I have two Audio related projects that need updating.
1. "rsynth" formant based text-to-speech synthesis
2. Audio::* Perl modules.
Both have existing /dev/dsp style backends at present, which have been working
fine. But recently (SuSE 9.0 install?) when run under ALSA emulation of
/dev/dsp they both started producing segfaults - "after program had exited",
(neither valgrind nor gdb can give any info on the fault).
So I decided it was time to do a native ALSA backend(s).
I have rsynth backend working, and perl Audio:: one almost working.
But before going forward I would like to solicit opinions on what
is the "right" API to use.
The Linux audio world seems to be in a state of flux with these options
(please tell me if I missed any):
1. Venerable OSS stuff
- Widely available :-)
- Some cards have quasi or truely commecial drivers rather than
free / opensource ones :-(
- emulation via ALSA (at least as shipped by SuSE) seems broken :-(
2. ALSA
- Reasonably widely available :-|, and improving
- Opensource :-)
- Documentation is lacking :-(
Everything hinges on the
"Configuration Space" concept, but I can't find an explanation of
how that works. This is the sticking point with perl code - I can't find
how to change the sample rate / channels, so _seems_ I need to
close and re-open.
3. JACK
- Gets lots of excited "this is cool" kind of coverage :-)
- realtime :-)
- Callback style not ideal for speech synthesis or play-from-file
of my simple apps. complex :-(
4. Enlightenment Sound Demon
- Seems to be used by KDE etc.
- haven't looked into it further.
5. Network Audio System
- Works on many platforms Linux/Solaris/X Terminals :-)
- X-like "imake" style rather than configure :-(
- Linux version still seems to be based on OSS - so recurse ;-)
6. Presumably there is some kind of telephony API as well, for sending
sound to incomming phone calls via modem / ISDN
Complexity and callbacks don't scare me - I do perl/Tk after all!
But this is a 2nd-string project so I don't want to do a lot of
complex stuff for an API that is vanishing - I would rather either
use a simple stable interface, or pitch in and help on the "comming"
complex API.
Suggestions anyone? (I just subscribed to both lists - so reply to your
favourite list.)
Hi,
is there any linux app available that lets me play around with my yamaha
db 50 xg daughterboard? In windows there was xgedit.exe [non freeware].
I tried to get it to run in wine, but the installer doesn't want to :)
So, i'm looking for an XG editor for linux.
Any hints?
--
music: http://www.soundclick.com/bands/9/florianschmidt.htm
Session Exchange 0.0.1 is available at:
http://www.piratesvsninjas.com/software/session_exchange.py .
It lets people easily manage their ardour sessions, specifically, with
sharing snapshots across the internet for collaboration.
It requires python2.2, pygtk2, and the latest version of twisted-matrix,
available at http://www.twisted-matrix.com .
It uses a three pane model. The first pane is for the session, the
second for the collaborator, and the third for the snapshot.
It is fundamentally done, since it exchanges the files correctly. The
user interface needs touching up, and it needs to be smarter about which
files it transfers. Overall though, I'm feeling pretty good about it.
Stay tuned for documentation. Patches and comments welcome.
Taybin
Hi All,
I've been having some weird problem with my new mainboard which has an
onboard VIA ac97 sound card
When I set everything for OSS drivers, xmms plays mp3 files like
a stuck record player. I can move around the music file but it gets
stuck in the first second.
When I try the piece of code I have written to use OSS techniques,
my simple program which is supposed to play an instrument file with
half a second of music recorded, the card plays the sound repeating it
indefinitely (until I abort the program).
The I switched to Alsa. When I use Alsa xmms plays fine but my piece of
code (unchanged, still using OSS calls) repeats the sample 32 times in 2
channels (stereo), 16 times in single channel (mono)
Linux is MDK 9.2. I tried the same things on a Dell notebook with
exactly the same Mandrake 9.2 (installed from the same network NFS
server) but ofcourse a different sound card, everything works fine.
Therefore it is certain that the problem is with the sound card driver.
From dmesg output:
--------------------------
Via 686a/8233/8235 audio driver 1.9.1-ac3
via82cxxx: Six channel audio available
PCI: Setting latency timer of device 00:11.5 to 64
ac97_codec: AC97 Audio codec, id: VIA97 (Unknown)
via82cxxx: board #1 at 0xE400, IRQ 18
--------------------------
From lsmod output
--------------------------
via82cxxx_audio 22112 0
ac97_codec 15828 0 [via82cxxx_audio]
uart401 8356 0 [via82cxxx_audio]
8139too 17384 1
mii 3864 0 [8139too]
sound 71528 0 [via82cxxx_audio uart401]
soundcore 6340 0 [snd via82cxxx_audio sound]
---------------------------
lspci output
--------------------------
00:00.0 Host bridge: VIA Technologies, Inc. P4M266 Host Bridge
00:01.0 PCI bridge: VIA Technologies, Inc. VT8633 [Apollo Pro266 AGP]
00:09.0 Ethernet controller: Realtek Semiconductor Co., Ltd.
RTL-8139/8139C/8139C+ (rev 10)
00:10.0 USB Controller: VIA Technologies, Inc. USB (rev 80)
00:10.1 USB Controller: VIA Technologies, Inc. USB (rev 80)
00:10.2 USB Controller: VIA Technologies, Inc. USB (rev 80)
00:10.3 USB Controller: VIA Technologies, Inc. USB 2.0 (rev 82)
00:11.0 ISA bridge: VIA Technologies, Inc. VT8235 ISA Bridge
00:11.1 IDE interface: VIA Technologies, Inc.
VT82C586A/B/VT82C686/A/B/VT8233/A/C/VT8235 PIPC Bus Master IDE (rev 06)
00:11.5 Multimedia audio controller: VIA Technologies, Inc.
VT8233/A/8235 AC97 Audio Controller (rev 50)
01:00.0 VGA compatible controller: S3 Inc. VT8375 [ProSavage8 KM266/KL266]
------------------------------------
I shall appreciate any hints or help
best regards
Can Ugur Ayfer
Ron,
S doesn't mean stereo, it means signed for 16-bit sample. The i810_audio support only signed 16-bit little-endian PCM. AFMT_MU_LAW is rarely supported by today's hardware.
ChenLi Tien
-----Original Message-----
From: Ron Shacham [mailto:rs2194@columbia.edu]
Sent: 2003/12/31 [星期三] 下午 05:27
To: linux-audio-dev(a)music.columbia.edu
Cc:
Subject: [linux-audio-dev] very limited formats
Hello, this is my first time posting.
I have been programming on my personal linux box (rh 8), and I've been
unable to set any audio formats on the sound device other than 16-bit
linear. For example, I try the following:
int format = AFMT_MU_LAW;
ioctl(audio_fd, SNDCTL_DSP_SETFMT, &format);
I set this right after opening the device.
Then, when I query for the format, it returns 16.
The following:
ioctl(audio_fd, SNDCTL_DSP_GETFMTS, &mask);
shows zeroes for everything except for AFMT_S16_NE
I have integrated audio hardware (intel i810).
I might also mention that I cannot get mono to work, but only stereo.
Any ideas?
Regards,
Ron
Hello, this is my first time posting.
I have been programming on my personal linux box (rh 8), and I've been
unable to set any audio formats on the sound device other than 16-bit
linear. For example, I try the following:
int format = AFMT_MU_LAW;
ioctl(audio_fd, SNDCTL_DSP_SETFMT, &format);
I set this right after opening the device.
Then, when I query for the format, it returns 16.
The following:
ioctl(audio_fd, SNDCTL_DSP_GETFMTS, &mask);
shows zeroes for everything except for AFMT_S16_NE
I have integrated audio hardware (intel i810).
I might also mention that I cannot get mono to work, but only stereo.
Any ideas?
Regards,
Ron
Introducing the initial release of ac3jack, bringing you realtime
AC3 stream encoding of any JACK audio.
http://essej.net/ac3jack/
ac3jack is a tool for creating an AC3 (Dolby Digital) multichannel
stream from its JACK input ports. Using this tool, an AC3 stream (up
to 5.1 channels) is encoded in realtime and either written to a file or
streamed to standard output.
When streamed to stdout and piped through the ALSA tool 'ac3dec -C',
the AC3 stream can be passed out the SPDIF port on your audio interface
for connection to a multichannel surround receiver. In this way,
you can achieve full 5.1 surround mixing and monitoring of your JACK
applications with a single digital cable, and no need for hardware
supporting discrete inputs and outputs.
AC3 is a compressed audio stream, so quality is somewhat compromised.
It is the price you pay for easy surround sound. After all, if it is
good enough for DVD and film soundtracks, it must be OK.
Please try it out, read the usage notes, and let me know of any
build or runtime problems you find.....
jlc
BEAST/BSE version 0.5.6 is available for download at:
ftp://beast.gtk.org/pub/beast/v0.5/
or
http://beast.gtk.org/beast-ftp/v0.5/
BEAST (the Bedevilled Audio SysTem) is a graphical front-end to
BSE (the Bedevilled Sound Engine), a library for music composition,
audio synthesis, MIDI processing and sample manipulation.
The project is hosted at:
http://beast.gtk.org
A mailing list is available at:
http://mail.gnome.org/mailman/listinfo/beast/
This new development series of BEAST comes with a lot of
the internals redone, many new GUI features and a sound
generation back-end separated from all GUI activities.
The most outstanding new features are the demo song, the effect and
instrument management abilities, the track editor which allowes
for easy selection of synthesizers or samples as track sources, loop
support in songs and unlimited Undo/Redo capabilities.
Overview of Changes in BEAST/BSE 0.5.6:
* New (or ported) modules:
BseEvaluator - highly experimental (available with --devel) expression
evaluator by Stefan Westerfeld
DavBassFilter - a low-pass resonant TB-303 style filter by David A. Bartold
* Added support for author and licensing information for plugins,
available as "Show Info" in the button3 popup menu on modules
* Started MIDI file import ability
* Started new undo-able parasite mechanism for BSE files
* Fixed default value serialization in BSE files
* IDL Compiler bug fixes and cleanups [Stefan Westerfeld]
* Started C++ Language Binding [Stefan Westerfeld]
* Added i18n support to IDL Compiler [Stefan Westerfeld]
* Added #include-impl support to IDL Compiler [Stefan Westerfeld]
* Added toplevel package tests in tests/ [Stefan Westerfeld]
* Support upper case note names
* Internationalized plugins
* Merged translation domains
* Updated Czech translation [Miloslav Trmac]
* Updated Dutch translation [Vincent van Adrighem]
* Updated German translation [Christian Neumair]
* Updated Serbian translation [Danilo Segan]
* Updated Spanish translation [Ismael Andres Rubio Rojas]
* Updated Swedish translation [Christian Rose]
* Added Catalan translation [Xavier Conde Rueda]
* Added Greek translation [Kostas Papadimas]
* Added Portuguese translation [Duarte Loreto]
* Complete rebuild of the GUI code by moving to XML based widget tree stencils
* Completely recoded menu generation, activation and sensitivity, based
on new simple action lists amd a stencil factory mechanism
* Rewrote all existing property entry fields and added new types
* Implemented a couple new widgets to improve GUI experience (GxkMenuButton
as GtkOptionMenu replacement, GxkSimpleLable for shortened widths, ...)
* Added accelerator support for popup menus
* Lots of overall GUI polishing
* Added "About" box
* Miscellaneous bug fixes, lots of code cleanups
---
ciaoTJ
>From: Fred Gleason <fredg(a)salemradiolabs.com>
>
> there are no sample-accurate positioning data in the disk sectors, just
>PCM samples. If the drive loses streaming, there's no reliable way to
>determine precisely where the read left off
Just to sum all up: as we have seen the drive did not loose the streaming;
it was cdparanoia which got out of sync.
So, only thing which fails is the cdparanoia.
>> Yes, I have a hardware problem, specially when using CDROM
>> but also with disks. Buggy IDE I think, or Linux does not support
>
>Then I wouldn't be spending time picking Paranoia apart until this issue is
>resolved.
But who said the problem was caused by the hardware problems?
Honestly I don't *know* if I have hardware problems or not.
I just noticed mails with subject "The trouble with disks"
in linux-audio-user. His problems sounds like the problems I have,
and he too has an MSI board. It could mean that MSI boards are
simply bad or that Linux requires special bios settings.
If anyone else has MSI board and believes it is perfectly ok, please
mail me and we will verify it by doing some tests (as believing is
not enough!).
Question: should the system clock go faster when CD-ROM device is used?
It goes faster here and thus my /etc/crontab has lines for reading
the hardware clock every ten minutes:
01 * * * * root hwclock -s --noadjfile --localtime
11 * * * * root hwclock -s --noadjfile --localtime
etc.
IRQ problems? Wrong settings in bios? Wrong settings in board switches?
Of course, it is always easier to blame the mysterious hardware
problems as one can then forget the possible real problems.
Regards,
Juhana
Hi!
gmorgan is a rhythm station. a full programable accompaniment tool in
real-time and also a pattern based sequencer.
Requirements:
---------------------
ALSA
FLTK
News on 0.20
--------------------
Convert midifiles to patterns.
Patterns and styles added.
Drastically reduced the amount of memory needed (67%).
Changed to Autotools-1.6.
gmorgan is availabe on:
http://gmorgan.sf.net
Thanks
Josep