blumlein records

Errata - Exotic Positions

noreply@blogger.com (Andrew Levine) — Fri, 09 Apr 2010 07:20:00 +0000

I just read Paul Stamler's Exotic Positions, and here are some necessary corrections:

3 out of 4 ways of recording in stereo...

Let me start with (the old master) Alan Dower Blumlein. While it is certainly correct that "he did most of the theoretical analysis necessary for the development of stereo recording" he was also very much into exploring the practical aspects of recording. And while "he developed a stereo miking system [the Blumlein setup] that solves many of the problems inherent in XY and ORTF techniques" it is important to note that not only ORTF came 30 years later, but that Mr. Blumlein invented all three coincident setups, including XY and M/S. The latter is listed under the "true exotica" section although it is the de facto standard of stereo in moving picture sound.

"XY"

Back to the start of the post. XY creates a stereo image strictly by differences in sound pressure level. The term "volume differences" is not accurate. And while often arranged at an angle of +-45, resulting in a huge recording angle of 196°, any angle can be used. It is important to note that the size of the inter-capsule-angle is inversely proportional to the size of the recording angle. Point the capsules father apart for a smaller soundstage!

Visit Eberhard Sengpiel's stereophonic playground for some experimentation.

Rather than saying that "typically, XY recording produces a narrow soundstage" I'd explain that using an XY pair with an inter-capsule-angle of 90° results in a semicircular pickup range. I use this configuration when I have to mike very close to an ensemble and don't want to go for AB, perhaps because I want to reduce the influence of the room or attenuate audience noise.

"Getting fancy: ORTF"

I'd personally place the near-coincident pair stereo technique after XY, Blumlein and M/S. Or, if I were to sort the approaches by their practical relevance to me it would be at the top.

The very nice stereo image that ORTF (and other near-concident techniques: NOS, DIN, EBS, ...) can produce relies on a combination of differences in sound pressure level volume and time of arrival cues. The reason that "sources are rendered in correct spatial perspective rather than in the narrow soundstage endemic to XY recording" is due to the fact that the recording angle is 98°, a little more than half that of XY (at +-45°).

"The old master: Blumlein"

I consider Blumlein to be a special case of XY, using bidirectional capsules / ribbon transducers angled at +-45°. While it is true that "Figure-8s tend to maintain their pattern at all frequencies" the statement that "at least some of them have excellent bass response" is doubtful. Look at the frequency charts of all bidirectional capsules for a pronounced proximity effect / LF-attenuation. As an example please compare the charts on the cardioid Schoeps MK 4 with that of the fig-8 Schoeps MK 8. David Royer has argued that this does not hold for ribbons (see the Royerlabs SF-1) and I have done a few one-point Blumlein recordings that seem to justify the assessment, but I am as yet uncertain on this point.

I don't agree with the statement that "because the pickup pattern is bidirectional the microphones will pick up lots of room sound, leading to a very wet recording." AB gives me much more spaciousness, but since AB (with two parallel omnis spaced at a distance of 51.5 cm) results in a recording angle of 180° I place it much closer to the ensemble than a Blumlen setup with it's recording angle of 76°. Therefore it is definitely true that the sonics of the recording space matter a lot when using a Blumlein setup.

"True exotica"

Read all about Tony Faulkner's Phased Array here: Part-1, Part-2, Part-3.

The Jecklin disk uses "a pair of omnidirectional mics spaced a foot or so apart with a large plastic disc between them." That much I can follow. The statement that "this creates the equivalent of a pair of cardioids pointed outward at 180˚" sounds wrong as (in the case of coincident capsules) this would result in a recording angle of 102°.

Regarding M/S the post states that "what’s most useful about this is a high degree of mono compatibility". In fact M/S is completely mono compatible: Use only the central channel...

Interesting to me: the "Swedien technique"--although in theory coincident yet angled true omnidirectional microphones can not generate differences in sound pressure level. Paul Stamler does say the setup pertains to the use of large diaphragm condensers though. I'll give it a try when I get the chance.

"Step up to the bar"

The sturdy yet inexpensive K&M 23550 stereo bar does a good job for XY & ORTF. Don't forget one or two K&M 218 thread adapters to be able position one mic above the other without undue vertical angling.

"What, when, and where"

Much of this is a matter of taste, but I'd usually not recommend using "a pair of good condensers overhead in ORTF formation (panned hard left and right) [...]" for recording drums. True, a stereo overhead and two mics, one for the snare (or a central position) and one for the kick is all you'll need, but if you don't use a coincident overhead you can not reduce the spread of the image in post without introducing comb filtering!

"Try a few true-stereo single pair recordings just to hear what the technique can do. And if you have the tracks, try using stereo miking techniques on multitracked projects. You won’t be sorry." I am happy to say that I wholeheartedly agree with that final statement. "Happy pairing" to you as well :-)

[Please check out Eberhard Sengpiel's website for detailed insights into various stereophonic configurations, recording angles etc.]

Do we need 24 bit audio for sonic nirvana?

noreply@blogger.com (Andrew Levine) — Thu, 08 Apr 2010 15:25:00 +0000

"The noise level in the avarage residence is about 43 decibels" [Harry Ferdinand Olson (1967): Music, physics and engineering], whereas a house in the country can be as quiet as 35 dB. Let's deduct 6 dB from that number as it is quite possible to discern musical content that level-wise lies within and seemingly should be masked by the noise floor.

16 bit audio has a theoretical dynamic range of 96 dB, which, if you add that onto the baseline 29 dB amounts to a peak of 128 dB, slightly below the threshold of pain. It seems to me that that span should suffice to adequately present the finest dynamics inherent in music, especially since the range I usually experience recording very high dynamic range avant-garde music lies at 54 dB, and many real-life concert venues have a noise floor at -60 dB FS. Of course less is highly preferable, but then it is also dependent on the quality of the ambient noise.

Note the use of the word "suffice". Clearly, having a theoretically usable resolution of 144 dB when working with (real) 24 bit audio is even better. It is tantamount when recording music while leaving adequate headroom--with no manual gain riding required and no need to use a compressor while tracking--, and during mixing to avoid introducing distortion while processing.

But once you are done, dithering carefully to 16 bit will be OK.

[December-2009]

John Watkinson in Resolution, March-2010 (p. 59): CD "[...] was put together by a skilled group of people who knew what they were doing, and it has stood the test of time. It's not broken and it doesn't need fixing."

Q&A: When Do I Normalize?

noreply@blogger.com (Andrew Levine) — Thu, 14 Jan 2010 12:31:00 +0000

Happy new year :-)

Please check out my latest post, and the 1st one with Ask a Sound Guy: Q&A: When Do I Normalize?

Andrew Levine's DAW-history

noreply@blogger.com (Andrew Levine) — Fri, 30 Oct 2009 07:17:00 +0000

In 1995, while in the process of obtaining my M.A. (in Computational Linguistics & Cognitive Psychology) I started developing multimedia applications. Being a dedicated Macromedia user I did all sound editing in SoundEdit 16 that came as part of the Director bundle. At some point DECK II joined the fray, originally developed by OSC for Digidesign, later sold to Macromedia and finally dropped by BIAS.

DECK was originally used for four track hard disk recording with synchronous MIDI recording and playback. Original DECK also offered moving fader automation, digital mix-to-disk, and unlimited, non-degrading track bounce. Among other awards, DECK (and DECK II) won the 1990 and 1993 MacUser Eddy awards for best music and sound software. [Link]

When in 2002 I got serious about "sound" I first delved into Protools LE that came with the 2ch MBox. While I liked the ergonomics of that GUI I quickly realized that

I wanted to move to OSX now that a UNIX-kernel had (finally) become the basis of the Mac OS
(I used to be a fan of the BeOS; the Wikipedia article states that "iZ Technology sells the RADAR 24, a hard disc-based, 24-track professional audio recorder based on BeOS 5."--amazing!)
I'd want to record more than two channels
Using a Protools system kept you locked to one hard-&software combination

In any case, I returned the MBox and after researching my options ordered a 2882+DSP by Metric Halo and Digital Performer by MOTU, which (as version 4.0) was the first OSX-based DAW on the market.

As with most complex programs it took me a while to get fluent in DP. I stayed with it for several years / iterations (DP 3.x, 4.x, 5.x, 6.0--that one I returned) until I ran into some serious issues, the worst being that with complex projects the engine would not play back the pieces of audio I saw on screen. You can imagine that it made editing quite painful, and I was in the middle of an editing-intense project (Debussy Préludes).

I managed to trudge along for a while, but when a friend suggested I try out the newly revamped Logic 8 I went for it. Luckily for me I used the DAW mostly for editing and as a "tape machine". I used to mix, process & master all projects mainly within the MIO's (Metric Halo's MobileIO interfaces) DSP-matrix. With the advent of the 2d-architecture that became even easier, even when I started moving into the surround realm. But still, the paradigm and handling of Logic was so much different from the one of DP that I enjoyed going back to old (working!) DP-projects from time to time.

And then came Reaper by Cockos.

Floris van Manen, a longtime net-acquaintance and friend of mine had been suggesting I give Reaper a try for ages, and it had been growing to be more and more feature-complete on OSX as time passed. So I finally decided to give it a spin with several new, not especially time-critical projects--and I got hooked!

Reaper features...

a newly designed audio engine, that sounds so good I moved from realtime-bouncing to DAW-rendering for most projects
a very comfortable GUI, that is even skinnable
an editing paradigm that is very smooth and that I got used to in next to no time
amazing support, both from the user community and the developers
all at an unbeatable price

How can you go wrong? The last iteration since I registered has introduced Redbook-CD-rendering, yet again streamlining my workflow, and fixed some small bugs that I had came accross. You just don't get that level of responsiveness from any of the big DAW-companies.

So, if you have not already done so, give Reaper a spin. I see myself happily sticking to this platform for a long time to come :-)

Tracking festivals

noreply@blogger.com (Andrew Levine) — Wed, 28 Oct 2009 05:44:00 +0000

I just noticed that it's been a while since my last blog entry. I guess there was too much "real" work to be done :-)

In any case, having just finished recording six 90 min concerts and one open podium event at the International Jewish Music Festival 2009 in Amsterdam I'll share some thoughts on covering festivals. In this case I was not only in charge of the recordings but also fed the guys handling the sound reinforcement all channels requested by the ensembles.

There has to be one, and only one person in charge of the overall procedure.
Organisation is key. The mikes have to be placed, patched to clearly documented lines & tested for functionality, the gain is determined for optimal signals, then the levels for the SR are set before running an overall soundcheck.
Prepare well in advance and still stay flexible.
Collect the ensembles' set lists right after each performance, make sure they are correct, and perhaps even ask for the musicians' opinions regarding which titles they felt especially good (or bad) about.

Always remember: the recording comes first--at least that's my take on things :-) The record of the music is the one thing that remains and can be re-listened (to), and as such it is always good practice to keep the FOH volume as low as possible and avoid stage monitor bleed in your main mic setup.

Mini-DV-Cameras with 5.1 surround?

noreply@blogger.com (Andrew Levine) — Sun, 19 Jul 2009 10:25:00 +0000

I have been seeing Mini-DV-cameras that are advertised as having a "5.1 channel surround sound microphone". The Sony ECM-HQP has been around as an add-on. SonyStyle says: "It features the capability to record up to four channels of high quality audio for clear, distinct voices and life-like sounds. [...] Offers 3 modes for recording: 5.1 channel Surround mode; 4ch MIC mode;" --mode 3 is sadly missing in the description.

Well, let me tell you it just won't work ;-)

Getting stereo "right" is not trivial, and achieving a soundstage that corresponds with the repertoire, ensemble & performance space gets even more complicated. This can not be achieved by any suit-all clip-on solution--and this goes even more for an enveloping surround soundstage. Recording music & ambience well requires accurate planning, based on plenty of experience, and adapted to the specific situation.

It's like the old joke about how to get to Carnegie Hall. The answer is _not_ to just grab an instrument & a map. Why, even a GPS will only do you so much good.

The C-trilogy (Part 2) - Center, where are you?

noreply@blogger.com (Andrew Levine) — Mon, 06 Jul 2009 06:30:00 +0000

[Continued from Part 1]

Like I said before: "What you want to do is find a spot where you'd place the one microphone required for a monophonic recording."

There are collegues that insist that the center is reserved for instruments that can be heard from the middle of the ensemble, but this makes sense primarily with small groupings. As soon as the ensemble is wide and deep it becomes difficult to determine--without examining the music in depth.

One possible creative idea would be to enhance transparency by placing various spot-miked sources in the center, delayed to let the original impression (from the space between the two front speakers) come first and steer the perceived source in that direction utilizing the Haas effect.

The single microphone used to derive the center will most likely be placed not far off the central axis, and rather close to the ensemble. Since we are dealing with a discrete channel Lou Burroughs' "3:1-rule" must not be taken into account--except for the situation in which you might want to experiment with adding some of that signal to a stereo mix.

The C-trilogy (Part 1) - Center, who are you?

noreply@blogger.com (Andrew Levine) — Tue, 30 Jun 2009 06:00:00 +0000

According to Wikipedia "It is the channel that is mostly, or fully, dedicated to the reproduction of the dialogue of an audiovisual program. [...] In material without accompanying visuals (eg music), the center channel simply reproduces sound intended to come from immediately in front of the listener."

The center channel serves to anchor the phantom image which widens the sweet spot for listeners. On the other hand leaving a discrete--and required--signal out of the L+R channels will cause a problem if the C channel is not configured correctly. Insofar many engineers also feed this material to L+R. See the discussion @ mixonline.

In the C-trilogy I'll do some thinking aloud about (1) which microphone to use, (2) where to place it and (3) how to add it to the mix.

Which microphone to use to supply a discrete signal for a dedicated center channel?

Sidetrack: I say "discrete signal" because with coincident main microphone setups (XY & Blumlein; M/S provides you with a dedicated C anyway) you can easily extract the sum of L+R and place that onto the C. This works slightly less good with near coincident arrangements (ORTF) and not at all with time of arrival based / AB setups.

What you want to do is find a spot where you'd place the one microphone required for a monophonic recording. True, this skill has been steadily devalued, but attempting it lets you appreciate stereo and surround all the more--and gives you something fitting the C.

Cardioid: A microphone of cardioid characteristic attenuates sources arriving off angle. Using one will work only if the ensemble does not expand far in width and depth. On the other hand you will attenuate sound sources from behind the microphone, e.g. favorably reducing audience noise.
Omni: I generally consider an omni, with a uniform frequency response in all directions, to be a good choice to track a dedicated center, especially if it can be placed close to or centrally within the ensemble.
Bidirectional: A fig-8 can also be used facing sideways in a close frontal location, but not centrally within a heterogenous soundfield, e.g. an instrumental ensemble. It works well for a choir, attenuating vocalists singing towards the mic as well as sound / noise coming from the direction of the audience (and conductor).

When recording repertoire with one soloist a close spot can also be fed to the C, suitably delayed (in relation to the main microphone) if you don't want to pull the soloist to the middle--which can be an option on occasion. You'll probably want use a directional mic in this case to have greater control over the pickup range, but if you are close enough an omni will work as well.

[Continued in Part 2]

Recording in Surround, _not_ in "5.1"

noreply@blogger.com (Andrew Levine) — Tue, 23 Jun 2009 18:30:00 +0000

Do you have to use a spaced-pair technique (AB or runtime-stereophony) when your speakers are placed apart?
Are you forbidden to toe in your speakers when the microphones that were used to make the recording were placed in parallel?
Do you have to stand one loudspeaker atop the other to play back a coincidence-stereophonic take (XY)?

The answer to every one of these questions is: No.

The configuration of the listening environment does have an influence on the requirements of the recordist.

The number of channels recorded or mixed up to / down to is--though not linearly--dependent on the number of speakers in use.
The balance of front and rear has an influence on the nature of information the recordist will want to capture.
If raised speakers are to be used in playback that has to be taken into account when tracking the session; currently a rather unusual requirement.

Still, when planning for a recording (primarily) intended e.g. for reproduction on a 5.x system your main microphone configuration does not have to mirror the constellation that will be employed for playback.

What you'll want to capture is a frontal soundstage, with adequate detail, combined with as much "room resonance" as required. For 5.x the latter will amount to a minimum of two microphones recording clean, uncorrelated ambience that can be positioned at the back left and right for spatial envelopment. For the (more common) horizontally level interpretation of 7.x you might decide to use four "ambient spots"--unless you intend to extend the soundstage to full left and right.

True one-point stereophonic systems, with the exception of the Blumlein-setup, employed preferably in great sounding spaces, in my opinion rarely can create a pleasant stereophonic image on their own. I feel the sound design often benefits by some "sweetening" through runtime / phase differences. One-point surround microphones exhibit the same problems--in my listening experience. But that is a separate topic.

Recording on the run

noreply@blogger.com (Andrew Levine) — Mon, 15 Jun 2009 13:50:00 +0000

What do you need to record the best possible sound (in stereo) while spending a minimum?

You need...

High resolution microphones
A pair of professional grade microphone preamplifier
A transparent Analog Digital Converter (ADC)
A digital recording device

With regards to microphones, small omnidirectional capsules are quite immune to wind noise and have a flat frequency response over the relevant spectrum. I am a big fan of DPA's miniature 4060 capsules. They have a healthy output, are reasonably quiet and don't color the sound.

As a 2-channel high quality micpre & ADC I am partial to Metric Halo's ULN-2. It can be battery (or FW-bus) powered and offers astounding sound quality. The 2d-expanded version is even more powerful than the original, but the former is less expensive while offering all a field recordist could wish for.

As for the final component, a digital recorder, you can either use a portable Apple Macintosh, connected to the ULN-2 via FireWire, or a Compact Flash recorder--with digital input! I hesitate to recommend M-Audio's MicroTrack II, but for the price nothing compares currently.

The above setup will cost you approximately...

0900,- US$ (2 x DPA 4060 + XLR-Adapters)
1200,- US$ ULN-2 (The current 2d-version retails for about 1700,- US$)
0250,- US$ (MicroTrack II)
------
2350,- US$

And before you go shouting please remember my initial premise: I aim for "the best possible sound" with no compromises. I didn't say it would be cheap, but this investment is well worth it :-)

For comparison purposes, here is a selection of professional integrated solutions (micpre, ADC, recorder) used in the film business, some of which are way outside of moderate budgets:

Aaton Cantar-X2
(5 x mic + 4 x line + 8 x AES/EBU)
Nagra LB
Sonosax MiniR82
(2 x mic or line or AES/EBU + 2 x line or AES/EBU + 4 x AES/EBU )
Sound Devices 702
Zaxcom ZFR100 (+STA100 for stereo)
(This one is actually quite inexpensive)

Sidenotes:

The 4060's are nearly invisible and easily and unobtrusively placed where ever you like, even over your earlobes for head-based recordings--not that I endorse that approach.
The 4060's will give you years of pleasurable recording, with all kinds of repertoire. OK, you'll be restricted to AB-stereophony, but you can always get other microphones later. Like 2 cardioids for ORTF, e.g. a matched pair of Rode NT5.
You'll be able to use the ULN-2 as an amazing sounding preamp for analog & digital sources, driving your power amp directly, with monitor level control in the analog domain.
The MicroTrack (usually) does what it's supposed to, but I've had several complaints about the original model that I am not certain have been resolved in the meantime. This calls for a separate blog-entry.

...

See also: The Digital Audio Field Recording Equipment Guide by Andy Kolovos.

Metric Halo (MH Labs) audio interfaces for high end consumer audio

noreply@blogger.com (Andrew Levine) — Tue, 09 Jun 2009 12:30:00 +0000

Why should sonophile music lovers not use the best equipment the professionals use as well? OK, some of them might not have need for microphone & line inputs as well as a multichannel ADC, but the rest of the package should appeal to you in any case.

2ch (ULN-2) or 8ch (2882 and ULN-8) output
digital inputs (S/PDIF and AES/EBU) with an amazingly transparent DAC
A great preamp to directly feed your power amp
A great headphone output
Pristine complex DSP-matrix processing, e.g. for modeling digital crossovers, speaker-tuning etc.
"Character"-options
FW-connectivity (Mac only)
In the case of the ULN-8, digital control of all analog output levels,
with a frequency- & phase response flat from DC to Nyquist.

Check them out!

The ULN-2 is the 2-channel incantation with hardware knobs, analog volume control and 10 user-definable presets.

The 2882 is the 8-channel incantation that offers complete digital control via a connected Mac. Great for multichannel (5.1, 7.1) setups.

The ULN-8 is the top of the line technically and audio wise. But if you don't audition it you won't mind missing out on the fun ;-)

One final note about 2d: This add-on card improves the accuracy of the built-in clock and of the PLL (for extracting clock from external sources) significantly, and frees up the internal DSP by handling the channelstrip-based mixer, but you probably could do without as a non-pro, so go for it if you want to save some money and can find a used interface that has not yet been upgraded. I have enjoyed their sound for years and believe you will to.

Errata - Beginners guide to microphones

noreply@blogger.com (Andrew Levine) — Sat, 30 May 2009 10:11:00 +0000

I just read J-Sun's beginners guide to microphones, and here are some necessary corrections:

"There are only really 3 general types of mics Dynamics, condensers and ribbon mics [...]"

Correct: Today there are two general types of microphones for recording music: dynamic and condenser. Ribbon microphones are a special case of the dynamic category.

"The size of the diaphragms in the condenser mics have a big impact on the sound. Larger ones are usually better and are considered to have a smoother, fuller sound."

Correct: Larger diaphragm capsules (usually) have greater electrical output and lower self-noise than smaller diaphragm capsules, but the latter (usually) have better transient response and less coloration of sound coming from off-axis.

"The sounds that are picked up from the sides of the cardioid mic are altered or colored sometimes […]"

Correct: Higher quality (and more expensive) microphones generally achieve a smooth frequency response all around. Please consider though that the polar charts (usually) only display the response for sound arriving on a uniform vertical plane! Sound nearly always gets bounced off the floor and ceiling also.

Regarding omnidirectional microphones it is said that "Picking up alot of the room (ambient sound) also causes a big increase in the low frequency response the closer the mic gets to the sound source."

This is completely wrong! Omnidirectional microphones exhibit no proximity effect--an increase of bass-pickup when the capsule is placed close to the sound source--, which is most prominent in bidirectional (fig-8) microphones and to a lesser degree in microphones of cardioid characteristic.

Sadly this goes on: "Due to this most omnidirectional mics have a high-pass filter (HPF) that reduces some of the bassiness."

Correct: Most omnidirectional capsules are flat towards frequencies you usually don't want to record, like the rumble of air conditionings, subways, planes overhead etc. A high-pass / low-cut filter spares the capsule's internal preamplifier from LF-overload.

The statement, that the use of omnidirectional microphones is beneficial because... "On the plus side it can make your vocals sound fuller."

...is wrong because of the fact that fullness is attributed to the proximity-effect that omnidirectional microphones do not exhibit.

"Some of more expensive mics have switches that allow you to switch between different pick-up patterns."

Correct: There are many expensive single-capsule single-characteristic microphones. Dual-capsule condensers have switches to select how the two signals are combined. E.g. electrically summing the signals of two cardiod capsules, facing in opposing directions and mounted back to back, results in an omnidirectional characteristic of the system.

Linguine with asparagus & red pepper in a cream sauce

noreply@blogger.com (Andrew Levine) — Thu, 28 May 2009 07:43:00 +0000

I recently came up with this receipe:

Peel 1 lb of asparagus and steam it in the pasta pot
take out the asparagus when it's done (test with a fork)
and save the water in a cup
cut the asparagus in half and quarter each stalk
start up the water for the Linguine (I like the De Cecco variety)
cut open & clean three red peppers, then slice them into thin stripes
heat up a chunk of butter and a generous amount of olive oil in a pan
insert the pasta into it's pot while stirring
gently fry the asparagus and pepper strips
--don't forget to regularly stir the pasta--
then, after about 6 minutes...
check the pasta regularly to see if it's al dente
turn off the heat to the pan,
add the asparagus cooking water, 125ml of cream
and some freshly ground black pepper
drain the linguine in time and
mix with the sauce immediately (!)
you might want to add some freshly grated parmigiano

Enjoy :-)

Recording at 32 bit (float)

noreply@blogger.com (Andrew Levine) — Mon, 25 May 2009 16:00:00 +0000

Researching the web for information on the benefit of tracking at 32 bit (float) rather than 24 bit (fixed) I came accross an informative page on Audacity's website:

"Normalised floating point values are quicker and easier to process on computers than fixed integer values and allow greater dynamic range to be retained even after editing. [...] With floating point, rounding errors during intermediate processing are negligible.

The (theoretically audible) advantage of this is that 32-bit floating point format retains the original noise floor, and does not add noise. For example, with fixed integer data, applying a compressor effect to lower the peaks by 9 dB and separately amplifying back up would cost 9dB (or more than 2 bits) of signal to noise ratio (SNR). If done with floating point data, the SNR of the peaks remains as good as before (except that the quiet passages are 9dB louder and so 9dB noisier due to the noise they had in the first place)." [Link]

Still, this benefit is only relevant when...

handling >24 bit (fixed) data
within an engine with a wider path than 24 bit (fixed)
that interfaces transparently with your system.

If you want to record the output of a 24 bit (fixed) ADC you need only track with 24 bit (fixed) accuracy.

As soon as you manipulate that data, before (!) recording it--which IMHO might not be advisable--you will increase the resolution, dependant on the internal bus depth of the engine / interface. In the case of Metric Halo's line of audio interfaces (2882, ULN-2, ULN-8) the bus is 80 bit (fixed), so you will gain some precision by recording the data at 32 bit (fixed). [Thanks to B.J. Buchalter for the clarification]

Music recorded with the Metric Halo ULN-8

noreply@blogger.com (Andrew Levine) — Wed, 13 May 2009 17:06:00 +0000

As part of the beta team I have had the privilege & joy of working with the Metric Halo ULN-8 since December 2006. Here are links to some of the music I have tracked (in stereo and surround) with one of these amazing interfaces:

12baritone (12 vocalists & piano) [The embedded player is at the bottom of the page]
Italian songs
Music by the ensemble wavegarden...
"Starlife"
"Waves of Orange"
Demonstration of my XYtri [5.1 DTS & stereo mp3]

Enjoy :-)

The ULN-8

noreply@blogger.com (Andrew Levine) — Thu, 07 May 2009 06:25:00 +0000

Today the ULN-8 by Metric Halo is officially launched. It's an amazing audio interface that I have been lucky to use for the past 2-1/2 years as part of the beta-program. Nearly half of the current releases of my edition have been recorded, mixed and mastered with this outstanding tool, beginning with the production of the ensemble 12baritone.

I am very glad to have been involved in this process as it is a testimony to the fact that there do exist companies focused on uncompromisingly developing new gear with the future in mind--while improving on the past. The original specs were posted on December 2005, after nearly two years of in-house development and testing. Beta-testing began in December 2006.

What's in the box?

In German we have the saying "Eier legende Wollmilchsau" which translates into "egg-laying wool-milk-sow". The ULN-8 is...

a remote & front-panel controllable 8-ch micpre with 92 dB of no-noise gain
(with real analog sends!)
and, as an electrically uncoupled alternative, 8 line-inputs
sampling at 44.1, 48, 88.2, 96, 172.4 and 192 kHz,
a digital routing system with 80 bit precision, outputting to 8 channels of AES/EBU
(also, 8 channels of digital input...)
an 8-ch preamp with digitally controlled analog gain stages,
managed by an easy to configure 2.0 to 7.1 master controller,
with a completely customizable DSP-matrix to boot.

Not to forget...
sample-accurate metering,
tactile control front-panel
with customizable gain readout options,
MIDI I/O,
SMPTE input & output,
Wordclock interfacing,
a dedicated headphone output
and two ultra high-Z DI-inputs on the front panel.
All in a portable, durable, 1 RU high enclosure
and at a great price!

Did I say it sounds amazing? Well, it does. When I first linked it into my studio I spent a day just listening to my favorite CDs and own recordings. There was depth and transparency like I had not heard before. This can possibly be explained by the fact that the unit is phase flat (as well as power flat) from 0 Hz to Nyquist. You get what's there, and what's there gets you :-)

A recordists take on sound reinforcement

noreply@blogger.com (Andrew Levine) — Sun, 19 Apr 2009 08:55:00 +0000

Once a year I do sound reinforcement for the ensemble wavegarden, recording the concert at the same time. The 2008 event, featuring guest musician Laraaji, appeared in my label as CD-A016.

Channel count is an interesting issue. As a recordist I'd have used four microphones as the main setup--central ORTF for good imaging while negating some of the audience noises plus two channels of uncorrelated ambience for a surround mix--, and two spots for the hang on the left and the flutes / singing on the right. In this case I used an extra 7 channels.

Two headset mics for Carlos & LeAnn
A close XY-pair to balance the small crystal singing bowls
A mic inside a monochord
Two lavaliers underneath the four waveharps

Naturally the room mics don't feed (back) into the PA system, consisting of four active speakers arranged 5 and 10 m from the stage. I placed a low-cut on the outputs and delayed them correctly so that most listeners wouldn't perceive them to be the sound sources.

Something I do, that is ingrained in me as a recording producer, is that I pan the sources according to their position on stage. Admittedly not as far as I'll do on the CD release, but significantly more than I have noted SR-people to do. I try to subtly reinforce the actual soundstage.

Last year a member of the audience approached me after the concert to comment on the fact that it was so nice that I hadn't used the speakers :-)

Dual Mixing

noreply@blogger.com (Andrew Levine) — Thu, 19 Mar 2009 05:57:00 +0000

I just came back from Paris with two live concerts I tracked using a surround setup: A Blumlein main microphone (2 custom built microphones by United Minorities), a dedicated omnidirectional center (Earthworks QTC-1 / QTC-40) and two ambient spots (DPA 4060) mounted on the walls on the left and right side of the stage.

[Add-on: have a listen]

The concerts, part of the 9th iteration of the festival Oboe, featured a fortepiano and an oboe on the first and a trio consisting of oboe, oboe d'amour and bassoon on the second evening. My goal was the uncompromised creation of both a stereophonic and a 5.1-mix.

In creating the realtime-bounces I used a technique I came upon when starting with surround-work and which I now termed "dual mixing". In effect I create two setups in parallel, one for stereo and one for 5.1, continuously comparing one to the other, matching density and detail for the two configurations.

One of the very interesting observations is that this approach is beneficial to both mixes. Auditioning the pure Blumlein-image, slightly enhanced by an addition of the uncorrelated ambient signals to both channels, which can be used to rebalance the sound stage, sometimes gives a more defined impression than the surround-mix. The 5.1-rendition on the other hand has greater density and conveys significantly more of the presence of the recording space than the stereophonic mix.

In the case of these recordings from the auditorium of the Schola Cantorum I increased the density of the 2.0-mix by adding some (-21 dBFS) parallel compression on the sum. The 5.1-mix holds it's own due to the added sound pressure the C- and surround-speakers provide. With this repertoire the significance of the subwoofer is not very great, but it's usage does stabilize the foundation.

Using Metric Halo's 2d-mixer I am able to intuitively and flexibly configure both paths, switch easily between both renditions, and bounce both mixes in parallel, saving a lot of time in the realtime-process. I just tweaked the mixer established for the 1st of the two concerts in five minutes to accomodate the 2nd recording.

My motto: get a grip on all the technical aspects to be able to just sit back and enjoy the music :-)

The saga continues - the Samson Zoom H4n

noreply@blogger.com (Andrew Levine) — Fri, 27 Feb 2009 12:44:00 +0000

After having pointed out the complete neglect of theoretical considerations and empirical observations with regards to the relationship between intra-microphone- and recording-angle in my theoretical appraisal of the Samson Zoom H2 I find the Samson Zoom H4n:

"The mics also adjust for variable recording patterns at either 90° or 120°. Our unique design lets you rotate each mic capsule from 90° (standard) to 120° (wide-angle) stereo for the ultimate versatility in any recording situation." [Link]

The surprises don't end here. Samson states that their "Unique X/Y mic design captures perfect stereo images." X/Y was invented in the 1930's by Alan Dower Blumlein. And who knows what the image labeled as "Conventional Stereo Mic" pertains to. It might be ORTF (2 cardioid capsules, facing +-55° off axis at an intra-capsule distance of 17 cm) but a more accurate specification would have been interesting.

[All © by their respective owners]

A theoretical appraisal of the Samson Zoom H2

noreply@blogger.com (Andrew Levine) — Wed, 25 Feb 2009 11:33:00 +0000

Samson states that "The H2 is the only portable recorder with 4 mic capsules on-board for 360° recording. [...] For maximum flexibility, you can record from the front of the H2 in a 90° pickup pattern or the rear of the H2 in a 120° pickup pattern [...]" [Link]

A view of the capsules can be found here.

As can be seen from the graphics Samson provides they expect the pair of forward-facing cardioids, arranged at +-45° off axis (for a total of 90°), to result in a stereophonic recording angle of 90°. This is not correct. According to Eberhard Sengpiel the recording area is much wider at 196°.

For the rear-facing microphones the arrangement of +-60° off axis (for a total of 120°) is said to result in a recording angle of 120°. This again is not correct. The calculated recording angle is 158° --less than the 196° found in the frontal configuration.

The fact is: a greater off-axis angle results in a smaller recording angle! This is not only mathematically correct but can be easily verified in practice.

In sum, the total recording area of the 4-Mic-setup in the Zoom H2 covers 354°, composed of a larger forward-facing and a smaller rear-facing area.

What should it sound like?

Both (near-) coincident pairs employ cardioids, which will result a high degree of diffuse-field correlation, perceptually compressing both the frontal and the rear-facing soundstage. Whereas I have not personally experimented with this recorder I believe a more complex setup is required to capture a realistically enveloping experience.

I find both the coincident approaches (Soundfield, Double-M/S) and runtime-based solutions (Atmos, Surround Decca-Tree, OCT/OCT2) interesting. My XYtri features a composite approach. But all of these are definitely way more involved than a compact, handheld solution. As such it might meet many user's needs.

[All © by their respective owners]

Innovations in recording technology

noreply@blogger.com (Andrew Levine) — Wed, 18 Feb 2009 09:10:00 +0000

In recording technology you have...

The microphone capsule
The microphone-internal amplifier
Gadgets to support the microphone and decouple it from structure-borne sound
Gadgets to protect the capsule from the elements, especially wind
The microphone-preamplifier (mic-pre)
Cabling
The sound recorder

Digital technology originally introduced an additional step before the sound recorder:

The Analog-Digital Converter (ADC)

In the case of digital microphones the ADC is placed either right after the capsule or after the microphone-internal amplifier.

I analysed the latest press releases by five of the main microphone manufacturers: AKG, DPA, Neumann, Schoeps and Sennheiser. This is what I found...

AKG

"The AKG C 414 has been successful for more than 30 years due to continuous improvement of technical specifications, useful features, and ease-of-use. [...] Our new C 214 was designed as a budget-friendly alternative to the leading-edge C 414 family." [Link]

"The new AKG Perception Series [based on the P 100, P 150, P 200 and P 400] includes the P 120, P 170, P 220 and P 420" [Link]

DPA

"DPA Launches 5100 Mobile Surround Microphone at IBC 2008. [...] Optimum channel separation and directionality are achieved through a combination of DPA’s proprietary DiPMic (Directional Pressure Microphone) technology, which mounts interference tubes on the L/C/R capsules, and the use of acoustic baffles that further preserve the accuracy of levels between the discrete analogue output channels. [...] The centre channel can be overridden by an external boom or lavalier microphone attached via a built-in XLR-F connector." [Link]

ProAudio Review (03-2008): "DPA has defined a new microphone sub genre [in Fall 2007]: 'the musical shotgun.'" [Link]

Mix Online (04-2005): The "DPA WINDPAC Folding Location Windscreen [...] incorporates new thinking about an old problem." [Link]

Neumann

Solution-D Digital microphones [Link]

Schoeps

"The 'Open Cardioid' directional pattern, occupying a point on the spectrum between the wide cardioid and classic cardioid patterns, was developed by SCHOEPS in 2008." [Link]

"The SCHOEPS CCM microphone series [...] has now been enhanced with a top-class elastic suspension. [...] Attenuation of solid-borne noise from the boom is achieved by the unique architecture of the CINELA OSIX series." [Link]

Digital Microphone Amplifier CMD 2U [Link]

Sennheiser

"With the launch of the MO 2000, the audio specialist has expanded its portfolio with a new type of transducer that represents acoustic signals on the basis of variations in light intensity." [Link]

"In the optical microphone, light from an LED is directed onto a reflective diaphragm via an optical-fibre cable (transmitter optical waveguide). The membrane reflects part of the light into a receiver optical waveguide. If the diaphragm is moved by sound signals, a displacement of the reflected light beam occurs, with the result that more or less light is coupled into the receiver waveguide. At the end of the receiver waveguide, a photodiode converts the light intensity variations into electric signals. [...] The optical principle is the only microphone principle in which the microphone head and the electronics can be located far away from each other." [Link]

According to the specifications this technology is not yet usable to record music, but as a completely novel approach it definitely deserves attention.

"Audio specialist Sennheiser is unveiling the MZD 8000 digital module at AES 2008." [Link]

MKH 800 TWIN studio microphone: "At its output, the dual capsule microphone provides both audio signals separately, allowing the mic’s pick-up pattern to be adjusted freely and remotely at the mixing desk, and then optimised later in the calm of the post-production studio." [Link]

Interestingly enough I had a discussion with a microphone specialist about outputting discrete signals from a dual capsule around this time. Besides the flexibilities in post I see a useful application in generating surround-channels from the sides of the microphones facing the audience.

Summary

Of the five microphone manufacturers I researched DPA looks to me to be the most productive in terms of in-house innovations pertaining to recording music. Sennheiser is a runner up with it's dual-output-capsule and for investigating a completely novel transducer-design. Neumann and Schoeps primarily announced their moving into the realm of digital microphones. What I like about Schoeps is their embrace of surround setups developed by forward thinkers, although DPA also sports a wide range of mounts for various surround configurations.

[All © by their respective owners]

Why double sample rates? (88.2 kHz / 96 kHz)

noreply@blogger.com (Andrew Levine) — Mon, 09 Feb 2009 07:15:00 +0000

I just read a short description of Why is 96/24 better than 44.1/16? in an otherwise quite knowledgeably assembled FAQ. While it is true that "a sampling rate of 96kHz gives a better digital representation of the waveform and yields a usable audio bandwidth of about 48kHz" this does not explain why recording made at double the usual sample rates (of 44.1 kHz / 48 kHz) should sound better.

Just consider...

What is the limit of your hearing, frequency-wise?
How many instruments have harmonic content above 20 kHz --that you really care to hear?
How many microphones record linearly --or at all-- above 20 kHz?
I recorded Gaita, an incredibly overtone-rich spanish bagpipe, optimized for open-air usage) twice. Once using a Blumlein-setup of two RoyerLabs SF-1 ribbon-microphones, with their characteristic, gentle high-frequency rolloff, and once using an AB-setup featuring two Earthworks QTC-1's (now called QTC40), practically flat from 4 to 40 kHz. Which do you think sounded more real, and which sounded more pleasant? ;-)

Well?

Listen to the Jata da Mahia for a sample of Gaita recorded with Blumlein-ribbons.

I believe it's more about the optimal SR the ADC was designed for, as well as the quality of the filters used, as all content above the Nyquist frequency has to be filtered out before the conversion.

Then again many audiophile are intrigued by the "24-96" label. If you are a true sonophile this may not be quite as much of an issue. Trust your ears!

Choosing microphones

noreply@blogger.com (Andrew Levine) — Sun, 01 Feb 2009 12:35:00 +0000

These are my thoughts on choosing the right microphones for a variety of scenarios, based on my way of approaching recording projects.

First of all, here are a few of the fundamental observations I have found helpful:

The more live a recording situation is, the easier it will be to make it sound live. Simple ;-)
The fewer microphones are used in the mix, the clearer the result will be. This is due to...
- Phase issues: the fact that all sounds occurring will to some degree be present in all microphones, but at slightly differing time; they might also be colored due to non-linearity in off-axis pickup.
- Additive background noise, which can also cause problems when spots are panned to a specific location.
Music is a temporal and spatial event. Musicians generally move (their instruments) and this causes a movement of the sound source which dynamically interacts with the recording space.
Never forget the importance of the microphone preamplifier & the ADC!
There is a distinct difference between far and close-up use. The character of a microphone's sound is much more significant in the latter case.
A microphone does not know what it is recording. Be suspicious about type-casting (guitar-mic, snare-mic etc).

Based on the above assumptions there are a few questions to be asked about what you wish to record.

Hauptmikrofon / main mike setup

You need a set of closely matched high quality microphones for this purpose. The key features to look for are neutrality, resolution and quietness. You don't want off-axis sound to be coloured, especially when using omnidirectional microphones in an AB-setup.

Blumlein-setup

The Blumlein pair requires two fig-8 microphones. Keep in mind that all bidirectional condensers exhibit (a pronounced proximity effect when used close up and) a significant low-frequency-rolloff in the far field. Interestingly ribbon microphones don't necessarily share that characteristic.

>= 88.2/96kHz-recordings

If you plan on recording high resolution audio it is advisable that the microphones (capsules and capsule-preamplifier) reach up to half of the sample rate, preferably in a linear fashion. Check out the Nyquist frequency for details.

Ambient microphones

Since these microphones will be way out in the reverberant field where the sound pressure level is relatively low, low self-noise is desirable. It is when you want to make the most out of directly approaching sound that the use of boundary layer microphones is indicated.

Spot microphones

Spot microphones is the area where recordists can get most creative. Any microphones' sonic character becomes more pronounced when used close to the sound source, especially with directional characteristics.

The blumlein records microphone closet

1 * Earthworks QTC-1 (QTC40)
This used to be one of a set, but a musician literally killed one in a session --tripping over a cable and making the mic crash tip-down. I often use it as the mid-channel for an M/S-setup.
2 * Earthworks QTC50 (matched pair)
Optimal for high-resolution AB-recordings. Their miniature capsule is extremely linear and amazingly fast.
2 * Elation KM201 MSP4 (matched pair) - 2 Capsules: cardioid & omnidirectional
A good general purpose spot microphone. I have also frequently used it as an AB-spot within choirs when recording oratorios.
2 * DPA 4060 (matched pair)
A beautifully unobtrusive microphone. It does have a self-noise of 23 dBA but works great as a close spot inside a bass, auto harp, udu etc. I also frequently used it as a boundary layer mic taped under a grand piano.
2 * Oktava mk102 UM-mod (matched pair)
The United Minorities tuned Oktava cardioid is fabulous for ORTF-recordings, as a mid-channel for M/S, and as a close mike for extremely quiet sources.
6 * Rode NT-5 (3 matched pairs)
This inexpensive microphone is a good spot for many applications.
Currently all six of these microphones are the basis of my XYtri-setup.
1 * Rode NT-4 (matched pair)
Using the same capsules as the NT-5 this microphone is quick to setup and does a good job. I have used the battery-powerable mic as an external microphone feeding my Mini-DV-camera for many interviews.
2 * RoyerLabs SF-1 (matched pair)
A beautiful ribbon microphone. Great for sonophile Blumlein recordings. Also a very nice spot for many overtone-rich instruments: gaita, trumpet, violin-overhead, ... due to it's gentle high-frequency roll off.
2 * UM-1 (matched pair) - Switchable: cardioid, fig-8, omnidirectional
Since I have received this United Minorities microphone it has been the staple for many of my recordings. It is very linear, extremely accurate in the time domain and eminently usable in a multitude of recording situations. I love it!

Disclaimer: I have occasionally used the famed microphones by Neumann and Schoeps, but I never felt the need to buy one of them. Equally good microphones are available --at a more reasonable price range. Let your ears be the judge :-)

Playing back Blue-ray-disks on an Apple Macintosh

noreply@blogger.com (Andrew Levine) — Fri, 23 Jan 2009 20:30:00 +0000

I have been researching high definition mediums, Blue-ray becoming the center of my focus. It offers high resolution video with 1920 × 1080 pixels and high resolution audio with up to 8 streams (for 7.1) of possibly uncompressed audio.

Whereas you'll want to output high definition video to your display, which likely is HDMI-interfaced and HDCP-compliant already, audio is another matter - especially for us sonophiles. It has been a fact from 2005 (Link) that there are difficulties involved in getting the pristine digital stream to a dedicated DAC.

How do you play Blue-ray-disks on an Apple Macintosh?

Short answer: Not easily.

Longer answer: It works if you...

Have a Blue-ray-capable drive connected.
Run Microsoft Windows (using Bootcamp, Parallels Desktop or VMware Fusion) on your computer. I think this is nasty ;-)
Run Slysoft AnyDVD HD to disable the copy protection, which I expressly do not recommend as it might possibly be considered a criminal activity!
Use a PC player software (e.g. VLC to access the M2TS containers).

Why would you want to do this?

You have a nice big (digitally accessible) screen anyway.
You want to route the audio streams to a DAC of your choice, connected to your Mac.

The cost will be around 250 US$ (110 US$ for the drive, 80 US$ for the emulation, 60 US$ for the access enabling software) - and then you have to get Windows to run --at a sufficient speed for Blue-ray-playback-- and route the audio back into Mac OS.

Is there an easier way?

The way I'll (try to) go is...

Get a standalone Blue-ray-capable player with S/PDIF audio output, coaxial or optical according to your input solutions.
Beware: You'll get 6 channels, no 7.1 currently, at resolutions of up to 96 kHz (with DTS).
Connect the HDMI-output to your (HDCP-compliant) screen.
Feed the S/PDIF into your Mac's or your audio interfaces' digital input.
If you did the former, use Soundflower to route the audio to (a decoder if you are dealing with a Dolby Digital or DTS-encoded stream and feed the linear PCM to) the DAC of your choice.

Please don't look confused! After all you wanted to know :-)

Sidenote

PC-based Blue-ray-playback of movies secured with AACS requires a Protected Video Path (PVP) as well as a Protected Audio Path (PAP) for all but linear-PCM-streams. This means that compressed high resolution audio >16 bit and >48 kHz (e.g. Dolby TrueHD or DTS-HD MA) must be downsampled to 16-48. [Link, 11/05/2008]

[All trademarks are the property of the respective trademark owners]

Open ideas vs. proprietary techniques

noreply@blogger.com (Andrew Levine) — Mon, 19 Jan 2009 16:25:00 +0000

Eberhard Sengpiel said: "Even after you've 'finished your studies' you have to keep improving your recording technique and everything related. It's nothing but a lifelong process." (Link in German)

In 2008 I developed a surround microphone system that I call "XYtri". I presented it at the VDT International Convention 2008 (Abstract) and am building a small XYtri-website with detailed information and sound samples.

Why "give away" something like this?

A website of a collegue of mine states that he has developed a proprietary main microphone setup which records the natural ambience of the recording space. That's all he says, so if you want to know what it is and what it sounds like you'd have to hire him. Another has developed a proprietary system to enhance ambience in headphone mixes. He let's you listen to many samples though and markets it as a service he renders to potential clients.

My incentive is the belief that only by exposing a new technique can you get your (competitors or) collegues to try it out, hopefully give you feedback to improve the design and possibly set in motion an exchange of ideas that can only improve the quality of everyone's work. This is why scientists publish. Peer review is good! I have frequently benefited from other Tonmeister's sharing their knowledge and experiences, so here I go too :-)