[Alsa-user] Native Instruments Komplete Audio 6 SPDIF-out issue
Hi all,
I just got a Native Instruments Komplete Audio 6 USB Audio Interface
which is connected to my AV-receiver via the SPDI-out as otherwise I
get lots of static noise.
While this works flawlessly in Windows 10, ALSA in Linux Mint 18.1
does not set up the SPDIF by default. Only an analog output with the
respective profiles are shown in the end in the Pulse Audio controls.
"aplay -l" lists the card as follows:
Card 1: K6 [Komplete Audio 6], Device 0: USB Audio [USB Audio]
Sub-Devices: 1/1
Sub-Device #0: subdevice #0
"aplay -L" includes also the SPDIF-out I want to use:
iec958:CARD=K6,DEV=0
Komplete Audio 6, USB Audio
IEC958 (S/PDIF) Digital Audio Output
But the attempt to play a wave file ends like this:
"aplay -D iec958:CARD=K6,DEV=0 test.wav"
Playback: WAVE 'test.wav' : Signed 16 bit Little Endian, Rate: 44100 Hz, stereo
aplay: set_params:1233: Sample-Format not supported
Available formats:
- S32_LE
Now my only attempt to define the device via /etc/asound.conf looks like this:
pcm.SPDIF {
type iec958
card 1
device 0
}
but does not work. I understand that I maybe have to define some
format conversion...
Can anyone give me hint if this is possible at all and if yes, how I
can set up ALSA for that?
Help is much appreciated!
Thanks and cheers,
Philip
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[Alsa-user] Native Instruments Komplete Audio 6 SPDIF-out issue
Hi all,
I just got a Native Instruments Komplete Audio 6 USB Audio Interface
which is connected to my AV-receiver via the SPDI-out as otherwise I
get lots of static noise.
While this works flawlessly in Windows 10, ALSA in Linux Mint 18.1
does not set up the SPDIF by default. Only an analog output with the
respective profiles are shown in the end in the Pulse Audio controls.
"aplay -l" lists the card as follows:
Card 1: K6 [Komplete Audio 6], Device 0: USB Audio [USB Audio]
Sub-Devices: 1/1
Sub-Device #0: subdevice #0
"aplay -L" includes also the SPDIF-out I want to use:
iec958:CARD=K6,DEV=0
Komplete Audio 6, USB Audio
IEC958 (S/PDIF) Digital Audio Output
But the attempt to play a wave file ends like this:
"aplay -D iec958:CARD=K6,DEV=0 test.wav"
Playback: WAVE 'test.wav' : Signed 16 bit Little Endian, Rate: 44100 Hz, stereo
aplay: set_params:1233: Sample-Format not supported
Available formats:
- S32_LE
Now my only attempt to define the device via /etc/asound.conf looks like this:
pcm.SPDIF {
type iec958
card 1
device 0
}
but does not work. I understand that I maybe have to define some
format conversion...
Can anyone give me hint if this is possible at all and if yes, how I
can set up ALSA for that?
Help is much appreciated!
Thanks and cheers,
Philip
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Re: [Alsa-user] Native Instruments Komplete Audio 6 SPDIF-out issue
Hi Ralf, thanks for the hints and forum-link! I am now reading some documentation on how to define a profile set for pulseaudio which hopefully will add the digital out to the selection. The command speaker-test -D iec958:CARD=K6,DEV=0 -F S32_LE -c 6 works and I get sound on channels 5 an 6 so I am confident. As I actually only need sound from Spotify and the browser in Linux I'll try this way first before switching to another sound server. The alternative to the Komplete Audio would have been the Focusrite Scarlett 6i6 but I couldn't find a reliable source that this one works better in Linux. Even a PCIe interface would do the job as I really only need a good ASIO driver (and the digital out)... well, I can still exchange the device if it won't work. Static noise might have been the wrong term; I meant a ground loop which I can't get rid of except using the digital out. This might well be related to the power lines in my appartment being fishy (using a specific light switch will cause audio drops at the AV-receiver...). Cheers, Philip 2017-08-09 21:51 GMT+02:00 Ralf Mardorf <ralf.mard...@alice-dsl.net>: > On Wed, 9 Aug 2017 20:42:30 +0200, Philip wrote: >>ALSA in Linux Mint 18.1 does not set up the SPDIF by default. Only an >>analog output with the respective profiles are shown in the end in the >>Pulse Audio controls. > > Hi, > > perhaps not an ALSA issue, but a pulseaudio issue? > > Seemingly the device should work OOTB: > > https://linuxmusicians.com/viewtopic.php?t=12161#p50901 > > Consider to subscribe to this forum and ask there. > > FWIW I'm not using the same device as you, but among PCIe and PCI audio > interfaces, I'm using one USB class compliant device, too. It's a > Focusrite Scarlett 18i20 2nd gen and S/PDIF works OOTB. > > I'm using ALSA and jackd 2 (not jackdbus) only. For packages with a hard > dependency on pulseaudio I build an empty dummy package > "pulseaudio". This works for at least Ubuntu and Arch Linux without > issues. > > On Arch Linux I'm using https://github.com/i-rinat/apulse for Firefox, > on Ubuntu I don't have Firefox installed and nothing else I'm using > requires pulseaudio. > > AFAIK you could disable pulseaudio temporarily, so for testing purpose > consider to at least disable pulseaudio, remove your asound.conf entry > and try again using plain alsa and/or jackd 1 or 2. > > Btw. neither my consumer, pro-sumer, nor my professional audio devices > cause static noise when using balanced as well as unbalanced analog IOs. > For some unknown reason the headphone output of my RME PCIe card had > unbearable computer noise in the audio signal, some day it occurred, > stayed for months and then disappeared. I suspect a cable issue, but > can't say for sure. However, I'm using a new headphone cable, as well > as a new computer, since my old computer anyway was fishy and outdated. > > Even if cheapest consumer gear does cause static noise, something is > fishy. > > Regards, > Ralf > > -- > 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 -- 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] (no subject)
Hello, i am trying to get the terratec DMX 6Fire 24/96 card to work on linux using alsa. the playback seems alright, but the Microphone In input doesn't work. Is that because i misconfigured something, or are there problems with the inputs on the DMX 6Fire? do the other analog inputs work? _ Send and receive Hotmail on your mobile device: http://mobile.msn.com --- This sf.net email is sponsored by: OSDN - Tired of that same old cell phone? Get a new here for FREE! https://www.inphonic.com/r.asp?r=sourceforge1refcode1=vs3390 ___ Alsa-user mailing list [EMAIL PROTECTED] https://lists.sourceforge.net/lists/listinfo/alsa-user
[Alsa-user] xrun error messages
Hi Folks, I have managed to successfully run the most uptodate alsa9 driver on a DELL latitude cpiA366XT with the Neomagic 256AV sound chip. However, I have a teething problem. When I run a dvd the flow is continually interrupted by:- alsa-space: xrun of at least 2.120 msecs. resetting stream 0.0% 0 0 0% alsa-play: xrun of at least 0.247 msecs. resetting stream% 10.1% 0 0 0% alsa-space: xrun of at least 106.110 msecs. resetting stream4.0% 0 0 0% alsa-space: xrun of at least 82.396 msecs. resetting stream14.1% 0 0 0% alsa-space: xrun of at least 154.871 msecs. resetting stream1.3% 0 0 0% This resetting of the stream upsets the video and sound output. Note I think the Neomagic chip is a combination of audio and video. Any ideas on how to stop this happening? Cheers -Phil -- Philip J. ClarkMS41 (Bristol), SLAC tel: 1-650-926-3761Stanford University fax: 1-650-926-3767P.O. Box 20450 [EMAIL PROTECTED] Stanford, California 94309 --- This sf.net email is sponsored by:ThinkGeek Welcome to geek heaven. http://thinkgeek.com/sf ___ Alsa-user mailing list [EMAIL PROTECTED] https://lists.sourceforge.net/lists/listinfo/alsa-user
[Alsa-user] xrun messages
Hi Folks, I have managed to successfully run the most uptodate alsa9 driver on a DELL latitude cpiA366XT with the Neomagic 256AV sound chip. However, I have a teething problem. When I run a dvd the flow is continually interrupted by:- alsa-space: xrun of at least 2.120 msecs. resetting stream 0.0% 0 0 0% alsa-play: xrun of at least 0.247 msecs. resetting stream% 10.1% 0 0 0% alsa-space: xrun of at least 106.110 msecs. resetting stream4.0% 0 0 0% alsa-space: xrun of at least 82.396 msecs. resetting stream14.1% 0 0 0% alsa-space: xrun of at least 154.871 msecs. resetting stream1.3% 0 0 0% This resetting of the stream upsets the video and sound output. Note I think the Neomagic chip is a combination of audio and video. Any ideas on how to stop this happening? Cheers -Phil -- Philip J. ClarkMS41 (Bristol), SLAC tel: 1-650-926-3761Stanford University fax: 1-650-926-3767P.O. Box 20450 [EMAIL PROTECTED] Stanford, California 94309 --- This sf.net email is sponsored by:ThinkGeek Welcome to geek heaven. http://thinkgeek.com/sf ___ Alsa-user mailing list [EMAIL PROTECTED] https://lists.sourceforge.net/lists/listinfo/alsa-user
[Alsa-user] YMF744 and OPL3 FM Synthesis.(Long)
-- I have been working on and off for sometime trying to get OPL3 FM Synth (NOT the WaveTable Synth) to work under ALSA with a AOpen (I'm fairly sure) Ymf744B Card. I have even gone as far as writing some simple code to communicate directly with the /dev/dmfm (/dev/sound/dmfm) device and thus at least verify that the OPL3 was reachable. My basic setup is the default asound startup script, modified to modprobe oss modules and snd-opl3-synth. The kernel driver is the latest CVS as of The End of August, and the sbiload is the latest cvs version. When I am trying to get the OPL3 FM Synth to work, I use a simple script to run sbiload with the std and drum patches as follows: /usr/bin/sbiload -p65:0 -v $1 --opl3 \ /usr/share/alsa/banks/opl3/drums.o3 \ /usr/share/alsa/banks/opl3/std.o3 Once the Patches are loaded, then I try to run pmidi and playmidi to see if ALSA and OSS midi is now working. pmidi -p 65:0 W2Humans.mid and playmidi [-r] W2Humans.mid Both seem to hang for 3 minutes(the length of the clip) and playmidi -r seems to indicate that indeed the player thinks the midi file is playing. No sound. :( It seemed like the midi channel might be muted. So I have double checked to make sure all the mixer controls were turned on and up, just in case. I recall some previous post mentioning a synth, midi, or a FM mixer control; none of which I can manage to find, and the ones I have found (master*,wave,pcm,external,cd,etc) are unmuted and have been tested turned all the way up. If anyone can help/share experiences it would be appreciated. It seems that there is at least one person out there on this list with a similar problem with the same card(first appearing in April), and I was curious how far he had managed to get with the OPL3. I have tried to include text/links of various pieces of information, which might help in a diagnosis. And a Test program for accessing the direct FM device, in a last ditch attempt to make sure I could talk to the OPL3 at least. The error output I get is included. Thanks Philip Thiem Loaded Modules -- snd-seq-midi snd-seq-oss snd-seq-midi-event snd-pcm-oss snd-mixer-oss snd-opl3-synth snd-seq-instr snd-seq-midi-emul snd-seq snd-ainstr-fm snd-ymfpci snd-pcm snd-mpu401-uart snd-rawmidi snd-opl3-lib snd-hwdep snd-seq-device snd-timer snd-ac97-codec snd soundcore /etc/modutils.conf(relavent) -- ### #ALSA Core ### # ALSA portion alias char-major-116 snd options snd snd_cards_limit=1 snd_device_mode=660 snd_device_gid=29 # OSS/Free portion alias char-major-14 soundcore ### #ALSA Drivers ### # ALSA portion alias snd-card-0 snd-ymfpci # OSS/Free portion alias sound-slot-0 snd-card-0 alias sound-service-0-0 snd-mixer-oss alias sound-service-0-1 snd-seq-oss alias sound-service-0-3 snd-pcm-oss alias sound-service-0-8 snd-seq-oss alias sound-service-0-12 snd-pcm-oss pmidi -l gives -- Port Client name Port name 64:0 External MIDI 0 MIDI 0-0 65:0 OPL3 FM synth OPL3 Port /proc/pci and /proc/ioports output -- Bus 0, device 9, function 0: Multimedia audio controller: Yamaha Corporation YMF-744B [DS-1S Audio Controller] (rev 2). IRQ 5. Master Capable. Latency=32. Min Gnt=5.Max Lat=25. Non-prefetchable 32 bit memory at 0xdb00 [0xdb007fff]. I/O at 0xdc00 [0xdc3f]. I/O at 0xe000 [0xe003]. dc00-dc3f : Yamaha Corporation YMF-744B [DS-1S Audio Controller] dc00-dc01 : OPL2/3 (left) dc02-dc03 : OPL2/3 (right) dc20-dc21 : MPU401 UART e000-e003 : Yamaha Corporation YMF-744B [DS-1S Audio Controller] e400-e43f : 3Com Corporation 3c905 100BaseTX [Boomerang] e400-e43f : 00:0a.0 /proc/asound/version -- Advanced Linux Sound Architecture Driver Version 0.9.0rc3. Compiled on Sep 4 2002 for kernel 2.4.19-xfs with versioned symbols. /prov/asound/seq/clients -- Client info cur clients : 4 peak clients : 5 max clients : 192 Client 0 : System [Kernel] Port 0 : Timer (Rwe-) Port 1 : Announce (R-e-) Connecting To: 63:0 Client 63 : OSS sequencer [Kernel] Port 0 : Receiver (-we-) Connected From: 0:1 Client 64 : External MIDI 0 [Kernel] Port 0 : MIDI 0-0 (RWeX) Client 65 : OPL3 FM synth [Kernel] Port 0 : OPL3 Port (-We-) Port 1 : OPL3 OSS Port (-we
[Alsa-user] alsa, ice1712 and aumix
Hi, i'm trying to get the Terratec DMX6Fire 24/96 to record via alsa oss compatibilty mode. For some reason aumix and oss can't find any recording source, i get: [me@leenawx]# aumix aumix: no device found and record says: [me@leenawx]# record mixer: havn't found device 'line' mixer: available: my modules.conf entries look like this: # ALSA portion alias sound snd-ice1712 alias char-major-116 snd alias snd-card-0 snd-ice1712 alias char-major-14 soundcore alias sound-slot-0 snd-card-0 alias sound-service-0-0 snd-mixer-oss alias sound-service-0-1 snd-seq-oss alias sound-service-0-3 snd-pcm-oss alias sound-service-0-8 snd-seq-oss alias sound-service-0-12 snd-pcm-oss post-install sound-slot-0 /bin/aumix-minimal -f /etc/.aumixrc -L /dev/null 21 || : pre-remove sound-slot-0 /bin/aumix-minimal -f /etc/.aumixrc -S /dev/null 21 || : post-install snd /usr/sbin/alsactl restore 0 playback via oss, and recordig and playback via alsa work fine, though, its only recording via oss that won't function. I think the reason it won't is that i've messed up my modules.conf entries. Can anyone help? thank you, philip _ Send and receive Hotmail on your mobile device: http://mobile.msn.com --- This sf.net email is sponsored by:ThinkGeek Welcome to geek heaven. http://thinkgeek.com/sf ___ Alsa-user mailing list [EMAIL PROTECTED] https://lists.sourceforge.net/lists/listinfo/alsa-user
[Alsa-user] alsa and mpeg4ip
Hi, i'm trying to get the Terratec DMX6Fire 24/96 to record via alsa oss compatibilty mode. For some reason aumix and oss can't find any recording source, i get: ice1712 has unconventional (as meaning of oss) mixer controls, and cannot be mapped well to oss mixer elements as default. try alsactl to save/restore the current state instead. you can modify the mapping by writing back the entry of /proc/asound/card0/oss_mixers. it would be not perfect but you can control some volumes at least. [me@leenawx]# aumix aumix: no device found and record says: [me@leenawx]# record mixer: havn't found device 'line' mixer: available: not sure what record program does. perhaps checking an appropriate mixer device? the other oss program, for example, rec included in sox should work. ciao, Takashi Hi again, thanks for the advice, oss compatibility is working quite well now. but when i try to record sound using mpeg4ip (mpeg4ip.sourceforge.net), the computer locks up. it seems to be an compatibility issue between mpeg4ip and alsa, as when i try to run mpeg4ip on another box with soundblaster awe64 value the same thing happens. the sb card works fine with mpeg4ip when i use OSS/Free, though. has anyone else had a similar problem with the computer locking up? any ideas what the reason could be? philip balinov _ Choose an Internet access plan right for you -- try MSN! http://resourcecenter.msn.com/access/plans/default.asp --- This sf.net email is sponsored by: Influence the future of Java(TM) technology. Join the Java Community Process(SM) (JCP(SM)) program now. http://ads.sourceforge.net/cgi-bin/redirect.pl?sunm0004en ___ Alsa-user mailing list [EMAIL PROTECTED] https://lists.sourceforge.net/lists/listinfo/alsa-user
[Alsa-user] Fix For all those YMF744 and OPL3 people
It was just brought to my attention that the OPL3 bug fix dicussed on alsa-devel has been added to cvs, so if you are interest in the OPL3 FM in the YMF7XX you might give the CVS tree a try. Remember this is NOT XG Sythnesis. It is the FM synthesis from the good old (MS/PC/DR)DOS days. Philip Thiem GPG Pub Key Archived at wwwkeys.us.pgp.net Isn't it obvious lumberjacks love traffic lights? --- This sf.net email is sponsored by: To learn the basics of securing your web site with SSL, click here to get a FREE TRIAL of a Thawte Server Certificate: http://www.gothawte.com/rd524.html ___ Alsa-user mailing list [EMAIL PROTECTED] https://lists.sourceforge.net/lists/listinfo/alsa-user
[Alsa-user] FC6 and SPDIF issues on EPIA SP-13000
I have an EPIA SP-13000 running FC6 (updated), but I
can't get SPDIF working to save my life.
I've got the following rpms:
alsa-lib-1.0.14-0.1.rc1.fc6
alsa-utils-1.0.14-0.1.rc1.fc6
alsa-lib-devel-1.0.14-0.1.rc1.fc6
and the default /etc/asound.conf. Plus I set my
.asoundrc to be:
# As suggested on
http://mythtv.org/wiki/index.php/Configuring_Digital_Sound_with_AC3_and_SPDIF :
pcm.!default {
type plug
slave {
pcm spdif
rate 48000
format S16_LE
}
}
What am I missing?
Thanks,
-Philip
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[Alsa-user] Staticky built-in mike on Gateway Product Name: LT30
People, I have uninstalled pulseaudio and am using: Linux 2.6.33.3-79.fc13.i686 The built in mike is working but with very bad sound and I can't get the external mikes working. alsa-info output is at: http://www.alsa-project.org/db/?f=822b709ba5cbc783a762d5229613cd81d5c53b43 Any help would be appreciated! Thanks, Phil. -- Philip Rhoades Pricom Pty Limited (ACN 003 252 275 ABN 91 003 252 275) GPO Box 3411 Sydney NSW 2001 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] MB with long-standing problem but I really just want to understand what is going on . .
People, I have been using this MotherBoard for years (it is probably the most reliable one I have ever had): http://www.intel.com/p/en_US/support/highlights/dsktpboards/dg45id but from the beginning I had trouble with ALSA and the Line In port - documented here: http://mailman.alsa-project.org/pipermail/alsa-devel/2009-February/014825.html Mostly this has not been a problem since I only use the Line In port on infrequent occasions but it is annoying that when I do need to use it, sometimes, the method I have found to activate the port does not work and now using it seems to have permanently messed up the configuration for Back Mic for normal use somehow. I have installed Fedora 22 on a new drive (currently using 21) and booting on the new disk cures the introduced problem so it appears not to be permanent sound card damage at all and just some sort of caching or other issue. My procedure for getting the Line In port used to be this (although it didn't work this time and just caused a problem with the Back Mic): - uninstall PulseAudio (or rename /usr/bin/pulseaudio) - stop all apps auto starting that use ALSA - reboot - systemctl stop alsa-state.service - echo 0x0c 0x01813021 /sys/class/sound/hwC0D2/user_pin_configs - echo 1 /sys/class/sound/hwC0D2/reconfig - chmod -R 777 /dev/snd - systemctl start alsa-state.service Going through this procedure showed up new controls in xfce4-mixer and everything worked fine for what I wanted to do - a reboot restored the original setup (it could also be done with echoing the original values). It has been some years since I used this technique but now I can't get it to work and some of the new controls that show up in xfce4-mixer (Capture 1, Input Source 1) don't disappear with a reboot - even if I: rm .config/xfce4/xfconf/xfce-perchannel-xml/xfce4-mixer.xml !? Also, from Audacity I can now no longer select Default for the microphone and I can't get any of the other options to record . . Is anybody able to tell me what is going on? I can just starting using F22 on the new drive and forget about using this procedure in the future but I don't like not knowing why things don't work reliably . . Any useful information would be appreciated! Regards, Phil. -- Philip Rhoades PO Box 896 Cowra NSW 2794 Australia E-mail: p...@pricom.com.au -- One dashboard for servers and applications across Physical-Virtual-Cloud Widest out-of-the-box monitoring support with 50+ applications Performance metrics, stats and reports that give you Actionable Insights Deep dive visibility with transaction tracing using APM Insight. http://ad.doubleclick.net/ddm/clk/290420510;117567292;y ___ 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,
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;
>>>>
>>>>
>>>>
>>>> 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
>>>&
Re: [Alsa-user] Fedora 23 x86_64; Pulseaudio removed; playing from multiple sources serially - a little progress . .
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.
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Australia
E-mail: p...@pricom.com.au
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Re: [Alsa-user] Fedora 23 x86_64; Pulseaudio removed; playing from multiple sources serially
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
Re: [Alsa-user] Fedora 23 x86_64; Pulseaudio removed; playing from multiple sources serially
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
[Alsa-user] Fedora 23 x86_64; Pulseaudio removed; playing from multiple sources serially
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 but because I can't reliably reproduce the problem - in order I do: - refresh tab - exit and restart Chrome - reboot in order to get audio going again on Chrome - I am presuming this is a Chrome issue more than an ALSA issue . . 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
Re: [Alsa-user] Fedora 23 x86_64; Pulseaudio removed; playing from multiple sources serially - a little progress . .
People,
On 2015-10-08 18:16, Clemens Ladisch wrote:
> Paolo Bolzoni wrote:
>> "Dmix is enabled by default for soundcards which don't support
>> hardware mixing."
>>
>> In my experience, this is a lie.
>
> It is enabled in the ALSA device named "default". That doesn't help
> with programs that hardcode a device name like "hw:0".
>
>
> You could try something like the following to find any programs that
> still try to use "hw":
>
> pcm.my_hw {
> @args [ CARD DEV ]
> @args.CARD {
> type string
> default 1
> }
> @args.DEV {
> type integer
> default 0
> }
> type hw
> card $CARD
> device $DEV
> subdevice -1
> }
>
> pcm.dmixer {
> slave.pcm "my_hw:1"
> ...
> }
>
> pcm.!hw = blow_up
>
>
> (You could also redefine "hw" to another valid device, but it would be
> a better idea to adjust the configuration of the respective program to
> use the correct device.)
This version has been rock-solid for a couple of months now:
pcm.!default {
type plug
slave.pcm "asymed"
}
# This is the audio output:
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
}
# This is the microphone
pcm.dsnooped {
ipc_key 1027
type dsnoop
slave.pcm "hw:1,0"
}
# This makes both channels work together.
pcm.asymed {
type asym
playback.pcm "dmixer"
capture.pcm "dsnooped"
}
I am very happy now! Thanks for all your help!
Regards,
Phil.
--
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PO Box 896
Cowra NSW 2794
Australia
E-mail: p...@pricom.com.au
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Re: [Alsa-user] A long shot I know: recording from a POTS phone for voicemail
Guys, On 2016-02-24 11:28, chris hermansen wrote: > Doug, Philip, list; > > On Tue, Feb 23, 2016 at 4:06 PM, doug <dmcgarr...@optonline.net> > wrote: > >> On 02/23/2016 03:10 PM, Philip Rhoades wrote: >>> People, >>> >>> I know this is a bit of a long shot but does anyone here have any >>> experience setting up a voicemail recording system for a POTS >> phone? I >>> thought it should be simpler than using say using Asterisk to set >> up a >>> whole PABX but maybe it isn't . . >>> >>> Yhanks, >>> >>> Phil. >> It depends on what you are trying to do. If you just want to record >> the >> audio on a telephone line, the easiest way is >> a small audio transformer and a capacitor. Say a 1 Ohm 1:1 audio >> transformer and about a 0.1 microfarad capacitor >> rated at 100 volts DC or better in series with the transformer on >> the >> phone line side. Connect the secondary to the recorder >> or computer line input. >> >> If you're trying to run voice in both directions from, say, a >> computer >> to the phone line, you need a telephone modem, which >> is explicitly designed to do that job. The main reason is to simply replace my answering machine with my computer - I don't need to make calls out or record my conversations or anything like that - the setup justs needs to answer the phone when I am not around and record a message as a WAV file or something . . > Most older laptops include a modem. Or if you don't have one of those > you can buy a cheapo modem > > Here is a reference you might find useful > http://www.linuxtoys.org/answer/answering_machine.html > > Here is another http://frank.harvard.edu/~coldwell/answering_machine/ > > Another > http://askubuntu.com/questions/513004/telephone-call-answering-machine-software > > Another > http://unix.stackexchange.com/questions/44624/is-there-a-way-for-linux-to-pick-up-the-phone > > You may also be able to do the job with vgetty > http://linux.die.net/man/8/vgetty I will have a look at all that - thanks! Regards, Phil. -- Philip Rhoades PO Box 896 Cowra NSW 2794 Australia E-mail: p...@pricom.com.au -- 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=272487151=/4140 ___ Alsa-user mailing list Alsa-user@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/alsa-user
[Alsa-user] A long shot I know: recording from a POTS phone for voicemail
People, I know this is a bit of a long shot but does anyone here have any experience setting up a voicemail recording system for a POTS phone? I thought it should be simpler than using say using Asterisk to set up a whole PABX but maybe it isn't . . Yhanks, Phil. -- Philip Rhoades PO Box 896 Cowra NSW 2794 Australia E-mail: p...@pricom.com.au -- 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=272487151=/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 - a little progress . . UPDATE
Ralf, On 2017-09-04 14:57, Philip Rhoades wrote: Ralf, On 2017-09-04 02:01, Ralf Mardorf wrote: On Mon, 04 Sep 2017 00:50:24 +1000, Philip Rhoades wrote: I now want to use "recordmydesktop" which is working fine with the mic but not recording sound from videos that are playing eg from YouTube or local mpv etc - hopefully an alsa guru will have a solution for me? Sorry, I can't help, but perhaps one of the following software is helpful: https://github.com/i-rinat/apulse https://github.com/vkohaupt/vokoscreen I don't know if vokoscreen With "Pulse" selected as audio input does not work . . although at least it was good to compare recordmydesktop to vokoscreen . or apulse vokoscreen Does not work either . . does work, at least apulse firefox works. Which doesn't help my situation unfortunately . . As an exercise I tried this: apulse google-chrome but it didn't work either. Thanks, Phil. Anyone on the list who sees this and needs to see my .asoundrc file should see my previous post. Thanks anyway, Phil. -- Philip Rhoades PO Box 896 Cowra NSW 2794 Australia E-mail: p...@pricom.com.au -- 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 -- Philip Rhoades PO Box 896 Cowra NSW 2794 Australia E-mail: p...@pricom.com.au -- 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] Fedora 23 x86_64; Pulseaudio removed; playing from multiple sources serially - a little progress . . UPDATE
Ralf, On 2017-09-04 02:01, Ralf Mardorf wrote: On Mon, 04 Sep 2017 00:50:24 +1000, Philip Rhoades wrote: I now want to use "recordmydesktop" which is working fine with the mic but not recording sound from videos that are playing eg from YouTube or local mpv etc - hopefully an alsa guru will have a solution for me? Sorry, I can't help, but perhaps one of the following software is helpful: https://github.com/i-rinat/apulse https://github.com/vkohaupt/vokoscreen I don't know if vokoscreen With "Pulse" selected as audio input does not work . . although at least it was good to compare recordmydesktop to vokoscreen . or apulse vokoscreen Does not work either . . does work, at least apulse firefox works. Which doesn't help my situation unfortunately . . Anyone on the list who sees this and needs to see my .asoundrc file should see my previous post. Thanks anyway, Phil. -- Philip Rhoades PO Box 896 Cowra NSW 2794 Australia E-mail: p...@pricom.com.au -- 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] Fedora 28 x86_64; Pulseaudio removed - using "recordmydesktop" to record sound from mic AND the computer (eg from YouTube)
People,
Is this possible? My current .asoundrc is below.
Thanks!
Phil.
pcm.!default {
type plug
slave.pcm "asymed"
}
# This is the audio output:
pcm.dmixer {
type dmix
ipc_key 1024
slave {
pcm "hw:0,0"
period_time 0
period_size 1024
buffer_size 4096
rate 44100
}
bindings {
0 0
1 1
}
}
ctl.dmixer {
type hw
card 0
}
# This is the microphone
pcm.dsnooped {
ipc_key 1027
type dsnoop
slave.pcm "hw:0,0"
}
# This makes both channels work together.
pcm.asymed {
type asym
playback.pcm "dmixer"
capture.pcm "dsnooped"
}
--
Philip Rhoades
PO Box 896
Cowra NSW 2794
Australia
E-mail: p...@pricom.com.au
--
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Re: [Alsa-user] aplay weirdness
Jay,
On 2018-08-31 01:42, Jay Foster wrote:
What version of aplay?
alsa-utils-1.1.6-1.fc28.x86_64
If it is version 1.1.6, you might be running
into this issue:
From: Takashi Iwai
Subject: [PATCH] aplay: Fix invalid file size check for non-regular
files
Looks like you nailed it - so I will have to build my own I guess . .
Thanks!
P.
aplay tries to check the file size via fstat() at parsing the format
headers and avoids parsing when the size is shorter than the given
size. This works fine for regular files, but when a special file like
pipe is passed, it fails, eventually leading to the fallback mode
wrongly.
A proper fix is to do this sanity check only for a regular file.
Reported-by: Jay Foster
Signed-off-by: Takashi Iwai
---
aplay/aplay.c | 3 ++-
1 file changed, 2 insertions(+), 1 deletion(-)
diff --git a/aplay/aplay.c b/aplay/aplay.c
index bbd7fffa04fc..63ec9efbebc1 100644
--- a/aplay/aplay.c
+++ b/aplay/aplay.c
@@ -2821,7 +2821,8 @@ static int read_header(int *loaded, int
header_size)
/* don't be adventurous, get out if file size is smaller than
* requested header size */
-if (buf.st_size < header_size)
+if ((buf.st_mode & S_IFMT) == S_IFREG &&
+buf.st_size < header_size)
return -1;
if (*loaded < header_size) {
Jay
On 8/30/2018 2:50 AM, Philip Rhoades wrote:
People,
This produces a crashing static sound:
espeak --stdout 'words to speak' | aplay
but this works as expected:
espeak --stdout 'words to speak' > ./t
aplay ./t
What is wrong with the first command?
Thanks,
Phil.
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[Alsa-user] aplay weirdness
People, This produces a crashing static sound: espeak --stdout 'words to speak' | aplay but this works as expected: espeak --stdout 'words to speak' > ./t aplay ./t What is wrong with the first command? Thanks, Phil. -- Philip Rhoades PO Box 896 Cowra NSW 2794 Australia E-mail: p...@pricom.com.au -- 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] aplay weirdness
Clemens, On 2018-08-30 20:29, Clemens Ladisch via Alsa-user wrote: Philip Rhoades wrote: This produces a crashing static sound: espeak --stdout 'words to speak' | aplay but this works as expected: espeak --stdout 'words to speak' > ./t aplay ./t Is there a difference in the output of "hexdump -C -n48 ./t" and "espeak --stdout 'words to speak' | hexdump -C -n48"? hexdump -C -n48 ./t 52 49 46 46 24 f0 ff 7f 57 41 56 45 66 6d 74 20 |RIFF$...WAVEfmt | 0010 10 00 00 00 01 00 01 00 22 56 00 00 44 ac 00 00 |"V..D...| 0020 02 00 10 00 64 61 74 61 00 f0 ff 7f 00 00 00 00 |data| 0030 espeak --stdout 'words to speak' | hexdump -C -n48 52 49 46 46 24 f0 ff 7f 57 41 56 45 66 6d 74 20 |RIFF$...WAVEfmt | 0010 10 00 00 00 01 00 01 00 22 56 00 00 44 ac 00 00 |"V..D...| 0020 02 00 10 00 64 61 74 61 00 f0 ff 7f e7 ff cd ff |data| 0030 Hmm . . four bytes at the end . . P. Regards, Clemens -- 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 -- Philip Rhoades PO Box 896 Cowra NSW 2794 Australia E-mail: p...@pricom.com.au -- 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] Problem with sending a reply to the list - previous replies were OK . .
lists, On 2020-08-03 17:13, lists wrote: I run my own email server. I use this to verify my server is working properly. https://dkimvalidator.com/ Thanks - I will have a look. I haven't been following this thread. If you aren't running your own email server, I don't see how you can effect SPF or DKIM. Yes I do run my own email server (IndiMail - an updated version of QMail) - the problem is that of the reports I get daily about my setup, only roughly 50% pass DKIM - it is better for SPF. I don't understand why it is not 0% or 100% . . SPF means your email comes from an IP address that is allowed to send email from your domain. Yes. This is handled by your DNS. DKIM is substantially more difficult to explain. It is a way to cryptographically sign your message. If you pass DKIM then the message has not been altered. How could my DKIM setup not be either 0% or 100% ? One way to fail SPF is to use a remailer. Well I don't have that problem . . Thanks for the info! Phil. Original Message From: p...@pricom.com.au Sent: August 2, 2020 11:45 PM To: alan01...@gmail.com Reply-to: p...@pricom.com.au Cc: alsa-user@lists.sourceforge.net Subject: Re: [Alsa-user] Problem with sending a reply to the list - previous replies were OK . . Alan, On 2020-08-03 01:43, Alan Corey wrote: I don''t remember the exact error message, it was something like "unable to establish secure connection to ...". I spent a lot of time digging into it and the simple fix was just to reboot the router which hadn't been done in months. My router then was a cell phone with battery and charger so it was mostly always up. It was affecting SSL connections to some web sites but not all. As expected, rebooting my modem did not help - I still get the DKIM error messages when I post to this list - I will just have to spend more time on it yet again to try and work out why some sites are failing . . it would be nice if there are some DKIM / DMARC / SPF gurus here who could help me . . Thanks, Phil. On 8/2/20, Philip Rhoades wrote: Alan, On 2020-07-30 14:44, Alan Corey wrote: When was the last time you rebooted your router? I've seen that solve weird things. Just did that now before sending this reply . . but I bet I will get this stupid DMARC / DKIM problem again . . Thanks, P. On 7/30/20, Philip Rhoades wrote: Alan, On 2020-07-29 23:58, Alan Corey wrote: Could it be because you changed the subject line? It wouldn't surprise me if something does a hash of the subject. I don't know a lot about secure email with SSL and all that. I don't think so - posting this note also caused me to receive similar error messages - even though in both cases the email made it onto the list . . must be some sort of DKIM problem I think . . Thanks, Phil. On 7/29/20, Philip Rhoades wrote: People, I tried to reply to my previous thread with a slightly modified Subject: SOLVED: Re: Anybody got Google Meet going on Linux (Fedora) with just ALSA (ie not with PulseAudio)? and received a response: This is an authentication failure report for an email message received from pricom.com.au on Wed Jul 29 10:43:32 UTC 2020 Received-SPF: pass (domain of lists.sourceforge.net designates 216.105.38.7 as permitted sender) Authentication-Results: atlas201.free.mail.ir2.yahoo.com; dkim=perm_fail header.i=@sourceforge.net header.s=x; dkim=perm_fail header.i=@sf.net header.s=x; dkim=perm_fail header.i=@pricom.com.au header.s=phr1; spf=pass smtp.mailfrom=lists.sourceforge.net; dmarc=success(p=NONE,sp=NONE) header.from=pricom.com.au; . . How can I fix this? 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 -- Philip Rhoades PO Box 896 Cowra NSW 2794 Australia E-mail: p...@pricom.com.au -- 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 -- 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] Problem with sending a reply to the list - previous replies were OK . .
Alan, On 2020-07-29 23:58, Alan Corey wrote: Could it be because you changed the subject line? It wouldn't surprise me if something does a hash of the subject. I don't know a lot about secure email with SSL and all that. I don't think so - posting this note also caused me to receive similar error messages - even though in both cases the email made it onto the list . . must be some sort of DKIM problem I think . . Thanks, Phil. On 7/29/20, Philip Rhoades wrote: People, I tried to reply to my previous thread with a slightly modified Subject: SOLVED: Re: Anybody got Google Meet going on Linux (Fedora) with just ALSA (ie not with PulseAudio)? and received a response: This is an authentication failure report for an email message received from pricom.com.au on Wed Jul 29 10:43:32 UTC 2020 Received-SPF: pass (domain of lists.sourceforge.net designates 216.105.38.7 as permitted sender) Authentication-Results: atlas201.free.mail.ir2.yahoo.com; dkim=perm_fail header.i=@sourceforge.net header.s=x; dkim=perm_fail header.i=@sf.net header.s=x; dkim=perm_fail header.i=@pricom.com.au header.s=phr1; spf=pass smtp.mailfrom=lists.sourceforge.net; dmarc=success(p=NONE,sp=NONE) header.from=pricom.com.au; . . How can I fix this? 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
[Alsa-user] This appears to be the problem I am getting with this list - and now other lists - DMARC / DKIM / SPF
People, FYI, this seems to explain (most of) my hassles mailing to this list: Yahoo breaks every mailing list in the world including the IETF's https://mailarchive.ietf.org/arch/msg/ietf/J-IsfA0Lb-6T_NeMD1ENKZyb9tA/ 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
Re: [Alsa-user] This appears to be the problem I am getting with this list - and now other lists - DMARC / DKIM / SPF
lists, On 2020-08-04 16:41, lists wrote: I just logged into my yahoo account and used the dkimvalidator. It passes SPF and DKIM perfectly. So does mine . . I really doubted the claim that they don't pass DMARC since they helped set the standard. Well that is an old article but it looks so much like my problem I figured it must be related . . As an aside, Google is happy with either passing SPF or DKIM. Right. From a spam viewpoint, if you pass SPF then you are authorized to send email from that server. There is a slight chance that the domain got spoofed, and that would make it trivial to spoof the SPF. Well, as I said before, despite passing the above checking site - I get reports like the attached which reports fails more often than not for DKIM less so for SPF . . DKIM is another story. It involves public and private keys. For postfix to sign the message with DKIM, you need the private key on the server. The public key is in the DNS so that links it to the domain. So if you own the domain and the private key, you must be legit. Mind you I have simplified DKIM slightly since it isn't worth me spewing a half dozen more paragraphs. See previous response. Try your account with the dkimvalidator: https://dkimvalidator.com/ I did when I had set up this stuff originally (as I said above - everything passed OK then and still does now) - something has recently changed - I am getting dozens of stupid Yahoo messages every time I mail to this list now . . The website creates an email address. You send a message to that email address. Don't leave the subject line or message block blank if you want a valid spam assassination score. Wait a minute or two, then click on the check results box. The output is a bit wordly, but you will see verification for spf and dkim. This is the website I use when setting up email servers. Yes, I had forgotten that I had also used it originally. Technically I have only proved SPF and DKIM work. That should be all that a mailing list requires. Apparently not anymore . . DMARC is a bit more complicated to explain, but it isn't worth talking about unless you fail SPF or DKIM. The email system administrator in theory processes reports from other email servers in order to see if any accounts on their server have been spoofed. I say in theory because there is no way to tell if the administrator reads the DMARC reports. Anyway, now that this problem appears to be greater than this list - I should pursue problem-solving in a more appropriate place! Thanks. On the bright side, I have had no ALSA problems in my latest opensuse upgrade! Great! - although am a Fedora person myself . . Regards, Phil. Original Message From: p...@pricom.com.au Sent: August 3, 2020 10:52 PM To: alsa-user@lists.sourceforge.net; us...@lists.roundcube.net Reply-to: p...@pricom.com.au Subject: [Alsa-user] This appears to be the problem I am getting with this list - and now other lists - DMARC / DKIM / SPF People, FYI, this seems to explain (most of) my hassles mailing to this list: Yahoo breaks every mailing list in the world including the IETF's https://mailarchive.ietf.org/arch/msg/ietf/J-IsfA0Lb-6T_NeMD1ENKZyb9tA/ 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 -- Philip Rhoades PO Box 896 Cowra NSW 2794 Australia E-mail: p...@pricom.com.au comcast.net!pricom.com.au!1596412800!1596499200.xml.gz Description: GNU Zip compressed data ___ Alsa-user mailing list Alsa-user@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/alsa-user
Re: [Alsa-user] Problem with sending a reply to the list - previous replies were OK . .
Alan, On 2020-07-30 14:44, Alan Corey wrote: When was the last time you rebooted your router? I've seen that solve weird things. Just did that now before sending this reply . . but I bet I will get this stupid DMARC / DKIM problem again . . Thanks, P. On 7/30/20, Philip Rhoades wrote: Alan, On 2020-07-29 23:58, Alan Corey wrote: Could it be because you changed the subject line? It wouldn't surprise me if something does a hash of the subject. I don't know a lot about secure email with SSL and all that. I don't think so - posting this note also caused me to receive similar error messages - even though in both cases the email made it onto the list . . must be some sort of DKIM problem I think . . Thanks, Phil. On 7/29/20, Philip Rhoades wrote: People, I tried to reply to my previous thread with a slightly modified Subject: SOLVED: Re: Anybody got Google Meet going on Linux (Fedora) with just ALSA (ie not with PulseAudio)? and received a response: This is an authentication failure report for an email message received from pricom.com.au on Wed Jul 29 10:43:32 UTC 2020 Received-SPF: pass (domain of lists.sourceforge.net designates 216.105.38.7 as permitted sender) Authentication-Results: atlas201.free.mail.ir2.yahoo.com; dkim=perm_fail header.i=@sourceforge.net header.s=x; dkim=perm_fail header.i=@sf.net header.s=x; dkim=perm_fail header.i=@pricom.com.au header.s=phr1; spf=pass smtp.mailfrom=lists.sourceforge.net; dmarc=success(p=NONE,sp=NONE) header.from=pricom.com.au; . . How can I fix this? 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 -- 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] Problem with sending a reply to the list - previous replies were OK . .
Alan, On 2020-08-03 01:43, Alan Corey wrote: I don''t remember the exact error message, it was something like "unable to establish secure connection to ...". I spent a lot of time digging into it and the simple fix was just to reboot the router which hadn't been done in months. My router then was a cell phone with battery and charger so it was mostly always up. It was affecting SSL connections to some web sites but not all. As expected, rebooting my modem did not help - I still get the DKIM error messages when I post to this list - I will just have to spend more time on it yet again to try and work out why some sites are failing . . it would be nice if there are some DKIM / DMARC / SPF gurus here who could help me . . Thanks, Phil. On 8/2/20, Philip Rhoades wrote: Alan, On 2020-07-30 14:44, Alan Corey wrote: When was the last time you rebooted your router? I've seen that solve weird things. Just did that now before sending this reply . . but I bet I will get this stupid DMARC / DKIM problem again . . Thanks, P. On 7/30/20, Philip Rhoades wrote: Alan, On 2020-07-29 23:58, Alan Corey wrote: Could it be because you changed the subject line? It wouldn't surprise me if something does a hash of the subject. I don't know a lot about secure email with SSL and all that. I don't think so - posting this note also caused me to receive similar error messages - even though in both cases the email made it onto the list . . must be some sort of DKIM problem I think . . Thanks, Phil. On 7/29/20, Philip Rhoades wrote: People, I tried to reply to my previous thread with a slightly modified Subject: SOLVED: Re: Anybody got Google Meet going on Linux (Fedora) with just ALSA (ie not with PulseAudio)? and received a response: This is an authentication failure report for an email message received from pricom.com.au on Wed Jul 29 10:43:32 UTC 2020 Received-SPF: pass (domain of lists.sourceforge.net designates 216.105.38.7 as permitted sender) Authentication-Results: atlas201.free.mail.ir2.yahoo.com; dkim=perm_fail header.i=@sourceforge.net header.s=x; dkim=perm_fail header.i=@sf.net header.s=x; dkim=perm_fail header.i=@pricom.com.au header.s=phr1; spf=pass smtp.mailfrom=lists.sourceforge.net; dmarc=success(p=NONE,sp=NONE) header.from=pricom.com.au; . . How can I fix this? 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 -- Philip Rhoades PO Box 896 Cowra NSW 2794 Australia E-mail: p...@pricom.com.au -- 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] Anybody got Google Meet going on Linux (Fedora) with just ALSA (ie not with PulseAudio)?
Peter, On 2020-07-26 19:39, Peter P. wrote: * Philip Rhoades [2020-07-26 10:11]: People, I am not sure what is going on - I seem to have had increased sound problems on recent versions of Fedora (30-31). I have been routinely uninstalling PA for some years and haven't had more than the usual number of problems with audio in that time. Currently: alsa-lib-1.2.1.2-4.fc31.i686 alsa-lib-1.2.1.2-4.fc31.x86_64 alsa-ucm-1.2.1.2-4.fc31.noarch alsa-utils-1.2.1-3.fc31.x86_64 alsamixergui-0.9.0-0.29.rc2.fc31.x86_64 qemu-audio-alsa-4.1.1-1.fc31.x86_64 wine-alsa-5.0-1.fc31.i686 wine-alsa-5.0-1.fc31.x86_64 http://alsa-project.org/db/?f=0da7d0299aadfe9dc87c66a526ee14c7806776c7 If I use the analog rear mic instead of "Default" on GoogleMeet people can hear me but I can't hear them - but the "Speakers" option can't be changed from "Default". Playing YT videos is fine and I can record from my mic here: https://online-voice-recorder.com Any suggestions about debugging would be greatly appreciated. Which browser ist this google-chrome-stable-83.0.4103.61-1.x86_64 and have you tried the apulse package (at least that's what it's called on Debian Doesn't that mean I would have to reinstall 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] Anybody got Google Meet going on Linux (Fedora) with just ALSA (ie not with PulseAudio)?
People, I am not sure what is going on - I seem to have had increased sound problems on recent versions of Fedora (30-31). I have been routinely uninstalling PA for some years and haven't had more than the usual number of problems with audio in that time. Currently: alsa-lib-1.2.1.2-4.fc31.i686 alsa-lib-1.2.1.2-4.fc31.x86_64 alsa-ucm-1.2.1.2-4.fc31.noarch alsa-utils-1.2.1-3.fc31.x86_64 alsamixergui-0.9.0-0.29.rc2.fc31.x86_64 qemu-audio-alsa-4.1.1-1.fc31.x86_64 wine-alsa-5.0-1.fc31.i686 wine-alsa-5.0-1.fc31.x86_64 http://alsa-project.org/db/?f=0da7d0299aadfe9dc87c66a526ee14c7806776c7 If I use the analog rear mic instead of "Default" on GoogleMeet people can hear me but I can't hear them - but the "Speakers" option can't be changed from "Default". Playing YT videos is fine and I can record from my mic here: https://online-voice-recorder.com Any suggestions about debugging would be greatly appreciated. Regards, 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
Re: [Alsa-user] Anybody got Google Meet going on Linux (Fedora) with just ALSA (ie not with PulseAudio)?
Peter, On 2020-07-26 21:18, Philip Rhoades wrote: Peter, On 2020-07-26 19:39, Peter P. wrote: * Philip Rhoades [2020-07-26 10:11]: People, I am not sure what is going on - I seem to have had increased sound problems on recent versions of Fedora (30-31). I have been routinely uninstalling PA for some years and haven't had more than the usual number of problems with audio in that time. Currently: alsa-lib-1.2.1.2-4.fc31.i686 alsa-lib-1.2.1.2-4.fc31.x86_64 alsa-ucm-1.2.1.2-4.fc31.noarch alsa-utils-1.2.1-3.fc31.x86_64 alsamixergui-0.9.0-0.29.rc2.fc31.x86_64 qemu-audio-alsa-4.1.1-1.fc31.x86_64 wine-alsa-5.0-1.fc31.i686 wine-alsa-5.0-1.fc31.x86_64 http://alsa-project.org/db/?f=0da7d0299aadfe9dc87c66a526ee14c7806776c7 If I use the analog rear mic instead of "Default" on GoogleMeet people can hear me but I can't hear them - but the "Speakers" option can't be changed from "Default". Playing YT videos is fine and I can record from my mic here: https://online-voice-recorder.com Any suggestions about debugging would be greatly appreciated. Which browser ist this google-chrome-stable-83.0.4103.61-1.x86_64 and have you tried the apulse package (at least that's what it's called on Debian Doesn't that mean I would have to reinstall PA? Doesn't appear to exist for Fedora anyway . . 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] SOLVED: Re: Anybody got Google Meet going on Linux (Fedora) with just ALSA (ie not with PulseAudio)?
People, I eventually narrowed down what the problem was: - a new test user on the same workstation worked fine - a different profile (attached to the non-working user) worked fine - using the Chrome reset account facility fixed the problem - reinstalling some Chrome extensions caused the problem again I have narrowed it down to either a YouTube video speed controller or an extension for preventing automatic playing of YT videos . . somehow one or other of these extensions, for Google Meet and Jitsi prevents the speakers from working at all and the mic needs to be changed from Default to Analog . . weird . . So now I have work-arounds anyway . . a somewhat interesting exercise I guess . . Thanks, Phil. On 2020-07-26 17:43, Philip Rhoades wrote: People, I am not sure what is going on - I seem to have had increased sound problems on recent versions of Fedora (30-31). I have been routinely uninstalling PA for some years and haven't had more than the usual number of problems with audio in that time. Currently: alsa-lib-1.2.1.2-4.fc31.i686 alsa-lib-1.2.1.2-4.fc31.x86_64 alsa-ucm-1.2.1.2-4.fc31.noarch alsa-utils-1.2.1-3.fc31.x86_64 alsamixergui-0.9.0-0.29.rc2.fc31.x86_64 qemu-audio-alsa-4.1.1-1.fc31.x86_64 wine-alsa-5.0-1.fc31.i686 wine-alsa-5.0-1.fc31.x86_64 http://alsa-project.org/db/?f=0da7d0299aadfe9dc87c66a526ee14c7806776c7 If I use the analog rear mic instead of "Default" on GoogleMeet people can hear me but I can't hear them - but the "Speakers" option can't be changed from "Default". Playing YT videos is fine and I can record from my mic here: https://online-voice-recorder.com Any suggestions about debugging would be greatly appreciated. Regards, 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] Problem with sending a reply to the list - previous replies were OK . .
People, I tried to reply to my previous thread with a slightly modified Subject: SOLVED: Re: Anybody got Google Meet going on Linux (Fedora) with just ALSA (ie not with PulseAudio)? and received a response: This is an authentication failure report for an email message received from pricom.com.au on Wed Jul 29 10:43:32 UTC 2020 Received-SPF: pass (domain of lists.sourceforge.net designates 216.105.38.7 as permitted sender) Authentication-Results: atlas201.free.mail.ir2.yahoo.com; dkim=perm_fail header.i=@sourceforge.net header.s=x; dkim=perm_fail header.i=@sf.net header.s=x; dkim=perm_fail header.i=@pricom.com.au header.s=phr1; spf=pass smtp.mailfrom=lists.sourceforge.net; dmarc=success(p=NONE,sp=NONE) header.from=pricom.com.au; . . How can I fix this? 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
Re: [Alsa-user] Anybody got Google Meet going on Linux (Fedora) with just ALSA (ie not with PulseAudio)?
Geordon, On 2020-07-27 04:20, 3DTechnics wrote: Peter, I switched to apulse from PA in gentoo.works well. Although I am using Fedora I installed OpenSuse's apulse (which should be OK I think): apulse-0.1.12-lp152.1.3.x86_64.rpm - now I get: "No microphone found" and for "Speakers", as well as "Default", I now also have: "default and only sync" but neither of those work with the speaker test still. I see at CLI: apulse google-chrome-stable /bin/apulse:10: DeprecationWarning: The SafeConfigParser class has been renamed to ConfigParser in Python 3.2. This alias will be removed in future versions. Use ConfigParser directly instead. conf_parser = configparser.SafeConfigParser() MESA-LOADER: failed to retrieve device information MESA-LOADER: failed to open i915 (search paths /usr/lib64/dri) failed to load driver: i915 MESA-LOADER: failed to open kms_swrast (search paths /usr/lib64/dri) failed to load driver: kms_swrast MESA-LOADER: failed to open swrast (search paths /usr/lib64/dri) failed to load swrast driver and when I try to test the speakers I get: [437680:437680:0727/121043.555607:ERROR:gles2_cmd_decoder.cc(3601)] ContextResult::kFatalFailure: fail_if_major_perf_caveat + swiftshader So I guess I could try building apulse from source for Fedora but I suspect that is not going to improve things . . Thanks, Phil. Geordon Peter, On 2020-07-26 21:18, Philip Rhoades wrote: Peter, On 2020-07-26 19:39, Peter P. wrote: * Philip Rhoades [2020-07-26 10:11]: People, I am not sure what is going on - I seem to have had increased sound problems on recent versions of Fedora (30-31). I have been routinely uninstalling PA for some years and haven't had more than the usual number of problems with audio in that time. Currently: alsa-lib-1.2.1.2-4.fc31.i686 alsa-lib-1.2.1.2-4.fc31.x86_64 alsa-ucm-1.2.1.2-4.fc31.noarch alsa-utils-1.2.1-3.fc31.x86_64 alsamixergui-0.9.0-0.29.rc2.fc31.x86_64 qemu-audio-alsa-4.1.1-1.fc31.x86_64 wine-alsa-5.0-1.fc31.i686 wine-alsa-5.0-1.fc31.x86_64 http://alsa-project.org/db/?f=0da7d0299aadfe9dc87c66a526ee14c7806776c7 If I use the analog rear mic instead of "Default" on GoogleMeet people can hear me but I can't hear them - but the "Speakers" option can't be changed from "Default". Playing YT videos is fine and I can record from my mic here: https://online-voice-recorder.com Any suggestions about debugging would be greatly appreciated. Which browser ist this google-chrome-stable-83.0.4103.61-1.x86_64 and have you tried the apulse package (at least that's what it's called on Debian Doesn't that mean I would have to reinstall PA? Doesn't appear to exist for Fedora anyway . . 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 -- 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] Anybody got Google Meet going on Linux (Fedora) with just ALSA (ie not with PulseAudio)?
Robert, On 2020-07-27 04:55, rm.ric...@jacob21819.net wrote: Have you experimented with different content in your .asoundrc file? Not in recent years . . I don't know about Google Meet and Chromium specifically, but other setup (the old Firefox Hello, for example) required a specific .asoundrc to make it work. IIRC, this is it: defaults.pcm.card 0 defaults.ctl.card 0 Tried creating a new .asoundrc with those lines in it - no improvement . . Thanks, Phil. HTH Robert Date: Sun, 26 Jul 2020 17:43:12 +1000 From: Philip Rhoades To: ALSA user Reply-To: p...@pricom.com.au People, I am not sure what is going on - I seem to have had increased sound problems on recent versions of Fedora (30-31). I have been routinely uninstalling PA for some years and haven't had more than the usual number of problems with audio in that time. Currently: alsa-lib-1.2.1.2-4.fc31.i686 alsa-lib-1.2.1.2-4.fc31.x86_64 alsa-ucm-1.2.1.2-4.fc31.noarch alsa-utils-1.2.1-3.fc31.x86_64 alsamixergui-0.9.0-0.29.rc2.fc31.x86_64 qemu-audio-alsa-4.1.1-1.fc31.x86_64 wine-alsa-5.0-1.fc31.i686 wine-alsa-5.0-1.fc31.x86_64 http://alsa-project.org/db/?f=0da7d0299aadfe9dc87c66a526ee14c7806776c7 If I use the analog rear mic instead of "Default" on GoogleMeet people can hear me but I can't hear them - but the "Speakers" option can't be changed from "Default". Playing YT videos is fine and I can record from my mic here: https://online-voice-recorder.com Any suggestions about debugging would be greatly appreciated. Regards, 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
[Alsa-user] Problem with kernel 2.6
Hello, all. I tried to search the archives for my problem, but couldn't access them -- connection kept timing out. I use a SoundBlaster 16(?) PCI sound card. Until yesterday I ran Gentoo with kernel 2.4.21 and with ALSA configured for the snd-ens1371 driver. Everything worked. Yesterday I decided to upgrade to kernel 2.6.0-test9. So, I stopped the alsasound service from coming up at boot and interfering with the kernel, and compiled Sound cart support as a module, ALSA as a module, RTC timer support as a module, and (Creative) Ensoniq AudioPCI 1371/1373 as a module. Having restarted, I did modprobe snd, which worked. Then I did modprobe snd-ens1371 which paused for about 3 seconds and supposedly loaded. However, when I viewed dmesg, I got the following: AC'97 0:0 does not respond - RESET ENS1371: probe of :00:0c.0 failed with error -6 Viewing /proc/asound/cards showed that the kernel is not aware of any sound cards. After I rebooted back into my old kernel and loaded the snd-ens1371 module manually, everything worked again. So, my question is, what can I do get my sound card, which worked with ALSA on the 2.4 kernel series, to work on the 2.6 series? Is it a problem with ALSA or with the kernel? (i.e. Where do I file the bug report, if it's even a bug?) Any ideas would be greatly appreciated! Philip --- This SF.Net email sponsored by: ApacheCon 2003, 16-19 November in Las Vegas. Learn firsthand the latest developments in Apache, PHP, Perl, XML, Java, MySQL, WebDAV, and more! http://www.apachecon.com/ ___ Alsa-user mailing list [EMAIL PROTECTED] https://lists.sourceforge.net/lists/listinfo/alsa-user
[Alsa-user] Changed from Fedora 33 + ALSA to Fedora 34 + Wayland + ALSA + Pipewire
People,
With my .asoundrc attached hereunder I used to be able to have inputs
and outputs from various sources without having to worry about using
things like Jack etc, but after the change in the Subject:
- I can record online here:
https://online-voice-recorder.com
happily while I use Jitsi (which can record to my DropBox account) but
when I do this CLI command as well:
arecord -D hw:0,0 -f S32_LE -c 2 ttt.wav
I get this result:
arecord: main:831: audio open error: Device or resource busy
Is some kind ALSA guru able to tell me if there is a change I can make
to my .asoundrc file that will allow me to have the convenience of my
previous setup of being able to have multiple source and destination
devices without needing to use extra software?
Thanks,
Phil.
.asoundrc:
pcm.!default {
type plug
slave.pcm "asymed"
}
# This is the audio output:
pcm.dmixer {
type dmix
ipc_key 1024
slave {
pcm "hw:0,0"
period_time 0
period_size 1024
buffer_size 4096
rate 44100
}
bindings {
0 0
1 1
}
}
ctl.dmixer {
type hw
card 0
}
# This is the microphone
pcm.dsnooped {
ipc_key 1027
type dsnoop
slave.pcm "hw:0,0"
}
# This makes both channels work together.
pcm.asymed {
type asym
playback.pcm "dmixer"
capture.pcm "dsnooped"
}
--
Philip Rhoades
PO Box 896
Cowra NSW 2794
Australia
E-mail: p...@pricom.com.au
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Re: [Alsa-user] Changed from Fedora 33 + ALSA to Fedora 34 + Wayland + ALSA + Pipewire
Extra info below:
On 2021-09-18 03:12, Philip Rhoades wrote:
People,
With my .asoundrc attached hereunder I used to be able to have inputs
and outputs from various sources without having to worry about using
things like Jack etc, but after the change in the Subject:
- I can record online here:
https://online-voice-recorder.com
Using FireFox.
happily while I use Jitsi
in Brave:
(which can record to my DropBox account) but
when I do this CLI command as well:
arecord -D hw:0,0 -f S32_LE -c 2 ttt.wav
I get this result:
arecord: main:831: audio open error: Device or resource busy
Is some kind ALSA guru able to tell me if there is a change I can make
to my .asoundrc file that will allow me to have the convenience of my
previous setup of being able to have multiple source and destination
devices without needing to use extra software?
Thanks,
Phil.
.asoundrc:
pcm.!default {
type plug
slave.pcm "asymed"
}
# This is the audio output:
pcm.dmixer {
type dmix
ipc_key 1024
slave {
pcm "hw:0,0"
period_time 0
period_size 1024
buffer_size 4096
rate 44100
}
bindings {
0 0
1 1
}
}
ctl.dmixer {
type hw
card 0
}
# This is the microphone
pcm.dsnooped {
ipc_key 1027
type dsnoop
slave.pcm "hw:0,0"
}
# This makes both channels work together.
pcm.asymed {
type asym
playback.pcm "dmixer"
capture.pcm "dsnooped"
}
--
Philip Rhoades
PO Box 896
Cowra NSW 2794
Australia
E-mail: p...@pricom.com.au
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