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Re: [rtcweb] compressed codec-free webrtc?

Christer Holmberg <christer.holmberg@ericsson.com> Tue, 29 October 2013 18:48 UTC

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From: Christer Holmberg <christer.holmberg@ericsson.com>
To: Dave Taht <dave.taht@gmail.com>, "rmcat@ietf.org" <rmcat@ietf.org>, "rtcweb@ietf.org" <rtcweb@ietf.org>
Thread-Topic: [rtcweb] compressed codec-free webrtc?
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Date: Tue, 29 Oct 2013 18:47:29 +0000
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Subject: Re: [rtcweb] compressed codec-free webrtc?
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Hi Dave,

As far as the standards is concerned, you are allowed to implement and use whatever codec you want.

The discussion is about whether there should be a codec that everyone MUST implement.

Regards,

Christer

-----Alkuperäinen viesti-----
Lähettäjä: rtcweb-bounces@ietf.org [mailto:rtcweb-bounces@ietf.org] Puolesta Dave Taht
Lähetetty: 29. lokakuuta 2013 20:33
Vastaanottaja: rmcat@ietf.org; rtcweb@ietf.org
Aihe: [rtcweb] compressed codec-free webrtc?

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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