Hi! On Thu, Feb 25, 2016 at 09:37:19AM +0530, Arun Raghavan wrote: > > > > > from v1: > > > Patch 1-3: Update code for the changed API and use some new features > > After updating the webrtc AudioProcessing code (http://arunraghavan.net > /2015/10/psa-breaking-webrtc-audio-processing-api/), this makes sure > we're using the API correctly, and then exposes a couple of new > features (an extended filter where you are willing to sacrifice some > latency/processing for accuracy, and an intelligibility enhancer which > is disabled till we can fix our infrastructure to support it). > Suddenly the world makes a whole lotta sense after reading the blog article and description above ;-) ... > > > Patch 21-24: Add beamforing support > > This adds the ability to use webrtc's beamformer, which can take > multichannel capture (and a description of the mic geometry), and use > that to "focus" the mic in a given direction, reducing (in software) > noise from either side. > > For example, my laptop has two mics to either side of the webcam, and > this could be useful there (for this to actually work generically, we > would need to expose a way for users to tell us what their mic geometry > looks like for each device -- which will hopefully one day be possible > via our configuration API). > Thanks a lot for describing 'beamforming' and its relation to pulse. Didn't know any of that before, too :-) > I hope this is helpful. Please do ask questions if something is unclear > or needs more explanation (I'll try to write up a more detailed summary > for general consumption as well). > Very very helpful. Yup, PulseAudio is quite big indeed and it probably needs a guide, something like the FreeBSD handbook, [*] on its own! [*] https://www.freebsd.org/doc/handbook/ Thanks again, -- Darwish http://darwish.chasingpointers.com
