similar to: basic include files for speex

It's not an include file missing. This is a link error. You need to link with libspeex (or whatever it's called when you build it). Jean-Marc Quoting Caroundw5h <caroundw5h@yahoo.com>: > Hi Jean, > > I'm trying to compile the encode.c and or the test programs that are on the > site and in libspeex respectively. however, i keep getting some undefined >

On Sat, Dec 14, 2002 at 01:46:19AM -0500, Jean-Marc Valin wrote: > Thanks for the patch. I applied it and it give me up to 15% in speed. > Doesn't seem to change the results, which is a good thing (though you > originally forgot a "used=0" in vq_nbest_sign). I'll check a thing or > two and I'll apply to CVS. D'oh. My carelessness, sorry! :) > Strange...

For a streaming application like VOIP, you collect 20 ms of samples, feed this through the encoder, stick it in an RTP packet, and send if over the network. On the receive side you feed packets through a jitter buffer to the decoder, and then copy the output audio to your output device. Speex runs efficiently on most compilers, so the real-time requirement is not a big deal, as long as you

Hi, This is Chandra. I have Polycom IP 501 phone. Its showing wrong time on the display screen. How can I set the "New York" time? What value I have to give to GMT offset value? Look forward to your response. Thank you. Regards, Chandra. --------------------------------- Ahhh...imagining that irresistible "new car" smell? Check outnew cars at Yahoo! Autos.

I have a patch for libspeex, which optimises some of the loops in vq_nbest and vq_nbest_sign that speeds up encoding - my results: test file: 10s wav file at 16000 Hz, mono encoding with wideband --quality 3, --comp 3 machine: PIII-900Mhz, 256MB RAM before: 2.78s after: 2.38s I'm still trying to grasp the code (I'm just a coder, no background in sound processing), and just optimised

I am trying to cross compile openssh for arm5b-jungo-gnu-gcc,, and also using openssl and zlib from same compiler arm5b-jungo-gnu-gcc but i am getting following error,,, /usr/local/openrg/armv5b-jungo-linux-gnu/bin/armv5b-jungo-linux-gnu-ld: warning: cannot find entry symbol _start; defaulting to 0000b9c8 /usr/local/openrg/armv5b-jungo-linux-gnu/bin/../armv5b-jungo-linux-gnu/lib/libcrypto.so:

Hi, For the openSpeak project we use PortAudio V19 & speex. So if you want some code examples, you can look an pick in our SVN (its GPL). ATM we are having some problems with PortAudio v19 (it might still be buggy somehow, it seems not to support the 32-kHz sampling rate used by speex ultra-wide-band mode) so if you want something stable you'd better use v18. And of course, if you

okay that questioned is answered, thank you. I am interested in using speex in a VOIP application. do i need to put it in into the ogg contianer format in order to encode/decode and send it? or will it work "as is"? if the latter then: "the packet is larger than the allocated buffer" message: whats your recomendaton for fixing that? i was thinking simply getting the size of

Hi All, I have just started looking at the code for Xen so please bear with me. A domU Linux kernel running with independent_wallclock=1 seems to sync its time with dom0 after every "xm unpause" (obviously preceded by an "xm pause"). I don''t see where the xtime variable is being updated after an "xm unpause", i.e., domain_unpause_by_systemcontroller().

> -----Original Message----- > From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users- > bounces@lists.digium.com] On Behalf Of Noah Miller > Sent: Wednesday, April 25, 2007 9:52 AM > To: Asterisk Users Mailing List - Non-Commercial Discussion > Subject: Re: [asterisk-users] My Polycom IP 501 is formatted its file > systemitself > > Hi Chandra - > >

i don't think so. this is my current code: for(i=0; i<samples;i++){ /*only get max 320 bytes at a time so bits can handle*/ Client->raw_audio[i] = *input++; Client->temp_audio[i] = Client->raw_audio[i]; Client->session_File[ndx] = Client->temp_audio[i]; ndx+=1; } speex_bits_reset(&ebits);

I am very new to linux and openSSH ,, i am trying to compile openssh for jungo openrg IXP45. but while configuring i am getting following error,, hecking for libgen.h... yes checking for getspnam... yes checking for library containing basename... none required checking for deflate in -lz... no configure: error: *** zlib missing - please install first or check config.log *** if anybody will me

> Using zsh's time function (gives real and user time - which are > pretty similar on this unloaded machine). I've put my sample wave > file at http://dagobah.ucc.asn.au/speextest/sample.wav if anybody > wants to compare times. > > I'm curious now why my machine is slower - perhaps it's something > about the way I've compiled it. (Compiles by default with

I've got a voicemailbox with one message store. When I try to read it, I get the followiing error: ast_openstream_full: File digits/1F does not exist in any format Obviously, I can just clear out that mailbox, but is this a bug that I should be reporting? /Per Jessen, Z?rich

Hi Everybody: I am doing the project in speex, and want to make the lib/dll in symbian platform. I checked there is one symbian project file, including bld and mmp files in speex-1.2rc1. But while I import the project and compile in Carbide. It showing so many errors: ***Invoking abld command perl.exe -S ABLD.PL \Symbian\Carbide\workspace\speex\symbian\ export make -r

Hello, I am trying to compile Speex on RealView Compiler and a lot of errors are indicated by compiler: libspeex\bits.c(48): error: #77-D: this declaration has no storage class or type specifier libspeex\bits.c(48): error: #65: expected a ";" libspeex\bits.c(139): warning: #12-D: parsing restarts here after previous syntax error libspeex\bits.c(141): error: #77-D: this declaration

--- Jean-Marc Valin <jean-marc.valin@usherbrooke.ca> wrote: > > Sorry if this a repost but I want to create the > > smallest Speex library possible to be put in TI's > > TMS320 DSP. I'm only interested in one > configuration: > > 5.97 Kbps narrowband. What part of source code > can I > > remove? Currently, when I compiled the version >

I am having trouble compiling ffmpeg to support speex, which didn't work with the ubuntu libspeex-dev package, but looks like it might with the Speex version 1.2rc1 tarball from http://speex.org/downloads/ How do I tell ffmpeg's configure and/or make to use the 1.2rc1 version of libspeex in /usr/local/include instead of the older debian/ubuntu libspeex-dev package in /usr/include/speex?

Hi, I'm one of the people working on the Rockbox project (http://www.rockbox.org) which is an open source alternative firmware for a range Digital Audio Players. Recently we integrated support for the Speex codec using libspeex and seems to work well. If you could add Rockbox to your list of software that supports Speex, that'd be great. So that's the introduction done. Now for

Hi, I have configured the below things: Extensions Trunk Outbound route Inbound route IVR Ring group If anybody call to my DID number, my IVR is responded. After that, if he press 1, then Ring group will be activated. All are working fine. My Problem: I want to avoid IVR for some phone numbers depends on their called IDs. If my common users will call to my DID