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CN Fletcher

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  1. The best way to "master" songs is to take them to a qualified "mastering engineer". While you can read a book or ask on an internet forum how to remove your appendix, there are some things that are best left to professionals. Mastering, while not quite as critical as surgery, is still one of those things that is best left to professionals. Scott Hull [Masterdisk], Brad Blackwood [Euphonic], Jeff Lipton [Peerless], Bob Olson [Georgetown] are the guys I would contact first... then there is Dave Collins, Ted Jensen, Howie Weinberg, Bob Ludwig, blah, blah, blah, blah, blah. These are seasoned professionals who can take your demo to come a hell of a lot closer to sounding like a record that you will be able to do at home on your best day. If nothing else, spend the money to book a session with one of these guys and have them at least begin to explain the process to you. They have decades of experience, are exceptionally good at what they do... but are also really cool guys who will be happy to take some time and at very least give you some of their knowledge with which you can begin to experiment. Peace.
  2. Okay, let's do the math. An M-Audio Audiophile sound card costs about $100 for two channels and is audibly transparent. So that's $50 per channel. My definition of audibly transparent means you can record audio through it and not hear any change when played back. By "you" I mean me, and the OP, and Fletcher, and everyone else reading this. If two channels isn't enough M-Audio has their 8-in/out rack-mount Delta 1010 for $500, but the "breakout cable" version costs only $200. So now we're down to $25 per channel for audio clarity that's as good as needed. It doesn't have to be M-Audio either - there are plenty of other brands and models in this price and quality range. I agree with you 100 percent that all home studio size control rooms need plenty of bass traps and other acoustic treatment to hear accurately. Of course, they also need good speakers. Good acoustic treatment can be had for only a few hundred bucks if someone is willing to put in the effort to research what is needed, then buy the materials and make their own. Buying high quality commercial treatment is more like $1,000 and up, and $3,000 isn't uncommon for really good results. But they should spend that much on speakers too! Ethan... you've said some pretty stupid {censored} over the years... but this may be the dumbest thing I've ever read by you. I can very much hear the difference. I can hear the difference between Digi converters, Apogee converters, iZ converters, Burl converters and my favorite of all [weighing in about $3,250 per channel] the JCF "Latte" converters. Anytime you want to come up to Foxboro with an "M-Box" and shoot it out with some "iZ" converters I'll make the arrangement for you. The "iZ" converters in my RADAR are only like $750/channel... but there are 24 channels to the unit [we have an extra 24 in our iZ "ADA" unit... which is used for "insert patching" when we're mixing "in the box"... I don't remember how much they go for... I think it's a little under $500/channel]. Then again, I spent around $60,000 on the construction of my room... so I guess it's easier for me to hear the difference. Peace.
  3. Around $100 per channel IMO. --Ethan ...as long as they spend several thousand on "room treatment" so they can hear why $100-/channel converters suck balls.
  4. I will be expanding my array of studio gear in the near future, and want to know how these more expensive interfaces make up for the steeper price - is it just the preamps? the software? Not "the pre-amps" per se... but the "analog front end" is exceptionally important. How the anti-aliasing filters affect the audio quality, how much headroom the input [or output] circuit has in the analog domain as well as things like over all frequency response [prior to the anti-aliasing filter] which affect phase response and distortion. These things all come into play, and are all "analog issues" prior to conversion. You also have clocking issues... the stability of the clock, where the instability of the clock sits in the audio spectrum, yada, yada, yada. In other words, yeah... the "interface" can and does make a HUGE difference... the unfortunate part is that every "inch" forward is generally exponentially more expensive than the last inch you improved. Peace.
  5. A "Champ" or a "Princeton" are great... I've had very good luck with my un-modified "Valve Jr."... but our speaker cabinet collection is pretty sick and as Dan pointed out... speakers are equally as important as the amp. Peace.
  6. There's no scientific proof of hearing (or sensing by some unknown means) of frequencies much above 20kHz. Bull{censored}. It has been proven in several university studies. I first found out about this phenomenon when I was out to dinner with Rupert Neve and his wife Evelyn. Rupert told a story about a desk that had been delivered to AIR Studios in London when he got a call from Geoff Emerick commenting that a couple of the modules sounded different than the others. The next day Mr. Neve and a few from his crew went down to the studio and pulled the modules that sounded "different". They found a manufacturing error where the modules were shipped with the transformers having been left in an unloaded state causing a VERY high frequency oscillation. The sound of these modules had a palpable difference from the other modules in the desk [and served as the basis for the Great River MP-2NV's loading switch as the transformers remain 'unloaded' until you hit that switch and flatten out the frequency response]. Peace.
  7. 96/24 generally sounds better than 192/24 as the converter chips for 96 sound better than the converter chips for 192... so while there are some recordings that are done by 'techno babble believers' at 192 I've found that most of the best work is being done at 96k. Now... why a higher sampling rate than what would lead you to 20kHz. First off, the ear may not hear above 20kHz [unless you're a young girl... and I doubt there are many young girls reading this forum]... HOWEVER we can indeed percieve harmonic content well above 20kHz. We proved that with the Great River MP-2NV as it's "flat" when you hit the "loading" button... but if you don't hit that button the transformers are 'unloaded' and ring at around 63kHz... on "airy" material you can hear a palpable difference. The second [and probably more important than the first] reason is that when you have a filter you create phase shift. This phase shift slows down the signal for an octave and change below the filter point... so, if you're filtering at 20.5kHz [1/2 of 44.1kHz] you'll hear the resultant phase shift in the audio down to about 5 maybe 8kHz [well within the range of audibility]... if you're filtering at 48kHz [1/2 of 96kHz] you'll only get the tail end of the effects of the phase shift that has been created in the upper regions of the audible range. One of the major differences in converters is the quality of "alias filter" design [the filter that keeps the digital noise out of the audio]... along with clocking and power supply, etc., etc., etc.... however these differences in this analog filter design give a very palpable difference to the quality and clarity of the audio. Make sense?
  8. 96/24 generally sounds better than 192/24 as the converter chips for 96 sound better than the converter chips for 192... so while there are some recordings that are done by 'techno babble believers' at 192 I've found that most of the best work is being done at 96k. Now... why a higher sampling rate than what would lead you to 20kHz. First off, the ear may not hear above 20kHz [unless you're a young girl... and I doubt there are many young girls reading this forum]... HOWEVER we can indeed percieve harmonic content well above 20kHz. We proved that with the Great River MP-2NV as it's "flat" when you hit the "loading" button... but if you don't hit that button the transformers are 'unloaded' and ring at around 63kHz... on "airy" material you can hear a palpable difference. The second [and probably more important than the first] reason is that when you have a filter you create phase shift. This phase shift slows down the signal for an octave and change below the filter point... so, if you're filtering at 20.5kHz [1/2 of 44.1kHz] you'll hear the resultant phase shift in the audio down to about 5 maybe 8kHz [well within the range of audibility]... if you're filtering at 48kHz [1/2 of 96kHz] you'll only get the tail end of the effects of the phase shift that has been created in the upper regions of the audible range. One of the major differences in converters is the quality of "alias filter" design [the filter that keeps the digital noise out of the audio]... along with clocking and power supply, etc., etc., etc.... however these differences in this analog filter design give a very palpable difference to the quality and clarity of the audio. Make sense?
  9. This is the best economic climate to get a divorce in the last 50 something years!!! If you have the notion that the time is right to pull the plug... sooner is better than later!! Peace.
  10. At the $100- mic level... if you have a pre-amp in something you already own it'll be good enough... no need to stress about it now. Peace.
  11. That phenomenon will pass as you gain confidence in your abilities and skills as an engineer... you'll actually be able to turn off your 'engineering ears' and listen to the music again. You'll hear songs instead of snare drums... you'll hear melodies instead of guitar sounds... on the 5th or 6th listen you'll start to hear the little subtle things that the production team included in the presentation... but you'll hear it in context with the song. While you can never entirely turn off the 'critical listening' aspect of your training, you will find a place where it is no longer your primary focus. Yeah, you'll hear 80-90% of the parts and how they're interacting with the song on the first listen through... but you'll hear the song more as a "presentation" than as a collection of various sounds. When this stuff gets to be old hat you'll only hear the engineering when it's exceptionally good or when it's so bad it interferes with the presentation of the music... other than that you'll find a place where you take the song at face value instead of the "hmmmm wonder why they did that?"... or worse, the "{censored}... I would have done _____ with that _____". All it takes is time... it's a bitch while you get there but eventually you will. Peace.
  12. There are different schools of thought on this, but mine is that the back wall will be the first hard boundary layer the low end energy will encounter. Yes, sub-100Hz frequencies are essentially omnidirectional, but they are still propagated *in* a particular direction based on the location of the transient emission source...the speakers. There are some speaker designs that go a different way (Martin Logan dipoles, for example), but point source speakers like yours will behave this way. I'd rather get the most absorption up on the back wall, which will be the source of some of the deepest valleys and highest peaks....the front wall will be the second boundary, so the 244's will attenuate the low end response further. Frank To a point... trapping heavily in the back of the room is more to kill the energy once you've heard it and to try to prevent it from building up in the room... however, you really need quite a bit of that trapping [on the order of feet, not inches] to do the job properly. Bass goes omni-directional around 250Hz and likes to couple to hard surfaces [like walls... which is why you always experience more bass when you're leaning against a wall as opposed to standing in the room]. One thing you can do that will be very effective is treat your side walls and ceiling as well as the back wall [the floor isn't really practical if you plan on moving around in the room... but a good, thick "shag" carpet can help so long as it's not on the 'dance floor' where you'll be moving your chair around]. As far as trapping at the back wall goes if you can spare a foot or three it's not too difficult to build frames out of 2"x 4" studs and then hang sheets [or strips] of 1/4" plywood from wire that you've covered with rock wool or fiberglass insulation. Make sure none of the strips are the same size!! and have them hanging in space. This way the low frequency energy will be absorbed as it tries in vein to move the piece of suspended light weight wood. It's really best to do these with them in a pattern of 3 running "east-west", 3 running "north-south" and then repeat the pattern [which is why it'll eat up a foot or three of depth in the back of your room]. In the meanwhile... albiedamned... check your PM's for other thoughts and feedback. Peace.
  13. I have an NTK... it's a fine mic. Then again... my NTK has a "non-RODE" amplifier in it that was custom built for me by a friend [and uses NONE of the original parts and has an output transformer]. As for "mic pre's for specific vocal types"... not really a consideration nearly as much as which mic to use for a specific vocal type. In my world the mic pre is decided by a myriad of factors... like which pre's have been used before it on a specific song [i try not to use the same pre twice if I can avoid it]... which mic is being employed, what we're [we as in the artist, producer, moi] are going for in terms of the sonic texture for the vocal... yada, yada, yada. If you have an arsenal of good tool... and an understanding of the capabilities of those tools then you can make an educated guess / decision [if you take the time to do a 'shoot out'] and find what best suits your sense of aesthetic... any thing else is a fun game but it's nothing more than uneducated conjecture. Peace.
  14. Digital reverb has been all the rage for more like 25 years than 15 years... the stuff you like from the 90's was almost entirely [until at least the late 90's] digital outboard effects boxes. The late 90's through modern times [say the last 10 years] have seen plug ins come to be. There are several plug ins I've tried [no, I can't remember their names] that I thought were pretty good... but no one has really been able to do a plate properly [which is kind of a drag]... but other than that, most of the reverb you hear on modern records is from the box. I'll add that the great majority of the boxes suck ass... as do the great majority of the plugins I've heard... you can "get away" with them... but you'll very often have to get into the menus and do stuff like roll off top end... compress the send to the reverb [a "transient designer" is usually my favorite treatment for a reverb send... there are hardware and software versions of them]. Will it "make or break" a recording? Hell no. The song will do that. It will however enhance or detract from the presentation which is [as engineers] our primary concern. Peace.
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