Last Update 10/21/08
New material, November 2008
My long association with Harman has come to an end as of November 21, 2008 The email address thus must change again. See the email address embedded in the picture.
A Finish student requested that I add the lecture slides for a talk I gave in October 2008 to the Finish section of the Audio Engineering Society. The subject was "Recording the Verdi Requiem in Surround and High Definition Video." These slides can be found with this link. However I notice that I had previously put most of these slides on this site - together with some of the audio examples you can click to hear. This version can be found below in the link to the talks give to the Tonmeister conference in 2006.
Last September I gave a paper at the ICA2007 in Madrid, which presented data from a series of experiments that I had hoped would lend some clarity to the question of why some concert halls sound quite different from others, in spite of having very similar measured values of RT, EDT, C80 and the like. A related question was how can we detect the azimuth of different sound sources when the direct sound - which must carry the azimuth information - is such a small percentage of the total sound energy in most seats?
The experiments first calculated the direct/reverberant ratio for different seats in a model hall (with sanity checks from real halls) and then looked for the thresholds for azimuth detection as a function of d/r. Some very interesting results emerged. It seems that the delay time between the arrival of the direct sound and the time that the energy in the early reflections had built up to a level sufficient to mask the direct sound was absolutely critical to our ability to obtain both azimuth and distance information. In retrospect this point is obvious. Of course the brain needs time to separately perceive the direct sound and the information it contains - else this information cannot be perceived.
The necessity of this time delay has consequences. It means that as the size of a hall decreases the direct to reverberant ratio over all the seats must increase if sounds are to be correctly localized, and the clarity that comes with this localization is to be maintained. To achieve this goal, it is desirable in small halls that the audience sits on average closer to the musicians, and the volume needed to provide a longer reverberation time be obtained through a high ceiling, and not through a long, rectangular, hall. Halls less than 1500 seats should probably look more like opera houses with unusually high ceilings. They should not be shoebox in shape. Fortunately, Boston has such a hall: Jordan Hall at New England Conservatory (1200 seats) is the best hall I know of worldwide for chamber music or small orchestral concerts. It has a near opera house shape with a single high balcony, and a very high ceiling. The sound is clear and precise in nearly every seat, with a wonderful sense of surrounding reverberaton. Musicians love it, as does the local audience.
Alas, the paper went over like a lead balloon. It was way too dry, and it seemed no one understood how it could possibly be relevant to hall design. So I quickly re-did the paper I was to present in Seville the next week to answer the many questions I received after the one in Madrid. The powerpoints for this presentation are here: "Why do concert halls sound different – and how can we design them to sound better?" Hopefully this gets the ideas across better.
At the request of a reader I have put the sound files associated with the lecture at the RADIS conference which followed ICA2004. "The sound files for RADIS2004" These include several (pinna-less) binaural recordings from opera houses, and both music and speech convolved with impulse responses published by Beranek.
David Griesinger is a physicist who works in the field of sound and music. Starting in his undergraduate years at Harvard he worked as a recording engineer, through which he learned of the tremendous importance of room acoustics in recording technique. After finishing his PhD in Physics (the Mösbauer Effect in Zinc 67) he developed one of the first digital reverberation devices. The product eventually became the Lexicon 224 reverberator. Since then David has been the principle scientist at Lexicon, and is chiefly responsible for the algorithm design that goes into their reverberation and surround sound products. He also has conducted research into the perception and measurement of the acoustical properties of concert halls and opera houses, and is the designer of the LARES reverberation enhancement system.
The purpose of this site is to share some of my publications and lectures. Most of the material on the site was written under great time pressure. The papers were intended as preprints for an aural presentation. Some of them are available as published preprints from the Audio Engineering Society. The papers should be considered drafts - they have not been peer reviewed.
Several power point presentations have also been added to the site. These presentations are often quite readable and informative. In general they are more coherent than the preprint of the same material, although they are naturally not as detailed. They can be read in conjunction with the preprint of a talk. These are available as .pdf downloads, although your browser may allow you to open them with Adobe Acrobat.
Some recent work includes:
"Perception of Concert Hall Acoustics in seats where the reflected energy is stronger than the direct energy - or… Why do Concert Halls sound different - and how can we design them to sound better? - A poster paper presented at the AES conference in Vienna, May 7, 2007. "Two powerpoints given at the Tonmeistertagung Nov. 2006, and the AES convention October 2006" The subjects are "Distance Effect and Muddiness" and "Recording the Verdi Requiem in Surround Sound and High-Definition Video" These lectures are presented in a .zip file so the audio examples can be included. It is recommended you unzip the package in a single directory, and then view the slides with powerpoint. The audio examles should work when clicked.
Recent work on headphone calibration [October 2008] - done partly with the students of Ville Pulkki at Helsinki University - has shown that all the types of headphones we tested have large variations in frequency response for different individuals AND that these differences make for very different perceptions of the sound quality of binaural recordings from halls. It is possible to compensate for these errors by matching headphones to listeners through noise band loudness matching. Thus before listening to the music examples in the above papers using headphones or typical computer speakers, please read the following: "The necessity of headphone equalization" This link has been substantially re-written. Simple loudness matching is not sufficient. A reference loudspeaker is required - and loudness differences must be compensated.
"Pitch Coherence as a Measure of Apparent Distance and Sound Quality in Performance Spaces" This is a preprint of a poster paper presented at the Institut of Acoustics conference in Copenhagen, May 2006. The preprint contains some audio examples, that can be heard by clicking on the links.
An effort to clean house resulted in scanning a few old papers: "Reducing Distortion in Analog Tape Recorders" JAES March 1975, "The Mossbauer Effect in Zn67" Phys Rev B vol 15 #7 p 3291 April 1 1977, and "Spaciousness and Localization in Listening Rooms - How to make Coincident Recordings Sound as Spacious as Spaced Microphone Arrays" JAES v34 #4 p255-268 April, 1986. I still find the paper on spaciousness to be interesting and insightful. The observations continue to be relevant to my work, particularly on the subject of the intereaction between loudspeakers and rooms at low frequencies. It took many years before I was able to explain the observed effects. For example, Dutton's work on the apparent localization errors with conventional stereo loudspeakers at high frequencies are fully explained in "Stereo and Surround Panning in Practice" The effects of low frequency room modes on spatial properties are fully explained in “Loudspeaker and listener positions for optimal low-frequency spatial reproduction in listening rooms”
The early paper on spaciousness and localization is much too optomistic about the possiblity of increasing the spaciousness of a listening room through increasing the low frequency separation. In most rooms where the low frequency modes do not correctly overlap the low frequency separation is inaudible. Increasing it only stresses the loudspeakers. The best solution is to drastically change the loudspeaker positions, or change the room dimensions.
New in October of 2005 are the slides from a lecture on recording technique,
given in Japan for the Audio Engineering Society, and in Schloss Hohenkammer for
the Tonmeister conference in October. I have also finally added "Griesinger's Coincident Microphone Primer", a paper
from 1987 that describes and mathematically analyzes much of the behavior of concident microphone arrays,
including the Soundfield microphone. New in May of 2005 is a paper on room dimensions and loudspeaker placement
for the reproduction of envelopment at low frequencies, presented at the
acoustical society meeting in There are more papers listed at the bottom of
the page... don't give up here! Bibliography (somewhat out of date as of 5/05): "Griesinger's Coincident Microphone
Primer" Oct. 1985 - available from the Author. "Spaciousness and Localization in Listening
Rooms and their Effects on the Recording Technique" JAES v34 #4 p255-268
April, 1986 "New Perspectives on Coincident Microphone
Arrays" presented at the 82nd convention of the AES preprint 2464 "Neue
Perspectiven fur koinzidente und quasikoinzidente Verfahren" Bericht 14.
Tonmeistertagung Munchen 1988 "Verbesserung
der Lautsprecherkompatibilitat von Kunstkopfaufnahmen durch herkommliche und
raumliche Entzerrung" Bericht 15. Tonmeistertagung Mainz 1988 "Practical Processors and Programs for Digital
Reverberation" proceedings of the AES 7th International Conference, Equalization and Spatial Equalization of Dummy-Head
Recordings for Loudspeaker Reproduction" JAES 37 #1/2 1989 p20-28 "Theory and Design of a Digital Audio
Processor for Home Use" ibid. p 20-29 "Binaural Techniques for Music
reproduction" Proceedings of the 8th international conference of the AES
1990 p 197-207 "Study of Acoustical Enhancement Systems,
leading to the use of time variant synthetic reverberation" ASA meeting PA
Mar. 1990 "Improving Room Acoustics through time variant
synthetic reverberation" AES convention "Room Impression Reverberance and Warmth in
Rooms and Halls" presented at the 93rd AES convention in "Measures of Spatial Impression and
Reverberance based on the Physiology of Human Hearing" Proceedings of the
11th International AES Conference May 1992 p114-145 "IALF - Binaural Measures of Spatial
Impression and Running Reverberance" Presented at the 92nd convention of
the AES March 1992, preprint #3292 "Analysis of Room Impulse Responses based on
Perception" 5/14/93 - available from the author "Quantifying Musical Acoustics through
Audibility" Knudsen Memorial Lecture, Denver ASA meeting, Nov. 1993 "Subjective Loudness of Running Reverberation in
Halls and Stages" Proceedings of the Sabine Memorial Conference MIT, June
1994 - available from the Acoustical Society of America "Progress in Electronically Variable
Acoustics" ibid. "Reverberation Level Matching
Experiments" W. Gardner and D Griesinger ibid. "Wie Laut ist mein Nachhall?" -
proceedings of the Tonmeister Convention, "Further Investigation into the Loudness of
Running Reverberation" proceedings of the "Optimum reverberant level in halls"
International Conference on Acoustics, "Feedback reduction and acoustic enhancement
in a cost effective digital sound processor" International Conference on
Acoustics, "Design and performance of multichannel time
variant reverberation enhancement systems" The proceedings of the 1995
International Symposium on Active Control of Sound and Vibration, Patents: "Sound Reproduction" - A directionality
enhancement system for converting encoded stereo signals into four output
channels #4,862,502 7/29/1989 "Sound Reproduction" - A directionality
enhancement system for converting encoded stereo signals into 6 or 7 output
channels #5,136,650 1992 "Electroacoustic System" - A system of
microphones and loudspeakers in conjunction with computer based electronics for
altering and improving the acoustics of spaces #5,109,419 4/28/1990 "A Spatial Impression Meter" 1993 David Griesinger is a physicist interested in sound
- the sound of music. He is particularly interested in translating subjective
impressions of sounds into the physics of sound propagation, and the
psychoacoustics of sound perception. He has found that although it is wonderful
to discover ways to improve the quality of a reproduced sound, it is far more
useful and powerful to understand exactly how the improvement was achieved. This interest started with work as a recording
engineer. Through college and graduate school I recorded concerts and made records
for student organizations. The need for better microphones led to work in
microphone design and construction, Starting in 1964 with the construction of
omnidirectional condenser microphones. In about 1985 I designed and constructed
a miniature Soundfield microphone (16mm diameter), and in about 1990 made a
dummy head microphone for classical recordings. Most of the work on microphones
has not been described in publications, but a paper did appear in the Journal
of the Audio Engineering Society on the equalization of dummy head microphones.
This paper is also available in German from the Deutsche Tonmeister Verband. "Verbesserung der
Lautsprechercompatibilität von Kunstkopfaufnahmen durch herkömliche und
räumliche Entzerrung" Bericht der 15. Tonmeistertagung, Early work in this field produced a paper in the
Audio Engineering Society journal on distortion reduction in magnetic tape
recorders, and a paper on image localization (as a function of frequency) from
two channel sound equipment in small rooms. Griesinger, D. "Spaciousness
and Localization in Listening Rooms - How to make Coincident
Recordings Sound as Spacious as Spaced Microphone Arrays" JAES v34
#4 p255-268 April, 1986 This paper is still interesting to me, although it
took more than 20 years for me to develop the knowledge and techniques to
predict the results from first principles. The work as a recording engineer also led to an
abiding interest in artificial reverberation, and this eventually resulted in
the development of the Lexicon digital reverberation devices. Alas, due to
problems with trade secrets this work remains unpublished. About in 1990 I started installing reverberation
units in spaces used for musical performances, in an effort to improve the acoustics
for live performances. This work led eventually to the development of the LARES
system for acoustic enhancement. This work is described in the paper "Improving Halls and Rooms with Multiple Time Variant
Reverberation" which is on this site. Unfortunately this paper is not
yet available to me with machine readable drawings, and is presented here
without them. Perhaps eventually we will have the complete paper. I still
consider this paper a classic - although the precise method of randomizing the
reverberation devices is deliberately not described (sorry... you have to draw
the line somewhere.) LARES works wonderfully well - but I learned quite
quickly that conventional acoustical measurement techniques were useless for describing
its performance. The glaring mismatch between what you could easily hear in a
hall and the measurements one could make resulted in a serious study into the
perception of acoustics. A flurry of papers resulted - all more or less wrong. I also did considerable work on technical methods
of room measurement. At least two interesting papers resulted - see the 1992
paper " Impulse response measurements using
All-Pass deconvolution and the later paper on occupied hall measurement. Beyond MLS - Occupied Hall Measurement With FFT Techniques I
am actually quite proud of both papers. The all pass deconvolution method is
amazingly clever and efficient. You simply play this strange time-reversed
signal into the room, and play the result through a simple reverberator.
Instant impulses result - quite amazing. The sweep method is actually much more
effective, but far less clever. Conventional measures were clearly missing the
point - but for a long time, so was I. About ten years ago this work started to
converge into a coherent (at least I think it is coherent) hypothesis about how
we perceive the acoustics of enclosed spaces. Griesinger, D. It turns out that acoustic perception relies on two
very different phenomena. The most basic is the detection of reflected energy
by the hearing system. This detection relies on fluctuations in the Interaural
Time Delay (ITD) and the Interaural Intensity Difference (IID). The
fluctuations are caused by interference between the direct sound from a source,
and delayed reflected sound. The creation of fluctuations is a physical process
- it can be easily modeled and predicted. The other piece of the puzzle takes
place much later in the neural process, and is related to the process of
separating incoming sound events into related streams of information, such as
the syllables of speech from a single person. It turns out there is neurology
for this separation process. This neurology organizes sound events into one or
more foreground streams. But there is also neurology that keeps track of the
loudness and the sound direction of background sound in the spaces between
sound events. Our perception of the background also forms a stream - but this
one is perceived as continuous, and has specific spatial properties. The
neurology associated with the background stream is the primary source of our
perception of musical envelopment, and so the spaces between musical notes are
vital to this perception. The separation of the background stream from the
foreground stream takes time. Reflected energy that arrives too soon after the
end of a note is perceived as part of the note itself, and does not contribute
to envelopment. It is only after 100ms or more that reflected energy really is
heard as background reverberation, and understanding this time delay is vital
to understanding how halls and operas are perceived with music. The whole
hypothesis is best described in the July 1997 article in ACTA Acustica. The
same material is contained in a somewhat longer preprint for JAES. "Spaciousness and envelopment in musical acoustics."
The JAES preprint also includes a section on how the hypothesis applies to the
practical improvement of halls and operas. This part has not yet appeared in
Acustica. The concept of interaural fluctuations has been used
to solve a very old riddle - the riddle of how many independent bass drivers
one needs in a sound system in a small room, and where should these drivers be
put. To make a long story short: you need at least two low frequency drivers,
and ideally they should be at either side of the listeners. This work is
described in the papers on small room acoustics. The latest paper on this
subject is the one
presented in Vancouver in 2005: “Loudspeaker and listener
positions for optimal low-frequency spatial reproduction in listening rooms”
This paper is highly recommended. Others include:
Much of the work described in the above paper was
done using the MATLAB language. Hardcore researchers might be interested in the
Code that was used. This is available with NO instructions, in the Following
file. Please email the author if you wish to use this code. For this purpose,
use the email address in the picture. The site also includes a recent paper on
reverberation enhancement. Be sure to check out the lecture slides for this
paper - they are much more interesting.: The next item is the lecture notes for a workshop at
the September 1999 Audio Engineering Convention. In this workshop I had about
two hours to cover the essentials of recording technique for surround sound. It
was a lot of fun - but a great deal of what was said is not in the notes. I
believe the AES made a cassette recording. This might be worthwhile. The following lecture slides were presented at the
meeting of the Acoustical Society in The AES conference in October 2000 was fun, but the
slides were prepared in more than the usual rush. Basically nothing new here,
particularly in the first one. Diehard fans might get something out of the
second, but the The Tonmeister conference in Alas, in most cases with large forces the actual
level of the "support" microphones in the final mix is larger than
the level of the "main" microphone, so in practice the roles are
reversed. Nothing intrinsically wrong with this confusion - but it leads to
some rather bizarre recommendations, such as delaying the output of the
"support" microphones so the time of arrival of the wavefront comes
after the signal from the "main" microphone. The remarkable thing is
that adding such a delay does not sound as strange as one might expect. But in
my experience it always sounds worse than no delay at all. Once again the
lecture slides may give the better picture, but you may want to look at both
the preprint and the slides. "Perceptual Modeling" was a term invented
by one of our advertising agents to describe the design of the reverberation
controls in the Lexicon 960. I don't think it means anything at all, which is
good for marketing. But the above paper is quite a useful description of how to
use reverberation to control the apparent distance of a sound source. We have
been doing this with our products for years of course, and the process is well
described in the Lexicon 480 manual with the "ambience" algorithm.
However, outside the manuals I made no real attempt to publish the concepts,
leading to some rather interesting claims by others of having discovered it
all. An interesting issue came up at this Tonmeister
conference. Gunther Theile played a tape made by some of his students, where
they compared the hall pick-up from four omni directional microphones spaced in
a square array at different distances. Unfortunately I was unable to understand
exactly the conditions of the experiment, but the closest set of microphones
used a spacing of ~25cm. In a quick listening test in the listening room at the
show, with about 50 people present, the closest spacing seemed to be preferred
generally over the wider spacings. The result seems to contradict an assumption that I
make in nearly all the work I have done - that uncorrelated reverberation
sounds better than correlated reverberation. The reasons for this result are
unclear. The suggestion offered at the time - that the closer spacing allowed
better imaging of the sides of the room - seems unlikely, among other things
for the fact that side imaging does not exist for a forward facing listener. In
an effort to resolve this issue - which I take to be of the highest importance
- I wrote a note to Eberhard Sengpiel. The note is included here for those who
think the issue is as important as I do. The next series of references are to slides for the
ICA 2001 conference. These references are from the conference of the
Audio Engineering Society in The next paper, on stereo and surround panning in
practice, is pretty good, I think. I wrote it because we were having difficulty
preserving the apparent horizontal direction (azimuth) of sound sources in a
two channel stereo image when we converted the two channel to 5 or 7 channels
with the Logic 7 algorithm. This is interesting because L7 was designed
assuming the standard sine/cosine pan law to be correct. We detected the
left/right balance of a front sound source, and used the sine/cosine law to
find the azimuth. We then adjusted the balance in the three front channels to
present it with the same azimuth. Alas, this does not work. We traced the
problem to the two channel sine/cosine pan law, which is seriously wrong for
most musical sources. (Curiously, the three channel version - that is panning
from a center speaker to either left or right - works quite well.) The reason
for all this is to be found in binaural theory. Turns out in two channel
panning the perceived azimuth is highly frequency dependent, with frequencies
above 1000Hz sounding much wider than the sine/cosine law would predict. Suitable
averaging over frequency gets the right answer.
For some reason this paper has remained undeservedly obscure. Another lecture on surround for the Tonmeisters. I
think both the message and the slides get better the more I do it. And now for something completely different... Being
currently over 60, and having in my youth studied information theory, I have a
low tolerance for claims that "high definition" recording is anything
but a marketing gimmick. I keep, like the Great Randi, trying to find a way to
prove it. Well, I got the idea that maybe some of the presumably positive
results on the audibility of frequencies above 18000Hz were due to
intermodulation distortion, that would covert energy in the ultrasonic range
into sonic frequencies. So I started measuring loudspeakers for distortion of
different types - and looking at the HF content of current disks. The result is
the paper below, which is a HOOT! Anytime you want a good laugh, take a read. Surround from stereo is my most complete
explaination of Logic 7 and its workings. Worth checking it out Slides from the
AES conference October 2003. Subject is converting stereo signals into
surround. And finally we get to something REALLY new. I had
been working for some time on ways of measuring hall acoustic properties from
binaurally recorded speech. It turns out to be pretty simple to learn a lot
about LATERAL reflections from a running IACC. But medial reflections are
trivial to hear in speech and music (at least when they approach the energy of
the direct sound) and the detection process (whatever it is) is very robust. I
decided to submit a preprint without knowing how to solve this problem,
figuring the pressure of the due date might make some progress. Sure enough,
the due date came around, with no solution. Two weeks of very hard work... and
I had an answer. Turns out, we detect medial reflections through their effect
on the audibility of pitch! This ability
(on signals from a single source with a defined pitch) is, I believe, the
primary distance cue. Ultimately I
believe the methods shown here will lead to a new (and quite useful) measure
for sound quality of rooms. The paper for the Be sure to check out the method of deriving
listenable sound examples from Leo Bernaek's published echograms! I am deeply honored that Leo Beranek chose me to share
his lecture to the Acoustical Society, in honor of it's 50th birthday, and
Leo's 90th. Spurred on by the thought that I could easily make an ass of
myself, I put together a pretty good lecture. Highly recommended. Bill Martins put together a little dog and pony
show about low frequency spatial reproduction in small rooms. In honor of this
I made this paper on how to determine the optimal room dimensions and speaker
placement for spatial reproduction at low frequencies. You can do it in an hour
on the back of an envelope if you know the room dimensions. Turns out square
rooms are pretty much impossible. If you have one, tear it down!
Progress in 5-2-5 Matrix Systems
Speaker placement, externalization, and envelopment in home
listening rooms
General overview of spatial impression, envelopment,
localization, and externalization
The .zip compressed Matlab code for experiments with
DFT and externalization. Requires a Working MATLAB C compiler to be practical.
Recent experiences with electronic acoustic enhancement in
concert halls and opera houses
Lecture notes from the September 27, 1999 AES workshop
Recent experiences with electronic acoustical
enhancement in concert halls, opera houses, and outdoor venues - the lecture
slides without pictures.