I haven't posted anything in a while, but I'm sure some of you remember
I made a few claims over the summer. Fear not, things are moving along.
Anyhoo, I had a spark of inspiration and banged out a little sort of folk tune
tracked with ardour. Audacity as a front end for LAME. Still getting my ears
calibrated to a subpar monitoring system so forgive the buried vox and any
tubbyness.
4 tracks.
1 scratch guitar & vocal (the limiting factor)
1 bassline
1 vox overdub
1 keyboard
http://65.125.227.61
click the title header for a link
l8r
J
Hi all:
I've been thinking some more about a prototyping/development board (not
a production board) for a free open multi-channel firewire audio interface.
The objective would be to keep initial development time and costs down
even at the expense of pushing component costs slightly up.
Here are my very rough, version 0, not-fully-thought-out thoughts:
~ Use off-the-shelf asics for the firewire link and physical layers
~ Use a small-ish xilinx spartan 3 fpga for the rest of the "signal
path" logic
~ Use a microchip dsPIC for everything that's off the "signal path"
~ Leave A/D, DA converters (or dig audio I/O) etc entirely off the board:
Just provide headers for adding them on a sub-board later.
There's a block diagram here:
http://www.sjenkins.pwp.blueyonder.co.uk/fwboard/block01.png
Some notes:
1. The smaller spartan 3 fpgas are supported by the free, downloadable ISE
WebPACK software.
2. The PCI block in the fpga doesn't need to be that fancy or versatile.
It just
needs to handle the one TI component fast enough to get data in and out of
its FIFOs on time plus the occasional (much less time-critical) interactions
between the dsPIC and the link layer. It doesn't need to do this stuff
as fast
as possible - there's nothing else on the PCI bus to contend the bandwidth -
it just needs to do it fast enough.
3. The dsPIC never, ever touches the audio data. The converter interface
engine directly handles all transfers between the link-layer fifos and
its own
buffering. It also directly handles any other time-critical interactions
with
the link layer.
4. The dsPIC does everything else: It...
~ stores and programs the fpga configuration data
~ performs all non-time-critical interactions with the link layer and, via
the link layer, with the driver.
~ initialises the converter interface engine ready for whatever formats of
audio I/O may be about to start
~ stores user configuration data
~ handles any LEDS, switches, pots, encoders, LCDs or whatever might
go on an interface front panel.
~ does any non-time-critical control of an analog I/O subboard. (eg if
analog
monitoring were implemented the dsPIC would do the necessary switching).
~ could handle a first-cut implementation of MIDI, although this might
better be migrated to the fpga.
Comments?
Cheers
Simon
I'm proud to announce a new stable release of Hydrogen Drum Machine!!
Features:
__General__
* Very user-friendly, modular, fast and intuitive graphical interface based
on QT 3.
* Sample-based stereo audio engine, with import of sound samples in .wav, .au
and .aiff formats.
* Support of samples in compressed FLAC file.
__Sequencer and mixer__
* Pattern-based sequencer, with unlimited number of patterns and ability to
chain patterns into a song.
* Up to 64 ticks per pattern with individual level per event and variable
pattern length.
* 32 instrument tracks with volume, mute, solo, pan capabilities.
* Multi layer support for instruments (up to 16 samples for each instrument).
* Ability to import/export song files.
* Unique human velocity, human time and swing functions.
* Multiple patterns playing at once.
__Other__
* JACK, ALSA and OSS audio drivers
* ALSA MIDI input with assignable midi-in channel (1..16, ALL).
* Import/export of drumkits.
* Export song to wav file.
* Export song to midi file.
Changes:
* New ALSA driver
* New french tutorial and manual page (thanks to Pierre 'AlSim' Chapuis)
* Bug fix
Download:
http://hydrogen.sourceforge.net
Vote hydrogen at the italian open source contest!
http://hydrogen.sourceforge.net/HowToVote.html
Happy drumming! :^)
--
Alessandro <Comix> Cominu
http://hydrogen.sf.net
e-mail: comix(a)despammed.com
Icq: 116354077
Linux User # 203765
Along with Common C++, I also made available the first release of the
new and much improved stand-alone (no longer requires GNU Common C++)
version of the GNU Common C++ Audio class framework; "ccaudio2". This
new framework, in addition to being more portable and fully endian aware
than it's predecessor (it builds and runs on platforms as varies as
GNU/Linux, various BSD's, OSX, and W32) also includes a new stand-alone
utility, "audiotool", which can do various basic manipulations with
audio files, and a revised system for supporting audio codecs as
plugins. ccAudio2 still needs additional work, particularly in support
of .mp3 and .ogg audio containers, and in getting additional codecs working.
We also had a w32 installer built using GNU GPL licensed "inno setup"
builder which installs all three class libraries together, along with a
new class reference manual, for those using a freedom challenged platform.
The one place all these things can be found together at the moment is:
http://sourceforge.net/project/showfiles.php?group_id=1523&package_id=41672…
Hi,
AFAIK:
OSS device on emu10k1 is always routed to front channels.
Routing can be changed through alsa mixer api (there are controls
controling channel routing), but this is not simple and there is
problem to know what oss stream uses what hw stream, because these are
alocated dynamicallly.
Other ways will require driver change.
Peter Zubaj
____________________________________
http://www.pobox.sk/ - najvacsi slovensky freemail
Hi,
I was wondering, is it possible to assign /dev/dspX devices to the
secondary and tertiary PCM devices on an emu10k1?
I have a SB Live! Platinum with LiveDrive IR. The stereo out is connected
to my regular set of speakers. The surround output is connected to an
earphone headset, it's mic is connected to the mic input on my sound card.
Also, another microphone is connected to the mic/line on the LIveDrive.
Now, I use skype, which is closed source. It uses OSS devices and aoss will
not work with it. I would like to have a /dev/dspX device that records from
the mic input and plays back to the surround output, so that skype, and
skype only, will use the headset.
The headset works fine in alsa mode and alsa apps can use it perfectly well.
I tried all /dev/dspX and /dev/adspX devices, to no avail. I tries aoss
with .asoundrc modifications, no luck. I even read the driver source, but
I'm not really conversant with the structure of the driver and couldn't
find anything useful. It would take me forever to figure it out from the
source code.
Is it possible, maybe with module parameters, to make alsa do this? Would
it need a patch, if yes, does someone have one? If no, what would have to
be done where to make that work?
Melanie
On Sun, 2004-11-28 at 14:03, tim hall wrote:
> Last Saturday 27 November 2004 21:36, Lee Revell was like:
> > On Sat, 2004-11-27 at 15:43 -0500, Lee Revell wrote:
> > > > Did this happen?
> > >
> > > Maybe not to them but look at Mackie and Behringer.
> >
> > Just to save people some googling here is a thread that documents the
> > long and colorful history of pro audio hardware manufacturers blatantly
> > ripping each other off, often leaving the victims with no legal
> > recourse:
> >
> > http://homerecording.com/bbs/archive/index.php/t-74439.html
> >
> > IMO the issue is not whether RME's concern is valid - clearly it is.
> > Sorry, but arguing otherwise makes us look stupid and naive. The issue
> > is how to address this concern. If that means a closed source Linux
> > driver, fine.
> >
> > Maybe the reason no firewire hardware is supported is because Behringer
> > and their ilk would instantly have all the info they need to copy the
> > design and mass produce it. Doesn't matter how cheap the device is to
> > design - it will _always_ be cheaper to rip someone off than design it
> > yourself. They can even sell at a loss, due to huge cash reserves -
> > they only need to sustain it long enough to put the competition out of
> > business. In the case of the "Swizz Army Tuner", the original designers
> > were ripped off by Behringer, but a lawsuit would have bankrupted them
> > _even if they won_ so could not take action.
> >
> > I think many people in this thread underestimate how cutthroat the
> > hardware business is.
>
> Yeah, If I was the MD of RME, after reading some of the responses on this
> thread I'd be thinking of flippin' the bird at all these ungrateful linux
> users.
I think it's about defending the position of open source and its nature.
And the work that people do here no matter whether for fun or not.
>From now on every company that doesn't do it like audioscience does, is
a plain loser to me, no matter whether they provide specs or not. It's
because other people do the actual work + support providing.
If MacOSX can have them, so can we, we have a greater marketshare.
Why the heck should we *always* understand them? Why can't they
understand *us*?
> We're a minority group and I think the onus is on us to convince RME
> to produce a driver for their firewire hardware, politely and if necessary,
> via the florists ;-). OK, so closed-source drivers are far from ideal, but
> better than a hole in the head.
http://www.audioscience.com
If they can, who can't? I can't see the difference, can anyone explain?
>
> It means that the drivers can't be bundled with distros and we won't be able
> to provide users & developers with technical support, which is a great shame.
>
> However, I suspect a certain amount of well-reasoned persistence will pay off
> here. Sure, our numbers on this list aren't great, but they are significant.
There are many audio hw customers outside of this list (see CK's post
for example, or judging form experience - somewhere on #gnome talking
about rme ;) plus tons of talks on #lad - Q: "hi, what's the best card
for audio under linux? A: "rme or if you don't have that much money,
maudio")
>
> OK, _very_ few people are using firewire technology for music, up till now I'd
> considered it the preserve of mac/motu users.
I think a majority of pc based audio hw will be fw based in the near
future. Every manufacturer will have at least one product. Scary.
> I think we should continue to
> support RME where licenses allow and look forward to the day that they
> release their firewire drivers :-).
That is going to be the day their hw becomes redundant on the market? Or
even discontinued? That's the problem i'm seeing.
> I think we should keep up the pressure on
> manufacturers like MOTU too. They'll see sense eventually. ;-]
I doubt it. They have their own sw products, like the DP. In their case
i can pretty much understand why they don't do that if they see linux
audio as a competition.
>
> Mine is an equally naive viewpoint, but with the knowledge that a little bit
> of positive thinking can go a long way, especially when backed up with a
> well-researched wish-list and plenty of patience.
2 years korg and now this. Trust me it's not possible to cope with that
for a long time :)
Marek
On Sat, 2004-11-27 at 11:06, MarC wrote:
> what about creating a wiki website to submit "soundcard experiences"?
This also seems like a good idea to me, and it would be cool to have a
knowledgebase like that.
However i think that doing a survey in order to measure how big the
linux audio market is is a different project.
Similar to what David did, i imagine that a RME (or M-Audio) customer
would submit his name, optionally email, choose type of his card from a
list, and enter approx. date of purchase so that we can track the growth
of the market(thus it's possible to roughly estimate its future growth).
Also we'd need to announce it to a broad range of people, LAD/LAU/LAA,
ALSA-dev/user/site, and Slashdot i suppose, so that we reach as many
such customers as possible.
Let's also not forget about OSS users which are also linux users.
(I personally reckon a few people from #lad having troubles with alsa,
they switched to oss for that reason).
However, since i've got no php experience and very little mysql
experience, i can't tell how hard it would be to do such a sumbit/log
system.
Marek
On Thu, 2004-11-25 at 20:50, Florin Andrei wrote:
> On Thu, 2004-11-25 at 04:22 -0500, Rick B wrote:
>
> > I kind of got the impression that the annoucement was just pertaining to
> > RME *Firewire* audio interfaces.
Consider that they have released some specs for their HDSP hammerfall
series, which uses a *proprietary* firewire protocol and that their
latest PC products were based on IEEE1394 except one or two PCI based
cards.
>
> That's what i thought. "RME is no more" seems a bit exagerated (although
> i feel for the person who bought the card thinking it's supported by the
> Linux drivers).
I don't think it's exagerated, see explanation above.
I knew exactly it wasn't at the time i bought it, i just took it for
granted. I talked to Thomas Charbonnel back in april at the ZKM and it
seemed that they were positive about alsa support for fireface.
>
> Anyway, beyond Linux support tribulations, the RME Fireface is a great
> card. I just read a review in the international Dec 2004 edition of
> Sound On Sound - it's really cool. It has all the things that i wish the
> Multiface had.
I can only agree with that. But that's even worse for us then. ;)
>
> Sadly, if there's no support for Linux, i guess i won't buy it. It's not
> like the world ends with RME or anything.
Well it's close to such situation in the linux pro-audio world. The two
major players in pro-audio hw market that supported ALSA development if
only indirectly by providing specs, were m-audio and... rme.
Have a look at the ALSA matrix, it's a pretty sad situation.
The only *real* hw manufacturer in my eyes is audioscience, they provide
their own ALSA drivers(that's how it should be) but produce only
broadcast cards.
Marek