onyx,
I would also check out the latency in the browser audio player. I somehow suspect that is where you are acquiring the largest portion of buffers and latency. I'm unsure, however, if you can even access and set the incoming buffer there. If not, you might have to write your own audio player...which would defeat your purpose of using the browser in a cell phone.
Flash does let you set the incoming audio buffer, ...but alas, many phones can't run flash.
WebRTC might be the best option on the browser side.
keep us posted!
suerte -august.
the audio side was just standard wireless hifi headphones, using lots of headphones but only two of the transmitters. The interesting part
technically
was the bike powered generators, but the 2 channel headphones dance floor
and
double DJ thing was lots of fun.
ah, nice. very clever! would love to check that out!
I may be making a bit of headway in looking at rtmp and hls streaming media servers like mist server and the rtmp module for nginx webserver. it accepts a raw connection to its input ports.
my question now becomes, which of the signal capable network objects can work without an associated in~ object? i have been toying with netsend~, udpsend~ mp3streamout~ streamout~ and mp3cast~, which works with the icecast server, but the others return errors so far. i have the greatest confidence in getting the netsend~ to eventually work, but does anyone have any experience with the others in regards to sending their outputs to non-pd or max based inputs like network ports set up to recieve other data?
Onyx
-- www.onyx-ashanti.com