similar to: Help about error in linking to libspeex.a

Hi All, i have configured speex1.2rc1 for xscale-elf (ARM architecture) ,then executed MAKE and MAKE INSTALL. so, i obtained libspeex.a in the /usr/local/lib with libogg.a compiled as well. but i when i link the library to my program (by adding LDFLAGS += -lspeex -lm ), and try to compile, i get this error: /usr/lib/gcc/xscale-elf/3.4.3/../../../../xscale-elf/bin/ld: cannot find -lspeex collect2:

Hi All, ?? I am a new member to the vorbis-dev mailing list. i hope that u receive the help that i've been searching for. I need to compress audio samples captured by wireless sensor node (16-bit PCM at 8Khz). can i use vorbis i such an embedded system environment that has the following HW/SW specifications: -416 MHz Microprocessor(ARM architecture, Intel Xscale family) -32 MB RAM Beside

Thanks for your appreciated help. i found casting fromunsigned short to short gives different value: unsigned short FFFF=65535 where short FFFF= -1. however, can i configure speex codec (specifically speex_encode_int function) to take input of type unsigned short instead of short so that i avoid casting errors ? Regards, Mashal ________________________________ From: Yanick Bourbeau

Thanks a lot for replying, The question now is , after subtracting 32767 from each unsigned sample, will the new signed samples represent the same original voice ? ???? For explanation, assume "Hi" is said in 8000 unsigned sample ,if i subtract 32767 from each sample and play the resulting 8000 sample as signed PCM samples,will they be "Hi" ?! Forgive me about my questions ,

Thanks for replying. actually 2^15 in decimal is the same as 0x8000 in hex. the idea is clear for me that i should scale down each unsigned sample by 0x8000 for becoming signed. I hope that will work correctly as input raw samples to Speex codec. do you think so ? Regards, Mash'al ________________________________ From: Steve Checkoway <s at pahtak.org> To: Mashal al-shboul

Hello, all. I am delightfully slogging my way through installing and configuring Asterisk 1.6.1.1 on CentOS 5.3. I'm learning lots and admiring the product but I'm having a problem getting speex to install and I would very much like to use it. It is not available in menuselect and the problem appears to be with speex_preprocess_ctl: [root at pbx01 asterisk-1.6.1.1]# grep -i speex

Hi, everyone! I'm a newbie of speex develog. I tried to encode and decode with the sample program in the speex manual. I compile those program with gcc sampleenc.c -o enc -lspeex gcc sampledec.c -o dec -lspeex I run these two program in this way and then speak loudly: ./enc /dev/dsp ? ./dec /dev/dsp I thought I would hear my voice. However, what i heard was a lot of noise. I

Hi all speex developers, ??? I have encoded 10 seconds buffer of Mic voice (raw) using speex on an embedded device, then transmitted it to a PC where i decoded the voice,in a way similar to the sampleenc.c and sampledec.c on website, ?BUT my problem is that the decoded voice is totally different from the original,what i hear is just noise !! what is possible to be the reason ? Any help is

Hi All, i need your help in determining the problem in the following sample code (taken from speex manual) for fixed point encoding. i tested encoding a 160 sample frame and then decoding it back But The problem is that the decoded frame is totally different from the original frame(see output below).What is the possible reason for this?.thanks for any help #include<stdio.h>

It seems that speex (speex 1.0.4 on OSX 10.3.9) is writing over the input buffer (with what looks like a sine wave). Included below is a short program that demonstrates the problem. Is this a bug, or am I doing something wrong? I don't see this behavior mentioned in the docs. I don't really need the input buffer, but I'm guessing this problem is related to a high pitched sound

Hi Mashal, 2012/3/14 Mashal al-shboul <shboul8989 at yahoo.com>: > Hi Blaise Potard, > Thank you for help. > Actually,i am sure that my samples are unsigned? , specifically, of type > uint16_t which is unsigned 2-bytes, these samples are provided by an audio > driver for an audio subsystem (microphone ADC...) in a wireless sensor > node(intelmote2 ), following is the

-----Original Message----- From: asterisk-users-bounces at lists.digium.com [mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Joshua Colp Sent: Thursday, September 24, 2015 6:23 AM To: asterisk-users at lists.digium.com Subject: Re: [asterisk-users] problems with PJSIP install on UBUNTU 14.04 On 15-09-23 10:25 PM, Ryan, Travis wrote: > Ok I did all that and it's still

From 1.2rc1 source code: preprocess.c: /* FIXME: This VAD is a kludge */ st->speech_prob = Pframe; if (st->vad_enabled) { if (st->speech_prob > st->speech_prob_start || (st->was_speech && st->speech_prob > st->speech_prob_continue)) { st->was_speech=1; return 1; } else { st->was_speech=0; return 0; } } else { return 1; } AND case

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?

I don't understand why you're subtracting 2^15 - 1. Assuming you really do have unsigned samples (which is very odd) that have 0x8000 as the mid point of the range, then you need to subtract 0x8000, as I believe I pointed out already. By mid point, I mean if you have a pure tone (i.e., a sine wave) that has unsigned range 0 to 0xffff, then the mid point is 0x8000. The mid point should be

On Sat, Jul 19, 2008 at 12:29 PM, ???? <oovoov008 at gmail.com> wrote: > Hi, everyone! > > I'm a newbie of speex develog. I tried to encode and decode with the > sample program in the speex manual. I compile those program with > > gcc sampleenc.c -o enc -lspeex > gcc sampledec.c -o dec -lspeex > > I run these two program in this way and then speak loudly:

Short question: is there way to tell EM to actually send data after send_data call? I''m building a file transferring app. I send Mashal.dump''ed metadata first, and then - the file contents (chunked). I found a silly bug: receive_data() gets marshalled metadata and the first chunk of the file in a single variable. Like that: c1.send_data("meta")

This works well. Thanks for the link. However I am curious about something. Speex requires a VAD for the VBR mode and the comfort noise preservation. Is the Speex portion for this different than the VAD result returned from the preprocessor call? Vas ________________________________ From: speex-dev-bounces at xiph.org [speex-dev-bounces at xiph.org] On Behalf Of Ken Smith [ken at alanta.com]

For what it's worth, I've had significantly better luck with the WebRtc VAD. It's pretty good. http://code.google.com/p/webrtc/source/browse/#svn%2Ftrunk%2Fsrc%2Fcommon_audio%2Fvad%2Fmain%2Fsource In general, the WebRTC voice engine seems to be more sophisticated and mature than the Speex preprocessor (as opposed to the Speex codec, which is pretty good). Ken Smith Cell:

I need to set the content to 0 because actually is not using malloc(), it uses calloc() that does set the memory to 0. Best Regards --------------------------------------------- Helios Cuevas L?pez Embedded Software Engineer Castle Care-tech Ltd. First Floor, 6 Bracknell Beeches, Old Bracknell Lane West, Bracknell, Berkshire, RG12 7BW Office - 01344 469470 Direct Dial - 01344 469485 FAX - 01344