Displaying 20 results from an estimated 100 matches similar to: "Trying to get cmdline going..."
2000 Jun 11
1
cmdline
I have got CVS version of vorbis.
In cmdline directory
# make
...
gcc -O20 -ffast-math -D_REENTRANT -D__NO_MATH_INLINES -fsigned-char \
-I. -I../lib getopt.o getopt1.o main.o -I. -I../lib \
../lib/libvorbis.a -o vorbis -lpthread -lm
main.o: In function `main':
main.o(.text+0x31c): undefined reference to `fill_buffer'
main.o(.text+0x332): undefined reference to `CheckWav'
2000 Sep 11
0
cmdline does not compile from cvs
When I try to compile "cmdline/vorbis", I get the following.
gcc -O20 -ffast-math -D_REENTRANT -fsigned-char -I. -I../lib -DUSE_ALLOCA_H -DUSE_MEMORY_H getopt.o getopt1.o main.o -I. -I../lib -DUSE_ALLOCA_H -DUSE_MEMORY_H ../lib/libvorbis.a -o vorbis -lm -lpthread
main.o: In function `main':
main.o(.text+0x31c): undefined reference to `fill_buffer'
main.o(.text+0x332): undefined
2004 Aug 06
0
Please 30 second to look a my code
Well, you seem to be using FRAME_SIZE but only defining frame_size.
Otherwise, the code looks OK, but it's always hard to tell. I suggest
you start from speexenc/speexdec or from the example I wrote in the
manual at: http://www.speex.org/manual/node12.html
Jean-Marc
Le ven 19/12/2003 à 05:22, Fabio a écrit :
> Hi
> i'm developing a sort of VoIP application
> for my
2007 Nov 14
0
Audio glitches/Configuration problem !!?
Hi all,
First of all, thanks to Speex developper for the all the job.
I am trying to implement my own Speex DirectShow fitlers for VoIP following
the documentation and sample code's.
I am facing audio glitches when encoding - decoding PCM data. The encoder
and decoder procedures are copied below.
What about the lookahead size ? how shouw we apply it in encoding stage ?
thanks in advance
2004 Aug 06
2
Please 30 second to look a my code
Hi
i'm developing a sort of VoIP application
for my ipaq using speex...
I'm still at beginning and i have many problems encoding and decoding my
wav files....output is only noise! Why?
I'm using
Libspeex 1.1.3,
Embedded VisualC++ 3.0,
Ipaq 3850(206 MHz Intel® Strong ARM 32-bit RISC Processor) PocketPC 2002 (Windows CE 3.0).
Libspeex is complied with the definition of
2000 Apr 11
1
Compiling the 'cmdline' demo
Hello,
I found out about Vorbis and Ogg recently and am very
interested in it. I grabbed the latest sources from CVS and have been
able to compile almost everything. The only thing (to my knowledge)
that I can't compile is the 'cmdline' demo.
Everything compiles correctly there error comes in the linking
stage. The error message is reproduced below:
make target
2000 Apr 11
1
Compiling the 'cmdline' demo
Hello,
I found out about Vorbis and Ogg recently and am very
interested in it. I grabbed the latest sources from CVS and have been
able to compile almost everything. The only thing (to my knowledge)
that I can't compile is the 'cmdline' demo.
Everything compiles correctly there error comes in the linking
stage. The error message is reproduced below:
make target
2004 Sep 10
2
Error initializing flac stream decoder.
Thanks for that email. The one lihe change I made is this :
from #define FLAC__MAX_RICE_PARTITION_ORDER (15u)
to #define FLAC__MAX_RICE_PARTITION_ORDER (6u)
and that seemed to make decoder_new() happy, but it's promptly crashing
after making a call to the read callback (below), then to the meta
callback. The meta callback did nothing but print a string and return.
I removed it, and
2005 Jan 18
0
AMP and Asterisk PSTN extension config
Hi,
I have configured an Asterisk server with TDM01P (1FXO) for testing purpose. The interface I'm using is AMP. I want to configure my extension so that when I dial from my mobile phone to the asterisk line, I want it to transfer the call to any extension, say 3042 and after a particular number of rings, transfer the call to voice mail so that I can record my message.
My Zaptel.conf is as
2007 Nov 04
3
WaveIn/WaveOut and Speex
Hello,
I know my question has been asked before because I spent the last week
searching the web for how to use Speex in combination with
WaveIn/WaveOut and I ran into a few posts, but none of them answer the
question. There is still a lot of confusion how to use WaveIn/WaveOut
and Speex by junior developers such as myself. Even after examining code
for SpeexDec and SpeexEnc, I cannot get clear
2007 Nov 04
0
WaveIn/WaveOut and Speex
I'm not sure what input/output format you're trying to use, but it looks
wrong. You're using the float functions that take +-32767 values and
you're feeding (or converting) chars. Unless your machine has very
special chars (which I doubt), a +-32767 value just isn't going to fit
in. This has nothing to do with Speex BTW, it's just handling the audio
data properly.
2007 Nov 04
0
WaveIn/WaveOut and Speex
> When I was going from Char to float and back looked very wrong to me
it not only *looked* wrong!
> as well, but I was just not sure (and still am) how to translate the
> Char* audio stream generated by WaveIn to a format that can be
> understood by Speex. Would using speex_decode_int and
> speex_encode_int instead of speex_decode and speex_encode be the
> answer?
Using
2005 Jan 26
1
Inbound analog Telco line not answered
I have an X100P clone hocked up to an analog line of my PRI. I can use it
to dial out.
but when I call the extension it answers and says "GOODBY"
I have a Livevoip DID which successfuly rings to ext 202
I am using asterisk@home and through the AMP inface the line should ring to
ext 202
Below are Asterisk Messages, Extensions.conf and Extensions_additional.conf
Extensions.conf
2007 Nov 04
2
WaveIn/WaveOut and Speex
Thank you for such a quick response.
The only reason I started with Char buffers is because WaveIn and WaveOut on Windows XP accept/emit WAVEHDR structures, which store audio data in LPSTR, which is Char*.
typedef struct {
LPSTR lpData;
DWORD dwBufferLength;
...
} WAVEHDR;
When I was going from Char to float and back looked very wrong to me as well, but I was just not
1999 Oct 04
3
Detailed decoder pseudocode (was: Re: ETA?)
> > Which part?
>
> Well, my biggest problem is dealing with files. As you have mentioned
> that fill_buffer() is obsolete, what has replaced it? ogg_sync_buffer()
> didn't seem to be what I was looking for, as far as I can tell... am I
> headed in completely the wrong direction?
Ah, OK, I understand where you're headed now.
The libvorbis API is different than
2007 Nov 05
0
Fw: RE: WaveIn/WaveOut and Speex
Begin forwarded message:
Date: Mon, 5 Nov 2007 07:27:21 -0500
From: "Evgueni Tsygankov" <eugenet@rusmex.com>
To: "Jean-Marc Valin" <jean-marc.valin@usherbrooke.ca>
Cc: speex-dev@xiph.org
Subject: RE: [Speex-dev] WaveIn/WaveOut and Speex
Again, thank you for helping me. I know this might seem like a trivial matter to you and other experts in the field, but believe
1999 Sep 12
3
Hello and questions
Hi.
So, I checked out libvorbis and it seemed pretty OK, but the cmdline
subdir didn't compile. I made changes to Makefile.in and added a
function or two (does "fill_buffer" live anywhere? I couldn't find
it) and eventually got a build that didn't do much.
Anyways, I guess I want real CVS access so I can check the changes in,
and I'd like to slap the cmdline thing
1999 Oct 26
2
building on openbsd...
the library builds with no problems, but the commandline tool stops at
this:
gcc -O20 -D__NO_MATH_INLINES -fsigned-char -I. -I../lib -Dsize16='short' -Dsize32='int' -Dsize64='long long' -c main.c
main.c: In function `main':
main.c:201: warning: assignment makes pointer from integer without a cast
main.c:253: warning: passing arg 1 of `CheckWav' from incompatible
2004 Dec 21
5
AMP - Fax Detections
Does anyone know of any obscur reference for detecting an incoming fax.
I currently have AMP running and everything else is working great.
Installed the spandsp patches and software... using the default AMP
extensions.conf, I start sending a fax, I hear it pick up and transfer
to voicemail after 20s.
Fax is set for system... Here is the detail from the extensions.conf
[global]
FAX_RX = system
2005 Jul 11
1
ASterisk@home + Broadvoice = Almost working installation...
Hello Guys,
I'm somewhat of a newbie and am desperately seeking for some help...
I've managed to get asterisk up and running on my server, and signed up with a
broadvoice account...
I'm having no problem dialing and communicating between extensions, but whenever
anyone tries to call my broadvoice account, they are greeted by no ring or
anything, but rather simply a direct to