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

From: lonce <lonce.wyse@zwhome.org>
Date: Fri, 14 Jun 2013 15:41:35 +0800
Message-ID: <51BAC92F.6020604@zwhome.org>
To: public-webrtc@w3.org

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 Friday, 14 June 2013 07:42:12 UTC

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