Greetings,
I have released a first version of my LADSPA host Soundtank which
holds instruments for you to play.
The project website is http://nymu.net/soundtank
In this program, you can use any LADSPA plugin with a pitch control as
a softsynth; multiple instances handle polyphony and MIDI control is
handled through user-customizeable Event Maps. As a perk, I have
included a command to automatically create Event Maps, ensuring you
instant gratification.
Soundtank is a shell for audio work. Instead of a directory heirarchy,
Signal Path objects hold other objects in ordered lists. The order
dictates the audio signal flow. There are alternate versions of the
shell commands you are familiar with, such as: ls, cd, rm, mv. This
should make it easy to learn to use.
This is a native ALSA app, with full duplex support in JACK, and an
ALSA Sequencer MIDI port is created for every active object. All audio
generation/filtering is handled by LADSPA plugins. Furthermore,
objects are saved to human readable, flat-file format XML files.
To see more, go to http://nymu.net/soundtank
thanks and best wishes,
--jacob
hi all,
i'm considering to add lock free queue support to packet forth, just to
follow the hype, and maybe also because i like the concept. but i'm
wondering if it would make much sense to go through all the trouble with
fast mutexes being available in 2.6
in other words, how much better (apart from being more elegant) are lock
free structures wrt a mutex approach where there is a minimal system
penalty? (not many collisions) did anyone look into this? or have i not
been lurking properly? ;)
cheers
tom
http://slashdot.org/articles/04/09/03/158210.shtml?tid=152&tid=13
"BionicFX has announced Audio Video EXchange (AVEX), a technology that
transforms real-time audio into video and performs audio effect processing on
the GPU of your NVIDIA 3D video card, the latest of which are apparently
capable of more than 40 gigaflops of processing power compared to less than 6
gigaflops on Intel and AMD CPUs."
--
Levi D. Burton
http://www.puresimplicity.net/~ldb/
Hi all,
I am pleased to announce the first beta release
(and the first public release, as well) of
Aqualung, a music player for GNU/Linux
--------------------------------------
Homepage: http://aqualung.sf.net
Aqualung is a new music player for the GNU/Linux operating system.
It plays audio files from your filesystem and has the feature of
inserting _no_gaps_ between adjacent tracks.
Aqualung is released under the GNU General Public License.
Features at a glance
====================
Supported file formats:
* Almost all sample-based, uncompressed formats (eg. WAV, AIFF, AU
etc.), files encoded with FLAC (the Free Lossless Audio Codec), Ogg
Vorbis and MPEG Audio files (including, but not limited to, MP3) are
supported. Naturally, any of these files can be mono or stereo.
Supported output devices:
* OSS and ALSA driver interface, as well as support for connecting to
the JACK Audio Connection Kit.
Key features:
* Continuous, gap-free playback of consecutive tracks! Your ears get
exactly what is in the files -- no silence inserted in between.
* Ability to convert sample rates between the input file and the
output device, in high quality. (Thanks to libsamplerate!)
* LADSPA plugin support -- you can use any suitable LADSPA plugin to
enhance the music you are listening to.
Some other niceties:
* Internally working volume and balance controls (not touching the
soundcard mixer).
* Support for multiple skins; changing them is possible at any time.
* Support for random seeking during playback.
* Track repeat, List repeat and Shuffle mode (besides normal playback)
* All windows are sizable. You can stretch the main window
horizontally for more accurate seeking.
* State persistence via XML config files. Aqualung will come up in the
same state as it was when you closed it, including playback modes,
volume & balance settings, currently processing LADSPA plugins,
window sizes, positions & visibility, and other miscellaneous
options.
In addition to all this, Aqualung comes with a Music Store that is an
XML-based music database, capable of storing various metadata about
music on your computer (including, but not limited to, the names of
artists, and the titles of records and tracks). This is much more
efficient than the all-in-one Winamp/XMMS playlist.
Hope you will like this program. Please report any problems.
Tom
Ingo Molnar <mingo(a)elte.hu> wrote:
>
> I took a
> look at latencies and indeed 2.6.7 is pretty bad - latencies up to 50
> msec (!) can be easily triggered using common workloads, on fast 2GHz+
> x86 system - even when using the fully preemptible kernel!
What were those workloads?
Certainly 2.6+preempt is not as good as 2.4+LL at this time, but 2.6 isn't
too bad either. Even under heavy filesystem load it's hard to exceed a 0.5
millisecond holdoff. There are still a few problem in the ext3 checkpoint
buffer handling, but those seem pretty hard to hit. I doubt if the `Jack'
testers were running `dbench 1000' during their testing.
All of which makes me suspect that the problems which the `Jack' testers
saw were not directly related to long periods of non-preemption in-kernel.
At least, not in core kernel/fs/mm code. There have been problem in the
past in places like i2c drivers, fbdev scrolling, etc.
What we need to do is to encourage audio testers to use ALSA drivers, to
enable CONFIG_SND_DEBUG in the kernel build and to set
/proc/asound/*/*/xrun_debug and to send us the traces which result from
underruns.
As for the patch, well, sprinkling rescheduling points everywhere is still
not the preferred approach. But adding more might_sleep() checks is a
sneaky way of making it more attractive ;)
Minor point: this:
cond_resched();
function_which_might_sleep();
is less efficient than
function_which_might_sleep();
cond_resched();
because if function_which_might_sleep() _does_ sleep, need_resched() will
likely be false when we hit cond_resched(), thus saving a context switch.
Unfortunately, might_sleep() calls tend to go at the entry to functions,
whereas cond_resched() calls should be neat the exit point, or inside loop
bodies.
by Kjetil Svalastog Matheussen <k.s.matheussenï¼ notam02.no>
Dave Robillard:
> On Wed, 2004-09-01 at 07:48, martin rumori wrote:
>> On Wed, Sep 01, 2004 at 11:31:01AM +0100, Steve Harris wrote:
>> > On Wed, Sep 01, 2004 at 10:03:18 +0100, Dave Griffiths wrote:
>> > > so if I'm writing a osc sequencer, is the best plan to leave the
>> > > mapping open for the user to modify?
>> >
>> > I would say so yes, its possible that an OSC schema spe will be
>> > standardised at soem point that would make it easier.
>>
>> not to mention the microtone-capabilities of your osc sequencer and
>> the sophisticated envelope control functions, which are hard to cover
>> with pure midi... :-))
>
> Imagine a sequencer where, instead of little straight bars representing
> notes, the 'piano roll' just allowed you to draw a line to represent
> frequency.. with any angle, straight or curved (bezier), etc. Wow..
>
> Control could be like that too, with overlay and everything, but having
> that for pitch would be amazing.. has something like this ever been done
> before?
>
You can draw lines to represent pitch (and everything else) in Radium, but
its not a piano-roll though. http://www.notam02.no/radium/ (latest
linux-(not-finished)-version: http://www.notam02.no/arkiv/src/)
Hi guys,
I'm coding a software implementation of a buffer for PIO-DA16 (icpdas).
I'd like to know what a stream of stereo PCM data looks like
and how a Digital-toAnalog-Converter understands, which
part of data belongs to which channel in a stereo PCM stream.
Rgds,
Anton