similar to: Speex speech coding is done in Time or Frequency Domain?

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, 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:

Hi Alex, How about you start by actually reading the manual ( http://www.speex.org/docs/ )? If you have any remaining questions after that, you can post them on the list. Jean-Marc ALEX NG a ?crit : > 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:

Dear all, I recently saw a graph on television that displayed selected words/phrases in a speech scaled in size according to their frequency. So words/phrases that were often used appeared large and words that were rarely used appeared small. The closest thing I can find on the web to approximate what I saw can be found here: http://stateoftheunion.onetwothree.net/ The example at that website

> Does anyone have an idea about the possibility to apply psychoacoustic > models as the ones in mp3 or AAC to a CELP coder? Thanks! > /Pontus This is (sort of) done in the decoder with the optional perceptual filtering. Speex tries to shape the noise so that it sounds more pleasant, I believe. To really use perceptual coding, one would require fine granular control over quantization

Hi, In order to make sure the Speex speech codec (http://speex.sourceforge.net/) is adapted to the needs of the greatest audience possible and because our project is in some way related to Vorbis, you are invited to take a few minutes to fill the following survey: https://sourceforge.net/survey/survey.php?group_id=46651&survey_id=13035 We would like to know what your needs are in terms of

Hi, Does anyone have an idea about the possibility to apply psychoacoustic models as the ones in mp3 or AAC to a CELP coder? Thanks! /Pontus <p>--- >8 ---- List archives: http://www.xiph.org/archives/ Ogg project homepage: http://www.xiph.org/ogg/ To unsubscribe from this list, send a message to 'speex-dev-request@xiph.org' containing only the word 'unsubscribe' in the

(This is almost out of topic but anyway...) It is surprising how little research effort have been put into psy-acou models for CELP. The basic problem lies in that it is not easy to alter the LP model without distroying the minimum-phase property (ie. the stability of the predictor). That leaves us with psy-acou modelling of the noise-part only. However, my own research is in constrained

Greetings list, What are people's experiences with WiFi SIP phones? When I last looked into them about 18 months ago, they were incredibly expensive, had very limited range and poor battery life. In the end, it worked out much more cost effective to simply use ATAs + DECT cordless phones where there was a requirement for portable devices. I assume things must have moved on somewhat since

Hi, We would like to announce the first release of the Speex project. Speex (http://speex.sourceforge.net) is an open-source (LGPL), patent-free compression format allowing an alternative to expensive proprietary codecs. Unlike Ogg Vorbis which compresses general audio, Speex is designed especially for speech. For that reason, Speex is meant to be a complement to Vorbis. Since it is specialized

> On make check the bit intrinsics test is still failing but it works from the > command line, strange ? Did Chris help you with this or is it still an issue? > What files do you want from the nightly report, as I am not familiar with it > ? in llvm-test, if you run the gmake TEST=nightly report command it will generated a report.nightly.txt file. Thats what I want to see. Thanks!

Hi, a set of AGI scripts that provide ASR and TTS for asterisk using the iSpeech API (http://www.ispeech.org/) are available on this page: http://zaf.github.io/asterisk-ispeech/ This is the first public release, updates will soon follow. Feel free to test and report. Regards, Lefteris Zafiris -------------- next part -------------- An HTML attachment was scrubbed... URL:

Just and FYI. In RPM form, works on 4.x and I will try on 5.x too. Speech is tacky, but if I can get something but the Gnome Festival synthesizer, maybe it will improve? http://www.adobe.com/products/acrobat/readstep2.html?promoid=BONRM -- Bill

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Hi! I need to use text to speech, what is the best application? Thanks! -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20070829/bc69eb9d/attachment.htm

Version 2.1 of app-espeak and app-flite modules just got released. Changes include voice selection for flite, support for 16kHz file playback in asterisk 1.6.x, use of libsamplerate for faster resampling, better error handling and cleaner code. You can get them here: Asterisk espeak module: http://zaf.github.com/Asterisk-eSpeak/ Asterisk flite module: http://zaf.github.com/Asterisk-Flite/

Hi I want to can we use asterisk as TTS server. Which can support mrcpv2 and ssml. Im looking for tts server with above requirement will asterisk 1.8 is useful for me. Any configuration available. Any opensource tts available. Amit-- -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi, Hoping someone can help with this most frustrating situation. I have a Linksys PAP2T registering with ADSL to my asterisk server which also sits behind a Mikrotik router. Thanks Shaun -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20080821/3d47a67a/attachment.htm

Hello All, I was wondering if anyone's tried to use OR currently use the Microsoft Speech Server or their UCMA 3.x SDK etc. as their ASR/TTS backend/engines etc. If yes, then what's their experience? Please Note, this does NOT need to be integrated with Asterisk ala MRCP or some module/plugin etc. I just wanted to know if someone's used it and and what their experience has been in

Hi, I need to plot a spectogram for a speech signal. Is there a package that can do this? -- Rajesh.J [[alternative HTML version deleted]]