Re: [Alsa-user] mp3 trouble - help from here?

2007-08-30 Thread Frank K
Rene Herman wrote:
 On 08/29/2007 06:52 AM, Frank K wrote:

   
 Thanks to both of you for your responses. Following Volker's advice I 
 downloaded lame-3.97-1.suse10.2.i586.rpm from Kradio. It's installed 
 now, but Xine and Banshee are still looking for an mp3 decoder. The man 
 page for lame shows it to be a converter of other sound files to mp3.
 

 More a producer than a converter. LAME's the (kind of) program that produces 
 MP3 files. It encodes raw, uncompressed digital audio such as found on CDs 
 for example (and sometimes also known as PCM) to the compressed audio 
 format MP3. As a bonus, it also accepts MP2 or MP3 files as input but 
 that's just for convenience (and a bad idea -- see below).

   
 Since I can't hear the mp3 files I have, I don't see how this helps.
 Man lame also mentions mpg123 and madplay. Neither are in the Suse10.2 
 distribution. These appear to be mp3 players.
 

 Yes, those are command line MP3 players. Amarok, XMMS, AlsaPlayer and many 
 more players are also available for the GUI.

   
 What is the decoder Xine and Banshee think they are missing?
 

 Try if this is useful:

 http://amarok.kde.org/wiki/MP3_on_SUSE_Linux_10.2

 More generally though, the answer's no idea, as this is completely 
 distribution specific and moreover not at all related to ALSA.

 See, the trouble is -- MP3 is a proprietary, patent encumbered format for 
 which (sometimes) licensing fees have to be payed to some unclear number of 
 parties and as such, free distributions have trouble supplying MP3 support 
 out of the box. The trouble you are having now is the kind of trouble usage 
 of these and similar formats land people in.

   
 Rene, I'm thinking that mp3 files would need to be converted to Ogg 
 Vorbis?
 

 No, I wouldn't really advice that. Both MP3 and Ogg Vorbis are lossy 
 compression formats, which means they both leave out some of the information 
 from the original uncompressed source audio. Exactly what and how much they 
 leave out is what makes them different. Obviously, if your source form is 
 one lossy format, any re-encoding to another lossy format could only leave 
 out more which means you should expect to end up with a worse sounding file.

 Generally it won't be bad, so for casual use transcoding MP3 to Ogg Vorbis 
 can be okay, but as a rule you shouldn't. Always encode directly from the 
 source, which normally means from the CD. If you don't have the source, 
 chances are fairly high the MP3 was illegally distributed to you in the 
 first place, so you don't get to complain about anything and nobody cares 
 about your needs anyway :-)

   
 Can OV do that or would that conversion require another application in
 need of an mp3 decoder?
 

 Dunno what OV is, but as said, I'd not transcode anything. The normal 
 application to encode from PCM to Ogg Vorbis is oggenc.

 Really off-topic this, so I hope the link above is useful. If not, you're 
 best of following up to some SuSE specific list/forum.

 Rene.


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Thanks again for the responses from Volker and Rene. I tried the Google 
search Volker suggested and found the How To on Overclock. That 
procedure went well until the load of XineLib. There the dependencies 
multiplied. Rene's link worked very well.
The last half of the last step doesn't work, but that seems related to 
different encoding format than mp3. As I type this I am listening to Al 
Gore's book on CD which is what I wanted. I count this thread a success!
Rene you are correct that this thread is/was off-topic. Thanks also for 
bending the rules slightly!
Frank K

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Re: [Alsa-user] alsa buffer size

2007-08-30 Thread paul blakeley
Thanks for replying.

I am writing an application that is generic enough to a number of
different drivers.  If thats possible!  I am trying to achieve a 2
second of buffer for a number of different sample rates.  This means
that for a sample rate of 44.1KHz I require a buffer size of 88.2K
frames  For all the different types of sound cards that I have
used I havent seen a buffer size  16K frames : (

I have noticed that various drivers have certain behaviour which makes
writing a application that is generic for all drivers very difficult.
For example for an intel chipset I have found that the driver was very
versatile and allowed me to choose various period sizes for different
buffer sizes.  Whilst the driver for the CS4299 chipset, although
allows a buffer size of upto 16K the period size has to be half that
value.  Therefore this makes it impossible for me to use with the ALSA
callback mechanism, which I need to use for the design to work!

By the way how do I find out which driver is assigned to a chipset? I
understand that I can use 'lsmod'.  Can a driver be used on multiple
chipsets?  You will have to excuse my knowledge on this : )

On 8/29/07, Clemens Ladisch [EMAIL PROTECTED] wrote:
 paul blakeley wrote:
  Is it possible to increase the buffer size of a ALSA driver?

 Yes. No. Er, maybe.

 Each driver has a certain limit for the buffer size.  The ALSA framework
 preallocates memory for the buffer, but usually only half the limit.
 You can increase the amount by writing a new value into
 /proc/asound/cardX/pcm*/sub*/prealloc.

 The limit is sometimes just an arbitrary value invented by the driver
 writer, but in many cases a hard limit of the hardware.

 What driver are you using?


 Regards,
 Clemens


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Re: [Alsa-user] alsa buffer size

2007-08-30 Thread paul blakeley
I have successfully increased the amount of buffer by writing a new value into
/proc/asound/cardX/pcm*/sub*/prealloc.

For a mono .wav file I was able to successfully change this from 64 to
512.  I was really happy with this.  I presume this value is in the
units of kBytes?  And that this value needs to be 2 to the n. ie
16,32,64 etc?  Dont know why but I couldn't change this to  512, do
you?

I also tried this change on a stereo .wav file but it didnt scale as I
thought.  In fact I could only set this to 32!  Which doesnt make
sense.  Do you know why?

On 8/30/07, paul blakeley [EMAIL PROTECTED] wrote:
 Thanks for replying.

 I am writing an application that is generic enough to a number of
 different drivers.  If thats possible!  I am trying to achieve a 2
 second of buffer for a number of different sample rates.  This means
 that for a sample rate of 44.1KHz I require a buffer size of 88.2K
 frames  For all the different types of sound cards that I have
 used I havent seen a buffer size  16K frames : (

 I have noticed that various drivers have certain behaviour which makes
 writing a application that is generic for all drivers very difficult.
 For example for an intel chipset I have found that the driver was very
 versatile and allowed me to choose various period sizes for different
 buffer sizes.  Whilst the driver for the CS4299 chipset, although
 allows a buffer size of upto 16K the period size has to be half that
 value.  Therefore this makes it impossible for me to use with the ALSA
 callback mechanism, which I need to use for the design to work!

 By the way how do I find out which driver is assigned to a chipset? I
 understand that I can use 'lsmod'.  Can a driver be used on multiple
 chipsets?  You will have to excuse my knowledge on this : )

 On 8/29/07, Clemens Ladisch [EMAIL PROTECTED] wrote:
  paul blakeley wrote:
   Is it possible to increase the buffer size of a ALSA driver?
 
  Yes. No. Er, maybe.
 
  Each driver has a certain limit for the buffer size.  The ALSA framework
  preallocates memory for the buffer, but usually only half the limit.
  You can increase the amount by writing a new value into
  /proc/asound/cardX/pcm*/sub*/prealloc.
 
  The limit is sometimes just an arbitrary value invented by the driver
  writer, but in many cases a hard limit of the hardware.
 
  What driver are you using?
 
 
  Regards,
  Clemens
 


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Re: [Alsa-user] snd-usb-audio and the Actiontec Internet Phone Wizard

2007-08-30 Thread Clemens Ladisch
TheOneKEA wrote:
 Was the lsusb output helpful in determining if the device does need a
 separate telephony driver?

It does have an extra interface that is independent of the three audio
interfaces.  I guess the driver uses that one to send vendor-specific
control messages.


Regards,
Clemens

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[Alsa-user] Front Output Jack problems with AD1986A/M5461

2007-08-30 Thread Kevin Hill
Hi, I'm using the latest ALSA version with Ubuntu Fiesty and I can't
seem to get my front audio jack on my computer to work.

I have an ASUS A8R-MVP mobo and my case has some jumpers to connect a
front audio output (and input but I don't use it) to my mobo. The rear
jack that is part of my mobo works great, but the front one never gets
any sound. Works find under my WinXP boot, but not under Linux, so I'm
pretty sure it's some ALSA setting I need to change.

I'm somewhat comfortable with Linux, but never had to worry about
sound issues before now, so not sure what information would be useful
here, but here's a stab at some useful stuff

$ cat /proc/asound/cards
 0 [M5461  ]: HDA-Intel - HDA ULI M5461
  HDA ULI M5461 at 0xcedf8000 irq 22

$ cat /proc/asound/pcm
00-01: AD198x Digital : AD198x Digital : playback 1
00-00: AD198x Analog : AD198x Analog : playback 1 : capture 1

$ cat /proc/asound/version
Advanced Linux Sound Architecture Driver Version 1.0.14.
Compiled on Aug 30 2007 for kernel 2.6.20-16-generic (SMP).

Ideally I would love to play the front port at an attenuated volume
compared to the back port, as I usually have headphones plugged into
the front and speakers in the back which can get a bit too lound on
headphones. But just getting sound out of that front port in any form
would be great. Thanks in advance.

-Kevin

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Re: [Alsa-user] EMU 1212M PCI

2007-08-30 Thread James Courtier-Dutton
Chris Aitken wrote:
 I installed fedora 7 so I'd have the kernel and alsa I need for the EMU 
 1212M PCI soundcard. I installed the card. I tried opening alsa mixer to 
 see if the soundcard is recognized and got:
 
 alsamixer: function snd_ctl_open failed for default: No such device
 

Please post the output of the following:
cat /proc/asound/cards
cat /proc/asound/version


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[Alsa-user] EMU 1212M PCI

2007-08-30 Thread Chris Aitken
James Courtier-Dutton wrote:
 Chris Aitken wrote:
   
 I installed fedora 7 so I'd have the kernel and alsa I need for the EMU 
 1212M PCI soundcard. I installed the card. I tried opening alsa mixer to 
 see if the soundcard is recognized and got:

 alsamixer: function snd_ctl_open failed for default: No such device

 

 Please post the output of the following:
 cat /proc/asound/cards
   
[EMAIL PROTECTED] ~]$ cat /proc/asound/cards
 1 [I82801BAICH2   ]: ICH - Intel 82801BA-ICH2
  Intel 82801BA-ICH2 with AD1885 at irq 20

 cat /proc/asound/version
   
[EMAIL PROTECTED] ~]$ cat /proc/asound/version
Advanced Linux Sound Architecture Driver Version 1.0.14 (Thu May 31 
09:03:25 2007 UTC).

Chris

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Re: [Alsa-user] Mixer problem with cmipci

2007-08-30 Thread Jan-Benedict Glaw
On Wed, 2007-08-29 13:53:07 +0200, Clemens Ladisch [EMAIL PROTECTED] wrote:
 Jan-Benedict Glaw wrote:
  On Tue, 2007-08-28 08:28:39 +0200, Clemens Ladisch [EMAIL PROTECTED] 
  wrote:
   Jan-Benedict Glaw wrote:
[EMAIL PROTECTED]:~$ amixer cset numid=35 off
numid=35,iface=MIXER,name='IEC958 In Monitor'
  ; type=BOOLEAN,access=rw--,values=1
  : values=on

Seems it cannot change this control's value?
   
   Strange.  Please show the contents of /proc/asounc/card0/cmipci.
  
  Since I just bought that card and never tested it somewhere else:
  Could this just be a hardware problem / defect?  From a look at the
  driver's sources, it seems that this mixer element directly maps to
  the low bit of CM_REG_MIXER1.
 
 Indeed.  (This is why I asked for the cmipci file and not the cards
 file, to see how the mixer control affects this bit in register 0x24.)

I didn't (yet) check that card in a Windows machine to test it with
vendor's drivers, but I really guess it's kind of a hardware problem.

Just out of interest, I ordered another, similar card (Ultron
Octosound 7.1, lspci: C-Media Electronics Inc CM8738 (rev 10),
/proc/asound/cards: C-Media PCI CMI8738-MC8 (model 68))

Plugged all in, fired up speaker-test, everything seems to work right
out of the box.

Fired up an alsaplayer to play some FLAC. Switched on Four Channel
Mode for a test, which nicely adds sound to the rear left/right
outputs.

After the FLAC ended, I fired up speaker-test again (and of course
switched off Four Channel Mode.)  This time, it's different than
before: front center/LFE, front left/right and side left/right are
okay, but rear right/left are highly distorted.

Unloaded/loaded the module once, but that didn't fix it.  Powercycle
(did also remove the power cord to really let it loss its power) fixed
it. So I guess there's still a state-handling problem somewhere?

Ran another test, right after another power-cycle:

  * speaker-test -twav -Dplug:surround71 -c8
- All okay.
  * alsamixergui 
  * speaker-test -twav -Dplug:surround71 -c8
- All okay.
  * alsaplayer 04_my_immortal.flac (for some seconds, Ctrl-C'ed out)
- Plays fine
  * speaker-test -twav -Dplug:surround71 -c8
- Distorted sound on both rear {left,right} speakers. All
other channels are still okay.

Maybe that rings a bell somewhere?

MfG, JBG

-- 
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Signature of:  Fortschritt bedeutet, einen Schritt so zu machen,
the second  :   daß man den nächsten auch noch machen kann.


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Re: [Alsa-user] ESI RoMIO 2 problem

2007-08-30 Thread Maciej Łoziński
Dnia 21-08-2007 o 19:28:47 Bill Unruh [EMAIL PROTECTED] napisał(a):

 On Tue, 21 Aug 2007, Maciej oziski wrote:

 Thank you very much! I'll try this.
 Unfortuantely my laptop has a lot of problems now. I upgraded it to  
 debian
 testing from stable to get the latest alsa drivers, and it started to  
 have
 soft lockups. Now I came back from work, and it doesn't even want to  
 power
 up :-/


 You do not have to install a whole new operating system to get a new  
 alsa.
 Just download the latest alsa, compile it on the machine you want to use  
 it
 on with the appropriate kernel source tree and config file, and install  
 it. (NOte you need to check to see if the files in
 /lib/modules/kernel-version/kernel/snd have a .o or .ko extention, or  
 are compressed ( eg .ko.gz) If so, then the
 alsa make install screws up and you need to run
 find /lib/modules/kernel-version/kernel/snd -name snd\*.\*o.gz|xargs rm
 first so as to remove the old modules.

 Note also that debian is the worst distribution to use if you want to
 support the latest hardware. They purposely trade off support for
 stability. That may be a good tradeoff for some, but support for sound
 hardware is not one of the places.


Yes, I know debian is not the best. But I know it better than others and  
so I use it.
I downloaded 64 Studio debian, and will give it a try as soon as  
possible :-)

What distro do you use? Which is the best for midi work?
I tried dyne:bolic and pure:dyne, but it's hard to compile new drivers  
there... Despite that I think they're great.

If it comes to the drivers, the patch Clemens Ladisch has provided link  
to, worked well. My interface now works. But unfortunately I can't use it  
due to problems with real time - when I do some work with interface (enter  
notes wit mouse etc), there are delays. Really nasty. So I'll wait for a  
RT kernel.

cheers
Maciek

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[Alsa-user] Best soundcard to buy for Linux

2007-08-30 Thread Helge Fredriksen
Hello!

I'm searching for the best soundcard to use with a JavaSound/VoIP
application on Linux. I even have to mix in noise/clips with the VoIP
packages received on the PC that will render the sound. So I thought better
to go for a card that has HW mixing capabilities, which let me open more
than one SourceDataLine (one for VoIP, one for noise and one for clip...)
and let mixing be done at the hardware. I got hold of an old Creative Audigy
1 card, which really gave me great sound, as opposed to the new Creative
Audigy SE cards with no HW mixing.

However, cards with HW mixing is really not easy to find on Linux. Creative
seem to have stopped producing the Audigy HW mixing cards. All I found
except that was the Mia MIDI card, but that had only passive mic input (had
to have a preamp on those mics!).

If my mixing can't be done in hardware, does any one have a good
recommendation on well performing single/double line I/O sound cards
supported on Linux?  Best would be with drivers supporting opening more than
one SourceDataLine on the same mixer.

Best regards,
Helge Fredriksen
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Re: [Alsa-user] Front Output Jack problems with AD1986A/M5461

2007-08-30 Thread James Shatto
 Hi, I'm using the latest ALSA version with
 Ubuntu Fiesty and I can't seem to get my
 front audio jack on my computer to work.

Have you checked all of your mixer settings?  The front one might be PCM2 or 
Headphone.  And might only be accessible in alsamixer.

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Re: [Alsa-user] Best USB based XLR3 interface to buy?

2007-08-30 Thread James Shatto
 Is there a list of still produced
 USB based XLR3 microphone interfaces
 that actually work with linux/alsa?

I would like to buy a USB 2.0 based soundcard that allows high quality input 
from an XLR microphone.  Preferably two XLR3 inputs for stereo recording.  Any 
recommendations?  For ones that work, or even ones that are known to NOT work, 
so I don't get one of those by accident.  

So far I've only be able to identify older Tascam US-122's to work, but newer 
US-122L's and US-144's don't work.  The usb-audio sources seem to indicate that 
the native instruments Audio 8 DJ and Audio Kontrol 1 to work with the caiaq 
driver.  But those don't have stereo inputs.

Is there a list of known devices that work?  Or do I just need to bump through 
all of the entries in the soundcard matrix and hope I run into a supported usb 
2.0 device that has stereo XLR mic interfaces?  The few vendor:device numbers 
I've researched yielded no entries in pciids.sf.net.  And many of the 
manufacturers of said devices have no manufacturer entry in the alsa soundcard 
matrix.

Thanks.

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Re: [Alsa-user] snd-usb-audio and the Actiontec Internet Phone Wizard

2007-08-30 Thread TheOneKEA
Mr. Ladisch,

On 8/30/07, Clemens Ladisch [EMAIL PROTECTED] wrote:
 TheOneKEA wrote:
  Was the lsusb output helpful in determining if the device does need a
  separate telephony driver?

 It does have an extra interface that is independent of the three audio
 interfaces.  I guess the driver uses that one to send vendor-specific
 control messages.

Hm.

So what else could be tried? I would really like to get this device
working in Linux, if at all possible.

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[Alsa-user] Audio Hardware for multichannel output

2007-08-30 Thread Jeromie Clark
Hi -

I've been looking around, but I can't seem to find a recommended list of
hardware for use with Alsa and multichannel output, and I'm wondering if
anyone on the list has hands-on experience with hardware that works reliably
in this application.

Specifically, I'm looking to drive between 16 separate mono outputs from a
single PC, using a custom application that I'll write in either C++ or
java.

It looks like alsa allows you to bind multiple consumer-grade cards as a
single device for this purpose, although I'm just as happy (and would
probably prefer) to buy a rock-solid, purpose-built, commercial-grade piece
of hardware.  I'd rather pay a little extra and avoid the nightmarish
headache of trying to get long-term consistency out of four soundblasters.

If you're not comfortable endorsing a particular venue or solution on the
list, please feel free to mail me directly.

Thanks in advance,
Jeromie Clark
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