search for: speech

...MF 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: https://wiki.asterisk.org/wiki/display/AST/Asterisk+15+Application_SpeechBackground So then there's https://wiki.asterisk.org/wiki/display/AST/Asterisk+15+Application_Record, which can terminate on any DTMF key with "y", but according to the docs, "RECORD_STATUS" only sets a flag of "DTMF" (A terminating DTMF was received ('#' o...

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/aste...

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...

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 up in encoding. Chandrakala, are these the encoding parame...

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 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&...

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?

...tream 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 coding is done in Time or FrequencyDomain? > Speex is time domain, just like pretty much every other speech codec. > > Jean-Marc > > sharanabasava matmari a ?crit : >> Hello, >> I am quite new to Speech. >> Can you please let me know whether Speex speech >&g...

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

...g 2015, Tiago Geada wrote: > > 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) >> > > I'm confused. Your subject says 'speech to text' but the URL you reference > does 'text to speech.' > > 'speech to text' is where your caller speaks and Asterisk gets the text. > > 'text to speech' is where Asterisk (your dialplan) has a string of text > and your caller hears the spoken words...

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 de...

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 F...

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? Is it at all possible to use Opus for stereo Speech( considering acceptable latency and...

...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<-Ter...

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...

...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 looking for a good speech to text solution, > either open source or reasonably priced commercial product, I?m open to > suggestions. > > Thanks; > > John V > > > > For a commercial option try Lumenvox, had very good results. For "free" > you can try google tts but you never know w...

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: https://raw.github.com/zaf/asterisk-speech-recog/master/README The script is available here: https://github.com/zaf/asterisk-speech-recog The code i...

...o: opus at xiph.org Sent: Tuesday, November 21, 2017 2:38:16 AM Subject: Re: [opus] Reg an issue with smoothing factor in VAD implementation 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 up in encoding. Chandrakala, are these the en...

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...