Re: [codec] [opus] Opus for ASR - update and questions

"Hoene" <hoene@uni-tuebingen.de> Thu, 29 November 2012 16:41 UTC

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From: Hoene <hoene@uni-tuebingen.de>
To: "'Young, Milan'" <Milan.Young@nuance.com>, opus@xiph.org, codec@ietf.org
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Date: Thu, 29 Nov 2012 08:41:18 -0800
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Subject: Re: [codec] [opus] Opus for ASR - update and questions
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Hello Milan Young,

 

thanks for doing the ASR testing.

We would be happy to consider your results to be in included in the Opus
characterization draft.

Please feel free to post them to the Opus IETF mailing list, too.

 

With best regards,

 

Christian Hoene

 

 

--

Universität Tübingen, Sand 13, 72076 Tübingen, Germany

Tel +49 7071 2970532, Fax +49 7071 5220

http://kn.inf.uni-tuebingen.de/staff/hoene.html

 

Von: opus-bounces@xiph.org [mailto:opus-bounces@xiph.org] Im Auftrag von
Young, Milan
Gesendet: Mittwoch, 28. November 2012 12:51
An: opus@xiph.org
Betreff: [opus] Opus for ASR - update and questions

 

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 done to you all!

 

As Nuance considers migrating to Opus, we’d like to consider the topic of
transport.  Traditionally we’ve relied on TCP for reasons of reliability.
Opus, with its packet redundancy features, offers an attractive real-time
alternative that we will soon be testing.  But in order to apply an
apples-apples comparison we need to model both data rates and latency in
real world scenarios.

 

For UDP, I’m assuming that the redundancy feature adds no additional
latency.  Correct?  On the data rate question, I see that the Opusenc tool
provides an “expec-loss” parameter with the value expressed as a percentage.
Could someone please describe how this is implemented?  Are you simply
removing some percentage of packets from the result, or is there a more
complex model underpinning the exercise?

 

Modeling TCP data rates and latency in similarly losssy scenarios seems much
more difficult since dropped packets have cascading effects.  Has anyone on
this list considered this class of comparison?  Any suggestions for modeling
software that could aid my search?

 

Thank you