<div dir="ltr"><div>Hi Dragos, thank you for your information, I noticed that with 44.1k greatest common factor is very high, 300 with 48k.<br><br></div><div><br>However do you think that we should expect that resampling would require more CPU then the Opus encoder for Silk for example ?<br><br>OR can we expect some alternate solution/improvement/tuning.<br><br></div><br></div><div class="gmail_extra"><br><div class="gmail_quote">On Tue, Mar 15, 2016 at 4:54 PM, Dragos Oancea <span dir="ltr"><<a href="mailto:droancea@yahoo.com" target="_blank">droancea@yahoo.com</a>></span> wrote:<br><blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex"><div><div style="color:#000;background-color:#fff;font-family:HelveticaNeue,Helvetica Neue,Helvetica,Arial,Lucida Grande,sans-serif;font-size:16px"><div dir="ltr">Hi Julien,<br><span></span></div><div><span><br></span></div><div><span>Quote from :<br></span></div><div dir="ltr"><a href="http://dspguru.com/dsp/faqs/multirate/resampling" target="_blank">http://dspguru.com/dsp/faqs/multirate/resampling</a><br><span></span></div><div dir="ltr"><br></div><div dir="ltr">"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!"</div><div><div><br></div><div>My guess is that Opus would perform similar to Speex if you'd have to have it resample to 44.1 khz.</div><div><br></div><div>Cheers,</div><div>Dragos<br></div><div dir="ltr"><br></div></div><div style="display:block"> <div style="font-family:HelveticaNeue,Helvetica Neue,Helvetica,Arial,Lucida Grande,sans-serif;font-size:16px"> <div style="font-family:HelveticaNeue,Helvetica Neue,Helvetica,Arial,Lucida Grande,sans-serif;font-size:16px"> <div dir="ltr"> <font face="Arial" size="2"> <hr size="1"> <b><span style="font-weight:bold">From:</span></b> Julien Chavanton <<a href="mailto:jchavanton@gmail.com" target="_blank">jchavanton@gmail.com</a>><br> <b><span style="font-weight:bold">To:</span></b> Jean-Marc Valin <<a href="mailto:jmvalin@jmvalin.ca" target="_blank">jmvalin@jmvalin.ca</a>> <br><b><span style="font-weight:bold">Cc:</span></b> <a href="mailto:opus@xiph.org" target="_blank">opus@xiph.org</a><br> <b><span style="font-weight:bold">Sent:</span></b> Tuesday, March 15, 2016 1:18 PM<span class=""><br> <b><span style="font-weight:bold">Subject:</span></b> Re: [opus] Question on opus_decoder output sampling rate<br> </span></font> </div><div><div class="h5"> <div><br><div><div dir="ltr"><div><div><div><div>Hi, another question on the same topic<br><br>Speex resampler at 44.1kHz seems to be very CPU intensive on Android (even more than the Opus encoder)<br><br></div>While Speex at 48kHz is just fine.<br><br></div></div>I wonder any alternate solutions or ideas ?<br>Improve it, look for alternate solution ...<br><br>I am guessing the NEON optimization are still used for both, etc. <br></div><br></div><div><br><div>On Thu, Apr 2, 2015 at 4:46 PM, Jean-Marc Valin <span dir="ltr"><<a rel="nofollow" href="mailto:jmvalin@jmvalin.ca" target="_blank">jmvalin@jmvalin.ca</a>></span> wrote:<br><blockquote style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">The encoder and decoder can handle, 8, 12, 16, 24 and 48 kHz<br>
input/output. If doesn't matter what it gets encoded to/decoded from.<br>
you can initialize a decoder at 8 kHz and it'll still decode 48 kHz<br>
audio fine (you just won't get the high frequencies obviously). For<br>
sampling rates other than 8/12/16/24/48, then you'll have to do<br>
resampling. Have a look at the speexdsp resampler if you don't already<br>
have one.<br>
<br>
Cheers,<br>
<br>
Jean-Marc<br>
<div><div><br>
On 02/04/15 10:42 AM, Julien Chavanton wrote:<br>
> Hi, is there any way to tell the decoder the output sampling Fz we want ?<br>
><br>
> opus_decoder_create = Sampling rate of input signal (Hz)<br>
><br>
> Considering this example (VoIP-out from WebRTC/RTP)<br>
><br>
> MICROPHONE(44.1/48kHz) >> [encoder created at 48kHz but with<br>
> internalSampleRate set to 8kHz]>> INTERNET >> [decoder(created with<br>
> 48kHz)] >> 48kHz(?) >> G.711(8kHz)<br>
><br>
> This leaves us with the only option to re-sample even if the internal<br>
> sample rate was set to 8kHz.<br>
><br>
> This may not seem like a big problem since we could simply resample but<br>
> on a server with a lot of load this could be significant ?<br>
><br>
> <a rel="nofollow" href="https://tools.ietf.org/html/draft-ietf-payload-rtp-opus-08" target="_blank">https://tools.ietf.org/html/draft-ietf-payload-rtp-opus-08</a><br>
> Is not very clear on these points.<br>
><br>
> Regards<br>
> Julien<br>
><br>
><br>
</div></div>> _______________________________________________<br>
> opus mailing list<br>
> <a rel="nofollow" href="mailto:opus@xiph.org" target="_blank">opus@xiph.org</a><br>
> <a rel="nofollow" href="http://lists.xiph.org/mailman/listinfo/opus" target="_blank">http://lists.xiph.org/mailman/listinfo/opus</a><br>
><br>
</blockquote></div><br></div></div><br>_______________________________________________<br>opus mailing list<br><a href="mailto:opus@xiph.org" target="_blank">opus@xiph.org</a><br><a href="http://lists.xiph.org/mailman/listinfo/opus" target="_blank">http://lists.xiph.org/mailman/listinfo/opus</a><br><br><br></div> </div></div></div> </div> </div></div></div></blockquote></div><br></div>