Posts Tagged ‘alsa’

festival “Linux: can’t open /dev/dsp” error fix

Wednesday, September 16th, 2009

I tried using festival today just to realize it doesn’t work anymore.
For instance # echo test |festival -tts would fail with the annoying
“Linux: can’t open /dev/dsp”  error message. I found the solution in ubuntuforums,
the solution is originally taken from “the Gentoo Speechd Howto“.
The solution is to create .festivalrc in your home as well as to the homes of all usersintending to use festival.
Here is how:

printf ";use ALSAn(Parameter.set 'Audio_Method 'Audio_Command)n(Parameter.set 'Audio_Command "aplay -q -c 1 -t raw -f s16 -r $SR $FILE")n" > .festivalrc

How to record microphone input sound (only) using good old ffmpeg

Tuesday, December 25th, 2012

The good old ffmpeg, along with being able to capture sound and video from your Linux Desktop or a certain Window and Skype whatever WebCamera input is also able to record sound from both camera or embedded laptop microphone. Here is how:

# ffmpeg -f alsa -ac 2 -i pulse   -acodec pcm_s16le -vcodec libx264 -vpre lossless_ultrafast -threads 0  -y  myVOICE.wav

This as you can see from arguments, uses GNOME's pulseaudio (audio service) and ALSA. Sound is first streamed through alsa and then the sound inflow is passed to be processed and multipled in a separate sound channel by pulseaudio. This method though said to be working fine on Ubuntu Linux is not working well on some other Linux distributions like Debian if one is using ALSA configured to use a software sound multiplexor via the so called – alsa dsnoop interface (previously I write how to use it in order to make Skype and other programs use SoundBlaster proper – article is here)

Below is the output warning I got whether trying ffmpeg with -f alsa and -i pulse arguments:

hipo@noah:~/Desktop$ ffmpeg -f alsa -ac 2 -i pulse   -acodec pcm_s16le -vcodec libx264 -vpre lossless_ultrafast -threads 0  -y  myVOICE.wav
FFmpeg version SVN-r25838, Copyright (c) 2000-2010 the FFmpeg developers
  built on Sep 20 2011 17:00:01 with gcc 4.4.5
  configuration: --enable-libdc1394 --prefix=/usr --extra-cflags='-Wall -g ' --cc='ccache cc' --enable-shared --enable-libmp3lame --enable-gpl --enable-libvorbis --enable-pthreads --enable-libfaac --enable-libxvid --enable-postproc --enable-x11grab --enable-libgsm --enable-libtheora --enable-libopencore-amrnb --enable-libopencore-amrwb --enable-libx264 --enable-libspeex --enable-nonfree --disable-stripping --enable-avfilter --enable-libdirac --disable-decoder=libdirac --enable-libschroedinger --disable-encoder=libschroedinger --enable-version3 --enable-libopenjpeg --enable-libvpx --enable-librtmp --extra-libs=-lgcrypt --disable-altivec --disable-armv5te --disable-armv6 --disable-vis
  libavutil     50.33. 0 / 50.43. 0
  libavcore      0.14. 0 /  0.14. 0
  libavcodec    52.97. 2 / 52.97. 2
  libavformat   52.87. 1 / 52.87. 1
  libavdevice   52. 2. 2 / 52. 2. 2
  libavfilter    1.65. 0 /  1.65. 0
  libswscale     0.12. 0 /  0.14. 1
  libpostproc   51. 2. 0 / 51. 2. 0
[alsa @ 0x633160] capture with some ALSA plugins, especially dsnoop, may hang.

where concrete programs, are run which take use of OSS (Open Sound System) – an already obsolete sound architecture. By the way on current Debian / Fedora etc. Linux-es OSS is managed and played only, whether few kernel modules are already  pre-loaded, below are the ones as pasted from my Debian Squeeze:

# lsmod | grep -i oss
snd_pcm_oss            32591  0
snd_mixer_oss          12606  1 snd_pcm_oss
snd_pcm                60487  3 snd_hda_intel,snd_hda_codec,snd_pcm_oss
snd                    46526  15 snd_hda_codec_analog,snd_hda_intel,snd_hda_codec,snd_hwdep,snd_pcm_oss,snd_mixer_oss,snd_pcm,snd_rawmidi,snd_seq,snd_timer,snd_seq_device

The oss processed sound recording from ffmpeg is not working, well on my Linux, cause I have my custom (non-Debian) native binary Firefox downloaded and installed from Firefox's website.The browser is compiled to open automatically /dev/dsp which in practice uses the above-mentioned OSS listed modules, which on their behalf when used break out the sound processed by alsa and respectively pulseaudio (those who use Linux for longer time should remember in the times of OSS only one certain sound stream was possible to be processed / played on Linux historically before ALSA come to scene to be "defacto" standard kernel sound processor. Well ofcourse firefox developers who compiled the Firefox for Linux probably was using Slackware or some other Linux distro which probably used to play sound still via OSS or maybe they compiled it so thinking OSS because of its historical importance is still supported by more Linux distributions than alsa is. I like the custom compiled Firefox to run on my Debian instead of default Debian Squeeze (IceWeasel) cause firefox.org ,Firefox version is much newer and supports better latest HTML5  as well as it includes ability to download and apply automatic updates to the latest version provided by Firefox team. However I fou

Thus for Linux users like me using latest firefox binary from firefox.org (in parallel) with opened Firefox browser to record sound from Webcam or Embedded notebook mic the obsolete OSS has to be used, here is how:

# ffmpeg -f oss -ac 2 -i /dev/dsp   -acodec pcm_s16le -vcodec libx264 -vpre lossless_ultrafast -threads 0  -y  my-recorder-VOICE.wav

Enjoy ;)

How to solve ALSA sound problems with old Linux programs and games depending on (OSS)’s /dev/dsp / fix wine games and pulseaudio problems – My few thoughts on OSS and ALSA

Wednesday, January 11th, 2012

 

ALSA OSS Pulseaudio ESD Some fixes workaround to gnu linux audio messI remember GNU / Linux, 11 years from now, times when ALSA was not standardly shipped with Linux.
Back then ALSA still lacked good support for many SoundCards and was still a "baby project".
In that time what we used to have sound on Linux was OSSOpen Sound System. OSS emerged right after the first ever Linux sound system VoxWare (formerly known as the Linux Sound Driver).

Back in those days OSS was used for multimedia support on both GNU / Linux and BSD based free OSes. It was few years later when I heard and used ALSA for a fist time and it wasn't really a love from first sigth.

One can easily find out by the name ALSA it is a system especially built for the Linux kernel and that's one of the reasosn why *BSD systems has their custom separate sound system.
There is plenty of reasons why OSS was substituted with ALSA. Main reason was its commercial like license, OSS wasn't completely "open source" GPLed (free software), there was resctions on use of OSS for commercial goals.

With its emerge ALSA started to push away OSS slowly. Somewhere in 2003, alsa has officially entered the Linux kernel source and until 2005 it was the default standard for all GNU / Linux operating systems.

As of time of writting ALSA has become the only sound system to have support for multiple sound card devices for Linux.
My experiences with ALSA, however ain't so nice if I take a look in my past experiences.
Since the very beginning of using ALSA, I had plenty of troubles with configuring properly my sound card not to mention, even after configuring it the MIDI support was not there.
Besides all the troubles main problems were stemming from the many applications still written to use OSS as sound system and hence with those sound was impossible with ALSA.The most problematic thing about apps written with OSS in mind was all of them tried to stream sound via /dev/dsp (OSS Digital Sound Processor), since alsa did not used /dev/dsp those programs was soundless.

On the other hand OSS was creating issues as well, one severe problem with OSS was the inability to stream multiple sounds simultaneously, because each sound stream required to pass voice through /dev/dsp and usually there was only one /dev/dsp.

The message;

/dev/dsp: Device or resource busy
and the proceeding irritation that used to annoy us in the early GNU / Linux days had of course some raw workarounds hacks but generally the workaround did not fix problems always.

Introduction of alsa free us from /dev/dsp issues but on the other handy has created a whole ocean of new BIG problems …
ALSA has modular structure and this imposes a great problem nowdays. The modular architecture is generally a good idea, however the way this was implemented within ALSA is far away from clear and easy to understand by the end user and therefore makes it very unintuitive and obscure.
Alsa misses simplicity which somehow was partially in the days of OSS. Thinking over the general situation with Linux multimedia nowdays, I believe it was exactly ALSA Project responsible for the so delayed mass Desktop Linux adoption.

Many long year standing Linux users had certainly had the alsa troubles during new system installs (correct me if I'm wrong).
The only fix to multiple soundcard initialization problems was to download alsa source and compile from source and hence made it hard and discouraging for people giving Linux a try.
This kind of ALSA "brokenness" pattern continues even to this very day (in Debian) Linux and probably building the alsa system from source is among the good practices to have a functional Linux sound system…

With all said the historic reason why ALSA was not quickly adopted and still is not a preferred default system for many applications ported to Free Software OSes by commercial company vendors is clear. Its simply not working out of the box …

Hope some ALSA developers will read this post work on changing the crazy structure of ALSA over complexity. ALSA needs automate way to solve issues with itself, the configuration should be more trivial and unified if Linux has to become more attractive for Desktop adoption.

Anyways, after the few words of history and indicating my pesonal observations on ALSA. I will proceed and explain few things on how ALSA can be configured to support and play nice with OSS dependant programs as well some basic explanations on common incompatibility between esd and pulseaudio and how this can be fixed;.

To assure nowdays OSS API built programs and games would work with Alsa its necessery to have installed;

ALSA wrapper for OSS applications

On Debian, Ubuntu, Fedora and most Linux distributions the Alsa OSS compatability layer comes under a (deb / rpm) package named alsa-oss

To install OSS compatability on Debian, Ubuntu and the like Debian based distributions issue:

debian:~# apt-get install alsa-oss alsaplayer-oss
...

On Fedora and other rpm based distributions install is with:

[root@fedora ~]# yum install alsa-oss alsaplayer-oss
...

alsa-oss provides with a command called aoss that should be used to work around some issues with old applications still depending on OSS:

hipo@debian:~$ aoss programName

Using aoss is helpful especially in situations if you have to run programs which deal with MIDI and others which somehow want to use /dev/dsp

There is also alternative way to enable alsa native support for MIDI and OSS by loading 3 kernel modules:

debian:~# modprobe snd-seq-oss
debian:~# modprobe snd-pcm-oss
debian:~# modprobe snd-mixer-oss

Note! The three modules has to be separately build using kernel source at most cases and does not come with most Linux distributions, so on many installations (including my current), they will be missing. If for you they load properly or you have customly build them add them also to load on system boot, like so:

echo 'snd-seq-oss' >> /etc/modules
echo 'snd-pcm-oss' >> /etc/modules
echo 'snd-mixer-oss' >> /etc/modules

The Linux sound situation becomes even more messy when ESD enters the scene. Many of the novice new Linux users certainly don't remember (Enlightened Sound Daemon) . ESD historically preceded PulseAudio . Hence it will be good to mention ESD was used for few years in GNOME and in around 2006-2007 it was substituted by PulseAudio.
Many applications, however who was ported or written for Linux especially (the proprietary ported ones) was already built to work with ESD and even though newer GNOME releases was fully using pulseaudio, this (non free software apps and games) were still depending on ESD.

The situation was partially fixed by creation of module for pulseaudio which added emulation support for esd . This was done by a module library for pulseaudio called libprotocol-esound.so
The package for almost all Linux distributions which does the esd emulation via pulse is pulseaudio-esound-compat . In latest Fedora Linux pulseaudio-esound-compat is installed by default.
In Debian and other Linux distributions it might need to be installed via apt with;

debian:~# apt-get install pulseaudio-esound-compat
...

pulseaudio-esound-compat solves some of the ESD app incompability but not always …
Handy tool also worthy to mention in solving PulseAudio, OSS incompatibility issues is padsp

padsp is helpful in solving obsolete issues with OSS applications (trying to access /dev/dsp) and therefore unable to communicate with Pulseaudio
padsp – is a PulseAudio OSS Wrapper.

An example where padsp is helpful is in case of /dev/dsp errors like:

/dev/dsp: Device or resource busy
Could not open /dev/dsp

Another common problem with sound on Linux is when running windows applications (running windows games with wine).
Quite often sound fails to work since wine tries to directly communicate with alsa and fails because alsa sound channel is taken by pulseaudio.

To workaround wine issues with pulseaudio, one of the solutions is to temporary stop pulseaudio, before running the wine emulated application:

hipo@debian:~$ pulseaudio --kill

Later on when the windows wine emulation is completed, pulseaudio has to be started once again in order to make Pulseaudio applications produce sound again, e.g. one has to issue:

hipo@debian:~$ pulseaudio --start
Alternative way to workaround wine sound issues is by using a script to kill pulseaudio every second. Here is fix_pulseaudio_wine_sound_probs.sh script

This script was reported by many people as fix to problems with wine games failing to play sounds and music, anyhow I personally prefer using the stop / start pulseaudio method.

The picture below is taken from Wikipedia and illustrates, clearly the intergalactical complexity of sound systems on Gnu / Linux and BSD

I just hope one day this (OSS, ALSA, esd, Pulseaudio) mess will be over! In the mean time I hope my suggested work arounds helps someone. If someone has a better more unified script or solution please share in comments

Solve ALSA audio and mic issues on Lenovo Thinkpads on Debian and Ubuntu Linux

Wednesday, January 11th, 2012

Since I've blogged about my recent skype issues. I've played a lot with pulseaudio, alsa, alsa-oss to experimented a lot until I figured out why Skype was failing to properly delivery sound and record via my embedded laptop mic.

Anyways, while researching on the cause of my Thinkpad r61 mic issues, I've red a bunch of blog posts by people experiencing microphone oddities with Lenovo Thinkpads

Throughout the search I come across one very good article, which explained that in many cases the Thinkpad sound problems are caused by the snd-hda-intel alsa kernel module. snd-hda-intel fails to automatically set proper sb model type argument during Linux install when the soundcard is initialized with some argument like options snd-hda-intel model=auto

Hence, the suggested fix which should resolve this on many Thinkpad notebooks is up to passing the right module argument:

To fix its neceessery to edit /etc/modprobe.d/alsa-base.conf .

debian:~# vim /etc/modprobe.d/alsa-base.conf

Find the line in the file starting with:
options snd-hda-intel model=

and substitute with:

options snd-hda-intel model=thinkpad

Finally a restart of Advaned Linux Sound Architecture (alsa) is required:

debian:~# /etc/init.d/alsa restart
...

At most cases just restarting the alsa via its init script is not enough, since the ssnd-hda-intel kernel module is already in use by some program or something, so its best to do a reboot to make sure the module is loaded with the new model=thinkpad argument.

My exact laptop sound card model is:

debian:~# lspci |grep -i audio
00:1b.0 Audio device: Intel Corporation 82801H (ICH8 Family) HD Audio Controller (rev 03)

After changing the module and using alsamixer and aumix to make sure mic is unmuted and its volume is high enough, mic sound rec works fine.

Upgrading Skype 2.0 to Skype 2.2 beta on Debian GNU / Linux – Skype Mic hell

Saturday, December 31st, 2011

Making Skype work with Alsa on Debian GNU / Linux

Though, I'm GNU / Linux user for many years now. I have to say, everything is not so perfect as many people present it.
Configuring even simple things related to multimedia on Linux is often a complete nightmare.
An example, today I've decided to upgrade my 32 bit Skype version 2.0 beta for Linux to 64 bit Skype 2.2 beta .
The reason I was motivated to upgrade skype was basicly 2.

a) My Skype run through 32 bit binary emulation with /usr/bin/linux32

b) I had issues with my skype if someone give me a Skype Call, while I have a flash video or some other stream in Browser (let's say Youtube).
Actually being unable to receive a skype call or initiate one while I have some kind of music running in the background or just some kind of Youtube video paused was really annoying. Hence until now, everytime I wanted to speak over skype I had to close all Browser windows or tabs that are using my sound card and then restart my Skype program ….

Just imagine how ridiculous is that especially for a modern Multimedia supporting OS as Linux is. Of course the problems, I've experienced wasn't directly a problem of Linux. The problems are caused by the fact I have to use the not well working proprietary software version of Skype on my Debian GNU / Linux.
I would love to actually boycott Skype as RMS recommends, but unfortunately until now I can't, since many of my friends as well as employers use Skype to connect with me on daily basis.
So in a way I had to migrate to newer version of skype in order to make my Linux experience a bit more desktop like …

Back to the my skype 2.0 to 2.2. beta upgrade story, the overall Skype upgrade procedure was easy and went smootlhy, setting correct capturing later on however was a crazy task ….
Here is the step by step to follow to make my upgraded skype and internal notebook mic play nice together:

1. Download 64 bit Skype for Debian from skype.com

For the sake of preservation in case it disappears in future, I've made a mirror of skype for debian you can download here
My upgrade example below uses directly the 64 bit Skype 2.2beta binary mirror:

Here are the cmds once can issue if he has to upgrade to 2.2beta straight using my mirrored skype:

debian:~# wget https://www.pc-freak.net/files/skype-debian_2.2.0.35-1_amd64.deb
...

2. Remove the old version of skype

In my case I have made my previous skype installation using .tar.bz2 archive and not a debian package, however for some testing I also had a version of skype 2.0beta installed as a deb so for the sake of clarity I removed the existing skype deb install:

debian:~# dpkg -r skype
...

3. Install skype-debian_2.2.0.35-1_amd64.deb downloaded deb

debian:~# dpkg -i skype-debian_2.2.0.35-1_amd64.deb
...

After installing skype, I installed pavucontrol A volume control for the PulseAudio sound server

4. Install pavucontrol

debian:~# apt-get install pavucontrol

PavUcontrol PulseAudio mixer screenshot

Pavucontrol has plenty of sound configurations and enables the user to change many additional settings which cannot be tuned in alsamixer

pavucontrol was necessery to play with until I managed to make my microphone able to record.

5. Build and install latest Debian (Testing) distribution alsa driver

debian:~# aptitude install module-assistant
debian:~# m-a prepare
debian:~# aptitude -t testing install alsa-source
debian:~# m-a build alsa
debian:~# m-a install alsa
debian:~# rmmod snd_hda_intel snd_pcm snd_timer snd soundcore snd_page_alloc
debian:~# modprobe snd_hda_intel
debian:~# echo 'options snd-hda-intel model=auto' >> /etc/modprobe.d/alsa-base.conf

In my case removing the sound drivers and loading them once again did not worked, so I had to reboot my system before the new compiled alsa sound modules gets loaded …
The last line echo 'options snd-hda-intel model=auto' … was necessery for my Thinkpard r61 Intel audio to work out. For some clarity my exact sb model is:

debian:~$ lspci |grep -i audio
00:1b.0 Audio device: Intel Corporation 82801H (ICH8 Family) HD Audio Controller (rev 03)

For other notebooks with different sound drivers echo 'options snd-hda-intel model=auto' … should be omitted.

6. Tune microphone and sound settings in alsamixer

debian:~$ alsamixer

Alsamixer Select Soundcard Debian Linux Screenshot
Right after launching alsamixer I had to press F6: Select Sound Card and choose my sound card (0 HDA Intel).

Following my choice I unmuted all the microphones and enabled Microphone Boost as well as did some adjustments to the MIC volume level.

Alsamixer My Intel SoundCard Debian Linux

Setting proper MIC Volume levels is absolutely necessery, otherwise there is a constant noise getting out of the speakers …

7. Use aumix to set some other sound settings

For some unclear reasons, besides alsamixer , I often had to fix stuff in aumix . Honestly I don't understand where exactly aumix fits in the picture with Alsa and my loaded alsa sound blaster module?? If someone can explain I'll be thankful.

Launch aumix to further adjust some sound settings …

debian:~$ aumix

Aumix Debian GNU Linux Squeeze Screenshot

In above screenshot you see, my current aumix settings which works okay with mic and audio output.

9. Test Microphone the mic is capturing sounds correctly

Set ~/.asoundrc configuration for Skype

Edit ~/.asoundrc and put in:

pcm.pulse {
type pulse
}
ctl.pulse {
type pulse
}
pcm.!default {
type pulse
}
ctl.!default {
type pulse
}
pcm.card0 {
type hw
card 0
}
ctl.card0 {
type hw
card 0
}
pcm.dsp0 { type plug slave.pcm "hw:0,0" }
pcm.dmixout {
# Just pass this on to the system dmix
type plug
slave {
pcm "dmix"
}
}
pcm.skype {
type asym
playback.pcm "skypeout"
capture.pcm "skypein"
}
pcm.skypein {
# Convert from 8-bit unsigned mono (default format set by aoss when
# /dev/dsp is opened) to 16-bit signed stereo (expected by dsnoop)
#
# We cannot just use a "plug" plugin because although the open will
# succeed, the buffer sizes will be wrong and we will hear no sound at
# all.
type route
slave {
pcm "skypedsnoop"
format S16_LE
}
ttable {
0 {0 0.5}
1 {0 0.5}
}
}
pcm.skypeout {
# Just pass this on to the system dmix
type plug
slave {
pcm "dmix"
}
}
pcm.skypedsnoop {
type dsnoop
ipc_key 1133
slave {
# "Magic" buffer values to get skype audio to work
# If these are not set, opening /dev/dsp succeeds but no sound
# will be heard. According to the ALSA developers this is due
# to skype abusing the OSS API.
pcm "hw:0,0"
period_size 256
periods 16
buffer_size 16384
}
bindings {
0 0
}
}
I'm not 100% percent if putting those .asoundrc configurations are necessery. I've seen them on archlinux's wiki as a perscribed fix to multiple issues with Skype sound in / out.

Onwardds, for the sake of test if my sound settings set in pavucontrol enables the internal mic to capture sound I used two programs:

1. gnome-sound-recorder
2. arecord

gnome-sound-recorder GNU / Linux Screenshot
gnome-sound-recorder

gnome-sound-recorder is probably used by most GNOME users, though I'm sure Linux noviced did not play with it yet.

arecord is just a simple console based app to capture sound from the microphone. To test if the microphone works I captured a chunk of sounds with cmd:

debian:~$ arecord cow.wav
Recording WAVE 'cow.wav' : Unsigned 8 bit, Rate 8000 Hz, Mono

Later on I played the file with aplay (part of alsa-utils package in Debian), to check if I'll hear if mic succesfully captured my voice, e.g.:

debian:~$ play cow.wav
cow.wav:
File Size: 22.0k Bit Rate: 64.1k
Encoding: Unsigned PCM
Channels: 1 @ 8-bit
Samplerate: 8000Hz
Replaygain: off
Duration: 00:00:02.75
In:100% 00:00:02.75 [00:00:00.00] Out:22.0k [-=====|=====-] Clip:0
Done.

By the way, the aplay ASCII text equailizer is really awesome 😉 aplay is also capable of playing (Ogg Vorbis .ogg) free sound format.

Further on, I launched the new installed version of skype and tested Skype Calls (Mic capturing), with Skype Echo / Sound Test Service
I'll be glad to hear if this small article, helped anybody to fix any skype Linux related issues ?. I would be happy to hear also from people who had similar issues with a different fixes for skype on Linux.
Its also interesting to hear from Ubuntu and other distributions users if following this tutorial had somehow helped in resolving issues with Skype mic.

How to fix multiple instance music streams with sound card (Intel 82801I ICH9 Family) alsa sound problems on Ubuntu 11.04 GNU / Linux

Thursday, October 27th, 2011

Ubuntu Logo Sound / Pulseaudio multiple sound channel issues

The Ubuntu Linux installed previously on Acer ASPIRE 5736Z on my sisters notebook works quite fine. However today she complained about an issue with her sound. The explanation of the problem she faced is:

When she plays a movie file and pauses it and then switches to a music player, suddenly the notebook sound disappears completely until she restarts all the running programs using the sound server. The Acer Aspire is used with a GNOME Desktop, hence my bet was the issues are most probably caused by some kind of mess happening inside Pulseaudio or the way Alsa loaded kernel drivers handles the multiple sound channel streams.

I’m using GNU / Linux for more than 11 years now and I have faced the same sound issues so many times, so when I heard about the problem I thought its pretty normal.
Anyways, what was really irritating in these situation is that when her laptop sound disappears a video or sound files which are to be played by Mozilla Firefox Browser or Chrome are also loosing the sound.
This causes big issues, especially taking in consideration the fact that she had no idea about computers and is a GUI Desktop user, who have no idea how to restart the pulseaudio server to fix the problem etc.

As a good brother, I took the time to check about the issues related to the specific model of Audio Module Hardware / Sound Card, first I checked the exact model of audio the Acer Aspire 5736Z is equipped with:

stanimiraaaa@Ubuntu-Aspire-5736Z:~$ lspci |grep -i audio
00:1b.0 Audio device: Intel Corporation 82801I (ICH9 Family) HD Audio Controller (rev 03)

I checked about any reported other users issues on the net and I found a user somewhere (lost the link), complaining he is experiencing the same sound oddities on his Acer ASPIRE

The fix he suggested is actually quite simple and comes to adding a simple line to /etc/modprobe.d/alsa-base.conf :

stanimiraaaa@Ubuntu-Aspire-5736Z:~$ sudo su -
[sudo] password for stanimiraaaa:
root@Ubuntu-Aspire-5736Z:~# echo 'options snd_hda_intel model=auto' >> /etc/modprobe.d/alsa-base.conf

Next I restartarted to make the new settings take effect. Its also possible to do it without restart, by unloading and loading the alsa module but I’m a lazy kind of person and the machine is notablyunimportant so why should I bother 😉

One important note here is that I removed also an .asoundrc file, that I created some long time ago and this file might have been creating also some sound issues, the content of ~/.asoundrc, before I delete it in her home user, was like so:

stanimiraaaa@Ubuntu-Aspire-5736Z:~$ cat ~/.asoundrc
pcm.!default {
type hw
card 1
device 0
}
ctl.!default {type hw
card 1
device 0
}
stanimiraaaa@Ubuntu-Aspire-5736Z:~$ rm -f .asoundrc

Doing this minor changes to the Ubuntu system erradicated the sound problems and now the sound with simultaneous sound channel streams works just perfect! Thx God 😉