DAD AX32 on the road


in early june I recorded choir Aspis’s last session with DAD AX32.

Thanks to Audinate Dante Virtual Soundcard I used it like a soundcard with my MacBook.

Session was in S. Pietro auditorium in Milan which it’s charaterized by a very dry sound.

I used two AKG C414 TL-II to ORTF stereo array and four Schoeps CMC6 MK4 as spots on sections.

Choir Aspis's session

Choir Aspis’s session

As I checked in my tests (part 1part 2) DAD AX32 has an amazing sound!

I used DAD AX32’s premicrophones which have a very pristine sound with a wide dynamic range. They sound very close to my Millenia Media HV-3R.

Record will be released in September.




Schoeps V4 U

Recently I had the pleasure to try Schoeps V4 U a new studio vocal microphone.

It’s a very interesting gear, lightly and small with several original technical solutions.

Shoeps V4 U

Shoeps V4 U in his wooden case

Available in grey and blue color Schoeps V4 U is a FET microphone trasformerless and free of coupling capacitors with a newly designed electronics which offer a very high maximum sound pressure level (144 dBspl).

Characteristic of this microphone is the beveled collar on the 33mm capsule which cause directivity to increase steadly and smootly at high frequencies, as in a large-diaphragm microphone.

Schoeps V4 U capsule head

Schoeps V4 U capsule head (source Schoeps website)

The classic look of the V4 U is based on the CM 51/3. This microphone was manufactured from Schoeps from 1951-1953.

The V4 U is available in two sets: The “V4 SGV set” contains the microphone, a wooden case and the SGV stand clamp, the “V4 USM set” differs from the other one for the elastic suspension USM-V4 (Rycote made).

The test

I tested it with a female singer tracking, my speech and a test signal coupled with SpectraFoo Complete spectral analyzer and I used my Neumann U87ai as reference to check V4 U nuances.

I made a cluster of three microphones to compare the different sounds on the same source, all microphones were plu into a Millenia HV-3C and a DAD AX32 AD/DA converter chained with my DAW Logic Pro by Dante ethernet protocol.

Three microphones cluster

Three microphones cluster

The test was carried out in three steps comparative:

1)  Spectral responses and output level (I fixed a reference gain) of all microphones with SpectraFoo complete.

2) Comparison between Neumann U87ai (worldwide vocal reference) and Schoeps V4 U to check on a female singer and male speaker differences.

3) Comparison between Schoeps Colette MK4 and Schoeps V4 U to check differences for at first glance similar capsules.


Spectral response

The spectral responses of microphones are showed bottom, the first graph is about the output level where U87ai is the reference.

After I overlapped spectral response of Schoeps CMC6@MK4 on Schoeps V4 U and I checked a better performance on low end by V4 U and a identical response on mid and high frequencies.

Neumann U87ai and Schoeps V4 U comparison was very interesting, because on mid and high frequencies V4 U has a flat and extended response while on low end it’s intermediate between the other two microphones.


Ok, after technical stuff I explain vocal tracking of  a female singer and a male speaker (myself).

All three microphones are amazing and professional gears, simply my opinion is about what I found ready to use”.

Neumann U87ai highlighted a slightly muddy sound on low end and a nastily mid-high frequency (about 3kHz); Schoeps CMC6@MK4 was smoothed on mid-high frequencies without harsh but with a lacking low end.

Schoeps V4 U captured a focused, depth and natural sound with a warm low end and it kept the same smoothed and extended high end of CMC6@MK4. Transient response was very fast.



Schoeps V4 U is a goal, it’s a very interesting alternative to vocal studio standard microphones (like Neumann, AKG, Brauner and others). It natural sound with a warm low end and a flat extended response is a powerful tool to capture vocal and acoustic instruments.

The lack of alternatives polar responses, filters and pads are a limitation for the multipurpose applications but it’s a tipical Schoeps phylosophy to maximize microphones perfomance.





Disclaimer: I tested this gear with care, nevertheless this test is inevitably affected by my opinion and possible analyzer gear and software imprecisions.

Copyright © 2013-2014 by iuatwest. All rights Reserved.
This material has been copyrighted,  feel free to share it with others; it can be distributed via social media or pingbacks or added to websites; please do not change the original content and, provide appropriate credit by including the author’s name @ and your readers shall not be charged by you under any circumstance.

Technical backstage: my analog consolle

It’s not a didactic section (when needed I’ll link related pages on manufacturer site) but so as to explain my approach to recording.

D&R Dayner in my control room

D&R Dayner in my control room


before talking about my analog consolle I want to expand my point of view on some questions about the summing and the mixing.


The result of summing is a single signal (electric or digital) from several sources(*), the problem is how to make it without quality drop. Obviously that signal can be mono, stereo or multichannel in order to destination.

Theorically analog or digital summing are the same, but in real world, analog gear has a non-linear response with typical distortions on second harmonics and cross-talking. Digital summing can emulate it or capture the nuances of analog gears with convolution to add “warmth” and depth to mix.

(*) example of Op-amp summing amplifieranother example


Assumed that you used a good microphones and preamplifiers to capture several instruments, it’s probably necessarily change their timbres, levels and dynamics to make a pleasure sum. This is a basic approach to mix.

Today it’s possible work in several ways:

– Full Analog (rare)

Analog tape recorder with analog gears

– Out The Box (box is the pc) or OTB

Computer as digital recorder with analog gears like consolle and audio processor to mix

– In The Box or ITB

Computer become a DAW (Digital Audio Workstation) and it’s added with audio card with AD/DA converter and specific softwares.

– Hybrid

DAW and several analog outboards are chained by a multichannel DA/AD

Nothing of their is the best choice but each can be a better way to work into a specific situation.

I chose two solutions: ITB and OTB.

To work ITB I chose Apple Logic Pro (since 1996 – 2.5.4 version) with many plug-ins like Waves, Softube, Metric Halo, Brainworx, SPL and Abbey Roads (this last is discontinued – ouch).

Alternative DAWs available are ProTools 10 and Harrison Mixbus 2.x.

ITB mix is a cheapest way to add many times the same expensive (but virtual) compressor or reverb, to create automation for all parameters available, to edit takes, to create incredible audio effects, to add and manage virtual instruments and to add samples to substitute or sum it with original recording. At last to save and restore the project with one “click”.

But after many years I realized an innate problem in ITB mixing.

If it’s true that when I add plug-ins the relative delays (into DAW) are automatical compensated, the delay due in conseguence how to CPU works (multiplexing) affects time alignment of all channels into sum (unlike analog consolle where all signals are process in a parallel way). That phenomena is audible in complex mix with many tracks and many plug-ins and it’s highlighted with a shifting the mix to muddy sound with a depth deterioration.

Let’s be clear ITB mixing is good choice but adding plug-ins without limits can shift your sound in trouble although your high perfomance pc system. Just the same of any machinery when is overfilled.

It’s odd to note how many phase or time alignment plug-ins are maded in last years although digital recording has less problems about phase correlaction compared with vynil cutting.

To realize OTB mixing I bought a second-hand consolle, directly from eighties, D&R Dayner.

Ok, now I go to present it.

D&R Dayner

This is a tipical studio in-line consolle, with direct outs, tape returns, eight busses and eight auxiliaries.

It has 24 channels and eight effects return and I did broaded it frame to insert Euphonix MC Mix control surface, Apple keyboards and third screen of my DAW.

Dayner peculiarity is it floating busses (named subs), that is the possibility to assign any bus to any channel (along left/right assignement). Tape output and monitor section are substitute by bus signal. It’s useful to send to recorder premixed channels.

Dayner input section - above the floating subs assign

Dayner input section – above the “from floating subs” assign

Then it’s possible to work from 24 channels without busses to 16 channels and 8 busses.

They have three kinds of channels: In-Line, Split and Tape/Effects return.

In-line: it’s a basic channel, it manage mic and line input with eq, aux sends and volume. Also it manage tape return on specific input and monitor section. It’s possible invert input with tape return to mix the latter (remix switch). Bus assignement send it to tape out and monitor section without possibility to assign it to main.

Split: it’s the dedicate channel to manage floating bus on mixing with assignement to main. It haven’t tape return section. I haven’t split channel but on In-line channel I chained tape out with line-in to replicate their. In this way I can applied insert and complete eq section on bus signal and mix it.

Tape/Effects return: it has four balanced line input. They are perfect to manage extra eight channel to mix. Today I use it chained with two channel strip Focusrite ISA220 and to input stereo effects return from DAW.

I tested Dayner bandwidth with Spectrafoo and DAD AX32 at 96kHz (see test here) and the eq too.

Bandwidth and phase response measured at 96kHz

Bandwidth and phase response measured at 96kHz

I like the eq of Dayner, it works in mellow way without artifacts. It sounds great on drums and electric bass.

The eight auxiliaries are routed to DAW to applied reverb, delay or modulation effect.

I can mix from 32 channels without group to 24 channels + 8 groups (busses).

I’m very happy for my Dayner consolle, I suggest it to mix rock, blues and jazz (acoustic and electric).



Copyright © 2013-2014 by iuatwest. All rights Reserved.
This material has been copyrighted, feel free to share it with others; it can be distributed via social media or pingbacks or added to websites; please do not change the original content and, provide appropriate credit by including the author’s name @ and your readers shall not be charged by you under any circumstance.

Technical backstage: Harrison Mixbus – an interesting alternative DAW

It’s not a didactic section (when needed I’ll link related pages on manufacturer site) but so as to explain my approach to recording.

“Other DAW mixers are designed by companies with experience in computer sound, but no pedigree in world-class recording facilities.  The Mixbus mixer is designed by Harrison: the maker of consoles used in the world’s most demanding music, film, and live performance facilities.” (from site)

When I read this statement (early 2010) I was very impressed and I thought “it’s true, I want try this software”. It was very low price (approximately $75 as I remember) and I bought Mixbus immediatly.

The first time I listened my records by Mixbus I perceived a better sound of mid and high frequency. Less harsh, more body and a superior depth. Wow, it’s ITB Valhalla? (ITB = In The Box otherwise audio mixed on computer)

Harrison implemented his digital summing algorithm (the same of their digital consolle) into a very cheap but powerful DAW Ardour, an open source project by Paul Davis.

He worked on another application too, available for Linux and OS X, named Jack (Jack Audio Connection Kit). It’s a virtual router and it can manage soundcard inputs and outputs and all virtual connections into computer. Also it can re-direct audio flush from an application to another without leave it from computer.

Features of Harrison Mixbus mixer are three band EQ and dynamics on every channels and busses, eight busses (from the version 2.x) with tape saturation and tone control, auxs and inserts freely assignable to pre or post fader, all parameters automation, freely assignable routing and master equipped with K-14 Meter(*), tape saturation, control tone and dynamic processor.

Unfortunately at that time Ardour and Mixbus too (obviously) wouldn’t managed MIDI narrow down their application.

(*) K-14 meter is a method to monitor audio level into digital system, Bob Katz made it and here he explained about it: part 1part 2

The first time I used Mixbus 1.x to summing tracks from Logic Pro. It was a live recording of acoustic ensemble composed by drums, double bass, strings quartet, piano, key synth, acoustic guitar and voice. I created a several group (drums, strings and keys) and I routed them with single tracks (double bass, acoustic guitar and voice) to Mixbus by Jack OSX to summing them without add other processes.

After I compared the same mix from Logic bounce and Mixbus record and the last sounded better.

I noticed better mid vocal frequency, a snare superior body and less harshed hi-hat and cymbals. Otherwise a natural, better acoustic sound.

About editing I like Ardour/Mixbus for his tools, clear graphics design and easy automation.

One year ago I mixed my band RSVP album (you can listen it on Youtube) with Mixbus 2.x, the project’s tracks included two acoustics drums, electric guitar, electric bass, keyboards, saxophone, trumpet, voiceover, movie audio sample, a pico-paso (analog oscillator) and other tools (like a drill).

I worked like an analog consolle (with automation) with few plug-ins unlike other DAWs. It sounded GREAT!

At the same time I suggested (and I taught) Ardour to my friends for record, edit and mix their latest news podcasts ( or Mixbus with Jack OSX to my customer Andrea Fedeli to make virtual summing from his DAW (with no added costs buiyng expensive brand analog gear).

Mixbus and Jack are availables for Linux, Mac OSX (PPC and Intel) and Windows.

Today Mixbus 2 still don’t manage MIDI but next major update surely will do it because Ardour 3 has already been implemented.



Below: Mixbus mixer

Copyright © 2013-2014 by iuatwest. All rights Reserved.
This material has been copyrighted, feel free to share it with others; it can be distributed via social media or pingbacks or added to websites; please do not change the original content and, provide appropriate credit by including the author’s name @ and your readers shall not be charged by you under any circumstance.

Logic Pro strikes back


another day with Logic Pro and his trouble…

I have still got problems opening logic project, despite I used AIF files…

To get it solved i changed all audiofiles names and I located them after opening the project….

At this point I had not more problems (I couldn’t believe) and I bounced (Logic worked very slow ) the project into stereo file.

Tomorrow I’ll generate (hope so) a different version of the master:

– Hi-End version (96kHz@24bit)

– Video version (48kHz@16bit)

– Cd version (44.1kHz@16bit)

– MP3 version (320kbits stereo)

Just authorized by my customer I’ll link one or more songs of this work.

…and the mix be with you  😉