similar to: frame size

Hi Jean-Marc, I'm implementing a VoIP client and I want to use your codec init. I need to ask you some questions about Speex because I can't get too much information about it on the web. I hope you can help me. 1.- First of all, I want to know if Speex provides a buffer for Jitter..If it does, what can I do to use it in my softphone? 2.- How does Speex manage the packet loss? (In a

"CALL FOR PAPERS" THE ISLAMIC SOCIETY IN SOUTH EAST ASIA CONFERENCE ON STATISTICAL AND MATHEMATICAL SCIENCES BANDUNG, 25 - 26 APRIL 2003 Fakultas Matematika dan Ilmu Pengetahuan Alam Universitas Islam Bandung Jalan Tamansari No. 1, Bandung 40116 Telepon +62 22 4203368 Ext. 136 or 135 Facsimile +62 22 4263895 E-mail: - fmipa_unisba at yahoo.com - wanda at

Thanks again Steve. I'll search for the term you mention below. What I really want is to take the output of the speex encoder and spit it out on the network via RTP. I haven't been able to find a library or code example that does that. On 11/01/2010 12:03 PM, Steve Kann wrote: > Jeff, > > It's in the manual: > >

Yes, I have made that search, but I'm restricted to Java. On 11/01/2010 12:21 PM, Steve Kann wrote: > Have you tried typing "speex rtp" into google code search? It gives lots > of examples of real applications which do exactly that. > > http://www.google.com/codesearch?as_q=speex+rtp > > > -SteveK > > > On 11/1/10 1:13 PM, "Jeff

Hi, I've compiled the speex codec and installed as per docs, speex works perfectly between Asterisk gateways using IAX protocol. My problem is between X-Lite softphone and a Asterisk Gateway users cant hear anything when the speex codec gets selected. Has anyone else experienced this with X-Lite or any other softphone client? Is their any other Softphone for Windows that makes use of the

Hi all, I am developing a VoIP application and have used SPEEX for that purpose; I successfully managed to port the codec in fixed-point to an ADSP-21364 processor, and the codec works fine in narrowband, wideband and ultra-wideband mode. Currently, I have chosen, in accordance with the speex manual, a 20 ms frame size in each of these modes: in other words, the input buffer for speex was set to

Joost Witteveen (joost@iliana.nl) wrote: > > So, each UDP package with 20 bytes speex-data, we send: > > 20 bytes speex > 12 bytes ogg headers (and others?) > 28 bytes UDP/IP headers (2 IP numerbers, 2 portnumbers, checksum, etc, etc) > > and, if it goes over the phone, each package has a few ppp headers. > > Am I overlooking something, or does this fixed frame

Thanks Jean-Marc. I do want to change Speex frame size to 320. In this case, I've changed frame size in init and modes (include framesize and subframesize). However, the audio quality becomes bad. It seems something else needs to be changed as well. Could you give a clue? Thanks, Dennis -----Original Message----- From: Jean-Marc Valin [mailto:jean-marc.valin@usherbrooke.ca] Sent:

hi, guys. I come from china, a student of BUPT. I have developed a softphone using osip&ilbc. Now, i want to add speex to my softphone. I heard speex have been a daft of rfc, but I just found draft of the format of rtp in ietf's website. Is speex dead? Who can tell me that? Thanks in advanced. yunyong

Hi, I hv a query. while decoding a voice session, i saw that one side of voice was using speex/16K with vbr=off and other with speex/16K with vbr=on. Packet Size in first case is 70 Bytes Packet Size in second case is 104/106/118 Bytes etc. when i send these packets for decoding the decoded ouput is 640 bytes in all the case. Now speex with vbr=off, is decoding well, but the other side is very

Have you tried typing "speex rtp" into google code search? It gives lots of examples of real applications which do exactly that. http://www.google.com/codesearch?as_q=speex+rtp -SteveK On 11/1/10 1:13 PM, "Jeff Ramin" <jeff.ramin at singlewire.com> wrote: > >Thanks again Steve. I'll search for the term you mention below. > >What I really want is to

Thanks Steve. Is there a document anywhere that shows how many bytes/bits of data are produced by the speex encoding process for a given amount of time sampling rate and quality setting? On 11/01/2010 09:41 AM, Steve Kann wrote: > Jeff, > > RFC-5574 is standards-track: http://tools.ietf.org/html/rfc5574 so, > while it's not an approved standard, it's more standardized

Thank you so much for your quick reply. It looks quite easy to change it. But, how about those codebook and excite table? I notice them have been fixed as the length of 160. Thanks again, Dennis -----Original Message----- From: Jean-Marc Valin [mailto:jean-marc.valin@usherbrooke.ca] Sent: Wednesday, February 13, 2008 4:03 PM To: Dennis Liu Cc: speex-dev@xiph.org Subject: Re: [Speex-dev] Can

Jeff, RFC-5574 is standards-track: http://tools.ietf.org/html/rfc5574 so, while it's not an approved standard, it's more standardized than a lot of interoperable traffic on the internets these days. The RFC specifies packetization guidelines, which is basically that you put one or more frames in a packet, and then pad the rest with 0 bits until you have a while number of octets.

Hi All, I am working on my master project to port the Speex Acousic Canceller to FPGA. But I am not able to get a satisfied result in its performance whenI integrated it to a SIP softphone as my first step. The SIP softphone calls to a PSTN gateway and lands to my land line phone. I can hear the echo of my every word while I talks on the land line phone. I am using G.711 codec, frame_size 80

[At the risk of educating you about something you might already know] Natural speech in most human languages gradually changes from one phoneme to the next. Concatenating phonemes together from a fixed, prerecorded, unflexible set would give rise to abrupt changes between them (both in phoneme quality and in pitch), and thus make the resulting speech hard to understand and/or uncomfortable

Hi Jean-Marc, Thanks for the suggesting. I tried the test wtih my recording file offline through the testecho. I got the same result(no prepocessor echo suppression) as my recording file for that live call. Basically I don't see the difference between input_frame and output_frame. As I mentioned in my original post, the echo return loss is about 20db. I guess the echo cancelleris not really

All, Wonder why all my posts has no line breaker when it displayed the forum... I used Hotmail(Windows Live Mail) with FireFox 2.0. I am posting this msg with IE 7.0 and see if it make any difference. Anybody experienced the same issue and how to workaround it? Thanks a lot! Regards, William From: espzzh@hotmail.comTo: jean-marc.valin@usherbrooke.caCC: speex-dev@xiph.orgSubject: RE:

Hi, I use SpeexEcho in my own softphone using MGCP protocol register to a PBX under WindowsXP in my laptop. I make a call: softphone ---> PBX ----> MediaGateWay ---PotsLine---> Traditional Phone. In the other side(TraditionalPhone), I did not here any echo voice, SpeexEcho works well. But, if I let laptop to make some sounds, such as play music, I heard all

Jeff, It's in the manual: http://www.speex.org/docs/manual/speex-manual/node10.html (table 3 and 4). However, if you're asking this, you're probably trying to do something wrong, or the hard way. You probably shouldn't be taking speex output, and trying to "count bytes". If you are using the API, then you will just get the bits out, and then you'll know how