similar to: Combining multiple speech files into one

Reed Hedges a ?crit : > > If you do decide to do things this way, decode from each of the three > streams seperately (with different decoder states) and combine them > somehow after decompression. The jump from one stream to the other will > sound poor, though, both from the cuts in the output, and also from > sending discontinous input to the decoder(s). That's the case

If you do decide to do things this way, decode from each of the three streams seperately (with different decoder states) and combine them somehow after decompression. The jump from one stream to the other will sound poor, though, both from the cuts in the output, and also from sending discontinous input to the decoder(s). Reed suhaib obeidat wrote: > Dear All, > > I am trying to

Hi All, I am a Speex novice but trying to use Speex in my Doctoral thesis. I installed Speex, was able to encode and decode files. Right now, I want to write my own cpp program that handle Speex-files. I tried to compile and run the two sample programs that come with the documentation (after having insalled the speex-devel-1.0.5-1.i386.rpm). However, when I compile. gcc sampleenc.cc Here is

Hi AlL, I ahve this problem that my objective function is discontinous in the paramaters and I need to use methods such as nelder-mead to get around this. My question is: How do i compute standard errors to a problem that does not have a gradient? Any literature on this is greatly appreciated. Jean,

> Hello, Hi :) First off, could you try to set your email client to break long lines before transmitting? In my (somewhat outdated) pine, the lines appear as VERY long lines when I try to reply, making it hard to read :) Minor detail though, I should probably fix pine. Some day. > The way you describe how the jitter buffer should be implemented makes me > wonder: How does the

Dear R users. I have data observed on the surface of a torus, and am trying to fit the nonlinear regression using the geodesic equation on a torus. Could anyone give me a helpful advise on this problem? I would definitely appreicate your reply. Sincerely, SUNGSU KIM [[alternative HTML version deleted]]

Hi, I'm trying to fit some data using a logistic function defined as y ~ a * (1+m*exp(-x/tau)) / (1+n*exp(-x/tau) My data is below: x <- 1:100 y <- c(0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0, 0,0,0,0,0,1,1,1,2,2,2,2,2,3,4,4,4,5, 5,5,5,6,6,6,6,6,8,8,9,9,10,13,14,16,19,21, 24,28,33,40,42,44,50,54,69,70,93,96,110,127,127,141,157,169,

> Re: [Speex-dev] Please help, very important (not for fun) I usually only help people who have fun doing what they're doing, but I'll try and make an exception here... suhaib obeidat a ?crit : > gcc sampleenc.cc > > Here is what I get: > > /tmp/ccXKa3LF.o: In function `main': > sampleenc.cc:(.text+0x1a): undefined reference to `speex_nb_mode' >

Hope someone can help me here. I have a daily time series, say 2003-02-01 2003-02-03 2003-02-07 2003-02-09 2003-02-14 .......... 2004-02-01 2004-02-04 0.4914798 -1.1857653 -1.6982844 -0.3559572 -0.2333087 ........... 0.44553 -0.45222 I need to calculate the statistics for the overlapping rolling yearly window with rolling step as 1 day so for each of the intervals: (2003-02-01 ~

Which speed recognition products will also recognize DTMF? In other words, I want to say "Please speak or dial the conference number". Does Vestec allow that? LumenVox? Any other way?

Hello, I am quite new to Speech. Can you please let me know whether Speex speech coding is done in Time or Frequency Domain? Regards, Vijay Send instant messages to your online friends http://uk.messenger.yahoo.com

I had been using google tts, but it started requiring a captcha for my browser, and via linux I can't access http://translate.google.com/translate_tts?q=test (redirects to captcha) as so, its not reliable On 27 August 2015 at 17:16, Carlos Chavez <cursor at telecomabmex.com> wrote: > On 8/26/15 1:15 PM, Tech Support wrote: > > All; > > I have a customer who is

Hi, I want to detect speech locations from raw WAV file. For example a 10 sec WAV file contains silence/noise for 1st sec, speech for next 4 seconds, then silence/noise for next 2 seconds and speech for next 3 seconds. I want to generate an information that this WAV file has speech from 1-5 seconds and from 8-10 seconds. How I can do that by using preprocessor utility of libspeexdsp? With Best

I was wondering how vorbis fares as a speech codec? I mean can it give similar or better quality as GSM/Toast at the same kind of bitrates/filesize? I'm not worried about streaming but I'd love to be able to create really small voice files. Can anyone give me an idea of filesize for recording time for pure speech based use? love Freya __________________________________________________

Hi Iam a newbie to NLP and I would like to get some directions on how to convert speech file to text Google search leads me to using GoogleLanguageR Package and API's but these need payments to be made for Google. Can someone suggest ways in which I can do the speech to text conversion in R studio ? Thanks for your help and time Regards Gayathri [[alternative HTML version deleted]]

Hello, I want information about the behavior of the --speech option in the opusenc program in opus tools package. The documentation only tells that it optimizes for speech, but what does this mean in terms of sampling frequency, bitrate, etc.? Thank you very much. Nicanor Garcia -------------- next part -------------- An HTML attachment was scrubbed... URL:

HI, I am new in speex. How can I use speex to implement the online voice chat? Can we stream the voice? Thanks Alex ----- Original Message ----- From: "Jean-Marc Valin" <jean-marc.valin@usherbrooke.ca> To: "sharanabasava matmari" <sharanabasavam@yahoo.com> Cc: <speex-dev@xiph.org> Sent: Monday, May 21, 2007 6:13 AM Subject: Re: [Speex-dev] Speex speech

I want to try using google for speech recognition in Asterisk and I found a ready made AGI: http://zaf.github.io/asterisk-speech-recog/ I have followed all the steps listed in the web site but I keep getting this error: <PJSIP/2001-0000006e>AGI Tx >> 200 result=99981 (timeout) endpos=22720 <PJSIP/2001-0000006e>AGI Rx << VERBOSE "Unable to get recognition

Dear All Can you let me know is this possible to if we are using Asterisk version 1.4 or 1.6 for incoming voicemail we can send as email in text formta. Means voice mesage converted into text message & send it to resp. email ids. is this possible. If yes. we can do the same with help of Asterisk or we require expertnal application need to isntall/integrate to work for speech to test. Please

Hi, I have a question regarding the stereo capabilities of Opus. I would like to establish a connection between two ends via Wi-Fi and the signals that are to be transmitted are of speech kind. It mean on both ends speech is both recorded and played back as stereo. Now would the delay and loss characteristics of the speech transmission at a certain bitrate be the same as Mono voice transmission?