We are a specialty studio that specializes in 2 things, audio mixing and audio mastering services.
We are unlike any other audio mastering and audio mixing studio you will find online. We will personally give your music the attention it needs in order for your songs, EP or album to sound their best as humanly possible. We take our time with every project we mix and/or master!
Here are what some of our past clients said about us, after hearing their completed songs:
"Dude wow. Sounds GREAT. I've had nothing but good comments about my song, Ur right about the drum panning...didn't really notice that they weren't very wide until u said something. It might have been the way I positioned the overheads.
My roommate was like "wow! how's he do that?! Thanks bro"
"Hello CJ, I have to tell you that you're mastering work is impeccable.
Not just saying this, because I mastered the same song at Abbey Road and that was not to this extent at all.
I monitored the master on B&W 802s and they sound really good. All the linear notes I gave were beautifully implemented into the master."
"CJ, this really rocks now! Great job. I really appreciate you're taking extra time in removing the breath noise from the lead vocal. Thanks! I'll definitely return with future mix and mastering orders."
Audio Mastering Tips, Mastering Facts, Mastering Techniques, Mixing Techniques, Mixing Tips, Recording Tips and Facts.
We tried to list all the popular questions someone may have regarding audio recording, mixing and mastering.
Its divided into 2 sections. One is about our processes and the other is all about techniques that will help you better understand all about music recording and mixing.
FAQ & Q/A PAGE
At Audio Mastering and Mixing™ We are the Premier and Leading Provider of Online Audio Mastering & Audio Mixing Services. Our goal is to provide the very best in Audio Mastering, Combined Mixing & Mastering and CD Replication. Because our quality control standards are high, our Audio Mastering, Audio Mixing, and CD Replication Services are among the best in the business.
We try to anticipate questions you might have about our mixing and mastering services and about general audio recording techniques, tips and procedures. So we provide the answers here. If you need additional information Contact us. We Are here to help you.
What I s Audio Mastering?
Mastering involves very sophisticated audio processing techniques, It enhances and prepares your final mix for CD replication and/or Duplication. Mastering is a form of audio post production. It's the process of transferring and preparing your audio from a source that contains your final mix to a data storage device, and that's called the master. The master is the source of all copies that will be replicated, duplicated or pressed. In short, it's a process that prepares and organizes your music/audio for mass distribution. It enhances the quality of your songs to make it acceptable for radio airplay, iTunes, CD replication and CD distribution.
WhatsThe difference Between Audio Mastering vs. Audio Mixing?
Many people get these 2 terms confused and use them interchangeably.
Mastering engineers work with a single stereo wave file that is kept at its original sample rate and bit depth. Often the bit depth will be converted to 32bit floating point. We do that during our mastering procedure, but you keep it at its original bit rate.
Mixing gets your sound by adding effects like reverb, equalization, delays, compressors, and other effects and you mix the tracks until your levels and sound is the way you want it.
There is 2 kinds of mixing - Track mixing and Stem Mixing
- Track Mixing uses the single tracks of different instruments and we process and combine those tracks to create your final mix
- Stem Mixing is a form of mixing audio that are in groups of audio tracks, that are mixed down. They are processed separately before combining them into the final mix.
Mastering takes that final mix and polishes and enhances the natural sound qualities of that mix.
It takes your mix to the next level and will increase the quality of your audio files. Click here for more info about our audio mastering services.
Ideally, the source material at its original sample rate and bit depth is processed using noise reduction, equalization, compression, limiting, leveling, Stereo Enhancement, fading in and out, pre-gaping and other audio enhancements. Audio restoration processes can be applied, if needed, as part of our audio mastering process.
Steps of the process can typically include, but not limited to:
- The transferring the audio into a digital work station.
- Arrange the sequence of the songs or tracks and the spaces in between them. Just like a CD or album.
- The processing of audio to enhance and maximize the sound quality.
- The transferring of audio to Red Book specifications in the final master format.
Why Have Your Songs Professionally Mastered By Us ?
- Mastering will give your music a way to compete with commercially released songs.
- All commercially released songs have been professionally mastered and so should yours.
- Mastering could be the difference between your project sounding like a demo and a commercially released CD.
- So you can level the playing field between you music and a commercially signed artist's music.
- It can be the difference between a record labels A&R Representative signing you or a publisher pitching your music to artist, TV shows, movies and commercials.
Hardware mastering gear includes: There are many tools that can and could be used for mastering, so here are just some of them.
- LA 3A Dual Optical Compressor
- Manley Variable Tube Compressor/Limiter
- Manley Pultec EQP-1A Tube EQ
- Rupert Neve Portico 5014 Stereo Field Editor
- TC-96 Finalizer Mastering Processor
- Lynx Aurora VT-16 A/D and D/A Converter
- Apogee Big Ben Master Word Clock
- Genelec 1031A Reference Monitors
- And more...
Software mastering gear includes:
- Fabfilter Pro-L
- Fabfilter Pro-Q
- Fabfilter Pro C
- Flux Epure II EQ
- Flux Solera II Dynamic Processor
- Flux Pure II Mastering Limiter
- Waves Linear Phase Multiband Compressor
- Waves Linear Phase Equalizer
- Waves Ren EQ
- Waves Q10 Parametric Equalizer
- Waves Z-Noise
- Waves X-Noise
- Waves S1
- Waves X-Crackle
- Paz Analyzer
- Sony Sound Forge Pro 10
- CD Architect
What Audio Level Should The Wave File Or AIFF Files Be Set At ?
Your Wave or AIFF files should peak no higher than -3dB. The optimal dB range is between -10dB to -3dB. If its any higher than -3db, we lose the headroom needed to master your stereo wave file. Note, if you have 1 or 2 small peaks over -3dB, that's OK. Just as long as the entire song isn't over -3dB.
What Sample Rate And Bit Depth Should I Send You ? Back to top
You keep your stereo wave files (or AIFF) at its original sample rate and bit depth. Do not mix them down to 16bit/44.1KHz. That is done in the final step of mastering. If your song was recorded at 24bit and 48KHz, that's what you send us. If it was recorded at 24bit and 44.1kHz, that is what you sent us. Do not change the bit rate and sample rate.
ISRC codes is a standard international code. It uniquely identifies your audio recordings and music video recordings. The ISRC code identifies a specific recording, not the song itself. If you have different versions of the same song, remixes or edits. You will need a ISRC code for each different version.
- The ISRC is separated into 4 distinct elements to ensure that the ISRC codes that you assign will be 100% unique.
- Since the recording rights can change ownership over a period of time, it can be dangerous to assume ownership based on the Registrant portion of the ISRC code.
- The Year of Reference now represents the year in which the ISRC was assigned. The date of copyright should not be inferred from this portion of an ISRC.
- The issuance of an ISRC doesn't imply the registration of a copyright.
To apply for an ISRC code, go to www.usisrc.org. You should have ISRC codes if your getting your CD replicated. You do not need to have one, but it is a good idea.
Can There Be Any Kind Of Sound Quality Loss (Data Loss) During The File Transfer ?
No. There is no loss of sound quality (data) during the upload function. There are checks in place to make sure the data is completely transferred. Its perfectly 100% safe!
Here are a few reasons why you should get it professionally mastered:
- If you're preparing your music for serious release
- Put on your website for others to listen to
- Upload to iTunes or any other digital distribution company
- If your going to sell your music
- One reason is because you're going to be partial to it and your too close to it.
- You also may not have a tuned room
- Not enough knowledge and experience to know how to master a song
- You may not have the equipment necessary to produce a professional sounding master
Will Mastering Fix A Song That Was Mixed Terrible ?
Mastering enhances and polishes what is already mixed. It makes what was recorded, sound better. By no means will it fix a terrible mix.
- Our mastering, mixing, and musician studio session rates are very competitive.
- There is nothing I would rather do than doing this. It's my passion.
- I will give you personalized service, from start to end and even answer any questions after wards about anything and everything.
- I have over 20 years experience in this field.
- Your music will translate on every sound system. Meaning it will sound good on every sound system.
- Did I mention there's nothing I would rather be doing?
- No more agonizing over your mix. Let us do that for you. (I already lost most of my hair)
- I'm dependable and will do right by your mixes
- Your mixes will sound allot better after you get them back from us.
- Did I mention I live for this ?
What Happens If I'm Not Happy With The Master Or Finished Product ?
We send you either the entire song or 30 to 50 second clips for you to listen to and approve. If there is anything you want changed, just email us or call us and we will make the changes for you. We want you to be 101% happy with your mastered song. We want you to return to us for more business and we want you to come back to us for all your mastering needs.
Should I Normalize Tracks Or The Entire Mix ?
Never normalize your tracks or mixes. There is nothing to gain, as far as quality when you normalize your tracks or mixes and your really losing audio quality by re-quantizing the audio. If you need to raise your gain, do it with your track faders. You do not need an extra DSP (Digital signal processing) step before mixing.
Should I Compress My Mix Before Sending It To You ?
No!! Hell No!! Do not compress your entire mix. Meaning, do not use a compressor or limiter on your master bus. That compression cannot be undone, if there where mistakes made with it. You can still use compression on single tracks, but not on the master bus.
OMF stands for Open Media Framework File. Its common purpose is for sharing projects between multiple applications.
Whats A Broadcast Wave File ?
It's a version of the standard wave file format. Its use is important because it allows files to be time stamped and this allows files to be moved form one recording platform to another recording platform, while its stays aligned to their correct point in time. This is great when I'm mixing someones project. All I do is import the wave files into my recording program and they get automatically aligned to their place in time.
What Are The Terms For Communication Of Musical Expression ?
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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 allot 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.
- Wash- Allot of re verb. 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 allot of other effects also.
Between 85 and 90dB SPL. This is according to the Fletcher-Munson Curve. At this level your ears will not fatigue as fast.
What Are The Frequency Range Groups For Audio ?
Basic Categories include:
- Highs - above 3.5kHz
- Mids - between 250Hz and 3.5kHz
- Lows - below 250Hz
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
What Is An Audio Compressor And How Does It Work ?
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 us able to bring the overall volume of the song up and in return, the softer sounds get louder. This enables us to print the entire track with a louder signal level. Its a fantastic tool for recording instruments with a wide dynamic range.
A limiter is a compressor that has its ratio settings greater than 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 compressors, it also says limiter. The only difference is the ratio setting.
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. Multiband compressors are used for mastering, because of the control you get withe each band.
The 5 Parameters of a Compressor:
- 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.
- 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.
- 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.
- 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.
- 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.
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 ratio controls the amount of compression and the knee settings determines 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 allot 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
Peak and RMS Compressor Settings:
- RMS stands for Root Mean Square. Its the average amplitude of an audio signal
- Peak stands for immediate amplitude of an audio signal
- The RMS and Peak setting on a compressor/limiter will determine if it responds to the peak or RMS changes in amplitude.
- RMS is more gentle than peak
- Peak is more for limiting.
- 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 your 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 from in the mix, 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 also.
- Multiband compressors are mostly used in the mastering process, to fix the mixes problem areas.
The Different Types Of Equalization And How They Work..
EQ is used for 2 different things:
- To boost (enhance) part of a tone we want
- 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
- 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. This kind of EQ techniques is called Complimentary EQing.
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.
How To Get A Great Vocal Track ?
Wouldn't it suck if you or anyone else belted out that premo 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 the primary focal point of the song. The vocal needs to have and maintain constant space in the mix.
1. 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 golden.
2. Mic placement is an important in getting that great vocal sound:
- There are two issue to consider. The position of the mic in the room. You need to find the 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.
- The other issue is 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 mic of choice for most singers is 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. Its a balancing act.
- If your 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.
- 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 sibilance.
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.
3. Things that can ruin a vocal take:
- Jewelry (necklaces, bracelets) can make allot of noise. If a singer cannot take them off, due to some reason, you can wrap the a towel around it and put some tape on it. Just as long as it doesn't move.
- Early reflections form a music stand that is too 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.
4. 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.
5. Double tracking vocals:
- 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.
6. Tips for 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 on 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.
7. 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 withe 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 compressor 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 3 and 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.
Gates Vs. Expanders. Whats The Diifference ?
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.
- 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. upward expanders are not common and they tend to be too noisy. 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.
In General, most guitars are unbalanced and most low impedance mics are balanced.
There are 3 terms when talking about balanced and unbalanced:
- Lead: Its just another term for wire.
- Hot Lead: Its 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 interferences by diffusing or absorbing or rejecting it.
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 - How Does It Work ?
Reverb (digital) is the simulation of sound that is 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.
2. 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.
3. Parameters of the reverb:
- Pre Delay is the delay in time that happens before you hear the reverb. 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.
- 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.
4. 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!
What is the Proper Distance For Setting Up Studio Monitors
- The distance between each speakers should be the same distance as your ears are to them. You and your speakers should form a 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 to loud or to soft. So you may need to move your speaker placement around a bit, before you settle on the right location.
How To Get That Big Guitar Sound ?
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. Its 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: Its 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 it:
- 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 it:
- 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.
- 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 interface/sound card manufactures web site 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 your 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.
What Sound Card/Interface Do You Recommend ? Back to top
By far, the most popular question asked. I don't know how many times I've been asked this question or have seen the question asked. So, I put together a short list of interfaces that are known to work good for home studio recording.
Fire Wire Audio Interfaces:
- Echo AudioFire 2
- Echo AudioFire 4
- Echo AudioFire Pre 8
- Motu UltraLite mk3 Hybrid
- Motu 896mk3 Hybrid
- Cakewalk FA-66
- RME Fireface 400
- RME Fireface 800
- Lynx Aurora 8/FW
- M-Audio Profire 2626
- M-Audio Profire 610
- Focusrite Pro 24 DSP
- Focusrite Saffire Pro 24
- Focusrite Liquid Saffire 56
- Focusrite Saffire Pro 40
USB Audio Interfaces:
- Cakewalk UA-25EX
- Cakewalk UA-101
- Cakewalk V-Studio 20
- Cakewalk V-Studio 100
- Cakewalk V-Studio 700
- RME Fireface UC
- RME Babyface
- M-Audio Fast Track Pro
- M-Audio Fast Track Ultra
- M-Audio Fast Track Ultra 8R 8x8
PCI and PCI-e Audio Interfaces:
- Emu 1616M PCI-e
- Motu 24/IO Core PCI-e
- Motu HD192 Core PCI-e
- RME Multiface II
Make Sure you look them up to see what features come with each one before blindly purchasing one of them. Back to top
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
Do You Use Any Ordinary CD For Your CD Masters ?
Absolutely Not!! The CD Replicator will reject the CD, if its not a master grade gold audio CD. These master grade Cd's contain real 24 karat gold that improves the data-audio accuracy and it improves the life span of CD. It They last over 300 years. Gold audio Cd's are also resistant to oxidation
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: They happen when more than one device shares IRQs.
- 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
IRQ conflicts can happen when you install new hardware or reconfigure hardware.
How To Detect and Fix IRQ Conflicts:
- 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.
To resolve conflicts with PCI cards or ISA cards, you can manually move them to other available slots in your motherboard. By changing the slot, you change the IRQ channel.
Snare Drum Recording and EQ Tips..
Recording The Snare Drum:
First, make sure your snare drums tuning pegs are tuned correctly. Its 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".
EQing 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 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.
Whats The Difference Between Peak and RMS Levels ?
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.
Can I be Involved In Your Audio Mastering and/or Audio Mixing Process ?
Yes and I really want you to be involved in the mastering and mixing processes and the studio musician services. Your input is important to us. Besides just giving us your artist/band name and an audio file, you can pick up the phone and talk me me one on one during normal business hours about your project. (about 11a -9:30pm E.S.T). We also answer emails after normal business hours in a swift manor.
If your in our area, you can attend your audio mastering or combined mixing and mastering sessions.
Communication is key in this business.