Re: [Alsa-user] Capture w/ M-Audio Fast Track USB?

2009-06-02 Thread Clemens Ladisch
Michael B Allen wrote:
 Does capture work with the M-Audio Fast Track USB?

Maybe.  Does some capture device show up in the output of arecord -l?

 Can someone give me some advice or a pointer to some documentation
 that explains how the whole ALSA tool chain works wrt external USB
 sound recording?

You can tell any ALSA program to use a specific device by using a device
name like plughw:x,y if you know the card and device numbers.

If you're using PulseAudio, you have to tell it to use that device.


HTH
Clemens

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Re: [Alsa-user] alsa-utils-1.0.20.12.g730e compile problem

2009-06-02 Thread Clemens Ladisch
Raena wrote:
 I also installed the snapshot alsa-lib and that was fine but alsa-utils 
 would not configure. configure: error: panelw library not found I have 
 not been able to find out what package has this library. Here is my 
 configure output;
 ...
 checking for ncursesw5-config... yes 

This indicates that ncursesw is installed ...

 checking for new_panel in -lpanelw... no
 configure: error: panelw library not found

... but actually it isn't.

What is the output of the following commands?

  ncursesw5-config --libs
  ncursesw5-config --libdir
  locate libncurses.so
  locate libncursesw.so
  locate libpanel.so
  locate libpanelw.so


Best regards,
Clemens

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Re: [Alsa-user] alsa-utils-1.0.20.12.g730e compile problem

2009-06-02 Thread Raena Lea-Shannon
Is there another ncurses-devel I have missed?


Raena wrote:
 I have libncurses5devel installed
 
 
 Geoffrey Leach wrote:
 Try ncurses-devel

 On 05/29/2009 09:21:04 PM, Raena wrote:
 Hi I am running 2.6.28-11-generic with ASUS M4A78T-E MOBO. I needed 
 to

 install the snapshot drivers to get my digital sound working and that 
 worked very well thanks a bunch.

 I also installed the snapshot alsa-lib and that was fine but
 alsa-utils 
 would not configure. configure: error: panelw library not found I 
 have

 not been able to find out what package has this library. Here is my 
 configure output;


 ra...@kubuntu:/usr/src/alsa/alsa-utils-1.0.20.12.g730e8$ sudo 
 ./configure
 [sudo] password for raena: 

 checking for a BSD-compatible install... /usr/bin/install -c 

 checking whether build environment is sane... yes 

 checking for gawk... no 

 checking for mawk... mawk 

 checking whether make sets $(MAKE)... yes 

 checking whether NLS is requested... yes 

 checking for msgfmt... /usr/bin/msgfmt 

 checking for gmsgfmt... /usr/bin/msgfmt 

 checking for xgettext... /usr/bin/xgettext 

 checking for msgmerge... /usr/bin/msgmerge 

 checking for style of include used by make... GNU 

 checking for gcc... gcc 

 checking for C compiler default output file name... a.out 

 checking whether the C compiler works... yes 

 checking whether we are cross compiling... no 

 checking for suffix of executables... 

 checking for suffix of object files... o 

 checking whether we are using the GNU C compiler... yes 

 checking whether gcc accepts -g... yes 

 checking for gcc option to accept ISO C89... none needed 

 checking dependency style of gcc... gcc3 

 checking build system type... x86_64-unknown-linux-gnu 

 checking host system type... x86_64-unknown-linux-gnu 

 checking for ld used by GCC... /usr/bin/ld 

 checking if the linker (/usr/bin/ld) is GNU ld... yes 

 checking for shared library run path origin... done 

 checking for CFPreferencesCopyAppValue... no 

 checking for CFLocaleCopyCurrent... no 

 checking for GNU gettext in libc... yes 

 checking whether to use NLS... yes 

 checking where the gettext function comes from... libc 

 checking for cross-compiler... gcc 

 checking for gcc... (cached) gcc 

 checking whether we are using the GNU C compiler... (cached) yes 

 checking whether gcc accepts -g... (cached) yes 

 checking for gcc option to accept ISO C89... (cached) none needed 

 checking dependency style of gcc... (cached) gcc3 

 checking for a BSD-compatible install... /usr/bin/install -c 

 checking whether ln -s works... yes 

 checking for ALSA CFLAGS... 

 checking for ALSA LDFLAGS...  -lasound -lm -ldl -lpthread 

 checking for libasound headers version = 1.0.16... found. 

 checking for snd_ctl_open in -lasound... yes 

 checking for alsa/pcm.h... yes 

 checking for alsa/mixer.h... yes 

 checking for alsa/rawmidi.h... yes 

 checking for alsa/seq.h... yes 

 checking for librt... checking for clock_gettime in -lrt... yes 

 checking for xmlto... yes 

 checking how to run the C preprocessor... gcc -E 

 checking for grep that handles long lines and -e... /bin/grep 

 checking for egrep... /bin/grep -E 

 checking for ANSI C header files... yes 

 checking for ncursesw5-config... yes 

 checking for curses library... ncursesw
 checking for curses header name... ncurses.h
 checking for curses compiler flags... -I/usr/include/ncursesw
 checking for curses NLS support... yes
 checking for sys/types.h... yes
 checking for sys/stat.h... yes
 checking for stdlib.h... yes
 checking for string.h... yes
 checking for memory.h... yes
 checking for strings.h... yes
 checking for inttypes.h... yes
 checking for stdint.h... yes
 checking for unistd.h... yes
 checking panel.h usability... yes
 checking panel.h presence... yes
 checking for panel.h... yes
 checking menu.h usability... yes
 checking menu.h presence... yes
 checking for menu.h... yes
 checking form.h usability... yes
 checking form.h presence... yes
 checking for form.h... yes
 checking for new_panel in -lpanelw... no
 configure: error: panelw library not found
 ra...@kubuntu:/usr/src/alsa/alsa-utils-1.0.20.12.g730e8$

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[Alsa-user] ymfpci drive capability - help needed with setup

2009-06-02 Thread robert crowther
Last night I tried an experiment. I took a second computer, installed a
Yamaha WF-192XG card, then loaded the i386 version of 64Studio on top.
To my shock (I'm still shocked) Rosegarden worked first time - very
promising.

Rosegarden seemed to be using Timidity for it's sound conversion. But
the WF-192XG, of course, has a wavetable on board. Not just any
wavetable, an XG wavetable with effects and Soundius on board. Much as I
love Timidity (and I love Timidity) I'd like to use the wavetable too,
but it wasn't showing.  I looked in proc, tried amidi -l, and tried
enabling in etcmodprobe.dalsa-base (though I don't know what I'm doing
there), but nothing showing. The only hint is in alsamixer, which has a
couple of unexplained audio outputs?

I searched the mail list, but found no reference to anything like this,
so I'm posting...

Questions,

Can ymfpci access the WF-192XG wavetable? 
(I tried looking in some recent source code, and though this was only a
brief glance, there did seem to be references to XG and XG mapping?)

If so, how would the wavetable appear?
(I assumed I'd have a labelled midi input, though by what name, I don't
know)

If the driver can access the wavetable, does anyone have any clues, or
examples, of how I would set up the module to make this happen?
(Ok, this is implementation specific, but I know nothing of modules and
modprobing, so something to work on would help)


Regards, and thanks in advance for any help,


Rob


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[Alsa-user] MOTU micro lite (USB, 5in, 5out)

2009-06-02 Thread Aidan Dixon
Hi,

I have a MOTU micro lite (USB, 5x MIDI IN and 5x MIDI OUT, Vendor ID
0x07fd, product ID 0x0001).  According to the kernel usb-audio device
driver source code, this device should be supported as its USB vendor
and product ID are listed in a table in linux/sound/usb/usbmidi.c.  Yet
the driver seems to take no notice of the deivce - even if I give the
vendor/product IDs on the modprobe command line. lsusb identifies it as
a Fastlane device, although I see from previous posts (one from 2006:
http://www.mail-archive.com/alsa-user@lists.sourceforge.net/msg15907.html)
that this is believed to be a 2in/2out device unlike mine.

Any ideas if this can/should work and how to do so?  What are the issues
beyond simply lack of information from MOTU?  Sadly, I also just noticed
(http://qbik.ch/usb/devices/showdev.php?id=3172) that apparently this
hasn't worked for sometime now.

Kind regards
Aidan Dixon


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[Alsa-user] Fragment size

2009-06-02 Thread Dennis Borgmann
Hello List-users!

I have been dealing with this for quite some time now and I cannot get
to the point.

I have a rather simple code of maybe 100 lines which just writes some
buffer to the soundcard using snd_pcm_writei(). Although the buffer is
just there within the program itself (no external source, just internal
in order to understand the code), I am running into underruns.

I don't see the point - how ca I run into underruns, if data is always
there?

In addition I would like to set the fragment count of the driver - I
think this term is derived from OSS. I would like to have at least three
fragments of each 256 frames, but I do not know how to set this. The
doxygen documentation is not a real good source I must admit...

Could anyone please help me?

Thanks in advance
Dennis

I attached my testcode.
#include includes.h
#include defines.h

struct globals global;

int main( int argc, char ** argv) {
	int transmitter_decision = 0;	/* simple flag for bool choice of server */
	snd_pcm_t *sound_handler;
	struct sched_param scheduler_parameter;
	char read_data[2048];
	char *pointer;
	int written_frames;
	int len=256;
	struct timeval time2,time3;
	scheduler_parameter.sched_priority = SCHEDULER_PRIORITY;
	sched_setscheduler(0, SCHEDULER_POLICY, scheduler_parameter);
	snd_pcm_stream_t stream;
	snd_pcm_hw_params_t *hwparams;
	snd_pcm_sw_params_t *swparams;
	char *pcm_name;
	unsigned int exact_rate;
	snd_pcm_access_t accesstype = ACCESS;
	snd_pcm_format_t format = FORMAT;
	snd_pcm_uframes_t buffersize = BUFFERSIZE_FRAMES;
	unsigned int periods = PERIODS_PER_INTERRUPT;
	int dir, err;
	snd_pcm_uframes_t val;
	snd_pcm_t *pcm_handle;

	memset( read_data, 0xff, sizeof( read_data ) );

	if (argc  2) {
		printf(usage: '%s SERVER-IP' for client or '%s 1' for server \n,argv[0],argv[0]);
		exit( -1 );
	}
	if( memset( global, 0, sizeof(struct globals) ) == NULL ) perror(memset);
	if( strncmp(argv[1],1,1) == 0 ) transmitter_decision = 1;

	pcm_name = strdup(hw:0,0);
	snd_pcm_hw_params_alloca( hwparams );

	stream = SND_PCM_STREAM_PLAYBACK;
	if( snd_pcm_open( pcm_handle, pcm_name, stream, 0 ) != 0 ) {
		fprintf(stderr, Error opening PCM device %s\n, pcm_name);
		return(-1);
	}
	if( snd_pcm_hw_params_any( pcm_handle, hwparams ) != 0 ) {
		fprintf(stderr, Cannot configure PCM device\n);
		return(-1);
	}
	/** Setting the previously tested capabilities */
	if( snd_pcm_hw_params_set_access( pcm_handle, hwparams, accesstype ) != 0 ) {
		fprintf(stderr, SET: Error setting access\n);
		return(-1);
	}
	if( snd_pcm_hw_params_set_format( pcm_handle, hwparams, format ) != 0 ) {
		fprintf(stderr, SET: Error setting format\n);
		return(-1);
	}
	exact_rate = RATE;
	dir = -1;
	if( snd_pcm_hw_params_set_rate_near( pcm_handle, hwparams, exact_rate, dir ) != 0 ) {
		fprintf(stderr, SET: Error setting rate %d\n, dir);
		return(-1);
	}
	if( RATE != exact_rate ) {
		fprintf(stderr, The rate is not supported by your hardware.\n \
		ATTENTION! Alsa-resampler being used!!! == Using %d_Hz instead of %d_Hz.\n, exact_rate, RATE);
	}
	if( snd_pcm_hw_params_set_channels( pcm_handle, hwparams, CHANNELS ) != 0 ) {
		fprintf(stderr, SET: Error setting channels\n);
		return(-1);
	}
	if( snd_pcm_hw_params_set_periods_near( pcm_handle, hwparams, periods, 0 ) != 0 ) {
		fprintf(stderr, SET: Error setting periods\n);
		return(-1);
	}
	// Set buffer size (in frames). The resulting latency is given by
	// latency = periodsize * periods / (rate * bytes_per_frame)
	// Buffersize in FRAMES Not in bytes!!
	if( snd_pcm_hw_params_set_buffer_size_near( pcm_handle, hwparams, buffersize ) != 0 ) {
		fprintf(stderr, SET: Error buffer size\n);
		return(-1);
	}
	if( buffersize != BUFFERSIZE_FRAMES ) {
		fprintf(stderr, BUFFERSIZE_FRAMES not set, check capture.cpp!\n);
		return(-1);
	}
	if( snd_pcm_hw_params( pcm_handle, hwparams ) != 0 ) {
		fprintf(stderr, SET: Error applying parameters\n);
		return(-1);
	}
	snd_pcm_sw_params_alloca(swparams);

	err = snd_pcm_sw_params_current(pcm_handle, swparams);
	if (err  0) {
		printf(Unable to determine current swparams %s\n,snd_strerror(err));
		return err;
	}

	snd_pcm_hw_params_get_period_size(hwparams, val, NULL);
	err = snd_pcm_sw_params_set_avail_min(pcm_handle, swparams, val);
	if (err  0) {
		printf(Unable to set avail min  %s\n, snd_strerror(err));
		return err;
	}
	err = snd_pcm_sw_params(pcm_handle, swparams);
	if (err  0) {
		printf(Unable to set sw params %s\n, snd_strerror(err));
		return err;
	}

	while(1){
		len=256;
		pointer=read_data;
		while (len  0) {
			while( ( written_frames = snd_pcm_writei( pcm_handle, pointer, len ) )  0 ) {
snd_pcm_prepare( pcm_handle );
printf( Buffer Underrun \n);
			}
			if(written_frames!=256)printf(%d\n,written_frames);
			pointer+=written_frames*4;
			len-=written_frames;
		}
	}
}
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[Alsa-user] no sound in ubuntu 8.10

2009-06-02 Thread sandip gangakhedkar
Hi all,

I running Ubuntu 8.10 on an AMD Athlon 64 3200+ machine. I have an Analog
Devices sound card for Creative Inspire 2.1 sound system. Few weeks ago, my
sound mysteriously stopped working.

/var/log/messages gives me this:

Jun  2 18:44:15 sandip-desktop pulseaudio[7147]: ltdl-bind-now.c:
Failed to find original dlopen loader.
Jun  2 18:44:15 sandip-desktop pulseaudio[7149]: main.c:
setrlimit(RLIMIT_NICE, (31, 31)) failed: Operation not permitted

Jun  2 18:44:15 sandip-desktop pulseaudio[7149]: main.c:
setrlimit(RLIMIT_RTPRIO, (9, 9)) failed: Operation not permitted

I have tried all alsa related fixes but to no avail - this seems like
a pulseaudio issue.

I wonder if anyone can help me on this one.

Thanks,

Sandip
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[Alsa-user] Bluetooth Headset

2009-06-02 Thread Memo5
Hi!

I am configuring a Bluetooth Headset (Nokia BH-101) with alsa. Pairing with 
the device seems to be ok, but the sound-output and input is not.

Playing a sound with
aplay -D btheadset -B 10 /usr/share/sounds/alsa/Front_Right.wav
produces a loud noise in the headset, only a recongnizeable distored sound is 
played in the background. It seems to me being a sampling-rate problem.

A similar problem exists when i try to record a sound with arecord, a sample 
file is here: http://members.aon.at/astronom/files/test_bt_rec.wav
It is an 16bit, mono PCM file with 8000Hz.
Can anybody speziefie the problem when litening to this file. (it is a 
count-up until 10 - in german)

My ~/.asoundrc:
pcm.btheadset {
type bluetooth
device 00:xx:xx:xx:xx:xx
profile voice
}

aplay --version
aplay: version 1.0.16 by Jaroslav Kysela pe...@perex.cz
(it is from Debian lenny)

thanks,
Steve

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Re: [Alsa-user] Capture w/ M-Audio Fast Track USB?

2009-06-02 Thread Michael B Allen
On Tue, Jun 2, 2009 at 3:25 AM, Clemens Ladisch cladi...@googlemail.com wrote:
 Michael B Allen wrote:
 Does capture work with the M-Audio Fast Track USB?

 Maybe.  Does some capture device show up in the output of arecord -l?

Yes. And I have been able to get capture to work but it is very faint
and distorted. Here is what I have done so far so that the other
people who were interested in this can play along:

# arecord -l
 List of CAPTURE Hardware Devices 
card 0: Intel [HDA Intel], device 0: AD198x Analog [AD198x Analog]
  Subdevices: 2/2
  Subdevice #0: subdevice #0
  Subdevice #1: subdevice #1
card 1: Track [Fast Track], device 0: USB Audio [USB Audio]
  Subdevices: 1/1
  Subdevice #0: subdevice #0

After inseting the Fast Track:

# cat /proc/asound/devices
  2:: timer
  3:: sequencer
  4: [ 0- 1]: digital audio playback
  5: [ 0- 0]: digital audio playback
  6: [ 0- 0]: digital audio capture
  7: [ 0- 0]: hardware dependent
  8: [ 0]   : control
  9: [ 1- 0]: digital audio playback
 10: [ 1- 0]: digital audio capture
 11: [ 1]   : control

So item 10 looks like card 1 subdevice 0 is digital audio capture.

But then again digital audio playback is also card 1 subdevice 0?

 Can someone give me some advice or a pointer to some documentation
 that explains how the whole ALSA tool chain works wrt external USB
 sound recording?

 You can tell any ALSA program to use a specific device by using a device
 name like plughw:x,y if you know the card and device numbers.

So this means the device identifier for the Fast Track with my
particular hardware setup is:

  plughw:1,0

 If you're using PulseAudio, you have to tell it to use that device.

Can I bypass PulseAudio and stick to the commandline?

If I do:

  # arecord -D plughw:1,0 /tmp/test.wav

and then do:

  # aplay -Dplughw:1,0 /tmp/test.wav
  Playing WAVE '/tmp/test.wav' : Unsigned 8 bit, Rate 8000 Hz, Mono

at first, this produced silence.

Then I realized I could do:

  # alsamixer -c 1

to run alsamixer with card 1. Now I can see Capture was turned OFF. So
if I go to Capture and hit the space bar to turn it ON and then turn
it up to 97%:

# amixer -c 1
Simple mixer control 'PCM',0
  Capabilities: pvolume pswitch pswitch-joined
  Playback channels: Front Left - Front Right
  Limits: Playback 0 - 256
  Mono:
  Front Left: Playback 192 [75%] [-31.50dB] [on]
  Front Right: Playback 192 [75%] [-31.50dB] [on]
Simple mixer control 'Mic',0
  Capabilities: cvolume cswitch cswitch-joined
  Capture channels: Front Left - Front Right
  Limits: Capture 0 - 128
  Front Left: Capture 124 [97%] [-3.00dB] [on]
  Front Right: Capture 124 [97%] [-3.00dB] [on]

and do arecord again just like before I can play it back and it works
but it is very faint and distorted.

So this proves capture signal is making it to ALSA.

However why is it so faint?

I suppose it is distorted because I have the level set to -3dB?

Mike

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[Alsa-user] ALSA 1.0.20 Speex PCM Plugin

2009-06-02 Thread Robert Krakora
Hello,

Has anyone successfully employed the ALSA 1.0.20 Speex PCM Plugin?  I
followed the speexdsp.txt document under the 'doc' directory but the
result was the following error:

[r...@vizioroom105 ~]# arecord -Dplug:mic poopy.wav
Recording WAVE 'poopy.wav' : Unsigned 8 bit, Rate 8000 Hz, Mono
ALSA lib pcm_params.c:2135:(snd1_pcm_hw_refine_slave) Slave PCM not usable
arecord: set_params:957: Broken configuration for this PCM: no
configurations available

Here is my asound.conf:

# MessageNet Systems ALSA sound configuration file
# Allowing multiple playback and capture

pcm.!default {
   type plug
   slave.pcm duplex
}
pcm.duplex {
   type asym
   playback.pcm dmixer
   capture.pcm analog_input
}
pcm.dmixer {
   type dmix
   ipc_key 1024
   ipc_perm 0666
   slave {
   pcm hw:2,3
   period_time 0
   period_size 1024
   buffer_size 8192
   }
   bindings {
  0 0
  1 1
   }
}
pcm.mic
{
  type speex
  slave.pcm {
  type hw
  card 1
  }
  agc 1
  agc_level 8000
}
pcm.analog_input
{
   type dsnoop
   ipc_key 2048
   ipc_perm 0666
   slave{
  pcm hw:1,2
   }
}
pcm.dsp0 {
   type plug
   slave.pcm dmixer
}
ctl.mixer0 {
   type hw
   card 0
}

Here are the results of alsa-info.sh:

http://www.alsa-project.org/db/?f=358e4d5d828dda3c44220382c8b480fd6019ab2a

Thanks in advance.

Best Regards,

-- 
Rob Krakora
Senior Software Engineer
MessageNet Systems
101 East Carmel Dr. Suite 105
Carmel, IN 46032
(317)566-1677 Ext. 206
(317)663-0808 Fax

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Re: [Alsa-user] No sound on LG-S1 QB01A9

2009-06-02 Thread Daren Krive
Hi Nigel,

Thanks for the suggestion.  I have tried what you suggested and still no
luck.  I do *sometimes* get static through my speakers when I boot up which
disappears and then  silence.

Here is the new output of the script after I did what you suggested...

http://www.alsa-project.org/db/?f=9e5b4548ded9b389661b0e669ecb5cd7364ff973

Regards,

Daren


On Sat, May 23, 2009 at 10:22 PM, Daren Krive daren.kr...@gmail.com wrote:


 On Sat, May 23, 2009 at 1:03 PM, Nigel Henry 
 cave.dnb2m9...@aliceadsl.frwrote:

 On Saturday 23 May 2009 17:03, Daren Krive wrote:
  Hi Arthur,
 
  Thanks so much for helping me.  I have downloaded the run the script.
  The
  output is here:
 
 
 http://www.alsa-project.org/db/?f=e2abef283d0e109086c3ce126a0db8ae6b967e21
 
  I am really excited about the possibility of getting sound to work.  I
 feel
  that with XP running under VirtualBox I wouldn't even need to dual-boot
  windows anymore if only I could get sound working.  Everything else
 works
  (almost).  Sound is the only thing I really can't live without anymore.
 
  Daren.
 

 Hi Daren.

 The PCM control is down at zero. Open alsamixer as user in Gnomes
 terminal, as
 below, which will show all controls for your soundcard, then push up the
 PCM
 slider, which hopefully will resolve your sound problem.

 alsamixer -D hw:0

 All the best.

 Nigel.


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[Alsa-user] No Sound (Realtek ALC650?)

2009-06-02 Thread Jamie Jackson
Hi I'm running 64-bit Ubuntu 9.04, and I have no sound from the
onboard jacks. I shouldn't say *no* sound, because there is some
low-level screeching coming from the line-out.

I don't know how to get sound out of this thing. I've read conflicting
accounts of which module to use, it's not obvious whether the ALC650
device is just an AC97 codec, or whether it's the motherboard chipset
that matters.

Please help me figure out how to get sound?

Here are the motherboard specs:
http://supermicro.com/Aplus/motherboard/Opteron/nforce/H8DCi.cfm

Here is lspci: http://jamiejackson.pastebin.com/f734d48ad

Please let me know what other info you might need to know.

Thanks,
Jamie

--
OpenSolaris 2009.06 is a cutting edge operating system for enterprises 
looking to deploy the next generation of Solaris that includes the latest 
innovations from Sun and the OpenSource community. Download a copy and 
enjoy capabilities such as Networking, Storage and Virtualization. 
Go to: http://p.sf.net/sfu/opensolaris-get
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