similar to: encoding lots of speech

I really don't know if this is the same problem that was reported by Ingo Saitz (I really couldn't say which one was 128kbit and which was the original wav when blind-testing. The original had some distortion that perhaps does somehting with my cheapo soundcard) but here's a description of what i've found: When doing some sample encoding with rc2 (rebuilt rpm with latest redhat

For my own needs I have created the .spec file needed for rpm packaging of the flac package. End users --------- If anyone is interested in the resulting rpm-files i have made them available on http://noa.tm/flac-rpms/ FLAC maintainers ---------------- I have attached the specfile along with hooks in Makefile.am and configure.in in unified diff format so that they are ready for inclusion in

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

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

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

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

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

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?

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:

Echo cancellation works like a charm, but it seems to confuse the preprocessor a bit. If listening to background music (properly fed through the echo cancellator), the music is removed but the result is still detected as speech even if almost silence remains in the signal. Also, the AGC keeps adjusting to the minute remains in the signal, meaning that sooner or later it will amplify the

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 Everyone, Has anyone got any opensource speech recognition software to work with Asterisk? Please only list WORKING ones. Not the "theoretically" should work ones! Thanks -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20100821/4d11d6c0/attachment.htm

Thanks Jurijs, Yes, in fact I'm already using that, and it works fine. The problem here is that I cannot find a way of recording speech AND listening for a DTMF digit being pressed as an alternative. That's where the problem lies. J.

Have you checked here: https://cran.r-project.org/web/views/NaturalLanguageProcessing.html Speech to text is a very complex, specialized task requiring, I would expect, a lot of IP. It would not surprise me if you have to resort to big time, specialized software products with or without R. "Speech to text with R" found some hits, but mostly the google package you refer to. You might do

Hi, Definitely! Since I have a glance and seems to me that speex is used for static compression but not for streaming. Can you suggest me which part should I read carefully? There are so many parts inside the doc. Many thanks Alex ----- Original Message ----- From: "Jean-Marc Valin" <jean-marc.valin@usherbrooke.ca> To: "ALEX NG" <alex@gorex.com.hk> Cc: