Comment 3 for bug 483812

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vocx (eliudcabrera) wrote :

Kurt,

It is only a regression if pulseaudio worked before but it doesn't work now. As per the comment #1, it may seem that this feedback capability is not in pulseaudio at all. To anyone reading this, remember that in previous versions of Ubuntu the default audio server was Alsa, and when it changed to pulseaudio it broke many things for many users (but not for me, if I must say).

But, well, I wouldn't know exactly, since I don't even have a microphone and only a few times I've tried to use one. I found this bug report accidentally while trying to solve a different problem, but I provide a workaround using the Jack audio server that may work.

1. Install "qjackctl", a Qt frontend for the Jack Audio Connection Kit.
2. It will pull some dependencies, specially the "jackd" package which is the Jack server/daemon proper.
3. Open "qjackctl" and start the server. Hopefully it will work with the default values, otherwise you may need to click "Setup" and adjust the settings.
4. There is some documentation for Ubuntu: https://help.ubuntu.com/community/HowToJACKConfiguration https://help.ubuntu.com/community/HowToQjackCtlConnections But perhaps a better documentation is found in the Jack Wiki http://trac.jackaudio.org/wiki
5. Essentially, adjust the controls (specially frame) to reduce the latency as much as possible, for instance from 50 ms to 11 ms. Also check that there are no XRUNs in the "Messages" and the "Status" windows in the main interface.
6. If the server seems to run okay, click "Connect".
7. In the "Audio" tab click on "system" in the "Readable Clients / Output Ports" and on "system" in the "Writeable Clients / Input Ports" windows. These represent the "sources" and "sinks" of audio.
8. Click "Connect". A line should be drawn indicating that the sound from the "capture" devices (microphone) is routed to the "playback" devices (speakers).
9. Every program that uses the Jack server will place different output and input ports and these may be wired as desired. For example, if Skype supports the Jack sever, it would be possible to use "capture" -> "Skype" and "capture" -> "playback", or perhaps "capture" -> "Skype" -> "playback", to send the sound to both Skype and the speakers.

In summary, this method worked for me but it may not be ideal for every case or may not work at all. Also, I think that if Jack is running and you have outputs routed to the playback devices, other audio programs may not work at all because the playback resources are already used. This happened for me with Audacious.