similar to: voice activated recording

Howdy, I'm reading your managing software with yum document and wanted to point out that there could be a problem with the examples that show 'su -c" some command"'. Su without the "-" or "- root" (e.g., "su -" or "su - root") will allow current environment settings to be inherited by the new root shell. This could lead to disaster

--- Harry Sack <tranzedude@gmail.com> wrote: > 2007/7/27, Josh Coalson <xflac@yahoo.com>: > > > > > > > But how is it possible then the FLAC encoder allows files which > have > > > a bad > > > resulting MD5 to be encoded? Is it because of the bad ram, ... > this > > > incorrect MD5 is not detected during encoding? > > >

Hi: I want to have conference call service and I use A102d sangoma's card.Do I should install ztdummy or app-conference? Best regards. --------------------------------- Yahoo! oneSearch: Finally, mobile search that gives answers, not web links. -------------- next part -------------- An HTML attachment was scrubbed... URL:

On 6/23/2015 2:05 PM, Frank Cox wrote: > I'll have to dig out the manual for my piano and do a bit of reading to see what's really going on with those plugs and whatnot. The only thing I've ever plugged into it so far is headphones. if you can plug in headphones, then you can plug it into a computer line-in jack with a suitable cable, just set the headphone volume on the piano

On 06/22/2015 07:20 PM, Frank Cox wrote: > The last time I wanted to record a voice announcement on my computer was a few years back, and as I recall I hooked up my Microsoft Lifechat headset and used Audacity to do the recording. > > Now I want to record a "recitation", which consists of me playing my piano and talking over it. The USB MIDI port won't give you audio, just

The last time I wanted to record a voice announcement on my computer was a few years back, and as I recall I hooked up my Microsoft Lifechat headset and used Audacity to do the recording. Now I want to record a "recitation", which consists of me playing my piano and talking over it. I have a Casio CDP-230 digital piano but I've never tried hooking it up to my computer and making a

I done it speex_preprocess_ctl(sppPreprocess, SPEEX_PREPROCESS_GET_PROB_START, &g.s.VADstart); speex_preprocess_ctl(sppPreprocess, SPEEX_PREPROCESS_GET_PROB_CONTINUE, &g.s.VADcontin); /*char *lisDebugCh = (char*) malloc(20); int decimalFcvt, signFcvt; static int firstDebug = 1; lisDebugCh = gcvt(g.s.VADstart, 20, lisDebugCh); if(firstDebug == 1) { ::MessageBoxA(NULL,

Sorry for top-posting but my webmail forces me to. I added -W to the APPEND line as suggested but I'm still getting the same result: Booting... Altiris, inc. X86PC PreBoot, PXE-2.x Enhanced Build ID=402 PXEPreZero: Invalid PXE Server list format. and the client PC freezes right there. Here's the full content of my dhcp.conf: max-lease-time 86400; ddns-update-style interim;

On Wed, Mar 5, 2014 at 1:55 AM, Vieri <rentorbuy at yahoo.com> wrote: > Sorry for top-posting but my webmail forces me to. Odd. It's been a while since I used Yahoo but I didn't think I had that issue. GMail does default to top-posting but clicking the ellipsis to look at the previous email is enough. > I added -W to the APPEND line as suggested but I'm still getting

What you want is simply a loudness threshold-based detector. It's not very complicated to do, but there's nothing in Speex that currently does that (not that I think it's really useful in practice). Jean-Marc Le jeudi 02 mars 2006 ? 04:26 +0100, Lis a ?crit : > Sorry. > > I forgotten the words volume or loudness. > But it is know as microphone stroke too, i think. >

Hi, I would like to know if one can set the outgoing caller ID within Asterisk when calls are going out through: 1) an analog POTS line (I suppose not) 2) a telco BRI line (I don't think so) 3) a telco PRI line (maybe) 4) a voip provider (surely) Thanks, Vieri ____________________________________________________________________________________ Moody friends. Drama queens. Your

Sorry. I forgotten the words volume or loudness. But it is know as microphone stroke too, i think. If something can tell me something about that procedure it would complete my pleasure. To bring back memories, i only wanted to know wheather i can change a variable that holds the sound intensity (loudness) needet to start "encoding >> sending" if the speex codec is in voice

Lis, I suggest you try tweaking Speex's VAD probabilities as Steve suggested. But consider a simple threshold-based approach as a backup option. Personally, I struggled with Speex's VAD algorithms (both encoder and preprocessor) for a long time, tweaked the probabilities, wrote special case code to work around the mistakes, and was still never satisfied with the results. In times

Hi, What does this message imply? [Apr 29 14:46:30] NOTICE[32175] channel.c: Dropping incompatible voice frame on IAX2/trunk1-9085 of format alaw since our native format has changed to 0x4 (ulaw) If voice frames have been dropped then I suppose that the call quality may be affected? Vieri

Hi...Tom, How to use the code you written? Can you show me some example? Thanks, -----Original Message----- From: speex-dev-bounces@xiph.org [mailto:speex-dev-bounces@xiph.org] On = Behalf Of Tom Grandgent Sent: Friday, March 03, 2006 12:57 AM To: Steve Kann; Lis Cc: speex-dev@xiph.org Subject: Re: [Speex-dev] Voice Activation Level (speex 1.1.11.1) Lis, I suggest you try tweaking Speex's

Hi, I'd like to get my microphone to work (mainly to use it with Skype). I've just been looking for a simple audio recording app, as I vaguely remember having seen such a thing before, only I can't seem to find it anymore. I'm using the default GNOME desktop with the latest CentOS 5. Any suggestions? Niki

Hi, Suppose you have 2 identical Asterisk servers and 1 alias IP address that you assign to either one, according to system failures, etc. Also suppose that all SIP clients register requests go to the alias IP address. Imagine server1 fails and server2 gets the alias IP address. Correct me if I'm wrong but I would have to wait at least 60 seconds before most SIP clients re-register to

I implemented the calcPower(). It works perfectly. The example is given you in just about 6 hours. Cant paste the whole source here and need to meet someone now. Thanks all (particulary tom). I try to figure out whitch problem exists with the #define SPEEX_PREPROCESS_SET_PROB_START 14 theese days ----- Original Message ----- From: "¼Õ½Â¿ø" <ssw0725@ncsoft.net> To: "Tom

I just found an old bug report at bugs.digium.com with exactly the same problem. It's really too bad this bug wasn't addressed: http://bugs.digium.com/view.php?id=7315 --- On Fri, 1/9/09, Vieri <rentorbuy at yahoo.com> wrote: > I'm trying to figure out how to reload iax2 (without > breaking existing calls) so it can listen on a new IP > address (like "ip addr

Hi, I have a Linux server at ip address 10.215.144.7 running DHCP, TFTP and syslinux. DHCP config contains the following: next-server 10.215.144.7; filename "/pxe/syslinux/pxelinux.0"; and the 'default' pxelinux.cfg contains: LABEL altiris ??? MENU LABEL ^7. Altiris ??? COM32 pxechn.c32 ??? APPEND 10.215.144.60::/BStrap/x86pc/BStrap.0 When a PXE client boots in my network