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.

Tracking

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.

 

Conclusions

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.

 

Cheers,

Lorenzo

 


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 @ http://iuatwest.com and your readers shall not be charged by you under any circumstance.

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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

Hi,

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

Summing

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

Mixing

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).

Cheers,

Lorenzo


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 @ http://iuatwest.com and your readers shall not be charged by you under any circumstance.

DAD AX32: test in progress in my studio – part Two

Ok,

after first approach based on comparison listen (see DAD AX32: test in progress in my studio – part One), I verified frequency and phase response with Metric Halo SpectraFooCompleteX.

All settings are controlled by DADman software (see figures bottom) like matrix or Mic/line switch. When you change Mic gain or Line out level you can hear the relay click on.

All audio connections use DB25 with Tascam pin-out.

In front AX32 has four buttons to change sample frequency and clock master, a little screen to check settings and a double series of sixteen leds to visualize carrier (digital signal) and audio presence.

Here DAD AX32, it look very nice 🙂


After I installed Dante controller and  Dante virtual sound card both by Audinate and I created an audio network where my computer (with audio application) and DAD AX32 are two clients.

Dante virtual soundcard has to work necessarly without a Dante PCI card compatible. It uses ethernet in your computer gateway. With Dante controller you can manage a flush of 64×64 channels audio at 48kHz@24bit or 32×32 channels audio at 96kHz@24bit with 1Gbit of bandwidth. You find all specific here.

I created a multicast and I managed audio interchange to and from AX32 with virtual routing.

Well, I opened SpectraFoo and I changed soundcard with Dante (it appears on available audio cards) and I ran signal generator with pink noise to test AX32.

I used Transfer Function window to visualize the difference between original signal (reference) and it after AX32 AD/DA (response). In this window it’s possible to visualize Power vs Frequency and Power vs Phase.

I tested AX32 at 96kHz and 48kHz.

At 96kHz I checked a bandwidth very width (8Hz to 43kHz) with an excellent phase response.

Also at 48kHz I checked a bandwidth very width (8Hz to 22kHz) although with a slight drift phase response at high end.

Also I tested Ax32 with FuzzMeasure, an audio and acoustical measurement application, and it confirmed SpectraFoo audio bandwidth analisys.

Here all SpectraFoo graphics:

My conclusions on the DAD AX32 is that it’s the best converter I’ve ever heard. It’s very valuable into mix and fundamental into mastering because very wide bandwidth with clean mid frequency and extended low and high frequency are a necessarly conditions to work fine.

Have a nice day,

Lorenzo


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 @ http://iuatwest.com and your readers shall not be charged by you under any circumstance.

Technical backstage: which audio monitor I use

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.


Hi,

this is a very hard question, which monitor choose and why?

When I go tracking I try different microphones to find the best matching with audio source. This is a obvious practice because each microphone has typical electronic and acoustic features.

Good, but you can take the same approach for monitoring? No, because their arrangement is a very delicate matter, a couple of professional monitor is expensive (but if possible it’s better to have two pairs) and unlike microphone above our reference listening should not be linked with any musical instrument or musical genre.

Essentially we need a pair (and matched) of monitors with frequency and phase flat response, correctly matched with control room. Wow…

All brands claim to fulfill these features. Ouch…

Ok, I tried several brand and model and I choosed Emes, a little german company.

In particulary I bought a pair of Violet HR (nearfield) and a pair of Blue HR coupled with sub Amber HR (main).

Violet Hr is a tipical two-way loudspeaker with the same amplifier (100W RMS) on drivers with a small woofer (18 cm – 7 inch.) and a silk dome tweeter.

Blue HR is designed as a D’Appolito configuration with the same kind of drivers of Violet HR. Three amplifier (two woofer and one tweeter) handling 100W RMS.

I like them because the portability to real world is almost perfect. Low end is good balanced and correct, high end very extended and the mids are really clear with speech frequency focused. Thanks to woofer small size they had a very fast response to transient and directivity.

All loudspeakers are sold with a frequency response matched with less than ±0,5 dB difference.

Violet HR are like a magnifying lens and allow you to listen with great precision individual parts.

I arranged the Blue HR in a large box suspended like a flush mount to eliminate audio emission back, minimize frame diffraction and relating them to the control room acoustics. The stereo image is very wide, clear and deep.

I used Amber HR into main system as a third way and furthermore to extend low end, also I can bypass it and listen to the Blue HR full-range.

During main system set-up I had checked phase alignment with audio measurement software SpectraFoo.

In addition I arranged a pair of classic Yamaha NS-10 studio and a pair of Avant electronics Avantone Mixcube (they are like Auratone) to achieve a listening poor.

Here my left monitors array

monitors arrangement

I sold to my customer several Emes loudspeaker system consists of a pair of Violet HR or a big system like Blue HR coupled to two sub Amber HR as a main monitor or a Hi-End system.

Well, now I wait for you in my studio to listen them  😉

Cheers,

Lorenzo


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 @ http://iuatwest.com and your readers shall not be charged by you under any circumstance.