similar to: voice delay after pause speaking

hi, this is a very basic question -- sorry for posing it: how can i force R to print 0.0001 instead of 1e-04??? .--------------------. | > 0.0001 | | [1] 1e-04 | `--------------------' i tried the functions format, formatC, ... and changed options()$digits with no success! thanks for advice, tomy -- no signature

hi, recently i discovered the functionability summary.formula, awesome! from the help page i understand that method=reverse allows to summarize all variables on the right hand side of formula (the help page on line 229 wrongly refers to the left? hand side variables) in categories which are determined by a single left hand side variable. my problem is that the argument fun seems not to be

I've been playing with LLVM for a couple of months now and really am enjoying myself. I finally got around to doing something useful and had started implementing bindings for Ruby. I've been working off the tip of trunk and realized my build didn't have shared libs. I've spent the better part of a day trying to build llvm with shared libraries with no luck on either Mac

Does anyone on this list know of any Linux based apps that will work with Dialogic or Brooktrout that provides voice-mail/Autoattendant only?? It seems that Panasonic, Avaya, and Mitel all use Unix/linux based OS on their firmware for their proprietary voice mails. My wish list would be; A software that provides all of the drivers for a dialogic or brooktrout board Voice Mail Messages in WAV

https://bugzilla.mindrot.org/show_bug.cgi?id=1999 Bug #: 1999 Summary: When speaking v2, send client version first to avoid long delay with some proxies Classification: Unclassified Product: Portable OpenSSH Version: 5.9p1 Platform: All OS/Version: All Status: NEW Severity: enhancement

https://bugzilla.mindrot.org/show_bug.cgi?id=1999 Damien Miller <djm at mindrot.org> changed: What |Removed |Added ---------------------------------------------------------------------------- Status|NEW |ASSIGNED Assignee|unassigned-bugs at mindrot.org |djm at mindrot.org Attachment #2145|0

[I have sent a message about SIP problems via gmane, but it seems the list is gatewayed one-way only...] The message was: I've been trying to use Asterisk as a SIP->PSTN gateway. It runs fine when the SIP client is on the local network and there is not packet loss. But now I've tried running a remote client (halfway around the globe) -- this works great until some packets get lost.

Hello, I'm using Speex Preprocessor to supress noise, eliminate echo etc. But I have another preprocessor state that I want to use ONLY to determine voice activity. int res = speex_preprocess_run(m_VADOnly, (spx_int16_t *) pStreamSampleBuffer); pStreamSampleBuffer is modified after call to speex_preprocess_run. I have manually turned off noise suppression and AGC but anyway - VAD modifies

Hey sorry to hijack this thread, but I just remembered a request I wanted to make to the speex devs. I tried using the activity detector, but I just couldn't get it working well. I ended up using my own, where I think it just considered voice on if it passed a certain threshold (I know, pretty primitive). I also tried one that checked for a signal, like if the strongest frequency

Yes, you can use it independently by using speex_preprocess(). This function can do noise removal, AGC, and VAD. I've been using it to do all three and it usually works very well. There can be a train going by outside, producing lots of sound that makes it through the noise filter, and yet the VAD knows it's not speech. However, sometimes the VAD seems to get into a bad state and

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...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

This must be a simple issue, but I cannot figure it out. I want to use VAD, but I don't know how to check if the actual frame has voice in it or not. So, in my code, I do: int tmp = 1; speex_preprocess_ctl(preprocess_state, SPEEX_PREPROCESS_SET_VAD, &tmp); speex_preprocess_ctl(preprocess_state, SPEEX_PREPROCESS_SET_DENOISE, &tmp); then later, for each frame

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

Hi All, I am running asterisk on Linux machine and trying to use confbridge application. Please have a look at Conf files. sip.conf ====== [general] context=default allowoverlap=no bindport=5060 bindaddr=0.0.0.0 srvlookup=yes disallow = all allow=ulaw allow=alaw defaultexpiry=100 [5001] type=friend nat=yes host=dynamic canreinvite=no context= conferences disallow = all

Hi, I'm experiencing some voice delay (2-3 sec) after outgoing call is setup. It means during the first 2-3 secs, audio is very choppy or nothing. So usually I can't hear the 'Hello". I use IAX2 for my ougoing calls with Grandstream phone as a client. Any hints to prevent this? Thanks, David

we are experiencing 30 second delay before voice is heard after answer when we ran wireshark it showed the problem between frames 634 (where the softphone answers) and 1366. Between those frames, asterisk receives RTP packets from both the softphone and the sip carrier, but doesn't forward them to each other.... Then we see a bunch of packets sent at once from the asterisk server, until

Hi all, This is my first post to the list. I have searched the net far and wide but can't find an answer to this problem. When I have call forward working or use the voicemail from a SIP phone, the first part of the message is always cut off. So instead of hearing "call forward cancelled" I hear "l forward cancelled". Or in voicemail I hear "edian mail"

Thanks for your reply. I changed my code to: if (speex_preprocess_run(preprocess_state, shortPointer) == 1) { speex_encode_int(enc_state, shortPointer, &enc_bits); } In the mobile version of the software, compiled against the mobile build of Speech, I get 1 and 0 based on whether the speech is detected. In the version of the software compiled against the Win32 version of Speex,

hi guys, i am trying to develop an application having a capability to detect voice activity in voice stream and then record only portion of this stream which contains the activity. i went through these steps. 1. Captured voice through Waveform Functions of Windows Multimedia API 2. Started Speex Preprocessing liberary 3. Turned on preprocessor by calling speex_preprocess_state_init and