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...size of the IP+UDP+RTP headers which is > 40 bytes ) . > By default the audio bandwidth (bandpass) setting is > OPUS_BANDWIDTH_FULLBAND , which will utilize more network bandwidth . > > > Regards, > Dragos > > ------------------------------ > *From:* Kelvin Chua <kelchy at gmail.com> > *To:* Benjamin Schwartz <benjamin.m.schwartz at gmail.com> > *Cc:* opus at xiph.org > *Sent:* Wednesday, March 4, 2015 2:27 AM > *Subject:* Re: [opus] adaptive bandwidth > > I am using libopus for my implementation. I wonder if anybody in the list > hav...

...conditions. For example, many WebRTC implementations do not > adjust the opus bitrate, because it is small in comparison to the video > bitrate. However, opus itself does support continuously varying the > bitrate over a wide range. > > On Tue, Mar 3, 2015, 12:58 AM Kelvin Chua <kelchy at gmail.com> wrote: > >> Hi guys, >> >> I have been reading a lot about the "adaptiveness" of opus and i quote: >> >> ... can still change, e.g. to adapt to changing network conditions. >> useinbandfec ... >> >> can somebody please en...

...a Voip call ) , but you can change the bitrate anytime.For example if you can read incoming RTCP packets , you can check if there's reported packet loss , and then lower the bitrate accordingly.Yes, the app has to be aware of the packetloss ?percentage. Cheers,Dragos From: Kelvin Chua <kelchy at gmail.com> To: Dragos Oancea <droancea at yahoo.com> Cc: Benjamin Schwartz <benjamin.m.schwartz at gmail.com>; "opus at xiph.org" <opus at xiph.org> Sent: Wednesday, March 4, 2015 11:02 AM Subject: Re: [opus] adaptive bandwidth Thanks Dragos, I assume I wi...

...he codec bitrate from the available network bitrate (by taking into account the size of the IP+UDP+RTP headers which is 40 bytes ) . By default the audio bandwidth (bandpass) setting is OPUS_BANDWIDTH_FULLBAND , which will utilize more network bandwidth . Regards,Dragos From: Kelvin Chua <kelchy at gmail.com> To: Benjamin Schwartz <benjamin.m.schwartz at gmail.com> Cc: opus at xiph.org Sent: Wednesday, March 4, 2015 2:27 AM Subject: Re: [opus] adaptive bandwidth I am using libopus for my implementation. I wonder if anybody in the list have any experience on how to make li...

given opus as a variable bitrate codec applied to voip rtp, i can verify that the bitrate really changes by a few kbps between max and min. as i understood, the bitrate variation is dependent on the audio source. are there any other factors which would affect this varying bitrate? like for example: packet losses, jitter, latency, etc. Will it automatically shift to lower bitrate / sampling rate

Hi guys, I have been reading a lot about the "adaptiveness" of opus and i quote: ... can still change, e.g. to adapt to changing network conditions. useinbandfec ... can somebody please enlighten me on this "adaptiveness"? whatever way I do our tests, it sticks to the same sampling rate and the same average bitrate, it would go up, down a bit but that's it. When we get

having a hard time communicating on IRC, thank you gmaxwell, very informative. anyway, we were discussing the proper implementation of FEC on the decoder side. well, encoder side is just a boolean thing so that's alright. i gave an example where the receiver lost 5 rtp packets, 1 2 3 4 and 5 during which, we call opus_decode with a null pointer and fec=0 for every packet lost. now, when it