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Post by lcr on Apr 13, 2014 18:29:39 GMT -6
Cool. Thanks for the feedback. Im wondering if I should start with Room or Vintage. I know, I know, get both... I kinda gravitate towards verbs that make the source bigger while somewhat subtle in the mix, I usually don't go for verb in your face.... Hmm, Unfortunately I can't demo either in the AAX64 format.
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Post by lcr on Apr 9, 2014 5:42:46 GMT -6
Anybody using the Valhalla AAX beta versions in Pro Tools 11 on OSX? Wanna try the Vintage Verb but would like to hear that its stable and functioning correctly.
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Post by lcr on Apr 6, 2014 6:47:22 GMT -6
Computers are fun.
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Post by lcr on Mar 20, 2014 17:19:31 GMT -6
Maybe it is more "mixing 101", but maybe it is still useful in this context... Does the order of leveling and equing instruments fit in this thread? May sound trivial, but first thing to get sound right - snare. Then kick. Both together. Rest of the drum kit. Bass. Bass + Kick together. Bass + drums together.... Whenever i proceed without getting the max out of this, i have to go back to it later, definitely, often beginning from scratch, losing time. Also - if i want results quicker, i have to force myself to not beeing afraid of beginning from scratch in favor of trying to fix a non-working mix. I try to imagine a live-mixing/FoH situation for faster mixing. Where you have to make the fundamental mixing decisions quick, and try to make things "just work" without losing focus. If it doesn't work, start over without hesitating, avoiding long discussions how to rescue it. Setting deadlines for a working mix and avoiding "open end" sessions by any means, where i might finish work but become highly ineffective and make more wrong decisions along the way because i lose concentration and because of ear fatigue... Whenever i lose concentration on sound work, getting ear fatigue, etc... i do editing, track organisation, basic stuff, to utilize the time i need for recovery to get my ears back to 'specs'... Hm, hope it's useful in this discussion....dunno... Great Suggestions. I used to start with (drum) room, Overheads, then drop snare in, Kick, Toms, etc. But when I focused on your suggestion of Snare and kick first, things happen better & faster. I usually start with kick, then snare, I should try starting with the snare. Good idea with using your time wisely to rest your ears. Question, Do you typically eq and compress the snare right off the bat, or do you get the whole kit working then start the eqing and compressing? And If you get a static kit mix first, do you process in the same order or different?
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Post by lcr on Mar 20, 2014 8:02:28 GMT -6
I use the same gear on the same channels, set very similar. Most of those settings on the outboard gear are for effect anyway, and the channel EQ is generally to make things fit together a bit better, so I tend to use common frequencies. I also use track and routing templates so I always start out with the most useful DAW setup. I figure that people want me to record them because they like the sound I get, so why should I change it by swapping around a bunch of gear each session? As for mixing the song, I take the Andy Wallace approach and find the most "happening" part of the song and get that sounding good. That not only sets my expectation for the rest, but also sets my optimal levels as well as gets me excited for the song. I've somewhat limited myself to a few eq's and compressors. Really trying to learn them. As far as having "a sound", I think everyone does wether they want to or not.
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Post by lcr on Mar 20, 2014 7:57:46 GMT -6
Many discussions regarding mixing tips to improve the quality of your mixes, which is great. But what about techniques/workflows that help complete a mix FASTER. The first thing I think of that speeded up my mixes is building a static balance of levels at the peak of energy/loudness of the song with all faders in DAW at 0, using the trim plugin in the last insert of the tracks and/or busses, usually the last chorus of the song. This is my starting point for all automation with faders at 0. For me, it's a cleaner more organized way of looking at automation and level changes in sections of the song. I then automate faders from this section to the end of song. Then I automate the previous section up to this last chorus making sure the balance/energy works as the sections flow into each other. This continues until I get to the beginning of the song, basically mixing the song backwards. This helps tremendously with headroom, avoiding hitting any 2 buss processing to hard(if you use any). Im sure this technique is nothing new, it has drastically improved my efficiency. Please share anything that help you complete a mix faster. Yup I do this as well but you described it way better than I could! Also mixing quietly works for me. and liberal use of preset track templates that were made from experience. Yep, CLA talking about monitoring levels(the king of fast mixing!) says he monitors a majority of the time so low he can hear his assistants typing in the background and its sometimes distracting! His philosophy is if its exciting at low levels it will always work loud, and also you don't wear out your ears for the day. I will say I try to keep it low volume, and recently I've been checking levels(only) on headphones. Building the mix on mains always, really everything on mains, but I check my balance of levels on headphones. I may do a slight adjustment on the headphones to levels then A/B that adjustment back on the mains and usually like what the headphones told me.
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Post by lcr on Mar 20, 2014 6:21:44 GMT -6
Many discussions regarding mixing tips to improve the quality of your mixes, which is great. But what about techniques/workflows that help complete a mix FASTER. The first thing I think of that speeded up my mixes is building a static balance of levels at the peak of energy/loudness of the song with all faders in DAW at 0, using the trim plugin in the last insert of the tracks and/or busses, usually the last chorus of the song. This is my starting point for all automation with faders at 0. For me, it's a cleaner more organized way of looking at automation and level changes in sections of the song. I then automate faders from this section to the end of song. Then I automate the previous section up to this last chorus making sure the balance/energy works as the sections flow into each other. This continues until I get to the beginning of the song, basically mixing the song backwards. This helps tremendously with headroom, avoiding hitting any 2 buss processing to hard(if you use any). Im sure this technique is nothing new, it has drastically improved my efficiency. Please share anything that help you complete a mix faster.
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Post by lcr on Mar 19, 2014 4:44:38 GMT -6
Can't you just print the track and then nudge it to line up with the original? Yes, but I am lazy.
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Post by lcr on Mar 18, 2014 13:21:51 GMT -6
lcr : I think you might be right: I have had some hair turn grey over the last years with OSX/FW & OTB processing as I haven't been able to replicate results w/o latency time after time. Although not with UA Appollo thought I'd share this info : I've done latency tests under OSX 10.6.8 with Logic Pro, with a vast array of different situations: sessions w/o plugins, sessions with heavy plugs, sessions without audio, with audio, with aux busses with and w/o inserts, different buffer ranges, sizes, compensating with sample calculations sending click tracks out and back in, even buying a latency compensator plugin that doesn't work. The ping function with the I/O plugin in Logic is everything but accurate. Trust me, I've lost hours trying to figure out by testing how the hell to get it to work, and have no concluding info regarding repeatability with succes. Two best options: do a session-specific roundtrip with a clicktrack ( as different sessions yield different results when using plugs/busses ), then enter the value of delay in the compensation options, and be done with it unless you will add other plugs and busses then you might have to recalculate. The last option that I use, just do a roundtrip of a few bars with your gear in bypass ( a compressor might change the shape of the displayed waveform so you cannot recognize and align ), align the recorded audio and write down the delay in samples, process your tracks, delay them by the number you just calculated. done......well I know this sucks and shouldn't happen...have a Metric Halo 2882 here with rocksolid drivers and use, so it's definitely not the hardware....didn't try with PCIe or USB though, might be better.... one last thing with Logic Pro 9 and outboard: sometimes it's impossible to exactly sample-align the new audio with the one already recorded, meaning intersample stuff just before or after grrrrr....although I cannot hear the difference at such minute zooming detail, it still messes with my brain not being able to see those tracks perfectly aligned....need to try still with Reaper to see if it's possible to obtain a set&forget setup, best of luck and patience ! Oh its def. an OSX issue. I wasted much time trying to get this to work properly in PT10, Cubase 6, and Studio One 2. Your suggestion of using a delay plugin will not work. If you save and reboot, the value of required latency will change. I've read this is a dedicated firewire issue in OSX. If possible(if you are interested), try USB and see if it works. Where I read others confirming this they blamed firewire only and stated all PT HD systems(PCIe based systems) were rock solid, it would be interesting to see if PCIe and USB function correctly.
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Post by lcr on Mar 13, 2014 19:28:49 GMT -6
This Pono thing has me thinking about PT sample rate. Am I shooting myself in the foot by recording at 48K, all because 30 years from now a bunch of people are gonna want to listen to the classic album that I haven't yet written, and if I track at 48, will people say, "cool tunes, but man, the recording engineer was an idiot! They should have fired that guy for tracking at such a crude sample rate! Everybody knows that 192 is THE rate! Even then . . . Must have been an old man producing the band. What's his name, anyway? Oh, the guitarist with the stupid last name that no one can pronounce. What an asshole!" Hey, you got to have a dream or two . . . Please let's not turn this into a sample rate debate, but I went 96k a year ago and like the results. What convinced me was a article I read about how plugins will perform more accurate, less upconverting, it does tax your machine harder. I even turned it up to 32 bit float 96k a few months ago, im not gonna switch back.
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Post by lcr on Mar 13, 2014 8:59:40 GMT -6
I hope this is successful. Not to sound negative, but I think the only people interested in this is us and audiophiles(are we audiofiles?) I don't think the world needs a new music player for your average consumer. I think your average listener doesn't even know about the dissadvantages of cd/mp3. But if this informs the consumer of better options than mp3 and has marginal success and educates people about quality, then Apple will finallly roll out the high res format they've mentioned. Most people still want to use Itunes and iphones. If anything, this will improve the quality of Iphones DAC and Itunes file format in the future. I hope this product is successful, i dont know if I would buy a media player at this price point.
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Post by lcr on Mar 9, 2014 8:09:17 GMT -6
Unless they've changed something since release, UA said the Apollo would not support the use of hardware inserts/compensation. I never understood how one could make that call, since that's on the DAW side and not the interface, but maybe that's the reason. I tried with a MR816 once and couldn't get a solid latency calculation in PT10 at the time. I was considering running a few hardware FX and was hoping maybe it was magically resolved. I believe I also tried Cubase and Studio One with the MR816, both DAW's ping the loop of the hardware and adjust to the calculation. If memory serves, when revisiting(rebooting) the session, the hardware would be out of phase. I wonder if OSX will ever fix this. I think Logic has a external hardware FX function including ping, one would think it would be resolved because of this. Again, i think this is a firewire problem. I like the Apollo really don't wanna change interfaces. Although regarding fixing this, Apples response would probably be that firewire is old technology, which makes me wonder if OSX has a solid latency with thunderbolt. I remember reading that Pro Tools HDX and HD native cards function correctly, I wanna say that I read other PCIex based systems functioned correctly under OSX.
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Post by lcr on Mar 8, 2014 12:27:44 GMT -6
Ive read where OSX firewire connection fluctuates latency. This would be a problem because after pinging or calculating the latency of the external hardware(depending on the DAW of choice) this time shifts slightly creating issues.
Anybody have success using hardware fx with the UA Apollo connected firewire 800 OSX with Pro Tools , or any other daw?
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Post by lcr on Mar 6, 2014 20:24:55 GMT -6
Concerning console angle. It has been said that part of the magic of the revered NS-10's was that they were usually perched atop an SSL 400o or similar. Thus the angle of the board actually helped them come into their sound so to speak. Couldn't say myself never using them with an SSL but say MY monitors sound better to me atop my desk than they do elsewhere. Food for thought. Interesting. Well, i am mixing the ITB mix in front of the console at Its designed angle.
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Post by lcr on Mar 6, 2014 11:20:31 GMT -6
For me, I would love a console and be working a hybrid set up, but it's not really feasible right now. I did the smaller desk as a experiment and found that it was an improvement in my situation. The worst part for me at the moment about using a console with outboard is easily getting to the outboards controls while still in the stereo field, I cant.. Its all off to the side and quite frustrating. I feel I am really not experiencing the full advantages of OTB because of this. I think having the rack gear top loaded on the sides of the desk would be ideal. Except for the reason this thread exists.. Reflection(s)
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Post by lcr on Mar 6, 2014 9:31:11 GMT -6
I was doing 2 "mastering" projects back to back and some where in the middle of it I changed my desk out. I went from around a 2' x 4' desk to a 1.5' x 2' desk and it was a significant improvement in monitoring clarity. Mostly in the hi/mid to hi's. My desk is now just big enough for my monitor and keyboard/mouse. My rack is under the desk, not a lot of leg room while working but the sonic improvement is worth it. Not to turn this into a ITB vs OTB discussion, but I agree with you slimming the desk down for less reflection is one great advantage of ITB. I am starting a mix ITB to see how it holds up to OTB thru the console. I am not going to mix the same track both ways and compare, I don't feel that is really a fair comparison. If I am happy with the results, I will complete a few more mixes ITB and then make some workflow considerations, which will drastically effect the size of my DIY desk(console vs no console). I know replacing the console with a summing box is also an option, I just wanna give ITB a try.
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Post by lcr on Mar 6, 2014 5:31:33 GMT -6
What about keeping the angle of the mixer and having 2 top loaded racks level on each side? Would the level racks still create nasty reflection?
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Post by lcr on Mar 5, 2014 19:52:51 GMT -6
Yeah, I don't think consoles had angles for audio reflection correctness in the beginning. I think they did so that you could reach knobs and things without your sleeves catching on everything like they would on a flat console. Honestly I think the reflection thing is overrated. There is so much other stuff that needs to be corrected in the typical home studio before worrying about reflection points on a console. Chances are you are sitting too close to your monitors to begin with if you have issues with reflections on the desk. My speaker triangle is approx 5 ft at the points. Ive treated the room with 101 treatment(early reflection, cloud, bass traps, etc.) i might get lazy and use the one rack i've already built and make everything level. My speakers are approx 1 ft from the wall, corners are treated however. Thanks for everyones opinions/help.
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Post by lcr on Mar 5, 2014 19:39:46 GMT -6
Moved it over here lcr. You'll get more help here anyway I think. Mine is on an angle for my console and I much prefer it that way. Easier to see the board for 40 year old dudes like myself. It's probably somewhere around a 10 to 15 degree angle. Just right. I can relate to being in your 40's and difficulty to see. Im upgrading my monitor to a wall mounted flat screen.
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Post by lcr on Mar 4, 2014 18:18:56 GMT -6
Speakers will be on DIY'd sand filled stands on the floor directly behind the desk.
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Post by lcr on Mar 4, 2014 18:15:39 GMT -6
Just realized this should probably be in "DIY"...
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Post by lcr on Mar 4, 2014 18:13:40 GMT -6
I'm not sure exactly what you are trying to describe. however... the idea of the angle is to limit reflections which disrupt the audio quality by reflecting the sound away from your ear. a lot comes down to where your speakers are located but I can say there is nothing more annoying then testing your mix position and finding that when you lay acoustic treatment on your mixer things sound clearer. Sorry If I didn't explain my question well. The mixers surface is at an angle when setting on a level desk. If I built my new desk so that the top surface of the mixer is flat instead of at an angle, is this bad even for a smaller mixer? I want to build the desk so the top of the mixer and both racks on the sides are at the same angle.
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Post by lcr on Mar 4, 2014 18:01:50 GMT -6
Is it bad to have your console and top mounted racks level instead of at an angle? I see desks built where you keep the angle of the console design and angle the racks on both sides(top mounted) at an angle vs having the console and the top loading racks level. I already built a level top loading rack, and was hoping I could get away with building another just like it, and build the console into the center. Its not a large console, approx. 28 inches wide x 23 inches deep. It will be 18 rack spaces per side. If I can get away with making everything level I could use 1 rack that Ive already built, but If its gonna mess up reflections, etc. I will start from scratch. Any opinions/help would be greatly appreciated.
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Post by lcr on Jan 28, 2014 17:40:31 GMT -6
check this out, if you got 2 hours: it's a video of a google hangout hosted by daniel ford, with a bunch of dudes from pensado's students facebook group. i got to sit in on this, it was cool. Thanks for this. I managed to get all the way through it (I had to pause/resume several times). I like his concepts on LCR. What I got from it, was like LCR or not, here is a way to look at panning, now go abuse it in a cool way(no rules!) which Is a great attitude toward things such as LCR. I still can't grasp how M/S hp/lp works. If the center is a sum of the sides, how do you hp/lp only the sides or only the center and not affect the other?
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LCR mixing
Jan 26, 2014 12:03:11 GMT -6
via mobile
Post by lcr on Jan 26, 2014 12:03:11 GMT -6
I went "full LCR" about a year ago. With the exception of sometimes toms and strings. I think its improved my mixes, however everytime we mix we SHOULD be getting better at it, I think LCR has def. helped. Speaking of improving your mixes/techniques, do you LCRers have a generic setting/starting point for HP/LP of middle and side material? Do you HP/LP MS differently? So, isnt the center channel of stereo just a sum of the L & R? If so, how do these plugins hp/lp the center without effecting the sides and vice versa? I know this is probably a very novice question explained 10.0000 times before, could anybody explain or share a link for how this works?
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