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

Hi,

     I would certainly be very grateful for any pointers to the right 
place or person to ask a few questions about the configurability of the  
audio streaming capabilities via the WebRTC API!

Thank you!
                         - lonce


On 2013-06-14 3:41 PM, lonce 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