Re: [Alsa-user] Could Alsa cause a hard crash?

[Paolo Bolzoni]

Kernel panic perhaps?

On Tue, Apr 14, 2015 at 5:21 PM, Clemens Ladisch
cladi...@googlemail.com wrote:
 Jeff Sadowski wrote:
 I was having issues with Mythtv hard crashing my system. I had
 previously had the output sound device set directly as my HDMI on my
 Intel Corporation 5 Series/3400 Series Chipset High Definition Audio
 I remembered in the past having issues with pulse audio and any other
 program fighting over the sound card. So I changed Mythtv's output to
 use pulseaudio in the hopes that it would work with pulseaudio instead
 of fighting it. This seems to have helped. I have not had a crash
 since. But the underlying problem worries me. Should fighting for
 sound card resources crash the computer?

 No.  The crashes are either a hardware problem, or a bug in one of the
 drivers used.

 What exactly do you mean with hard crashing?


 Regards,
 Clemens

 --
 BPM Camp - Free Virtual Workshop May 6th at 10am PDT/1PM EDT
 Develop your own process in accordance with the BPMN 2 standard
 Learn Process modeling best practices with Bonita BPM through live exercises
 http://www.bonitasoft.com/be-part-of-it/events/bpm-camp-virtual- event?utm_
 source=Sourceforge_BPM_Camp_5_6_15utm_medium=emailutm_campaign=VA_SF
 ___
 Alsa-user mailing list
 Alsa-user@lists.sourceforge.net
 https://lists.sourceforge.net/lists/listinfo/alsa-user

--
BPM Camp - Free Virtual Workshop May 6th at 10am PDT/1PM EDT
Develop your own process in accordance with the BPMN 2 standard
Learn Process modeling best practices with Bonita BPM through live exercises
http://www.bonitasoft.com/be-part-of-it/events/bpm-camp-virtual- event?utm_
source=Sourceforge_BPM_Camp_5_6_15utm_medium=emailutm_campaign=VA_SF
___
Alsa-user mailing list
Alsa-user@lists.sourceforge.net
https://lists.sourceforge.net/lists/listinfo/alsa-user

Re: [Alsa-user] Could Alsa cause a hard crash?

[Paolo Bolzoni]

mplayer allows it, try the option: -ao alsa:device=devicename

Just replace any ',' with '.' and  any ':'  with '=' in the ALSA device name.

For example: mplayer -ao alsa:device:hw=Audigy2


On Thu, Apr 16, 2015 at 11:35 PM, Jeff Sadowski jeff.sadow...@gmail.com wrote:
 On Thu, Apr 16, 2015 at 1:40 AM, Clemens Ladisch
 cladi...@googlemail.com wrote:
 Jeff Sadowski wrote:
 I think it is a kernel panic but the display is locked with the GUI.

 Would you be able to switch back to the console before the crash?

 I need a way to play audio to both pulseaudio and then play something
 to the hdmi directly. From command line. Then I could do it from the
 console. Maybe play 2 mp3s with mplayer? I think mplayer has switches
 to send to different sound devices?
 I could be wrong all together about what was causing the crashes but I
 would like to try that.


 Regards,
 Clemens

--
BPM Camp - Free Virtual Workshop May 6th at 10am PDT/1PM EDT
Develop your own process in accordance with the BPMN 2 standard
Learn Process modeling best practices with Bonita BPM through live exercises
http://www.bonitasoft.com/be-part-of-it/events/bpm-camp-virtual- event?utm_
source=Sourceforge_BPM_Camp_5_6_15utm_medium=emailutm_campaign=VA_SF
___
Alsa-user mailing list
Alsa-user@lists.sourceforge.net
https://lists.sourceforge.net/lists/listinfo/alsa-user

Re: [Alsa-user] Could Alsa cause a hard crash?

[Paolo Bolzoni]

It looks a grave hardware failure to me

On Wed, Apr 15, 2015 at 12:40 AM, Jeff Sadowski jeff.sadow...@gmail.com wrote:
 I think it is a kernel panic but the display is locked with the GUI.
 No messages are displayed. It is what ever image was on screen as it
 crashes. I try to ssh in with no luck. Mouse and keyboard do nothing.
 I restart and I don't see anything in the logs about what happened.

 On Tue, Apr 14, 2015 at 2:53 AM, Paolo Bolzoni
 paolo.bolzoni.br...@gmail.com wrote:
 Kernel panic perhaps?

 On Tue, Apr 14, 2015 at 5:21 PM, Clemens Ladisch
 cladi...@googlemail.com wrote:
 Jeff Sadowski wrote:
 I was having issues with Mythtv hard crashing my system. I had
 previously had the output sound device set directly as my HDMI on my
 Intel Corporation 5 Series/3400 Series Chipset High Definition Audio
 I remembered in the past having issues with pulse audio and any other
 program fighting over the sound card. So I changed Mythtv's output to
 use pulseaudio in the hopes that it would work with pulseaudio instead
 of fighting it. This seems to have helped. I have not had a crash
 since. But the underlying problem worries me. Should fighting for
 sound card resources crash the computer?

 No.  The crashes are either a hardware problem, or a bug in one of the
 drivers used.

 What exactly do you mean with hard crashing?


 Regards,
 Clemens

 --
 BPM Camp - Free Virtual Workshop May 6th at 10am PDT/1PM EDT
 Develop your own process in accordance with the BPMN 2 standard
 Learn Process modeling best practices with Bonita BPM through live exercises
 http://www.bonitasoft.com/be-part-of-it/events/bpm-camp-virtual- event?utm_
 source=Sourceforge_BPM_Camp_5_6_15utm_medium=emailutm_campaign=VA_SF
 ___
 Alsa-user mailing list
 Alsa-user@lists.sourceforge.net
 https://lists.sourceforge.net/lists/listinfo/alsa-user

--
BPM Camp - Free Virtual Workshop May 6th at 10am PDT/1PM EDT
Develop your own process in accordance with the BPMN 2 standard
Learn Process modeling best practices with Bonita BPM through live exercises
http://www.bonitasoft.com/be-part-of-it/events/bpm-camp-virtual- event?utm_
source=Sourceforge_BPM_Camp_5_6_15utm_medium=emailutm_campaign=VA_SF
___
Alsa-user mailing list
Alsa-user@lists.sourceforge.net
https://lists.sourceforge.net/lists/listinfo/alsa-user

Re: [Alsa-user] Fedora 23 x86_64; Pulseaudio removed; playing from multiple sources serially - a little progress . .

[Paolo Bolzoni]

3. Using the .asoundrc above prevents Mumble and YouTube in Chrome working
Mumble is a phoning program right? That asoundrc does not setup any
input device, in my examply I combine dmix for the output and the
hardware for input.
I did that because I don't need multiple input streams at the same
time, but I need multiple outputs.

On Wed, Oct 7, 2015 at 4:12 PM, Paolo Bolzoni
<paolo.bolzoni.br...@gmail.com> wrote:
> 6. I found this: "NOTE: For ALSA 1.0.9rc2 and higher you don't need to
> setup dmix for analogue output. Dmix is enabled by default for
> soundcards which don't support hardware mixing. You still need to set
> it up for digital outputs."
>
> In my experience, this is a lie.
>
> On Wed, Oct 7, 2015 at 2:13 PM, Philip Rhoades <p...@pricom.com.au> wrote:
>> Paolo, Anders,
>>
>>
>> On 2015-10-07 02:44, Paolo Bolzoni wrote:
>>>
>>> I use this default. It allows to select any pcm via environment variable
>>> pcm.!default {
>>>   type plug
>>>   slave.pcm {
>>> @func getenv
>>> vars [ ALSAPCM ]
>>> default "pcm.PCH"
>>>   }
>>> }
>>>
>>> As you can see the default is pcm.PCH I told you before. But it is
>>> useful to be able to change the sound pcm with ease because some
>>> programs do not like speak with dmix and want to speak with the
>>> hardware directly. (e.g., wine)
>>>
>>> So, for that programs I use ALSAPCM=hw:PCH.
>>>
>>> But, once again, names are probably different on your system.
>>
>>
>>
>> OK, once I saw your comments about dmix I started to try and get that to
>> work - what I have found so far:
>>
>> 1. With NO .asoundrc file - only ONE mplayer instance works
>>
>> 2. Using this .asoundrc:
>>
>> pcm.!default {
>>   type plug
>>   slave.pcm "dmixer"
>> }
>>
>> pcm.dmixer  {
>>   type dmix
>>   ipc_key 1024
>>   slave {
>>   pcm "hw:1,0"
>>   period_time 0
>>   period_size 1024
>>   buffer_size 4096
>>   rate 44100
>>   }
>>   bindings {
>>   0 0
>>   1 1
>>   }
>> }
>>
>> ctl.dmixer {
>>   type hw
>> # card 0
>>   card 1
>>   device 0
>> }
>>
>> I can get TWO mplayers going at the same time.
>>
>> 3. Using the .asoundrc above prevents Mumble and YouTube in Chrome working
>>
>> 4. With NO .asoundrc file and xfce4-mixer and alsamixer settings for Front
>> Mike, Front Mike Boost, Rear Mike and Rear Mike Boost ALL muted, Mumble
>> still works fine! - it looks like it is bypassing ALSA and talking directly
>> to the port or something?
>>
>> 5. It looks like dmix setups can get quite complex - more reading is
>> required I think . .
>>
>> 6. I found this: "NOTE: For ALSA 1.0.9rc2 and higher you don't need to setup
>> dmix for analogue output. Dmix is enabled by default for soundcards which
>> don't support hardware mixing. You still need to set it up for digital
>> outputs." here:
>>
>>   http://alsa.opensrc.org/Dmix
>>
>> - so I'm not sure what that means for me - I am only using analogue output .
>> .
>>
>> 7. It looks like with some correct dmix parameters for custom pcms for
>> Mumble and Chrome/YouTube, I should be able to get those to work?
>>
>> 8. There is stuff about snd-aloop and asym that also might be relevant but I
>> am a bit lost now . .
>>
>> 9. It looks like these problems go back a while:
>>
>>   https://bugzilla.redhat.com/show_bug.cgi?id=130593
>>
>> but my problems seems like they should be soluble by now at least - even if
>> I have to set up a custom pcm for each input source . .
>>
>> Thanks,
>>
>> Phil.
>> --
>> Philip Rhoades
>>
>> PO Box 896
>> Cowra  NSW  2794
>> Australia
>> E-mail:  p...@pricom.com.au

--
Full-scale, agent-less Infrastructure Monitoring from a single dashboard
Integrate with 40+ ManageEngine ITSM Solutions for complete visibility
Physical-Virtual-Cloud Infrastructure monitoring from one console
Real user monitoring with APM Insights and performance trend reports 
Learn More http://pubads.g.doubleclick.net/gampad/clk?id=247754911=/4140
___
Alsa-user mailing list
Alsa-user@lists.sourceforge.net
https://lists.sourceforge.net/lists/listinfo/alsa-user

Re: [Alsa-user] Fedora 23 x86_64; Pulseaudio removed; playing from multiple sources serially

[Paolo Bolzoni]

I use this default. It allows to select any pcm via environment variable
pcm.!default {
  type plug
  slave.pcm {
@func getenv
vars [ ALSAPCM ]
default "pcm.PCH"
  }
}

As you can see the default is pcm.PCH I told you before. But it is
useful to be able to change the sound pcm with ease because some
programs do not like speak with dmix and want to speak with the
hardware directly. (e.g., wine)

So, for that programs I use ALSAPCM=hw:PCH.

But, once again, names are probably different on your system.


On Tue, Oct 6, 2015 at 5:39 PM, Anders Genell <anders.gen...@gmail.com> wrote:
>
>
>
>
>> 6 okt 2015 kl. 16:41 skrev Philip Rhoades <p...@pricom.com.au>:
>>
>> Paolo,
>>
>>
>>> On 2015-10-06 22:12, Paolo Bolzoni wrote:
>>> Sorry, my bad. I understood you needed to use a program that needed
>>> pulse audio without it; in this case apulse could help
>>>
>>> However, if your problem is allowing to use the sound output to
>>> multiple programs at the same time. I had a similar problem and I
>>> solved it using dmix.
>>>
>>> In my .asoundrc I had this pcm:
>>> pcm.PCH {
>>>  type asym
>>>  playback.pcm {
>>>type plug
>>>slave {
>>>  pcm {
>>>type dmix
>>>ipc_key 9175930
>>>ipc_key_add_uid true
>>>slave {
>>>  pcm "hw:PCH"
>>>}
>>>  }
>>>  }
>>>  }
>>>  capture.pcm "hw:PCH"
>>> }
>>
>>
>> I replaced my simple .asoundrc:
>>
>> pcm.!default {
>> type hw
>> card 1
>> device 0
>> }
>>
>>
>> with yours but with no improvement (I even rebooted to be sure
>
> In yours you had !default which means that is the device used when no other 
> is specified. If you change pcm.PCH to pcm.!default it might work.
> Also, the above refers to the hardware as "hw:PCH" which is valid for many 
> computer/soundcard combinations but not all. If you run the command
> aplay -l
> in your terminal, you should get a list of audio hardware, and may figure out 
> how to refer to the correct one in the asoundrc file.
>
> Regards,
> Anders
>
>
>
>> - I have
>> had so much problem with audio in the past):
>>
>> - mplayer works fine in isolation
>>
>> - YouTube works in isolation
>>
>> - Mumble works in isolation
>>
>> - If I have Mumble open, mplayer does not work
>>
>> - If I have YT running, mplayer does not work
>>
>> etc
>>
>> It would be nice to pause whatever I am using ie NOT have to exit the
>> program, and use another audio app and then come back to where I left
>> off with the first app . .
>>
>> Thanks,
>>
>> Phil.
>>
>>
>>> On Tue, Oct 6, 2015 at 12:38 PM, Philip Rhoades <p...@pricom.com.au>
>>> wrote:
>>>> Paolo,
>>>>
>>>>
>>>>> On 2015-10-06 21:34, Paolo Bolzoni wrote:
>>>>>
>>>>> It is meant to use skype without pulse audio, but it might help you?
>>>>> https://github.com/i-rinat/apulse
>>>>
>>>>
>>>>
>>>> I am not sure how Skype came into the discussion - I don't use it . .
>>>>
>>>> Thanks,
>>>>
>>>> Phil.
>>>>
>>>>
>>>>
>>>>
>>>>> On Tue, Oct 6, 2015 at 12:28 PM, Philip Rhoades <p...@pricom.com.au>
>>>>> wrote:
>>>>>>
>>>>>> Chris,
>>>>>>
>>>>>>
>>>>>>> On 2015-09-27 00:23, chris hermansen Sat wrote:
>>>>>>>
>>>>>>> Phil and list,
>>>>>>>
>>>>>>>> On Sep 26, 2015 03:21, "Philip Rhoades" <p...@pricom.com.au> wrote:
>>>>>>>>
>>>>>>>>
>>>>>>>> People,
>>>>>>>>
>>>>>>>> Years ago when I needed to simplify things to solve audio hardware
>>>>>>>> problems, I had to remove PA - and for every new version ever
>>>>>>>> since I
>>>>>>>> have automatically uninstalled it to continue to keep things as
>>>>>>>> simple
>>>>>>>> as possible - which generally works well for me.  Mostly I play
>>>>>>>>

Re: [Alsa-user] Fedora 23 x86_64; Pulseaudio removed; playing from multiple sources serially

[Paolo Bolzoni]

Did you actually told all the applications to use that pcm?

On Tue, Oct 6, 2015 at 4:41 PM, Philip Rhoades <p...@pricom.com.au> wrote:
> Paolo,
>
>
> On 2015-10-06 22:12, Paolo Bolzoni wrote:
>>
>> Sorry, my bad. I understood you needed to use a program that needed
>> pulse audio without it; in this case apulse could help
>>
>> However, if your problem is allowing to use the sound output to
>> multiple programs at the same time. I had a similar problem and I
>> solved it using dmix.
>>
>> In my .asoundrc I had this pcm:
>> pcm.PCH {
>>   type asym
>>   playback.pcm {
>> type plug
>> slave {
>>   pcm {
>> type dmix
>> ipc_key 9175930
>> ipc_key_add_uid true
>> slave {
>>   pcm "hw:PCH"
>> }
>>   }
>>   }
>>   }
>>   capture.pcm "hw:PCH"
>> }
>
>
>
> I replaced my simple .asoundrc:
>
> pcm.!default {
> type hw
> card 1
> device 0
> }
>
>
> with yours but with no improvement (I even rebooted to be sure - I have had
> so much problem with audio in the past):
>
> - mplayer works fine in isolation
>
> - YouTube works in isolation
>
> - Mumble works in isolation
>
> - If I have Mumble open, mplayer does not work
>
> - If I have YT running, mplayer does not work
>
> etc
>
> It would be nice to pause whatever I am using ie NOT have to exit the
> program, and use another audio app and then come back to where I left off
> with the first app . .
>
>
> Thanks,
>
> Phil.
>
>
>> On Tue, Oct 6, 2015 at 12:38 PM, Philip Rhoades <p...@pricom.com.au>
>> wrote:
>>>
>>> Paolo,
>>>
>>>
>>> On 2015-10-06 21:34, Paolo Bolzoni wrote:
>>>>
>>>>
>>>> It is meant to use skype without pulse audio, but it might help you?
>>>> https://github.com/i-rinat/apulse
>>>
>>>
>>>
>>>
>>> I am not sure how Skype came into the discussion - I don't use it . .
>>>
>>> Thanks,
>>>
>>> Phil.
>>>
>>>
>>>
>>>
>>>> On Tue, Oct 6, 2015 at 12:28 PM, Philip Rhoades <p...@pricom.com.au>
>>>> wrote:
>>>>>
>>>>>
>>>>> Chris,
>>>>>
>>>>>
>>>>> On 2015-09-27 00:23, chris hermansen Sat wrote:
>>>>>>
>>>>>>
>>>>>> Phil and list,
>>>>>>
>>>>>> On Sep 26, 2015 03:21, "Philip Rhoades" <p...@pricom.com.au> wrote:
>>>>>>>
>>>>>>>
>>>>>>>
>>>>>>> People,
>>>>>>>
>>>>>>> Years ago when I needed to simplify things to solve audio hardware
>>>>>>> problems, I had to remove PA - and for every new version ever since I
>>>>>>> have automatically uninstalled it to continue to keep things as
>>>>>>> simple
>>>>>>> as possible - which generally works well for me.  Mostly I play audio
>>>>>>> and video stuff from the CLI with mplayer but on odd occasions, like
>>>>>>> when I want to listen to a long audio book, it is more convenient to
>>>>>>> use
>>>>>>> QuodLibet which remembers where I am up to on the MP3.  However I
>>>>>>> have
>>>>>>> found that if I forget to exit QL, then mplayer does not work . . I
>>>>>>> guess there is no solution to having multiple players open at the
>>>>>>> same
>>>>>>> time - but not playing at the same time - and being able to switch
>>>>>>> between them without reinstalling PA?
>>>>>>>
>>>>>>> I also quite frequently have problems with audio on Chrome...
>>>>>>
>>>>>>
>>>>>>
>>>>>> Phil, my environment and use of it is somewhat different than yours.
>>>>>>
>>>>>> I use Ubuntu; leave Pulse in place; use the standard video application
>>>>>> (totem, I think) for video and Guayadeque for audio rather than
>>>>>> mplayer
>>>>>> and
>>>>>> QuodLibet, talking directly to Alsa and to an external DAC;
>>>>>
>>>>>
>>>>>
>

Re: [Alsa-user] Fedora 23 x86_64; Pulseaudio removed; playing from multiple sources serially

[Paolo Bolzoni]

Besides, do you changed the hw:PCH with the name of your sound card?
(from aplay -l)

On Tue, Oct 6, 2015 at 5:28 PM, Paolo Bolzoni
<paolo.bolzoni.br...@gmail.com> wrote:
> Did you actually told all the applications to use that pcm?
>
> On Tue, Oct 6, 2015 at 4:41 PM, Philip Rhoades <p...@pricom.com.au> wrote:
>> Paolo,
>>
>>
>> On 2015-10-06 22:12, Paolo Bolzoni wrote:
>>>
>>> Sorry, my bad. I understood you needed to use a program that needed
>>> pulse audio without it; in this case apulse could help
>>>
>>> However, if your problem is allowing to use the sound output to
>>> multiple programs at the same time. I had a similar problem and I
>>> solved it using dmix.
>>>
>>> In my .asoundrc I had this pcm:
>>> pcm.PCH {
>>>   type asym
>>>   playback.pcm {
>>> type plug
>>> slave {
>>>   pcm {
>>> type dmix
>>> ipc_key 9175930
>>> ipc_key_add_uid true
>>> slave {
>>>   pcm "hw:PCH"
>>> }
>>>   }
>>>   }
>>>   }
>>>   capture.pcm "hw:PCH"
>>> }
>>
>>
>>
>> I replaced my simple .asoundrc:
>>
>> pcm.!default {
>> type hw
>> card 1
>> device 0
>> }
>>
>>
>> with yours but with no improvement (I even rebooted to be sure - I have had
>> so much problem with audio in the past):
>>
>> - mplayer works fine in isolation
>>
>> - YouTube works in isolation
>>
>> - Mumble works in isolation
>>
>> - If I have Mumble open, mplayer does not work
>>
>> - If I have YT running, mplayer does not work
>>
>> etc
>>
>> It would be nice to pause whatever I am using ie NOT have to exit the
>> program, and use another audio app and then come back to where I left off
>> with the first app . .
>>
>>
>> Thanks,
>>
>> Phil.
>>
>>
>>> On Tue, Oct 6, 2015 at 12:38 PM, Philip Rhoades <p...@pricom.com.au>
>>> wrote:
>>>>
>>>> Paolo,
>>>>
>>>>
>>>> On 2015-10-06 21:34, Paolo Bolzoni wrote:
>>>>>
>>>>>
>>>>> It is meant to use skype without pulse audio, but it might help you?
>>>>> https://github.com/i-rinat/apulse
>>>>
>>>>
>>>>
>>>>
>>>> I am not sure how Skype came into the discussion - I don't use it . .
>>>>
>>>> Thanks,
>>>>
>>>> Phil.
>>>>
>>>>
>>>>
>>>>
>>>>> On Tue, Oct 6, 2015 at 12:28 PM, Philip Rhoades <p...@pricom.com.au>
>>>>> wrote:
>>>>>>
>>>>>>
>>>>>> Chris,
>>>>>>
>>>>>>
>>>>>> On 2015-09-27 00:23, chris hermansen Sat wrote:
>>>>>>>
>>>>>>>
>>>>>>> Phil and list,
>>>>>>>
>>>>>>> On Sep 26, 2015 03:21, "Philip Rhoades" <p...@pricom.com.au> wrote:
>>>>>>>>
>>>>>>>>
>>>>>>>>
>>>>>>>> People,
>>>>>>>>
>>>>>>>> Years ago when I needed to simplify things to solve audio hardware
>>>>>>>> problems, I had to remove PA - and for every new version ever since I
>>>>>>>> have automatically uninstalled it to continue to keep things as
>>>>>>>> simple
>>>>>>>> as possible - which generally works well for me.  Mostly I play audio
>>>>>>>> and video stuff from the CLI with mplayer but on odd occasions, like
>>>>>>>> when I want to listen to a long audio book, it is more convenient to
>>>>>>>> use
>>>>>>>> QuodLibet which remembers where I am up to on the MP3.  However I
>>>>>>>> have
>>>>>>>> found that if I forget to exit QL, then mplayer does not work . . I
>>>>>>>> guess there is no solution to having multiple players open at the
>>>>>>>> same
>>>>>>>> time - but not playing at the same time - and being able to switch
&g

Re: [Alsa-user] Fedora 23 x86_64; Pulseaudio removed; playing from multiple sources serially - a little progress . .

[Paolo Bolzoni]

6. I found this: "NOTE: For ALSA 1.0.9rc2 and higher you don't need to
setup dmix for analogue output. Dmix is enabled by default for
soundcards which don't support hardware mixing. You still need to set
it up for digital outputs."

In my experience, this is a lie.

On Wed, Oct 7, 2015 at 2:13 PM, Philip Rhoades <p...@pricom.com.au> wrote:
> Paolo, Anders,
>
>
> On 2015-10-07 02:44, Paolo Bolzoni wrote:
>>
>> I use this default. It allows to select any pcm via environment variable
>> pcm.!default {
>>   type plug
>>   slave.pcm {
>> @func getenv
>> vars [ ALSAPCM ]
>> default "pcm.PCH"
>>   }
>> }
>>
>> As you can see the default is pcm.PCH I told you before. But it is
>> useful to be able to change the sound pcm with ease because some
>> programs do not like speak with dmix and want to speak with the
>> hardware directly. (e.g., wine)
>>
>> So, for that programs I use ALSAPCM=hw:PCH.
>>
>> But, once again, names are probably different on your system.
>
>
>
> OK, once I saw your comments about dmix I started to try and get that to
> work - what I have found so far:
>
> 1. With NO .asoundrc file - only ONE mplayer instance works
>
> 2. Using this .asoundrc:
>
> pcm.!default {
>   type plug
>   slave.pcm "dmixer"
> }
>
> pcm.dmixer  {
>   type dmix
>   ipc_key 1024
>   slave {
>   pcm "hw:1,0"
>   period_time 0
>   period_size 1024
>   buffer_size 4096
>   rate 44100
>   }
>   bindings {
>   0 0
>   1 1
>   }
> }
>
> ctl.dmixer {
>   type hw
> # card 0
>   card 1
>   device 0
> }
>
> I can get TWO mplayers going at the same time.
>
> 3. Using the .asoundrc above prevents Mumble and YouTube in Chrome working
>
> 4. With NO .asoundrc file and xfce4-mixer and alsamixer settings for Front
> Mike, Front Mike Boost, Rear Mike and Rear Mike Boost ALL muted, Mumble
> still works fine! - it looks like it is bypassing ALSA and talking directly
> to the port or something?
>
> 5. It looks like dmix setups can get quite complex - more reading is
> required I think . .
>
> 6. I found this: "NOTE: For ALSA 1.0.9rc2 and higher you don't need to setup
> dmix for analogue output. Dmix is enabled by default for soundcards which
> don't support hardware mixing. You still need to set it up for digital
> outputs." here:
>
>   http://alsa.opensrc.org/Dmix
>
> - so I'm not sure what that means for me - I am only using analogue output .
> .
>
> 7. It looks like with some correct dmix parameters for custom pcms for
> Mumble and Chrome/YouTube, I should be able to get those to work?
>
> 8. There is stuff about snd-aloop and asym that also might be relevant but I
> am a bit lost now . .
>
> 9. It looks like these problems go back a while:
>
>   https://bugzilla.redhat.com/show_bug.cgi?id=130593
>
> but my problems seems like they should be soluble by now at least - even if
> I have to set up a custom pcm for each input source . .
>
> Thanks,
>
> Phil.
> --
> Philip Rhoades
>
> PO Box 896
> Cowra  NSW  2794
> Australia
> E-mail:  p...@pricom.com.au

--
Full-scale, agent-less Infrastructure Monitoring from a single dashboard
Integrate with 40+ ManageEngine ITSM Solutions for complete visibility
Physical-Virtual-Cloud Infrastructure monitoring from one console
Real user monitoring with APM Insights and performance trend reports 
Learn More http://pubads.g.doubleclick.net/gampad/clk?id=247754911=/4140
___
Alsa-user mailing list
Alsa-user@lists.sourceforge.net
https://lists.sourceforge.net/lists/listinfo/alsa-user

Re: [Alsa-user] Fedora 23 x86_64; Pulseaudio removed; playing from multiple sources serially

[Paolo Bolzoni]

It is meant to use skype without pulse audio, but it might help you?

https://github.com/i-rinat/apulse

On Tue, Oct 6, 2015 at 12:28 PM, Philip Rhoades  wrote:
> Chris,
>
>
> On 2015-09-27 00:23, chris hermansen Sat wrote:
>> Phil and list,
>>
>> On Sep 26, 2015 03:21, "Philip Rhoades"  wrote:
>>>
>>> People,
>>>
>>> Years ago when I needed to simplify things to solve audio hardware
>>> problems, I had to remove PA - and for every new version ever since I
>>> have automatically uninstalled it to continue to keep things as simple
>>> as possible - which generally works well for me.  Mostly I play audio
>>> and video stuff from the CLI with mplayer but on odd occasions, like
>>> when I want to listen to a long audio book, it is more convenient to
>>> use
>>> QuodLibet which remembers where I am up to on the MP3.  However I have
>>> found that if I forget to exit QL, then mplayer does not work . . I
>>> guess there is no solution to having multiple players open at the same
>>> time - but not playing at the same time - and being able to switch
>>> between them without reinstalling PA?
>>>
>>> I also quite frequently have problems with audio on Chrome...
>>
>> Phil, my environment and use of it is somewhat different than yours.
>>
>> I use Ubuntu; leave Pulse in place; use the standard video application
>> (totem, I think) for video and Guayadeque for audio rather than mplayer
>> and
>> QuodLibet, talking directly to Alsa and to an external DAC;
>
>
> But I prefer the CLI . .
>
>
>> use Firefox for
>> the web in general and for YouTube and Vimeo (though not much) and use
>> Chrome only for Netflix.
>
>
> That doesn't really suit me . .
>
>
>> Given those differences, my experience with Pulse in the last several
>> releases has been problem-free. In particular, no problems of the type
>> you
>> describe.
>>
>> So my advice to you would be to try Pulse out again.
>
>
> Maybe I should at least try reinstalling Pulse and see how it goes but I
> don't like that it introduces another layer of complexity when it might
> not be necessary . . I was hoping some ALSA guru here could tell me how
> to conveniently allow multiple sound sources without PA . .
>
> Thanks,
>
> Phil.
> --
> Philip Rhoades
>
> PO Box 896
> Cowra  NSW  2794
> Australia
> E-mail:  p...@pricom.com.au
>
> --
> ___
> Alsa-user mailing list
> Alsa-user@lists.sourceforge.net
> https://lists.sourceforge.net/lists/listinfo/alsa-user

--
___
Alsa-user mailing list
Alsa-user@lists.sourceforge.net
https://lists.sourceforge.net/lists/listinfo/alsa-user

Re: [Alsa-user] Fedora 23 x86_64; Pulseaudio removed; playing from multiple sources serially

[Paolo Bolzoni]

Sorry, my bad. I understood you needed to use a program that needed
pulse audio without it; in this case apulse could help

However, if your problem is allowing to use the sound output to
multiple programs at the same time. I had a similar problem and I
solved it using dmix.

In my .asoundrc I had this pcm:
pcm.PCH {
  type asym
  playback.pcm {
type plug
slave {
  pcm {
type dmix
ipc_key 9175930
ipc_key_add_uid true
slave {
  pcm "hw:PCH"
}
  }
  }
  }
  capture.pcm "hw:PCH"
}



On Tue, Oct 6, 2015 at 12:38 PM, Philip Rhoades <p...@pricom.com.au> wrote:
> Paolo,
>
>
> On 2015-10-06 21:34, Paolo Bolzoni wrote:
>>
>> It is meant to use skype without pulse audio, but it might help you?
>> https://github.com/i-rinat/apulse
>
>
>
> I am not sure how Skype came into the discussion - I don't use it . .
>
> Thanks,
>
> Phil.
>
>
>
>
>> On Tue, Oct 6, 2015 at 12:28 PM, Philip Rhoades <p...@pricom.com.au>
>> wrote:
>>>
>>> Chris,
>>>
>>>
>>> On 2015-09-27 00:23, chris hermansen Sat wrote:
>>>>
>>>> Phil and list,
>>>>
>>>> On Sep 26, 2015 03:21, "Philip Rhoades" <p...@pricom.com.au> wrote:
>>>>>
>>>>>
>>>>> People,
>>>>>
>>>>> Years ago when I needed to simplify things to solve audio hardware
>>>>> problems, I had to remove PA - and for every new version ever since I
>>>>> have automatically uninstalled it to continue to keep things as simple
>>>>> as possible - which generally works well for me.  Mostly I play audio
>>>>> and video stuff from the CLI with mplayer but on odd occasions, like
>>>>> when I want to listen to a long audio book, it is more convenient to
>>>>> use
>>>>> QuodLibet which remembers where I am up to on the MP3.  However I have
>>>>> found that if I forget to exit QL, then mplayer does not work . . I
>>>>> guess there is no solution to having multiple players open at the same
>>>>> time - but not playing at the same time - and being able to switch
>>>>> between them without reinstalling PA?
>>>>>
>>>>> I also quite frequently have problems with audio on Chrome...
>>>>
>>>>
>>>> Phil, my environment and use of it is somewhat different than yours.
>>>>
>>>> I use Ubuntu; leave Pulse in place; use the standard video application
>>>> (totem, I think) for video and Guayadeque for audio rather than mplayer
>>>> and
>>>> QuodLibet, talking directly to Alsa and to an external DAC;
>>>
>>>
>>>
>>> But I prefer the CLI . .
>>>
>>>
>>>> use Firefox for
>>>> the web in general and for YouTube and Vimeo (though not much) and use
>>>> Chrome only for Netflix.
>>>
>>>
>>>
>>> That doesn't really suit me . .
>>>
>>>
>>>> Given those differences, my experience with Pulse in the last several
>>>> releases has been problem-free. In particular, no problems of the type
>>>> you
>>>> describe.
>>>>
>>>> So my advice to you would be to try Pulse out again.
>>>
>>>
>>>
>>> Maybe I should at least try reinstalling Pulse and see how it goes but I
>>> don't like that it introduces another layer of complexity when it might
>>> not be necessary . . I was hoping some ALSA guru here could tell me how
>>> to conveniently allow multiple sound sources without PA . .
>>>
>>> Thanks,
>>>
>>> Phil.
>>> --
>>> Philip Rhoades
>>>
>>> PO Box 896
>>> Cowra  NSW  2794
>>> Australia
>>> E-mail:  p...@pricom.com.au
>>>
>>>
>>> --
>>> ___
>>> Alsa-user mailing list
>>> Alsa-user@lists.sourceforge.net
>>> https://lists.sourceforge.net/lists/listinfo/alsa-user
>
>
> --
> Philip Rhoades
>
> PO Box 896
> Cowra  NSW  2794
> Australia
> E-mail:  p...@pricom.com.au

--
___
Alsa-user mailing list
Alsa-user@lists.sourceforge.net
https://lists.sourceforge.net/lists/listinfo/alsa-user

Re: [Alsa-user] dmix/software mixing doesn't work

[Paolo Bolzoni]

This older thread might help you, Philip Rhoades had a similar problem.

https://www.mail-archive.com/alsa-user@lists.sourceforge.net/msg31403.html

On Wed, Feb 3, 2016 at 5:16 AM, Andoru Ekkusu  wrote:
>> Run pavucontrol, and set the B85 as the default.
>>
>> Please note that PulseAudio saves the last used card for each application.
>
> Thanks for the reply. I forgot to mention in the original message that I'm
> not using PulseAudio, so it's just ALSA to set up. Any way to set the
> default soundcard, and not cause it to disable dmix?
>
>
> --
> Site24x7 APM Insight: Get Deep Visibility into Application Performance
> APM + Mobile APM + RUM: Monitor 3 App instances at just $35/Month
> Monitor end-to-end web transactions and take corrective actions now
> Troubleshoot faster and improve end-user experience. Signup Now!
> http://pubads.g.doubleclick.net/gampad/clk?id=267308311=/4140
> ___
> Alsa-user mailing list
> Alsa-user@lists.sourceforge.net
> https://lists.sourceforge.net/lists/listinfo/alsa-user

--
Site24x7 APM Insight: Get Deep Visibility into Application Performance
APM + Mobile APM + RUM: Monitor 3 App instances at just $35/Month
Monitor end-to-end web transactions and take corrective actions now
Troubleshoot faster and improve end-user experience. Signup Now!
http://pubads.g.doubleclick.net/gampad/clk?id=267308311=/4140
___
Alsa-user mailing list
Alsa-user@lists.sourceforge.net
https://lists.sourceforge.net/lists/listinfo/alsa-user

Re: [Alsa-user] two soundcards/headset. Help please

[Paolo Bolzoni]

I had a similar problem and I solved this way, in .asoundrc I put this lines:

pcm.!default {
  type plug
  slave.pcm {
@func getenv
vars [ ALSAPCM ]
default "hw:PCH"
  }
}

(Ensure the default makes sense in your system)
And so I simply change the env variable ALSAPCM to decide what sound card to use

On Wed, Jun 29, 2016 at 12:30 PM, Kristoffer Gustafsson
 wrote:
> Hi.
> I've got two soundcards.
> my usb headset, and my integrated intel card.
> but I want to change chard sometimes.
> for example, when I've got my headset I want to use that.
> can you help me?
> /Kristoffer
>
> --
> Kristoffer Gustafsson
> Salängsgatan 7a
> tel:033-12 60 93
> mobil: 0730-500934
>
> --
> Attend Shape: An AT Tech Expo July 15-16. Meet us at AT Park in San
> Francisco, CA to explore cutting-edge tech and listen to tech luminaries
> present their vision of the future. This family event has something for
> everyone, including kids. Get more information and register today.
> http://sdm.link/attshape
> ___
> Alsa-user mailing list
> Alsa-user@lists.sourceforge.net
> https://lists.sourceforge.net/lists/listinfo/alsa-user

--
Attend Shape: An AT Tech Expo July 15-16. Meet us at AT Park in San
Francisco, CA to explore cutting-edge tech and listen to tech luminaries
present their vision of the future. This family event has something for
everyone, including kids. Get more information and register today.
http://sdm.link/attshape
___
Alsa-user mailing list
Alsa-user@lists.sourceforge.net
https://lists.sourceforge.net/lists/listinfo/alsa-user

Re: [Alsa-user] Please increase size limit of mailing list

[Paolo Bolzoni]

On Thu, Apr 13, 2017 at 3:49 PM, Paul Menzel  wrote:
> Dear ALSA folks,

> Yesterday I sent a message, which got moderated, because my attachment
> containing debugging information is too big.

>> Message body is too big: 178116 bytes with a limit of 60 KB

> Could the limit please be increased to 500 KB or something similar?
> That’d be great.

Dear Menzel,

While I do agree that 60Kb is quite restrictive, what about simply upload the
debug information elsewhere and link them in the email?

Cheers.
Paolo

--
Check out the vibrant tech community on one of the world's most
engaging tech sites, Slashdot.org! http://sdm.link/slashdot
___
Alsa-user mailing list
Alsa-user@lists.sourceforge.net
https://lists.sourceforge.net/lists/listinfo/alsa-user

Re: [Alsa-user] Please increase size limit of mailing list

[Paolo Bolzoni]

On Thu, Apr 13, 2017 at 4:04 PM, Paul Menzel <pmen...@molgen.mpg.de> wrote:
> Dear Paolo,
> On 04/13/17 15:57, Paolo Bolzoni wrote:
>> On Thu, Apr 13, 2017 at 3:49 PM, Paul Menzel wrote:

>>> Yesterday I sent a message, which got moderated, because my attachment
>>> containing debugging information is too big.

>>>> Message body is too big: 178116 bytes with a limit of 60 KB

>>> Could the limit please be increased to 500 KB or something similar?
>>> That’d be great.

>> While I do agree that 60Kb is quite restrictive, what about simply upload
>> the
>> debug information elsewhere and link them in the email?

> That would indeed be a workaround, with the following disadvantages in my
> opinion.
>
> 1.  Using “small” attachments, the sender normally doesn’t think of the
> limitation, and therefore has to resend the message.
> 2.  The information is not in one place. Having external links is bad
> for people reading messages offline.
> 3.  What upload service should be used, which everyone trusts or can
> access?

Dear Menzel,

I have to underline I am not one of the mailing list administrators.
Mine was nothing more a suggestion for a workaround.

About point (3) however, there is an obvious solution. Sign both the
file and your email via PGP, if the file get tampered in any way it
will be detected.

Cheers,
Paolo

--
Check out the vibrant tech community on one of the world's most
engaging tech sites, Slashdot.org! http://sdm.link/slashdot
___
Alsa-user mailing list
Alsa-user@lists.sourceforge.net
https://lists.sourceforge.net/lists/listinfo/alsa-user

[Alsa-user] Using compressor + limiter with 6 channel sounds

[Paolo Bolzoni]

Dear list,
This is my .asoundrc, and the pair compressor + limiter works fine for
stereo input:

--- 8
pcm.ladcomp_compressor {
  type ladspa
  slave.pcm ladcomp_limiter;
  path /usr/lib/ladspa;
  plugins [{
  label dysonCompress
  input {
  #peak limit, release time, fast ratio, ratio
  controls [0 1 0.5 0.99]  }
  }]
}

pcm.ladcomp_limiter {
  type ladspa
  slave.pcm plughw:Audigy2;
  path /usr/lib/ladspa;
  plugins [{
  label fastLookaheadLimiter
  input {
   #InputGain(Db) -20 - +20 ; Limit (db) -20 - 0 ; Release
time (s) 0.01 - 2
   controls [ 20 -1 0.8  ]  }
  }]
}
8 ---

Unfortunately, it does not work for 6 channels.

I.e., this one works fine (I setup my system to use a certain PCM
via environment variable):
$ ALSAPCM='pcm.ladcomp_compressor' speaker-test -c 2 -t wav -l 5

This one does not and you hear only the left and right channels.
$ ALSAPCM='pcm.ladcomp_compressor' speaker-test -c 6 -t wav -l 5

Of course the hardware is wired correctly and this one works as expected:
$ ALSAPCM='pcm.surround51' speaker-test -c 6 -t wav -l 5


Since normally you have to select surround51 manually I thought that the
problem could be in the output pcm, and I rewrote like this. But it is the same:

--- 8
#[...] analogous 51 compressor omitted

pcm.plug51 {
  type plug
  slave.pcm surround51
  slave.channels 6
}

pcm.ladcomp_limiter51 {
  type ladspa
  slave.pcm plug51
  path /usr/lib/ladspa
  plugins [{
  label fastLookaheadLimiter
  input {
   #InputGain(Db) -20 - +20 ; Limit (db) -20 - 0 ; Release
time (s) 0.01 - 2
   controls [ 20 -1 0.8  ]  }
  }]
}
8 ---

Is there a way? Any insight?

Yours faithfully,
Paolo

--
October Webinars: Code for Performance
Free Intel webinars can help you accelerate application performance.
Explore tips for MPI, OpenMP, advanced profiling, and more. Get the most from 
the latest Intel processors and coprocessors. See abstracts and register 
http://pubads.g.doubleclick.net/gampad/clk?id=60134071iu=/4140/ostg.clktrk
___
Alsa-user mailing list
Alsa-user@lists.sourceforge.net
https://lists.sourceforge.net/lists/listinfo/alsa-user

Re: [Alsa-user] Using compressor + limiter with 6 channel sounds

[Paolo Bolzoni]

I guess I can split the six channels in three limiters using the multi
plugin, but how I can join them back?
In the ladspa are other limiters, but they do not have the gain
control so they do not actually increase the
volume. Just adding plain 20db of amplification distorts the sound...
Maybe there is an limited amplifier
or something?

I am trying to set-up a chain of plugins to always get the max volume...


On Sun, Oct 13, 2013 at 12:58 AM, Uwe upu...@googlemail.com wrote:
 the problem with the configuration, I think, is: the plugins can handle a
 maximum of 2 channels.

 unfortunately, I cannot tell you exactly how, but it *should* be possible to
 route 3 pairs of channels to separate instances of the plugins.

 such a setup might also make sense musically. it might be a good idea to
 setup separate plugins for center and bass because the signal in the
 channels differs significantly from the left/right channels in frequency
 range and level.

 have fun, Uwe


 2013/10/12 Paolo Bolzoni paolo.bolzoni.br...@gmail.com

 Dear list,
 This is my .asoundrc, and the pair compressor + limiter works fine for
 stereo input:

 --- 8
 pcm.ladcomp_compressor {
   type ladspa
   slave.pcm ladcomp_limiter;
   path /usr/lib/ladspa;
   plugins [{
   label dysonCompress
   input {
   #peak limit, release time, fast ratio, ratio
   controls [0 1 0.5 0.99]  }
   }]
 }

 pcm.ladcomp_limiter {
   type ladspa
   slave.pcm plughw:Audigy2;
   path /usr/lib/ladspa;
   plugins [{
   label fastLookaheadLimiter
   input {
#InputGain(Db) -20 - +20 ; Limit (db) -20 - 0 ; Release
 time (s) 0.01 - 2
controls [ 20 -1 0.8  ]  }
   }]
 }
 8 ---

 Unfortunately, it does not work for 6 channels.

 I.e., this one works fine (I setup my system to use a certain PCM
 via environment variable):
 $ ALSAPCM='pcm.ladcomp_compressor' speaker-test -c 2 -t wav -l 5

 This one does not and you hear only the left and right channels.
 $ ALSAPCM='pcm.ladcomp_compressor' speaker-test -c 6 -t wav -l 5

 Of course the hardware is wired correctly and this one works as expected:
 $ ALSAPCM='pcm.surround51' speaker-test -c 6 -t wav -l 5


 Since normally you have to select surround51 manually I thought that the
 problem could be in the output pcm, and I rewrote like this. But it is the
 same:

 --- 8
 #[...] analogous 51 compressor omitted

 pcm.plug51 {
   type plug
   slave.pcm surround51
   slave.channels 6
 }

 pcm.ladcomp_limiter51 {
   type ladspa
   slave.pcm plug51
   path /usr/lib/ladspa
   plugins [{
   label fastLookaheadLimiter
   input {
#InputGain(Db) -20 - +20 ; Limit (db) -20 - 0 ; Release
 time (s) 0.01 - 2
controls [ 20 -1 0.8  ]  }
   }]
 }
 8 ---

 Is there a way? Any insight?

 Yours faithfully,
 Paolo


 --
 October Webinars: Code for Performance
 Free Intel webinars can help you accelerate application performance.
 Explore tips for MPI, OpenMP, advanced profiling, and more. Get the most
 from
 the latest Intel processors and coprocessors. See abstracts and register 

 http://pubads.g.doubleclick.net/gampad/clk?id=60134071iu=/4140/ostg.clktrk
 ___
 Alsa-user mailing list
 Alsa-user@lists.sourceforge.net
 https://lists.sourceforge.net/lists/listinfo/alsa-user




 --
 這只是警告訊息。

 --
 October Webinars: Code for Performance
 Free Intel webinars can help you accelerate application performance.
 Explore tips for MPI, OpenMP, advanced profiling, and more. Get the most
 from
 the latest Intel processors and coprocessors. See abstracts and register 
 http://pubads.g.doubleclick.net/gampad/clk?id=60134071iu=/4140/ostg.clktrk
 ___
 Alsa-user mailing list
 Alsa-user@lists.sourceforge.net
 https://lists.sourceforge.net/lists/listinfo/alsa-user


--
October Webinars: Code for Performance
Free Intel webinars can help you accelerate application performance.
Explore tips for MPI, OpenMP, advanced profiling, and more. Get the most from 
the latest Intel processors and coprocessors. See abstracts and register 
http://pubads.g.doubleclick.net/gampad/clk?id=60134071iu=/4140/ostg.clktrk
___
Alsa-user mailing list
Alsa-user@lists.sourceforge.net
https://lists.sourceforge.net/lists/listinfo/alsa-user

Re: [Alsa-user] Using compressor + limiter with 6 channel sounds

[Paolo Bolzoni]

Sorry, I am not much in this stuff... But I am reading now that AGC is
Automatic gain control
and it seems what I would like to do.

I am trying to have a more or less constant volume of the speaker
output. The configuration
I posted in the first email works in this sense for stereo; it does
not work for 5.1 audio.

On Sun, Oct 13, 2013 at 11:03 PM, Robert M. Riches Jr.
rm.ric...@jacob21819.net wrote:
 I probably don't know any answers, but would like to make sure I
 at least understand the question.  Are you trying do AGC on a
 pair-wise basis?  Or, is it something else you're trying to do?

 Thanks,

 Robert Riches


 Date: Sun, 13 Oct 2013 15:02:42 +0200
 From: Paolo Bolzoni paolo.bolzoni.br...@gmail.com
 To: Uwe upu...@googlemail.com
 Cc: alsa-user@lists.sourceforge.net

 I guess I can split the six channels in three limiters using the multi
 plugin, but how I can join them back?
 In the ladspa are other limiters, but they do not have the gain
 control so they do not actually increase the
 volume. Just adding plain 20db of amplification distorts the sound...
 Maybe there is an limited amplifier
 or something?

 I am trying to set-up a chain of plugins to always get the max volume...


 On Sun, Oct 13, 2013 at 12:58 AM, Uwe upu...@googlemail.com wrote:
  the problem with the configuration, I think, is: the plugins can handle a
  maximum of 2 channels.
 
  unfortunately, I cannot tell you exactly how, but it *should* be possible 
  to
  route 3 pairs of channels to separate instances of the plugins.
 
  such a setup might also make sense musically. it might be a good idea to
  setup separate plugins for center and bass because the signal in the
  channels differs significantly from the left/right channels in frequency
  range and level.
 
  have fun, Uwe
 
 
  2013/10/12 Paolo Bolzoni paolo.bolzoni.br...@gmail.com
 
  Dear list,
  This is my .asoundrc, and the pair compressor + limiter works fine for
  stereo input:
 
  --- 8
  pcm.ladcomp_compressor {
type ladspa
slave.pcm ladcomp_limiter;
path /usr/lib/ladspa;
plugins [{
label dysonCompress
input {
#peak limit, release time, fast ratio, ratio
controls [0 1 0.5 0.99]  }
}]
  }
 
  pcm.ladcomp_limiter {
type ladspa
slave.pcm plughw:Audigy2;
path /usr/lib/ladspa;
plugins [{
label fastLookaheadLimiter
input {
 #InputGain(Db) -20 - +20 ; Limit (db) -20 - 0 ; Release
  time (s) 0.01 - 2
 controls [ 20 -1 0.8  ]  }
}]
  }
  8 ---
 
  Unfortunately, it does not work for 6 channels.
 
  I.e., this one works fine (I setup my system to use a certain PCM
  via environment variable):
  $ ALSAPCM='pcm.ladcomp_compressor' speaker-test -c 2 -t wav -l 5
 
  This one does not and you hear only the left and right channels.
  $ ALSAPCM='pcm.ladcomp_compressor' speaker-test -c 6 -t wav -l 5
 
  Of course the hardware is wired correctly and this one works as expected:
  $ ALSAPCM='pcm.surround51' speaker-test -c 6 -t wav -l 5
 
 
  Since normally you have to select surround51 manually I thought that the
  problem could be in the output pcm, and I rewrote like this. But it is the
  same:
 
  --- 8
  #[...] analogous 51 compressor omitted
 
  pcm.plug51 {
type plug
slave.pcm surround51
slave.channels 6
  }
 
  pcm.ladcomp_limiter51 {
type ladspa
slave.pcm plug51
path /usr/lib/ladspa
plugins [{
label fastLookaheadLimiter
input {
 #InputGain(Db) -20 - +20 ; Limit (db) -20 - 0 ; Release
  time (s) 0.01 - 2
 controls [ 20 -1 0.8  ]  }
}]
  }
  8 ---
 
  Is there a way? Any insight?
 
  Yours faithfully,
  Paolo
 
 
  --
  October Webinars: Code for Performance
  Free Intel webinars can help you accelerate application performance.
  Explore tips for MPI, OpenMP, advanced profiling, and more. Get the most
  from
  the latest Intel processors and coprocessors. See abstracts and register 
 
  http://pubads.g.doubleclick.net/gampad/clk?id=60134071iu=/4140/ostg.clktrk
  ___
  Alsa-user mailing list
  Alsa-user@lists.sourceforge.net
  https://lists.sourceforge.net/lists/listinfo/alsa-user
 
 
 
 
  --
  
 
  --
  October Webinars: Code for Performance
  Free Intel webinars can help you accelerate application performance.
  Explore tips for MPI, OpenMP, advanced profiling, and more. Get the most
  from
  the latest Intel processors and coprocessors. See abstracts and register 
  http://pubads.g.doubleclick.net/gampad/clk?id=60134071iu=/4140/ostg.clktrk
  ___
  Alsa-user mailing list
  Alsa-user@lists.sourceforge.net
  https://lists.sourceforge.net/lists/listinfo

Re: [Alsa-user] Using compressor + limiter with 6 channel sounds

[Paolo Bolzoni]

Ideally they should be all synced.

On Mon, Oct 14, 2013 at 12:39 AM, Ralf Mardorf
ralf.mard...@alice-dsl.net wrote:
 On Mon, 2013-10-14 at 00:25 +0200, Paolo Bolzoni wrote:
 On Sun, Oct 13, 2013 at 11:03 PM, Robert M. Riches Jr.
 rm.ric...@jacob21819.net wrote:
  I probably don't know any answers, but would like to make sure I
  at least understand the question.  Are you trying do AGC on a
  pair-wise basis?  Or, is it something else you're trying to do?

 I am trying to have a more or less constant volume of the speaker
 output. The configuration I posted in the first email works in this
 sense for stereo; it does not work for 5.1 audio.

 IIUC Robert want's to know if you want channel pairs synced to each
 other, or if all channels should compress independent or if all channels
 should be synced to each other.



 --
 October Webinars: Code for Performance
 Free Intel webinars can help you accelerate application performance.
 Explore tips for MPI, OpenMP, advanced profiling, and more. Get the most from
 the latest Intel processors and coprocessors. See abstracts and register 
 http://pubads.g.doubleclick.net/gampad/clk?id=60134071iu=/4140/ostg.clktrk
 ___
 Alsa-user mailing list
 Alsa-user@lists.sourceforge.net
 https://lists.sourceforge.net/lists/listinfo/alsa-user

--
October Webinars: Code for Performance
Free Intel webinars can help you accelerate application performance.
Explore tips for MPI, OpenMP, advanced profiling, and more. Get the most from 
the latest Intel processors and coprocessors. See abstracts and register 
http://pubads.g.doubleclick.net/gampad/clk?id=60134071iu=/4140/ostg.clktrk
___
Alsa-user mailing list
Alsa-user@lists.sourceforge.net
https://lists.sourceforge.net/lists/listinfo/alsa-user

Re: [Alsa-user] re-route sound output to line in (for e.g. recording, audio visualization)

[Paolo Bolzoni]

On Sat, Nov 16, 2013 at 11:09 PM, D.T.au dan...@iki.fi wrote:
 i managed earlier to create a loopback device, and it worked so-so.

 but i have one stubborn application that insists on using the default 
 soundcard's first capture device (sndpeek).

I use this default device, that can actually be pointed to any other
via environment variable.
I am not sure if it can help for your other problems, but in my system
all the application
used the correct pcm indicated by ALSAPCM.

(Change the hw:Audigy2 with your default)

 pcm.!default {
   type plug
   slave.pcm {
 @func getenv
 vars [ ALSAPCM ]
 default hw:Audigy2
   }
 }

--
DreamFactory - Open Source REST  JSON Services for HTML5  Native Apps
OAuth, Users, Roles, SQL, NoSQL, BLOB Storage and External API Access
Free app hosting. Or install the open source package on any LAMP server.
Sign up and see examples for AngularJS, jQuery, Sencha Touch and Native!
http://pubads.g.doubleclick.net/gampad/clk?id=63469471iu=/4140/ostg.clktrk
___
Alsa-user mailing list
Alsa-user@lists.sourceforge.net
https://lists.sourceforge.net/lists/listinfo/alsa-user

[Alsa-user] Avoid that Pulse blocks the hardware

[Paolo Bolzoni]

Dear list,

I would like to setup my card so programs, like pulse, do not
lock the hardware.

I am not sure why it happens, this is my asound.conf file:

pcm.!default {
type plug
slave.pcm {
@func getenv
vars [ ALSAPCM ]
default pcm.PCH
}
}


pcm.PCH {
  type asym

  playback.pcm {
type dmix
ipc_key 9175930
ipc_key_add_uid true
slave {
  pcm hw:PCH
}
  }

  capture.pcm {
type dsnoop
ipc_key 2412430
ipc_key_add_uid true
slave {
pcm hw:PCH
channels 2
}
  }
}

The idea is to use a software mixer both for microphone and
speakers via dsnoop and dmix respectably. While being able to
select the hardware directly for programs who needs it (for
example wine programs don't like to speak to dmix it seems)

$ lspci | grep -i aud
00:03.0 Audio device: Intel Corporation Xeon E3-1200 v3/4th Gen
  Core Processor HD Audio Controller (rev 06)
00:1b.0 Audio device: Intel Corporation 8 Series/C220 Series
  Chipset High Definition Audio Controller (rev 05)

I see the problem mainly when I try to use Mumble and
Simplescreenrecorder at the same time. I do this because I want
to share the screen with a colleague when we are far away.

Any idea how to fix or it is simply better to buy a cheap usb
card to connect my microphone to it?


Yours faithfully,
Paolo

--
Download BIRT iHub F-Type - The Free Enterprise-Grade BIRT Server
from Actuate! Instantly Supercharge Your Business Reports and Dashboards
with Interactivity, Sharing, Native Excel Exports, App Integration  more
Get technology previously reserved for billion-dollar corporations, FREE
http://pubads.g.doubleclick.net/gampad/clk?id=157005751iu=/4140/ostg.clktrk
___
Alsa-user mailing list
Alsa-user@lists.sourceforge.net
https://lists.sourceforge.net/lists/listinfo/alsa-user

Re: [Alsa-user] Function providing current output level (like a VU meter)

[Paolo Bolzoni]

I am missing something,  how plugins like the ladspa
fastLookaheadLimiter can work if they cannot detect current output
intensity?

On Tue, Mar 3, 2015 at 10:58 AM, Ray r...@renegade.zapto.org wrote:
 Am 2015-03-03 10:47, schrieb Clemens Ladisch:
 Ray wrote:
 is there some ALSA function which I can use to get the current music
 output level from the soundcard output?

 No; be default, samples are written by the application directly into
 the
 DMA buffer, and there is no opportunity for anybode else to listen in.

 Ok, this puts an end to my efforts in userland, I suppose. How about
 kernel-land then?

 Best,
 Ray

 --
 Dive into the World of Parallel Programming The Go Parallel Website, sponsored
 by Intel and developed in partnership with Slashdot Media, is your hub for all
 things parallel software development, from weekly thought leadership blogs to
 news, videos, case studies, tutorials and more. Take a look and join the
 conversation now. http://goparallel.sourceforge.net/
 ___
 Alsa-user mailing list
 Alsa-user@lists.sourceforge.net
 https://lists.sourceforge.net/lists/listinfo/alsa-user

--
Dive into the World of Parallel Programming The Go Parallel Website, sponsored
by Intel and developed in partnership with Slashdot Media, is your hub for all
things parallel software development, from weekly thought leadership blogs to
news, videos, case studies, tutorials and more. Take a look and join the 
conversation now. http://goparallel.sourceforge.net/
___
Alsa-user mailing list
Alsa-user@lists.sourceforge.net
https://lists.sourceforge.net/lists/listinfo/alsa-user