mic by hitting it to hard and blowing and moving the diaphragm. Dynamic mics seldom have a max SPL rating. Condenser mics come with SPL rating because their built with electronic circuitry that can overload and that overload can cause audible distortion.
When you're trying to find the correct placement of your mic, you need to factor in the sound of the room. Putting the mic closer to the instrument, will diminish the sound of the room (environmental interference). This kind of Microphone technique is called close Micing or tight Micing. A good technique is to place your ear directly to where the microphone is at and this will give you a perspective of what the mic hears and in return records..
Each and every mic was built with a specific application intended for it. The characteristics of a mic help you decide what to use it for. Like the diaphragm size, the pick-up pattern and the frequency response. All those things along with understanding of their specifications will affect your mic choice.
Omnidirectional is a mic pattern that picks up all directions equally. It doesn't reject sound form any angle. For this characteristic, the omnidirectional mics are great for capturing room ambiance and groups of instruments. Its great for picking up sound from a distance. These mics are not for live use as they can produce feedback more easily than any other pick-up pattern.
Bidirectional is a mic pattern that doesn't hear form the edges, but it hears equally form both sides. This mic is a great choice for recording two sound sources into one track by positioning the mic between the two sound sources. Another name for this is called the figure-eight pattern.
Unidirectional and often called cardioid pickup or directional has a heart shaped pattern with it most sensitive part being the part you sing into, facing the mic capsule. This mic is great for isolating sounds. Its great for when you're recording with a group of people. Because when you point the mic at one instrument, it will pick up less to none of the sounds from the other instruments in the opposite direction. The disadvantage of using this mic is you need to be up close to get the full sound. After a 12 inches or more, your sound will get very thin compared to the sound you're recording with the mic. This mic is great for live sound as it produces way less feedback than other microphone patterns, such as the omni and the bidirectional.
The Five Directional Pickup Patterns:
Cardioid has full response at the front of the mic. It decreases in sensitivity of around 25 to 30dB at 180 degrees off axis. This has the heart shaped pickup pattern and like I said before, its great for recording a single sound source when you have many sound sources in the same room. That's due to its pickup pattern. Its more sensitive in the front and not so sensitive in the back or anywhere else.
is more directional up front than a cardioid pattern. It has a decreased sensitivity of around 170 degrees on the sides of the mic.
Hypercardioid has a very high degree of up front direction. It decreases around 10 to 14 dB on the sides and is less sensitive at 110 degrees off axis.
Ultra Cardioid has a very focused and directional pattern in the front. It also has a very small area of sensitivity at 90 degrees and at 180 degrees.
Subcardioid has a much wider and extends more out front than the cardioid pattern. This pattern is close to a non directional- omnidirectional mic.
The Three Basic Categories of Microphones:
1. Condenser Microphones are the most accurate of the three. They are more precise in responding to fast attacks and transients than any other microphone and usually adds less tonal coloration than the other ones. Condenser mics can be a large diaphragm or a small diaphragm. You use a condenser mic when you need to capture the purest sound of a voice or instrument. Condenser mics need phantom power.
Here are some popular condenser mics
- AKG 451,353, C1000, C3000, and C-12
- Neumann U47, U67, U87, U89, KM83 and KM84
- Shure KSM 27, KSM 32, KSM 44, KSM 141, and SM 82
- Sennheiser MKH 40 and MKH 80
- Audio-Technica 4033, 4041, and 4047
- Blue Microphones Cactus, Mouse, Dragon, Kiwi and Bluebird
In Omni pattern, the condenser mic will capture a more precise broad range of frequencies at a greater distance than the other two mics. This trait is the reason the condenser mic is widely used in the recording studio, because it can capture the sound source and some of the room ambiance. The further the sound source, the more natural room ambiance it picks up.
2. Ribbon Microphones by far are the most fragile of all the mics and for this reason alone they are the least popular choice for live use. The capsule in the ribbon mics are bidirectional. The front and back are equally sensitive as the sound from the 90 degrees off axis cancels out. If a ribbon mic has its back enclosed and it becomes unidirectional. Ribbon mics have a characteristic of having a warm and smooth sound with close mic recording. When used at a distance, these mics sound thin. Do not drop a ribbon mic. Handle them with extreme care
Here are some popular ribbon mics
- Royer SF-12 and SF -24
- RCA 77-DX and 44-BY
- Beyer M160 and M500
Ribbon mics require a solid preamp. Ribbon mics have a lower output than condenser and moving coil (dynamic) mics. The preamp needs to give you at least 60 to 65dB of pristine gain to bring out the mics better qualities. That's why not all preamps should be used with ribbon mics. Active ribbon mics can use phantom power, but passive ribbon mics can be damaged by the use of phantom power.
3. Moving Coil Microphones (Dynamic Microphones) got their name because they are made with a movable induction coil. This coil is inserted in the magnetic field of a magnet that is attached to the diaphragm. Moving coil mics do not really do good in capturing transients, but they are the most durable mic out of the three types. As for being the most durable, they are can also endure the most volume before they start to distort the audio signal. With that said, this mic are optimal for live use cause they tend not to feedback as easily as the other 2 types.
Moving coil mics color the sound more than a condenser mic. The frequencies affected by this coloration are generally between 5kHz and 10kHz. That frequency range is known to add edge, clarity and presence to its sound source, like vocals and guitar. When placed more than a foot away from the sound source, these mics will have a thin sound. With that said, these mics should be used for close mic-ing situations.
Here are some popular moving coil mics
- Shure SM7, SM57 and SM58
- Sennheiser 441 AKG D12, D112, and D1000E
- Beyer M88
Complimentary EQ techniques involves the cutting and boosting of frequencies in specific tracks. I use these techniques in all my mixing projects. You should use these techniques for any instruments that are in the same frequency range. Like Bass Guitar and Kick Drums and Guitars and Piano. The Bass Guitar and Kick Drum are in the same frequency range and so is the Guitar and Piano. Both of these instruments tend to mask each other in a mix.
Complimentary EQ techniques will make it so you can hear both of these instruments clearly in a mix, by cutting unwanted & unused frequencies, and boosting certain key frequencies that where cut in the other tracks.
I will use the Bass Guitar and Kick Drum for this example, since I always see questions about these two. Let's say I boost the kick at 65Hz and 2kHz and then cut at 250Hz. Now since I boosted those 2 frequencies for the kick drum track, I will cut those 2 frequencies in the bass guitar track and then boost the bass guitar at 250Hz.
I did the complete opposite in both tracks. If I boost 65Hz in the kick drum track, I will cut 65Hz in the bass guitar track. If I cut 250Hz in the kick drum track, I will boost 250Hz in the bass guitar track. Note: these frequencies I'm cutting and boosting are only examples and these may not be the frequencies you should cut and boost in your mixes. Each mix will be different.
You also cut out all the frequencies in each instrument that is not needed
A vocal track normally doesn't use any frequencies below 80Hz to 100Hz, so you can set a high pass filter so it cuts everything below 80Hz to100Hz.
You do this for every single track in your project and you will have a clearer mix that is less muddy and a mix that you will be able to make hotter, since all the unwanted frequencies are cut out of it.
The Sample rate is the number of times per second a recording platform either digitizes the incoming sound or converts the digital sound back to an analog signal.
The standard sample rate for CD format is 44.1kHz. (44,100 samples per second) If you are recording at 48, 88.2, 96 or 192kHz, you must convert your sample rate to 44.1kHz if you want to burn your song to CD.
The standard sample rate for video is 48kHz (48,000 samples per second)
The greater the sample rate, the more accurate your recording program will capture your sound. But if your target media is going to be CD and MP3, then recording at 44.1 or 48kHz is suggested.
According to the Nyquist Theorem, the highest frequency a system can handle is equal to half its sample rate.
The Bit Depth will determine how many values are available to describe the amplitude level of an audio signal being recorded at any given moment. The number of bits determines the resolution of each and every sample. The more bits, the greater the resolution and the better the sound quality.
Just like the sample rate, the more bits used to capture a sound, the more accurately the sound will be represented.
Anything under 16bits is not considered professional. CD standard is 16bits 44.1kHz..
The recommended bit depth to record at is 24bits. Recording in 24bits will give you 256 times the resolution of 16bit recording. Even when you convert the 24bits to 16bits for CD burning, you will not lose the effects of originally recording in 24bit.
To convert from 24bit to 16bit, you need to apply dither to it.
Floating Point Bit Depths:
32 bit floating point processing means that after you record your audio at 24bit, you can convert it to 32bit floating point. This means that it will add the extra bits after it is recorded. These extra bits that get added onto the file after its recorded, will give you more headroom for processing (audio calculations).. I always convert all my files to 32bit floating point when working with my clients audio.
Having 32 bits, rather than 24bits is going to render a more accurate result. Some recording platforms do not have this option yet.
Dither is just low level noise that gets added to a digital audio signal. This low level noise helps mask and get rid of quantization errors.
Dithering should only be applied at the very last stage of the mix, when you're going form a higher bit depth, to a lower bit depth. Like going form 24bit to 16bit. This goes for fixed point and floating point bit depths. If you are in 16bit and you have a plugin that processes your audio signal with a 64bit floating point engine, you need to dither upon mix down of that track or mix.
You do not dither when you go from a lower bit depth to a higher bit depth. You only need to dither when going down in bit depths. So, if you go from 24bit to 32bit floating point, you do not need to dither.
All audio devices are made to convert and /or output samples at a high rate of speed. A standard CD player must read and output nearly 85,000,000 bits of data per minute. When you digitally link 2 or more devices together, the smallest of discrepancies in each of their sample rates can and will cause major audio problems.
To fix this problem with multiple audio devices, you should have a single master clock that uses a specialized timing signal called word clock. This word clock contains no audio data or any other types of markers. Its just a timing reference, that makes sure each piece of gear moves processes audio at the same speed.
Included in this section are some of my tips that I want to share with you on recording a choir.
Planning Ahead: Make sure you have enough music stands and when you get a lot of people in one area, the environment will heat up quickly, so turn down your A/C so the choir singers do not get uncomfortable. You want them as comfortable as possible. You also should have a lot of bottled water for them. The water should be kept at room temperature. Its better for them, then cold water.
Monitoring: If the music is pre-recorded, an FM transmitter can be fed by your headphone mix and then this is fed to your FM transmitter. The choir would plug their headphones into their portable radio and dial into the correct frequency. I think this is the cheapest way to have a large group of people hear a pre-recorded mix. It cost between $100 and $200. All you need to do is ask each choir member to bring their own portable FM radio.
Arranging Your Singers: You should separate the altos, sopranos, tenors and basses into separate sections and then spread them out left to right. If the choir has a conductor, it will be his job to position the choir members
Mic Placement: The most important and hardest thing is having your mics at equal distance from each other. Microphone placement is crucial as you don't want to hear one voice more than the others. You want to hear the choir as one.
These are the 3 most popular stereo mic techniques for recording a choir.
Choir should be in a U shaped or half moon shaped circle for these techniques.
- ORTF Stereo - This uses 2 direction microphones, that point away form each other. It makes a wide V shape
- XY Stereo - This uses 2 directional microphones angled at 90 degrees with one capsule over the other
- Spaced Omni Stereo -This uses 2 omnidirectional microphones spaced far apart.. This is for larger groups
For a 3 or 4 mic set up, you can place mics evenly across the front of the choir. This shouldn't be used if the choir is in a semi circle.
Now that you have your microphones positioned and your singers in place, the next step is to balance the group.
Usually one section will be louder than the other section and in most cases the choir will even it out themselves. If it can't be fixed on their own, then you may need to move some singers around until the situation s fixed.
Do Your Sound Checks - Make sure there is a careful balance of choir. It should sound as one.
How did flanging get its name? Well, in the 1960's, audio engineers learned when they recorded the same exact sound on two different tape machines and played them at the same time, one would slightly delay to the other. These two audio sounds would react with each other and cause frequency cancellations between the both of them.
Flanging occurs when delay times go into the millisecond range and into the single digits. One way to describe flanging is to say its like a whooshing sound or a sound a jet airplane makes. The sound it makes is a kind of filtering or resonance that moves audibly up or down in frequency. The way it moves depends on what's going on with the modulation.
Let's say a sound source is delayed by 6 milliseconds and then you combine that with the source sound, especially when feedback is added, some frequencies cancel each other out. This will result in a pattern of peaks and valleys across the frequency spectrum. When you apply modulation, the delay time can either shorten or lengthen and in return the patterns of the peaks and valleys will shift up or down.
The shifting you hear is the flanging you hear. As the delay times shorted, the flanging will appear to go up and when they lengthen, the flanging can seem to go down.
By touching the flange on one of the tape machines, this would make the machine slow down, hence causing the delay to change in one direction and this will also change the way the sound reacts with one another. This technique is considered true flanging, because it allows 2 sounds to cross in time.
Flanging is a kind of spacial effect that has a distinctive sound. This sound can be so bold, that’s its easily over used. If you where to record a 12 song album or CD, you should only use the flanger in one or maybe 2 songs. Anymore than that and its overkill.
We (Audio Mastering And Mixing) embed the CD information such as band and artist names, along with album, track titles, ISRC codes into your master CD. This is known as CD Text and PQ Codes.
You need special players that are compatible in order to display the CD text. These compatible players include, many of the new car stereo systems, the newer potable CD players and some other professional hi fi players.
In order for your PC to display this CD Text, you need to submit it to the Online CD Data Bases. (the people who developed the computer software screwed up when programming the coding and this is why it doesn't automatically show the CD Text.).
These online data bases include:
- Gracenote - this is what iTunes and a few other programs use
- AMG Lasso - this is what Windows Media Player uses
- Freedb - a number of other programs use
- MusicBrainz - a number of other programs use
Websites for the online data bases
Gracenote - http://www.gracenote.com/company_info/FAQ/FAQs/#5a
AMG Lasso - http://allmusic.com/cg/amg.dll?p=amg&sql=32:amg/info_pages/a_product_submissions.html
Freedb - http://www.freedb.org
Musicbrainz - http://www.musicbrainz.org
To create an unbelievable stereo image, the sound stage needs to be wide and the mix has to sound good in mono. This is one of the hardest things to do. That said the stereo spread should not be so wide that the image is unrealistic. The best way is to try to place each instrument as you would hear them play live onstage from the view of the audience.
For example, panning instruments hard left and hard right. This goes especially for guitars and drum toms.
Another way is to pan them from the drummer's point of view. The kick drum would be dead center, right in the middle of the mix. The snare drum would be left of center and the hi-hat will be placed slightly to the left of the snare drum. You would place the toms and cymbals exactly how they would appear on the drum set. Then the guitars would be panned at around 10 and 2 o'clock and the vocals and bass would be right up the middle. The back-ground vocals would be placed just right of center. With this way, you have all your important information in the center and in mono and your supporting cast would be on the outside. Just remember, this is not the only way to do it, as the stage can hold a wide variety of musicians that are playing instruments.
You also use effect like reverb and delays to create a 3D stereo image
Using these effects, along with panning can place each instrument in its own stereo field and you can place them not only left , right and center, but you can place them back left, back right, back center, front left, front right, front center, med front left, mid front back and so on. This is what makes a great 3D stereo field.
Please Note: There are no set rules, just like everything else in mixing. So regardless of where you place each instrument, its very important to maintain an accurate representation of all the instruments. Just keep in mind that when you pan instruments hard left and hard right, you can introduce phasing problems when the mix is played in mono or pseudo stereo.