similar to: Re:

It might be good to use the (uncompressed) samples on the opus page, as a common starting point? http://www.opus-codec.org/examples/ On Dec 21, 2013, at 9:43 AMEST, Stuart Marsden wrote: > I have run a few more test at different bitrates and 1.1 is looking even worse in terms of speed compared to previous versions. > > I have shared a google sheet which has the raw data and charts for

Hi guys, as MP3 and Opus have very similar objectives, I think the original poster's question was a valid one: Why does Opus have more artefacts in the lower frequency ranges than MP3 has? The spontaneous suspect that lower frequency artefacts may be more noticeably than higher frequency artefacts seems plausible, also. Is it a matter of energy (which is higher for higher frequencies)? When

If I understand correctly, Ogg Vorbis is a lossless format. But then why does it support bitrates? Ogg Vorbis does not delete any information. When I encode a 128 kbit MP3 file to 32 kbit OggVorbis, the resulting file has a bitrate of about 110 kbit. And when I encode it to 320 kbit, the file will be bigger, but the quality won't improve because of the quality of the original. So what purpose

I have to admit that I am impressed by your results -- making 1.1 look slower than 1.0 is by no means an easy task. On the other hand, it's a great tutorial on how not to use Opus, so for the benefit of everyone, this is a summary of what we learned in this exercise: 1) When running on ARM, the fixed-point build is usually faster than floating point. This is true on the majority of ARM archs

Cliff, Yes it would be good, but very hard to get a figure for the quality. At 6kbps I assume it does not bother trying to figure what mode to use as at that rate it can only use SILK. When I run some other bitrates it may get a bit slower trying to decide whether it is voice or music. I started with low bit rate because I am only really interested in Voice and very low bit rate. I think there

Hi, I had R working since version 1.4. Then I bought a new HD and installed a RH 7.3 on it and since then I can no longer install any R package. Here is the failure message I obtain: ... g77 -fPIC -O2 -m486 -fno-strength-reduce -g -c sortm.f -o sortm.o gcc -shared -o fields.so css.o csstr.o cvrcss.o cvrf.o dchold.o dcopy.o ddot.o dlv.o

Just a word to say thanks for your help. Yes, in order to install R packages I needed ncurses-devel which was not installed. Many thanks Rachid -- Dr. Rachid Cheddadi Centre universitaire Arles Tel: 00.33.(0)4.90.96.18.18 European Pollen Database Fax: 00.33.(0)4.90.93.98.03 CNRS - UMR 6116 rachid.cheddadi at wanadoo.fr 13200 Arles - France rachid.cheddadi at

I only have mp3s and it doesn't make sense to me to change all of them to ogg. My MDA(windows mobile phone) plays mp3s with no problem. It uses them as ringtones. Its more widely used in my circles. Why would I and everyone else want to switch to ogg? What is the benefit? Why the shun on mp3? And I can stream at 64 and 32 which is great for those streams that will get a lot of

Hi, I have a 4 column file (long/lat/elev/variable) and I tried to fit the values of my variable to the XYZ space using Tps and I keep getting the following message: Warning messages: 1: GCV search gives a minumum at the endpoints of the grid search in: Krig.find.gcvmin(info, lambda.grid, gcv.grid$GCV, Krig.fgcv, 2: GCV search gives a minumum at the endpoints of the grid search in:

Hi Rob The stream format has nothing to do with your audio files. I probably have 70% MP3's and 30% Ogg Vorbis files, and 1 or 2 WAV files for jingles on my hard drive. I only encode new CD's to Ogg Vorbis. But my station is streaming in AAC+ format, and I have also streamed in Ogg Vorbis format simultaneously. There is no way I would bother streaming in MP3 format. It's 15+

I can understand wanting to boost the adoption of Xiph codecs, but I think restricting the directory to vorbis streams only will not help this endeavour. I can imagine it will just piss a lot of people, including myself, and give Xiph a bad name. I get very few listeners of my AAC+ stream now since this occured. The restriction is not an insentive to switch to Ogg Vorbis at all. The incentive

Dear Vorbis Devteam, My name is Michael Feilen and I've been working on Digital Radio Mondiale (DRM) transmitters and receivers for quite a while now. DRM uses HE-AACv2 by Dolby to encode the audio content (see http://www.drm.org/uploads/media/es_201980v030101p.pdf - pages 23 ff). As I think Vorbis is an excellent alternative, I'd like to implement and define an interface for Vorbis

hy all, i'm trying to make an acl so a local unix user 'sie' can access exalead mboxes. my exalead mboxes are stored in : /opt/exalead/mail/sie/ mailboxes are automatically created every week like sie.2007.W17 for the 17th week of the year. how can i do this ? i've tried several things but none succedded. dovecot version 1.0.rc15 # /etc/dovecot.conf ddIEffective uid=65534,

Alfred E. Heggestad wrote: > <...> > > If we don't get any comments in 1 week (by 22. May 2007) we will go ahead > and submit it to the IETF. Of course you can comment on it also after it > has been submitted, but we would like to get the input from the Speex > community first.. > thanks for all the input. please find attached an updated version of the draft. I

Hello, HawkVoice doesn't have a 6.3kbps codec for CELP, it has a 4.5kbps CELP codec and I do not believe it is being used by TeamSpeak. The 6.4kbps CELP being used in TeamSpeak, to which you are referring I believe comes from Lernout & Hauspie's LHACM.ACM file which it appears you are redistributing (I assume TeamSpeak has a license and permission to do this). The only people I

Hi Please find an updated version of the Speex I-D attached. The only change is addition of the copyright conditions in Appendix A, as requested by Ivo. Many thanks for your input. I will give you a few more days before submitting to AVT working group /alfred Ivo Emanuel Gon?alves wrote: > Do not forget to add the "Copying conditions" to the RFC. > > Check

Looks good to me. Jean-Marc Alfred E. Heggestad a ?crit : > Hi > > Please find an updated version of the Speex I-D attached. The only > change is addition of the copyright conditions in Appendix A, > as requested by Ivo. > > Many thanks for your input. > > I will give you a few more days before submitting to AVT working group > > > /alfred > > Ivo

Hi! Excuse the delay, but I had to deal with a corrupted NTFS file system that ate many important files on an USB stick... The FLAC version of the original is almost 6MB and it can be downloaded slowly from this time-limited link: https://sbr5vjid0jgmce4q.myfritz.net:40262/nas/filelink.lua?id=0ba5a10529a6fe7b On the meaning of a logarithmic sweep: If you use foobar2000 and the

Q: Is there any testing against a collection of known "hard-to-encode" clips before new releases? It would be an obvious thing, if you want to be serious about quality. I brought this up because I tried latest cvs version of oggenc on one of these standard clips I have. It's a 6 sec long clip of an applause. Heavy noise is easy to hear with qualities 0 to 5,99. (This corresponds

Hello, I have a queue with only one element and one agent member. I want that my call leave the queue after 30s. My problem is that my call stays 60s in the queue and my agent is called 2 times. Can you say me how can i do it please?? -------------------------------- [queue] music=default strategy=ringall timeout=30 maxlen=1 context=mbdsys announce-frequency=0 announce-holdtime=no