On more than a few occasions, I've wanted to be able to software-record the sound coming out of my computer speakers.  Many users faced with similar problems resort to using pulseaudio as their sound system, which is reasonble, because it provides a very extensible/pluggable framework for sound.  Unfortunately, my experience with pulseaudio in the past has been "meh", probably due in large part to my heavy use of Pd.  So I've stuck with ALSA through the years when doing simple stuff, resorting to jack when doing more complicated routing between applications.  Simply recording what's playing seems simple enough...right?  Not so much...

I guess some (nicer?) sound cards provide a built-in hardware recording channel that can mix back in the currently playing audio.  Most built-in ones, like the one in my aging T410, do not.  After some sleuthing, I discovered that ALSA's plugin system does, in fact, provide a way to do this.  I'll describe the process here, but it's basically ripped from this thread where kokoko3k serves up the right approach:  https://bbs.archlinux.org/viewtopic.php?id=147852

There's an ALSA kernel module called snd_aloop that "provides a pair of cross-connected devices, forming a full-duplex loopback soundcard".  With just a little fiddling, you can create a "looprec" device that has loops back the audio output into a new recordable ALSA device.  The steps, just like in the above-mentioned post, are:

1. $ sudo modprobe snd_aloop
   (this inserts the relevant kernel module into the kernel)

2. create/edit ~/.asoundrc and paste in the following (a bit of alsa black magic):

   pcm.!default {
     type asym
     playback.pcm "LoopAndReal"
     #capture.pcm "looprec"
     capture.pcm "hw:0,0"
   }
   
   pcm.looprec {
       type hw
       card "Loopback"
       device 1
       subdevice 0
   }
   
   pcm.LoopAndReal {
     type plug
     slave.pcm mdev
     route_policy "duplicate"
   }
   
   pcm.mdev {
     type multi
     slaves.a.pcm pcm.MixReale
     slaves.a.channels 2
     slaves.b.pcm pcm.MixLoopback
     slaves.b.channels 2
     bindings.0.slave a
     bindings.0.channel 0
     bindings.1.slave a
     bindings.1.channel 1
     bindings.2.slave b
     bindings.2.channel 0
     bindings.3.slave b
     bindings.3.channel 1
   }
   
   pcm.MixReale {
     type dmix
     ipc_key 1024
     slave {
       pcm "hw:0,0"
       rate 48000
       #rate 44100
       periods 128
       period_time 0
       period_size 1024 # must be power of 2
       buffer_size 8192
     }
   }
   
   pcm.MixLoopback {
     type dmix
     ipc_key 1025
     slave {
       pcm "hw:Loopback,0,0"
       rate 48000
       #rate 44100
       periods 128
       period_time 0
       period_size 1024 # must be power of 2
       buffer_size 8192
     }
   }
   

That's it!  Your recording software should now have a device available called "looprec", and if you record from it you'll get whatever is playing on your speakers.  You can make this permanent by adding the snd_aloop module to /etc/modprobe.d/sound.conf.

Since you've made it this far, I'll share what I was trying to record:  http://websdr.ewi.utwente.nl:8901 -- which is pretty much the raddest thing ever.

Hmmm. Way too complicated probably, but if you can actually get the device to pair and sink it as an audio device, then you can do this: mplayer -ao alsa:device=bluetooth and it may or may not require ~/.asoundrc that looks like

pcm.bluetooth {
    type bluetooth
}

I heard audio at least one time from my headphones via mplayer that way...at least before the batteries ran out. Open question: Why doesn't it just show up like a normal alsa device? Meh.

• reference 1
• reference 2
• Lots of people link here, so Why is this link down??
• some hints?

Ok, enough is enough. You gotta help me out. I can't be the only person running Linux on a Lenovo Thinkpad T410 (Conexant CX20585) that has audio volume problems. How is it that I've owned this machine for 6+ months and have tried kernel upon kernel, scoured message boards, compiled alsa from trunk, tweaked endless permutations of snd-intel-hda kernel module options, yet still can't get a reasonable audio volume from this Intel HDA audio card on my T410!? It's unthinkable! I know, I know, the driver is still relatively "new", and the good work that has been done has been based on hacking some undocumented codecs without support from the manufacturers...but come on! Output volume is critical! You see, sound works just fine...but the output power is lacking. Something, somewhere, is preventing the output amplifier from being turned on or otherwise being configured correctly. I have to crank the alsa master and pcm output to 100% for music or movies to even be listenable, and even then it's not quiet loud enough. Thinkpads are still considered "good", Linux friendly laptops, so how is it that people are living with this problem? Certainly I can't be the only one! I've tinkered with hda-analyzer and hda-verb, and have some vague indication that an output amplifier is muted, but there's no apparent way to enable it. I guess its time to dig in and try and grok the source...but I kinda dread it. Don't get me wrong, I love hacking source and learning and trying things, but sometimes I just want stuff to work. I assumed 5 or 6 years ago that the "oooooh, audio on Linux is so hard" problem had been solved...but I guess I was wrong. After so many years of ALSA "just working" and doing what I needed, I guess it is actually a problem again...and that sucks. If you have any insights into this issue or want to steer me down a path, please send me an email...otherwise wish me luck.