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14

I think the reason this isn't working for you is because that interface has been deprecated. You normally can't write audio using /dev/dsp anymore, at least without being tricky. There is a program that will accomplish this for you on your system: padsp. This will map the /dev/audio or /dev/dsp file to the new Audio Server system. Fire up the terminal and ...


14

You can use amixer. It's in the alsa-utils package on ubuntu/debian. Run it without parameters to get an overview about your devices. amixer Then use the set command to set the volumn. For example to set the master channel to 50%: amixer set Master 50


12

As @Teresa-e-Junior pointed out pactl is the tool to use: First of all we might want to get the IDs of our PA sinks. On my system this is what I get: $ pactl list short sinks 0 alsa_output.pci-0000_01_00.1.hdmi-surround module-alsa-card.c s16le 6ch 44100Hz SUSPENDED 1 alsa_output.pci-0000_00_1b.0.analog-stereo module-alsa-card.c ...


4

I'm not exactly sure what you meant with "ALSA or PulseAudio", I assume you meant PulseAudio over ALSA. I'm also in the dark, in regards to your distribution, so I'm prevented from being very specific. If you provide your distro + version, I can let you know if this problem has known workarounds. GNU/Linux audio has improved, but it's not on level with ...


4

You can use udev rules. For this, you'll need to find the device attributes that distinguish your USB headset once it is plugged in. This can be usually done in two steps: Run udevadm monitor --udev befor plugging the device. Then plug it in. There will be a bunch of output lines - just pick one and copy the device path from it - it is something like ...


4

I finally found a solution to this, it took me a while to find so I post it here in case it might help others. Edit the file: /etc/pulse/default.pa Look for the line: load-module module-udev-detect And change it into: load-module module-udev-detect ignore_dB=1


4

I use: cvlc --no-one-instance --volume 150 <soundfile> to play short sounds (< 1 second) as notifiers for program activities. The --no-one-instance makes sure this playing does not get scheduled after something that might be running in my 'normal' vlc (like music) and which is setup to have a single instance and for which additional invocations ...


4

That's because /proc (and /sys) are kernel interfaces. Nothing in there is a real file on disk. The information comes directly from the OS. The individual files are kind of like socket interfaces, and when you read them you are making a request for data. The proc filesystem apparently originates with UNIX 8 (although the Linux implementation, like most ...


4

OSS is a relict of old kernel 2.4 times. Since kernel 2.6 (over 10 years ago!) ALSA is the default sound system in linux. If you really need to use OSS, there should be a alsa-oss kernel module, which provides some OSS functionality on top of ALSA. Have a look at speaker-test this is even better than your dd.


4

ALSA stands for Advanced Linux Sound Architecture, I'd encourage you to poke around their project website if you're truly curious. Specifically I'd take a look at the "I'm new to ALSA pages & tutorials. The ArchLinux wiki probably describes it the best. The Advanced Linux Sound Architecture (ALSA) is a Linux kernel component which replaced the ...


3

Pulse Audio has the functionality you're looking for. See the Network Documentation. CentOS packages are available in EPEL.


3

Most likely it's pulse that's not set-up correctly. speaker-test is a handy utility here. If pulse is your default audio device then all audio programs including speaker-test will go through it for audio. Try adding default-sample-channels = 6 to /etc/pulse/daemon.conf to tell pulse to use 5.1 audio. speaker-test -c6 will test all 6 channels ...


3

Remember that Alsa is not removed when installing Pulseaudio. Pulseaudio uses Alsa in quite some extent and you could say it's a layer on top of Alsa. There's pavucontrol (on Debian/Ubuntu also the package name). It allows you to do a little bit more configuration on the Pulseaudio server and Pulseaudio-managed things like Bluetooth audio profile selection. ...


3

I finally found an answer to this problem based on this post from another forum (see "Attempt 1b" for reference). According to it, one can specify the module id, even if the driver used is the same: alias char-major-116 snd alias snd-card-0 snd-hda-intel alias snd-card-1 snd-hda-intel options snd cards_limit=2 options snd-hda-intel id=SB index=0 options ...


3

If I understand correctly, you want playback on your build in sondcard and capture (microphone) from external USB device. Your external device is listed as card 2: device 0 and your build in soundcard as card 0: device 0 I think your asound.conf should look something like this: pcm.!default { playback.pcm { type hw card 0 device 0 } ...


3

To just change the default cards, replace that asound.conf with this: defaults.pcm.card 2 defaults.pcm.device 0 To be independent from changing card numbers, use the card ID instead: defaults.pcm.card PCH defaults.pcm.device 0


3

I think you can do this with PulseAudio. I found this tutorial that shows how, titled: Redirect Audio Out to Mic In (Linux). General Steps Run the application pavucontrol. Go to the "Input Devices" tab, and select "Show: Monitors" from the bottom of the window. If your computer is currently playing audio, you should see a bar showing the volume of the ...


3

If this is a feature of the device and not just the software it came with (it's probably the former), then it has to be implemented in the kernel driver.1 It appears to me that normally, Realtek PCI sound cards actually use the Intel HDA driver with a patch to support the ALC codecs (for anyone who's interested: [src]/sound/pci/hda/patch_realtek.c). ...


2

Maybe you removed alsa related packages or you messed up their installation. Since there are no snd_* modules loaded, it's probable that /etc/modprobe.d/alsa-base.conf is wrong or missing. Try reinstalling alsa and reboot: apt-get --reinstall install alsa-base alsa-oss alsa-utils gstreamer0.10-alsa Sometimes, upgrading a package, with many dependencies, ...


2

It seems like I found a solution, at least for this particular case. Since I knew the card and device number assigned by ALSA, I just had to open /etc/pulse/default.pa. in editor and change this line #load-module module-alsa-sink into this load-module module-alsa-sink device=hw:2,7 where 2 and 7 are my particular instances of card and device numbers. ...


2

You can create your own alsa config file ~/.asoundrc which overrides /etc/alsa.conf. It is possible then to create your own aliases for PCM devices. For example, 'PCM' in mixer is just an alias for the device ie. hw:0,1


2

The basis of the problem is you have to change which sound card is being used. How to do this depends on whether you are using a gui or something else. I know next to nothing about pulse audio so I can't help with directions directly for that. But if you're using KDE you can open your "K" menu -> system settings -> multimedia -> phonon -> Audio Hardware ...


2

You can use amixer to better control the volume. But it really depends on the channel/card. For example, my card only has 255 levels, so even if I issue amixer set PCM '0.1dB-', the volume is reduced by a full 0.2 dB. Btw, it's a command line program, not graphical control. See man amixer or amixer -h.


2

After asking this questions multiple times on gentoo IRC and finally the forums, I was pushed in the right direction and able to solve the problem. ~ $ aplay -l **** List of PLAYBACK Hardware Devices **** card 0: PCH [HDA Intel PCH], device 0: ALC892 Analog [ALC892 Analog] Subdevices: 1/1 Subdevice #0: subdevice #0 card 0: PCH [HDA Intel PCH], device 1: ...


2

I get just about same output from arecord -l. I've used audacity to record a song from a website (yes, I tried to buy the song from the site, but...). Need to get the right adjustment of muting the microphone (if possible) and then cleaning up any static/background noise. So for you, I would start audacity, start your program, set to record to a new ...


2

From what I know you can use the softvol plugin for ALSA and set the max_dB value. Something in direction of: pcm.!default { type plug slave.pcm "softvol" } pcm.softvol { type softvol slave { pcm "hw:0,0" # pcm "dmix" or this or the like. } control { name "PreAmp" card 0 } ...


2

I've done this in the past and have had pretty good success with boosting the volume up. The only problem with it is if you turn it down, then you'll have to come back into the system preferences under the speaker icon to turn it back up to 150%. However with the steps below should remain "sticky" from reboot to reboot. ...


2

You can pipe your music to the speakers of the other computer. See something similar here, but with the microphone: http://www.commandlinefu.com/commands/view/350/output-your-microphone-to-a-remote-computers-speaker


2

I found solution here: http://thenerdshow.com/index30e5.html there I've found : $ amixer -c0 cset iface=MIXER,name='Input Source',index=1 'Front Mic' # (Record from Front Mic) slightly modified according to my sound-card and setup (default sound-card, different items ordering) : $ amixer cset name='Input Source',index=0 'Rear Mic'


2

This has nothing to do with CONFIG_SND_SEQUENCER. The particular patch that caused this regression has been found. The fix has been posted and should appear in some stable kernel eventually.



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