W3C home > Mailing lists > Public > public-webrtc@w3.org > July 2013

Re: Audio and Opus configurability for speed and quality (at the expense of bandwidth)

From: Cullen Jennings (fluffy) <fluffy@cisco.com>
Date: Wed, 24 Jul 2013 22:18:36 +0000
To: lonce <lonce.wyse@zwhome.org>
CC: "public-webrtc@w3.org" <public-webrtc@w3.org>, Jean-Marc Valin <jmvalin@mozilla.com>
Message-ID: <C5E08FE080ACFD4DAE31E4BDBF944EB11360DBE2@xmb-aln-x02.cisco.com>

I suspect Jean-Marc (Cc'd) might be best person to answer this 

On Jun 14, 2013, at 1:41 AM, lonce <lonce.wyse@zwhome.org> wrote:

> 
> Hello -
> 
>     I have a couple of questions I have not been able to answer myself after looking over published docs. I am interested in maximum speed and uncompromised quality transmission (for musical purposes), which leads to these questions:
> 
> 1) What exactly is the strategy of the "components to conceal packet loss".  Is there a strategy specifically for audio packet loss?
> 
> 2) Can the audio echo cancellation (AEC), automatic gain control (AGC), and noise reduction, be turned off (not used)?
> 
> 3) Can compression by turned off completely (to avoid the algorithmic delay of coding/endcoding)?
> 
> 4)  If you cannot bypass the compression algorithm, what is the minimum delay one can achieve?  It appears to me (from http://www.webrtc.org/reference/architecture and http://en.wikipedia.org/wiki/Opus_%28codec%29 ) that analysis frame sizes down to 2.5ms (CELT layer) and 10ms (SILK layer) are possible. This, in addition to "look ahead"  and algorithm delay puts the minimum delay at at least 20 ms, right?
> 
> 5) Does one have control over how many analysis frames are sent per packet (could I set it to 1)? 
> 
> Musicians have been using a system called JackTrip (CCRMA, Stanford University) which suuports uncompressed transmission, and sub-millisecond frames (and packet) size. To recover from UDP losses, it sends redundant streams, and the receiver takes the first packet that arrives with the time stamp it needs next to reconstruct the audio on the receiver. My questions above are all about how close WebRTC can come to achieving the same performance. 
> 
> Thanks!
>                  - lonce
Received on Wednesday, 24 July 2013 22:19:22 UTC

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