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Yamaha Mg16XU mixer, first impression


Theo Verelst

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Will it work for the purposes I set for it: mixing keyboards/synths/workstations with control over internal and external effects, adding quality microphone with (partially parametric) eq and build in compression, in a neutral way, as well as recording and playing back high quality and usable 192kHz/24bit Usb audio, preferably over Linux ?

 

After unpacking and trying some channels, some effects and the USB audio: it's very good quality, and the USB audio interface is perfectly recognized on Linux, which is a great relief. And it's not too heavy and pretty, and sliders work nice.

 

Next I'll try the keyboards, try effects and eq and compressors more accurately, compare with my high quality 384kHz DAC, vary the type of channel strip and run groups through channels with compression and eqs and then try some multi compression and studio effect tricks I have in mind on it.

 

So far, so good.

 

T.V.

 

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Testing a good mixer is fun, especially if it's cheap, and even cheaper because of an extra discount.

 

http://www.theover.org/Keybdmg/dsc00676bmc.jpg

 

The 8 compressors are usable, the mid sweeps are thus far clean, 4 aux sends and 2 stereo sub groups too. Feeding through channel strips twice is no problem, noise and hum figures are good (digital 1/3 octave analysis of build in 192kHz AD input shows S/N mostly over 100dB). Build in effect are strong and usable.

 

Balanced CP4 in is no problem, more testing on the way. An unbalanced connected Lexicon experiment went well, it was late so I should try at more normal volume still.

 

Cubase AI and drivers installed on the 4.5 GHz 6 core machine fine, runs ok, nothing much interesting new since the last decade, but it claims it records and plays back sample accurate, which is cool, and it sounds like that's true.

 

T.

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An experiment I did last night was to make a stereo gate circuit with sort of side chain eqs by using two pairs of channel strips in parallel, a straight strip, and an exactly mirrored one right of it (CH 1+2 for left, CH 5+6 for right), with as only difference the compressor is dialed in, and the left of the these channel strip pairs is in counter phase with respect to the right one, courtesy of the master group output (balanced).

 

http://www.theover.org/Keybdmg/20150405_165517bmc.jpg

 

Channel 7+8 are Lexicon effect return, 11/12 gets stereo input from my high quality (DIY) 384kHz DAC, which get's the averaged mid frequency expanded signal from a digital signal graph running on a connected Linux system, the USB DAC from the MG16XU gets pretty much the normal music at it's USB input, at 192/24. Group 1 is the output to the amps, group 2 is Lexicon send, and then there's a slightly circular signal path through the build in effects and the Lexicon.

 

Result ? I can create a sort of "phone" quality signal from good (A grade) mixes through the improvised analog "gate" which adds to signal clarity and even to get less sampling (reconstruction) errors in the sensitive mid-rage. Bit playing around, but it works and sounds interesting.

 

T.V.

 

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  • 2 weeks later...

Not really "first" impression anymore, I've tried to get sample reconstruction errors beat somewhat by running my mid low and mid frequency averaging software (on Linux) with the mixer connected over USB (runs good, at 192/24), and running the USB output channel back to other channel strips, for using the parametric eqs and compressors, and came up with a normal "mix" related setup that starts to sound like records used to when there was no digital. Not perfect, but the first time I got a Mike Stern CD to sound like a good show, and some High Definition tracks (from James Taylor and some reggae to the latest AC/DC) good enough to take notice of.

 

Recording processing results through the mixer worked good too. There's -70dB roughly of meter noise through the USB AD converter, but in the sensitive sense there's not much per frequency band, and it sounds good. I took a ffmpeg recording from the 182/24 AD converted mix result down to 96/24 files, accessed them from a notebook (running Linux) to upsample that up to 384 kHz to the DIY USB converter I've shown here, and it sounded great on my medium sized bi amped monitors, even at high volume.

 

Making a few keyboard setups with mixing and adaptations for some of the analog mix effects I have in mind is next. Recording an acoustic guitar proved ok, but those digital synts can pose a real problem to get as good as I sometimes can get get them, which I like make into a generally usable setup.

 

T.

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So I did some heavy post processing on some of my latest (and pretty great, no negative comments!) HDTracks studio quality digital masters (most 96kHz/24bit stereo, some 192kHz), making the very fast 4.5 GHz 6 core machine run up to 60 degrees Celsius by doing 10 band multiband bass decompression, 9 mid-low FFT averaged subband smoothening, mid averaging, high peak smear-outs (using seperate FFT processing), and a couple more things like some equalizing. I made the mixer play the resulting sound, with some channel strips doing mild compression (and analog smoothing of all kinds of digital tainted transients, and used the AD convertor (running perfectly at 192/24 on the machine running Linux) to feed certain averaging signal path inputs, so that they were as it were a bit part of an DA->analog_processing->AD loop, changing the mid and mid-low averaged master parts. That worked, and using 3 to 6 channel strip passes and a couple of mix buses for analog processing (a couple of frequency bands with various (analog) compressor settings, and a few pre-post asymmetrical eq settings, as well as stereo Lexicon effect in/out and internal (Yamaha) effect in/out, the resulting recording from the main bus it's AD converter are really good. I used the result on my big monitoring system as small PA into the garden+little field, and it sound great.

 

Also, I installed the Cubase AI 7 that came with the mixer on the Linux partition of the big PC, and tried using a CP4 connected over USB as Midi controller for a "Dexed" I put in the VST directory, and that worked too, tight enough fine sound. Funny though that a whole well working audio interface of a machine of very little over 400 euros can work fine on Linux, while a major instrument like the CP4 cannot even in some way connect it's UAB midi to Linux, while over a decade ago, that was already pretty much the norm, also with Yamaha instruments.

 

T.

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  • 2 weeks later...

Nothing overly complicated, the multicable run to and from the Lexicon, cheap patch cords are used to connect two synths, I connected a mic via a Lexicon pre-amp and my own high Q buffer into a line level channel, I record the 192kHz/24bit Usb signal from the Yamaha mixer, and monitor with a 384 kHz upsampled DAC (electrically isolated from the amp system).

 

Some new sounds I'd worked on a while ago:

 

[video:youtube]MLd-S_Ii36I

 

Sorry about the audio getting out of sync, I forgot to turn the computer TV off, or put the processor power higher, it had some audio drops. If this were commercial, I'd need to do it again, but for an impression the setup can work, it's ok.

 

T.

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After some have looked at this, a little nerd friendly info. The sounds I used were from the respective instruments, the Kurzweil PC3 on top, and the Yamaha CP4 placed at the bottom, both played live, on their own keyboards, no MIDI, not additional sound processing: both have their stereo outputs in balanced fashion connected to the MG16XU inputs.

 

The sounds on the CP4 are layered sounds, with a lot of reprogramming having been done in the effects and EQ portion, so they're not close to the ROM sounds, not easy to compare. The PC 3 synthesizer sound is a setup with some layers which have been tuned to provide the sound properties I've been looking for (like the mixing sub-bands I've posted about in the past), and aren't at all set up like the ROM sounds, but appeal to the considerable processing power present in the VAST technology, to emphasize some amount of sampling error corrections that are in the machine. The use of the filter parameters is still a bit artificial, but comes close to the liveliness and sonic appeal of some analogs, which is pretty hard (and I don't know any software that can do that a bit ok at the level I prefer).

 

The sound was recorded via the 192/24 AD converter build in the mixer, which like I reported get USB recognized by my Linux machine, where it was reduced (via "ffmpeg") from 192/24 to 96 kHz 24bit, before I encoded it into the youtube movie.

 

The AT2020 mic was pre-amped (and phantom powered) by the Lexicon Omega, of which the insert-out was buffered by the high quality pre-amp/signal buffer (with high Z input) I've made a little while ago, and fed into a channel strip on the Yamaha mixer with parametric equalizer, which was set to slightly compensate for the mic's frequency response.

 

T.

 

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Thanks for providing all of this information, Theo; it's been helpful. These newer Yamaha MG mixers seem to offer good quality and sonic clarity for a great price.

 

I just picked up an MG10XU for live use, and used it on a gig this past Saturday. It's attached to the side panel of my DXR12 cabinet, using Velcro. With the DXR kicked-back slightly on an amp stand, it's a best-of-both-worlds keyboard mixing/monitoring rig. I especially like the option of having one, stereo channel available for USB audio input/monitoring; very thoughtful addition, Yamaha !

And when I need to provide a simple PA for an acoustic type of gig (w/vocals and guitar added), it'll be perfect for that too.

'Someday, we'll look back on these days and laugh; likely a maniacal laugh from our padded cells, but a laugh nonetheless' - Mr. Boffo.

 

We need a barfing cat emoticon!

 

 

 

 

 

 

 

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