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Klarinet Archive - Posting 000822.txt from 1997/12

From: "line ringuette" <lringuet@-----.net>
Subj: Re: klarinet-digest V1 #476
Date: Wed, 17 Dec 1997 14:03:57 -0500

Re: recent discussions concerning gender and sex in music... this is not at
all new!!!!
Metaphors of gender and sex have been used throughout music's history by
composers, and have been identified and encouraged by many theorists (ie:
A.B. Marx, Schenker, Schoenberg, J.A. Hiller, A. Sorge). I personally have
found this area of music theory very deep and rewarding to research, and I
do feel that it also has expanded my interpretational capacity. If anyone
on this list is interested, try reading:
"Feminist Theory, Music Theory, and the Mind/Body Problem" by Suzanne
Cusick. Perspectives of New Music [I believe this article appeared in 1994,
but it could be anywhere from 1991-1996 - sorry, I didn't note the exact
date!!!!]
"Feminine Endings" by Susan McClary. Minneapolis: U. of Minnesota Press,
1991.

Line Ringuette
Wayne State University
lringuet@-----.net

----------
> From: klarinet-digest <owner-klarinet@-----.us>
> To: klarinet-digest@-----.us
> Subject: klarinet-digest V1 #476
> Date: December 5, 1997 7:08 PM
>
>
> klarinet-digest Friday, December 5 1997 Volume 01 : Number
476
>
>
>
>
> Re: Bass Clarinet Stand
> Re: Reed cases
> Re: Music & Sex was Re: De Peyer
> Sampling Parameters in Digital Audio Systems
> Re: Student Level Clarinets
>
> ----------------------------------------------------------------------
>
> Date: Fri, 05 Dec 1997 18:52:21 -0500
> From: Bill Hausmann <bhausman@-----.com>
> Subject: Re: Bass Clarinet Stand
>
> At 01:37 AM 12/5/97 -0500, Craig Romanec wrote:
> >
> > I'm looking for a bass clarinet stand for a Selmer bass clarinet
> >with low Eb (not to low C) with a peg. Any brand names or manufacturers

> >of such would be appreciated, as well as any ideas for making one (e.g.
> >using a cymbal stand or something?)
> >
> >Thanks.
> >
> I made one using a microphone stand and a U-shaped tool hook. I screwed
> the tool hook into a small piece of wood, and used a small electrical
clamp
> (for attaching conduit to a wall) to attach that assembly to the shaft of
> the stand. The instrument rests on its peg (rubber tip required),
slightly
> inclined toward the mic stand, and is cradled just below the neck by the
> tool hook. It worked very well when used for quick changes between bass
> clarinet and bari sax in a pit band last summer. You may have to adjust
> where the hook grabs the horn on a Selmer, due to its register mechanism;
I
> have a Noblet.
>
>
>
>

> Bill Hausmann
bhausman@-----.com
> 451 Old Orchard Drive
http://www.concentric.net/~bhausman
> Essexville, MI 48732
http://members.wbs.net/homepages/z/o/o/zoot14.html
> ICQ UIN
4862265
>
> If you have to mic a saxophone, the rest of the band is too loud.
>
> ------------------------------
>
> Date: Fri, 05 Dec 1997 18:04:46 -0600
> From: Dee Hays <deerich@-----.net>
> Subject: Re: Reed cases
>
> I use the Reedguards and have never had a problem with denting, falling
out, mold,
> or warping. It has never been difficult for me to tell when I had the
reed inserted
> into the guard. As far as mold or warping, I always wipe as much
moisture as
> possible before I put it in the reedguard. If it is still really soggy
or I have
> used several reeds, I leave the reed case out where the reeds can dry
before putting
> it back in my clarinet case.
>
> Dee Hays
> deerich@-----.net
>
> ------------------------------
>
> Date: Fri, 05 Dec 1997 18:09:59 -0600
> From: Dee Hays <deerich@-----.net>
> Subject: Re: Music & Sex was Re: De Peyer
>
> This is really new in the music world. Mothers were warned not to let
their
> innocent daughters listen to some of Beethovan's works as they were too
> "stimulating."
>
> Jacqueline Eastwood wrote:
>
> > On Thu, 4 Dec 1997, Dan Leeson: LEESON@-----.edu wrote:
> >
> > > For example, the sarabande is not a dignified dance in the least. It
> > > is slow, true. And it is in triple meter true. But the accent is
> > > on the second beat and the dance was sexual in nature. Supposedly
> > > some body movement occurred on the second beat that was to give the
> > > impression of sexual congress. Now before you start talking about
> > > style as limiting itself to interpretations of articulation types
> > > and patters or to tempi, you first have to understand the nature of
> > > the thing you are playing.
> > >
> > > =======================================
> > > Dan Leeson, Los Altos, California
> > > Rosanne Leeson, Los Altos, California
> > > leeson@-----.edu
> > > =======================================
> > >
> > Dan, this reminds me of a stunning revelation told to me by my 10th
grade
> > English teacher when I was preparing notes for a speech against the
> > burning of rock 'n' roll records. He said, "You know, when they're
> > singing about rocking & rolling, they don't really mean dancing!" I
was
> > 14 or 15 at the time -- what a shocker! (OK, maybe I was a little
naive,
> > but times were different then.) Once I knew "the code", this kind of
> > music took on a whole new dimension for me.
> >
> > Decades later, Dan Leeson has managed to shock me once again by
informing
> > us all that the Sarabande is actually representative of something not
as
> > stately or courtly as I had previously imagined. I will never listen
to
> > a Sarabande in the same manner again. In fact, all works containing
> > sarabande movements will now be organized next to the Led Zeppelin and
> > Aerosmith in my collection.
> >
> > (Didn't know I was a crazy rock 'n' roller, didja, Dan? Or that I have
> > sex on the brain permanently? I especially enjoyed the story about the
> > clarinet lesson that was, um, postponed!)
> >
> > Have a nice weekend....
> >
> > Jacqueline Eastwood
> > University of Arizona/Arizona Opera Orchestra
> > eastwooj@-----.edu
>
> ------------------------------
>
> Date: Fri, 5 Dec 1997 16:20:49 -0800 (PST)
> From: Josias Associates <josassoc@-----.com>
> Subject: Sampling Parameters in Digital Audio Systems
>
> Jonathan:
>
> During recent conversations about digital audio technology, I
> posted a message reporting 12-year-old findings of an audio expert
> asserting that commercial digital audio sampling rates (44.1 kHz) and
> quantizing resolution (16 bits) were inadequate for the medium. When you
> disputed those claims, implying that they were typical of those advanced
> by "audiophile zealots," I prepared a draft of a detailed quantitative
> reply on the subject. But, because the debate on the thread was then
still
> at its height, I decided against the knee-jerk reaction of posting an
> immediate response and opted to wait until the dust settled. During this
> hiatus, I've thought additionally about the subject and have solicited
> comments outside the list about some of the message traffic.
>
> Although I'd prefer not spending further time on this matter, I
> view it as unfinished business requiring at least an interim attempt at
> closure. Here, then, is my reply, tempered by a few weeks of reflection
> and also by the inclusion of information previously unknown to me.
>
> Before going any further, I should comment on what motivated me
> to delve into this subject in detail. It certainly wasn't a passion about
> audio engineering, because that's not one of my interests or professional
> fortes. However, as a developer of sophisticated scientific
instrumentation
> for more years than I'd care to admit, data-sampling similar to what is
> used in digital audio systems has been a frequent element of my stock in
> trade. With that background, I was struck by the seeming inconsistency
> of strongly held opinions on the subject. For example, I agreed with a
> number things that Jerry Korten said that you rejected. In my case, with
> all the admiration I have for your musicianship and your breadth of other

> knowledge, I felt particularly hurt that your comments seemed to scold me

> as though I were a callow student who was collossally ignorant of basic
> science and mathematics. I refer to your comments about my relayed report

> about undersampling, underresolution, and high-frequency phase distortion

> as being inconsistent with the Sampling Theorem. And yet, I believe that,

> however you chose to couch your disagreement, your statements about the
> Sampling Theorem were valid....as far as they went. While I also commend
> Mark Charette for his superb and valuable work as keeper of the flame,
> I was troubled with his negative comment about Jerry Korten, whose
> postings had merit. Those are the things that prompted me to write this
> message.
>
> I regret loading up the in-box memories of disinterested people,
but
> the idea of taking this outside the list is not practical. (I have
already
> offered an apology in advance about this to Neil Leupold, who has
> expressed strong feelings about such specialized threads.) For those
> people only mildly interested in the subject, I preface my discussions
> with a comparatively brief summary. Those more interested in the subject
> matter or provoked by the summary claims, as I believe you will be,
> Jonathan, should read further before deleting.
>
> SUMMARY
>
> 1. SAMPLING RATE - Based on the expectation of a useful hearing
> range of 20 kHz, analog reconstruction of audio waveforms at sampling
> rates of 44.1 kHz produces non-negligible amounts of several types of
> high-frequency distortions that would be reduced at higher sampling and
> reconstruction rates;
>
> 2. SAMPLING THEOREM VALIDITY - I have no quarrel with your
> arguments about the sampling theorem. However, the analog
> reconstruction process, which provides a delayed staircase approximation
> to the source waveform (because it is convenient and expedient to do it
> that way), departs from the kind of curve fitting to discrete data points

> or from spectral synthesis that permits the perfection contemplated by
the
> Sampling Theorem;
>
> 3. FREQUENCY RANGE OF HEARING - While I argue that there are deficiencies

> in the ability of commercial digital audio systems to reproduce audio
> signals accurately out to 20 kHz, there is growing evidence that human
> auditory response goes beyond 20 kHz. As an example, human auditory
> perception through bone conduction extends to beyond 40 kHz (Ref. 1:
> Lenhardt et al). There is also other evidence of sound perception above
> the audible range (Ref. 2: Oohashi et al);
>
> 4. COMMERCIAL USE OF HIGHER SAMPLING RATES - Much of
> the recording industry has been aware for some time of correctable
> deficiencies in recording practices. In one example, studios are already
> using higher sampling rates, and digital video discs will soon do the
same
> for the audio portions;
>
> 5. QUANTIZING RESOLUTION - Louis Fielder, now Chief Engineer
> at Dolby Labs, reports in a series of related papers that the dynamic
range
> of live music dwarfs the dynamic ranges of analog and 16-bit-digital
> systems. Results of tests performed about 10 years ago by the British
> Broadcasting Company demonstrate that 99% of the population can hear
> quantizing granularity down to a 22-bit resolution threshold (where the
> granularity becomes concealed), and 1% can hear better than that; and
>
> 6. DOUBLE BLIND TESTS - Although I suspect that a number of
> comparative analog/digital double-blind tests have been run, especially
at
> manufacturers' facilities, I do know without speculation that one such
> double-blind test was performed off the same feed at Caltech in 1982. I
> witnessed a convincing demonstration of the Caltech comparisons in 1985.
>
> The balance of this posting is devoted mainly to a discussion of
the
> kinds and levels of distortions that result from undersampling.
>
> DISCUSSION
>
> If I am correct about the existence of high-frequency anomalies in
> digital audio systems, how then can such effects be reconciled with your
> valid statements about the Sampling Theorem?
>
> I aim to answer this question and then describe quantitatively the
> consequences to the reconstructed waveforms of sampling too slowly.
> Within the constraints of the Sampling Theorem, one can presumably
> curve fit the original source waveform perfectly about discrete sampling
> points, preserving both amplitude and phase information. Alternatively,
> one could conceivably synthesize periodic waveforms using fast Fourier
> transforms, except that audio waveforms are inconveniently aperiodic.
>
> While sampling procedures used in recording digitizers are
> consistent with the Sampling Theorem up to that point, the analog
> reconstruction playback process is not, because it employs a flat step
> between reconstituted samples and not a discrete point value against
> which curve fitting or some other synthetic procedure could be used. At
> frequencies much lower than F/2 (where F is the sampling frequency),
> this reconstruction process has a negligible effect on waveform
> reproduction. But, at higher frequencies, this type of system not only
> produces high-frequency phase distortion, it also produces non-negligible

> amounts of harmonic distortion and amplitude modulation together with
> attendant spurious sidebands.
>
> These pernicious high-frequency effects are artifacts of the
non-ideal
> reconstruction process and have nothing to do with imperfect ADCs or
> DACs. I have satisfied myself that the artifacts do exist, and others
have
> demonstrated that the artifacts can be heard.
>
> The following section examines the distorting effect of the delayed
> staircase approximation to curve fitting used in reconstructing sampled
> audio signals. You questioned my understanding of the Nyquist Theorem
> as follows:
>
> On Thu, 13 Nov 1997, Jonathan Cohler wrote:
>
> > This is not correct. The Nyquist Theorem says that if an analog
> > signal is digitally sampled at a frequency F then using those
> > digital samples one can PERFECTLY reconstruct all frequency
> > components of the original signal up to a frequency of 1/2 * F.
>
> You highlight the word "perfectly" with caps, which is valid, but you
> should also highlight the preceding word "can." That is, because,
although
> one "can" presumably reconstruct perfectly all frequency components
> up to F/2 using digital samples, the reproductive process employed in
> digital audio systems "does not" do a perfect reconstruction. While it
may
> be possible to approach such perfection with another system, the
> complexity and costs of such a reconstruction processor would be much
> greater and less convenient than the comparatively simpler and more
> straightforward approach now in use. As a result, there are important
> limitations to being able to perfectly reconstruct waveforms with respect
> to phase, amplitude, and structure, as I'll demonstrate.
>
> Assuming perfect ADCs and DACs, the receiver DAC will extrapolate
> a flat level until receipt of the next digital number. At audio
frequencies
> well below the sampling frequency, one can expect faithful waveform
> reproduction. But that is not the case in reconstructing the top end of
the
> sampled-data spectrum.
>
> As an extreme example, consider a phase-synchronous sinusoidal input
> to the ADC at the Nyquist Frequency. The output will be a square wave
> at the Nyquist Frequency. If sampling occurs near the nodes, the output
> square wave will be nearly in phase with the source signal, except that
the
> amplitude will be highly attenuated. Thus, like doppler radars, which are
> blind to moving targets of certain velocities, sampling systems produce
> two blind phases at F/2 -- phases where the signal disappears. If ADC
> sampling occurs at the waveform maxima, a full-amplitude square wave
> will be produced, but phase shifted by 90 degrees. Thus, reconstructed
> amplitude and phase are not automatically that of the source signal. At
> nearby lower frequencies, signals will slip by the sampler so that a
> maximum can be detected for purposes of a spectral display. But, in terms
> of real-time audio near F/2, there will be a square-wave component
> modulated at the slip rate with a peak-to-peak modulation index of 100%.
> Viewed on a spectrum analyzer, one would see the signal frequency plus
> side bands. At frequencies higher than F/2, aliasing occurs which
> produces false signals at apparent frequencies below F/2. We'll assume,
> however, that the audio input to the ADC is band limited to attenuate any
> residual signals above the Nyquist Frequency.
>
> Now consider a similar source signal at 1/4 of the sampling
frequency,
> which would be near Jerry Korten's 10 kHz. For the sake of illustration,
> sample this waveform at the nodes and maxima. Beginning at the first
> node, one gets zero for 90 degrees, plus maximum for the next 90
> degrees, zero for the next 90 degrees, and finally negative maximum for
> the last 90 degrees of that cycle. This full-amplitude bipolar pulsed
> waveform is displaced by 45 degrees and, worse still, is rich in harmonic
> content that was never there in the first place. But, fast edge
transitions
> and higher harmonics will undoubtedly be attenuated with a post-DAC
> low-pass filter, which might possibly produce a bumpy poor-man's
> triangular wave of the kind Jerry Korten saw.
>
> Sampling of the same F/4 waveform at 45 degrees, 135 degrees,
> 225 degrees, and 315 degrees produces a square wave with 70.7% full
> amplitude, also phase shifted by 45 degrees lagging. As before, signals
at
> nearby frequencies, both above and below, produce artificial amplitude
> modulation (in this case with a peak-to-peak modulation index of 29.3%)
> and unwanted sidebands. The analog reconstruction approaches
> distortionless perfection asymptotically as the signal frequency becomes
> a smaller fraction of the sampling frequency.
>
> The question of phase distortion was raised again in your
> critique, as follows:
>
> You said:
> > Wrong. "The digitizing process" has no high-frequency phase
> > distortion associated with it. Some bad A-to-D converters may
> > have some high-frequency phase distortion. Certainly, phase
> > distortion problems are much more prevalent in analog recording
> > equipment.
>
> The problem is not with the digitizing process; the problem is with
> the decoding or reconstruction process. The phase distortion of the
> high-frequency components has nothing to do with imperfections of the
> ADCs or DACs. It is, moreover, a byproduct of the relationship of the
> sampling and signal phases at higher frequencies and the staircase
> approximation of the reconstructed signal to the original signal. Also,
> the problems associated with the composite output signal are not
> exclusively those of phase displacement, because undesirable
> attenuations, modulations, and harmonic distortions occur, as well.
>
> Furthermore, if predigitized signals are band limited to prevent
> frequency foldback of aliased signals above F/2 after sampling, something

> that is usually done in sampled-data systems, reconstructed signals will
> then not only have the previously mentioned distortions attributable to
> the playback process, there will be an additional phase lag from the
> pre-digitizing band limiter in the recording setup. If, for example, a
> double-pole filter at a corner frequency of F/2 (22 kHz) were used ahead
> of the ADC, the phase shift at F/4, 11 kHz, would be -53 degrees, and
> that would be in addition to the -45 degrees resulting from
> reconstruction-induced phase distortion.
>
> While the industry still has a way to go to reach a Utopian ideal in
> digital audio signal processing, I have no doubt that digital processing
> quality, which is already superior in noise performance, can eventually
> equal that of analog systems. As illogical and surprising as it may seem
> to some who are first becoming aware of it, knowledge of defects in
> digital audio reproduction due to undersampling and underresolution has
> existed for a long time in the industry. I am pleased to learn that
> manufacturers are moving in a direction to eliminate these deficiencies,
> something they would not be doing unless they believed that there were
both
> room for improvement and competitive pressure to do so.
>
> Regards,
>
> Connie
>
> Conrad Josias
> Consulting Engineer
> La Canada, California
>
> Reference 1: Martin L. Lenhardt, Ruth Skellett, Peter Wang, Alex M.
> Clarke, "Human Ultrasonic Speech Perception." Science, Vol. 253, 5 July
> 1991, pp. 82-85.
>
> Reference 2: Tsutomi Oohashi, Emi Nishina, Norie Kawai, Yoshitaka
> Fuwamoto, Hiroshi Imai, "High-Frequency Sound Above the Audible
> Range Affects Brain Electric Activity and Sound Perception." Audio
> Engineering Society preprint No. 3207 (91st Convention, New York
> City).
>
> ------------------------------
>
> Date: Fri, 5 Dec 1997 17:22:11 -0700 (MST)
> From: Josh-Boy <joshcole@-----.Edu>
> Subject: Re: Student Level Clarinets
>
> The Artley line of clarinets has many problems. I owned one for my first
> 5 years of playing clarinet. For the longest time, I thought the
problems
> were 100% me! Then I discovered that my intonation problems and
> difficulties crossing the several breaks of the clarinet were only 50%
me.
> And I, too, have experienced the wonderful cracking of the plastic
> clarinet. The Artley is good if you have nothing else. Besides, if one
> can sound good on an Artley, imagine what he or she will sound like on a
> Concerto or a Prestige?
>
> Josh
>
> JoshStillNeedsANewNameJoshStillNeedsANewNameJoshStillNeedsANewNameJosh
> Visit Mr. Josh at http://web.nmsu.edu/~joshcole
>
> ------------------------------
>
> End of klarinet-digest V1 #476
> ******************************

   
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