similar to: CCITT A-Law to Speex conversion

Hi, Jean-Marc Valin wrote: >> I haven't tracked the Assignment of the SubModeID through the source >> code, so I just wanted to ask, if I can assume "speexenc -n" to use >> SubModeID 5 always? > > Only if you never change the bit-rate. But speexenc -n was a dumb idea > in the first place... I know ... Thats why I downsampled the WAV-File to 8kHz (I got

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hello For some reasons, I need to encode in mono a stereo wave file, as with the --downmix option from oggenc. I searched in speexenc man page and in the on line documentation, unsuccessfully. Of course I could first use something like sox to down mix my file, but it would be more practical to have an option directly in speexenc. - -- RIQUER

this is not really a development question, but i didn't find another mailing list to ask it. any idea why size of wav file doubles when it is encoded to speex and back to wav: $ ls -ls testi.wav 40 -rw-r--r-- 1 foo foo 40674 May 4 14:38 testi.wav $ speexenc --denoise --agc --quality 10 testi.wav testi.spx Encoding 8000 Hz audio using narrowband mode (mono) $ ls -ls testi.spx 20

I'm trying to link sox to the ogg vorbis static libraries but I'm getting the following error: gcc -L. -L./gsm -o sox sox.o -lst -lgsm -logg -lvorbis -lvorbisfile -lvorbisenc -lmad -lmp3lame ld: Undefined symbols: _oggpack_bytes _oggpack_get_buffer _oggpack_reset _oggpack_writeclear _oggpack_writeinit _oggpack_read _oggpack_readinit _oggpack_write _oggpack_adv _oggpack_look

Witam! I have a question about speex. how to prepare wave file to use with sample code that is on your site: http://www.speex.org/manual2/node12.html . If u can please answer me soon. -- Pozdrawiam Marcin Sliwa mailto:marcin_sl@wp.pl

I'm a new Asterisk user. I've managed to set it up to do everything I want except sound good. Currently, Asterisk sounds considerably worse than my cell phone. I know VOIP can be _better_ than my cell phone, because I've heard Skype do it. (Using 32k iLBC, I believe.) I did an experiment with audio quality: 1) I made a recording which was pretty good. I used an iSight

Hello, I do test runs of speex using speexenc -n [name].wav [name].spx The wav-files (6 different speech files) hereby are sampled at 44100Hz. Most interestingly, all Frames have SubModeID 5. I haven't tracked the Assignment of the SubModeID through the source code, so I just wanted to ask, if I can assume "speexenc -n" to use SubModeID 5 always? Ciao, Bj?rn -- You will

I'm trying to get speex to encode a bit faster, mainly by rewriting a few functions in SSE and translating the GCC __asm__ to VC __asm. There's 2 functions I'm targeting, first is vq_nbest which consumes 40% of the time at high complexity and split_cb_search_shape_sign. Which consumes just over 30%. I've split out two functions from: cb_search_precompute_energy - loop at the

> Hi Daniel, > > I'm intrigued by this and wanted to try it out - but I'm wondering how > you get Asterisk to call sox at all during Voicemail()? Our server > doesn't even have sox installed, so I'm not sure how to go about > tricking Asterisk into running a different one. To do anything useful you would have to get sox installed on your server. But to get

Hi, Can someone please comment about the below issue [root at host0040 kaushal]# file obd-demo.mp3 obd-demo.mp3: MPEG ADTS, layer III, v1, 256 kBits, 44.1 kHz, Monaural [root at host0040 kaushal]# sox obd-demo.mp3 -e stat sox: Failed reading obd-demo.mp3: Do not understand format type: mp3 [root at host0040 kaushal]# sox -V obd-demo.mp3 -r 8000 -c 1 -t ul -w vm-intro.ulaw sox: Failed reading

Hi, Can someone please comment about the below issue [root at host0040 kaushal]# file obd-demo.mp3 obd-demo.mp3: MPEG ADTS, layer III, v1, 256 kBits, 44.1 kHz, Monaural [root at host0040 kaushal]# sox obd-demo.mp3 -e stat sox: Failed reading obd-demo.mp3: Do not understand format type: mp3 [root at host0040 kaushal]# sox -V obd-demo.mp3 -r 8000 -c 1 -t ul -w vm-intro.ulaw sox: Failed reading

For those of you who want to be able to decode Ogg Vorbis files into other formats, SoX (the swiss army knife of audio programs) now has Ogg Vorbis support in CVS. It can read Ogg Vorbis files and export the audio in any format SoX supports. You can also write Ogg Vorbis with it, but, because of limitations in the design of SoX, only output at 128kbps is supported. Note: Instructions on how to

On this macro I keep getting exited non-zero on s,3, but s,3 is doing what it is suppose to do but the macro stops. Is there a way to make a macro ignore errors and continue to s,4? I have the lattes ver of sox 12.17.4. Also if I just run this line from the command line I don't get an error. [root@redhat monitor]# sox in.wav in-rev.wav reverse [root@redhat monitor]# [macro-record-cleanup]

> On Dec 17, 2009, at 8:50 PM, dathead2 at gmail.com wrote: > > On Dec 17, 2009, at 8:42 PM, Ron Decline <rutlecorps at gmail.com> wrote: > >> Is it possible to switch the Left / Right channel when encoding in FLAC? >> (I have some flac files with incorrect left/right channel assignment that I would like to re-encode with flac) >> _____________________________

Hello, all. I'm running Asterisk 1.6.1.6 on CentOS 5.3 in a multi-tenant environment with IMAP voice mail storage on Zimbra. One of our clients is having a problem when transferring voice mails from one mailbox to another (option 8 in the standard voice application menu) using their Snom 320 and 360 phones. The end results is the final recipient cannot listen to the voicemail. We also email

Hi. I'm wondering if anyone out there can give me a little help trying to figure something out: I presently am running CentOS4.1 with GNOME as my desktop (It runs quite well, I must say!). I just have a little curiosity: I have a few audio files (.wav, .mp3 and .ogg) and I like to "preview" them. When run my mouse over the .wav files (and mp3 files) the

I thought I would post this in case it might be of any use to anyone. Not anything special but it does work. Keep in mind you need sox and wmix. Here is some relevant exerpts of my extensions.conf using John Todds macro. [globals] CALLFILENAME=foo FOO=foo CALLERIDNUM=foo [default] exten => 287,1,Macro(dial,SIP/agent20002|20) exten => 287,2,Voicemail(u287) exten =>

hey Andy, you could use sox something like sox old.mp3 new.mp3 fade t 3 0 0 you need to first compile sox with libmad or libmp3lame http://sox.sourceforge.net/ adam ..on Mon, Mar 06, 2006 at 10:30:27PM -0000, Andy Woolley wrote: > Hi All, > > Sorry about the off topic post but I'm having trouble finding anything that > will work, thought one of you guys might have a

Thanks. sox was exactly what I was looking for. -----Original Message----- From: volsung@asu.edu [mailto:volsung@asu.edu] Sent: Thursday, February 08, 2001 2:06 PM To: vorbis-dev@xiph.org Subject: Re: [vorbis-dev] Conversion API for computer telephony systems (Dialogic Mu-law wa v format to PCM encoded wav format) On Thu, 8 Feb 2001, Giovanni Sanfelici wrote: > I am working on a project

Hello: I would like to produce some GSM Prompt audio files for a Telephone Directory Project-- and have hired a freelance audio engineer to record, and edit the actual files-- However the GSM files he gives me to upload into asterisk DO NOT work when played back throgh "Stream File" or "Get Data" in my agi. It seems that there may be more than one GSM file type (with