audio bridge sampling rate and quality

375 views
Skip to first unread message

Kaplan

unread,
Feb 28, 2017, 8:43:38 PM2/28/17
to meetecho-janus
Hi there,
I been using the audio bridge for some time on its default sampling rate of 16000.  I rtp_forward the audio to a janus streaming plugin endpoint (audio only), that is created like this:
var body = {
                        "request": "create",
                        "type": 'rtp',
                        "id": 1234,
                        "is_private": true,
                        "audio": true,
                        "video": false,
                        "audioport": 5001,
                        "audiopt": 96,
                        "audiortpmap": 'OPUS/48000/2'
                    }
Note the opus/48000/2.  
I've heard complains that the quality is not as good as my other system (flash speex codec, 22k).

What can I do to improve the audio quality?  Would setting the audio bridge to sample at 48k help? is the issue with the miss match of rates? i.e. 48k on the opus rtpmap vs 16k on the audio bridge?
Will lowering the audiortpmap to OPUS/16000/1 work ??

Most of the people connect to the streaming endpoint, only a few connect to the audio bridge (presenters).  So as far as bandwidth utilization, I think I am already using 48k (stereo) !

Ideas? thoughts?
any help appreciated!

Mirko Brankovic

unread,
Mar 1, 2017, 3:03:30 AM3/1/17
to meetecho-janus
well my 2¢,  if presenters are muxing to a 16k audio bridge you can forward it with what ever higher you want but you won't increase the quality of initial 16k.
I think you need to change it on audio room if you want to have better quality.

mirko

--
You received this message because you are subscribed to the Google Groups "meetecho-janus" group.
To unsubscribe from this group and stop receiving emails from it, send an email to meetecho-janus+unsubscribe@googlegroups.com.
For more options, visit https://groups.google.com/d/optout.



--
Regards,
Mirko

Lorenzo Miniero

unread,
Mar 1, 2017, 3:14:12 AM3/1/17
to meetecho-janus
Do NOT touch the opus/48000/2 part: that's standard, and the same whatever sampling rate you use due to the way Opus works.

Yes, if the mixer works at 16kHz, whatever you send you'll receive back 16kHz. The same applies to Streaming listeners too, as they get the audio from the mixer originally, which as Mirko explained means a source you can not "improve" again even if you re-encode. You can configure each room to mix at a different sampling rate: a higher rate will get you better quality, but also more CPU consumption. I personally feel 16kHz to be good enough (wideband) but I understand the need for something better. If a 22050hz room works good for them, you may want to try 24000 as a sampling rate.

Lorenzo

Kaplan

unread,
Mar 1, 2017, 8:35:38 AM3/1/17
to meetecho-janus
Yup, that is what is precisely what I assumed too :) I was wondering if it was worth touching the 48000/2 part, but Lorenzo explained what that is and to leave it alone.
Thanks to you both. I will experiment with 24000...
Thank you both...
To unsubscribe from this group and stop receiving emails from it, send an email to meetecho-janu...@googlegroups.com.

For more options, visit https://groups.google.com/d/optout.



--
Regards,
Mirko
Reply all
Reply to author
Forward
0 new messages