Cacophony Forums
Unofficial Mackie User Forums => DL1608/DL806/DL32R/ProDX Mixers => Topic started by: RoadRanger on January 14, 2014, 02:33:56 AM
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So, who's been doing this and what luck have you been having with getting a decent demo recording? I was planning on never doing a board recording but one of my regular bands can't afford me bringing in my multitrack capable board and really needs audio that they can mate up with video they are having someone shoot at an upcoming gig.
I plan on doing a "set and forget" PA mix during the first few songs and using my high isolation headphones to mix the recording on-the-fly. The DL1608 will be sidestage but I do have some 25 foot headphone extensions I can use so can set up out front - or maybe in another room so I can concentrate on the recording? The problem with high isolation headphones is that they don't isolate bass frequencies very well I suppose I can EQ the bass later if needed?
I'm really hoping some of you have become proficient at this and can give me some of your hard-won wisdom :) .
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I'd spend my time on getting a good FOH sound and recording it in stereo (2 mics out front and to either side not in the FOH Mix) than to try to mix on the fly for a demo CD. I used to drag a Yamaha AW1600 around to collect stems and post mix but looking back it really wasn't worth it. Of course if they need a lot of help then you may not have a choice if you know what I mean. Timing tracks of some sort are quite useful. A lot will depend on the room and the equipment your willing to bring. Do you have Auria? Do you just want to use the DL and iPad? What's the setup (# of mics etc)? Will aux 6 be enough for balance or do you want to separate tops from bases with some control?
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I have done this and it worked pretty darn good. As you know recording the mains only has a big set of problems in getting the sound right. Stage volume makes the live mix almost nothing like what we would want for a recording. I was the first person to come up with the aux as mains idea way back when the old forums existed. Beno even thanked me for that.
Of course to get this to work great you need everything mic'd up, or plugged in direct, which I am sure you are aware of. Another good idea to try is a pair of wireless in ears, or headphones, and get away from the band. I did a mix for a band a while ago at the four seasons in south Florida. I had a room to sit in while they were playing to get away from the private party. I still had full working control of the band. If I had plugged in my wireless IEMs then I could have ait and mixed the recording without noise issues.
I did sound at a halooween party for a band. They wanted me to record and luckily I had them all mic'd. The room was big enough for this. Because I had no idea at first they wanted a recording I didn't use the aux setup. It was brought up a few minutes before they had to play. The recording sounded good. I knew it would. I did say if I knew this was going to happen I would have set the recording up separately. The drummer asked me what I needed to do. So during a break I rewired, and reset so everything was mains from auxs, and a separate recording mix. I also added an overhead to the drums. I was not able to monitor the signal at this point but using the level meters I was able to create a seperate mix. When we listened back to the recodings it was a lot better the second time using a seperate mix. Imagine if I actually could hear it how much better I could have made it.
So as far as my experience goes it works very well.
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Let me go into a little more detail for some of you that may be new to this. Given that RR wants to walk away with recordings of 2 channels worth and TIME= $ in this case otherwise he would have brought more equipment that he has. I take that to mean that the mix-down will be done live. To make this manageable and to have something other than a dual mono from the existing mics possibly artificially contrived (panned) stereo let the two mics I mentioned give you the real thing a true stereo image without any effort on your part other than a good mic placement. These are the only inputs on L&R and that's what you will hear in your earphones and record. You can adjust the FOH mix in aux 6 to suite the recordings and your audience will most likely appreciate it as well. You could also use the new stereo link on aux 5 and 6 and balance the speakers beyond the DL at setup time. Room acoustics and other factors will have tremendous influence on the overall sound. Yes you also don't need to have everything individually mic'd this way. RR knows this I'm sure but JL keep reading it's in chapter 2 of recording 101. :) For post if you even want to spend the time you can deal with stereo width, compression, dynamics etc. things that can't be done in real time that are typically done in mastering a CD or DVD. A timing track will help the Video mix guy to sync audio with his video. You won't get SMTE out of a DL but it can be added post to the original in an editor. I've done it both ways not with a DL (early on) and settled on this as a good compromise solution given the time constraints and equipment limitations. I think you can swing two SD condensers mics and mic stands on this. Eyes sometimes help from a FOH position in dealing with the mix as well. This all requires 2 iPad's that I know you have. If that's too far away then you'll have to use you Galaxy IEM for the earphones. I know more gear to shlepp. Dave Rat has a unique method for mixing that I like even better but unfortunately it requires feature not on this mixer.
http://www.youtube.com/watch?v=MMMmR1u0CFk
Even small venues can benefit from this.
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Well, tonight was the night and it worked pretty good IMO :) .
A snippet: http://www.aspinock.com/foo.aac
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A snippet:
http://www.aspinock.com/foo.aac
Is this with post processing and you final take? Low AAC bit rate and artificial stereo. Got it.
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A snippet:
http://www.aspinock.com/foo.aac
Is this with post processing and you final take? Low AAC bit rate and artificial stereo. Got it.
No, that's a piece cut out of the original file as recorded 160 kbs AAC which is as high as Voice Record Pro will go - some say it is almost as good as a 320 kbs MP3, sounds OK to me. There's a "large plate" on there on the vocals, acoustic, and snare/highhat. There were only two mics on the drums.
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OK first the good news. Thank heavens I don't use iPad stuff for recording. My H2 will definitely outperform anything I've heard from iPad apps and can certainly provide higher quality raw tracks. I've used Voice Record for signals and playback but that's the extend. Those that believe 320MP3 and 160AAC are even close well it just goes to show you that this is an opinionated business but then there's math.
Now for the bad news. I don't particularly like a lot of reverb especially plate, my opinion. There are however the basics. The tracks recorded were at -33dB rms with a dynamic range of about 25dB, way to low. That may have been due to the app settings or the need to avoid feedback. This is where separate mics not in the mix will earn their keep. Max signal to noise without worry about feedback and a real stereo image. The usual goal for a good raw track is about 50dB above noise. You had about 30dB so a fair amount of detail is lost. Throw it into an editor and see what I mean. I also ran a FFT on the snippet and it appears that you really cut significantly starting at about 200 Hz, so much for lows. The first 35sec went OK but then something happened in the last 10sec which sounded like a torn driver. That's my take YMWV.
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As for the signal level, this was a "throw and go" situation (tables moved one hour before showtime) so I didn't have a chance to optimize that. I can't remember if the L&R fader affects the record level or not? I did have all the vocals faders at "0" and their levels bumping the yellow.
I was expecting to have issues with the low end, even high isolation headphone don't keep out the lows so I couldn't judge it that well - and I was literally sitting 6 inches from the sub cluster :o . This is just for some web demos so a bit of EQ and normalization should make it good enough. I sure wish I had the space and time to set up my Phonic Summit instead which can do multitrack :( . I'm sure looking forward to V3 with the multitrack recording upgrade to the DL1608 (;)) or at least a separate stereo bus to record from (or maybe matrices? 8)) .
Using aux 5 and 6 for the mains and subs was pretty weird - as they were both pre-fader I had to adjust anything in the subs on both auxes. I ran the 'verb pre-fader too. I would have liked to use a different 'verb on the snare and acoustic but such is life :-\ . I wonder if an upgrade to four "real" fx engines is possible to better compete with the competition?
In any case I'm not super happy with it but not super unhappy either - I was prepared for worse :) .
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RR can you find a more unfriendly environment to record in? Yes the main reason I don't record with the Ipad now that I remember is the lack of level control in the apps. Even a lowly Phillips cassette recorder had more control. The Faders and level controls are after the iPad data (see the infamous Pg. 162 in the MANUAL). The Levels as I remember are not available in most apps. The only pseudo control is the comp in the output LR path. Yuk!!! 15 dB at best. No indication of what goes to the iPad. When you hit record the iPad meter should reflect the level being recorded but I guess that's for V4 if at all. My take on the preliminary v2.1 is that all the output EQ's etc should be switched to the input channels where they belong. Give the output more PE's and be done with it. Band-aids anyone. Admit you like the HPF's. Let's keep the primary reason for the mixer in place and leave the weird secondary uses where they belong, elsewhere.
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it sounded pretty good. the panning was a little extreme. and of course you said only 2 mics were on the drums. but, all in all, not bad.
the best thing you can do is what i did. order the x32 rack. i already have auria and now will be able to record up to 24 racks and mix them down to my liking.
now all i have to do is wait for my dl to sell. i also have the mackie case as a bonus.
i may keep it becasue i now live in two different countires. i am thinking about leaving one in each so i dont have to travel with a mixer.
but if it sells.....nah nah nah nah, nah nah nah nah, hey hey hey, goodbye.
i'll probably still hang out here to keep you guys company. i dont want you to miss me.
curious if anyone is thinking about going the x18 route? i was debating on waiting for it, but in the last video i saw the behringer guy said it would be sept/october berfore it is ready. but to compare two boards the x18 blows away the dl in every way. it is direct competition, with a lot more features, and a cheaper price. i'm making sure to sell my dl now because when the x18 comes out the used price of the dl will go way down. its already dropping just based on the announcemnt at namm. for those of you who love your dl's, and refuse to make a latteral move that is actually a huge jump up, i feel sorry for you.
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it sounded pretty good. the panning was a little extreme.
On purpose, you can always reduce the width in post but it's tough to widen it.the best thing you can do is what i did. order the x32 rack.
MAP just dropped to $1299!now all i have to do is wait for my dl to sell.
They seem to be holding their price well used. I'm hoping to eventually pick up a spare for $500.
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Hmm... I wonder if there would have been any real advantage to recording at 24/48 instead of 16/44?
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Hmm... I wonder if there would have been any real advantage to recording at 24/48 instead of 16/44?
Not without fixing bigger problems first. I converted the 136 AAC (not that it mattered) into a wav file. The peak values where around 10% of what 16 bits can handle (64K). That's about a 13 bit recording. Philips cassette's are around 12 bits as a reference point. It would however have given a little more detail and would have helped when normalizing (yuk) and down-converting to 16 bits for CD quality. Oh and 44.1k would have been a better choice (no time conversion involved). This is assuming that your target was 16/44.1k CD.
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Given a choice I record at 24/44.1 (unless it's for video only when it'd be 24/48) the extra bit depth give a huge increase in headroom so you don't have to push the levels.
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Yah, as the DL1608 doesn't have a recording level adjustment and records WAY low it makes sense to record at 24 bits so you can boost it in post. And my target is web videos so either bitrate is probably OK - but yes, video tends to be at 48.
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Hmm... I wonder if there would have been any real advantage to recording at 24/48 instead of 16/44?
Not without fixing bigger problems first. I converted the 136 AAC (not that it mattered) into a wav file. The peak values where around 10% of what 16 bits can handle (64K). That's about a 13 bit recording. Philips cassette's are around 12 bits as a reference point. It would however have given a little more detail and would have helped when normalizing (yuk) and down-converting to 16 bits for CD quality. Oh and 44.1k would have been a better choice (no time conversion involved). This is assuming that your target was 16/44.1k CD.
I thought about this a bit and decided to run some simple test to confirm my suspicions. Turns out there is no difference between 16 and 24 bit recording using Voice record. Sounds strange but in reality it's quite simple. The DL converts an analog signal via the 24bit A/D converter and deals with it internally with at least 24 bits. What bit depth you store it in only matters if the number is large, in this case greater than 16 bits. This is not the case with the DL recording. My settings were 0 meter and fader setting on input (using a stable 1kHz sine wave) and you end up with -24 dBFS rms on the recording. That's under 10% of a 16 bit depth, in other words the number 6400 peak out of a 64K range. The 24 bit depth would contain the same number (as it does based on the recording) in a number range of 0-16.7 Million. No more detail and other than using more storage no real value. The recording settings for Voice Record were 160 kbps (AAC-LP) at 48K sample rate and a bit depth of both 16 and 24, stereo and max encode quality. I adjusted both the input fader and the output compressor gain and could get a +10dB (input fader) or +20dB (compressor) gain to get a reasonable level for recording. That's a -14dBFS rms on the fader and -4.5dBFS rms on the compressor adjustments. None of these caused clipping although the meters would indicate otherwise. Back on the old Mackie forum I mapped the meters and found them to be off as much as 10 dB ( fairly useless for accuracy, good for ball-parking).
Sam and many of you may want to take a look at the explanation on headroom in the digital world here. Keep in mind that any mixer can still be measured for dBu in and out. They're just all unique.
http://www.sengpielaudio.com/calculator-db-volt.htm
Conclusion: Due to Mackie's false indication of input levels the recorded output is too low. The best S/N ratio of course can be gained at the preamp settings not at the tail end compressor. Yes that means a "hot" input signal based on their meters even if it's not so. This will also give you max dynamic range. Remember that the meters are PPM meters not RMS or VU meters and show peak values. A while back there were a lot of complaints that recordings from CD's were to "hot" when played back on the iPad, they're right on the money but the meters not so much. Meters on the DL are problematic. If you have a good rms meter you can verify this.
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RR if your target was YouTube then the current recommended specs are 16:9 720p at 44.1K and 16 bits. Currently the audio ends up at 128kbps AAC. The upload converters are giving this kind of results at present but it has changed over time as they improve things. :)
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I thought about this a bit and decided to run some simple test to confirm my suspicions. Turns out there is no difference between 16 and 24 bit recording using Voice record. Sounds strange but in reality it's quite simple. The DL converts an analog signal via the 24bit A/D converter and deals with it internally with at least 24 bits. What bit depth you store it in only matters if the number is large, in this case greater than 16 bits. This is not the case with the DL recording. My settings were 0 meter and fader setting on input (using a stable 1kHz sine wave) and you end up with -24 dBFS rms on the recording. That's under 10% of a 16 bit depth, in other words the number 6400 peak out of a 64K range.
-24dBFS in volts is ~1/16 so I'd expect a 20 bit signal in that 24 bit recording? Yes, it would only be 12 bits in a 16 bit recording. Internally everything is floating point AFAIK. On the recording a 16 bit recording would be the 24 bit recording with the bottom 8 bits thrown away - not the other way around where the 14 bit recording would be the 16 bit one with 8 zero bits added on top as that would make the 14 bit recording -72dBFS vs the -24dBFS of the 16 bit recording :o .
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I thought about this a bit and decided to run some simple test to confirm my suspicions. Turns out there is no difference between 16 and 24 bit recording using Voice record. Sounds strange but in reality it's quite simple. The DL converts an analog signal via the 24bit A/D converter and deals with it internally with at least 24 bits. What bit depth you store it in only matters if the number is large, in this case greater than 16 bits. This is not the case with the DL recording. My settings were 0 meter and fader setting on input (using a stable 1kHz sine wave) and you end up with -24 dBFS rms on the recording. That's under 10% of a 16 bit depth, in other words the number 6400 peak out of a 64K range.
-24dBFS in volts is ~1/16 so I'd expect a 20 bit signal in that 24 bit recording? Yes, it would only be 12 bits in a 16 bit recording. Internally everything is floating point AFAIK. On the recording a 16 bit recording would be the 24 bit recording with the bottom 8 bits thrown away - not the other way around where the 14 bit recording would be the 16 bit one with 8 zero bits added on top as that would make the 14 bit recording -72dBFS vs the -24dBFS of the 16 bit recording :o .
OK the math is like this (the reason I gave the Sengpielaudio link above) and we've all fallen for the analog thinking about digital at one time. The DL has a 0dBFS at 21dBu as per spec. The -24dBFS works out in theory to -3 dBu (21-24=-3). This amounts to 0.54V rms or 1.5V PP. The above 6400 example (10% of 64K) had nothing to do with actual values just a explanation (sorry for the confusion). The point I was trying to make is that if you input a voltage that converts to 6400 (we don't really know what Mackie is using internally fixed or floating point both are available) and you loose nothing in internal processing in precision one would expect that same number out, converted to the same voltage. If however you exceed that max number internally you would clip. Now for reality. Mackie looses about 3dB from input to output (all fader at 0 and no processing input signal at 0) if you use the visual scale and on top of all that it's nonlinear (as much as +/-10dB) if you plot it against dBu or dBFS. The point I made on the old Mackie forum. Their centerpiece on the MF and they can't get it right. Your correct if you down convert a full 24bit value to 16 bits you loose precision the least significant 8 bits. I converted the 24 bit recording to 16 bit (in Audacity) and no precision was lost due to the low values and a smart conversion.
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I'm 99% sure they use floating point so no loss of precision internally.
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I'm 99% sure they use floating point so no loss of precision internally.
I'm not that sure and both have the same precision so no loss either way and fixed still beats float for speed.
Correct that the fixed point is 32 bits (higher precision than 32bit floating point equal to extended 40 bit Float) but we can't create something out of nothing. 24 bits in is the max (A/D) internal math is not going to change that.
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It's actually 24 bits per input, so 2 inputs gets you 25 bits, 4 inputs gets you 26 as you sum - but I dunno where the noise floor is? In any case it's a LOT easier to do all the EQ and such in floating point so you don't have to worry about over or under flow and you retain precision no matter how far down you are. You can attenuate something by 200 dB and somewhere else amplify it by 200 dB and still have your 24 (or whatever) significant bits. No challenge at all LOL.
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It's actually 24 bits per input, so 2 inputs gets you 25 bits, 4 inputs gets you 26 as you sum - but I dunno where the noise floor is? In any case it's a LOT easier to do all the EQ and such in floating point so you don't have to worry about over or under flow and you retain precision no matter how far down you are. You can attenuate something by 200 dB and somewhere else amplify it by 200 dB and still have your 24 (or whatever) significant bits. No challenge at all LOL.
I understand but if you look at the Sharc data flow structure It has a 64 bit data transfer bus and 80 bit multiplication registers. They made sure that most things won't run into a limit. So 8 bits or 2 to the power of 8 = 256 channels. You think 16 is a problem? If they were clever they would use both for the right job.
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For those of you reading this don't get the impression that 24 bit recording is no better than 16 bit recording because it is. The key in this case was the under-utilization of the hardware. The manufacturer sets the range of the A/D converter for the 24 bits. If the range is set too low then you are converting a lot of your noise floor. This does nothing for your dynamic range which is the max level the A/D or preamp can handle minus the noise floor. The trick here is to utilize as much of your equipments range as possible without hitting it's limits (High and low). The goal is to get 50-60 dB above noise floor for great dynamics but that's not as easy as it sounds since some music only has a 20 dB dynamic range. The advantage to 24 bit recording is that it is more forgiving in trying to reach that compromise in settings. If however your dynamic range is only 30 dB there is nothing either 16 or 24 bit recording can do to get it back or improve it, it's gone. Hope this helps if the stuff above is too techie.
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So if I'm reading this right, we should be setting the trims as high as possible without clipping to get the most out of the dynamic range that exists. (I've been getting crap for using db numbers on another forum. I'm trying to break that habit, but sometimes it just seems the only way to actually talk about things with any accuracy.) I've noticed the tolerance of the DL's pre-amps. There sure seems to be a large range there. Anyway with most recordings rarely ending up with much in the way of dynamic range, it might not be as important in the end. (One very well done exception is "The Letters" by King Crimson from the Islands album. That one probably has a 30 db range from the beginning at a whisper to the solos in the middle that are screaming. Don't listen in a car with the engine running. Best with headphones set barely audible at the beginning. And they did that with analog equipment. Amazing.) It's hopeful that Mackie could massage more recording possibilities out of the DL. I'm hoping they don't give up on software development, but unless they charge for significant upgrades, there isn't much incentive for development of this mixer. (And we have all gotten used to free upgrades.) I would be willing to pay for an upgrade that included multichannel recording. (I'm happy with the improvements that have occurred already.) If most of the products shown at NAMM come onto the market, Mackie's not going to have much incentive to continue working on the DL and instead start their next mixers. On the other hand, what they've learned about programming this mixer should be easy to port to new products. (How long have most of these companies had software development departments?) Thanks for the discussion, even if parts of it were solidly above my head. Very interesting.
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Wynnd this was meant for your recordings if your looking for quality work. There is nothing that can be done for other peoples work it is what it is. This of course leads into the "Loudness war" discussion something for another time and place.
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The loud band is gone from my life. Currently only working on a low volume project. Might actually keep my hearing for some years to come. (Hope so. My Mother who never was exposed to loud music had two hearing aids at the end of her life. She couldn't hear the phone ring without them.)
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Going to try this over the weekend, my own band on Friday and a very loud tribute act on Saturday.
Saturday will be easier (recording wise) as only the vocals will be actually coming out of the rig, if past experience is anything to go by, but everything will be mic'd up.