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2016 Mar 15
3
Question on opus_decoder output sampling rate
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
2015 Jan 05
1
FEC monitoring
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
2016 Mar 15
0
Question on opus_decoder output sampling rate
...e 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 CPU intensive o...
2014 Oct 27
0
Codec setting using fmtp maxaveragebitrate and OPUS_SET_BITRATE
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
2015 Apr 02
2
Question on opus_decoder output sampling rate
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