search for: speeche

Briefly: I want to be able to have "press or say (number)", with Asterisk listening for a spoken number, but accepting a DTMF digit, too. I'm posting everything I found so far, here, partly to show working, but also in case anyone else finds it useful. So, moving on.... This looked hopeful for a moment until I realised that it doesn't do DTMF:

Does anyone know of a speech recognition module (like say yes or no, or numbers) I guess due to the complexity of speech recognition it might just be found in commercial applications or am I wrong like always? -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20050128/119168cb/attachment.htm

Dear All, I am trying to send speech across a wireless link. I want to change which speech compression/rate to use according to some algorithm that I have. To be more specific, I have a speech file. In real time, I want to keep changing at which rate speech will be sent next. Let us say, I am considering three rates. I will take the original speech file, compress it using the three rates, which

Just for fun, I tried to reproduce such an overflow. I turned on all debug macros, assertions, and checked arithmetic and then encoded 2 hours of mixed speech/audio with these parameters: Sample rate = 48000 Channels = 1 Application = OPUS_APPLICATION_AUDIO Bitrate = 24 KB/s Force Mode = MODE_SILK_ONLY Signal Type = OPUS_SIGNAL_AUTO Complexity = 10 Frame size = 480 samples (10ms) No errors came

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 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

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

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?

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

Hi, I'm using Cepstral as a TTS Engine for Asterisk with Swift application. It works fine when I have just 1 voice installed. Now I have 2 voices in the same language installed but I can't seem to find the way to select which voice to use in Swift's application in Asterisk. Does anyone know?? Thank you, Pim

I have a similar situation here, I want to include TTS in my asterisk IVR system. Could someone give suggestion(s) please, I prefer open-source thanks in advance! Chatila, A. C. P. O. Box 365, Kihesa Street, Njombe, Tanzania. *Mob:* +255 765 154 235 *Whatsapp:* +255 653 258 608 *Website:* chax.me.tz On Thu, Aug 27, 2015 at 9:07 PM, Steve Edwards <asterisk.org at sedwards.com> wrote:

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]]

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, 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?

...f length zero In addition: Warning message: package 'rJava' was built under R version 2.9.1 Error : .onLoad failed in 'loadNamespace' for 'rJava' Error: package/namespace load failed for 'rJava' > > #Set documents directory > DIR <- "G:/TextSearch/Speeches" > > #Load corpus > speech <- Corpus(DirSource(DIR), readerControl = list(reader = readPlain, + language = "en_US", load = TRUE)) > > #Remove stopwords > speech <- tmMap(speech, stripWhitespace) > speech A corpus with 2 text documents > tdm<-Term...

I have RSI (hurty hands) and need software for recognition of continuous speech. Linux has nothing workable. Dragon was bad even on Windows. What's needed is a Linux speech engine, so that the linux crew can get making a good GUI. So I'm trying to bring over the Vista Speech Engine. Here's my (failed) attempt: http://womblezone.blogspot.com/ I'm trying to bring it over with

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

Hello, I have written an agi script that uses google voice API for voice recognition. The script records from the current channel untill the pound key (#) is pressed or the timeout (15 seconds) is reached. The recording is send over to google speech recognition service and the returned text string is assigned to a channel variable. More info and dialplan examples can be found in the README file:

Hi, Can anyone let me know if this is a bug? Thank you, Chandrakala ----- Original Message ----- From: "Logan Stromberg" <loganstromberg at gmail.com> To: "Chandrakala Madhira" <chandrakala.madhira at soctronics.com> Cc: opus at xiph.org Sent: Wednesday, November 22, 2017 12:12:39 PM Subject: Re: [opus] Reg an issue with smoothing factor in VAD

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: