opus - Oct 2014 - runtime reconfiguration of opusenc

Hello All,

This is my first appearance to the opus mailing list. My name is Michael 
Mehari and i am a researcher at Ghent University mainly on cognitive 
radio networking. Recently i am working on a demo scenario on self 
adjustable wireless audio conferencing and i selected opus for the audio 
encoding and decoding part. specifically the applications (i.e opusenc 
and opusdec) from opus-tools are what i was looking for.

However, i have one problem when using the opusenc application. The 
input parameters are start time configurable but i was looking for 
realtime reconfiguration for example changing bitrate on the fly. With 
in the context of my demo scenario, a wireless audio conferencing works 
in a dynamic environment and the audio quality at the reciever end is 
impacted by the interference level on top of the (opus) encoding 
degradation. I have read so far that opus can adjust its bitrates 
seamlessly (i.e depending on the audio input) but i don't know if this 
functionality is exposed to the user. For example, is it possible to put 
"opusenc" with in an optimization framework such that it accepts 
configuration inputs (i.e. bitrate and framesize) from the framework 
during runtime?

I hope i make myself clear and please let me know if there exists such a 
thing or direct me how i can do it myself.

Thank you,
Michael

-- 
Michael Mehari
Department of Information Technology
Internet Based Communication Networks and Services (IBCN)
Ghent University - iMinds
Gaston Crommenlaan 8 (Bus 201), B-9050 Gent, Belgium
Mob: +32 476 38 49 07
Tel: +32 (0)9 33 13800
Fax: +32 (0)9 33 14899
E-mail: michael.mehari at intec.UGent.be
Web: www.ibcn.intec.UGent.be

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On 2014-10-06 12:17 AM, Michael Mehari wrote:
> However, i have one problem when using the opusenc application. The
> input parameters are start time configurable but i was looking for
> realtime reconfiguration for example changing bitrate on the fly.
Hi Michael. You're correct that opusenc doesn't expose anything for
dynamic control of the encoding parameters. It's designed for encoding
static files or for http streaming.

The reference encoder in the libopus library does have this dynamic
switching ability. The framesize is controlled by how much audio data is
submitted to each call of opus_encode(). See


https://mf4.xiph.org/jenkins/view/opus/job/opus/ws/doc/html/group__opus__encoder.html#details

for an overview of the calling conventions.

Overall bitrate, redudancy, expected packet loss, and many other
parameters can be controlled dynamically through the opus_encoder_ctl()
interface. Just make calls to this in between submitting data for each
packet. A complete list of options is available in the API docs at


https://mf4.xiph.org/jenkins/view/opus/job/opus/ws/doc/html/group__opus__encoderctls.html
or
  http://opus-codec.org/docs/html_api-1.1.0/group__opus__encoderctls.html

You could modify opusenc.c to call these in response to commands read
from a socket, for example.

However, while opusenc is fine for internet radio applications,
conferencing usually uses the RTP protocol for its prioritization of
realtime interaction over complete delivery. I'm not aware of a
stand-alone tool for that with opus. You'd need to implement something
yourself here as well.

There's an experimental 'opusrtp' program in the opus-tools source
package you can look at, but it's only useful over an uncongested network.

For a sophisticated implementation of Opus in RTP see the webrtc.org
codebase. You can find the wrappers their engine calls to adjust encoder
parameters here:


https://code.google.com/p/webrtc/source/browse/trunk/webrtc/modules/audio_coding/codecs/opus/opus_interface.c

You can also look at open source voip clients like mumble:


https://github.com/mumble-voip/mumble/blob/master/src/mumble/AudioInput.cpp#L675

Hope that helps, and thanks for your interest in Opus.

 -r