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Oddly, this works. This is now my configuration
Simple mixer control 'Master',0 Capabilities: pvolume pvolume-joined pswitch pswitch-joined penum Playback channels: Mono Limits: Playback 0 - 64 Mono: Playback 51 [80%] [-13.00dB] [on] Simple mixer control 'Headphone',0 Capabilities: pswitch penum Playback channels: Front Left - Front Right Mono: Front Left: Playback [on] Front Right: Playback [on] Simple mixer control 'PCM',0 Capabilities: pvolume penum Playback channels: Front Left - Front Right Limits: Playback 0 - 255 Mono: Front Left: Playback 255 [100%] [0.00dB] Front Right: Playback 255 [100%] [0.00dB] Simple mixer control 'Front',0 Capabilities: pvolume pswitch penum Playback channels: Front Left - Front Right Limits: Playback 0 - 64 Mono: Front Left: Playback 64 [100%] [0.00dB] [on] Front Right: Playback 64 [100%] [0.00dB] [on] Simple mixer control 'Front Mic',0 Capabilities: pvolume pswitch penum Playback channels: Front Left - Front Right Limits: Playback 0 - 31 Mono: Front Left: Playback 31 [100%] [12.00dB] [off] Front Right: Playback 31 [100%] [12.00dB] [off] Simple mixer control 'Surround',0 Capabilities: pvolume pswitch penum Playback channels: Front Left - Front Right Limits: Playback 0 - 64 Mono: Front Left: Playback 64 [100%] [0.00dB] [on] Front Right: Playback 64 [100%] [0.00dB] [on] Simple mixer control 'Center',0 Capabilities: pvolume pvolume-joined pswitch pswitch-joined penum Playback channels: Mono Limits: Playback 0 - 64 Mono: Playback 64 [100%] [0.00dB] [on] Simple mixer control 'LFE',0 Capabilities: pvolume pvolume-joined pswitch pswitch-joined penum Playback channels: Mono Limits: Playback 0 - 64 Mono: Playback 64 [100%] [0.00dB] [on] Simple mixer control 'Line',0 Capabilities: pvolume pswitch penum Playback channels: Front Left - Front Right Limits: Playback 0 - 31 Mono: Front Left: Playback 0 [0%] [-34.50dB] [off] Front Right: Playback 0 [0%] [-34.50dB] [off] Simple mixer control 'CD',0 Capabilities: pvolume pswitch penum Playback channels: Front Left - Front Right Limits: Playback 0 - 31 Mono: Front Left: Playback 0 [0%] [-34.50dB] [off] Front Right: Playback 0 [0%] [-34.50dB] [off] Simple mixer control 'Mic',0 Capabilities: pvolume pswitch penum Playback channels: Front Left - Front Right Limits: Playback 0 - 31 Mono: Front Left: Playback 31 [100%] [12.00dB] [off] Front Right: Playback 31 [100%] [12.00dB] [off] Simple mixer control 'IEC958',0 Capabilities: pswitch pswitch-joined cswitch cswitch-joined penum Playback channels: Mono Capture channels: Mono Mono: Playback [off] Capture [on] Simple mixer control 'IEC958 Default PCM',0 Capabilities: pswitch pswitch-joined penum Playback channels: Mono Mono: Playback [on] Simple mixer control 'Capture',0 Capabilities: cvolume cswitch penum Capture channels: Front Left - Front Right Limits: Capture 0 - 31 Front Left: Capture 22 [71%] [19.50dB] [on] Front Right: Capture 22 [71%] [19.50dB] [on] Simple mixer control 'Capture',1 Capabilities: cvolume cswitch penum Capture channels: Front Left - Front Right Limits: Capture 0 - 31 Front Left: Capture 22 [71%] [19.50dB] [on] Front Right: Capture 22 [71%] [19.50dB] [on] Simple mixer control 'Channel Mode',0 Capabilities: enum Items: '2ch' '6ch' Item0: '2ch' Simple mixer control 'Digital',0 Capabilities: cvolume penum Capture channels: Front Left - Front Right Limits: Capture 0 - 120 Front Left: Capture 95 [79%] [17.50dB] Front Right: Capture 95 [79%] [17.50dB] Simple mixer control 'Input Source',0 Capabilities: cenum Items: 'Mic' 'Front Mic' 'Line' 'CD' Item0: 'Front Mic' Simple mixer control 'Input Source',1 Capabilities: cenum Items: 'Mic' 'Front Mic' 'Line' 'CD' Item0: 'Mic'In alsamixer, I muted the front mic in 'Playback' and for some reason, the digital control in the 'Capture' controls the mic volume. I can confirm that this set up also works in Skype. I don't understand why the 'digital' control affects the mic volume - it's an analogue device.
ADDED: Thanks, gromloc.
Oddly.... it seems I'm not the only one confused by alsa.
"Mic" in Playback [F3] is what controls how loud you hear the input of your mic output. It's the playback of your mic - but as evident as this is being spelled out like this, it's quite confusing in practice. You see "Mic" and you immediately think of input. That wouldn't be alsamixer's fault or problem and we'd figure it out eventually. I'd suggest renaming it to MicPB or something...
But there's a "Mic Boost" control right next to it. I doubt you're not thinking input again, this time rightfully. Why would I boost the ouptut/playback of my microphone ? I need to amplify input since the output isn't usually even on my side, (the output is someone else's skype), and is controlled on that computer.
I'm being framed into thinking I'm dealing with input by seeing "Mic boost" next to "Mic". You then scratch your head wondering what's with all the noise in your speakers and look into "Capture" [F4], but find that "Mic" doesn't have a volume bar and it's only tag-able with a red "L CAPTURE R". You say ok... this is how we select what input device we capture and then you find a control called "Capture" instead of "Master" or something else, that is itself tagged "L CAPTURE L". Do you mean MUTE/UNMUTE all input...?
To make matters worse you have a "Mic select" in "Playback" allowing you to select between mic1 and mic2 (which are input devices), but there's no mic2 in "Capture" to select.
As a side note, pulseaudio is miss represented as a sound card, and while I understand it's a pseudo-soundcard, alsamixer keeps this information for itself.
As a bonus while trying to write this I pressed F5 with the wrong window in focus having to rewrite this whole thing. This isn't alsamixer's fault but I'm not sure is F3,F4 and F5 where the best options.
Feel free to move this to off-topic, I'm basically curious if anyone else here had headaches like me, I'm not here to report a bug, unless the community decides this is actually an issue.
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Here I was, headphones clamped tight, head'n into a holiday weekend, slide'n thru a '75 live concert on stereo bluetooth headphones (thanks to Bluez and ALSA; performed entirely through the Linux command line)... when... for some insane reason I check the Arch forums. Must be some kind of right brain/left brain thing. Anyway... I hit your post and just had to chuckle.
You're totally correct. alsamixer has a very schizophrenic nature.
But...
Take an insane amount of time to dive in below the covers.
For example, let's just assume that you're using a typical modern motherboard that has an Intel HDA chip. That chip is only half of it (and the very simple half). There's the codec chip it controls... lot's of 'em:
http://kernel.org/doc/Documentation/sou … Models.txt
Now, download the datasheets for those codec chips. It won't take too long to determine that they all have very specific requirements. The "nodes" are different. The capabilities are different... 2 channels vs 7.1 for instance. Amplifiers are on different nodes... hmmm... what about jack-sense (automatically determining if you plugged something into your box)?
But don't ignore the basics. What format does the hardware require??? If you're the developer you need to determine that this codec chip must have (for example) 2 channels (stereo), S16_LE (signed 16 bit little endian), PCM formatted data, at either 44100, or 48000, or 96000 samples per second... or something entirely different for another chip.
So perhaps we should have some sort of "plugin" architecture:
http://www.alsa-project.org/alsa-doc/al … ugins.html
as goofy as that may be documented.
We also got's SPDIF and HDMI (Hollywood's *&#$@ attempt to plug the "analog hole") to contend with.
But let's not forget that this was just a "corner case". There's more... so let's do a very shallow dip into "professional" audio equipment.
Is it on the PCI, PCMCIA, Firewire, or USB bus?
Does it do MIDI?
Don't forget that latency is a huge issue in the pro environment... so we better hook into Jack with some sort of master clock while bringing that down into the low millisecond range on a kernel that has multiple schedulers (you do know about run vs wait queue timing don't you?).
oh, jeeze... facepalm... you were talking about alsamixer.
You're totally right. They absolutely need to determine the playback vs capture controls. It might only take a simple flag buried somewhere in the internals.
Patches are alway welcome... and welcome to open source... now I'm going to head on over to some floyd and listen to "Welcome to the Machine"
And (very truly) thanks for providing one of the most down to earth posts I've read on this forum... ever.
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