similar to: Hardware based DSP

Hi, Are there any optimized codecs for Analog Blackfin DSP? If yes, from where we can download it? We are looking for Speech, Audio and Video codecs. Best Regards, Miroslav Nachev

Hi, I would like to find some way for hardware coding instead software (using the Host CPU). Are there any PCI boards just with codecs (DSP) or other way? Best Regards, Miroslav Nachev COSMOS Software Enterprises, Ltd. Tel: (+359-2) 983-32-62 Mobile: (+359-88) 897-31-95 E-Mail: m_natchev@yahoo.com miro@space-comm.com http://www.space-comm.com

Hi, Thank you. I will try it. Do you know some G.72x, GSM, and iLBC optimized for Blackfin ? I mean open source. -- Best regards, Miroslav mailto:miro@space-comm.com Wednesday, March 23, 2005, 9:05:11 PM, you wrote: JMV> Hi, JMV> As far as I understand, the last patch (for TI C5x) I merged in SVN also JMV> allows Blackfin to work, but I haven't

Dear Jean, The source code for G.729 can be download from ITU for free. Also, some developer can do yourself as open source G.729 codec without any help. In this case each who use this codec which source code is free and open source must pay, but not to the developer. Best Regards, Miroslav Nachev JMV> Le jeudi 24 mars 2005 ? 10:08 +0000, John Villar a ?crit : >>

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I am having trouble building appradius from http://appradius.minitelecom.org/ I configure, make, make install cpprad-1.0, but when I configure, then make appradius I get :- obelix:/usr/src/appradius/appradius1.0 # make make[1]: Entering directory `/usr/src/appradius/appradius1.0/lib' make[1]: Nothing to be done for `all'. make[1]: Leaving directory

Hi Everybody, I am new to Asterisk. Can anybody suggest me some link where I can find architecture level detail of this system. My aim is to find out how easy it is to port it on a new hardware (T1/E1 and POTS)? Any input is highly appreciated. Regards Arun

Hi, As far as I understand, the last patch (for TI C5x) I merged in SVN also allows Blackfin to work, but I haven't tested. Jean-Marc Le mercredi 23 mars 2005 ? 10:01 +0200, Miroslav Nachev a ?crit : > Hi, > > Are there any optimized codecs for Analog Blackfin DSP? If yes, > from where we can download it? > We are looking for Speech, Audio and Video codecs. >

Hi! All I would appreciate if someone could advice me on how stable is sip-h323 & h323-sip translation as well as how many calls can it handle when doing such translation.( assuming single 2.8Ghz intel processor & 1GB RAM) Regards, John -- ___________________________________________________________ Sign-up for Ads Free at Mail.com http://promo.mail.com/adsfreejump.htm

== Manager 'sendcron' logged off from 127.0.0.1 vicidialnow*CLI> dial 919545090201 -- Executing AGI("OSS/dsp", "agi://127.0.0.1:4577/call_log") in new stack -- AGI Script agi://127.0.0.1:4577/call_log completed, returning 0 -- Executing Dial("OSS/dsp", "SIP/19545090201 at sip203||tTor") in new stack -- Called 19545090201 at sip203 Feb 2 13:36:38

Hi, i'm not an expert in dsp but i've seen there is some code to run speex on omap C5X dsp so i was wondering what is the status of this code. Actually my goal is to run a voip application which uses gstreamer on an OMAP 5912 board. As there is already a speex codec for gstreamer, i was wondering if this plugin can already use the dsp power or if there is the need for a new plugin. thank

Hello , I'm a student in Shahed university in IRAN . Now I'm become a member of our uni IT-Center . We are going to have an organized network with a en-bloc user authentication and proxy server. In comparison between MS AD and Samba , i want to choose Samba server ( cause of my belief in Open Source ) . Our plan have a Forest(root : shahed.ac.ir) and trees for faculties (like :

Hi, I am a newbie in asterisk, i could compile it and run it with no problem on a RedHat 9. In the same box, i got a linejack and a phonejack cards and i downloaded the CVS driver from quicknet. This 2 card were working in a openh323 (openphone and pstn) project with gnugk on a RedHat 9. I am using the default samples, and i tried /dev/phone0 and /dev/phone1, but when i run asterisk, i get this

Email : miro@space-comm.com FirstName : Miroslav LastName : Nachev Company : COSMOS Software Enterprises, Ltd. Phone : (+359-88) 897-31-95 Fax : Address : P. O. Box 941 Address2 : City : Sofia State : Outside the US, Mexico, or Canada Zip/Postal

(REPOST, sorry if you get this more than once.) Hello all, (Not sure if this is more appropriate for user or dev list) Does asterisk have any sort of "standards based" api that can enable an application to do call control on the switch ? For example, if I am developing a call center application using asterisk, I would like to be notified of inbound calls and then be able to route

Hi, I'm porting speex as a DSP task to run using the DSP gateway project. I have some results, but I still need some improments, because I'm still getting some error messages in the mail box system... If someone wants to help... it would be useful. 2006/1/16, Ralph Giles <giles@xiph.org>: > On Mon, Jan 16, 2006 at 04:43:56PM +0100, Christophe Augier wrote: > > >

Hi, JMV> ... none of these codecs can ever have open-source JMV> implementations due to patent issues. We use now these codecs for x86 in "C" source code running under Linux. They works very well. My problem is that we are not so familiar with Blackfin DSP. Concerning of patent issues, you are not right. The patent is for their using, not for the source code. Every one

Hi All, I am wondering if I will be breaking the GPL, if I write for example, a channel driver or make some modifications to the astrisk source code, to interface at RUN TIME, through sockets, with a proprietary system. Eg. 1. I write chan_xxx + modify asterisk source (make changes + new code publicly available) 2. chan_xxx supports hardware by XXX Corp. 3, XXX Corps interface is

I noticed a little while back that Andrew@teledesign.co.uk posted he was going to port the speex onto a TI DSP. Any news on this or similar developements? --- >8 ---- List archives: http://www.xiph.org/archives/ Ogg project homepage: http://www.xiph.org/ogg/ To unsubscribe from this list, send a message to 'speex-dev-request@xiph.org' containing only the word 'unsubscribe' in

Hi, I'm running the latest CVS HEAD version of asterisk, and I'm experiencing hangups during voice conversation. This happens quite regularely and often. The problem is in dsp.c, line 1235, where it says accum /= len; But `len', at this point, is 0, resulting in a SIGFPE. The routine ast_frame *i4l_read() in channels/chan_modem_i4l.c:411 is setting p->fr.datalen to