search for: chavanton

...#39;t get the high frequencies obviously). For > sampling rates other than 8/12/16/24/48, then you'll have to do > resampling. Have a look at the speexdsp resampler if you don't already > have one. > > Cheers, > > Jean-Marc > > On 02/04/15 10:42 AM, Julien Chavanton wrote: > > Hi, is there any way to tell the decoder the output sampling Fz we want ? > > > > opus_decoder_create = Sampling rate of input signal (Hz) > > > > Considering this example (VoIP-out from WebRTC/RTP) > > > > MICROPHONE(44.1/48kHz) >> [encode...

...le, 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 Speex if you'd have to have it resample to 44.1 khz. Cheers,Dragos From: Julien Chavanton <jchavanton at gmail.com> To: Jean-Marc Valin <jmvalin at jmvalin.ca> Cc: opus at xiph.org Sent: Tuesday, March 15, 2016 1:18 PM Subject: Re: [opus] Question on opus_decoder output sampling rate Hi, another question on the same topic Speex resampler at 44.1kHz seems to be very...

Hi, is there any way to tell the decoder the output sampling Fz we want ? opus_decoder_create = Sampling rate of input signal (Hz) Considering this example (VoIP-out from WebRTC/RTP) MICROPHONE(44.1/48kHz) >> [encoder created at 48kHz but with internalSampleRate set to 8kHz]>> INTERNET >> [decoder(created with 48kHz)] >> 48kHz(?) >> G.711(8kHz) This leaves us with

I have a concern with Dial command, I want to enable a secondary route with a remote partner, if the first route fails then we use the second one : Solution1: it will try both (there will be 2 simultanious actives calls ringing) this is not clean when calling an endusers exten => _X.,1,Dial(SIP/${EXTEN}@remote-sip1,5 <mailto:SIP/${EXTEN}@remote-sip1,5> ) exten =>

I have to register asterisk/sip with a sipproxy that does not support authentication, I do not know how to tell Asterisk not to send authentication request? # sip.conf [general] insecure=very permit=207.148.115.10/255.255.255.0 [myproxy] type=friend host=217.118.115.10 context=from-sip # Logging: <--- Reliably Transmitting (NAT) to 207.148.115.10:5060 ---> SIP/2.0 407 Proxy

I was looking for a (http socket module / mysql module) not using AGI(perl/php/shell) for asterisk in order to do intensive database / web server interactions as needed without performance to much overhead. Is there a real benefit in using : "Asterisk cmd MYSQL" app_addon_sql_mysql , I see it require perl DBD so it is not using Mysql C API, then there must be an instance of the perl

Hi, I would like to monitor FEC usage in order to include it in RTCP EX or calculate MOS estimation, etc. However the Opus codec library does not seem to expose such information. "Was LBRR found and used or was it PLC ?" I saw in WebRTC that they are using a technique to parse the "frame header" WebRtcOpus_PacketHasFec() It this something that is supported ? What would you

...;ll still decode 48 kHz audio fine (you just won't get the high frequencies obviously). For sampling rates other than 8/12/16/24/48, then you'll have to do resampling. Have a look at the speexdsp resampler if you don't already have one. Cheers, Jean-Marc On 02/04/15 10:42 AM, Julien Chavanton wrote: > Hi, is there any way to tell the decoder the output sampling Fz we want ? > > opus_decoder_create = Sampling rate of input signal (Hz) > > Considering this example (VoIP-out from WebRTC/RTP) > > MICROPHONE(44.1/48kHz) >> [encoder created at 48kHz but with >...

I have an installation 100% dialplan and mysql-addons. I find out that mysql-addons is working great, but I suspect there may be a memory leak involved. Anyone else facing a memory leak recently ? Regards -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20100608/d4ae4e6e/attachment.htm

Hi Folks, thanks for the great work, not sure if this is the right list for this type of quesiton. We are looking to use only Opus as "one codec for all", with VoIP-out obviously we want to tune it. I am planning to use fmtp in SDP to control server/client Opus settings. Something like : - *maxplaybackrate*: a hint about the maximum output sampling rate that the receiver is