On 07/22/2010 03:36 PM, Fons Adriaensen-2 wrote:
> On Thu, Jul 22, 2010 at 09:31:09PM +0200, lieven moors wrote:
>
> > Hi Fons, I'm a fool to even try to answer this question.
> > But I couldn't resist...
>
> :-)
>
>
> > Let's suppose we have two sounds A and B,
> > and sound B has been measured as being twice as loud as A,
> > by somebody. In order to be able to say that, that person needs
> > some kind of reference measurement unit, the equivalent of a
> > measurement stick. That unit has to satisfy two requirements.
> > It has to be big enough, so that people can agree some difference
> > is being measured, and it has to be small enough, so that a multiples
> > of that unit fit into a realistic range. There is a requirement of
> maximum
> > precision (the smallest value we can measure), and a requirement of
> > minimum precision. The question is, what kind of measurement stick
> > is being used by that person.
>
> Not really. If A is 'twice' B, either A or B can act as the reference.
Yes, but we can never agree that A is twice B, unless we agree on
how precise the measurement could/should be.
>
> I'm pretty sure that if you'd do the experiment to find out when
> people think that an object B is twice as big as another object A
> (without introducing optical illusions), you'd find that it's quite
> close to a factor of 2. This is because we can easily imagine two
> A's side by side, which would be 'twice as big' as one A.
I don't think this is easy. Imagine a ruler lying on your desk, and
try to imagine the point where the ruler would become twice as
long. I think you will find that your brain is continually adjusting
that distance, and that it requires significant effort.
It also occurs to me, that by doing this, I am actually
determining the smallest observable difference, and that this
distance is proportionate to the length of the ruler.
>
> Can we do something similar with 'loudness' ? As I wrote, the
> only option I see is to consider two equal sources to be 'twice
> as loud' as one of them, but that doesn't work out.
It could work if we would agree on it. But apparently our brains
have made up their own mind :-)
That's why I propose to reconsider how we look at measurement.
The process of measuring might not be as simple as we think.
>
> Given this, what you write does make sense - there must be some
> 'stick' rather than a real comparison of A to B. But what is it
> based on ? If most people do agree on some value for 'twice as
> loud', even with a large variation, there must be some physical
> ground for this. But what is it ? And a related question: iff
> there is some 'unit' even a variable one depending on frequency
> etc., why can't we imagine that unit ? Why don't we 'see' the
> stick ?
When we see the length of something, we don't see the stick either.
The only thing we see is that one thing is longer than the other.
The stick is just a short thing, which we can compare to all other
things. But even when you agree on such a reference, you still have to
go through the process of measuring.
The problem of 'twice as loud', shows us that we can measure things
even without an agreed-on unit. Somehow we are able to dynamically
create that unit when we need it.
This is how it could work for example:
we have a range, and something within that range: | x
|
we try to cut the range in halves and keep the part with x: | x |
The precision of cutting in halves is based on the smallest
observable difference between the two halves, and is
proportionate to the size of the range. (yes I know this is
a problematic statement, I have to think more about this)
We do the same thing
again: | x |
Now, if we would halve the range again, we would be unable
to distinguish x from that point, and we have some kind of
measuring
stick: |
| | | |
> > First of all, we can assume that the length of that stick will be
> depend
> > on the range of possible input values that we observe, and that we want
> > to measure. If we want to measure the size of a road, we will probably
> > use kilometers, instead of meters. In the same way, when our ears want
> > to measure the amplitude of a sound, our ears will use smaller or
> bigger
> > units, depending on the ranges observed. What are the ranges we
> observe?
> > Let's assume that humans are perfect, and observe everything that we
> > can observe with SPL meters. We could do a statistical investigation
> > on a number of people, and make charts of everything they hear.
> > In these charts we would see what frequencies they are exposed to,
> > and what the minimum and maximum SPL's are for that frequencies.
> > After more analyses, we would have one chart that could be
> > representative for most people.
>
> This is basically what has been done more than 50 years ago, with
> the known results: the objective ratio corresponding to 'twice as
> loud' depends on frequency, absolute level, etc.
>
> > From that chart we could get an estimate of the size of the measurement
> > unit. Frequencies with with bigger SPL variations would be measured
> > with bigger units, and visa versa. And from this we could deduce what
> > the minimum precision is for a certain frequency, when we say it is
> twice
> > as loud. To satisfy the requirement of maximum precision, we should
> > take into account the smallest observable differences for every
> frequency
> > in the spectrum.
>
> 'Smallest observable difference' has been measured as well. It should
> relate in some way to 'twice as loud', but I haven't verified this.
> OTOH, knowing the smallest observable difference does not help to
> define what 'twice as loud' is supposed to be.
As I said above, I think it plays a major role, in our ability to
measure things
Could you elaborate on this a little bit more?
Greetings,
Lieven
>
> Another poster mentioned that he found it quite difficult to work
> out what 'twice as loud' means for him - and I do believe that is
> touching on the real problem: if you start *thinking* about it
> rather than just following your 'gut feeling', how sure can you
> still be of your impression of 'twice as loud' ? How stable is it
> in the face of doubt ?
>
> Keep on thinking !
>
> Ciao,
>
> --
> FA
>
> There are three of them, and Alleline.
>
> _______________________________________________
> Linux-audio-dev mailing list
> Linux-audio-dev@...
> http://lists.linuxaudio.org/listinfo/linux-audio-dev
---------- Forwarded message ----------
From: Marc Hohl <marc(a)hohlart.de>
Date: Sat, Jul 24, 2010 at 8:19 PM
Subject: [tablatures] [Request/Bounty] Tablature bends
To: Lily-Devel List <lilypond-devel(a)gnu.org>, "tablatures(a)lilynet.net" <
tablatures(a)lilynet.net>
Hello all,
after about half a year I have to admit that I don't have a) the time
and b) the programming skills to implement bends in lilypond[1]. Sorry.
Bends are important for guitarists, so I offer 75$ for this issue and
I am posting to the tablature list, because perhaps someone else there
is willing to donate some money too, to get this task done.
During winter 2009/2010, Carl and I discussed the syntax and implementation
details very intensively in about a dozen mails or so, and as things got
nailed down, I wrote a roadmap how the bend engraver and the tab note head
engraver have to be built/extended. There is also a file bend.ly
which uses slurs to mimic bends - the graphic routines for bends are already
there,
so I offer my help providing and explaining the roadmap. A skilled developer
should have no problems to code the new engraver and transform the graphic
routines
into c++ code.
During the bend.ly story, I got in touch with developers of tuxguitar, so I
think
the whole bend story will make lilypond more attractive as a front end to
tuxguitar users and of course to professional music typesetters working on
contemporary
guitar music.
Greetings
Marc
[1] It was really hard for me to accept b), but to be honest, I still
struggle
with some basic concepts of c++ in combination with lilypond, which is
*very*
frustrating.
I suggested this idea on linux-audio-dev list: http://lists.linuxaudio.org/pipermail/linux-audio-dev/2010-July/028590.html, but still have no any answers.
Unfortunally, as I just saw, it was included into other thread: http://lists.linuxaudio.org/pipermail/linux-audio-dev/2010-July/028588.html, I can guess, it is why there are still no answers. I will not confuse, if this message will also be included into some other thread... At least, you could point me other reasons to not answer to that post.
It causes a slight letargy, when question hits into black hole.
Initially I planned to make a special controller for Aeolus, which should allow to write stops toggling to sequencer, but then thought, that it is like a sycle building, and a such thing could be more compact - e.g., controller could be integrated into instrument GUI. A feature to generate correct MIDI signal, written to sequencer, could be disabled, even on the fly, to save performance.
...continuation of truncated mail (does anyone know why this happens?)
>From that chart we could get an estimate of the size of the measurement
unit. Frequencies with with bigger SPL variations would be measured
with bigger units, and visa versa. And from this we could deduce what
the minimum precision is for a certain frequency, when we say it is twice
as loud. To satisfy the requirement of maximum precision, we should
take into account the smallest observable differences for every frequency
in the spectrum.
Greetings,
Lieven
Hi everybody,
I need some help on using the mailing list.
The problem is that I don't know why my posts
are creating new threads in the mailing list
archives. And I am not sure what the subject
line should be, so the post gets attached to the
existing thread. I use "Re: subject" in the subject
line, but this doesn't seem to be enough (although
I think this used to work)
Part of the problem might be that I choose not to
receive all the mailing list mail in my mailbox.
So I am mostly replying to feeds.
Thanks for your help,
Lieven
JACK NetSource GUI is renamed to JACK Network Manager (jack-netmanager-gtk), since jack_netsource is only a command line front-end for jack netmanager module.
Changes:
- Name changed from jack-netsource-gui to jack-netmanager-gtk (humany name - JACK Network Manager), since jack_netsource is just a wrapper, controlling jack "netmanager" module.
- Presets support, deprecating creation of script with tray support, due to ability to reuse ordinary scripts by JACK Network Manager itself. During of preset saving name is requested and approriate script is stored in presets directory.
- Ability to start sources at LADISH rooms.
- ladish_launch is deprecated. ladish_control snewapp and rnewapp commands used instead.
- Improved desktop launcher
- New system tray icons; added application icon, unwantedly removed from 0.2.2
- Fixed help text
- Options for jack_netsource from various jack versions are stored in separated file now
- Internationalization of bash scripts through gettext
- Build system improvements: all text files are configured - e.g., instalation prefix now is working. Configured files are stored into separate build directory, which is autonomous and can be distributed. Also, uninstalation is available.
- Internationalization improvements: added update-locales script, which updates template with translations at once, and localization of bash scripts through gettext.
Note for distribution maintainers: if you place menu items for special audio software into extra submenus inside standart Audio menu, it would be nice to have JACK Network Manager item in one place with QJackNet (at least) and QJackCtl.
And create dummy jack-netsource-gui package for smooth upgrade.
Page at GTK-Apps: http://gtk-apps.org/content/show.php/JACK+Network+Manager?content=122327
Latest screenshot is third.
A lot of kids wish to have a kill switch for their guitars.
A kill switch is a short circuit, to 'stop' the audio signal.
I'm not fine with this solution, but the kids argue, that e.g. an
interruption does cause unwanted noise, especially for over drive
sounds. IMO even using opto-electronics won't solve the issue, because
the noise of the transistor overdrive effect still would be hearable,
while for a short circuit there is silence.
Has anybody an idea to solve this without a short circuit?
I'm really not a fan of short circuits. Note, it's not possible to do an
interrupt all the times behind the latest noise generator and even an
interrupt could cause noise itself, while a short circuit indeed is a
good way to cancel sound.
Hello!
Both Ardour and Qtractor are capable sequencers. Ardour is getting midi
soon, Qtractor is getting automation. All of this is good news.
I was wondering if the devs are interested in implementing routing signals
in a mixer as the next important priority. This would allow not only effects
chains which is usually very useful, but also usage of vocoders.
At the moment it is impossible to use a vocoder plugin inside Ardour or
Qtractor.
The beauty of vocoders is that they can be used not only on voice, but as
electronic instruments as well. I am often using them to create beautiful
textures, but in order for the result to be good,
I need to automate parameters of the vocoder. LADSPA Vocoder, for instance,
is capable of generating very beautiful sounds, but only if you can change
the bands in real time, fade the in and out.
Of course, sometimes it can be recorded live, but in many cases it needs to
be automated and stored in a project.
Tell me what you think.
--
Louigi Verona
http://www.louigiverona.ru/
On 07/22/2010 05:31 AM, Philipp Überbacher wrote:
> Excerpts from Philipp Überbacher's message of 2010-07-22 03:16:00 +0200:
>
> > Excerpts from fons's message of 2010-07-22 02:24:04 +0200:
> > > On Thu, Jul 22, 2010 at 01:05:01AM +0200, Philipp Überbacher wrote:
> > >
> > > > I think the word loudness is a problem here. Afaik it usually
> refers to
> > > > how it is perceived, and twice the amplitude doesn't mean twice the
> > > > perceived loudness. It may mean twice the sound pressure level,
> energy,
> > > > or intensity (if we ignore analogue anomalies, as you wrote in
> some other
> > > > answer).
> > >
> > > Subjective loudness is a very complex thing, depending on the
> > > spectrum, duration, and other aspects of the sound, and also
> > > on circumstances not related to the sound itself.
> > >
> > > For mid frequencies and a duraion of one second, the average
> > > subjective impression of 'twice as loud' seems to correspond
> > > to an SPL difference of around +10 dB.
> >
> > I had a brief look at the section about loudness in musimathics and it
> > mentions 10 dB based on the work of Stevens, S.S. 1956,
> > "Calculation of the Loudness of Complex Noise" and 6 dB based on
> > Warren, R. M. 1970,
> > "Elimination of Biases in Loudness Judgments for Tones.".
> > I think I've encountered the 6 dB more often in texts, which doesn't
> > mean it's closer to the truth, if that's possible at all.
> > Knowing a 'correct' number would be nice for artists and sound
> > engineers, but if it varies wildly from person to person, as Gareth Loy
> > suggests (no idea where he bases this on) then this simply isn't
> > possible. Picking any number within or around this range is probably as
> > good as any other.
> >
> > > I often wondered what criterion we use to determine which
> > > objective SPL difference sounds as 'twice as loud'. We don't
> > > have any conscious numerical value (there may be unconscious
> > > ones such as the amount of auditory nerve pulses, or the amount
> > > of neural activity), so what it this impression based on ?
> > >
> > > The only thing I could imagine is some link with the subjective
> > > impression of a variable number of identical sources. For example
> > > two people talking could be considered to be 'twice as loud' as
> > > one. But that is not the case, the results don't fit at all (it
> > > would mean 3 dB instead of 10).
> >
> > I never thought about that to be honest. It's immensely complex. It
> > might have to do with each persons hearing capabilities, for example
> the
> > bandwidth of loudness perception or the smallest discernible loudness
> > difference. If it really is very different from person to person, then
> > an explanation that takes the different hearing capabilities into
> > account could be sensible, don't you think?
>
> I did find some more approaches to the problem, but those are just
> ideas. From my personal experience I have to say that I have a very hard
> time saying when something is twice as loud. A musically well trained
> person might have an easier time, I wouldn't know, but for me twice as
> loud is something that is very vague. This might already explain the
> large deviation between subjects as described in musimathics. It lead me
> to another idea though, the evolutionary perspective. Evolutionary it
> likely never was important whether a sound is twice as loud. The only
> situation I can imagine where judging loudness probably was important
> is judging distances. How far is the animal I can't see, be it prey or
> predator, away from me? We know that this takes more than the SPL into
> account, and 'twice as loud' doesn't have relevance in this context. So
> maybe the loudness perception is linked with spatialization.
I think this is a very interesting idea. Could this be linked to some
kind of
avarage SPL of all the sounds human beings are exposed to (and this variable
changes throughout history). Because when we try to judge the distance of
a barking dog, our brain would use the knowledge of all other dogs we heard
barking before, to estimate the distance of that dog. If we never heard
a dog
before, maybe we would use the sounds of other animals as a reference,
and so on...
greetings,
Lieven
>
> My other ideas are rather stupid, just ways to get the right numbers for
> your two person idea.
> I simply used ln instead of log and got 7, but that's not even Neper and
> has no relevance.
>
> The other idea of that kind is to assume a field quantity, which would
> result in 6 dB. I'm still easily confused about 10*log and 20*log, but I
> think 20*log is usually used for sound pressure, but maybe not for
> psychoacoustic effects.
> --
> Regards,
> Philipp
>
> --
> "Wir stehen selbst enttäuscht und sehn betroffen / Den Vorhang zu und
> alle Fragen offen." Bertolt Brecht, Der gute Mensch von Sezuan
>
> _______________________________________________
> Linux-audio-dev mailing list
> Linux-audio-dev@...
> http://lists.linuxaudio.org/listinfo/linux-audio-dev
Hi all, i'm new to this list.
I'd like to ask some advice about a small multitrack recorder program i
wrote, and have been using for some time. Basically, what it does is to:
- simultaneously capture sound from several consumer-grade soundcards.
- use libsamplerate to stretch the audio streams, re-syncing them to the
one chosen as "master". The stretch ratio is continuously re-calculated
to make the overall frame count of the stretched stream match the
overall frame count of the master.
- write the "corrected" streams plus the "master" stream to parallel
.wav files using libsndfile.
The purpose is the same as the quite famous "El-Cheapo Howto" (
http://quicktoots.linuxaudio.org/toots/el-cheapo ), just with no
soldering involved :-)
Of course, i know the solution is far from perfect, but i use it to
record some friends of mine who play in a blues/punk band, and the
result is not that bad.
Now, the question is: do you think this piece of code can be of any
interest for someone out there?
Do you think i should i publish it on an open source repository ? Or
maybe there's already some other software i'm not aware of, that does
the same thing?
thanks for your patience, please excuse my bad english.
bye
alberto