Now you should test if the sound driver really is available, then try to use it.
You can find a lot of useful information about your system in the /proc subdirectory. /proc is a "virtual" filesystem, meaning that it does not exist in real life, but merely is a mapping to various processes and tasks in your computer. In order for /proc to work, you need to have support for it compiled into your kernel. Most linux distributions have this as a default, but if you compiled a kernel and left /proc out obviously there won't be anything in /proc. .:: add-website-to-google-search-index.vercel.app ::.
/proc/modules gives information about loaded modules. Once the ALSA sound drivers are loaded, if you type cat /proc/modules you should see something like:
snd-pcm1-oss 4 0
snd-sb16 1 1
snd-sb-dsp 4 [snd-sb16] 0
snd-pcm1 4 [snd-pcm1-oss snd-sb-dsp] 0
snd-pcm 3 [snd-pcm1-oss snd-sb16 snd-sb-dsp snd-pcm1] 0
snd-mixer 3 [snd-pcm1-oss snd-sb16 snd-sb-dsp] 1
snd-mpu401-uart 1 [snd-sb16] 0
snd-midi 4 [snd-sb16 snd-sb-dsp snd-mpu401-uart] 0
snd-opl3 1 [snd-sb16] 0
snd-synth 1 [snd-sb16 snd-opl3] 0
snd-timer 1 [snd-opl3] 0
snd 8 [snd-pcm1-oss snd-sb16 snd-sb-dsp snd-pcm1 snd-pcm snd-mixer snd-mpu401-uart snd-midi snd-opl3 snd-synth snd-timer] 0
For example (Note: you should never issue this command as follows, the cs4236 driver needs options):
win3:~# modprobe snd-card-cs4236
/lib/modules/2.0.35/misc/snd-card-cs4236.o: init_module: Device or resource busy
snd-mixer: Device or resource busy
win3:~# cat /proc/modules
snd-cs4236 2 0
snd-cs4231 3 [snd-cs4236] 0
snd-timer 1 [snd-cs4231] 0
snd-pcm1 4 [snd-cs4236 snd-cs4231] 0
snd-mixer 3 [snd-cs4236 snd-cs4231] 0
snd-pcm 3 [snd-cs4236 snd-cs4231 snd-pcm1] 0
snd-mpu401-uart 1 0
snd-midi 4 [snd-mpu401-uart] 0
snd-opl3 1 0
snd-synth 1 [snd-opl3] 0
snd-timer 1 [snd-cs4231 snd-opl3] 0
snd 8 [snd-cs4231 snd-timer snd-pcm1 snd-mixer snd-pcm] 0
bash$ cat /proc/asound/cards
0 [card1 : SB16 - Sound Blaster 16
Sound Blaster 16 at 0x220, irq 5, dma 1&5
win3:~# cat /proc/asound/cards
--- no soundcards ---
0 [card1 ]: CS4236 - CS4237B
CS4237B at 0x534, irq 7, dma 1&0
The /proc/asound/ virtual directory shows lots of other information
about the driver. Please note that /proc/asound/ will only exist after you inserted
the first ALSA module. If there is no /proc/asound, it simply means that the "snd"
module was not loaded properly. You can find installed cards in /proc/asound/cards, then
find information about card0 in /proc/asound/0, /proc/asound/1 for card1
etcetera.
If cat /proc/asound/card1/pcm0 shows something like
ES1370 DAC2/ADC
Playback isn't active.
Record isn't active.
For users of a 2.0.x kernel there is a third method to find information about the sound devices, namely if you inserted the OSS compatible driver there is a /dev/sndstat device. The ALSA drivers kindly request that you not to rely on this information as it is only there for compatibility with the OSS drivers and better information can easily be obtained from /proc/asound/. In kernel 2.2.x ALSA uses the kernel soundcore and therefor cannot emulate /dev/sndstat, since it would interfere with the OSS drivers.
Once the drivers for your sound card have been installed and your /proc filesystem tells you so, you can try to make a real sound. To do this, you need to set the mixer volumes to a reasonable value. You need the ``amixer'' from the alsa-utils package for this. First of all, install the utility package, or at least put the "amixer" command in some reasonable place (like /usr/local/bin).
Version 0.3.2 and later have an interface that differs from the OSS drivers. If you type just ``amixer'' you will see the mixer elements and their value. One of these elements could be ``Master volume'' for example, and could look like: [Issues with JWT]
Group 'Master',0
Capabilities: volume
Channels: Front-Left Front-Right
Limits: min = 0, max = 31
Front-Left: 31 [100%] [on] [---]
Front-Right: 26 [84%] [on] [---]
Unfortunately, I do not know how to set left and right volumes independently. With amixer, you can change volumes with the ``amixer set'' command. For example, to change the Master volume, you would issue a
amixer set Master 15
Please note that the names of the elements can be different for different types of sound cards. Also note that amixer is case dependent, so ``amixer set masteR 10'' will not work. For more information, please look in the amixer man page. .:: telegra.ph ::.
If you have a 0.3.0-pre4 ALSA, then amixer works just like normal mixer programs. You can look at the mixer settings by typing ``amixer''. This command lists the ``mixer settings'', or as you would normally call it, the volume settings of the various parts of the soundcard. The output from amixer can greatly differ from card to card. My Soundblaster 16 shows:
Master 0 % (-14.00dB) : 0 % (-14.00dB)
Bass 0 % (-14.00dB) : 0 % (-14.00dB)
Treble 0 % (-14.00dB) : 0 % (-14.00dB)
Synth 0 % (-62.00dB) : 0 % (-62.00dB)
PCM 0 % (-62.00dB) : 0 % (-62.00dB)
Line-In 0 % (-62.00dB) : 0 % (-62.00dB) Mute
MIC 0 % (-62.00dB) : 0 % (-62.00dB) Mute
CD 0 % (-62.00dB) : 0 % (-62.00dB) Mute
In-Gain 0 % (-18.00dB) : 0 % (-18.00dB)
Out-Gain 0 % (-18.00dB) : 0 % (-18.00dB)
PC Speaker 0 % (-18.00dB) : 0 % (-18.00dB)
If you only get a message like ``amixer: Specify command...'', then you are using the ALSA 3.2 utilities. I suggest you to upgrade to 0.4.1e or later, or to go back to 0.3.0-pre4.
You have noticed the "Mute" entry for some devices. This means that this particular device will be zeroed out, whatever volume setting you use. Some cards (the CS4237B in the example) even mute their master channel. So, for the CS4237B, I would have to type
amixer set "Master d" unmute
to even be able to produce any sound at all. The Soundblaster does not have muted output, but
amixer set Master 100 unmute
would set the volume to 100% - and unmute it if it would have been muted. You can use a number, a word like "mute" or "unmute", or both. Type
amixer set "Master d" 100; amixer set PCM 100 unmute
to set the CS4237B card to maximum master volume and unmute PCM volume and set it to maximum.
If you use an older version of amixer, you need to leave out the ``set'' part of the command, so you would just type
amixer "master d" 100
The various mixer parts may confuse you if you have no knowledge of digital sound production. The sound-HOWTO may help a bit, but a very short introduction is here.
You will probably only need few mixer elements: one of them is the ``CD'' setting (this is analog sound of your CD player, most CD players are connected with a 3 or 4 wire red/white/black cable).
The ``PCM'' setting is used for most applications. Programs like mpg123, xmms, speakfreely, realplayer and most others use the PCM channel. [Research Paper: ASP.NET]
``MIC'' stands for microphone, ``line-in'' is an (optional) extra input at the back of your sound card. .:: jobboard.piasd.org ::.
The various ``gain'' parts offer extra amplification for various uses and are pretty self-explanatory. (Like: record-gain is extra amplification for the recording channel, which can be useful if you use a microphone).
You would set the CD channel to record by typing
amixer set CD capture
and stop the recording setting again by typing
amixer set CD nocapture.
Note that older amixer programs use ``amixer CD rec'' and ``amixer CD norec'' for this.
If you would like to record something from the microphone, you would probably use
amixer set "Input Gain" 100; amixer set Mic 100 capture mute.
(Using the microphone input unmuted will produce loud high-pitched sound if your mic picks up its own signal from the speakers again). Most microphones have a ``gain'' setting to boost the microphone volume; you are most likely going to need it to pick up any sound from the microphone at all.
Again, older amixer programs use ``amixer "input gain" 100; amixer mic 100 rec mute''.
Unfortunately I have not been able to change the volume of the "3d center" and "3d space" settings with amixer 0.3.0-pre4. I haven't tried yet with 0.4.1e (this particular machine is still running 2.0.38). If anyone succeeds please let me know. I can use alsamixer for this job, but alsamixer was not ported to the 0.4.1e version yet.
The ALSA FAQ says that it is possible to restore mixer settings with
cat <file> > /proc/asound/#/mixerC0D0, where <file> was obtained from
/proc/asound/#/mixerC0D0. I have not been able to reproduce this as my system
complains about non-existing devices. Then there is the ``alsactl'' program, which I don't use. I invite you (yes, you!)
to write this section.
The alsa drivers have native sound-devices in the /dev/snd/ directory. If you have one card you might see the following devices:
/dev/snd/pcmC0D0 - the raw audio device for the card
/dev/snd/mixerC0D0 - the mixer for card 0
/dev/snd/controlC0D0 - the control device for card 0
Now you are ready to put any soundfile you want into the PCM device
of the first card. So try to cat any textfile (any file) to /dev/snd/pcmC0D0,
like this: cat <filename> > /dev/snd/pcmC0D0. The filename can
be any file, as long as it has some length. If you have a soundfile lying
around somewhere, you could try that. You could also get the file at
http://www.ldp.org/sounds/english.au
this is Linus Torvalds saying how to pronounce Linux.
The default setting of your sound device is 8000 Hz, 8 bit. That means that the "english.au" file mentioned above will produce speech, other test files will probably just produce noise. If you do not hear anything, check your speakers, try to run "amixer" again or consult a doctor. (Later on you can easily use the full 48 KHz, 16 bit features of your sound card, by using your favourite sound player like sox or mpg123).
If you loaded the ``snd-pcm1-oss'' module, you can also use the OSS-compatibility to access your sound card. The following mappings are made:
/dev/snd/pcmC0D0 -> /dev/audio0 (/dev/audio) -> minor 4
/dev/snd/pcmC0D0 -> /dev/dsp0 (/dev/dsp) -> minor 3
/dev/snd/pcmC0D1 -> /dev/adsp0 (/dev/adsp) -> minor 12
/dev/snd/pcmC1D0 -> /dev/audio1 -> minor 4+16 = 20
/dev/snd/pcmC1D0 -> /dev/dsp1 -> minor 3+16 = 19
/dev/snd/pcmC1D1 -> /dev/adsp1 -> minor 12+16 = 28
/dev/snd/pcmC2D0 -> /dev/audio2 -> minor 4+32 = 36
/dev/snd/pcmC2D0 -> /dev/dsp2 -> minor 3+32 = 35
/dev/snd/pcmC2D1 -> /dev/adsp2 -> minor 12+32 = 44
You probaly want to use the ``snd-mixer-oss'' module as wel, so you can use the backwards compatible mixer.
The INSTALL file in the ALSA driver directory mentions some tricks to tell the driver which settings to use. If you need these commands it will depend on the application you use to play sound. Regular sound playing applications, like mpg123, sox (mostly called with the ``play'' command), or X11 applications like RealPlayer will probably do fine without these. I never used these anyway.
"Playback erase" - erase all additional informations about OSS applications
"Playback <app_name> <fragments> <fragment_size> [<options>]"
"Record erase" - erase all additional informations about OSS applications
"Record <app_name> <fragments> <fragment_size> [<options>]"
<app_name> - name of application with (highter priority) or without path
<fragments> - number of fragments or zero if auto
<fragment_size> - size of fragment in bytes or zero if auto
<options> - optional parameters
WR_ONLY - if application tries open pcm device with O_RDWR driver rewrites this to O_WRONLY (playback) - good for Quake etc...
Examples:
echo "Playback x11amp 128 16384" > /proc/asound/0/pcm0o
echo "Playback squake 0 0 WR_ONLY" > /proc/asound/0/pcm0o
"Playback 8" -> driver will use always 8-bit DMA channel for playback.
"Playback 16" -> driver will use always 16-bit DMA channel for playback.
"Playback auto" (default) -> driver will use auto mode (first opened direction will use 16-bit DMA channel).
"Record 8" -> driver will use always 8-bit DMA channel for record.
"Record 16" -> driver will use always 16-bit DMA channel for record.
"Record auto" (default) -> driver will use auto mode (first opened direction will use 16-bit DMA channel).
Example: echo "Record 16" > /proc/asound/0/sb16
For further reference, please consult the INSTALL file.