Did your mastering engineer tell you?

Once mixed, your songs will sound best after the mastering process if:

They are well balanced dynamically:
Make sure it sounds like nothing in the mix is compressed more than anything else to the extent that it sounds unnatural. For example, take the relationship between vocals and the rest of the instruments - vocals should not be so dynamic in contrast to the instruments so as to make it sound like you are listening to a live vocalist singing over a compressed backing track of instruments. Another example is the relationship between vocal 'ess' sounds and the hihats or cymbals - make sure that the relationship between them is such that the mastering engineer will not be tempted to turn one of them down at the expense of the other.

They are well balanced sonically:
When soloing instuments on the mixer each individual instruments EQ should not be radically different to how the instrument sounds in real life. This is especially true of vocals. Be careful to fill the mixes with a broad spectrum of frequncies as contributed naturally by the instruments in the mix.

They are well balanced from an ambient or spacial perspective:
Avoid too much reverb and use it to create a sense of depth to the music by applying more reverb to those instruments or suonds you want to appear further back in the sound field or 'sound-stage'. Avoid putting a lot of reverb on vocals while keeping drums dry for example - unless of course you are specifically trying progressive production ideas.

Individual tracks (like a tom-mic channel for example) are muted when the instrument it is intended to capture is not being played at the time:
The cumulative noise from numerous open channels adds unecessary noise and ambience (reverb) to a mix. Rather than simply gating individual channels, go through every track and delete sections of the waveform where there is no signal from the intended source.

They are well-managed in terms of phase-relationship between the left and right channels:
Badly managed recordings or mixes can result in your left and right speakers cancelling out each others respective frequencies making the music sound thin and oily. It will also have adverse effects on FM transmission of the song causing it to break up on the edges of the transmitter's footprint.

They are not limited or compressed at the final master bus:
Leave this to the mastering engineer, making your music louder while preserving important musical detail is his speciality and he should have invested unreasonable amounts of money on equipment to do it really well.

They do not peak over 0dBfs anywhere in the track:
There is nothing worse than digital distortion to make mixes fatigueing and irritating. Avoid it like the plague.

They are tracked, mixed and bounced at high word length and sample rates:
Long word lengths (24 bit as opposed to 16 bit) leave more detail for digital processors to do mathematical calculations on. There are less mathematical 'loose-ends' so-to-speak. Higher sample rates move audible distrotion from digital equipment filters higher up the spectrum where it becomes less and less audible (see the section below on 96kHz sample rates)

They are burned on good optical media at optimal write speeds:
This is where you should spend the money on a good brand of CD-R for the purpose of delivering your master to the mastering engineer. Taiyo Yuden is an excellent and highly recommended brand of CD-R. Make sure you buy A-grade CD-Rs and burn the pre-master onto the disc at nothing higher than 16 speed to minimise write errors. Don't burn too low either since modern CD-Rs are optimised for writing at higher speeds and might not benefit from very low speed writing.

 

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Monitor levels: Interpreting the Fletcher-Munson Curve

Fletcher and Munson did research into the response of the human ear to a wide range of frequencies over a wide range of listening volumes. They came up with a set of graphs based on feedback from people who were asked to report when two different frequencies were perceived to be at the same volume. They found that the ear does not hear all frequancies equally well at different listening levels. The graphs show that the ear is less sensetive to bass and high frequencies at lower listening levels and is always most sensetive to frequencies between 1kHz and 6kHz (vocal frequencies). What this means for audio professionals is that consistent results can be achieved (particularly when adjusting EQ) by monitoring at a consistent listening levels and at a level where your judgement is least affected by the ear's frequency response. 83dB is reported to be the ideal listening volume. This is quite convenient since you can listen for fairly long periods without damaging your ears.

You can check your listening levels using a standard SPL meter. A convenient SPL meter is available from Radioshack.com

Once you have found a level close to 83dB, make a note of your monitor level settings on your sound card or monitor control and always use the same settings for every session. Making good EQ judgements will be easier than ever!


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Why 96 kHz?

If human hearing really only spans up to around 22kHz, what is the point of sampling frequencies high enough to reproduce frequencies up to 48kHz (as is the case with a 96kHz recording)? More importantly, what is the point of recording at 96kHz when it is all going to be played back through a speaker system that would probably only be capable of reproducing frequencies up to about 20kHz?

The answer is not a simple one, and if you want a heavily technical explanation you'd be best served by visiting the Lavry website where you can find a tutorial by Mr Lavry himself about this very topic, but if a short summary of the important reasons is what you're after, read on...

There are exceptions, but for the most part, everything important about the benefits of 96kHz recordings is wrapped up in the fact that distortions of the recorded signals in the highest bands of the recorded spectrum are present in the audible bands of the spectrum for recordings made at 44.1kHz, but these distortions shift higher up the spectrum for recordings made at 96kHz - high enough to be moved out of the range of perceptable human hearing. Even recordings at 48kHz offer some advantage over those at 44.1kHz for the same reason.

It is worth mentioning that well designed equipment operating at 44.1kHz may have very low audible distortion and conversely, poorly designed equipment operating at even 192kHz might have significant audible distortions. Ultimately it is a function of how much research and development goes into producing the equipment.

There is a lot of debate as to why 96kHz sounds better (and most agree - it does in fact sound better) but one thing is for certain -since most of our sound systems play back these precious recordings leaving out everything above 22kHz the explanation for the audible benefits of 96kHz sampling rates has to lie in how the equipment designed for higher sampling rates affects the audible part of the spectrum. This is where is gets controvertial and heavily technical and is something Mr Lavry is best qualified to elaborate on. For more information please visit: www.lavryengineering.com, visit the 'support' page and download his bold tutorial on Sampling theory. Enjoy...