[linux-audio-dev] Re: (1) Jack -- busted? (2) jack.udp -- busted? (3) jack-osx -- binary-only?

lazzaro lazzaro at eecs.berkeley.edu
Sat Sep 24 16:58:50 UTC 2005


On Sep 24, 2005, at 9:02 AM, linux-audio-dev- 
request at music.columbia.edu wrote:
> Is anyone interested in collaborating on a common sample streaming
> protocol (possibly based on a somewhat simplified version of SDIF or
> the SC3 protocol)?


I'd recommend using RTP as your base protocol, and defining your
SPIF or SC3-like payload as an RTP payload format.  You'll pick up
the entire IETF multimedia protocols for free this way, including RTP  
MIDI:

http://www.cs.berkeley.edu/~lazzaro/rtpmidi/index.html

I think when it comes to networking, the writing is on the wall when
in comes to packet loss being a part of the environment you need
to live in.  Most new purchases of computers are for laptop computers,
most of those users want to use WiFi as their network, and the Internet
layer sees 1-2% packet loss on WiFi.  Also, we live in an era where
people want to run LAN apps on the WAN and WAN apps on the LAN,
and packet loss is also an unavoidable part of the WAN Internet  
experience.
Finally, modern applications want to use link-local Internet multicast.

RTP was built for letting payload formats handle packet losses in
a way that makes sense for the media type -- RTP MIDI is an extreme
example of this, but the audio and video payload formats are loss
tolerant in more subtle ways.  RTP is also multi-cast compatible.

Finally, with RTP there's a standardized way to use RTSP and SIP
for your session management if you wish, or if you prefer, you can
just build RTP into whatever session manager you have committed
to (like jack).

---
John Lazzaro
http://www.cs.berkeley.edu/~lazzaro
lazzaro [at] cs [dot] berkeley [dot] edu
---





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