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...< channels; n++) {? ? if (frame_data[0][0] & (0x80 >> ((n + 1) * (frames + 1) - 1)))? ? ? return 1;? } Is this the way to detect is an Opus-encoded packet has FEC info about the previous packet ? Does Opus support FEC n+2 too, like SILK ? Thank you for this great codec! Regards,Dragos Oancea -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.xiph.org/pipermail/opus/attachments/20150205/d5b35f50/attachment.htm

Hi Dragos, The issue is that not all packet loss maybe congestion related. Often, reducing bitrate seems to have no impact on improving packet loss. Thanks, Manpreet. On Tue, Dec 16, 2014 at 2:09 AM, Dragos Oancea <droancea at yahoo.com> wrote: > > Hi > > You can start a VOIP call with 50 kbps bitrate and reduce the bitrate if > there is packet loss. You know if there's packet loss if you receive RTCP > . > Linphone does this . > > Regards, > Dragos Oancea > >...

...if connection gets too lossy, how will opus adapt to it? Can it automatically shift bitrate down to minimize impact? Mark from IRC suggests that the app has to be aware of the losses and change it on the fly. Has anybody on the list tried this? Kelvin Chua On Wed, Mar 4, 2015 at 5:53 PM, Dragos Oancea <droancea at yahoo.com> wrote: > Hi Kelvin, > > You can use something like : > opus_encoder_ctl(enc,OPUS_SET_BITRATE(bitrate)); > opus_encoder_ctl(enc,OPUS_SET_MAX_BANDWIDTH(bandpass)); > bandpass is the audio bandpass , eg: OPUS_BANDWIDTH_WIDEBAND . > > You will need...

Hi, Although this maybe considered out of scope here, but I'll ask anyway. Opus has remarkable flexibility for changing encoder bitrate during a call. Are there any suggestions about how bandwidth/capacity between the two endpoints can be measured/estimated during a call so that the outgoing bitrate can be adjusted accordingly? Thanks, Manpreet. -------------- next part -------------- An

...more loss there is, the lower the threshold for enabling FEC. Overall, the encoder attempt to make the best decision on a packet-per-packet basis as to whether FEC is useful considering the bit-rate, percentage of loss, and the exact audio content. Cheers, Jean-Marc On 05/02/15 11:46 AM, Dragos Oancea wrote: > Hello, > > Is FEC supposed to work in NARROWBAND mode (with > maxaveragebitrate=12000; maxplaybackrate=8000 ) ? > I am having some confusing results, it appears that FEC is enabled in > the encoder, but the decoder cannot find any packet with FEC. > > I am also w...

...he 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 will be setting those parameters durin...

Hi, another question on the same topic Speex resampler at 44.1kHz seems to be very CPU intensive on Android (even more than the Opus encoder) While Speex at 48kHz is just fine. I wonder any alternate solutions or ideas ? Improve it, look for alternate solution ... I am guessing the NEON optimization are still used for both, etc. On Thu, Apr 2, 2015 at 4:46 PM, Jean-Marc Valin <jmvalin at

...using complexity 10 .What is the impact of this setting on the decoder, when for example a device is encoding with complexity level 0 , but it's receiving a stream encoded with level 10 ??? We plan to use Opus with OPUS_BANDWIDTH_NARROWBAND or OPUS_BANDWIDTH_WIDEBAND . Thank you. Regards,Dragos Oancea -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.xiph.org/pipermail/opus/attachments/20141117/ff79a11b/attachment.htm

Hi You can start a VOIP call with 50 kbps bitrate and reduce the bitrate if there is packet loss. ?You know if there's packet loss if you receive RTCP .?Linphone does this . Regards,Dragos Oancea From: Manpreet Singh <manpreets7 at gmail.com> To: opus at xiph.org Sent: Tuesday, December 16, 2014 7:54 AM Subject: [opus] Estimating bitrate during a real-time voip call Hi, Although this maybe considered out of scope here, but I'll ask anyway. Opus has remarkable flexibi...

I am using libopus for my implementation. I wonder if anybody in the list have any experience on how to make libopus dynamically adjust its bitrate? On Mar 3, 2015 10:42 PM, "Benjamin Schwartz" <benjamin.m.schwartz at gmail.com> wrote: > It sounds like your software isn't adjusting the opus bitrate in response > to network conditions. For example, many WebRTC

Hi Kelvin, You can use something like :opus_encoder_ctl(enc,OPUS_SET_BITRATE(bitrate));opus_encoder_ctl(enc,OPUS_SET_MAX_BANDWIDTH(bandpass)); bandpass is the audio bandpass?, eg: OPUS_BANDWIDTH_WIDEBAND . You will need to calculate the 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

Hi Julien, Quote from : http://dspguru.com/dsp/faqs/multirate/resampling "The problem is that for resampling factors close to 1.0, the interpolation factor can be quite large. For example, in the case described above of changing from the sampling rate from 48 kHz to 44.1 kHz, the ratio is only 0.91875, yet the interpolation factor is 147!" My guess is that Opus would perform similar to