This is a bugfix and feature improvement release.
Madej added a song title in the title bar option.
Several bugs was fixed. Among them, libmad was fixed in configure and a
possible horrible crash when upgrading from a gtk1 AlsaPlayer version was
fixed. Every user is encouraged to upgrade.
Please contribute with bug reports and artwork. See
http://www.alsaplayer.org/artwork.php3 for existing artworks.
Discussion about which artwork(s) to include with the next stable release will
be done on the mailing list. See the website for details:
http://www.alsaplayer.org
Dominique
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
Dear all,
we are happy to announce the call for papers, music and installations
for the 6th International Linux Audio Conference (LAC2008). LAC2008
will take place in Cologne (Germany) Feb 28th to March 2nd 2008.
The Academy of Media Arts, Cologne, is proud to host the 2008 issue of
this event, where once a year programmers and artists, musicians,
composers and practioneers gather to discuss and explore new and
important developments in making music and sounds with Linux and Open
Source software.
The Linux Audio Conference is both a meeting of developers writing
audio software for Linux as it is a music festival, where artists from
all over the world show how free software can create fresh and
exciting new sounds.
Find the various calls for papers, music and installations attached to
this mail. Further informations and an overview over all calls can be
found at
http://lac.linuxaudio.org
As mentioned in the calls in detail, the deadline for both paper and
music submissions is December 1st 2007, while the deadline for
installations is already October 19th 2007.
We are looking forward to many interesting submissions for the 6th
International Linux Audio Conference 2008 and we hope to see you in
Cologne 2008!
Please feel free to forward this e-mail to anybody who might be
interested.
On behalf of the LAC2008 organisation team,
Frank Barknecht and Martin Rumori
>>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.
I have two questions about DVD-Audio
a)
can you help me to reading the DVD-Audio specification? They are not on
the net as far as I've seen. I want to do some modifications at
dvd-audio.sf.net .
b)
A GUI for authoring DVD-Audio is wanted.
Regarding this whole audio over Cat5 thread, do we actually understand what the objective is of this exercise / project? Is the original authors use of terminology such that we know what is actually supposed to be achieved?
When I first read this, I thought the idea was to produce some sort of network-transported audio stream, like that of a video stream, something which could be "listened to" by some network-connected client computer with its own audio output hardware. Since routers themselves have no speakers, I could not reach any other conclusion. Now it sounds more like a hack to get packets to somehow generate audio as a byproduct ... kind of like the audio version of streaming video via ASCII on a character terminal. Or a better analogy would be getting the old PC "speaker" to emit discernible sounds by sending digital data to it's IO port.
It reminds me of a story my uncle the television engineer told me about how there was a scrap pile of old bolts and cables lying on the floor in the corner of the antenna-transmitter shack of the VHF TV station where he was chief engineer for a number of years before he retired. One day, after replacing a couple of guy wire mounting brackets on the tower (a tower of about 1200'), he threw the scrap hardware on the junk pile in the shack. Suddenly, he could hear the audio signal for the station coming out of the junk pile. He tried to leave the pile undisturbed after that to preserve the effect, but eventually someone bumped or otherwise changed the pile and it stopped "working".
Sorry, I digress. So tell me again ... what is the real goal of this project?
- Doug H
Hi list.
I' m using alsa-lib v1.0.14rc2 on gentoo-linux (v2.6.20-gentoo-r8)
my soundcard is RME Hammerfall HDSP9632 (snd-hdsp driver)
i have been using it for a while - all fine
i only run a few applications for my sound experiments.
one of those is Snd (from CCRMA), it was okay until i have done a few
updates lately - kernel from 2.6.17 to 2.6.20 and alsa-lib as well.
i'm not sure about either of these but i check Snd src-tree and the
mailing list - nothing was changed ..
the error msg that i get is:
:(play)
;status of hw:0
hw_params status of hw:0
ACCESS: MMAP_NONINTERLEAVED RW_NONINTERLEAVED
FORMAT: S32_LE
SUBFORMAT: STD
SAMPLE_BITS: 32
FRAME_BITS: [128 384]
CHANNELS: [4 12]
RATE: [32000 192000]
PERIOD_TIME: (333 256000]
PERIOD_SIZE: [64 8192]
PERIOD_BYTES: [2560 393216]
PERIODS: 2
BUFFER_TIME: (666 512000]
BUFFER_SIZE: [128 16384]
BUFFER_BYTES: [2560 786432]
TICK_TIME: 1000
sw_params status of hw:0
start_mode: DATA
xrun_mode: STOP
tstamp_mode: NONE
period_step: 0
sleep_min: 0
avail_min: 0
xfer_align: 0
silence_threshold: 0
silence_size: 0
boundary: 0
;Invalid argument: hw:0: access type RW_INTERLEAVED not available
can't play : Invalid argument: hw:0: access type RW_INTERLEAVED not
available
#f
:
i had tried to modify audio.c in Snd's tree to use both ACCESS types
given above , but it didn't help - showed device is busy error instead.
i'm only studying C, so can't be sure ..
anyway here is a problem which i could not solve yet and haven't seen
anyone helpful around #alsa, neither i could google anything out.
for a ref.: Cmdist mailing list Cmdist(a)ccrma.stanford.edu http://ccrma-mail.stanford.edu/mailman/listinfo/cmdist
and our discussion is here:
http://ccrma-mail.stanford.edu/pipermail/cmdist/2007-July/004221.html
could anyone have a look at this at somepoint, please .
--
thank you,
Ilya. D
errordeveloper|*@*|gmail|*dot*|com
Hello,
Some interesting products based on Ethernet <-> audio applications may be
found here
http://audioscience.com/
Linux drivers ara available too.
Maybe this can help :)
vedran
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
