[Freeswitch-users] FS6947 - Tuning Opus bandwidth

Emrah lists at kavun.ch
Sat Oct 15 00:18:16 MSD 2016 

In my FS 1.7 I still see this:
2016-10-14 20:16:44.109370 [NOTICE] switch_loadable_module.c:213 Adding Codec opus 116 OPUS (STANDARD) 8000hz 40ms 2ch (VBR)
2016-10-14 20:16:44.109370 [NOTICE] switch_loadable_module.c:213 Adding Codec opus 116 OPUS (STANDARD) 8000hz 40ms 1ch (VBR)
2016-10-14 20:16:44.109370 [NOTICE] switch_loadable_module.c:213 Adding Codec opus 116 OPUS (STANDARD) 8000hz 120ms 1ch (VBR)
2016-10-14 20:16:44.109370 [NOTICE] switch_loadable_module.c:213 Adding Codec opus 116 OPUS (STANDARD) 8000hz 100ms 1ch (VBR)
2016-10-14 20:16:44.109370 [NOTICE] switch_loadable_module.c:213 Adding Codec opus 116 OPUS (STANDARD) 8000hz 80ms 1ch (VBR)
2016-10-14 20:16:44.109370 [NOTICE] switch_loadable_module.c:213 Adding Codec opus 116 OPUS (STANDARD) 8000hz 60ms 1ch (VBR)
2016-10-14 20:16:44.109370 [NOTICE] switch_loadable_module.c:213 Adding Codec opus 116 OPUS (STANDARD) 8000hz 20ms 2ch (VBR)
2016-10-14 20:16:44.109370 [NOTICE] switch_loadable_module.c:213 Adding Codec opus 116 OPUS (STANDARD) 8000hz 20ms 1ch (VBR)
2016-10-14 20:16:44.109370 [NOTICE] switch_loadable_module.c:213 Adding Codec opus 116 OPUS (STANDARD) 8000hz 10ms 2ch (VBR)
2016-10-14 20:16:44.109370 [NOTICE] switch_loadable_module.c:213 Adding Codec opus 116 OPUS (STANDARD) 8000hz 10ms 1ch (VBR)
2016-10-14 20:16:44.109370 [NOTICE] switch_loadable_module.c:213 Adding Codec opus 116 OPUS (STANDARD) 48000hz 40ms 2ch (VBR)
2016-10-14 20:16:44.109370 [NOTICE] switch_loadable_module.c:213 Adding Codec opus 116 OPUS (STANDARD) 48000hz 40ms 1ch (VBR)
2016-10-14 20:16:44.109370 [NOTICE] switch_loadable_module.c:213 Adding Codec opus 116 OPUS (STANDARD) 48000hz 20ms 2ch (VBR)
2016-10-14 20:16:44.109370 [NOTICE] switch_loadable_module.c:213 Adding Codec opus 116 OPUS (STANDARD) 48000hz 20ms 1ch (VBR)
2016-10-14 20:16:44.109370 [NOTICE] switch_loadable_module.c:213 Adding Codec opus 116 OPUS (STANDARD) 48000hz 10ms 2ch (VBR)
2016-10-14 20:16:44.109370 [NOTICE] switch_loadable_module.c:213 Adding Codec opus 116 OPUS (STANDARD) 48000hz 10ms 1ch (VBR)

> On Oct 14, 2016, at 10:16 PM, Emrah <lists at kavun.ch> wrote:
> 
> Since when? I don't see it in FS 1.6.10.
> 
> Emrah
>> On Oct 14, 2016, at 6:50 PM, Brian West <brian at freeswitch.org <mailto:brian at freeswitch.org>> wrote:
>> 
>> We do have 12khz and 24khz I added the specifically when we added SILK, we can resample to and from both rates. :)
>> 
>> On Fri, Oct 14, 2016 at 7:27 AM, Dragos Oancea <dragos.oancea at athonet.com <mailto:dragos.oancea at athonet.com>> wrote:
>> Hi
>> 
>> Why would you say opus at 8000@20i does not sound very good ? What do you
>> mean exactly ? Your call quality is bad or you mean you don't like the
>> fact that the sampling rate is low ?
>> 
>> You can play with the sampling rates by setting maxplaybackrate and
>> sprop_maxcapurerate and enabling asymmetric_samplerates (which is an
>> experimental feature so far) in your opus.conf.xml  .
>> 
>> FS does not have 12 khz or 24 khz because they are not much used for
>> Voip. As for opus @ 16 khz perhaps it will be added.
>> 
>> But if you use opus @ 48 khz and you just change maxaveragebitrate and
>> maxplaybackrate accordingly you should get only WIDEBAND from the
>> encoder anyway .
>> The decoder should decode at any sample rate.
>> We're working on a document (sort of manual) for the Opus module and
>> hopefully it will be released soon.
>> 
>> As for the issue why codec settings cannot be set from FS's dialplan , I
>> think its a missing feature that affects other audio codecs too and I
>> think it would be very useful.
>> 
>> We needed opus at 8000hz for transcoding and I tried to explain some things
>> here (see my comment at the bottom of the page ):
>> https://freeswitch.org/confluence/display/FREESWITCH/mod_opus <https://freeswitch.org/confluence/display/FREESWITCH/mod_opus>
>> 
>> 
>> Basically if you do heavy transcoding to PCMA / PCMU which is 8000 hz
>> you'll want to avoid resampling 48 khz <-> 8 khz  - we did tests and by
>> avoiding resampling we were saving 20-30 % CPU .
>> 
>> Regards,
>> Dragos
>> 
>> 
>> 
>> On 14/10/2016 14:02, Emrah wrote:
>> > Hi there,
>> > Revisiting this issue. I see that I can set my Opus codec with
>> > Opus at 8000@20i, but 8khz seems to be the only alternative profile I can
>> > use. I see only 2 extreme options when the module is loaded or unloaded
>> > that it's either 48khz, mono or stereo and packet size, or 8khz, mono or
>> > stereo and packet size. Can someone clarify why there is nothing in
>> > between? And what exactly this setting does? Opus at 8000h@20i definitely
>> > doesn't sound very good. I'd rather have a compromise for tough network
>> > conditions.
>> > Since these are parameters I can dynamically set on my dialplan, the
>> > question then becomes why can't I fully manipulate my Opus stack from
>> > the dialplan?
>> >
>> > Thanks!
>> >> On Jun 2, 2015, at 6:23 AM, Emrah <lists at kavun.ch <mailto:lists at kavun.ch>
>> >> <mailto:lists at kavun.ch <mailto:lists at kavun.ch>>> wrote:
>> >>
>> >> Hi there,
>> >> @Mike: yes, but in a commonsensical approach the Opus library on the
>> >> client's side would resample and therefore optimize the codec and the
>> >> bandwidth accordingly up to FS.
>> >> @Julien, I saw the setting for Opus globally, but it defeats the
>> >> purpose. I don't want to limit the bandwidth of Opus for all
>> >> instances. I'd like to optimize Opus on a per call basis.
>> >>
>> >> Thanks for the replies
>> >>> On May 31, 2015, at 2:13 PM, Michael Jerris <mike at jerris.com <mailto:mike at jerris.com>
>> >>> <mailto:mike at jerris.com <mailto:mike at jerris.com>>> wrote:
>> >>>
>> >>> Side note, opening at the different rate I believe just makes the
>> >>> opus library do the re sampling instead of FreeSWITCH.
>> >>>
>> >>> On Sunday, May 31, 2015, Julien Chavanton <jchavanton at gmail.com <mailto:jchavanton at gmail.com>
>> >>> <mailto:jchavanton at gmail.com <mailto:jchavanton at gmail.com>>> wrote:
>> >>>
>> >>>     Hi Emrah,
>> >>>
>> >>>     The settings exist but they are not available from the dialplan,
>> >>>     right now they can only be set globally .
>> >>>     https://freeswitch.org/confluence/display/FREESWITCH/mod_opus <https://freeswitch.org/confluence/display/FREESWITCH/mod_opus>
>> >>>
>> >>>     You can control the bandwidth using maxplaybackrate and
>> >>>     maxplaybackrate this will control the local encoder and also adds
>> >>>     the corresponding FMTP parameters to the SDP to be used by the
>> >>>     remote encoder (if it does implement the following draft, the
>> >>>     draft is evolving but I think it as not changed)
>> >>>
>> >>>     https://tools <https://tools/>
>> >>>     <https://tools/ <https://tools/>>.ietf.org/html/draft-ietf-payload-rtp-opus-11 <http://ietf.org/html/draft-ietf-payload-rtp-opus-11>
>> >>>
>> >>>     Maybe something like :
>> >>>
>> >>>     maxaveragebitrate 24000
>> >>>     maxplaybackrate 8000
>> >>>
>> >>>     The discussion was getting slightly more complicated when we
>> >>>     where discussing about unnecessary resampling this was not a
>> >>>     problem but it was just adding extra load on the server.
>> >>>
>> >>>     On Sun, May 31, 2015 at 6:09 AM, Emrah <lists at kavun.ch <mailto:lists at kavun.ch>
>> >>>     <javascript:_e(%7B%7D,'cvml','lists at kavun.ch <mailto:lists at kavun.ch>');>> wrote:
>> >>>
>> >>>         Hi list,
>> >>>
>> >>>         I re-read FS6947 and don't understand how this problematic
>> >>>         was addressed and the issue fixed.
>> >>>         The scope is simple. There should be a setting in the
>> >>>         dialplan that allows downsampling of Opus for applications
>> >>>         that do not require the 48khz / 2 channels framework. I.e.:
>> >>>         terminating to the PSTN with Opus to take advantage of low
>> >>>         bandwidth and great PLC.
>> >>>         There seems to be a lot of confusion around bandwidth in
>> >>>         general there. It doesn't matter if the internal clock of the
>> >>>         device is always sampling at 48khz / 2ch. There are settings
>> >>>         that can facilitate a lower bandwidth consumption for
>> >>>         particular use cases, and it seems the reason it is not being
>> >>>         implemented in FS is just a matter of being confused about
>> >>>         the intent of the 48khz 2ch base.
>> >>>         Please revisit this issue. FS should allow tuning of Opus
>> >>>         audio / network bandwidth in the dialplan. It would optimize
>> >>>         greatly lots of use cases.
>> >>>         If I'm calling the PSTN, I'd rather have my client downsample
>> >>>         and stream at a lower bandwidth, even if my audio capture
>> >>>         would still be at 48khz / 2ch as per the RFC, and save on
>> >>>         bandwidth, than transcode the full 48khz spectrum into PCM on
>> >>>         my FS and minimize processing power on the client's side.
>> >>>
>> >>>         Jira here: https://freeswitch.org/jira/browse/FS-6947 <https://freeswitch.org/jira/browse/FS-6947>
>> >>>
>> >>>         Emrah
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