- From: Jonathan Raoult <jesuisjonathan@gmx.fr>
- Date: Thu, 14 May 2015 11:41:51 +1000
- To: public-media-capture@w3.org
Sorry guys I just jumped in this thread.
I'm very interested in this discussion specially on the low latency side. I recently hit the "optimum buffer size for everyone" wall with getUserMedia and I would need something to adjust latency on capable platform at least.
What I noticed in music creation softwares (and other audio API as well) is the use of frame count as input to adjust latency. Then the result in ms calculated but only for display purposes. It would fit well with sampleRate and sampleSize from MediaTrackSettings which are already low level enough for the user to infer the latency in ms. It also have the advantage of being precise, there not rounding or calculation to make for the implementation.
So to come back to the example, something like that is another solution:
{ sampleCount: { max: 20 } }
Jonathan
Received on Friday, 15 May 2015 10:16:58 UTC