similar to: Opus for ASR

For the last couple months, Nuance has performed extensive testing on how the Opus codec performs in the speech recognition task. I'm hoping to publish a full report in the coming months, but until then all I have is a teaser. Opus performed within about 1% of the WER (Word Error Rate) of unencoded audio. This is compared to about 5% for Speex, which was the previous codec of choice. Well

Hello Luca, Thank you for your response. I?m familiar with speech recognition and TTS, but new to MRCP. Yes, the 100k options is used for names in a directory listing. In the pre-MRCP support, Nuance ASR used API events/methods for the application to tell ASR when the prompt was playing and when it stopped. If ASR detected speech, it would signal an event so we would stop playing the prompt.

Is it correct that the unimrcp is the best approach for Asterisk and ASR/TTS? Could anyone provide pros/cons for the various ASR options for Asterisk? We need the ability for very large grammars (over 100,000 options). Because of this, my initial thought is Nuance or Lumenvox. Does this sound correct? Have a great day! Dan -------------- next part -------------- An HTML attachment was

Hello, Question for developers: what is the asterisk way to integrate with ASR (speech recognition)? Question to the community: has someone done anything in this direction? On the first glance that shouldn't be too hard. One part is delivering audio to the engine (for example, main ASR players Nuance and Speechworks will be happy with RTP) - can be done via RTP forking. The other part is

Hello, Is there anyone who has ever done a setup of VoiceXML combined with some licenses from Nuance for the ASR/TTS engine within Asterisk ? I'm currently working with VoiceGenie, for the VoiceXML + ASR/TTS engine, but we are having a couple of issues which I guess are caused by VoiceGenie. If there's an alternative, it would be very interesting for us. Thanks, -- Eric Rousse

Hello, I'm in the early stages of designing an Emergency calling service IVR application. The IVR application asks simple one or two questions like "which is the postal code of the area you are currently calling from ?" "Is the correct ?". The expected values are a 5-digits number like "twenty-five-thousand-two-hundreds-twelve" or

Hi all Arnaud, Some newly purchased Ellipse ASR-model UPS find their way to my office for a first evaluation test. It concerns the Ellipse 1500 and Ellipse 1000 models. As you know we are still working with hidups. Peripheral recognition, start of NUT, mains short interruptions management, all that work fine... as expected ;-) But changes occured on the delayed shutdown procedure after the

Is there any program Asterisk users use to calculate ASR and ACD ?? Thanks for any comments. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20090910/af1f9656/attachment.htm

Is there a ready add-on to asterisk that will display the ACD/ASR per channel, source & destination? Thanks. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20091014/48076c6b/attachment.htm

Hello, I have been working on designs for two different projects, where both of them would need to use the IBM Watson streaming ASR service. Based on our discussion at AstriDevCon, I know there is currently no support for that. However, there may be some workarounds I am not aware of. Would it be possible to write out the audio frames as they get recorded? Watson supports 16 bit signed little

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

Matt Riddell wrote: > >> On 17/10/2016, at 3:43 PM, Luca Pradovera <luca.pradovera at gmail.com >> <mailto:luca.pradovera at gmail.com>> wrote: >> >> I have been working on designs for two different projects, where both >> of them would need to use the IBM Watson streaming ASR service. >> >> Would it be possible to write out the audio frames

Hello everybody, Our supoprt team is facing a problem which froze the CPU. They suspect that this happens after an automatic battery test, which lets the battery goes down to 30% and then stop the test. At this level of charge, our kernel also triggers a shutdown (CPU and UPS). I saw (in the doc found here : tools.*mge*ops.net/download/intl/products/*ellipse*-asr/28-umeasr_08.pdf) that this

Hello, I have Ellipse ASR 1500 UPS and now I'm trying to monutor it with NUT from FreeBSD. It seems that there are some problems with this UPS: =================================== ugen0: MGE UPS SYSTEMS ELLIPSE, rev 1.10/42.41, addr 2 > usbdevs -v Controller /dev/usb0: addr 1: full speed, self powered, config 1, OHCI root hub(0x0000), NEC(0x0000), rev 1.00 port 1 powered port 2

Hello guys, I have to manage an Eaton Ellipse ASR 750 VA UPS under a FreeBSD 8.2 server. I used nut from ports collection but I'm unable to detect the UPS by USB cable. Here is the output of 'usbconfig -u 4 -a 2 dump_device_desc' : ugen4.2: <ELLIPSE EATON> at usbus4, cfg=0 md=HOST spd=LOW (1.5Mbps) pwr=ON bLength = 0x0012 bDescriptorType = 0x0001 bcdUSB = 0x0110

Hello, I understand the encoder provides an option for FEC to provide some protection against packet loss, but I don't understand the details of this arrangement. I'd appreciate answers to the following: * Adding FEC seems to change the encoded audio bit-stream itself, i.e., it doesn't just add additional protection bits, but also changes the encoded bits. This is easy to show by

Hi Rhishi, PQevalaudio is very unreliable and buggy. I have compared to PEAQ and - as a result - now I am not using it anymore. With best regards, Christian Hoene Von: opus-bounces at xiph.org [mailto:opus-bounces at xiph.org] Im Auftrag von Rhishikesh Agashe Gesendet: Freitag, 4. Oktober 2013 12:35 An: opus at xiph.org Cc: Rasmi Mishra Betreff: [opus] ODG (Objective Difference

Greetings list, I am working on a project on which we wish to use Speex with Google Automatic Speech Recognition (ASR) to transcribe Speex audio being sent on to Google ASR service and return us the text of the spoken audio in the Speex audio stream. However, Google ASR's Speex support requires the off-standard Speex-with-header-byte format, and my group cannot find any worthwhile

Benjamin, Thanks for the prompt response. Are there other recommended methods to verify encoder implementations? Regards, Chris From: benjamin.m.schwartz at gmail.com [mailto:benjamin.m.schwartz at gmail.com] On Behalf Of Benjamin Schwartz Sent: Sunday, November 10, 2013 1:53 PM To: Wang, Chris Cc: opus at xiph.org Subject: Re: [opus] Questions Regarding Opus Test Vectors On Sun, Nov 10,

Greetings list, I am working on a project on which we wish to use Speex with Google Automatic Speech Recognition (ASR) to transcribe Speex audio being sent on to Google ASR service and return us the text of the spoken audio in the Speex audio stream. However, Google ASR's Speex support requires the off-standard Speex-with-header-byte format, and my group cannot find any worthwhile