blumlein records

Stereo, quad, 5.1, 7.1,… Ambisonics to the rescue

2010-12-29T11:23:00.008+01:00

Prelude

Imagine you did a nice recording of an oratorio. You created a fine stereo mix when the client calls you...

"Hey, did you see the guy with the video camera at the concert? That was just the angle for details. He also had a camera on a tripod, and his montage looks great. Wouldn't it be great if we had a 5.1 up-mix of your recording?"

No problem as you had two outriggers for ambience. Lucky they were there! So you dutifully do a 5.1 mix that will shake things up. Soon afterwards your client rings you up again.

"Nice work on that! The video guy is very happy. Actually I'd love to hear it as well, but I've just got this quad setup from way back. I wired it all up now so could you please give me a 4.0 mix?"

OK, back to mixing it is, and guessing as to how the balance will work for him, as you are loath to tear apart your standard conforming monitoring rig.

"Thanks mate, I feel like I'm back in the 70's, but with a much better sound of course! Now, the video guy said he'd love to release this recording--not on DVD but on Blue-ray, do you dig it? He'll want 7.1 for that though, can you please do that?"

OK, stereo, 5.1, quad, now 7.1, every time requiring a dedicated mix to accomodate the discrete speaker feeds. So you do the 7.1 as well, moving to a friends mixing space that takes some getting acquainted to, but you get it done in the end.

"Lovely man! Your 7.1 mix really rocks! I get great feedback from everyone who saw the clips, and even the local TV wants to broadcast the performance. Say, is your mix mono compatible?"

Mixing to B-format

Ambisonics, developed in the early 1970's by the mathematician Michael Gerzon has a solution for this problem: Mix in B-format.

"In the basic version, known as first-order Ambisonics, sound information is encoded into four channels: W, X, Y and Z. [...] The W channel is the non-directional mono component of the signal, corresponding to the output of an omnidirectional microphone. The X, Y and Z channels are the directional components in three dimensions. They correspond to the outputs of three figure-of-eight microphones, facing forward, to the left, and upward respectively." [Wikipedia]

For a lot of music (and most playback configurations; really, how many setups with height channels have you encountered so far?) the Z-channel can be omitted leaving the need for three channels for 1st order B-format.

What do you need to do when recording?

If the ensemble is rather close and not too expansive a Nimbus-Halliday (developed by Dr. Jonathan Halliday, research director at Nimbus Records) can do the job. If not a main mic setup based on symmetry or better yet arranged in an a isotropic configuration can be used. Use directional spot microphones according to taste.

What do you need to do when mixing?

Configure your DAW for a 3 (or 4) channel main bus. Put an Ambisonic decoder (e.g. by Bruce Wiggins / WigWare or Daniel Courville) onto that bus. Now use an Ambisonic panner (see previous links) to pan the sources to specific locations, and that's about all there is to it.

To be continued...

The "4x8" approach to quadraphonic miking

2010-08-31T08:45:00.010+02:00

How do you capture a concert harp in an amazing sounding acoustic space in surround, playing solo pieces, compositions with additional playback through 6 widely distributed speakers as well as a duo performance involving both the main as well as several surrounding cavern like spaces? This was the specific case for one of the fine concerts during Berlin's __tiefKLANG festival.

I came up with what I named the "4x8" approach to quadraphonic miking: four microphones of a fig-8 characteristic are placed in four corners of a square centered around the main performer. All microphones are angled 45° inward and facing downward towards the center, kind of like the cross-hairs of a reticle, rotated sideways by 45°.

All four microphones were placed at a height of 170 cm. Regarding the distance between the corners of the "4x8" I went for 220 cm in this case, but like with the flexible DECCA-tree you'll need to listen and adjust spacing and height to get it just right :-)

What are the benefits of "4x8"?

Assuming the shape of the room is a true rectangle and the performer sits in parallel in relation to the walls at the sides and perpendicularly / orthogonally in relation to the walls at the front and back the 0-axis of the fig-8 capsule is equally distant from any parallels between walls. Tilting the microphones downwards towards the performer distances the vertical axis of maximal pickup from parallels between floor and ceiling and also places the audience closer to the vertical 0-axis.

I attached a small omnidirectional condenser to the resonant base of the harp that provides the C-channel of my 5.1 mix and taped an omnidirectional boundary layer microphone to the wall, centrally behind the quadraphonic configuration, that gives me the LFE after appropriate low-pass filtering. You do need to be able to compensate for the LF-rolloff of condenser fig-8 designs!

The "4x8" approach served me well for this setting, with a stationary central sound source and the necessity of capturing ambient sounds in a 360° radius. be careful when employing it with an instrument that moves a lot in space and has strongly directional qualities at any frequency range. You will quite likely end up with the central sound source jumping around sideways and front to back on quadraphonic playback.

2d, 3d, how many dimensions really?

2010-06-11T07:50:00.024+02:00

I've done some more thinking after having posted What is 3d audio…

Strictly speaking a dimension is "a spatial characteristic of an object; that is, length, width, or height." (I'll skip the fact there was a best-selling shampoo named alike; thanks Wikipedia ;-)

So a static image has two dimensions whereas reproduced sound originally has only one: the lateral location of a transducer--as it changes over time, resulting in the propagation of a sound wave through space by compression and rarefaction of air.

You can see that it's necessary to add the parameter of "time" to the available physical dimensions.

Moving pictures, be they film- or video-based, add the aspect of change over time to their dimensionality. So a traditional movie has three dimensions. As explained above sound can not exist without time--even sound pressure level (SPL) readings need a reference value which they are based on--, so it requires two dimensions.

"3d"-movies are four-dimensional as they include depth,
stereophonic sound is three-dimensional as you require _two_ sources of correlated sound waves, but the result is a forth dimension: depth can now be perceived (when located at the correct position: usually one point of an equilateral triangle, with both speakers facing in the observer's direction).

You can now add more dimensions to the sound by introducing more sound sources…

L+R+Surround in the rear
Quad (L+R+Ls+Rs)
5.x (L+C+R+Ls+Rs)
etc.

…or you can consider the introduction of a second sound source (with relation to a listener's position) as being the establishment of an array of sources on a two-dimensional plain, which would result in the observation that adding another speaker does _not_ increase the number of dimensions.

To summarize, so-called "3d" in movies or video, as well as surround sound, require four dimensions (three physical ones plus time), but only surround sound has the capacity to envelop the listener and create a truly three-dimensional experience, placing the observer at the center of all action. In that respect "3d" movies are about where stereophonic sound was in the early 1930's ;-)

Disclaimer: I very much enjoy watching good movies (with well crafted soundtracks--preferably in surround) on DVD!

What is 3d audio?

2010-06-07T12:08:00.004+02:00

The traditional movie theater / home dvd (or video for ye old-timers ;-) experience is flat. Conventional screens have two dimensions, width and height, but no depth. What I mean by that is that while a picture certainly can seem deep, any perceived depth is just that: perceived. It is quickly recognized as such, and does not change when one moves one's eyes or head.

Audio on the other hand, especially simple stereophonic recordings, have always been able to convey depth--but no height. The latter may be due more to the playback setup than the recording technology itself! I had an amazing experience of height information encoded in a simple AB recording when attending a presentation of the "Bloomline loudspeaker system" (formerly named the "inaudible loudspeaker") by Leo de Clerk at 2008's VDT International convention. [Abstract]

So what makes video three dimensional? It is either (1) depth or (2) envelopment. And what puts audio in the same position: either (1) height or (2) envelopment. So the question at the start can be answered in that surround sound, be it quad(raphonic) or 5.0, 7.0 etc. already satisfies the requirement.

Surround sound is 3d audio!

And so it appears we have had 3d audio long before moving pictures went 3d. BTW, adding height to the equation introduces yet another dimension with little or no sweat ;-)

The album lives!

2010-05-12T08:54:00.005+02:00

According to Nate Anderson's The death of the album "40 percent of all [Tunecore] sales were single-track downloads, 57 percent were streams, and a mere 2.3 percent were full albums." [Original markup] Please note that the percentages refer to the strange numbers on the graph in relation to 65.18!

I think the real question is: How much income did Tunecore-sellers generate?

Turning sales-percentages into dollars the situation looks very different to me: streams (costing 1c) made 5.70$, albums (costing 10$) 23$, and single tracks (costing 1$) 40$. With albums & tracks a commission of roughly 1/3 is deducted from the profits. In the end people selling albums generated more than twice the amount of money that people selling tracks did, and the overall direct income from streaming seems comparatively negligeable.

All of this besides the fact that one has to closely examine the demographics of the potential clientele. E.g. lovers of the opera and symphonic music will rarely purchase a single aria or movement--and Wolfgang Spahr writes: "Classical music was the big winner at retail in 2009 according to a statement issued by the German Federal Music Industry Association (BVMI). Double-digit growth was achieved in both volumes and revenues." [Classical Music Sales Soar In Germany]

Errata - Exotic Positions

2010-04-09T09:20:00.000+02:00

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?

2010-04-08T17:25:00.000+02:00

"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?

2010-01-14T13:31:00.002+01:00

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

2009-10-30T08:17:00.007+01:00

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

2009-10-28T06:44:00.003+01:00

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?

2009-07-19T12:25:00.001+02:00

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?

2009-07-06T08:30:00.001+02:00

[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?

2009-06-30T08:00:00.001+02:00

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"

2009-06-23T20:30:00.005+02:00

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

2009-06-15T15:50:00.003+02:00

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

2009-06-09T14:30:00.001+02:00

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

2009-05-30T12:11:00.007+02:00

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

2009-05-28T09:43:00.004+02:00

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)

2009-05-25T18:00:00.010+02:00

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

2009-05-13T19:06:00.008+02:00

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

2009-05-07T08:25:00.007+02:00

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

2009-04-19T10:55:00.000+02:00

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

2009-03-19T06:57:00.002+01:00

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

2009-02-27T13:44:00.004+01:00

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

2009-02-25T12:33:00.006+01:00

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

2009-02-18T10:10:00.003+01:00

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)

2009-02-09T08:15:00.000+01:00

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

2009-02-01T13:35:00.003+01:00

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

2009-01-23T21:30:00.005+01:00

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

2009-01-19T17:25:00.001+01:00

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 :-)

What kind of audiophile are you?

2009-01-09T06:52:00.007+01:00

I've been thinking about the so-called "audiophile" crowd lately and come to the conclusion that they are a composite of many different predilections, some of which are listed here in alphabetical order:

Fetishist
Do you buy only 24k gold CD's and use the most esoteric of digital interconnects?
Assuming you employ a jitter-immune DAC the digital source / medium & connecting cables are of no importance. Please test double-blind if in doubt.
Freak
Do you believe that vinyl records sound better than digitally encoded music converted back into the analog domain?
Assuming a high-quality, very accurately clocked ADC is used and the high-quality, accurately clocked DAC employed is jitter-immune, an exact digital "copy" of an analog source, fed to the same amplifier stage, is indistiguishable from the analog source. Please test double-blind if in doubt.
Geek
Do you feel that the technical specs of gear are all that counts?
The well tuned ear is the ultimate judge. Technical specs are obviously also very important, but, besides recommendations from people (with ears) you trust they should primarily help deciding what gear to audition.
Touch&Feel
Do you believe in "you get what you pay for" beyond what most people --or even audio professionals-- would consider a reasonable price, enjoy the look & feel of extra heavy gear with a thick metal frontplate, wooden sidepieces and custom made knobs?
It is true that good sounding gear takes a lot of R&D, which must be payed for, and that high quality components don't come cheap, but, as a music enthusiast, please use the resolution of what your own ears can discern as the landmark and give the money you save to a charity.
Sonophile
Do you love to listen? Do you enjoy the vibe of a live performance, the sound of a "real" space, the incredible span of real-life dynamics, the tinyest details that may still be discerned?
A true sonophile is in love with things that can be enjoyed when listening to, no matter which genre. Great musicians, performing extraordinarily well in a nice sounding space, recorded optimally and presented (mixed and mastered) in the best possibly way, retaining the excitement of "being in the moment".

So, what kind are you?

I confess to find myself reflected to varying degree in several of the above categories. I am no audio fetishist and do trust in a completely digital workflow, but there surely is great analog gear about and there are people who can use it to achieve optimal results. To some degree it's a question of what environment you are comfortable with. I do like to browse gear by looking at it's specs, and I often feel tempted by numbers, but due to budget constraints I don't run amok ;-) The same goes for the looks of some equipment. Nice...
But in the end I am a sonophile. I just _love_ to listen. And I hope there are many others around, because even though I don't know how to explain this I believe it will help make this world a better place. Listen closely, listen well, enjoy!

One point stereo A-B?

2009-01-04T10:44:00.005+01:00

While researching audiophile labels on the web I came across Waon RECORDS. Their offerings look interesting although most of the website is in Japanese only, which sadly I can't read.

Besides session snapshots the website informs that their recordings have been produced using "One point stereo A-B". This is an incorrect use of the term.

AB-stereophony takes it's name from a line connecting the most likely distinct points "A" and "B". It is runtime based, which means it employs differing time-of-arrival cues / phase differences embedded in the signals arriving at both microphones. Quite obviously this requires that the locations of the capsules are spatially distinct.

When you separate the microphones by about 50 cm a signal coming from one side of the configuration arrives at the opposite microphone delayed by about 1.5 msec, which, according to the rule of the 1st wavefront, results in it being localized towards the side of the speaker which first emitted it.

One-point stereophonic recordings can only be the result of coincident setups, where both capsules occupy (approximately) the same spot in space. This goes for XY (with two cardioid capsules), Blumlein-stereophony (with two bidirectional / fig-8 capsules) and M/S (with one bidirectional and usually one cardioid capsule).

This is not to say that you can't craft great recordings using a spaced pair of omnidirectional microphones. Well designed and built omnis can be extremely precise in the temporal domain, have a linear frequency response, don't colour sound arriving off axis and don't exhibit a proximity effect. Used in quiet, good sounding venues they can be used to capture the depth of the space very convincingly. Localization is impaired when compared to coincident setups, but that is a compromise the recordist might want to make.

On good monaural sound

2008-12-17T06:37:00.007+01:00

Andrew Lutes just said @andrew_levine Personally, I love mono. I love surround too but mono seems to get right to the point. I feel so too, if it is _good_ mono, meaning the sound is...

Based on a good mix
Intelligible
Wide dynamic range
Wide bandwidth

The issue is that you can only get that by design. You can not e.g. condense two tracks based on runtime-stereophony to mono without comb-filtering intruding. You have to work with good material to start with and then carefully craft the mix, and then all will be well :-)

Making a better latte

2008-12-12T09:25:00.003+01:00

Expanding and refining my 11 steps to a fine latte...

Go for a double. Making a double-shot really is much easier than preparing a single. I think it's due mainly to the larger basket that masks any inaccuracies when dosing. Also, with the Rancilio Silvia and baskets by Lamarzocco (Rancilio's single does not work for me), it is easier to tighten the portafilter using the double-shot basket.

These are my usual steps:

Make sure the machine is heated up well
Press the steam switch
Fill a large (250 ml) cup with milk nearly halfway
Let out steam until it comes out dry, but don't let the lamp go off
Stretch the milk to the top
Depress the steam switch
Sink the wand and heat the milk thoroughly
Bump the cup to get rid of larger bubbles
Wipe the steam wand clean
Insert the portafilter into the (doser-less) grinder
Test the brewgroup to see if the temperature is right (no extreme steaming)
Dose into the portafilter
Tamp well
Brew the shot (about 27 sec for 30/60 ml - single/double)
Old - Pour it through the milk
Old - Rinse the shot glass with some extra hot water
Clean the group head, basket & portafilter
Enjoy

I have now found that the result is _much_ better when steaming the milk in a steaming pitcher and pouring the result onto the espresso, brewed into the preheated cup. Now I can enjoy my single-shot as much as one half of a double shot. Surprising but replicable.

BTW, I very much enjoy my Ilsa Anniversario lattiera. It makes steaming even more fun.

Someone did invent Stereo

2008-12-08T09:30:00.011+01:00

We take the stereophonic listening experience, with images between two loudspeakers and it's convincing illusion of width and depth for granted, but it has been used commercially for less than fifty years. It is also very interesting to note that many of the advancements in this area were fuelled by motion pictures. People were seen to move from left to right, and the sound should match. Sound is understood as an enveloping experience, and --simplifying a bit-- along comes Surround...

1857
Monaural sound is first transcribed by Édouard-Léon Scott de Martinville
1877
Thomas Edison's phonograph cylinder enables the recording & playback of sound
1880
Stereophonic sound, realized by Clément Ader, has a cursory appearance
1930
Binaural Sound is studied in depth by Alan Dower Blumlein and Arthur C. Keller
1960
The age of Stereo begins
1970
RCA Records introduces Quadraphonic sound
1970
Michael Gerzon extends on Blumlein's work with the development of Ambisonics
1970
Walter Murch thinks up 5.1-Surround for "Apocalypse Now"

Alan Dower Blumlein, born on June 29, 1903 in Hampstead, London, invented two out of the three "pure" techniques for recording sound in stereo [stereos = "solid" and phōnē = "sound"]: XY (subsuming the special case of the Blumlein-configuration) and M/S. Both are termed coincident stereophony because the capsules of the two microphones involved are located at (about) the same spot. Blumlein's work has a profound influence on Michael Gerzon who developed Ambisonics as a followup to Quadraphonics.

The third traditional approach to recording in stereo, AB-stereophonics, is based not on differences in the sound pressure level of signals reaching the two microphones but rather on differences in time-of arrival (or the "runtime" or phase) of the signals. It was developed at Bell Labs by Arthur C. Keller under the supervision of Dr. Harvey Fletcher. Stokowski was delighted: ''Listening monaurally,'' he reported to Bell Laboratories, ''gives me the sensation of the music being choked and crushed together. Binaurally (i.e., in stereo) the music sounds free, spacious, and the choked sensation is gone." [Link]

Coincident and runtime-based configurations give rise to near-coincident techniques, one of the most mentionable being ORTF, developed by and named for the Office de Radiodiffusion Télévision Française.

5.1 for Apocalypse Now?

Movie-sound before the 1970's mainly utilized stereo, plus an effects channel at the rear of the audience. Walter Murch wanted to be able to spread the rear ambience to better enfold the listeners. And then there was the matter of the voiceovers...

The target of "Apocalypse Now", big movie theaters, were not suited for Quadraphonic playback. With stereo there is a more or less defined sweet spot in the center between the two speakers. If the voice of the actors or a disembodied voiceover came from the phantom-middle, that is if it were mixed into the front left and front right stream in equal parts, it would appear to come from the side the audience were located on. Members of the the audience on the far left would hear the voice coming from there. Placing a center speaker behind and in the middle of the big screen anchors the voice to that position more firmly.

The establishment of this setup as a mainstream commodity doesn't mean recordings have to be made with an explicit 5.1-mapping though. But this is (yet) another topic, to be discussed soon :-)

[Thanks to the Wikipedia & Steven Schoenherr]

[100204 - Quadraphonic sound came before Ambisonics]

Analysis - Erleuchtung garantiert

2008-12-05T09:30:00.001+01:00

The Geman movie Erleuchtung Garantiert (Enlightenment Guaranteed) by Dorris Dörrie features Solweig Bores as the sound designer and Chris Price (I) as "sound"-person.

I was positively surprised by it's nice rich 5.1 ambience that did have a very real and unpolished feel to it. And this in a movie that uses quite a lot of camcorder-imagery. Have a listen!

Analysis - Terminator 2

2008-11-17T13:15:00.005+01:00

Analysis - Terminator 2

Terminator 2 by James Cameron features a huge crew of sound people and Gary Rydstrom as sound designer and sound re-recording mixer. Three assistand sound designers credited are Scott Chandler, Tom Myers and David Slusser.

The sound during the credits and introduction is already very nice. Clear sounding, highly dynamic, with well-done music and dramatic voice-over. It is interesting to note that the surrounds channels are used sparingly. I imagine they will be more developed at the start of the feature.

Sure enough, when Schwarzenegger has appeared (06'30) the ambience and the sounds of the visual scanning process are presented in an enfolding manner. There is a slight change when he opens the door to the bar and an interesting phasing effect that more firmly established the character of the in-the-head mode.

The incidental music is faded out as the mechanical drone and orchestral score move to the foreground in waves. All very subtly done.

At 09'15 the now fully dressed Schwarzenegger leaves the bar to a rock-like tune. The ambient channels carry an extra large reverb that links back to the industrial feel of the drone. Some very nice foley work follows as the hero acquires the sun glasses.

The appearance of the T-1000, played by Robert Patrick, is also accompanied by a matching industrial soundscape, in which the backdrop of the score is composed out of effects. There is no "traditional" music for this robot.

The 1st scene with Edward Furlong as the young John Connor starts with a stereophonic spread that goes full depth when the camera approaches him in the garage. Moving back into the house limits the panorama, which widens again as the foster father tries to talk John into cleaning up his room.

The transition to the hospital is again very interesting. A bouncing reverb cuts to an enveloping ambience and then to the silence in Sarah Connor's cell. The editing intercuts quit scenes from within the cell with reverberant scenes of Dr. Silberman approaching. When he talks with her her breath can be heard on the rear channels.

Music comes back as two male nurses enter Sarah's cell, accompanied by a soft yet menacing electric buzz that has no visual origin as yet. After she has been forced to swallow a tranquilizer an acoustic effects cue, like the movement of a piston, kind of like a sonar probe, connects to the T-1000's last appearance. Then there was a much longer slap delay-like interval between the two beats, but now the robot is much closer to John's foster-home.

There is the drone from the T-1000's last scene, then the sole sound of the car approaching and a door opening expands into a peaceful urban soundscape, with a hint of thretening chords reappearing in the background. The once industrial soundscape assumes a more musical texture.

The cut to the boy's in action at an ATM reestablishes a by now known technique. The enveloping soundscape unfolds gradually at the scene changes. Contrasting to this the cut to the boy's on the way to "spend some money" jumps right into a full-depth fast-paced musical score. The soundtrack turns incidental during the cut to a street perpendicular to the passage of the motorcycle.

Then the rumble of the Terminator's motorcycle takes over. Industrial ambience has a brief appearance. The motorcycle leaves the screen but the rumble of it's passing in the distance prevails over the scene cut to Sarah's cell and merges into musical chords.

The dreamlike appearance of Kyle Reese is accompanied by a long musical backdrop, the end of which is signified by a piston-like sound overlaying the pads. Enveloping industrial rumble is opposed by the sound of naked feet running. A preview of Sarah's vision of the apocalypse follows, mainly in the frontal array, the creaks of playground-toys punctuae the strings of the soundtrack, a crescendoing chorus leads to the brief sound of destruction before a visually and sonically abrupt cut back to the cell.

This review covers the first 23 minutes of the (ultimate edition) DVD release.

Here is some information I researched on the web pertaining to Gary Rydstrom's involvement with the soundtrack on DVD:

Q: Typically, how involved do you get in the DVD release of a movie that you worked on theatrically?

Rydstrom:I've done a couple of DVD mixes. I did one for Terminator 2, and for Punch Drunk Love. I love to. It's a matter of time. Because there's things you want to do. There's things I want to do myself; I don't want someone else to do it. It's a different listening experience; I want to make the balance adjustments myself. Not as often as I'd like. [...] when I could do a DVD mix of a mix that I did I was very happy. There were always those things that I wish I'd done in the final mix anyway --do this little balance adjustment to make this cleaner, make this more punchy, make this more subwoofer.

Analysis - Primary Colors

2008-10-26T00:45:00.002+02:00

The movie Primary Colors by Mike Nichols features quite a few people in the sound-department, but no sound designer. The two re-recording mixers credited are Michael Barry and Ron Bochar.

Although the soundtrack on the DVD is said to be DD 5.1 it is a very sad listen. The ambience in the surrounds is only noticeable when accompanying large spaces, and there to the detriment of the overall impression. The dialog is clear and there is some music, but that is about the extent of the positive comments I can offer.

Analysis - Daredevil

2008-09-29T13:00:00.000+02:00

The movie Daredevil by Mark Steven Johnson features Steve Boeddeker from Skywalker Sound as sound designer.

Right from the beginning, with credits morphng ot of braille, the DVD presents an enfolding soundscape with some touches of LFE. Thunder, an exhaust mutating into a chord, 1 sec long cutting visual and sonic flashbacks, sirens, then helicopters circling.

Next you enter a church for some rest in the quiet resonance. A voiceover takes you back in time for the prologue, the ambience of which is defined by trains and dogs barking in the distance of hell's kitchen. Music is nicely interwoven.

The scene where Matt looses his eyesight is very well defined, leading to silence before his shocking discovery of his enhanced perception. He experiences a sensory overload when first confronted with the world of noise all around, finding rest only when the clang of chuch bells focuses his attention.

I like the idea of Matt resting in a double-walled, water-filled coffin that is sealed airtight. The incidental music he turns on in the morning to accompany the routine of his daily preparation - contrasting to his nightlife, to be revealed later - comes from a surround receiver. And well-crafted surround it is to be.

It is a foley-artists' dream, all sounds, big and especially small, being prominent to Matt and the viewer / listener. His is a world of enveloping and defining sound, and so is ours, at least while the movie lasts.

Well done!

Digital Microphones

2008-09-22T08:45:00.001+02:00

Disclaimer: At present I don't own a digital microphone and have not used one either. I have worked with the Nexus installation at the Konzerthaus in Berlin in 2004, and it was a very interesting experience.

How does a "digital microphone" work?

You take the output of the capsule and feed it directly to a high-resolution ADC [Analog Digital Converter] that is designed to optimally complement the capsule. Optional DSP processing allows for transient suppression, limiting etc, and since the standard also calls for information flow to the microphone, various parameters are remote-controllable, including the ability to change the characteristic of dual capsule systems.

Some history...

In 1996 Beyerdynamic premiered the MCD 100 based on an 24 bit Stage Tec TrueMatch ADC - with a true dynamic range of 22 bit. The Berliner Reichstag was equipped with 200 copies of a simpler version, the MCD 803, after which Beyerdynamic retired from that stage. The first microphone to implement the AES 42 standard was the Neumann D-01 in 2001.

An important concept in this regard is gain ranging: Every ADC has an optimal range for incoming signal-levels. In the case of Stage Tec's TrueMatch technology the signal is amplified by +-00, +20, +40 and +60 dB and fed each to it's own 28 bit ADC. A subsequent DSP-unit then picks the optimal representation and digitally scales it to match the continuous output.

Here is a list of companies currently manufacturing digital microphones (in alphabetical order):

Microtech Gefell MV 230 digital
- Measurement microphone

Neumann Solution-D
- Uses a two-tiered approach to optimizing the ADC-input
- Connected though the DMI-2 digital microphone interface

Oktava MK-012 USB preamp
- USP output only, no support for AES 42

Schoeps CMD 2U
- Puristic (non-DSP) approach
- No adaptive gain ranging
- No Mode 2 support for reasons of simplicity

Sennheiser MZD 8000
- Designed from the go to support 2 channels, according to AES 42

The DMC-842 by RME is an 8-channel interface and controller designed to work with all AES 42 compatible digital microphones.

Any concerns anyone?

I wonder if the capsules themselves are linear with regards to their frequency response. When you have a look at the companies manufacturing modular systems this should not be much of a problem as their preamp is designed to work with their selection of interchangeable capsules. With other manufacturers it could be an issue though.

I personally am very happy with my interfaces by Metric Halo.

They feature very well designed microphone preamps that I don't have any issue with sonically or handling-wise.
The adjacent ADC's are very transparent.
Metric Halo's 2d has had a great impact on my workflow, optimizing it in many respects.

I can not image myself lugging around a dedicated controller for digital microphones, but having a few that I can run into my interfaces' digital inputs is conceivable.

OTOH, have a look at Zaxcom's wireless digital systems...

It's not how much stuff you log around...

2008-09-15T14:45:00.000+02:00

A while back I arrived to record a choral concert with three choirs and >100 singers in one of Hamburg's churches. I carried my bag (a Crumpler "Very busy man") on my back, a mic stand bag with two stands and 6 lengths of 10m mic-cable over my shoulder. My bike was parked outside.

A collegue of mine was just tearing down. He had recorded a girl's choir in the morning and was now lugging three huge trunks into his van. He seemed a bit unnerved my appearance and asked me if I was doing a stereo recording. I said no. I would be setting up a six-channel recording. That's why I was so loaded...

On my website I advertise my enterprise as an "environmentally friendly studio (traveling by bike, train and bus)", and this is what I do for most events, the sole exception currently being my biannual trips to London, although I have meanwhile discovered a way to comfortably get there and back by train, which I will soon put into practice.

The reason it is possible to work lightweight is the availability of mature digital audio solutions. I am using very high end interfaces by Metric Halo as mic-pre's and AD's, I verify my headphone-based sonic evaluation by running SpectraFoo, and microphones don't take up much space. What's left to lug around?

Tip: You can save on mic-stands by using the K&M 240/5 microphone holder's whenever you can clamp a mic (or pair of mic's) to a rail or so. I also like using the DPA 4060 that can be employed as a boundary layer mic, taped to walls, easily flown by it's cable or placed as an inconspicuous spot.

In the end the measure of how much stuff you have to lug around depends on the type of music, the venue, the proposed use of the recording and the recordist's expertise. A good selection of gear, a lot of experience with your mobile setup and meticulous planning help you optimize your rig, minimize setup and tear down time and maximize the quality of your work. Go for it!

Surround sound on DVDs - Part 2

2008-09-08T11:15:00.000+02:00

Who is in charge of (re-)mixes for DVD?

As I detail in my analysis of the soundtrack of the Contact-DVD I was disappointed with the transfer and could not believe it had been overseen by the movie's sound designer, Randy Thom, who's work I have often enjoyed. I am glad to say it is possible that this process is not alway in the hands of the designer and / or re-recording mixer.

Gary Rydstrom states that "It's a different listening experience; I want to make the balance adjustments myself. Not as often as I'd like. [...] when I could do a DVD mix of a mix that I did I was very happy. [...] you're making balance adjustments based on a whole different standard listening level and environment." [IGN (2005): An Interview With Gary Rydstrom]

Music on DVD-V - Part 1

2008-08-29T11:24:00.004+02:00

I had an interesting talk with the owner of one of Hamburg's high end audio suppliers. He said flat out that Super Audio CD (SACD) and DVD Audio (DVD-A) were dead. He does have hope for Blu-ray as a carrier for stereo / multichannel losslessly compressed high-resolution audio.

Talking about the future, what's with the present?

Check out the prices for the manufacturing of small runs of DVD-5 as compared to CD-media. Using two online calculators (AltonaMedia and DiscOnDemand) I found an increase of about 50% in replication cost. And this is what you get with a DVD-5:

- You can store the equivalent of 5 CD-A's on one DVD-5 (4.3 GB vs. 800 MB)
- You can deliver high resolution audio: 24 bit as opposed to 16 bit, up to 192 kHz sample rate as opposed to 44.1 kHz
- e.g. 66 min of uncompressed PCM @ 24-192
- You can deliver surround encoded audio (Dolby Digital and / or dts)
- You can offer...
- Additional information about the music, ensemble, recording etc.
- Navigational menus
- Still images or video
-Links to (dynamically available) material on the internet

[2009-02-02] Discovered an interesting read regarding the advantages and disadvantages of the audio portion of a DVD-V for music .

I didn't yet find numbers comparing the amount of standalone and computer-based CD-players vs. DVD-V-players in use worldwide, but I assume that even in the latter case the installed base is sufficiently big to encompass a sizeable chunk of the audiophile / avantgarde / classical / jazz clientele, which for me would be fine. What do you think?

What I like about digital audio…

2008-08-24T09:22:00.009+02:00

Here is my comment on Jarome Matthew’s blog on Poor Quality Sound: Now Standard! in which Jarome lays down his dislike for the digital age of sound. While it is true that the new paradigm has created many ways of distorting audio it is not necessarily evil. I personally am definitely a proponent of digital audio.

For me digital audio technology (digital software technology?) in it’s current incantation does many good things:

- it is affordable, even considering the worlds best interfaces
- it gives me a very low noisefloor, which…
- let’s me dial in a 12 dB headroom to prevent accidental clipping
- it eliminates mechanical considerations
- it makes multitrack high end audio portable
- it does not necessarily induce coloration

True, a CD-A can only contain frequencies up to 22.05 kHz, but everything in that (audible) range is preserved. I doubt that _anything_ on vinyl can sound better than an optimal (!) 1:1 digital transfer of the same (!) source-material. AFAIK there is no reason for it.

Then again, using a SR of 192 kHz gives you a range up to 96 kHz Nyquist. Even a top of 48 kHz is sufficient in my experience. Analyzing the spectrum of a Galician Gaita can open up your eyes.

You can find a nice example of that instrument's sound: Jata da Mahia played by Javier Celada, recorded in Blumlein-configuration with two RoyerLabs SF-1's at Berlin's Heiligkreuzkirche in 2004. I recorded another concert featuring Gaita and Organ in 2004 using an AB-setup with two Earthworks QTC-1's - and trust me, you'll want to use a mic with a gentle HF-rolloff for these pipes :-)

Ich bin ein Ohrwerker

2008-08-04T12:31:00.005+02:00

A while ago I stopped at a coffeeshop in Berlin (DoubleEye) on the way to a recording. I had the usual large bag on my back, a long bag with mic stands and mic cables slung around my shoulder. I had a galao outside. A boy studied my appearance and asked me if I was an artisan. I answered that I was an "Ohrwerker", an artisan working with his ears.

For some reason this incident made me think about language. In German we have only the word "Handwerker" for someone who does manual labour. In English it is the artificer, artisan, craftsman, handy(crafts)man, workman, etc. Still the general idea is that of a person working with one's hands, which in my case is true, except for the fact that the focus is on what my ears and brain perceive of.

The job really is a mix of science, artistry and craftsmanship. You need to want to learn about every space you record in and the characteristics of the instruments as they appear in this environment. At the same time it's all about the music that is being performed, the intrinsic dynamics, the "color" of the sounds (Klangfarbe in German) and the flow of it all. And then there is the craft of capturing it well.

I love it :-)

How much should a recording cost?

2008-08-04T07:25:00.002+02:00

I like to present people with an overview of the prices I charge. Personally I favor businesses that do likewise over one's that say they'll have to make me an offer. It is clear that this constitutes a conservative estimate, taking some of the factors and dropping others, but it defines the "ballpark" of the service provider.

Which factors do I pick?

- The target mode of the recording: stereo or surround (5.0, 7.0)
- The number of microphones used as a measure of the complexity of the recording situation: more microphones likely means a longer setup time, more HD-space needed, more choices in post etc.
- The duration of the event to be tracked
- The potential use of the recording, which effects the sample rate (44.1 kHz for CD, 48 kHz for video, higher rates for probable EQ-ing in post)
- The projected workload in post: head & tail editing, tweaking of the preview-mix, real-time processing etc.
- The designation of the recording as non-profit or for-profit

Sound is my life! No matter what the parameters are, if I'm preparing for and executing a payed job or a pro bono venture, I approach every recording project with a clean slate. My goal is to craft a recording that gives the listener the impression of being there, which is to say it must sound at least as good as it sounded live.

Why analyze movie's surround soundtracks?

2008-08-04T01:35:00.000+02:00

It's quite simple: I enjoy watching and listening to good, well crafted movies. And if I review them to learn what I like about the way in which the surround panorama has been realized I might as well take down some notes. And if I take down some notes I might as well post them, as other people might be interested in my thoughts - and possibly even submit some feedback.

I'd like to emphasize that last point: Posting a critical analysis into the void is of value only if said analysis is complete, which I don't assume it will be, or if it can be refined with the help and the input of other people who read it, possibly verify and / or falsify my observations, and submit their assessment to the general web populace. Please do tell me what you think. Thanks!

If a movie ends up being talked about in a negative way it's because an in-depth review of a feature that I enjoyed watching, and which I believe to have a well-crafted soundtrack, turned out to exhibit a serious flaw. I start out liking it, which is why I pick it in the first place, but it turns out I didn't listen closely enough at the start. Which is a good thing as a good story lies at the root of every good film, and that story should carry you painlessly across any inadequacies - unless you pause, dig deep and reconsider.

Analysis - Contact

2008-07-29T14:31:00.013+02:00

Contact by Robert Zemeckis features Skywalker Sound's Randy Thom as Sound Designer and re-recording mixer.

The DVD is said to contain a 5.1 mix. The enveloping nature of the surround soundfield is used well during the zoom away from earth and into the galaxy at the beginning of the movie. Afterwards there is a more monophonic feel, perhaps as a contrast to the sonic caleidoscope. Still, I feel the stage should widen again when there is a definite purpose.

The party scene near Arecibo relies heavily on the frontal image as does the following sequence, in which the jungle noises sadly don't originate from the surrounds. It is interesting to note that ambience is located in the L and R but also perceptively in the center channel. The dialog on the other hand is _only_ placed in the center channel, with no trace in the other speakers - at least at that point.

The music takes up the L+R channels. The effects and therefore also the environmental envelopment are neglected.

19'30 into the movie the surrounds are once again activated. The (overly long) reverb from young Ellie's shouts is located in the back - with a rise towards the fading out. She then goes on to shout without any reverb. Perhaps this application is supposed to signify the transition into the (tunnel of) past? When Ellie enters the house the music is doubled in the surround channels. It carries on to smooth the transition to the creaking of threes in the wind in the frontal channels.

When Ellie walks up the stairs at her father's funeral (22'40) there is an increase in hiss in the surrounds, especially the left one, that then abates. I take this as yet another sign that the remix to Dolby Digital 5.1 for DVD was not executed carefully and decide to skip further into the movie, to the point when Ellie travels to Hokkaido (01'37).

The ambient channels are used to illustrate the flight of an airplane from rear right to frontal left, but the fury of the storm on the water takes place only in the front. This is yet another space where surround would very much enhance the illusion of "being there". Also, when the camera tracks around Ellie as she steps into the rain would be a great moment to move the weight of the rain to the rear.

Inside the ship there is the pulsing of the engines and rain hitting the window. None of it finds it's way into the ambient channels. I am happy to say that at least when Ellie enters the machine (01'42) the surround channels are used. When she talks with mission control the rear channels carry some reverb at very low, nearly imperceptible levels. This is mixed in with the dry music track.

At 01'46 mission control talks about some low frequency noise. Too bad it did not find it's way into the mix... The rear channels are used during Ellie's trip through the wormholes (01'50) and in the silence they carry the stadium-like reverb of Ellie's voice. The second episode of the travel features only occasional effects in the rear.

When Ellie arrives at the beach (01'55) the sound of the sea is conspicuously absent from the rear channels. They do carry the reverb tail of her breathing and at 01'57 the very low volume reverb on Ted's and Ellie's voices. Still I wish for a better sense of envelopment which definitely is possible.

It must be said that this rendition of the movie would benefit immensely by a careful acoustic design, as I am certain the movie in it's realization for the big screen contains. It remains unclear why the DVD-release sounds this poor.

The comments above pertain to the Dolby Digital 5.1 soundtrack of the German special edition (p) Jonathan Gaines / Acoustic Visions [Sorry, I didn't find any links].

Analysis - Ghost Dog

2008-07-24T14:42:00.009+02:00

Ghost Dog by Jim Jarmush features Anthony J. (Chic) Ciccolini III as Sound Designer and Sound Effects Editor.

The DVD contains a very nice 5.1 mix in which the environment is well represented. The frontal soundstage is fully developed. Effects, like the last remaining pigeon that appears in the middle of the film to fly around the audience, are effective. The - often incidental - soundtrack by RZA with a strong bass is powerful and drives the action. The reverb of the monophonic mix is placed mainly in the surround channels.

It is noteworthy that the passages where Ghost Dog (Forest Whitaker) recites from Yamamoto Tsunetomo's "Hagakure" are not only placed in the Center but also in the surround channels. This might serve to create a stronger link to the interior monologue.

I was a bit surprised to experience dense white noise when I turned up the surrounds experimentally once in a while. It's volume is modulated by the overall mix, primarily from the front channels. I don't have an explanation for this observation as yet.

Surround sound on DVDs - Part 1

2008-07-24T08:33:00.003+02:00

Since I set up my 5.1 monitoring system two months ago I have evaluated the surround soundtracks of several movies on DVD. BTW, it is surprising how many releases feature only 2.0, as movies have been produced with at least one rear channel for decades; maybe one can be glad that the consumer is not expected to own a monophonic system… In any case, I found most productions claiming to feature Dolby Digital 5.1 to be severely lacking in enveloping ambience.

What does 5.1 provide me with? I talked about that in an earlier blog, but to summarize briefly:

- Enveloping ambience
- Stability of the central image - as opposed to a phantom
- LFE extension, primarily for effects

With many of my recordings of acoustic music, tracked with surround in mind but only now remixed to 5.1, I feel that the sense of "being there" is the most prominent feature. With some symphonic works the LFE is absolutely essential to transfer the excitement of live timpanies or a bass section let loose, but being enveloped by the natural ambience for me is the key.

So why are many movies, that after all represent _major_ undertakings, focusing and condensing the passion of many contributors and generally consuming considerable amounts of time & money, lacking in that respect?

I believe it may be due to the nature of the process:

- The location recordist is primarily concentrating on capturing the narrative
- Ambience recording on set is a compromise as:
- Picture comes before sound
- There is a lot (of hustle and noise) going on around the scene being filmed
- There are many takes that feature slightly disparate acoustic angles

Speaking of angles, how do the final spectators / auditioners of the movie react to ambience cuts, in sync with visual edits, or is a (recreated) continuous ambience used to smooth out visual breaks?

- It is less costly to overdub than to reshoot after primary shooting is done
- In many cases the creation of sound effects and of the musical score & soundtrack are independent
- Sound is assembled when the final edit is done

In the end it is likely that the sonic realization of a DVD will not be more refined than the one created for the big screen.

Disclaimer: all this is my personal view as a sound person not directly involved in the movie industry. As such I represent the informed audiophile consumer's perspective. I welcome any feedback and corrections from collegues more intimately involved.

The Zen of Recording in real spaces

2008-06-15T09:25:00.003+02:00

Arrive in time -
Unload your gear, preferably before the space -
Empty your mind of your thoughts on how to setup for this gig -
Breathe -
Walk into the space slowly and take in the layout with your eyes -
Close your eyes as you step into the acoustic realm -
Breathe -
Listen carefully for the sound of your breath, your steps, sounds from without, the ambience within -
You are ready to go.

In 1971 Michael Gerzon observed: "The rule here is so astonishing and difficult that many engineers seem to think that balancing up twenty or so mikes is a lot quicker. The rule is to walk around during a rehearsal or first run-through, find out what position makes the music sound best live, and then place the microphones at that precise point at ear height!"
[Why Coincident Microphones? In: Studio Sound Vol. 13, pp. 117, 119, 140; from www.michaelgerzonphotos.org.uk ]

Why bother with accuracy?

2008-05-01T06:47:00.014+02:00

I like the DPA 4060 miniature microphone a lot and take my stereo pair to nearly every gig. Although I have not used any other of their microphones I respect the company and it's reputation. I was not able to determine who authored the DPA-branded booklet (with accompanying SACD) I will be referring to. The German copy e.g. is (c) Mega Audio, which to me seems to be mainly a distributer for audio equipment. BTW, you can receive a copy of the SACD with accompanying booklet free of charge from DPA's local distributor.

So, what's up with "Miking a Grand Piano"?

The picture on the front page shows a strange sight. There is a nice looking grand in a fantastic hall, but the microphone pair, positioned right at the piano, seems to be aligned rather haphazardously. I assume they are cardioids since they are not mounted in parallel, on the other the hand they are quite far apart and angled sideways only slightly.

The booklet states that it can only provide guidelines, not recipes, but these guidelines should make sense! Talking about positioning an AB pair 1 to 2.5 m from the piano is one thing, showing a picture where the stand is about 30 cm from the lid of the grand is another, and all examples on the SACD are very close miked. Secondly, I don't believe aligning omnidirectional microphones in parallel to the fully open lid does not negate the acoustical influence of the lid.

Next comes ORTF, where the microphone pair is shown positioned as close as makes sense for an XY pair @ +-90°. But considering the much smaller stereophonic recording angle of the former, 96° as opposed to 190°, and the fact that modifying the panning of left and right source from full left & right will induce frequency-dependant phase-irregularities, this location should not be one's first choice.

A few pages later there is mention of an "oversized ORTF-configuration": Two cardioids at a distance of 40 cm and at an angle of 70°. There is no such thing! ORTF is defined as a distance of 17 cm from capsule to capsule and an angle of +-55° from the central axis. Anything else should be described by reference to a different framework, e.g. in the context of Williams' Stereophonic Zoom, according to which the Stereophonic Recording Angle of the above mentioned system would amount to 80°. IMHO this is far too shallow an angle to be located "over or just ahead of the curve of the piano".

And that pretty much sums it up with regards to the needs of the recording engineer.

To be fair, there is brief mention of utilizing a far AB setup "for recordings in large environments with favorable acoustics", which luckily goes for most places in which a grand is featured for a live performance. The positioning suggested is 2-3 m distant and at a height of 3-4 m. There are no samples on the SACD featuring a similar configuration, which is sad since Radio Denmark's concert hall is likely to be a formidable acoustic venue.

So what is missing?

- What does a grand sound like to the audience?
- What are the acoustic effects of the position of the lid?
- In what way does the grand project?
- At what distance should a distant AB pair be positioned relative to the piano?
- What distance should be between the microphones mounted in parallel?
- At what height should the setup be located? I personally stay in the range from between the edge of the grand and the height of the lid.
- What about positioning microphones under the piano? I used an XY pair once to good effect, boundary layer style miking on other occasions.

And on and on…

So, how to mike a grand?

Currently, on many occasions, I use an AB setup with two accurate omnidirectional microphones at 51.5 cm, placed 1.5 to 2 m from the piano, augmented by two parallel ribbon spots suspended within the grand, not too close to the hammers, and angled so the vertical 0°-axis is parallel to the lid.
Using two microphones holders that can be carefully clamped to the piano (e.g. K&M's 240/5) saves me from lugging along two extra stands with booms.
In quiet, reverberant environments I will probably also place two omnidirectional microphones farther back and far left and right to have a return on two channels of uncorrelated natural ambience and as an aide in rebalancing the image of the sound stage. A time-delay measurement allows me to align all six tracks in post.

Move it, baby! Or: Mike it in stereo, dear recordist.

2008-04-18T09:06:00.008+02:00

Compare many studio recordings with well recorded, minimalistically miked live performances and you will notice that the latter often contain more "space", and movement within that space, making them sound even more live. Why is that so?

- When music resonates in a space the intensity of the sound at any one spot changes over time.
- In real life most musicians move while they are playing.
- When musicians move, their instruments move with them, with the obvious exception of musicians performing on a fixed instrument or set: pianists, percussionists, organ players come to my mind.
- When an instrument moves in space it continuously projects (different frequencies) in different directions.
- This movement is as much part of the musical performance as the notes played, the texture of the sound, the dynamics and timing etc.

Set up a single microphone to mike a musician's instrument and it will record only varying degrees of sound pressure as the musician moves closer and farther, on and off axis. Most of the acoustic effects of any movement in space will be lost.

What to do?

- Recording in a good sounding acoustic environment.
- Position the ensemble so that you can capture a balanced sound stage with a main stereo microphone.
- Use (few) stereo pairs for spot miking.

Then the only thing left to do is for the musicans to move it :-)

Don't always go for 100%

2008-04-09T11:00:00.009+02:00

I'm just now listening to a stack of so called audiophile CDs I borrowed from a friend and notice that on quite a few titles the entire width of the sound stage is used in a way that places direct sources full left or right. I don't think that is a good idea - in general.

The way I perceive it a sound stage that leaves some room left and right creates a more realistic space. All sound having a phantom location and being localized from the area circumscribed by the set of loudspeakers can never be as well defined as sound that originates from one of the speakers only.

- Use the appropriate configuration. XY at +- 90° and AB at about 50 cm cover an area of approximately 180°, ORTF 96° and Blumlein 76° (see an overview for more detail).

- When tracking, the main stereo setup should be placed so that (a little) more width than the original sound stage is covered.

- Coincident configurations allow you to safely(1) manipulate the width of the recorded stage in post.

- In many cases additionally tracking a flanking pair of parallel omnidirectional microphones can help restore the balance.

- Regarding the positioning of musicians it is important to place them linear with regards to the distance between one another rather than linear with regards to the angle as seen from the main microphone setup.

- When panning sources to create an artificial space, the difference in level or in sound pressure (16 to 20 dB) or the delay of one side relative to the other (1 to 2 msec), or a combination of both factors, should be chosen to not be "extreme".

This is my opinion, so YMMV. Let your ears be the final arbiter.

(1) You don't experience (as much) comb filtering due to phase differences in the signals being combined.

11 steps to a fine latte

2008-03-31T11:19:00.006+02:00

01. Visit a local roaster and pick a flavorful, balanced espresso blend.
03. Buy whole beans that have been roasted two to three days ago, or buy freshly roasted ones and let them sit a bit.
04. Don't take home more beans than you can use up in up two weeks.
05. Have the machine fully heated up.
06. Steam the milk before pulling the espresso - which is an art in itself.
07. Use a high quality burr grinder(1).
08. Grind the right amount of beans at the right setting fresh into the heated portafilter's basket(2).
09. Tamp with enough pressure - around 15 kg.
10. Pull your shot while counting down the seconds (25 to 30 sec for a 30 ml single or 60 ml double shot)
11. Use the knock box, clean the group head and quick rinse the portafilter.

Enjoy :-)

(1) Invest as much into the grinder as into your (semi automatic) coffee machine.
(2) Watch a good barista as he experiments with the grind setting to optimize his shots.

[Many thanks to Mark Prince, the CoffeeGeek (http://coffeegeek.com/) for his wisdom]

[Expanded on in Making a better latte]

ORTF off axis

2008-03-29T17:26:00.003+01:00

Yesterday I tracked a choral concert in the Catholic Church St. Sophien in Hamburg (http://www.sankt-sophien.de/). Here are some initial observations about the situation:

- The church is highly reverberant.
- The rows of benches are placed on a raised wooden pedestal that is sensitive to shuffling and kicking.
- There were four children's choirs ranging from about 30 to 80 singers performing in differing configurations.
- The altar area was populated asymmetrically as there is a big pulpit to the right.

The smaller choirs I saw rehearse were located more to the left, so I placed the ORTF-setup (two cardioid Oktava MK-102's modified by Attila Czirja'k) about two meters high, pointing at the middle of the choir when the pedestals were used, near the left hand row of benches and about five meters from the front line.

Why five meters? I assumed that the larger setups would take up about eight meters from left to right, but I couldn't be sure they wouldn't spread out a bit more. According to Eberhard Sengpiel (http://www.sengpielaudio.com/AufnahmebereichWichtigeWerte.pdf) ORTF covers an angle of 96°, so your distance from the source has to be about 53% of the width of the sound stage. Then there was a grand piano (with closed lid) close to the left hand side of the choir. I wanted to be a bit more distant from that instrument to not let it sound too prominent.

Since I wanted to have the option to increase the natural reverberance and potentially producing a quadraphonic mix I taped two ambient pickups (DPA 4060) to the walls of the doorways left and right of the altar, in back of the choir podiums. At this location audience noise is minimal.

During the rehearsals I noticed that the raised row of singers of the larger ensembles did fill the space behind the pulpit. Some solos were also performed on the right hand side, and in one song (what I believe to have been) a Conga was also placed there. I decided to turn the ORTF-setup to face a little bit to the right of it's originally central focus.

So what?

Direct an ORTF-config off axis only when considering the consequences ;-) You don't move the center of the sound stage in the direction you are turning towards but increase the effect of sound pressure reaching one side of the system while decreasing it for the other side.

Luckily the problem was solvable for my recording. I was able to reduce the left hand side by 3 dB and added 3 dB more of the natural reverb to the right hand side. The balance was restored without compromising the overall quality of the sound stage. Still, when the first sopranoes are placed on the left hand side it is more prominent to start with.

What could have also been a solution?

- I could have tracked closer to the ensembles with an AB-setup. That way the vertical localization would not have been as clearly defined. OTOH I would have picked up more of the reverberance and audience noise.
- I could have used flanking omnidirectional microphones to the left and right of the sound stage, perhaps at the closest columns, to be able to manipulate the balance.

In the end it is highly advisable to have a say regarding the placement of the ensemble to be recorded. If it is symmetrically arranged things are much easier. And if the vocals are balanced in themselves, with (especially) the sopranoes singing in a controlled way, and an adequate presence of lower voices, one approaches optimal conditions.

What's good and what's bad about Stereo?

2008-03-28T09:29:00.006+01:00

I enjoy tracking musical events stereophonically and I enjoy listening to well recorded stereophonic recordings. Still, there are certain issues with this approach, over which I will reflect. I'm trying to keep these thoughts simple yet accurate.

1. How to record in stereo?

Even though Eberhard Sengpiel pulls our nose by stating that two times mono is also stereo (http://www.sengpielaudio.com/Unterlagen09.htm ;-) what I am talking about is the art of using an integral system comprised of two microphones to recreate a soundstage for the listener.

The sound that microphone A transforms, thereby translating acoustical into electrical impulses, is mapped to the left loudspeaker, the sound microphone B transforms is mapped to the right loudspeaker.

There might be additional microphones used to

- help set right inadequacies in room acoustics - or microphone placement - capture nuances from various sound sources that would not be discernible otherwise - capture uncorrelated ambience from disjoint positions

[Uncorrelated ambience: sound transduced by two or more microphones, placed not in the direct but in the reverberant sound field, does not change significantly when mixed together. Even though the spectral (frequency-related) content is probably similar there is enough difference in phase to make the various sources dissimilar]

Mixing some of the information from the spot-microphones with the sound obtained through the primary setup (Hauptmikrofon), compensating for the runtime-difference, because the sound arrives earlier at closer microphones than at those farther distant, can increase the perceived clarity of the recording. The uncorrelated ambience can be used to make the stereophonic recording sound more spacious.

2. The advantages of stereophonic recordings

A straightforward assumption is that the less microphones the recordist uses the clearer the resultant sound will be.

There are so many reflections in any acoustic space that, even when carefully planning the setup and working primarily with symmetry axis' and equal distances for the placement of spot microphones, any microphone captures a slightly disjoint representation of all that is going on sonically. Obviously there are some aspects which should be perceived differently at each microphone position, but there are aspects that should be identical or similar, and these are converted with differences in spectral and phase composition.

Well executed stereophonic recordings are able to recreate a soundstage that lets the listener perceive width and depth when played back on only two loudspeakers.

3. The disadvantages of stereophonic recordings

- The vertical and horizontal placement of sound sources, as well as phantom images that appear to be located in the middle of the two loudspeakers, are only accurately perceivable by a listener who is located at the right position relative to the setup.

- The listener is not enveloped by the ambience of the recording space.

- A loudspeaker built to reproduce the full spectrum of audible frequencies is difficult to construct. There are many design issues involved, including the construction of inadvertently phase- manipulating crossover networks, the projective abilities of the loudspeaker at different frequencies, internal resonance etc.

- The farther the loudspeakers are placed from the listener the more the quality of the sound being played back is dependant on the sonic characteristics of the listening space, the one aspect the loudspeaker- developer can't get close to solving - unless he creates a near-field system.

4. Surround sound

- LCR (Left Center Right) setups create a real sound source in the middle of the loudspeakers placed left and right of the listener's position in place of the finnicky phantom image.

- Quadraphonic setups enable the creation of enveloping ambience.

- Having a dedicated, centrally placed subwoofer literally takes (a lot of) the load off the other speakers that don't have to create large amounts of sound pressure by moving large amounts of air.

In my opinion 5.1 Surround is a good mix of the solutions to create an even better listening experience. Certainly 7.1 is even neater, as you are able to create three stereophonic zones before the listener combined with enveloping ambience from the back, but in my opinion 5.1 not only will do for quite some time but there is also the aspect of the quality of the components used - and the issue of optimizing the speaker's placement. Go for a good 5.1 system and enjoy :-)

Starting up - again

2008-03-28T08:21:00.003+01:00

I set up a blog in 2006, diddled a bit in 2007, but just was not up to the challenge. You have to be persistent, draft ideas as you go along, publish regularly etc. I did manage to create two podCast-episodes, but basically they talked about the stuff that would be forthcoming, which then didn't. Let's see if I'm off to a better start in 2008 :-)