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Re: [rtcweb] [rmcat] compressed codec-free webrtc?

"Mo Zanaty (mzanaty)" <mzanaty@cisco.com> Wed, 30 October 2013 05:16 UTC

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From: "Mo Zanaty (mzanaty)" <mzanaty@cisco.com>
To: Dave Taht <dave.taht@gmail.com>, "rmcat@ietf.org" <rmcat@ietf.org>, "rtcweb@ietf.org" <rtcweb@ietf.org>
Thread-Topic: [rmcat] compressed codec-free webrtc?
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Date: Wed, 30 Oct 2013 05:15:59 +0000
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Subject: Re: [rtcweb] [rmcat] compressed codec-free webrtc?
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Give app developers yuv, udp and native-speed js, and they won¹t need all
the webrtc plumbing at all. No chance of negotiation failure as icing. :)

Mo


On 10/29/13, 2:33 PM, Dave Taht <dave.taht@gmail.com> wrote:

In having my eyes glaze over at the codec debate I found myself
wondering to what extent anyone was pursuing truly low latency video
and audio, along the lines of what the lola project has been doing for
collaborative concerts.

See:

http://www.conts.it/artistica/lola-project

They ship raw audio and raw video, they actually use cameras where
they can get at scanlines and ship that (saving 16ms)

So I'm ignorant of what webrtc can do is there a codec selection (yuv?
48 bit audio? for the rawest video and audio possible?) All the extra
encoding steps we take today induce extra latency, and available
bandwidth continues to increase... and from a cc perspective if you
drop some bits from a scanline that's not a problem, but from audio
it's a huge deal....


-- 
Dave Täht

Fixing bufferbloat with cerowrt:
http://www.teklibre.com/cerowrt/subscribe.html

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