Hi,
For some unknown reason, I seem to be unable to set SND_PCM_FORMAT_FLOAT
format flag when initialising alsa driver. This happens with my both cards
(on is the laptops AC'97 Audio Controller and the other a similar onboard
cart)
This line used to work with alsa 0.9 but return an "Invalid argument" error
with alsa 1.0.14a :
snd_pcm_hw_params_set_format(alsa_pcm_handle, alsa_hwparams,
SND_PCM_FORMAT_FLOAT_LE)
If I remove this line, the driver gets initialised but as integer sample
format, which, of course, generates noise as I'm sending 32bit floats to the
output...
Example code at lines 1031-1041 :
http://cvs.savannah.nongnu.org/viewvc/freecycle/src/soundplayer.cpp?annotat…
Please, please help !
Predrag
http://freecycle.redsteamrecords.com
>>Well, you have a problem in that the wrt doesn't do floating point very
well.
MAD says it supports 100% fixed-point (integer) computation, and i
am using libmad as of now. to do my decoding
>> Otherwise I'd suggest netjack.
I checked out netjack, but i dont think i want that, as i said i want
the audio from the router directly to the speaker through CAT5 and balun
assembly. Wat i am confused about is after the decoding how, wud i send it
over the ethernet and how wud analog signals travel over ethernet.
Hi,
I am an undergrad student working on a project related to playing audio on a
Linksys WRT54GL Router through cat5 cable. I have the following setup.
A Linksys WRT54GL router with OpenWRT firmware on it. What i want to do is
to play some audio file (prefrably mp3 format), and stream the output over
Ethernet through CAT5 cables and convert it through a balun. I have broken
the process in following steps
1. Get the audio file and decode it (through libmad or a similar library)
2. Convert this to analog form
3. Send it over Ethernet
I understand the first part, and have already started the implementation,
but the other two steps are where i have problems.
To play it directly on a speaker i would need to convert it to Analog, but
how do make those analog signals travel over ethernet?
As you can i dont have much knowledge about audio programming, so i would
appreciate any help i could get on this mailing list.
Thanks.
Hallo list!
I am just thinking about the right strategy for denormal handling in a
floating point (single or double prec) audio application (and yes I
already read the docs of the different methods at musicdsp and so on ...)
Basically my question is, if it is enough to simply turn on the
Flush-to-zero and Denormals-are-zero mode and then compile everything
with -msse -mfpmath=sse ?
I know it won't run on older Pentium3,2 etc. - but for the machines
which support this feature, is this enough ?
Thanks for any hint,
LG
Georg
Hi!
I am trying to create a userspace driver with module uinput and I am
getting into trouble when I terminate the program that creates the
device. After a while the machine will crash ...
The object is to create a joystick device, and I might be missing some
important bits of documentation. So far I am not writing any data to the
interface, but js_demo will find it and open it, showing the expected
number of axis. The name of the interface looks like random though,
which worries me and gets me to suspect that something is wrong
Any idea why the machine is crashing or has a pointer to some example I
could study? As long as I just let the program run, there is no
problem ... Only on termination.
This is what I have gotten so far in the uinput department:
--8<-------------------------------------------------
#include <fcntl.h>
#include <stdio.h>
#include <stdlib.h>
#include <string.h>
#include <signal.h>
#include <linux/input.h>
#include <linux/uinput.h>
int create_uinput() {
int fd = -1;
fd = open ("/dev/uinput", O_RDWR);
if (fd <= 0)
{
fprintf (stderr,"could not open uinput device\n");
exit(EXIT_FAILURE);
}
memset (&uinp, 0, sizeof (struct uinput_user_dev));
strncpy (uinp.name, "NoJoy!", 7); // FIXME!
uinp.id.version = 4;
uinp.id.bustype = BUS_USB;
ioctl (fd, UI_SET_EVBIT, EV_ABS);
ioctl (fd, UI_SET_EVBIT, EV_KEY);
ioctl (fd, UI_SET_ABSBIT, ABS_X);
ioctl (fd, UI_SET_ABSBIT, ABS_Y);
ioctl (fd, UI_SET_ABSBIT, ABS_Z);
ioctl (fd, UI_SET_ABSBIT, ABS_THROTTLE);
ioctl (fd, UI_SET_KEYBIT, BTN_TOP);
ioctl (fd, UI_SET_KEYBIT, BTN_TOP2);
ioctl (fd, UI_SET_KEYBIT, BTN_BASE);
ioctl (fd, UI_SET_KEYBIT, BTN_BASE2);
ioctl (fd, UI_SET_KEYBIT, BTN_BASE3);
ioctl (fd, UI_SET_KEYBIT, BTN_BASE4);
// Create device
write (fd, &uinp, sizeof (uinp));
if (ioctl (fd, UI_DEV_CREATE))
{
fprintf (stderr,"could not create uinput device.\n");
exit(EXIT_FAILURE);
}
sigset(SIGINT,destroy_uinput);
sigset(SIGTERM,destroy_uinput);
return fd;
}
void destroy_uinput()
{
fprintf(stderr,"NoJoy says: BYE!\n");
if(uinp_fd > 0)
{
ioctl (uinp_fd, UI_DEV_DESTROY);
close (uinp_fd);
}
exit(0);
}
--
Fons Adriaensen <fons(a)kokkinizita.net> sez:
>
> On Thu, Sep 27, 2007 at 09:03:32PM -0700, Maitland Vaughan-Turner wrote:
>
> > There was something like a 50% success rate in choosing the audio
> > source correctly.
>
> Which means it was just a random selection...
>
Obviously... (not that it proves anything either way)
I just thought it was an interesting coincidence that this article
appeared right on the heels of our discussion of the subject.
I *did* say maybe you're right and maybe it *is* all in my head.
hehe, no need to rub salt in it. :)
~Maitland
Hiho,
I am having a discussion on the supercollider front about what is the proper
way for dynamic linking.
as far as I know, you use ldconfig and have the library location that programs
dynamically link to defined in /etc/ld.so.conf
but what is supposed to happen if the user just installs the program to a
directory in his home directory?
how should the dynamic linking be defined?
esp. if the user does not have the root rights to change anything
in /etc/ld.so.conf or to run ldconfig.
I did not find anything quick on the net about this, so maybe one of you can
enlighten me what is the "proper" way of dealing with this.
sincerely,
Marije
Hallo!
> our experience with ardour has been that DC bias is measurably more
> effective at reducing CPU load than DAZ, FTZ or both combined. DAZ and
> FTZ do both help significantly, however.
One more question: is it not necessary to deactivate DAZ, FTZ again
after the application (or operation) ?
Or is this done automatically ?
(because I cannot see it e.g. in your ardour code)
Because e.g. in this document:
http://developer.apple.com/documentation/Performance/Conceptual/Accelerate_…
they set it back afterwards.
Thanks,
LG
Georg
Dominique Michel <dominique.michel(a)citycable.ch> sez:
>
> Personally, I don't like it. I prefer very much a good stereo sound in the
> original language (with some kind of text if it is a language that I don't
> understand) like on the Swedish TV.
>
> For that PCM-DSD stuff. I prefer PCM because we can archive a good sound
> quality with a much lower bandwitch. DSD was fine at the beginning of digital
> recording because it was nothing else (for what I know), but for today's
> professional audio, DSD is a waste of resources because of the huge needed
> bandwitch.
>
> Dominique
>
This hits near something I was wondering: Wouldn't it be pretty easy
to do lossless compression on a DSD stream. Since it's only one bit,
It seems like you could use simple run-length encoding to achieve
pretty good results.
~Maitland
Gordon JC Pearce <gordonjcp(a)gjcp.net> sez:
>
> On Tue, 2007-09-25 at 19:31 +0200, Fons Adriaensen wrote:
>
> > A nice variation on this theme occured years ago at an AES conference.
> > The speaker wanted to demonstrate that 'digital' sound was crap, by
> > using the familiar 'push down the extended arm' test. Test persons
> > listening to analog sound could easily resist, while they lost all
> > force when listening to a digital recording.
> >
> > What the speaker didn't know was that the PA system used to play the
> > tracks was fully digital...
>
> I once helped prepare the equipment for a double-blind test of speaker
> cables. All the golden-eared audophiles picked out one cable as being
> far superior to the others, with better clarity and definition in the
> upper harmonics and tighter more defined bass or some such bollocks.
>
> I did have to buy my Mum a new extension lead for her lawnmower, though.
> Sixty feet of Black and Decker's finest, with Speakon plugs soldered to
> it.
>
> Gordon
That's pretty freaking funny. Reminds me of the Penn and Teller where
they sell the diners in a trendy restaurant water from a garden hose
on the patio. haha http://www.youtube.com/watch?v=XfPAjUvvnIc it's
really funny stuff; worth watching if you have a couple minutes.
Incidentally, I get my water from a mountain spring up the road (can't
do much better than that, eh?), although I *have* bought a bottle of
two of Evian in my day :)
As for the double blind audio test, I reckon you guys have all seen
the new AES journal by now (if not, go to your mailbox). There is a
double-blind test where people were played DVD-A and SACD's, but some
of them were passed through an extra A/D/A stage at 16bit-44.1khz.
There was something like a 50% success rate in choosing the audio
source correctly.
Maybe it *is* all in my head...?
Although, I wonder if ear training has anything to do with it. I'm
super dorky sometimes, and I used to mix multitrack projects to
different bit-depths/sample-rates and then try to train myself to hear
the differences between them. hahaha, I figure most people don't do
that sort of thing...
~Maitland
