Score: 3. The first edition of Bobby Owsinski's The Mixing Engineer's Handbook destroyed that myth forever, breaking the craft of mixing down into discrete, understandable steps and showing musicians, audio engineers, and producers exactly how to get great results in the studio.
The book has since become the go-to text on mixing for recording programs in colleges and universities around the world. Learn the art of mixing from start to finish, and pick up tips and techniques from the pros, with The Mixing Engineer's Handbook, Fourth Edition. It follows the broad outline of its predecessors, but has been completely recast for the benefit of today's training in recording and its allied arts and sciences. Digital recording and signal processing are covered in detail, as are actual studio miking and production techniques -- including the developing field of surround sound.
As always, the traditional topics of basic stereo, studio acoustics, analog tape recording, and the stereo LP are covered in greater detail than you are likely to find anywhere except in archival references. This book has been completely updated with numerous new topics added and outdated material removed.
Many technical descriptions are now presented in Sidebars, leaving the primary text for more general descriptions. Handbook of Recording Engineering, Fourth Edition is for students preparing for careers in audio, recording, broadcast, and motion picture sound work. It will also be useful as a handbook for professionals already in the audio workplace.
This newly revised and expanded third edition of an Artech House bestseller offers invaluable insights and tips for every stage of the selling process. This third edition features a wealth of new material, including new chapters on business-driven discovery, white boarding, trusted advisors, and calculating ROI.
This invaluable book equips new sales engineers with powerful sales and presentation techniques that capitalize on their technical background—all spelled out step-by-step by a pair of technical sales experts with decades of eye-popping, industry-giant success under their belt. Sales Engineers' Handbook covers all of the key areas of selling high-technology products, including detailed action plans to establish personal excellence in key performance drivers in technical sales.
This comprehensive volume teaches you how to be more successful as an individual contributor, helping to better ensure promotion within your sales organization, or advancement elsewhere within your company. The book gives you the practical guidance you need to sharpen your skills in sales and technology.
Moreover, for the technical manager it explains how to build an infrastructure to support continuous high sales growth. T-RackS is a popular stand-alone audio mastering application that includes a suite of powerful analog-modeled and digital dynamics and EQ processor modules that also work perfectly as plug-ins during mixing.
While T-RackS is an extremely powerful tool for improving the quality of your recordings, all of that power won't do you much good if it's misused. Through his expert guidance, you'll learn the tips and tricks of using T-RackS processor modules to help bring your mixes to life and then master them so they're competitive with any major label release.
At the end of each chapter, there are a number of questions that will help you to better understand some of the principles of mixing and mastering. This newly revised and expanded third edition of an Artech House bestseller offers invaluable insights and tips for every stage of the selling process. This third edition features a wealth of new material, including new chapters on business-driven discovery, white boarding, trusted advisors, and calculating ROI.
This invaluable book equips new sales engineers with powerful sales and presentation techniques that capitalize on their technical background—all spelled out step-by-step by a pair of technical sales experts with decades of eye-popping, industry-giant success under their belt. Bob Katz explains audio concepts in a simple, holistic manner in this guide to producing a compact disc from scratch.
With the advent of cheap computers many amateurs are interested in learning this skill but the book will also interest professionals for its many useful tips and hints.
Sarah Jones presents a comprehensive guide to being an effective and creative assistant studio engineer. However, this book is unique. Rather than offering pages and pages of technical jargon, Jones discusses ways to approach the industry itself including How to network and achieve the best internship Knowing exactly what will be required of you and what to expect from your role The standards of knowledge and technical education you may require Pragmatic ways to approach 'difficult' sessions Preparing for interviews Effective strategies and career management The book is packed with useful contacts, organisations and support.
A host of interviews and anecdotes from established industry figures offer help and advice, giving you the best opportunity to beat the competition and land the engineering job you deserve. Every high-tech sales team today has technical pros on board to "explain how things work," and this success-tested training resource is written just for them.
After reading this book, you will be able to: - Walk into ever demo feeling confident and prepared - Include the one critical moment that must be in every demo - Hit that home run and know how to set it up - Master the art of answering difficult questions - Leverage the power of saying NO with ease You will definitely: - Avoid late nights and long sales cycles - Accelerate pipeline velocity and close more deals - Learn and apply the best practices in the business - Know exactly what to say and do before, during and after a demo - Achieve the technical win alarming, predictable consistency - This book addresses the root causes of the most common mistakes made by sales engineers.
This invaluable book equips new sales engineers with powerful sales and presentation techniques that capitalize on their technical background--all spelled out step-by-step by a pair of technical sales experts with decades of eye-popping, industry-giant success under their belt. The singer-songwriter, someone who writes and performs their own music, is an ever-present and increasingly complex figure in popular music worlds. He sends it off to an editor, and the editor reads it with a fresh set of eyes, just like a mastering engineer hears it with a fresh set of ears.
I can walk into this room in the morning and know if my monitors are right or wrong just by listening to a track from yesterday. It is the impartial ear that you get from your mastering engineer that is valuable. Sample Rate and Word Length Sample rate and word length determine the quality of a digital audio signal.
To understand the significance of sample rate and word length and how they affect quality, a brief discussion is in order. Remember, this is a brief discussion that will only give you the general concepts of digital audio. If you really want to get under the hood of digital audio, refer to a book such as Principles of Digital Audio by Ken Pohlmann. The more samples per second of the analog waveform that are taken, the better digital representation of the waveform that occurs, resulting in greater bandwidth for the signal.
Audio on a CD has a sampling rate of 44, times a second or A sampling rate of 96 kHz gives a better digital representation of the waveform because it uses more samples, and it yields a usable audio bandwidth of about 48 kHz.
A kHz sample rate yields a bandwidth of 96 kHz. Therefore, the higher the sampling rate, the better the representation of the signal and the greater the audio bandwidth—which means it sounds better!
The more bits in a digital word, the better the dynamic range—which means it sounds better! Every bit means 6 dB of dynamic range. The higher the sample rate, the greater the bandwidth, and therefore the better the sound. The longer the word length more bits , the greater the dynamic range, and therefore the better the sound. What all this means is that a mixing engineer now has a choice of sonic resolutions to mix to that was never available before.
This applies even if the ultimate delivery medium is to be a lower resolution CD or MP3. Standard Audio File Formats This section discusses the types of files found on a typical digital audio workstation and their differences. This is the process of sampling an analog waveform and converting it to digital bits that are represented by binary digits ones and zeroes of the sample values.
When LPCM audio is transmitted, each one is represented by a positive voltage pulse and each zero is represented by the absence of a pulse see Figure 2. LPCM is the most common method of storing and transmitting uncompressed digital audio. Because it is a generic format, it can be read by most audio applications, similar to the way a plain text file can be read by any word-processing program.
This is a file format for storing LPCM digital audio data. It supports a variety of bit resolutions, sample rates, and channels of audio. The format was developed by Apple Computer and is the standard audio format for Macintosh computers, although all platforms can read almost any file format these days.
AIFF files generally end with an. This is another file format for storing LPCM digital audio data. WAV files are indicated by a. The WAV file format supports a variety of bit resolutions, sample rates, and channels of audio. Perhaps the most significant aspect of BWFs is the feature of time stamping, which allows files to be moved from one DAW application to another and easily aligned to their proper point on a timeline or edit decision list.
These files end with a. It is the successor to the original monophonic Sound Designer I audio file format. When used on a PC, the file must use the extension of. Data Compression Linear PCM files are large and, as a result, painfully slow to upload and download, even with a dedicated high-speed connection. As a result, data compression was introduced to keep a certain amount of sonic integrity how much is in the ear of the beholder while making an audio file imminently transportable.
Data compression reduces the amount of physical storage space and memory required to store a sound, and therefore reduces the time required to transfer a file. That being said, the tools that are required are very unique to the genre, and in the analog days, they were often custom-made.
Even today there are custom mastering versions of some very popular outboard recording units again, mostly analog. These mastering versions have many of the most used controls detented and selectable, which is a rather expensive feature. We have all custom wire in the console. We build our own power supplies as well as everything else—the equalizers, everything.
Common Elements All tools for mastering, regardless of whether analog or digital, have two major features in common—extremely high sonic quality and repeatability. The sonic quality is a must in that any device in either the monitor or signal chain should have the least effect possible on the signal. The repeatability is important although less so now than in the days of vinyl in that the exact settings must be repeated in the event that a project must be redone as in the case of additional parts or changes being called for weeks later.
These additions add seriously to the cost of the device. Image courtesy of George Massenburg Labs. Figure 3. Image courtesy of Avalon Designs.
That being said, the overall signal path is kept as short as possible, with any unneeded items removed so the signal remains unaffected. So I have a bunch of tube equipment. And I have an Avalon compressor and Avalon equalizer, which is a little bit more specific.
That feeds an all-custom analog console. I also use that TC dB Max. Ninety-nine percent of what I do is done between those two devices. For example, with this one-inch two-track that I am working with, if I decide I need an analog EQ I will come through a Millenniia Dual the mastering version with the detents on it , then run into my Prism AD2 converter, and then come into the rest of the mastering chain bit digital.
Then we will store it bit digital and do anything else that we have to do at 24 bits internally. Then on the way back out the door, I can now loop out and back in and pick up my Z-Sys equalizer, using the power of POW-R word length reduction if I need to. So I have got the ability to handle it whichever way is most appropriate for the music. The output of that goes into either a dCS, Pacific Microsonics or sometimes Apogee analog-to-digital converter.
The bw, which has the kHz de-esser in it as well, is complete with a mixer, compressor, and equalization. As you can see, the analog path is somewhat of a hybrid in that it starts out in the analog domain but eventually enters the digital. This is essentially a digital router or patchbay that allows patching one digital device to another or many others at the push of a button. The unit functions as a digital audio patchbay, a distribution amplifier, a router, a format converter, and a channel switcher, all in one box see Figure 3.
Image courtesy of Z-Systems. For more information go to www. More than any one device, these are the main link of the mastering engineer to both the reference point of the outside world and the possible deficiencies of the source material.
More great pains go into the monitoring system than almost any other piece of gear in the studio. Because of this, more time, attention, and expense are initially spent on the acoustic space than on virtually any other aspect. No matter what you do, they will still suck, and you will still have problems. Wide frequency response is especially important on the bottom end of the frequency spectrum, which means that a rather large monitor is required, perhaps with a subwoofer as well.
This means that many of the common monitors used in recording and mixing, especially near-fields, will not provide the frequency response required for mastering. Smooth frequency response is important for a number of reasons. First, an inaccurate response will result in inaccurate equalization in order to compensate. Large monitors with a lot of power behind them are not for loud playback, but for clean and detailed, distortion-free level.
These monitors never sound loud; they just get bigger and bigger sounding and yet reveal every nuance of the music. Although the selection of monitoring is a very subjective and personal issue just as in recording , there are some brand names that repeatedly pop up in major mastering houses.
Image courtesy of Lipinski Sound. I think the phase coherency is just unsurpassed. In this particular case, my Dual 15s have been custom-modified for the room to some degree, and using them is just a great treat.
We have it all mixed in with different elements that we feel are going to give us the best sound. We just use a two-way speaker system with just one woofer and one tweeter so it really puts us in between near-fields and big soffited monitors. ON THE BOTTOM Getting a project to have enough low end so that it translates well to speaker systems of all sizes is one thing that mastering engineers pride themselves on, and one of the reasons that near-field or even popular soffit-mounted large monitors are inadequate for mastering.
The only way that you can properly tune the low end of a track is if you can hear it; therefore, a monitor with a frequency response to at least 40 Hz is definitely required. A great debate rages as to whether a single subwoofer or stereo subwoofers are required for this purpose. Those who say stereo subs are a must insist that enough directional response occurs at lower frequencies to require a stereo pair. There is also a sense of envelopment that better approximates the realism of a live event with stereo subs.
Either way, the placement of the subwoofers is of vital importance due to the standing waves of the control room at low frequencies. Feed pink noise only into the subwoofer at the desired reference level. Walk around the room near your main monitor speakers until you find the spot where the bass is the loudest. For more level, move it toward the back wall or corner, but be careful because this could provide a peak at only one frequency. Keep in mind that this method is for single subwoofer use.
To Calibrate the Subwoofer Using only one main speaker, feed pink noise in at a desired level say 80 dB, for example with the subwoofer disconnected. Listening only to the subwoofer, set its level 6 dB less than the main speaker 74 dB. Adjust the phase of the subwoofer to the position with the most bass. This can be done by adjusting the phase control on the unit or by simply reversing the wires on the input connector. Adjust the crossover point until the transition between the subwoofer and satellite is the most seamless.
It is not uncommon to see amplifiers of well over 1, watts per channel in a mastering situation. Because many speakers used in a mastering situation are rather inefficient as well, this extra amount of power can compensate for the difference. Image courtesy of Krell Industries. These are the amps that will put out like 6,watt peaks. Because each brand has a slightly different sound just like most other pieces of gear , most mastering facilities have numerous versions of each type available for a particular type of music.
Image courtesy of Lavry Engineering. I have a dB Technologies converter and I have one that the guys at JVC were fooling around with for awhile, which is excellent. Mastering equalizers differ from their recording counterparts in that they usually feature stepped rather than continuously variable controls in order to be able to repeat the settings. The steps may be in increments as little as 0. Some of the more popular digital hardware equalizers are the Weiss EQ-1 see Figure 3.
Image courtesy of Manley Labs. Image courtesy of Weiss. Image courtesy of Sonnex. Image courtesy of Massenburg DesignWorks. Although during recording this is usually the same unit that can be selected to function either way, mastering requires two separate units.
Generally speaking, the compressor is used to shape the dynamics of a song by adding punch and strength, whereas the limiter is used to raise the apparent level of the song by controlling the musical peaks. Hardware compressors that are often found in major mastering facilities include the analog Manley Vari-Mu see Figure 3.
Some of the popular software compressors and limiters include the Oxford Dynamics w and the Waves L1 Ultramaximizer see Figures 3. Image courtesy of Waves Audio Ltd. Analog Tape Machines Although the use of analog tape is very limited these days, you still see it used occasionally for the final mix, particularly on big-budget superstar sessions.
It is not uncommon for the electronics of these machines to be highly modified to improve the signal path. It should be noted that neither machine is currently in production, meaning that they draw premium prices on the used market.
When analog tape was at its peak in the mids, a format that was briefly used was the 1" two-track. Again, this is a 1" headstack mounted on an Ampex or Studer transport. At one point in time, cassette decks were an important part of the mastering facility, with huge banks of decks used for artist and label check copies.
But since the advent of the inexpensive CD burner, cassettes have nearly gone the way of the dinosaur. I have that connected to a Studer C37 classic vintage transport with the extended low-frequency heads that John French put in, made by Flux Magnetics. We also have one of the Tim de Paravicini 1" two-track machines with his fantastic tube electronics. I also have a Studer I usually work with two different A-to-D converters. I have a dB Technologies converter, and I have one that the guys at JVC were fooling around with for awhile, which is excellent.
You never know when the info might come in handy. Since the beginning of the CD, the was the standard format that the mastering facility used to deliver the master to the replicator because of its low error rate. Although every facility at one time had a least one, and they once drew premium prices on the used market, the has long been obsolete.
Many major mastering facilities still have one around, though, just in case they have to retrieve a master in this format. CONSOLES Although mastering consoles sometimes referred to as transfer consoles at one time were much more sophisticated and were the centerpiece of the master studio, these days mastering consoles can be as simple as a piece of wire with relays in the middle to connect the various pieces of gear and control the monitor level.
A mastering console differs from a normal recording console in that there are only two inputs for stereo four at most for manual crossfades between songs and no channel or track assignments. Manley Labs designs custom-built analog-based consoles, while Weiss with their now-standard modules , Crookwood, and SPL Labs manufacture console modules for the digital domain.
Image courtesy of Sound Performance Labs. Image courtesy of Crookwood. Originally the audio division of Sonic Solutions the company with the DVD authoring software , Sonic Studio was spun off into its own company in A series of VPIs are specifically designed for mastering, restoration, and signal analysis, which is why the AudioCube has gained favor with mastering engineers worldwide recently.
Although the modern mastering studio is loaded with peak-reading digital meters, most mastering engineers still like to use a good old-fashioned VU meter as well. This is because the VU gives a more accurate indication of the relative loudness than a peak meter. The classic example of this is the human voice, where a very quiet voice can have an extremely high peak level. Image courtesy of Dorrough Electronics, Inc. Although this function is sometimes available within the DAW, this is a complicated DSP task requiring massive calculations that tends to change the sound.
Therefore, most mastering engineers prefer to use a dedicated system for this task. Image courtesy of Weiss Electronics. As the name implies, a de-esser limits the amount of S sounds that might occur in a vocal track. Excessive high-frequency content is sometimes a by-product of compression and is known as sibilance.
A deesser is a frequency-dependent compressor that only triggers when excessive selective frequency content is present. What really separates the upper-echelon mastering engineers from the rest is the ability to make the music any kind of music as big and loud and tonally balanced as possible, but with the taste to know how far to take those operations.
Level The amount of perceived audio volume, or level, without distortion on an audio file, CD, vinyl record, or any other audio delivery method yet to be created is one of the things on which many top mastering engineers pride themselves. Notice the qualifying words without distortion, since that is indeed the trick—to the make the music as loud as possible and thereby competitive with other products while still sounding natural. Since then it has been the charge of mastering engineers to make any song intended for radio as loud as possible in whatever way they can.
And of course, this applies to situations other than the radio as well. Take for instance the iPod, the CD changer, or, in the very old days, the record jukebox. In the days of vinyl records, if a mix was too loud the stylus would vibrate so much that it would lift right out of the grooves, and the record would skip. When mixing too hot to analog tape, the sound would begin to softly distort, and the high frequencies would disappear although many engineers and artists actually like this effect.
A hypercompressed track has no dynamics, leaving it loud but lifeless and unexciting. Figure 4. Still, both mixing and mastering engineers try to cram more and more level onto the disc, only to find that they end up with either a distorted or an over-compressed product. The tradeoff with excessive compression to me is the blurring of not only the stereo image, but blurring the highs too.
An over-compressed program sounds pretty muddy to me. In the quest to get the level, they end up EQing the heck out of these tracks, which of course induces even more distortion between the EQ and the compression. Never in the history of mankind has man listened to such compressed music as we listen to now. What happens is everybody is right at that ceiling level as high as you can go, so now guys without a lot of experience try to make things loud, and the stuff starts to sound god-awful.
You can hear some pretty bad CDs out there. If you switch anything in at all, it just absolutely turns to dust. I wish all mastering engineers would speak out about this because it sucks. I buy CDs that I really want to listen to, and they are so fatiguing. We had level wars in vinyl right near the end of it, where everybody was trying to get the vinyl hotter and hotter and hotter. I have to say that the CDs that always please me the most sonically are not the real hot ones when I bring them in here and look at them on the meters.
If you put a compressor in the circuit, not even compressing, you will hear a difference, and it will sound worse. Just as important is the fact that every song on the disc must be perceived to be just as loud as the next.
The compressor is used to increase the small and medium level signals, whereas the limiter controls the instantaneous peaks. Remember, though, that the sound of the compressor and limiter will have an effect on the final audio quality—maybe for the worse—especially if you push them hard.
In general, the highest peak of the source program determines the maximum level that can be achieved from a digital signal. But because many of these upper peaks are of very short durations, they can usually be reduced in level by several dB with minimal audible side effects. By controlling these peaks, the entire level of the program can be raised several dB, resulting in a higher average signal level. Most digital limiters used in mastering are set as brick-wall limiters. This means that no matter what happens, the signal will not exceed a certain predetermined level, and there will be no digital overs.
Thanks to the latest generation of digital limiters, louder levels are easier to achieve than ever before because of more efficient peak control. Lookahead delays the signal a small amount about two milliseconds or so so that the limiter can anticipate the peaks in such a way that it catches the peak before it gets by.
Because there is no possibility of overshooting, the limiter then becomes known as a brick-wall limiter. By setting a digital limiter correctly, the mastering engineer can gain at least several dB of apparent level just by the simple fact that the peaks in the program are now controlled. Generally speaking, transient response and percussive sounds are affected by the attack control setting.
Release is the time it takes for the gain to return to normal or zero gain reduction. In a typical pop-style mix, a fast attack setting will react to the drums and reduce the overall gain.
If the release is set very fast, then the gain will return to normal quickly, but can have an audible effect of reducing some of the overall program level and attack of the drums in the mix. As the release is set slower, the gain changes that the drums cause might be heard as pumping, which means that the level of the mix will increase and then decrease noticeably. Each time the dominant instrument starts or stops, it pumps the level of the mix up or down.
Compressors that work best on full program material generally have very smooth release curves and slow release times to minimize this pumping effect.
EDDY SCHREYER: You go as loud as you can and you begin listening for digital clipping, analog grittiness, and things that begin to happen as you start to exceed the thresholds of what that mix will allow you to do, in terms of level.
You go for the level and properly control it with compression, then you start to EQ to achieve this balance. Of course, it all depends on the type of mix, how it was mixed, the kind of equipment that was used, how many tracks, the number of instruments, and the arrangement. I overdrive two, sometimes three, and even four pieces of gear, one of them being an A-to-D converter, and the other ones being digital level controls. Normalization looks for the highest peak of the audio file and adjusts all the levels of the file upward to match that level.
Although that seems like a very simple and easy way to adjust levels, it is seldom, if ever, used. As stated before, what normalization does is look for the highest peak of the audio file and adjust all the levels of the file upward to match that level. Even the smallest adjustment inside the DAW can sometimes cause massive DSP recalculations, all to the detriment of the ultimate sound quality.
But the biggest problem of normalizing is that it just looks at the digital numbers involved and not at the content of the music. The first one has to do with just good old-fashioned signal deterioration.
Every DSP operation costs something in terms of sound quality. It gets grainier, colder, narrower, and harsher. Adding a generation of normalization is just taking the signal down one generation. The ear responds to average level and not peak levels, and there is no machine that can read peak levels and judge when something is equally loud. Whereas in recording you might use large amounts of EQ from 3 to 15 dB at a certain frequency, in mastering you almost always work in very small increments usually in tenths of a dB to 2 or 3 at the very most.
What you will see is a lot of small shots of EQ along the audio frequency band, but again in very small amounts. This means that rather than applying a large amount of EQ at a single frequency, you add small amounts at the frequencies adjoining the main one. This lowers the phase shift brought about when using analog equalizers and results in a smoother sound.
You find yourself dipping 42 Chapter 4 The Mechanics of Mastering and boosting and trying to simulate air and openness and clarity and all the things that high end can give you, and so you have to start modifying the bottom a lot. DAVE COLLINS: I guess when we were talking about the philosophy of mastering, what I should have added was that one of the hardest things—and it took me forever to get this—is knowing when to not do anything and leave the tape alone.
As I have gained more experience, I am more likely to not EQ the tape, or just do tiny, tiny amounts of equalization. I go for a balance that is pleasing in any playback medium that the program may be heard in.
And obviously I try to make the program as loud as I can. That still always applies. But there are also limiting factors on what balance can be achieved. Some mixes just cannot be forced at the mastering stage because of certain ingredients in a mix.
If something is a little bottom-light, you may not be able to get the bottom to where you would really like it. You have to leave it alone so it remains thinner because it distorts too easily. Processing on Load-In Depending on the program, the elite mastering engineers may do some of their level adjustments and equalizing outside the workstation and then record that result into the DAW.
This is mostly a sonic issue, since the dedicated outboard devices may sound better than what can be offered within the DAW for that particular type of music or program. Editing Editing during mastering has gone through a complete metamorphosis in just a few short years. Until the mids, when most mastering entered the digital age, most editing was still done by hand using a razorblade and splicing magnetic tape on an analog two-track recorder. As with most mastering operations, what may seem easy can be enormously difficult without the proper knowledge of how to apply the proper tools.
FADES Almost anyone with a workstation knows how to apply fades, but does that mean that they are the right fades? Another one of the main elements of professional mastering is making sure that the fade not only happens, but sounds smooth as well.
As a result, the mastering engineer is frequently called upon either to do the fade entirely or to help it out. Even in these days of automated mixing and drawn-in fades in the workstation, many mix engineers still actually leave the master fade-out completely up to the mastering engineer.
Regardless of which type of fade is chosen, the principle is to get rid of count-offs, coughs, and noises left on the recording before the song begins. Although this seems to be an easy procedure, you must use care in order to maintain the naturalness of the downbeat.
The biggest problem with the headfades is that people just cut them off. The breath at the beginning of a vocal is sometimes very important. The temptation is to use a linear curve to make a fade, as in Figure 4. However, an exponential curve see Figure 4. Even when a fade is made during the mix, it sometimes needs some help due to some inconsistencies. Although this might seem to be quite arbitrary in many cases, the savvy mastering engineer usually times the spread to correspond with the tempo of the previous song.
In other words, if the tempo of the first song was at beats per minute, the mastering engineer times the very last beat of the first song to stay in tempo with the downbeat of the next. The number of beats in between depends upon the flow of the album. Please note that this might not be appropriate in all cases because each project is unique. It is a place to start, however. Many times a smooth flow between songs is not desirable, and a longer space is far more appropriate.
The spread in that case is replaced with a two-, three-, or four-second area in between songs to keep them disconnected. The EDL, which was originally developed for video editing, makes it easy to change the order of songs at any time.
The EDL is the list of all the elements that make up the final result and the positions those elements will take in the final sequence. Those elements are usually songs and will also be described in some fashion, usually by the name of the song.
This is partly due to the proliferation of digital audio workstations, where a poorly chosen fade is used prior to mastering. Most mastering engineers prefer to add any effects in the digital domain, both from an ease-of-use and from a sonic standpoint, so a reverb plug-in like the Audio Ease Altiverb is chosen.
Sometimes this is done by sending the output of the workstation into the effects device, then recording the result back into the workstation on two different tracks. The resultant effects tracks are then mixed in the proper proportions in the workstation. Among the devices used are the Lexicon PCM 91 and , and the TC Electronic M and , although any high-quality processor that operates in the digital domain will do.
I might use it on five percent of all my jobs. After you spend a couple of hours fine-tuning it, it can sound just like an EMT. Sometimes mixes come in that are just dry as a bone, and a small amount of judicious reverb can really help that out. Nothing is as exasperating to all involved as not knowing which mix is the correct one or forgetting the file name.
Bring the highest resolution mixes you can and make the other formats after mastering. In general, mastering engineers can do a better job for you if your mix is on the dull side rather than too bright or too big.
Hypercompression deprives the mastering engineer of one of his major abilities to help your project. Squash it for your friends. Squash it for your clients. But leave some dynamics for your mastering engineer. Matching levels between songs is one of the reasons you master your mixes.
Leave it to the mastering engineer to get the hot levels. Leave a little room and let the mastering engineer perfect it. The documentation expected includes any flaws, digital errors, distortion, bad edits, fades, shipping instructions, and record company identification numbers. Make sure you document them properly, though. Check it and fix it before you get there. After that, a bond of trust develops, and they will simply Chapter 5 Preparation for Mastering send the mix master with any instructions.
That being said, you should go to all of the mastering sessions if possible because things will always sound a bit different and probably better than what it sounded like during mixdown.
Attending the session also allows for some final creative decisions that only you can make. Sequencing the order in which the tunes appear on the CD or vinyl record is especially important, and doing this beforehand will save you a bunch of money in mastering time.
There should be one additional session, however, known as the sequencing session. This is really important if you will be releasing in multiple formats, such as CD and vinyl yes, there are still some diehards , or in different countries or territories because they will probably require a different song order due to the two sides of the record. Most CDs have a total time of just under 80 minutes , to be exact , although it is possible to get an extended-time CD.
But be careful—you may have replication problems. Obviously the available time decreases if you choose to include additional files on the ROM section of the disc. Cumulative time is important because the mastering engineer must know the total time per side before he starts cutting, due to the physical limitations of the disc.
You are limited to a maximum of about 25 minutes per side if you want the record to be nice and loud. Cutting vinyl is a one-shot deal with no undos like on a workstation. There has always been a difference of opinion on the sound of DAWs, and since there are more Pro Tools systems than anything else, I thought it would be a good idea to get some insight into DAW fidelity directly from the source. What are the common things that you see that cause a decrease in fidelity in the DAW?
One of the things is over-compressing and using way too much processing in order to get that CD sound too early in the process. I see mixes that are totally squashed and maximized up to the top of the digital word leaving the studio, heading to mastering.
GK: Exactly. I always recommend for people to leave 3 to 6 dB of headroom or even more depending upon the kind of music in their recorded files in their mix. Coming back to over-processing, do you have a recommended method for keeping everything as clean as possible? GK: As I said, part of the sound today is to make things compressed and loud, but I think what people do is over-compress. People try to get to the finished sound too quickly.
A little bit at a time is the key. One other thing about making a cleaner mix: Filtering makes a difference. How about the theory that you degrade the sound if you move the faders off of unity gain?
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