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Sunday, December 12, 2010
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Friday, December 10, 2010
Read The Manual BEFORE You Buy
Brandon Drury quit counting at 1,200 recorded songs in his busy home recording studio. He is the creator of RecordingReview.com and is the author of the Killer Home Recording series. I've been saying it for years. Recording gear companies are f'ing scumbags, for the most part. They'll imply that their gear has more inputs and more features than it actually has to make a buck. Some gear comes with all sorts of new features, but they don't work if you choose to use some of the other features the unit has. This is the equivalent of General Motors claiming their Camaro has both a cd player and an air condition. However, they don't tell you that you can't use the cd player AND the air conditioner at the same time. That would never fly, obviously.
A recent example is my Steinberg MR816csx. The thing PROMISES..... I want to emphasize that when the features are listed for a given gadget at Musician's Friend or the manufacturer's website, it IS a PROMISE......that it has 8 channels of inputs via ADAT Lightpipe and 2 channels of ins via S/PDIF. This combined with the stock 8 analog ins SHOULD give a total of 18 inputs. (Yes, they do call the unit a MR816 and you can read that it is a 16-channel interface. However, it's never really explained how what appears to be 18 inputs is actually only 16.) Well, it turns out that you can't use all 8 of the ADAT channels and the 2 S/PDIF channels at the same time. You can use 6 ADAT inputs and 2 S/PDIF inputs. This isn't the end of the world, but it's generally accepted in recording land that this isn't an either/or situation. It should be stated how/why they arrived at the “16” number.
The problem is there are 700 choices AFTER you've narrowed down your choices. I just took a look at my spreadsheet I created to help me with selecting a new audio interface. Yes, I had to create a spreadsheet to manage all the freakin' possibilities, features, and requirements. It's so easy to get hung up on this phase with tail chasing research, rethinking your needs, rethinking your budget, trying to speculate future problems with chipsets, operating system bit depth, etc that many of us just look at one, throw our hands up in the air, and say “That one!”. You pull out the credit card, throw a Hail Mary, and sign your life away.
The very first thing MOST of us do when dealing with a fancy piece of gear (no, we aren't talking DVD players here) is fire up the manual. We know that audio interfaces don't fit in “asking for directions” territory. (Reading the manual is a necessity to getting things done. Asking for help while driving is immoral.) If more guys could discern the difference between reading manuals for recording gear and asking a gas station attendant where Clark St is those people would be cranking out dramatically more and dramatically better recordings.
Anyway, the first thing we SHOULD do is read the damn manual. Then, when are free from bs marketing jargon like “pristine quality” we get the real truth. This is where we find ourselves saying “Whoops!” when we have severe conflict issues. Common issues now are audio interfaces that won't run on certain operating systems due to issues with the bit depth of that operating system, incompatible chipsets, features that require you to buy more stuff, etc.
As stated above, this is where you also find out which features the unit actually has and which will work in your specific situation. The fancy DSP plugins found in the MR 816 CSX don't work if you use the S/PDIF output on the interface. The Cubase intergration is reduced to “not much” in this situation, as well. That's stuff you won't see in the ad. Guess where you do find it. The manual.
So, after 10 years of practicing this craft, it finally occurred to me to go ahead and fire that manual (that I'm going to read anyway) just BEFORE I pull out the credit card. The manual is not going to be a literary marvel, but it will at least be honest with you. It's not going to sugar coat all the limitations of unit. It's going to explain how stuff works and that means they are going to tell you what the unit can not do. I find that all the red flags end up being in bold anyway.
You'll find that disabled features are highlighted. You'll find how many inputs the damn thing actually uses ahead of time. Basically, you'll encounter what you should have been told by the bs marketing. I guess that's why it's called “bs marketing” and not “facts” or whatever. Hell, this gives a person an excuse not to read the manual once the interface shows up! That sound pretty damn manly to me!
Tags: Audio Interface, Yamaha MR816
This entry was posted on Thursday, August 12th, 2010 at 1:05 pm and is filed under Audio Engineering Principles. You can follow any responses to this entry through the RSS 2.0 feed. You can leave a response, or trackback from your own site.Healing Music Sessions
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Presonus Firestudio 2626 Review
Brandon Drury quit counting at 1,200 recorded songs in his busy home recording studio. He is the creator of RecordingReview.com and is the author of the Killer Home Recording series. First off, I want to point out that this is only my personal experience with one single Presonus Firestudio. You can count on your experiences being quite different....hopefully better. I suspect my experiences may have been different if they would have grabbed a different one from the shelf.
Secondly, it's worth noting that my rig is setup about as good as a person can get. My computer is totally optimized, dedicated for recording, and I didn't even use internet on it until a few months ago. (The amount of time needed to activate the numerous amounts of plugin and software I review became staggering so I decided to take a gamble and plug the thing into the web.) I've catered to all requirements from Presonus. I'd guess the average home recorder is not using such an ideal system.
Third, I record day in and day out for money. If my rig goes down, I lose cash. Period. I expect greater performance than most.
The Firestudio is an audio interface with 8 analog ins (with preamps), 8 analog outs, up to 16 total channels of ADAT ins and outs, and S/PDIF. It normally sells for a street price of $599-699, give or take.
For more details, check out the official Firestudio page.
Tons of I/O
For those of you who don't get excited by military-grade acronyms, I/O stands for inputs and outputs. The Firestudio is well equipped for this task if you use ADAT ins such as the M-Audio Octane or Presonus Digimax D8 http://www.recordingreview.com/blog/mic-preamps/presonus-digimax-d8-review/ . There aren't many people who are going to need more than 26 simultaneous inputs and there aren't much more affordable solutions than the Firestudio in this department.
Sounds Alright
The preamps in the Firestudio 2626 do not compare to Neve, Great River, etc. No shit! I think this is a no brainer. Are they acceptable for a person who doesn't want to spend the price on a new car for 8 preamps? Definitely!
What you really want to know is how the preamps in the Firestudio 2626 compare to other interfaces out there. People rave and rave and rave about the sound of the Yamaha MR816. I've used that interface a ton in the past month and even made myself use it's preamps. I'd give a SLIGHT edge to the MR816, but those pres didn't blow my mind either. (For what it's worth, no preamp blows my mind! My Martech MSS-10 doesn't suck. End of story.) For a person wanting to get in at this price point, the Firestudio 2626 is not going to turn heads with it's sonics, but neither does the highest of high end systems either. So if the Firestudio's price makes sense, it's sonics are more than adequate.
I want to note that I use the S/PDIF input with my high end pres and Mytek converters for all my overdubs and the Firestudio 2626 sounds as good as anything in this situation.
If you are really worried about the sound of the Firestudio 2626, you may want to check out The Interrogator Sessions in Killer Home Recording. You'll hear it up against numerous other preamps ranging from Great River to Martech to Manley to an M-Audio Octane.
Excellent Routing
Without a doubt, the routing on the Firestudio 2626 is definitely one of its selling points. They got that right. Its DSP routing matrix (which does take a bit to get used to and does have some redundancy going on in not-so-intuitive areas) allows you to route any signal to any output you choose. I must admit that they've spoiled me in this department.
Routing the main outputs from Cubase to my 4-channel crusty/trusty Behringer headphone amp, my Mackie HR824 monitors which essentially serve as a miniature PA system, and to Focal monitors via S/PDIF out into a Mytek DA96 have made life easy. When I tried out the Yamaha MR816, it was very frustrating when I couldn't do this. In fact, I had to rethink my whole setup. I never quite got it where I wanted even with the Control Room features in Cubase 5. So I'd give the routing possibilities in the Presonus an A+.
Routing the stereo out to multiple sources is one thing, but I'd guess most people don't need this. Being able to route the individual stereo mixes for headphones is another. As a headphone mixer, this thing is extremely powerful and highly recommended. I generally don't need a bunch of specific mixes for individuals in live band situations, but when I have, the Firestudio has pulled it off extremely well. If they added reverb and maybe compression to their features list, I'd say they had this perfected.
Again, these are my own experiences, for whatever they are worth. I'm not sugar coating this. If you can't handle the truth, YOU CAN'T HANDLE THE TRUTH! (Sorry, I'm not good at typing good Jack Nicholson impressions.)
After using the Presonus Firestudio for 2.5 years, I can say that, without a doubt, my Firestudio has been totally unreliable. If it were a woman, the cops would have found her dead in a ditch a long time ago. (Don't ask me why I have a higher tolerance for audio interface problems than woman problems.) My most reliable era required me to leave my recording computer and Firestudio on indefinitely. Simply turning the computer off could cause problems. We'll get to that.
Loses Sync
I may have great luck with the Firestudio for a month. The, for no apparent reason, and with no obvious change in my system, the Firestudio would lose sync with the computer and the little red light would begin to flash. This phase of random working and not working would last between 3 days and 3 weeks and then the unit would work flawless for a while. For this period, I would be hijacked from my recording computer indefinitely. The solution?
This is where it gets interesting. There is no solution because there is no clear, obvious problem other than the damn thing simply going on strike. I wish I could have fed it $5 bills to work. I would have gladly done so on many occasions. It would be the extortion scam of the century, but maybe then Presonus could afford to build a product I can freakin' count on.
The solution is to restart the computer and see if that fixes it. When it failed, I'd turn off the Firestudio and restart the computer. I'd try turning the computer and Firestudio off for 30 seconds and firing them both up. I'd try turning the Firestudio on and then the computer. I'd try turning on the computer and then the Firestudio. Nothing.
Hell, just last night I had to end a session 2 hours early (lost time and lost money!) because the stupid thing wouldn't sync up. 4 hours later, nothing changed. NOTHING! I fired up a mix with no trouble.
A person may want to blame this on user error. Luckily, I've been doing this long enough to KNOW it's probably user error. That's why I've became pretty damn good at tracking down my screw ups. In fact, I'd go so far as to say I'm awesome at finding my screw ups. To this day, I can find no pattern and no trend. When the Firestudio 2626 wants to be a damn woman, it becomes a damn woman. (Note: A “damn woman” is opposed to a “nice woman” who deserves bunnies and chocolate.......and a bunch of wild screwing.)
I have noticed that if the power button gets pressed somehow (you'd be surprised how many times this has happened accidentally even though it's recessed in my rack) with Cubase running, all hell breaks lose. Trying to get it to sync up after that, even with 15 restarts is nearly impossible. The best solution I've found is not to care. Turn everything off, make a sandwich, and see if there is a Star Trek rerun on.
Just for the hell o f it, I killed the Yamaha MR816 with Cubase open. Cubase immediately says, “Hey! Where'd the interface go?” When I turned it back on and told Cubase to calm down (by re-selecting the Yamaha driver) all was well. This is clearly a Firestudio-specific situation. No doubt about it. I award Presonus minus a billion points for this one.
Chews Up CPU Power
I remember when my M-Audio Delta 1010s went to the audio interface dumpster in the sky and I switched to the Firestudio 2626. The first thing I noticed was the fact that I was suddenly out of CPU power on mixes that had plenty of headroom before. I got used to it, upgraded to a Quad Core, and never thought about it much. Now that I had a chance to use the Yamaha MR816 for a month, once again, I'm finding that the CPU meter FLYING up on mixes where it really shouldn't. The Presonus Firestudio is definitely a CPU hog.
Actually, not only is it a CPU hog, it's a ram hog as well. Okay, maybe not a HOG. Maybe I'm too geared towards 2001 RAM standards, but I can't accept an audio interface requiring 40MB of RAM. I can't figure out what it's doing that would require such RAM usage. While 40MB is not the end of the world in the an era when most of us have 4GB, the designers of the Firestudio clearly knew that audio a zero-tolerance, high performance kind of ballgame. The fact they pissed away 40MB of RAM when other interface companies do not says something and I don't think it's a good something.
Multiple Control Panels
This one ain't the end of the world, but it's always bugged me. I never understood why they chose to use two control panels. One control panel allows you to change latency, clock source, etc. Then if you make a few clicks in that control panel, you'll get to a new control panel that allows you to control routing, individual mixes, etc.
What I never understood was why they broke this up. They all should have been under one control panel with maybe a few tabs added. It's entirely unintuitive and I always feel like I waste 4 seconds every time I need to make a change.
I always mix at 2048 samples (high latency) and track and much lower latencies. I usually have 8 projects going on so it seems like I need to change the latency for every session. Pushing a bunch of buttons and going through a bunch of menus to do something super simple is annoying.
I'm shocked that this has not been addressed as they have released updated drivers. End of the world? No. Annoying? Definitely!
Okay Latency Settings
While maybe this one isn't a “screw up”, the Firestudio's latency is a bit slow compared to other Firewire interfaces on the market. On a good day I can get 128 samples without too many pops and clicks. I usually have to resort to 192 samples however. For vocals, this is flat out unusable. Latency As Vocal Producing A person can switch to direct monitoring, but then when you need reverb, you need an outboard reverb unit and you'll need to get a bit clever with your sends / returns. It can be done, and is probably worth the trouble. However, on most days it would be cheaper and easier if you could just turn the latency down low enough to make this a non-issue.
I had no problems getting the Yamaha MR816 reliably down to 64 samples as long as I wasn't pushing my rig too hard. The Firestudio was never able to get down lower than 128 samples even with zero CPU load on a Quad core, XP 32-bit rig. It was clear with my hardware and operating system that we had pushed the limits of the Firestudio.
Random Latency Permissions
On some days, I'll start mixing and realize that I still have my latency set to a super low setting and need to crank it up. When I open the menu, the latency may be grayed out, which is a nice way of Presonus to say, “Go F yourself, San Diego.” I then have to stop what I'm doing, kill Cubase to get my permissions back, and then restart the mix. I lose 45 seconds because of some Presonus error.
It wouldn't bother me if this was how it was all the time. I could live with being forced to set my latency before opening Cubase. The problem is half the time I CAN change the latency whenever I want with Cubase running. This glitch bugs the hell out of me.
Random Wordclock Changes
This won't affect you guys with simple setups, but it drives me nuts. I use my Mytek AD96 as a master clock. I run that clock into my M-Audio Octane and that feeds my Presonus Firestudio 2626. This is a very common setup. For no particular reason, the Firestudio will switch its inputs to the second ADAT input (which I'm not using). This causes it to lose sync. As long as they menu isn't grayed out I can quickly change this back to ADAT #1. Unfortunately, it is often grayed out. Why? Either way, this is another damn thing I have to think about when I have a billion other things on my plate.
Why Is That Light Blinking?
As a dude who has dabbled in web programming, I know that it's fairly straight forward to develop error codes. If something screws up, a window should pop up and say “Error #554”. Then I can look online and see what that means and actually fix it. I don't have to guess and I don't have to look like an idiot restarting a computer 15 times.
I don't know anything about interface drivers, but I suspect that displaying error codes would require one programmer to work one extra week. The fact that they haven't done this means that Presonus is willing to compromise in areas that I flat-out do not believe should be compromised.
It's clear I'm willing to pay more for an interface that makes my life easier. I've certainly paid by going with the Firestudio.
Is It My Fault?
I've really grown to like the eastern philosophy that says, “Everything is your own fault”. In this particular case, I've went round and round about what I (I want to emphasize “I” here) could be doing to make the Firestudio work better for me. Simply put, I've got nothing! Nada. Zip. The only thing I know to try is a Windows 7 rig with 8GB of RAM even though it's generally considered a much better idea to stick with the trusty ol' operating system. XP should be more stable than Windows 7 right now.
The fact that the MR816 never had a single issue with syncing up (or any other session stoppers) illustrates that my rig is stable.
For hobbyists, the purchase of an interface is rarely looked at as the kind of thing that can pay for itself. However, for those of us who are charging by the hour, when we lose 2 billable hours due to a shitty interface design, that's easily quantifiable cash that we'll never get back. In my opinion, it adds to the cost of the piece of gear causing the trouble. I wish I had only lost 2 billable hours per month. I'd say I've lost dramatically more than that. At 2.5 years, I could have have bought an interface that costs 3x as much as still came out ahead.......Or I could have went on a cruise or tw.
For anyone who is counting on an interface to work day in and day out, there is no way I can recommend the Firestudio 2626. When it works, it's a fine interface at a very good price. Maybe you'll have better luck with the reliability end. However, for me, I wish I would have had the balls to slap a fancy interface on the credit card long ago. 25% interest would have been cheaper than the hell it has put me through and the time it has wasted.
I also want to point out that I paid $700 for mine in early 2008. On Ebay these things are going for a fraction of the price. This poor resale value is semi-common with computer recording gear, but it's definitely common with gear that people want to get rid of. If you are looking for a “nice” interface, at this point, I'd recommend the MR816. It's not perfect, either, but it's a product that I would and have taken to battle.
Brandon
Tags: Audio Interface, Presonus Firestudio, Yamaha MR816
This entry was posted on Thursday, August 12th, 2010 at 12:33 pm and is filed under Recording Equipment Reviews. You can follow any responses to this entry through the RSS 2.0 feed. You can leave a response, or trackback from your own site.Friday, November 19, 2010
The Daily Adventures Of Mixerman Book Review
Brandon Drury quit counting at 1,200 recorded songs in his busy home recording studio. He is the creator of RecordingReview.com and is the author of the Killer Home Recording series. I had some guy griping that his username was already taken here at the RecordingReview.com forum. I deal with this sort of thing daily, but this time the guy actually claimed to have the name trademarked. I thought, “Hmmm. This is interesting”.
The username in question was “Mixerman”. A quick search in Google pointed me to a certain book called The Daily Adventures of Mixerman, which promised to be a true insiders look of the day-to-day goings on in the major label big boy land. It seems our Mixerman was a real engineer doing the kind of records that people used to pay for.
Mixerman and I shot the shit a bit through emails, he gave some advice on consoles (coming soon), and he helped me out considerably. In return, I snagged a copy of his book from Amazon and off we went on our separate paths.
I generally try to read a book a week, but I usually have 4-5 books I'm reading at any one time. It took a while for my queue to make it to the Mixerman book.
Unfortunately, things didn't go as planned.
You see, I'm a very busy dude. I always seem to attempt to squeeze more than is physically possible into any given 24-hour period. The end result is I end up getting mad and demoralized when I feel the rippage from the overstuffing. If you've got some dirty ideas in your head, you are not too far off track.
As I turned from page 1 to page 2, I find myself being reminded of that chapter in Mind Of The Market (Michael Shermer) where he explains how the brain's pleasure center juice causes us to not want to stop certain behavior. In short, this damn book turned me into a temporary crack addict. I simply could not rationalize putting it down. I felt I was committing treason every time I stopped reading it. My only savior was the finite number of pages. I don't find many books that keep me THIS hooked....and I read a lot of freakin' books.
The Daily Adventures of Mixerman is a daily journal of everything that happened on a major label gig from an audio engineer's perspective. No, he didn't document compression ratios and attack times, unfortunately, but he certainly painted a vivid account in a non-holds-barred fashion as to what it's really like being a “big boy”.
I think, deep down, all of us home recorders who've been beat up and down by our local clientele have always wanted to sneak a peak at what the big guys are really up to. All the sessions I've attended didn't get much done. Now I know why.
I came very close to seriously “going for it” in big boy studio land (another story for another time) and always wondered what it would have been like if I had chosen that path. The are certain delusions of grandeur that the greener grass of big boy land intoxicates as we struggle with often closet-sized room, micron-sized budgets, and even smaller-sized talents.
I can't think of a way to get any closer view of the engineer's perspective in big-boy land than The Daily Adventures of Mixerman.
Mixerman is a certain breed of human......a hair cynical, highly intolerant of bullshit, even less tolerant of idiocy, quite intelligent, and maybe even a bit smug. To put it frankly, he's an asshole. At the risk of insulting Mixerman, I've got to call a spade a spade. Mixerman is honest. He says what he thinks and I laughed HARD dozens of times.
It's no wonder he had people flipping over his once-publicly viewable journals. Honesty is about as welcome as birth control in a Catholic orgy these days with the one exception (which just happens to be this book's core demographic): other assholes. (It taking one to know one may apply.)
The Daily Adventures of Mixerman is a MUST READ for anyone who owns a microphone. The humor is on the crude side (which MAY offer some explanation as to why I enjoyed it so much) and the old Make Twain quote about the difference between fiction and non-fiction being fiction has to actually be believable may apply here. In short, this story is INSANE.....therefor proving it's validity.
Just one note: Because this is a journal, don't expect a giant Lord Of The Rings battle at the end with the good guy coming out on top. The ending isn't a bad one, it's just of a Sony Picture Classics-style ending as opposed to however Bruckheimer ended his last “movie”.
Just a second note: I did attempt to contact Mixerman again before posting my review, but either he didn't get it or he doesn't respond to private messages with the subject consisting of a 4-letter explicative followed by “you”. I'm not sure why my tact didn't warrant a response. Oh well.
Tags: book review, Mixerman, The Daily Adventures Of Mixerman
This entry was posted on Monday, October 25th, 2010 at 11:15 pm and is filed under Audio Engineering Principles. You can follow any responses to this entry through the RSS 2.0 feed. You can leave a response, or trackback from your own site.Thursday, November 18, 2010
World's Cheapest Strongest Recording Equipment Rack
Brandon Drury quit counting at 1,200 recorded songs in his busy home recording studio. He is the creator of RecordingReview.com and is the author of the Killer Home Recording series.
If there is one area where you can blow outrageous amounts of cash with no real hope of retaining resale value, it has to be gear racks. Take a look around. For a “thingy” that does little more than hold your gear in place, a person can easily shell out $300. (The cheapest I found was this, but that one didn't feel right.) I'm talking plain jane, no-frills, no shock absorption racks.
I can't figure out why this price is justified. One can argue about aesthetics, but the kind of racks I like to look at have nothing to do with audio gear. When dealing with ugly racks, I find that it doesn't take much work to get a few cheap pieces of wood to look “pretty”, if that's your interest.
In terms of strength, the rack gear itself rarely needs much in the way of support. In fact, one could argue that they ARE the support. Unless you reside on the San Andreas fault or have a guy who looks mysteriously like Tesla performing resonance experiments under your apartment, I can't see how strength is a huge priority anyway. I figure as long as drop kicking your rack does more damage to you than your gear, you should be in good shape.
I'm fairly certain a person won't be able to top this rack for strength, size, or budget. Hell, it probably ranks quite well in the ease of construction department as well. I have the construction skills of the guy down the street with seven fingers. If I can build this thing in 20 minutes, you Stag drinkers could build it in two. Even if you are a vegetarian, have never shot a gun, and talk with a lisp, the longest this would take would be 22 minutes assuming you had the right tools. (Note: I'd bet you HGTV style lisp-speakers will give the Stag drinkers a run for their money.)
2 - 2x4's cut to 33.5”
2 – 2x4's cut to 17”
2 – 18” rack rails (found at Musician's Friend (#1 , #2) or Parts Express)
2.5” drywall screws
1” drywall screws
Total Price: $35 (approx)
Build Time: 2-22 minutes
I highly recommend that you angle the rack rails so they point slightly upward. This has improved visibility immensely for me and I'll require it on all racks I use.
Some guys worry that using too large of rack could cause unwanted direct reflections from your monitors. You definitely don't want this. If you are in a position where the monitor's sound is not going to slam directly into the rack's side and into your face, you can go pretty large (18 space and larger). If you think this may be a concern, a 12-space rack may be about as high as I want to go.
This rack design is extremely strong. When you give it a good push (with plenty of gear in it) it won't budge. It feels extremely stable.
The rack has great ventilation. Some of my gear runs hot. By keeping the sides open, ventilation is improved dramatically over racks with solid sides.
Obviously, the price is right.
As is, the rack is unfinished. I've found that a little stain and polyurethane go a LONG way as you can see by the racks I use with my current setup.
The sides are exposed and this can look messy. At the time I built this rack, I wasn't concerned with aesthetics. If you aren't worried about ventilation, covering the sides with a prettier material is a no-brainer. On my latest design, I wanted to retain ventilation so I used fabric to cover up the sides. It does the trick and people who are into visual crap think it looks good.
This rack is not the most portable thing (whether within your room or taking it out on the road).
This rack design relies on the gear itself for strength. If you only have a few pieces, it may not be quite as strong as I've implied here. With it full, the thing is a tank. I've added a back support brace on my latest design to keep the rack square as I've found the back likes to expand outward a bit.
Tags: cheap, rack
This entry was posted on Monday, October 25th, 2010 at 11:47 pm and is filed under Audio Engineering Principles. You can follow any responses to this entry through the RSS 2.0 feed. You can leave a response, or trackback from your own site.10 Myths About Normalization
The process of normalization often confuses newcomers to digital audio production. The word itself, “normalize,” has various meanings, and this certainly contributes to the confusion. However, beginners and experts alike are also tripped up by the myths and misinformation that abound on the topic.
I address the 10 most common myths, and the truth behind each, below.
First, some background: While “normalize” can mean several things (see below), the myths below primarily involve peak normalization.
Peak normalization is an automated process that changes the level of each sample in a digital audio signal by the same amount, such that the loudest sample reaches a specified level. Traditionally, the process is used to ensure that the signal peaks at 0dBfs, the loudest level allowed in a digital system.
Normalizing is indistinguishable from moving a volume knob or fader. The entire signal changes by the same fixed amount, up or down, as required. But the process is automated: The digital audio system scans the entire signal to find the loudest peak, then adjusts each sample accordingly.
Some of the myths below reflect nothing more than a misunderstanding of this process. As usual with common misconceptions, though, some of the myths also stem from a more fundamental misunderstanding – in this case, about sound, mixing, and digital audio.
Myth #1: Normalizing makes each track the same volume
Normalizing a set of tracks to a common level ensures only that the loudest peak in each track is the same. However, our perception of loudness depends on many factors, including sound intensity, duration, and frequency. While the peak signal level is important, it has no consistent relationship to the overall loudness of a track – think of the cannon blasts in the 1812 Overture.
Myth #2: Normalizing makes a track as loud as it can be
Consider these two mp3 files, each normalized to -3dB:
The second is, by any subjective standard, “louder” than the first. And while the normalized level of the first file obviously depends on a single peak, the snare drum hit at 0:04, this serves to better illustrate the point: Our perception of loudness is largely unrelated to the peaks in a track, and much more dependent on the average level throughout the track.
Myth #3: Normalizing makes mixing easier
I suspect this myth stems from a desire to remove some mystery from the mixing process. Especially for beginners, the challenge of learning to mix can seem insurmountable, and the promise of a “trick” to simplify the process is compelling.
In this case, unfortunately, there are no short cuts. A track’s level pre-fader has no bearing on how that track will sit in a mix. With the audio files above, for example, the guitar must come down in level at least 12dB to mix properly with the drums.
Simply put, there is no “correct” track volume – let alone a correct track peak level.
Myth #4: Normalizing increases (or decreases) the dynamic range
A normalized track can sound as though it has more punch. However, this is an illusion dependent on our tendency to mistake “louder” for “better.”
By definition, the dynamic range of a recording is the difference between the loudest and softest parts. Peak normalization affects these equally, and as such leaves the difference between them unchanged. You can affect a recording’s dynamics with fader moves & volume automation, or with processors like compressors and limiters. But a simple volume change that moves everything up or down in level by the same amount doesn’t alter the dynamic range.
Myth #5: Normalized tracks “use all the bits”
With the relationship between bit depth and dynamic range, each bit in a digital audio sample represents 6dB of dynamic range. An 8-bit sample can capture a maximum range of 48dB between silence and the loudest sound, where a 16-bit sample can capture a 96dB range.
In a 16-bit system, a signal peaking at -36dBfs has a maximum dynamic range of 60dB. So in effect, this signal doesn’t use the top 6 bits of each sample*. The thinking goes, then, that by normalizing the signal peak to 0dBfs, we “reclaim” those bits and make use of the full 96dB dynamic range.
But as shown above, normalization doesn’t affect the dynamic range of a recording. Normalizing may increase the range of sample values used, but the actual dynamic range of the encoded audio doesn’t change. To the extent it even makes sense to think of a signal in these terms*, normalization only changes which bits are used to represent the signal.
*NOTE: This myth also rests on a fundamental misunderstanding of digital audio, and perhaps binary numbering. Every sample in a digital (PCM) audio stream uses all the bits, all the time. Some bits may be set to 0, or “turned off,” but they still carry information.
Myth #6: Normalizing can’t hurt the audio, so why not just do it?
Best mixing practices dictate that you never apply processing “just because.” But even setting that aside, there are at least 3 reasons NOT to normalize:
Myth #7: One should always normalize
As mixing and recording engineers, “always” and “never” are the closest we have to dirty words. Every mixing decision depends on the mix itself, and since every mix is different, no single technique will be correct 100% of the time.
And so it goes with normalization. Normalizing has valid applications, but you should decide on a track-by-track basis whether or not the process is required.
Myth #8: Normalizing is a complete waste of time.
There are at least 2 instances when your DAW’s ‘normalize’ feature is a great tool:
Myth #9: Normalizing ensures a track won’t clip
A single track normalized to 0dBfs won’t clip. However, that track may be processed or filtered (e.g. an EQ boost,) causing it to clip. And if the track is part of a mix that includes other tracks, all normalized to 0dB, it’s virtually guaranteed that the sum of all the tracks will exceed the loudest peak in any single track. In other words, normalizing only protects you against clipping in the simplest possible case.
Myth #10: Normalizing requires an extra dithering step
(Note: Please read Adam’s comment below for a great description of how I oversimplified this myth.) This last myth is a little esoteric, but it pops up sporadically in online recording discussions. Usually, in the form of a claim, “it’s OK to normalize in 24 bits but not in 16 bits, because …” followed by an explanation that betrays a misunderstanding of digital audio.
Simply put: A digital system dithers when changing bit depth. (i.e. Converting from 24-bits to 16-bits.) Normalizing operates independent of bit depth, changing only the level of each sample. Since no bit-rate conversion takes place, no dithering is required.
Normalizing can mean a few other things. In the context of mastering an album, engineers often normalize the album’s tracks to the same level. This refers to the perceived level, though, as judged by the mastering engineer, and bears no relationship to the peak level of each track.
Some systems (e.g. Sound Forge) also offer “RMS Normalization,” designed to adjust a track based on its average, rather than peak, level. This approach closer matches how we interpret loudness. However, as with peak normalization, it ultimately still requires human judgment to confirm that the change works as intended.
Wednesday, November 17, 2010
Vocal EQ Tips
Here are some tips and techniques for treating vocal tracks with EQ while mixing.
Most importantly: Every voice is different, and every song is different. That advice bears remembering, even if you’ve heard it dozens of times. When you find yourself approaching a vocal mix on auto-pilot, applying effects “because they worked last time,” consider disabling the EQ altogether to gauge just how badly the adjustments are needed.
Reasons to EQ: The 3 main reasons to filter a vocal with EQ are
1) to help the voice sit better in the mix,
2) to correct a specific problem, and
3) to create a deliberate effect, like “A.M. radio voice.”
If you’ve EQ’d a vocal track for some other reason, be sure the result is improving the mix.
Gentle boosts: The “cut narrow, boost wide” guideline applies to vocals perhaps more than any instrument. Our ears have evolved remarkable sensitivity to the sound of human speech. (Consider how easily we pick up a single conversation in a crowded noisy room.) So we’re immediately, instinctively aware when a voice has been processed unnaturally.
High-pass: Most vocals – though of course not all – benefit from a low cut filter. The average fundamental frequency in an adult male voice is 125Hz, and often you can roll off up to 180Hz without affecting the sound. (If your mic or preamp has a low-cut filter, consider engaging it when recording vocals, as most subsonic audio in a vocal track consists of mic-stand noise, breath rumble, popping, and other undesirable sounds.)
Bypass: Especially with high-pass filters, it’s easy to remove too much body from a vocal, as our ears adjust so quickly to new sounds when mixing. If your EQ has a bypass option, use it periodically to make sure you haven’t gone too far with an adjustment.
Common fixes:
To reduce a nasal sound, try dipping a few dB around 1kHz, and moving the center frequency slightly up or down to find the most effective point.To treat popping P’s and T’s, cut everything below 80 Hz.For a little extra clarity and presence, try gently boosting the “vocal presence range” between 4kHz and 6kHz.Reasons NOT to EQ: EQ can’t make your voice sound like someone else’s.
See Also: Better vocals improve your recordings, Great free vocal pluginsFor more home recording tips,
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Friday, November 12, 2010
All Piano Sheet Music Has Buried Treasure In It
Vast musical riches lie beneath the surface of every song -- like buried treasure. But you'll never find it if you use sheet music the same old way. I will teach you to use sheet music like a map.
Check it out!
Thursday, November 11, 2010
How To Become a Musical Mind Reader!
How To Know Which Chords To Use In a Song Before You Play The Song! Become a Musical Mind Reader
Check it out!
Wednesday, November 10, 2010
Insider Music Business Members Club
Join today and receive tele-class training, downloadable Music Journals, expert interviews, members forum and more. Plus your bonus gifts are waiting inside too. Run by music business expert Kavit Haria
Check it out!
Tuesday, November 9, 2010
Free Online Music Promotion - A Starving Artists Must Have!
"a Musician's Guide to Growing Your Web Presence: Zero to Hero Without Spending a Dime" is a nearly 3 hour video series showing musicians how to forge a web presence that markets and even gives them a way to sell their music For No Cost. Priced to move.
Check it out!
Monday, November 8, 2010
Auto-Tune Abuse in Pop Music – 10 Examples
Pitch correction software has applications from restoration and mix-rescue to outright distortion of a voice or instrument. I’ll discuss some of the more tasteful uses of these auto-tune tools (whether the original from Antares, or a variant like the free GSnap) below. But first I thought I’d highlight their misuse to illustrate the effects we usually try to avoid.
So, listen here to 10 of pop music’s most blatant auto-tune abuses:
If you’re unfamiliar with Auto-tune, and especially if you listen to much pop and rock, you might not hear it initially. When overdone, the effect yields an unnatural yodel or warble in a singer’s voice. But the sound is so commonplace in modern mainstream music that your ears may have tuned out the auto-tune!
The songs in this clip, in order, and the phrases most affected by auto-tuning to help you spot them:
Dixie Chicks – The Long Way Around – Noticeable on “parents” and “but I.”
T-Pain – I’m Sprung – Especially obvious on “homies” and “lady.”
Avril Lavigne – Complicated – Listen to “way,” “when,” “driving,” “you’re.”
Uncle Kracker – Follow Me
The whole vocal sounds strained, but especially the word “goodbye.”
Maroon 5 – She Will Be Loved – Listen for “rain” and “smile.”
Natasha Bedingfield – Love Like This – “Apart” and “life.”
Sean Kingston – Beautiful girls – “OoooOver” doesn’t sound human.
JoJo – Too Little Too Late – Appropriately, “problem” stands out.
Rascal Flatts – Life is a Highway
Every vocal, foreground and background, is treated, but “drive” in particular.
New Found Glory – Hit or Miss – “Thriller”, and every time Jordan sings “I.”
When used noticeably, an auto-tuner produces what most call “The Cher Effect“, named for her trademark sound in the song Believe*. (In essence, we named the effect like scientists naming a new disease after its first victim.) Treated this heavily, a vocal track sounds synthetic, and obviously processed.
But not all auto-tuning is so blatant. In the sample above, it’s harder to hear the pitch correction on Uncle Kracker and Avril than on T-Pain and Bedingfield.
As with any tool, a little care can yield great results. Some simple things to keep in mind about pitch correction tools:
Performance: Most importantly, an auto-tuner isn’t a shortcut to a perfect performance. If you can’t sing the song properly, no amount of post-processing will make it sound like you did. So when your pitch matters, and you don’t want to correct it with an effect, you’ll need to work on your performance until it’s right.Less is more: The fewer notes you correct, the less obvious your use of an auto tuner will be. Consider automating the plugin so it acts only when most needed.Graphical mode: If your pitch correction software offers a graphical mode (like Antares Auto-Tune and Melodyne,) learn how to work with it. The default “auto” modes are OK for basic corrections, but often produce noticeable yodeling.Backing vocals: In general, you can get away with more pitch correction on backing vocals than lead vocals.Outdated: Obvious vocoder-style autotuning is dated, and borders on kitschy. The synthetic warbling vocal sound marks songs as having come from a specific era, the same way gated-reverb on drums instantly places a song in the 1980’s. Remember: If you make the auto tuner obvious, people will say your song uses “the Cher effect.” Let this be a guideline.Two songs have auto tuners on my mind today: Snoop’s Sensual Seduction (because of Anil Dash’s ruminations on the death of the analog vocoder,) and Natasha Bedingfield’s Love Like This, which I heard on the radio. In the former, the auto tuner is clearly a gimmick. But every time I hear Bedingfield’s song, I’m struck by the same question: Why do that to her voice?
She’s a fantastic singer, and once you’ve heard the song without the cheesy auto tuner effect, it’s hard to take the radio single seriously.
And there’s a lesson in that for home recordists, (even those of us who don’t write pop music,) which echoes the rule of mixing: If an effect significantly changes the sound of a track, especially one so important as the lead vocal, be sure that change improves the song before committing it to the mix.
See Also: The Rule of MixingFor more home recording tips,
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Friday scraps
A few Homerecording.com regulars debate the merits of dithering. The conversation could easily have devolved into a flame war, but the participants kept it civil, and offered some great food for thought.
Some engineers even argue over which type of dither is best, claiming this algorithm is more airy sounding that that one, and so forth. But just because everyone believes this, does that make it true?
That quote comes from Ethan Winer’s great summary of his position on the matter – he’s squarely in the “dithering is usually not needed” camp.
I tend to agree with Ethan. Responsible mixing engineers don’t apply processing to a mix if they themselves don’t hear the effect of the processing. Simply put, if you can’t hear a difference, don’t make the change.
Unmitigated awesome: Daved Lee Roth’s vocal track from Runnin With The Devil, solo’d.
Converting Ikea bedside tables into studio racks: “the Rast bedside table makes a snug rack for music machines.”
Two unrelated sites feature famous songwriters discussing what went on behind the scenes as they wrote:
First, Joni Mitchell on the writing and recording of her most recent album:
When I recorded it, I was sick so a doctor prescribed some penicillin, which I had an allergic reaction to. I was delirious, stressed out, and we worked all night long. I was so delirious that I was playing way back on the beat… [I]n January 2007, I had demos of the Shine songs with me and played them to some friends at a party afterward. James Taylor told me that he had to play on this song. I wasn’t sure if anyone could because it was created in such a rare spirit. But James came in anyway and I asked him to play short figures like a saxophone. So you can hear fractions of James’ guitar playing here.
Jim Vallance’s site has some fantastic insight into the mind of a professional songwriter. Jim, who’s worked with Aerosmith, Ozzy, Bryan Adams, The Scorpions, and Thornley, meticulously lists every song he has ever written. The site is full of anecdotes and details about his creation process.
Tags: arrangement, hearing, myths, vocalsOn our very first basement demo of “Summer of ‘69? we started the song with the 12-string riff, exactly like the “break down” section in the middle of the song … but on subsequent demo’s we replaced the 12-string with a chunky 6-string intro. In fact, we toiled over the musical arrangement for several weeks, maybe longer. We recorded the song three or four different ways, and we still weren’t convinced we had it right! Bryan even considered dropping the song from the Reckless album.
Now, 20 years later, when I hear “Summer of ‘69? on the radio, I honestly can’t remember what bothered us.
Sunday, November 7, 2010
Quick Home Studio Monitor Tests
I keep a collection of audio samples designed to help check my monitor setup. Test tones, essentially, that I use after I’ve moved my speakers or desk, to ensure the speakers still behave as they should.
I’ve included 4 of the samples below, and I hope you find them useful – and possibly enlightening. Each tests a facet of the two most common monitoring problems in home studios: Uneven bass response, and poor stereo imaging.
Contents: A sine wave sweeping from 40Hz to 300Hz.
Use this to test for: Bass response, sympathetic vibrations.
Unless you’re outdoors, or listening on headphones, you’ll notice the volume rising and falling as the audio plays. That’s normal, although the level doesn’t actually change. (Open the MP3 in your DAW to confirm this.) Rather, you’re exposing the acoustic response of your room.
Use this test as a rough gauge of how extreme the acoustic issues are in your space. (You can flatten the response somewhat, but acoustic treatment is a topic unto itself. For some more information, check the quick backgrounder on home studio acoustics.)
Additionally, the sweep can expose low-frequency dependent rattles, buzzes, or other sympathetic vibrations happening in the area around you. With this test, I once discovered the casing on an overhead light shook at exactly 140Hz, after puzzling with a mix for 15 minutes, unable to isolate the odd rattling sound.
Contents: Consecutive semitones from G1 (46.2Hz) to F3 (174.6Hz)
Use this to test for: Bass response, specific problem notes.
Here, the tone ascends through a chromatic scale. Certain notes will jump out or disappear, for the same reasons as above. Remember these notes, as they’re important to the character of your mixing space. Specifically, when you know that, for example, the B at 61Hz drops in volume in your space, you can reconsider when you find yourself reaching for the fader every time the bass guitar plays B.
Contents: 5 bursts of white noise at different pan positions.
Use this to test for: Coarse panning issues.
This file plays sound at the center, hard left, hard right, half left, and half right. If you don’t hear 5 separate panning locations, you’ve got stereo issues!
Most stereo imaging problems are caused by incorrect speaker configuration (i.e. the speaker aren’t equal distances from your ears,) and poor room acoustics.
Contents: White noise at 3 different pan positions.
Use this to test for: Fine panning issues.
This file plays a sound at 50% left, then hard right, then 25% left. (The jump to the right distracts your ear so it can’t track the sound moving from 50% to 25%) The 3 sounds then repeat on the other side.
Most listeners can reliably distinguish 5 or 7 distinct pan positions. So if your stereo imaging is clear across 9 points, i.e. 25% increments, you’re in good shape (for mixing in a home studio, at any rate.)
On the other hand, if the difference from 50% to 25% isn’t clear in your monitors, or is more defined on one side, you might want to consider using headphones to verify your important panning decisions.
Note: Since these test don’t require high fidelity, MP3s should be fine for checking your setup. However, here are links for WAV versions of the test:
Sine Wave Sweep – 40Hz – 300Hz
Consecutive semitones from G1 (46.2Hz) to F3 (174.6Hz)
White noise at 5 pan positions
White noise at 3 pan positions
For more home recording tips,
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Using Delays for 3D Sound Placement
The easiest way to move a track “back” in a mix is to lower its volume. This works because in our everyday lives, sounds get quieter as they recede from us, so we’re accustomed to the effect.
But our brains also use other cues to determine distance. For example, human hearing excels at matching a sound with the echoes and reflections it causes, to localize its source. And we can apply this principle to add realism when creating the sound stage in a mix.
Consider this picture, and the accompanying audio samples below.
In the scenario illustrated above, sound from the guitar reaches the listener almost immediately, whereas the reflections off the rear wall make a 40-foot round trip, and therefore arrive 40ms later. (Sound travels approximately 1 foot per millisecond.) With the drum kit, on the other hand, the direct and reflected sounds arrive at almost the same time.
The series of events goes something like this:
Guitar sound arrives at listenerDrum sound arrives at listenerDrum sound reflected off rear wall arrives at listenerGuitar sound reflected off rear wall arrives at listener
Our ears and brain are sensitive to these differences in sound arrival time, and use the information (along with other cues, like volume) to judge where a sound source is located in the space around us. Our brains know that sounds and reflections arriving together at our ears must have originated close to a wall, where sounds that arrive much before their reflections must be close to our ears.
Here are two short instrumental samples, both mixed from the same raw tracks, to illustrate how this can apply in a mix.
In the first sample, I’ve placed the drums closer by adding a delay between the direct drum sound and the reverb, so the reflections arrive 40ms later than the direct sound – which tricks our ears into hearing a 20ft distance between the drums and rear wall, as with the guitar in the above diagram:
In the second sample, I’ve simulated moving the drums further back by having the direct sound and reverb occur together, both 40ms later than the guitar.
Note that the levels are the same in each clip. I changed the delay times only, to illustrate the effect.
Caveat: The illustration above is grossly over-simplified. Sounds in a real room reflect off all the walls and surfaces, not just the rear wall. And our ears depend on much more than just timing differences to determine distance. But for the technique at hand, those complications generally aren’t important. The idea here is to trick listeners’ brains by exploiting a property of their sense of hearing, and whether there’s one wall or 4, human ears and brains interpret reverberant sounds the same. (If your listeners are mostly non-human, then all bets are off.)
Implementation: In Sonar, I configure sends (i.e. busses) with delay plugins for each delay time that I need, and I route tracks accordingly. But any platform that allows bussing or routing the signal can accomplish the same end result. So long as you can independently control the delay on the direct sound and on the reverb, you can manipulate the relationship between the two as described above.
Other levels: In practice, you’ll also reduce the level of the drum kit somewhat to make it sound more distant, and adjust the reverb level as required to make the effect more obvious.
As an addendum: Most reverb units and plugins have a pre-delay setting for controlling the delay between the input sound and the reflections it generates. Pre-delay serves exactly the same function as placing a delay between the direct sound and the reverb. In essence, it “moves” the sound further from the simulated reflecting surface. So if your reverb unit or plugin supports pre-delay, you can accomplish much of the above technique without a separate delay plugin.
And remember this simple guideline when using reverbs for realistic 3d sound stages: To bring a sound forward in the mix, increase the pre-delay.
See Also: Reverb on kick drum, Reverb possibilitiesFor more home recording tips,
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Saturday, November 6, 2010
Improve Your Recordings and Mixes, on the Cheap
Some of the easiest ways to improve your recordings are also the cheapest. In fact, the most effective techniques require no money at all.
Here’s a collection of tips you might find helpful the next time a pricey piece of gear stands between you and great recordings.
Have a friend perform: Home recording, especially for singer/songwriters and electronic musicians, often involves a single musician writing and recording all the music. But artists in this situation can find themselves too close to the song, at mix time, to make decisions critically.
Working with other musicians might initially complicate recording and mixing. However, creating a great mix depends, in part, on your ability to remove unnecessary details, and most of us are more comfortable objectively critiquing someone else’s work. So asking a friend (or some professionals) to perform a track or two will ultimately make mixing easier, and more effective.
Get more ears on the mix: With any task requiring attention to detail, it’s easy to lose the forest for the trees. And so it goes with mixing. A second or third opinion can draw your attention back to details you’ve glossed over.
And outside opinions needn’t come from other musicians and engineers. (Although the homerecording.com MP3 mixing clinic is a great source for free advice.) Often, regular listeners give the best feedback because they don’t think in technical terms about the production, and instead form their thoughts on how the song makes them feel. And some of the best mix feedback I’ve gotten has come from children, who are unconditioned by musical convention.
Listen on multiple systems: Hearing a mix through different speakers is a little like getting a second opinion. And professional mixing engineers rely on this technique. Chris Lord Alge, for example, keeps a portable radio near his console for checking mixes:
[E]very client who comes in here wants to hear their mixes on it. If it doesn’t sound good through 2-inch speakers on your little boom box, what’s the point? It’s got to sound big on a small speaker.
Avoid dogma: Our hobby (or profession, if you’re lucky) is plagued with religious arguments, like “tube gear sounds better,” and “analog sounds warmer than digital.” Regardless of each argument’s merit, these dogmatic issues over-complicate the recording process, and distract us from the importance of technique – which, of course, costs nothing!
Cut. Ruthlessly: As musicians, our egos push us to put everything we’ve got into every part we record. But virtuoso performances and great recordings don’t necessarily go together. The whole, as they say, is often greater than the sum of the parts.
In most song arrangements, over-instrumentation usually just leads to clutter. And along with being more difficult to mix, clutter rarely sounds good.
The so-called “car test,” checking a mix though car speakers, helps gauge the overall balance of a mix rather than the translation of small details. So instead of burning a CD of every mix you want to check, transfer the mixes to a cheap MP3 player. You may lose tiny details with the MP3 compression, but you’ll still be able to judge if the bass is too loud or the vocals are too quiet, and you’ll save time and money in the long run.Make every part do work: Ensure that every part competing for the listener’s attention is supposed to compete for the listener’s attention.
Practice your performance before hitting record: The benefits of practice should be obvious to all musicians, but home recording fosters a “write as you record” approach to song creation.
Practice takes time. But it needn’t hamper the creative process; and in most cases it will ultimately save time. Though the tracks may take longer to record, it’s far easier – and quicker – to mix a set of well-performed, polished performances.
Not only do the performances themselves benefit from practice, but the final mix will sound more professional.
Use reference CDs: No single technique will do more to improve the quality of your mixes. Working with a reference mix is, in some ways, like getting a free lesson on mixing from a professional engineer.
Practice mixing when you’re not in the studio: Every mixing engineer should spend time listening critically to professional mixes. Set aside some time every day, say 10 minutes, to immerse yourself in a mix someone else has done. Consider the panning, which instruments take your focus, and how the focus changes as the song evolves. Try to determine the effects in use, and why they were chosen. In modern pop and rock mixes, the interplay between the lead vocal and the snare drum is particularly important, as is the bass guitar/kick drum relationship, so spend some time analyzing these parts in detail.
See Also: Create more professional home recordingsFor more home recording tips,
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7 Questions from Amateur Mix Engineers
Over time, I’ve noted several questions that arise repeatedly on the web’s home recording forums. Each question reads as though it should have a simple answer, but none of them do. And indeed, the questions themselves betray their askers’ lack of experience with the subject.
In effect, posing one of these questions tells the world you’re an amateur. But I hope that by explaining why the questions don’t have the simple answers a rookie expects, you’ll appreciate how an experienced engineer thinks about each problem, and be better equipped to identify gaps in your own knowledge.
1. What are the best EQ settings for guitar?
Or its many variants: “What are the best compressor settings for vocals,” “what reverb settings should I use for mastering,” and so on.
This question has a straightforward answer: The best settings are the ones that sound right. But for most beginners, who haven’t yet learned critical listening skills, this advice seems trite.
Unfortunately, any other answer is meaningless. Every track, in every song, has its own unique requirements. And the best settings, for EQ or compression or any effect, are dictated solely by the requirements of the song. (See the Rule of Mixing for more.)
2. Which is the best microphone?
We’d all love to own a U87 or a C12. But engineers covet those mics because they’re reliable and versatile, not because either is inherently superior. In fact, there are as many ways to define “best” (and for that matter “worst”) as there are sounds to record. As with the question above, what’s best ultimately depends on what fits the song.
3. How do I record my song to sound like The Foo Fighters?
This question stems from the misconception that The Foo Fighters, or any band, sound the way they do because of their equipment. Acquire the same instruments and mics, the thinking goes, and you can duplicate their recordings.
Most professional recordings have deceptive clarity. They sound, at least to listeners unfamiliar with the process, as though they should be easy to reproduce. But the question above has only one honest answer. To sound like The Foo Fighters:
Buy quality instruments, and learn how to play them well.Write songs suitable for the genre.Arrange those songs to support Foo Fighters-style production.Practice. Lots. Record in a great live room.Spend time on microphone selection and placement.Play every part till you get it right.In other words, there are no shortcuts, and it’s not easy. Great recordings take time and talent.
4. What vocal chain does Paul Simon use?
Also commonly worded as “I want to sound like John Mayer. Which microphones and settings should I use?”
Beginners ask this question assuming that we can recreate a track by knowing how it was recorded. Unfortunately, even if you bought Paul Simon’s complete signal chain, you’d have little success matching his recordings. His voice, and John Mayer’s voice, and of course the voice of any famous musician, is unique, as are his performances.
To sound like Paul Simon, in short, you need to have him sing your vocal
5. How do I remove the room’s ambiance from a recording?
Conceptually, it makes sense that since we use reverb to add depth, there must be some way to reverse the process.
There isn’t. If you don’t notice until you’re mixing that a guitar track has too much room sound, you have 2 options: Live with the sound, or re-record.
6. Is this mix finished?
Rookie engineers like to think there’s a golden standard sound to which they aspire, and once they’ve attained that sound, their mixes will thereafter be perfect.
We should be so lucky! In truth, our learning never stops. We continue (hopefully) to improve, but none of us is ever done acquiring knowledge, as true of recording and mixing as it is of life. But this is OK. Learning, after all, is the fun part!
To the question: As a general guideline, a mix is finished when it best represents the song. Of course, “best” is open to interpretation here as it is everywhere in recording. You need to use your ears and your gut, and make the call when it feels right. In other words, only you know when the mix is finished.
Unless someone has paid you, in which case the mix is done when the deadline arrives.
Finally, a surefire question to signal your newbie status to the world:
7. How do I use this $1,200 plugin that I just happen to have installed on my machine?
Answer: You read the manual, which comes with the software when you buy it legally.
You’ll out yourself as a novice by asking these questions of an experienced engineer. But really, there’s nothing wrong with that. In some senses, we’re all amateurs.
Take the colleague of my friend Paul, who once asked him, “what does a compressor do?” The question seems innocent enough until you learn that this colleague has been a film industry sound engineer for over 20 years, and has worked on dozens of major motion pictures. Of course, Paul now has difficulty taking his colleague seriously as an audio professional. But the guy still works on movies as a sound engineer, so the anecdote should be comforting for the rest of us amateurs!
See Also: Tips for more professional recordingsFor more home recording tips,
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Friday, November 5, 2010
EQ – “Cut narrow, boost wide” explained
This tip arises in most discussions of good equalizer technique: “Use narrow adjustments when cutting frequencies, and wide adjustments when boosting.”
There are some great reasons to heed this advice. But they’re not immediately obvious, especially if you’re unfamiliar or uncomfortable with parametric EQs, and they’re rarely fully explained. I’ll explain and demonstrate below, and you can use the information to improve your EQ adjustments, and in turn your mixes.
In brief, equalizers alter the tonal quality of audio by applying gain to a specific frequency range. (For something a little less brief, Sound On Sound’s Equalisers Explained is the best EQ primer I’ve read.)
Every EQ filter has 3 settings: Frequency, Gain, and Bandwidth.
Frequency determines where in the tonal spectrum an adjustment occurs. Low frequencies correspond to bass sounds, high frequencies to treble.
Gain determines the magnitude of the adjustment. Positive values increase the signal level at the specified frequency, and we call this a “boost.” Negative gain values decrease the signal level, and we call this a “cut.”
Bandwidth allows us to choose the range of neighbouring frequencies that our adjustment affects. Bandwidth is usually called “Q” (for esoteric reasons from filter theory.) Higher Q values affect fewer frequencies, and we refer to this as a “narrow” filter. Low Q values, on the other hand, yield “wide” filters that affect many frequencies.
This is easier to understand as a visual:
The diagram above shows 4 key combinations. From left to right:
#1 – A narrow cut – Note the high Q value, and negative gain.
#2 – A narrow boost – Note the positive gain.
#3 – A wide cut – Note the low Q value.
#4 – A wide boost.
Your EQ plugin may not look the same (for comparison here’s the above illustration using Reaper’s EQ) but all parametric equalizers support the same 3 basic options: Frequency, Q, and gain. And using these options, we can “cut narrow, and boost wide.”
In practice, wide EQ cuts remove more signal, and therefore more of a sound’s defining characteristics. Remove too much signal, and the audio you’re treating no longer sounds like itself. This can certainly produce interesting effects, but it won’t yield accurate mixes.
Narrow surgical cuts, on the other hand, remove only specific frequencies, and as such leave the signal largely unchanged. The narrowest cuts can be practically inaudible, as they remove so little from the sound. Often, we use narrow cuts to remove only “problem frequencies,” such as ringing overtones from a drum or boomy resonance from an acoustic guitar, without affecting the overall character of the sound.
It might seem the same should be true of boosting – that narrow boosts are the least audible. But in fact, because of how our ears work, narrow EQ boosts usually sound unnatural and jarring, where wide boosts are much less obvious. (The reasons behind this involve science a little beyond the scope of this article. Summarized: Human brains evolved an innate understanding of the harmonic series, and narrow EQ boosts affect specific harmonics, producing timbres that we sense can’t possibly have occurred naturally.)
The effect should be clear in the examples below. These 5 audio files illustrate the various extreme EQ adjustments. First, an untreated track:
In the next sample, I’ve used a narrow boost at 2060Hz. [diagram] 
The ringing is immediately apparent, and sounds unnatural and distracting. (Your ears and brain sense, based on the other frequencies, that there shouldn’t be a loud harmonic at that frequency.)
Now, here’s a wide boost at 2060Hz. [diagram] 
While the sound might not be great, the ringing effect introduced above isn’t apparent, because the boost affects so many other frequencies:
The next example illustrates a wide cut at 2060Hz. [diagram] 
Notice how much of the guitar’s character disappears:
Finally, in this example the narrow cut is barely audible at 2060Hz.[diagram] 
All we’ve done is remove the ringing frequency, though since it wasn’t readily apparent in the original sample, its removal is hard to hear.
These examples were contrived to illustrate an effect. (i.e. You’d never actually apply at 14dB boost at 2060Hz to an acoustic guitar track.) However, the principle applies regardless of the audio with which you’re working.
Note, too, that this technique is relevant only to adjustments made with parametric equalizers. Graphic EQs have a fixed bandwidth at each frequency, so “narrow” vs. “wide” cuts aren’t possible.
Finally, and perhaps most importantly, the advice is generally useful but NOT a set-in-stone rule. Sometimes, a ringing effect or hollowed-out sound is exactly what a mix requires. As with everything in audio engineering, let your ears be the final judge of what works best.
See Also: The Rule Of Mixing, General EQ GuidelinesFor more home recording tips,
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Thursday, November 4, 2010
Who Needs Fancy Preamps?
Brandon Drury quit counting at 1,200 recorded songs in his busy home recording studio. He is the creator of RecordingReview.com and is the author of the Killer Home Recording series. For some time I've been trying to make sense of this preamps mess, which I find exceptionally boring, and figure out how to teach it here at RecordingReview. It's a tricky mess for a million reasons, but in this article I'm going to try to add the proper context to this thing so that you younger guy, older guys, whoever can figure out where the hell you stand on preamps.
First off, you've got to figure out what your engineering goals are. Are you just wanting to have a little fun making noise? Are you looking to be the flat-out best engineer you can possibly be? Are you looking to put together a pro facility in which you could record tip-top bands? Are you somewhere in the middle.
I think the biggest reason so many people are mislead on this whole preamp thing comes from not understanding a person's needs. I've been struggling for a while in explaining why I've acquired some fancy pantsy preamps but then rarely recommend this upgrade to anyone here at RecordingReview.
#1 I'm of the opinion that a person can definitely make exciting music with gear that is “just okay”. Stock interface preamps fall into this category, for example. Even in a not-so-hi-fi state (which high end pres in no way guarantee), exciting music means the recording IS exciting. This was the motivation for this blog: Preamps Don't Matter?
I'd prefer to listen to great music with cheapo preamps over stale, boring crap any day of the week. Everyone with a pulse is this way. If they say differently, avoid them like a gay rapists with big guns.
Good New For: Broke and talented people
Bad News For: Skilled engineers recording uninspired noise, anyone untalented
#2 Dumb metaphor time: If cash is no object, you get the lightest gear possible for 10x the price when climbing Everest because too many people have died trying and that last 100 ft is harder than it looks.
Even the best engineers are going to have a hard time with that “last 100ft”....the thing that separates the ultra men from the men without the fancy preamps. Of course, #1 is still applies here.
Good News For: Skilled engineers with deep pockets who want to be ultra-skilled engineers, hobbyists who aren't attempting to “climb Everest” and therefor don't need idiot-gadgetry anyway
Bad News For: Broke people climbing Everest
#3 There are a billion places a person can screw up a recording (with or without the high end preamps). Fancy pres don't bail you out of bad sounding situations....ever!
Good News For: People who've really got their shit together......which is basically no one
Bad News For: Anyone trying to make a butt kicking recording
#4 The two main problems with cheapo pres is they sound “blurry”....kinda like a VCR tape where stuff is kind of smudged/smeared or they take on the megaphone effect a bit. How much? Not THAT much, but it's certainly something no one I know would prefer.
You can clearly hear what I'm talking about in The Interrogator Sessions in Killer Home Recording. Once your ears are acclimated, it doesn't take much work to hear how the cheapo pres compare to the high end pres. There's never a time when the cheapo pre would win. The top notch pres always have this “extra excitement” in them while the cheapo pres seem more “sluggish”.
I divide the preamp thing into three categories: cheapo, adequate, and fancy. Once you get passed the cheapo stuff, the adequate pres do the job just fine. They may not add anything interesting, but they don't murk up anything and they don't add the megaphone effect. The True Systems pres fall into this category, among others. I thought the pres in the Yamaha MR816 weren't TOO far from being adequate, but they didn't quite make it. I've rumors that the pres on the RME Fireface 800 may be playing not too far from this ballpark. So when you read reviews about these upper-range interfaces sounding “amazing”, they really just sound “almost adequate” if we take the Total Idiot stance from above.
Adequate preamps do a fine job and could be used on everything without any real issues.
The special pres do a certain something extra. This “extra” thing could be good or bad depending on what you are going for (we'll get into that) and this is where knowing exactly what each preamp is ideal for comes into play. Certain special pres push you into a corner a little bit.
#5 Not all fancy mic preamps are ideal for use on everything. Big, dark sounding preamps can cause big problem when using them on everything. Too many “big” sources makes mixing a challenge. It's best to use the big sounding pres sparingly on the bigger stuff and use the tighter pres for everything else.
When I started out with my Vintech 1272 on the very first recording I had ever done (I was told I absolutely NEEDED it!....asshole!) Anyhow, that preamp is more on the bigger/thicker side of the fence. It doesn't have the hi-fi sparkle that many pres do, but it does have a the Neve-style harmonic in the upper mids. All Neve-type pres have this and it's a dead giveaway in shootouts. The Vintech does not have the hi-fi Neve thing that the Great River Neve-style pres have. It's more of a darker sound.
This is not a preamp I would recommend starting with. Even though I do pull it out for certain things some of the time. The Vintech 1272 can be great on vocals that you don't need to be ultra bright (particularly with an SM7b among others) but want to sound big. The lack of sparkle paints the 1272 into a corner that's only useful at certain times. It kind of reminds me of when mom puts the special tablecloth on at Christmas. It's only great once a year.
For what it's worth, the pres in the Yamaha MR816 don't have near the sparkle of a Manley TNT solid state channel, Martech, or Great River either, but the more neutral approach to their design makes them more usable across the board. There's something impractical about certain high end pres as a daily driver. A person would be better off with a new Ford Focus than a 500Hp Hemi Cuda if they only had one car. For a person who only has single-car garage, there are powerhouse BMWs and Mercedes that can do the daily driver thing and can do it with super high performance. This is where the high end sparkly preamps mentioned above come in.
In short, spending big bucks on some random pre is not the solution and could actually get you into a place that is worse than decent interface pres.
Good News For: People with a wide variety of pres who know how and when to use them, People who have a single faster/tighter preamp they use on all overdubs
Bad News For: Guys overdubbing exclusively with one fancy preamp that may not be ideal for across the board use
Hobbyists – If you are a guy just having fun, don't buy a fancy preamp. If you want your recordings to sound good but have limited time and aren't going to get upset if your productions don't blow away the big boy bands, just have fun and don't let anyone convince you need to make a huge investment.
Crazed Hobbyists – If you do this for fun, but have a few bucks to blow, have a little fun with adequate preamps first. You may not need a big selection and your needs will depend on if you are entirely doing overdubs or if you are recording multiple tracks at once. However, in this world, I recommend preamps on the sparkly side of the fence with tight low mids. You saw a few examples above. The True Systems stuff definitely gets the job done in this realm, but if you want to go all out the solid state Manley TNT channel is my go-to preamp for sparkly stuff. The Great River gear works really well. I definitely wouldn't go with something not-so-tight in the low mids for my single pre again. Those were some long years!
Total Idiots – If you are going for the top and climbing Everest, I'll tell you what I've done. I just bought a Toft ATB32. It supposedly has okay pres to hold me over when I run out of fancy stuff. I picked up an API 3124, a Wunder PaFour, and a Focusrite ISA 428. I'm keeping my Manley TNT and most likely selling everything I've had previously. (I haven't decided about my Chameleon 7602s yet.)
I expect the API to end up on close mics on drums and probably most things I'm overdubbing. It doesn't have a super sparkly top end and it's quite colored. It'll smooth out shrill stuff. My Manley TNT solid state is definitely sparkly and doesn't smooth out anything. The X factors here are the PaFour and ISA 428. I'll have to report back when I've put them to the test, but the Wunder should do more of the Neve thing with the extra harmonic in the upper midrange. The ISA should be sparkly like the Manley TNT SS, probably a little less colored, maybe a little faster. I'm speculating.
For what it's worth, there are only a few qualities that are that important. Why a person needs 22 different models of preamps is beyond me.
Meaty TransientsSparkly preamp with tight low mids (for brighter vocals, acoustic guitars)Tamed Upper Mids, Harmonic Content (for most vocals, electric guitars)You can make it much more complicated if you want to, but I don't see the reason. I really do think the old view of just using a console worked on most the cds in my collection. I've leave the optional hair/atom-splitting up to you. I've got noise to make.
The tighter, more focused pres are where it is at for day to day tracking. There are times when you want the bigger, darker, and less-focused sound but the most part I'm not into it. In the tighter/focused category you have all kinds of options to choose from and various attributes that make them more or less ideal.
Here's a 1992 Chicago Bulls basketball analogy that I think sums it up. Michael Jordan is API. BJ Armstrong is Great River. We are recording my “sparkly” acoustic guitars again. (Work with me, I know this sounds, and is, stupid.) BJ Armstrong is the point guard and so it's his job to bring the ball up the court. Is it going to really be the end of the world if Michael Jordan has to handle the ball? Probably not. Basically, we get an extra point in the sparkly department by selecting the Great River. We don't necessarily lose anything with the API. Your girlfriend probably won't notice and the bass player definitely will not notice.
Then again, when all this stuff aligns and you select the pres that give you what you want at various times (and combine that skill with an equal amount of across-the-board engineering skill) the band will definitely notice.
Broke home recorders don't have to worry too much unless their aim is absolute robo recordings. A hobbyist will usually have goals that are more modest than the guy engineering the next INSERT BIG BAND's album. Some hobbyists are dealing with limitations in music, musicians, instruments, room, and engineering ability that hold them back and in any of those situations, the fancy preamp thing is unnecessary.
For the crazed hobbyist, If the wrong pre is used throughout a recording, a person can have their work cut out for them in mixing. Because of this, I highly recommend tighter, more focused pres to be used overall and then, in time, adding a big sounding pre for special situations (vocals, kick, etc).
For the guys going for the absolute echelon of ultra recordings, there are specialized tasks for various pres than can make life a little more fun. Understanding when to use what is part of the skill of a great engineer. While there have certainly been numerous recordings made with a console only, and great music music will always come through, most people agree that there are sound quality benefits to specializing.
Tags: API, Great River, Manley TNT, Mic Preamps, Vintech 1272
This entry was posted on Thursday, October 14th, 2010 at 3:05 am and is filed under Audio Engineering Principles, Mic Preamps. You can follow any responses to this entry through the RSS 2.0 feed. You can leave a response, or trackback from your own site.
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