similar to: regarding CELP/ACELP/others patentes

And the main point of interest is the following: how much difference in SPEEX terms would it be if it were developed with ACELP in mind? (not that i would do an aSpeex codec implementing ACELP, just to know and point my efforts) -- John Villar Gerente de Proyectos Computadores Flor Hard Soft 2058 C.A. www.florhard.com -------------- next part -------------- A non-text attachment was

Hello Jean-Marc: I've been watching the speex development from its inception because I and several Tech Startup Connection members have a very important application for this voice encoding/decoding. Further, we are quite familiar with ACELP as implemented in G.729. As far as I know Speex is also ACELP ... yes/no? Question ... what do you see as the advantages of ACELP compared to MP3? I

Hi, We would like to announce the first release of the Speex project. Speex (http://speex.sourceforge.net) is an open-source (LGPL), patent-free compression format allowing an alternative to expensive proprietary codecs. Unlike Ogg Vorbis which compresses general audio, Speex is designed especially for speech. For that reason, Speex is meant to be a complement to Vorbis. Since it is specialized

> I've been watching the speex development from its inception because I > and several Tech Startup Connection members have a very important > application for this voice encoding/decoding. Further, we are quite > familiar with ACELP as implemented in G.729. As far as I know Speex > is also ACELP ... yes/no? No. Speex is CELP (Code-excited linear prediction), but not ACELP

> -----Original Message----- > From: Jean-Marc Valin [mailto:jean-marc.valin at usherbrooke.ca] > Hi > > On 10-06-21 07:24 PM, Pavel Pavlov wrote: > > I'm just curious, who and why came up with that strange way to write includes: > > I did. Because I think that's the way to go. I'm just curios, why? Is there some compiler/platform that requires it to be

Dear list, I need to convert this data.frame > names(codesM) [1] "key" "AMR.pa1.M" "AMR.pa2.M" "AMR.pa3.M" "AMR.pa4.M" [6] "AMR.pa5.M" "AMR.pa6.M" "AMR.pa7.M" "AMR.pa8.M" "AMR.pa9.M" [11] "AMR.pa10.M" "AMR.ta1.M" "AMR.ta2.M" "AMR.ta3.M"

I was wondering how vorbis fares as a speech codec? I mean can it give similar or better quality as GSM/Toast at the same kind of bitrates/filesize? I'm not worried about streaming but I'd love to be able to create really small voice files. Can anyone give me an idea of filesize for recording time for pure speech based use? love Freya __________________________________________________

Hi all, I have a project for the 3G related, AMR and AMR-WB support. I'm using the client develop suite from the PortSIP(http://www.portsip.com), as their said support the AMR, AMR-WB with RFC4867. Now I have to setup a SIP server/SIP PBX in our Lab for test, does the Asterisk support these codecs and RFC4867 ? If no, there has any plugin to support this ? Also, any other Server/PBX which

Hi, I'm having problems with an asterisk server that's not offering Codecs for ulaw and alaw as it should. I've three servers in total: a1, a2 and "b" A1 and A2 have pretty much the same config files, except IP address info changes Server B is configured to accept all inbound invites. Calls from A1 to B, all work fine, and in a sip debug session I can see

Hi all, supposing to have speex working at 3.95, when VAD/CNG is employed, which is the size of those "silence" samples respect to voice samples? I am wondering if it could be possible to lower even the transmission of VAD/CNG samples by specifying "outbound" in the communication channel only 1 bit instead of sending the whole VAD/CNG packet. I am doing the same for AMR

Hello, I'm trying to upgrade from 5-stable to 6-stable; after rebooting to the new kernel, the amr(4) module (amr.ko) refuses to load with this message: link_elf: symbol cam_simq_alloc undefined However, a quick nm(1) on /boot/kernel/cam.ko does show: ... 0000f0ac T cam_sim_set_path 0000efc4 T cam_simq_alloc 0000efd4 T cam_simq_free ... and cam.ko is already loaded before amr.ko.

In order to develop a Voip application, today i should make it robust to bit-errors over wireless networks. This is an actually lack of Speex, infact what i've understood is that if a packet arrives corrupted, i must pass NULL to the decoder in order to let it know. My target is to use UDP (with checksum field disabled) and exploit also corrupted packets giving them to the AMR codec. Otherwise

Hi, R-users, data is a matrix like this AMR BS GE HR MO UK SP500 1974 -0.3505 -0.1154 -0.4246 -0.2107 -0.0758 0.2331 -0.2647 1975 0.7083 0.2472 0.3719 0.2227 0.0213 0.3569 0.3720 1976 0.7329 0.3665 0.2550 0.5815 0.1276 0.0781 0.2384 1977 -0.2034 -0.4271 -0.0490 -0.0938 0.0712 -0.2721 -0.0718 1978 0.1663 -0.0452 -0.0573 0.2751 0.1372 -0.1346

Hi I am using first time speex library, and this is my first problem. I need to decode AMR-NB packet to PCM. I read all manual instruction and I wrote these simple lines of code. bool CMMediaObj::AmrInit() { speex_bits_init(&bits); destate = speex_decoder_init(&speex_nb_mode); int tmp=1; speex_decoder_ctl(destate, SPEEX_SET_ENH,

Hi Jean-Marc, I thought at quality 3 (wideband) - wb_submode1 that the 4-8k band was not using a codebook table. From the code I can see that some sort of "lsp" encoding is performed. What exactly does this encode? (I assume lsp means line-spectral pairs) The reason I am asking is I'm comparing the "effective" spectral bandwidth of Speex against the

Hi all, I'm to hit about aprox the same encoding settings (ie bitrate,filesize, quality) with talk radio shows that are currently encoded with Real Producer in 16kbit acelp (maybe cs-acelp). Currently 2 hours of talk radio encoded in a .rm container yields about 16.3 meg file. With speex in wideband vbr I'm getting very good quality results, but with a larger filesize of 19.1 meg.

Has anyone successfully compiled the AMR codec into an Asterisk install, and if so, what steps did you take? -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20100825/b3db9f37/attachment.htm

Jean-Marc, I am building a tool for producing the highest possible quality Internet interviews for "podcasting" applications. The goal is to produce a perfect recording of an interview or conference -- and giving the participants a glitch-free experience is secondary. My approach, therefore, is to build a Windows "wave" file asynchronously by using a streaming

I was posed this question: A T1 set up for voice carries 24 conversations on a circuit that is 1.544 megabits/second. Right? Well, if you set that T1 up to carry data and run a link between two IP networks over it, how many SIP conversations could it be expected to carry? How about IAX? How would one extend this calculation to varying bandwidth circuits and various VOIP protocols (MGCP,

Hi, according to http://www.freebsd.org/cgi/cvsweb.cgi/src/sys/dev/amr/amr.c?only_with_tag=RELENG_6 it seems MegaMGR for linux now can work. Any experience? -- Cris, member of G.U.F.I Italian FreeBSD User Group http://www.gufi.org/