Hi!
The aim is trivial:
- get some wav file,
- burn it to CD-R(W),
- rip it, get a copy,
- compare the original file with the copy.
If I understand well, the main problem here is to determine "the same
starting point" for each of files. Is there some kind of software which
may be useful in such comparison.
(probably, some of answer may be like this: "learn scheme/ruby and use
snd". I'm interested in more short way :-)
Andrew
Seems to be pretty good. The audio samples were flawless in XMMS. I
played the video files in the VideoLAN client. They seemed to be pretty
good, but the video was quite a bit jumpy in high-res mode (could be on
my end, of course).
On Wed, 6 Apr 2005 22:09 , Erik de Castro Lopo <erikd-lad(a)mega-nerd.com> sent:
>On Wed, 06 Apr 2005 00:31:15 -0500
>Shane lists(a)itsagesolutions.com> wrote:
>
>> Hey everyone. I have a bland but important question for everyone. Say
>> hypothetically a company is developing an audio product using lots of
>> GPL source, but for whatever marketing reasons asks for NDA concerning
>> the codebase. Lots of GPL work is referenced and at least dynamically
>> linked, and though the company has directly stated that it will release
>> the codebase publicly with the product release (once it is complete).
>>
>> I am curious as to the general feel in the community on such practices.
>> Would this 1) be a violation of the GPL,
>
>Yes.
>
>> 2) if it is how tolerant would
>> the OSS community be, considering the general good intent of the
>> project, and
>
>If any of my code is involved, I will prusue the matter.
>
>
>> 3) if I were asked to sign such a NDA would that document
>> be a binding agreement even if the NDA itself might be a violation of
>> the GPL since it is inherently counterintuitive to the intent of the
>> GPL.
>
>The GPL is pretty clear about this. The GPL comes into action when
>the binaries or code derived from GPL code is **distributed**.
>That means that you and the company can hack on whatever GPL code
>you like as long as they don't release a binary (under NDA or not).
>
If you are part of the company or a contractor to the company and the product
is not distributed outside the company you can do whatever you want with the code
without having to release it. If you are outside the company and have to sign an
NDA then I think that would be considered distribution and would therefore
violate the GPL. As Erik said, once it's distributed then it has to be released.
Jan
>
> From: Alfons Adriaensen <fons.adriaensen(a)alcatel.be>
> Date: 2005/04/06 on PM 01:00:45 GMT
> To: "The Linux Audio Developers' Mailing List"
> <linux-audio-dev(a)music.columbia.edu>
> Ämne: Re: [linux-audio-dev] FOSS-tools for realtime audio analysis?
>
> You may mave a look at JAAA, (JACK and ALSA Audio Analyser). This is
a
> realtime 'technical' (rather than musical) spectrum analyser,
designed
> to make accurate measurements. It will also generate sines and white
> Gaussian noise. It's at
>
> <http://users.skynet.be/solaris/linuxadio>
>
> You will also need some small shared libs available from the same
> place.
>
> I'm considering some additional functionality, and if you have any
> specific requirements I'd be interested to know.
>
Thanks! Actually, it's a friend of mine interested in something like
this. One thing that I know he was curious about was if it would be
possible to develop some kind of test-case for external hardware, which
could give an indication of whether something is wrong with it or not.
>From just taking a peek at it, I'm getting the impression that this is
mainly a graphical tool (which is really cool). Anyway, he's not afraid
of hacking if this is what he's looking for. Are features like these
anything you've considered/already implemented/interesting?
> I am also working on a Room Impulse Response measurement system using
> a sine sweep as the stimulus. First release expected in a about two
> months.
>
> --
> FA
>
>
>
>
>
> I would like to invite you all to have a look at The Freesound Project.
>
> -> http://freesound.iua.upf.edu/
>
> This project was started in the context of ICMC 2005, and is a website for creative-commons licensed audio material exchange. If you want to know more about the project (why it was started and what it's goals are) have a look at this page:
>
> -> http://freesound.iua.upf.edu/whatIsFreesound.php
>
> Don't hesitate to contact me (bdejong(a)iua.upf.edu) for more information or feedback.
>
>
> kindest regards,
>
> Xavier Serra
> ICMC 2005 Program Chair
>
> PS: please feel free to forward this message to relevant mailing lists or interested people.
>
>
--
/**********************************
********* Xavier Amatriain ********
****** Music Technology Group *****
** IUA- Universitat Pompeu Fabra **
****** www.iua.upf.es/~xamat ******
***********************************/
Hi!
Are there any tools/libraries around which makes it possible to do
something like this?
1) Feed an audio signal (f.ex. a sinus wave) into external hardware
2) Record the signal returned from the external hardware
3) "Compare" the two signals (realtime would be neat, if you know what
latency you have btwn output/input I hope this might be achieved w
Jack) and perform some kind of analysis of what the external hardware
does to the signal (S/N ratio, damping, etc etc).
Open source stuff would of course be preferable
/Mathias
Hi Steve,
Was testing this scenario today, which may look familiar to you:
- I created a server with some methods: '/ping/documentation' and
'/osc-schema/documentation', both of which use the same handler 'doc'.
- I send query '/*/documentation' to the server.
- 'doc' got called twice with path '/*/documentation'.
The unexpanded path makes it impossible for the handler to determine
what container was intended. In my interpretation the OSC spec is not
completely explicit on this point, but I think the intention was to
send the expanded path to the handler. The spec is specific in stating
that '*' is invalid in an OSC method, and it talks about the "OSC Address
of a OSC Method".
Initially I hacked liblo to always send the expanded path. However, the
result was also that the generic method handler (oh, I used that too :)
got called with a NULL path. Since that is useless to that handler it
seems best to me to use the unexpanded path for this case.
The resulting patch is applied. Please let me know your view on this.
Cheers,
--
Martin
---------------------------------------------------------------------------
Index: liblo/src/server.c
===================================================================
--- liblo.orig/src/server.c 2005-03-02 19:15:47.000000000 +0000
+++ liblo/src/server.c 2005-03-26 11:33:41.000000000 +0000
@@ -463,6 +463,7 @@
lo_address src = lo_address_new(NULL, NULL);
char hostname[LO_HOST_SIZE];
char portname[32];
+ const char *pptr;
free(msg->types);
msg->types = types;
@@ -542,7 +543,13 @@
}
}
- ret = it->handler(path, types, argv, argc, msg,
+ /* Send wildcard path to generic handler, expanded path
+ to others.
+ */
+ pptr = path;
+ if (it->path) pptr = it->path;
+
+ ret = it->handler(pptr, types, argv, argc, msg,
it->user_data);
} else if (lo_can_coerce_spec(types, it->typespec)) {
@@ -572,7 +579,13 @@
}
endian_fixed = 1;
- ret = it->handler(path, it->typespec, argv, argc, msg,
+ /* Send wildcard path to generic handler, expanded path
+ to others.
+ */
+ pptr = path;
+ if (it->path) pptr = it->path;
+
+ ret = it->handler(pptr, it->typespec, argv, argc, msg,
it->user_data);
free(argv);
free(data_co);
Hi all,
First of all, please accept my apologies for cross-posting. I simply feel
that this issue may be of interest to all these groups.
After some research and experimenting with jacklaunch (libjackasyn) as well
as alsa .asoundrc jack plugin I found out that very few apps actually
cooperate with either of these.
Jack plugin for asoundrc practically works only with aplay/arecord and
freebirth. Audacity doesn't cooperate with either, while jacklaunch even
though it has been reported to work with sweep doesn't on my machines (mdk
10.0 and 10.1).
I've tested latest jack, sweep, Alsa 1.0.6 and 1.0.7 (I did not go beyond
that as 1.0.8 has apparent issues with hdsploader) and had no luck
whatsoever.
Hence, I am just wondering at this point whether asoundrc jack is still in
the testing phase (some of the errors that apps reported trying to use jack
plug were due to inability to probe some info from the alsa driver which
implies possible incompleteness of the implementation or weird probing from
the app's side) and also whether anyone has had better luck than myself
using either of these with audacity, sweep, and other non-jackified, yet
important Linux audio apps via either of the aforementioned ways.
FWIW, it would be really nice to see some united effort to jackify some of
the other critical Linux audio apps even if their original authors do not
feel like that is one of their priorities (I know, I know, if I am so
interested in such developments then I might as well set an example by doing
what I am advocating, which is something I would love to do, but
unfortunately currently the notion of "free time" in my life is but a joke).
Many thanks!
Best wishes,
Ivica Ico Bukvic, composer & multimedia sculptor
http://meowing.ccm.uc.edu/~ico/
--
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Checked by AVG Anti-Virus.
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Hi all,
Is there a chance that signals generated from software could damage live PA
equipment? I'm guessing not, as there is obviously (I assume) some limiting
built into soundcards, but I'm just wondering what applications do to, if
anything, to help.
It's just something that worries me sometimes when playing live, and
generating particually nasty noises by accident, or design :)
cheers,
dave
Andres Cabrera wrote:
>>Overheating of the amp is nonsense. Most amplifiers operate in class A
>>or AB anyway, and maybe in class D. Heat dissipation is independant of
>>the input signal for these amps. The types that might build up a problem
>>with heat (e.g. class H) aren't used in PA enviroments. And before the
>>amp overheats, your speakers will be dead.
>>
>>
>
>Thanks for clarifying. I remembered I heard about overheating somewhere
>with DC, and thought it was on the amp.
>Adding a small DC offset is one easy way of getting rid of denormal
>problems (in programs like csound and pd), do you know if this small DC
>offset is able to produce these effects?
>
>
I don't know about the output stage of a soundcard. (Almost) every
soundcard uses a delta-sigma DAC, which needs some sort of low pass
filtering. It needs an output buffer to transform the rather high
impedance of this filter to a low output impedance, so the question is
if there is some sort of DC decoupling impemented in this buffer. I
don't know, but using a voltmeter (or scope) while applying a DC signal
to the card will answer that.
if you take a look at the specs of the EMU1820M it states "Frequency
Response: 0.0/-.35dB, 20Hz - 20kHz" which would mean that it doesn't
output DC signals.
But as far as I know, most audio input stages are capacitively coupled,
thereby removing such DC offsets, so there is little chance that this
type of signals will propagate through the entire signal chain.
Greets,
Pieter