freem hormbleen | 5 Mar 2009 02:26
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static when using mpg123 in to lame out

Any idea why this might happen?  The input file is this:
http://www.earnityourself.com/input.mp3

when I run

/usr/local/bin/ecasound -i input.mp3  -o static.mp3

i get

http://www.earnityourself.com/static.mp3

it sounds like it's not random data - the beats sort of match the original, but it's full of static and messed up.

and my ecasoundrc file looks like the below

ecasound-version = 2.5.2
midi-device = rawmidi,/dev/midi
default-output = autodetect
default-audio-format = s16_le,2,44100,i
default-to-precise-sample-rates = false
default-to-interactive-mode = false
default-mix-mode = avg
bmode-defaults-nonrt = 1024,false,50,false,100000,true
bmode-defaults-rt = 1024,true,50,true,100000,true
bmode-defaults-rtlowlatency = 256,true,50,true,100000,false
resource-directory = /usr/local/share/ecasound
resource-file-genosc-envelopes = generic_oscillators
resource-file-effect-presets = effect_presets
ladspa-plugin-directory = /usr/local/lib/ladspa
ext-cmd-text-editor = pico
ext-cmd-text-editor-use-getenv = true
ext-cmd-wave-editor = ecawave
ext-cmd-mp3-input = /usr/local/bin/mpg123 -b 0 -q -s  %f
#ext-cmd-mp3-input = /home/ernst/mpg123-1.6.2/src/mpg123 --stereo -r %s -b 0 -q -s -k %o %f
ext-cmd-mp3-output = /usr/local/bin/lame -r -b %B -s %S -x - %f
ext-cmd-ogg-input = ogg123 -d raw -o byteorder:%E --file=- %f
ext-cmd-ogg-output = oggenc -b %B --raw --raw-bits=%b --raw-chan=%c --raw-rate=%s --raw-endianness 0 --output=%f -
ext-cmd-mikmod = mikmod -d stdout -o 16s -q -f %s -p 0 --noloops %f
ext-cmd-timidity = timidity -Or1S -id -s %s -o - %f
ext-cmd-flac-input = flac -d -c %f
ext-cmd-flac-output = flac -o %f -f --force-raw-format --channels=%c --bps=%b --sample-rate=%s --sign=%I --endian=%E -
ext-cmd-aac-input = faad -w -b 1 -f 2 -d %f
ext-cmd-aac-output = faac -P -o %f -R %s -B %b -C %c -

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Lance Hoffmeyer | 6 Mar 2009 22:36

delta66, chains, and ecasound

Hey all,

Wondering if anyone has any experience with delta66 and ecasound?

I have two mics plugged into the delta control box.  Usually I just simply
type ecasound -i alsa,plughw:2 -o file.wav and start playing. (Delta66 
is the
second sound card on my system, hence the plughw:2).

Now, I am wanting to record with another person and am wondering how
I chain the two mics (I think chain is the term I want to use) so that I 
have
a chain for each mic and can then go back at some point and manipulate
each mic seperately (volume, pan ...)?

Anyone have experience with this?

Thanks in advance,

Lance

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Julien Claassen | 7 Mar 2009 15:07
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Re: delta66, chains, and ecasound

Hello Lance!
   I'musing a delta 1010 LT and in the old days I worked with ALSA. But I 
abandonned that a long time ago. Now I use JACK, which is highly preferable, 
especially with these big cards.
   If you need to stay with ALSA, you could use the ~/.asoundrc or 
/etc/asound.conf files to achieve, what you want. I used ttable and dshare (or 
dsnoop) not sure. Youcan find more about them at:
http://www.alsa-project.org
   There's a wiki.
   If on the other hand you could live with JACK, get back here and I'm sure, I 
won't be the only one able and willing to help. It's not too dificult.
   Kindest regards
          Julien

--------
Music was my first love and it will be my last (John Miles)

======== FIND MY WEB-PROJECT AT: ========
http://ltsb.sourceforge.net
the Linux TextBased Studio guide
======= AND MY PERSONAL PAGES AT: =======
http://www.juliencoder.de

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Joel Roth | 7 Mar 2009 19:13
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Re: delta66, chains, and ecasound

On Fri, Mar 06, 2009 at 03:36:02PM -0600, Lance Hoffmeyer wrote:
> Hey all,
> 
> Wondering if anyone has any experience with delta66 and ecasound?
> 
> I have two mics plugged into the delta control box.  Usually I just simply
> type ecasound -i alsa,plughw:2 -o file.wav and start playing. (Delta66 
> is the
> second sound card on my system, hence the plughw:2).
> 
> Now, I am wanting to record with another person and am wondering how
> I chain the two mics (I think chain is the term I want to use) so that I 
> have a chain for each mic and can then go back at some point and manipulate
> each mic seperately (volume, pan ...)?
> 
> Anyone have experience with this?
> 
> Thanks in advance,
> 
> Lance

Hi Lance,

Whether you access the device through JACK or ALSA, 
it is possible to record two or more channels of sound
from Ecasound, and later apply volume and pan effects
to the playback.

You can do this using Ecasound's interactive command
language (by starting Ecasound with the -c flag),
command-line syntax, or chain setup files. You can also use
a high-level  application like Tkeca or Nama to tickle
Ecasound as appropriate. 

Regards,

--

-- 
Joel Roth

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Lance | 8 Mar 2009 07:29

Re: delta66, chains, and ecasound

Decided to go ahead and try to use jack instead of/in addition to alsa. Jack will only allow me to use alsa,hw and not alsa,plughw with the Delta 66. I don't really know the diff between hw and plughw (I did read up on this, see bottom of email but descriptoin of diff) but I have always used plughw with ecasound. So, I guess my first step is to get the Delta 66 to work with alsa,hw2 as well as alsa,plughw:2. What should I try/do so that I can play sound files using alsa,hw2 using ecasound? Lance $ecasound -i Rag.mp3 -o alsa,hw:2 ******************************************************************************** * ecasound v2.4.6.1 (C) 1997-2007 Kai Vehmanen and others ******************************************************************************** - [ Session created ] ---------------------------------------------------------- - [ Chainsetup created (cmdline) ] --------------------------------------------- - [ Connecting chainsetup ] ---------------------------------------------------- (eca-chainsetup) 'rt' buffering mode selected. (eca-chainsetup) Audio object "Rag.mp3", mode "read". (audio-io) Format: s16_le, channels 2, srate 44100, interleaved. ERROR: Connecting chainsetup failed: "Enabling chainsetup: AUDIOIO-ALSA: Audio ... format not supported." $ecasound -i Rag.mp3 -o alsa,plughw:2 ******************************************************************************** * ecasound v2.4.6.1 (C) 1997-2007 Kai Vehmanen and others ******************************************************************************** - [ Session created ] ---------------------------------------------------------- - [ Chainsetup created (cmdline) ] --------------------------------------------- - [ Connecting chainsetup ] ---------------------------------------------------- (eca-chainsetup) 'rt' buffering mode selected. (eca-chainsetup) Audio object "Rag.mp3", mode "read". (audio-io) Format: s16_le, channels 2, srate 44100, interleaved. (eca-chainsetup) Audio object "alsa", mode "write". (audio-io) Format: s16_le, channels 2, srate 44100, interleaved. - [ Chainsetup connected ] ----------------------------------------------------- (eca-control-objects) Connected chainsetup: "command-line-setup". - [ Controller/Starting batch processing ] ------------------------------------- - [ Engine init - Driver start ] ----------------------------------------------- ecasound -i foo.wav -o jack,remote_client







Julien Claassen wrote:
Hello Lance! I'musing a delta 1010 LT and in the old days I worked with ALSA. But I abandonned that a long time ago. Now I use JACK, which is highly preferable, especially with these big cards. If you need to stay with ALSA, you could use the ~/.asoundrc or /etc/asound.conf files to achieve, what you want. I used ttable and dshare (or dsnoop) not sure. Youcan find more about them at: http://www.alsa-project.org There's a wiki. If on the other hand you could live with JACK, get back here and I'm sure, I won't be the only one able and willing to help. It's not too dificult. Kindest regards Julien -------- Music was my first love and it will be my last (John Miles) ======== FIND MY WEB-PROJECT AT: ======== http://ltsb.sourceforge.net the Linux TextBased Studio guide ======= AND MY PERSONAL PAGES AT: ======= http://www.juliencoder.de ------------------------------------------------------------------------------ Open Source Business Conference (OSBC), March 24-25, 2009, San Francisco, CA -OSBC tackles the biggest issue in open source: Open Sourcing the Enterprise -Strategies to boost innovation and cut costs with open source participation -Receive a $600 discount off the registration fee with the source code: SFAD http://p.sf.net/sfu/XcvMzF8H _______________________________________________ Ecasound-list mailing list Ecasound-list <at> lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/ecasound-list







Re: [Alsa-user] "hw" or "plughw"

stan
Tue, 20 Nov 2007 09:21:42 -0800

Vladimir Mosgalin wrote: > Hi paul blakeley! > > On 2007.11.20 at 12:26:29 +0000, paul blakeley wrote next: > > >> Can someone please explain the differences between these? What impact >> they have on the application? >> > > plughw supports much more sample formats / channel configurations > than underlying hardware supports natively, and performs conversion if > needed. hw performs no conversion, but supports less configuration, > sometimes only very obscure ones, but when used you can rest assured > that no conversion takes place. > > Mostly you'd want these conversions to take place, like mono->stereo > conversion or S16LE->S32LE conversion etc (all depending on your > hardware) > > >> If I need to drive the sound card directly should I use 'hw'? >> > > You can use hw, but it isn't really recommended unless you REALLY know > how to use it and have support of every weird format in all possible > combinations. That's about underlying hardware details, and most > applications don't want to deal with them. > > For example my soundcard supports only S24_3BE format; you can't open hw > device in any other mode, if you want to output S16LE (most applications > never heard of S24_3 formats, let alone BE variations), you must use > plughw, there is no other choice. Or p16v device on audigy2 supports > only 8-channel modes; you can't output stereo signal to it, no matter > how you try. So you either can output 8 channels to hw device or let > plughw to do stereo->8ch conversion for you automatically. > > Unless you want to take care of all these little details, using hw is > probably not a good idea. Though it's required for some applications > because you don't actually know if/what kind of conversion takes place > when you use plughw, most application would trouble users much less if > they were to use plughw instead of hw. Actually, almost all application > shouldn't even use plughw, sticking to "default" device, to allow > software mixing, jack/pulse routing plugins, user choosen conversions to > take place. If you do anything else, you create problems for users, so > you must have really good reasons to do so.. > > Valadimir, I agree with what you say, but if you want to use the exotic features of alsa to control the sound device you have to use plughw: instead of default. default uses dmix which always uses 48000 frame rate. So if you want to use a card at say, 96000 frame rate, you can't do it with default but can do it with plughw. The sound servers don't implement this exotic alsa functionality (they are 'dumb' regarding alsa). For casual use, great because people aren't interested in sound device performance, they just want to hear sound, and want it to be easy to manipulate. For average mp3 use, sound quality is irrelevant, for anything else, not good. To the original poster, you can of course control the hardware directly using hw: as well, but you are responsible for all the details as Vladimir pointed out above. Usually what you want is to avoid software mixing by using a frame rate supported by the hardware. With plughw: you can do this and let alsa take care of format conversion for you. With hw: you have to do the format conversion to the card's internal format. If you run on more than one card or expect to run in the wild, this means you duplicate the functionality of alsa for format conversion. Counterproductive. On the other hand, if you are building an embedded device with a single sound interface always, hw: might be the way to go.
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Julien Claassen | 8 Mar 2009 12:00
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Re: delta66, chains, and ecasound

Hello Lance!
   If you use ALSA, you don't want to use hw, you DEFINITELY WANT plughw. 
Plughw does some convenience conversions and detections(?) for you. Like 
sampling rate and the like.
   With jack use something like:
jackd --timeout 4500 -R -d alsa -d hw:2 -r 44100 -z shaped -p 128
   If you like 48000 Hz samplingrate, jst say so, where 44100 is written now. 
This will do exactly what you want.
   Then you can use:
tty# jack_lsp
   To see the ports (ins and outs). They have names in the format:
client:port_name
   The client is the name of the program (system or alsa_pcm for the soundcard 
ports) and port_name is the name of a "jack" where you can plug in.
   Assume you've started ecasound:
ecasound -f:16,1,44100 -a:1 -i jack -o mic1.wav -a:2 -i jack -o mic2.wav -c
[copious output]
ecasound ('h' for help)> engine-launch
[change console, tty]

tty# jack_lsp
   Now you should see:
system:capture_1
...
system:playback_1
...
ecasound:in_1
ecasound:in_2
   Now you want to connect your mics:
tty# jack_connect system:capture_1 ecasound:in_1
tty# jack_connect system:capture_2 ecasound:in_2
   Then press 't' in ecasound and press 's' when you want to stop. See the 
small help 'h' inside ecasound for some basic commands.
   The same goes for outputs. You can just connect them the same way. Always be 
sure to connect them like this:
jack_connect sound_producer sound_consumer
   Ecasound still supports the object jack_alsa. So to play some sound to the 
first - who knows how many outputs you chose in -f - use:
ecasound [options] -i input -o jack_alsa
   No need to connect anything, ecasound does it for you. There's also a new 
jack object. But I keep forgetting, I use nama, which is pretty good and 
helpful for doing good old fashioned multitrack recording and processing, 
mixing.
   Very long winded, for which I appologise, but hopefully helpful.
   Kindest regards
         Julien

--------
Music was my first love and it will be my last (John Miles)

======== FIND MY WEB-PROJECT AT: ========
http://ltsb.sourceforge.net
the Linux TextBased Studio guide
======= AND MY PERSONAL PAGES AT: =======
http://www.juliencoder.de

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Julien Claassen | 8 Mar 2009 12:02
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Re: delta66, chains, and ecasound

Sorry, one small addition:
use JACK instead of ALSA. JACK will use you soundcard (hw:2) exclussively. And 
you don't want/need any more ALSA in the same picture. Just trust in that for 
the moment.
   Warm regards
         Julien

--------
Music was my first love and it will be my last (John Miles)

======== FIND MY WEB-PROJECT AT: ========
http://ltsb.sourceforge.net
the Linux TextBased Studio guide
======= AND MY PERSONAL PAGES AT: =======
http://www.juliencoder.de

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Lance | 8 Mar 2009 17:50

Re: delta66, chains, and ecasound

Not long winded at all! Perfect amount of info and got 
jack+ICE1712+ecasound working with 2 mics

A couple of followup questions:

1) jack seems to be set root:root so that I have to use sudo jackd ...
to start jack and hence, I also have to run ecasound via sudo ecasound.
I would prefer to run these as a user instead of sudo.  How do
I change jack configuration to run as user?

2 No idea what
jack_connect sound_producer sound_consumer means.  I have seen something 
similar
to this in envy24 (Hardware settings, Professional vs Consumer, is it 
the same thing?)

3) Since I will be using this configuration (or something similar) 
whenever I run
ecasound and I am wonder if it is possible to configure all of this into the
.ecasoundrc file so that when I type:

ecasound -f:16,1,44100 -a:1 -i jack -o mic1.wav -a:2 -i jack -o mic2.wav -c

everything is ready to go.  No need to type engine-launch, jack_connect 
system:capture_1 ecasound:in_1 ...

Could it be as simple as?

.ecasoundrc
jackd --timeout 4500 -R -d alsa -d hw:2 -r 44100 -z shaped -p 128
engine-launch
jack_connect system:capture_1 ecasound:in_1
jack_connect system:capture_2 ecasound:in_2
jack_connect system:playback_1 ecasound:out_1
jack_connect system:playback_2 ecasound:out_2

A summary for any future reference

To get Delta66+ecasound to work one can use

ecasound -i alsa,plughw:1 -o foobar.wav (record using delta66)

but one doesn't get a lot of control over two mics without additional
configuration to a .asoundrc file in one's home directory.

This requires three programs:
envy24
alsa
ecasound

The approach we are using to get chain control with 2+ mics uses 4 programs
envy24
alsa
jack
ecasound

Julien Claassen wrote:
> Hello Lance!
>   If you use ALSA, you don't want to use hw, you DEFINITELY WANT 
> plughw. Plughw does some convenience conversions and detections(?) for 
> you. Like sampling rate and the like.
>   With jack use something like:
> jackd --timeout 4500 -R -d alsa -d hw:2 -r 44100 -z shaped -p 128
>   If you like 48000 Hz samplingrate, jst say so, where 44100 is 
> written now. This will do exactly what you want.
>   Then you can use:
> tty# jack_lsp
>   To see the ports (ins and outs). They have names in the format:
> client:port_name
>   The client is the name of the program (system or alsa_pcm for the 
> soundcard ports) and port_name is the name of a "jack" where you can 
> plug in.
>   Assume you've started ecasound:
> ecasound -f:16,1,44100 -a:1 -i jack -o mic1.wav -a:2 -i jack -o 
> mic2.wav -c
> [copious output]
> ecasound ('h' for help)> engine-launch
> [change console, tty]
>
> tty# jack_lsp
>   Now you should see:
> system:capture_1
> ...
> system:playback_1
> ...
> ecasound:in_1
> ecasound:in_2
>   Now you want to connect your mics:
> tty# jack_connect system:capture_1 ecasound:in_1
> tty# jack_connect system:capture_2 ecasound:in_2
>   Then press 't' in ecasound and press 's' when you want to stop. See 
> the small help 'h' inside ecasound for some basic commands.
>   The same goes for outputs. You can just connect them the same way. 
> Always be sure to connect them like this:
> jack_connect sound_producer sound_consumer
>   Ecasound still supports the object jack_alsa. So to play some sound 
> to the first - who knows how many outputs you chose in -f - use:
> ecasound [options] -i input -o jack_alsa
>   No need to connect anything, ecasound does it for you. There's also 
> a new jack object. But I keep forgetting, I use nama, which is pretty 
> good and helpful for doing good old fashioned multitrack recording and 
> processing, mixing.
>   Very long winded, for which I appologise, but hopefully helpful.
>   Kindest regards
>         Julien
>
> --------
> Music was my first love and it will be my last (John Miles)
>
> ======== FIND MY WEB-PROJECT AT: ========
> http://ltsb.sourceforge.net
> the Linux TextBased Studio guide
> ======= AND MY PERSONAL PAGES AT: =======
> http://www.juliencoder.de

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Julien Claassen | 9 Mar 2009 10:46
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Re: delta66, chains, and ecasound

Hello Lance!
   Unfortunitely no answer to the first question. I alwasy run my audio stuff 
as root.
2. No it has nothing to do with professional and consumer. Let's see:
Your soundcard input (the microphone Jack in question) gets audio from your 
microphone. Right? So what does the system:capture_1 port do: It offers you 
the audio from the microphone. Still following? So for the computer software 
site system:capture_1 produces sound. That's a sound_producer in my example. 
Jack_connect's help sasy:
jack_connect <source> <destination>
   But back to my example: What does ecasound:in_1 do? It reads audio. It sits 
there and waits for audio to come in, so ecasound can write it to disk, run it 
through effects or simpl play it back for you. So it consumes the audio, that 
- system:capture_1 in our example - produced. So I called it consumer. You 
could imagine, that instead of system:capture_1 we could take the output of a 
virtual synthesizer, like
fluidsynth:right
   or an mplayer output:
MPlayer [5392]:out_1
   It would all do the same.
   The only thing that had me confused a bit in the early beginnings was, that 
for the soundcard the producers (outputs) are called "capture" and the 
consumers (the inputs) are called "playback". All other JACK_clients had it 
the right way round. Why? Simple if you think about it: the soundcard port 
name are chosen from the outside perspective (yours as the user), and you sit 
in the real world, with which the computer doesn't bother. The rest is correct 
for outside and inside perspective, because, it's all inside the computer. So 
you both have the same view.

3. No there's no way to put this into ecasoundrc. You could write a small 
shell script like this:

#!/bin/sh
ecasound -c -a:2 -i jack_multi,system:capture_1 -o mic1.wav -a:2 -i \
jack_multi,system:capture_2 -o mic2.wav

   The jack_multi syntax was, what I was looking for last time. You could also 
do:
ecasound -c -i play.wav -o jack_multi,system:playback_3,system:playback_4
   So you can decide (based on your recording format)
a) Which jack_port to connect to
b) If more then mono, which port(s) to connect to the next ecasound ins or 
outs.
   Meaning, your not bound to use: system:playback_1 and system:playback_2 for 
stereo anymore, but system:playback_1 and system:playback_4.
   Btw.: I just remembered an alsa-way of doing it, but still not prefferable 
(at least not to me):
ecasound -f:16,12,48000 -a:in -i alsahw,2 -o loop,1 -a:2 -i 
loop,1 -cymove:2,1 -f:16,1,48000 -o mic2.wav
   I don't know, how many times you should specify the -f-options and if I 
placed the chmove in the correct spot. It's the way nama does it internally.

   Summeray: envy24control is just there to configure your card (volumes, 
muting and routing) so we don't count it. :-)
   Besides that: Yes, your old way you have : ALSA and ecasound
   The new way you have: ALSA (invisible) JACK and ecasound.
   But ALSA will always be there, because it's your soundcard driver. JACK is 
just something to be on top of a driver. It guarantees synchronous audio 
(especially important, if you start using other linux audio software with 
ecasound), low latency (if you set it to do so) and it has a flexible system 
for connecting pieces of software and hardware. My startup command has good 
settings for lowlatency. If you want to optimise ecasound's behaviour you 
could include these options in a script:

#!/bin/sh
ecasound -B:rtlowlatency -b:64 -r 30 -z:mixmode,sum -c $ <at> 

   What this does: hand over all these options to ecasound and then there's the 
$ <at> , which means after that take all the other option a user gives.
   Assume this script is stored in a file called rteca . Then you would do:
tty# rteca -a:1 -i [and so on]
   It would start ecasound with your chainsetup and the lowlatency options.
   Hope that helps!
   Kindest regards
          Julien

--------
Music was my first love and it will be my last (John Miles)

======== FIND MY WEB-PROJECT AT: ========
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William Goldsmith | 9 Mar 2009 19:15
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Ecasound with no hardware?

Hi there Ecasound wizards,

I've been using Ecasound for over 10 years now as part of a radio  
automation system. I use it as a wrapper for the playback of MP3  
files, executing fade-ins and fade-outs on the files. Each file  
playback event launches a new instance of ecasound, so that during  
song transitions -- or when DJ breaks are overlaid over the beginning  
or end of a song -- two, or sometimes three, instances of ecasound  
will be playing simultaneously. I use OSS sound card drivers (or   
OSS's vmixer utility) to accept multiple sources and sum them for  
output via a s/pdif interface.

This all works fine,  but in some cases I don't export the audio at  
all, but rather feed it directly to an MP3 stream encoder like ices or  
sc_trans. In that case, I'm using the sound card & sound drivers  
basically as a mount point & summing network.  There *must* be some  
way I could play back multiple files to a single output, and pipe that  
output to the input of a stream encoder, without needing the audio  
hardware.

Any thoughts?

thanks
-bg
-----
Bill Goldsmith
Radio Paradise
www.radioparadise.com

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Open Source Business Conference (OSBC), March 24-25, 2009, San Francisco, CA
-OSBC tackles the biggest issue in open source: Open Sourcing the Enterprise
-Strategies to boost innovation and cut costs with open source participation
-Receive a $600 discount off the registration fee with the source code: SFAD
http://p.sf.net/sfu/XcvMzF8H
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Gmane