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<p class="MsoNormal"><span style="font-size:11.0pt;font-family:&quot;Calibri&quot;,&quot;sans-serif&quot;;color:#1F497D">Thank you Benjamin, a few follow-up question please:
<o:p></o:p></span></p>
<p class="MsoNormal"><span style="font-size:11.0pt;font-family:&quot;Calibri&quot;,&quot;sans-serif&quot;;color:#1F497D"><o:p>&nbsp;</o:p></span></p>
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<![if !supportLists]><span style="font-size:11.0pt;font-family:Symbol;color:#1F497D"><span style="mso-list:Ignore">&middot;<span style="font:7.0pt &quot;Times New Roman&quot;">&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;
</span></span></span><![endif]><span style="font-size:11.0pt;font-family:&quot;Calibri&quot;,&quot;sans-serif&quot;;color:#1F497D">Is &#8220;packet&#8221; the right word to be using in these discussions?&nbsp; Perhaps I&#8217;m wrong, but I always thought of a packet as a bundle of information on the
 wire.&nbsp; If so, a copy of the previous packet in the current might contain many &#8220;segments&#8221; of audio (because the previous packet itself may have contained redundancy).&nbsp; If there is a better term (eg &#8220;segments&#8221;), please let me know.&nbsp; If not I&#8217;ll continue using
 &#8220;packet&#8221;. <o:p></o:p></span></p>
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<![if !supportLists]><span style="font-size:11.0pt;font-family:Symbol;color:#1F497D"><span style="mso-list:Ignore">&middot;<span style="font:7.0pt &quot;Times New Roman&quot;">&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;
</span></span></span><![endif]><span style="font-size:11.0pt;font-family:&quot;Calibri&quot;,&quot;sans-serif&quot;;color:#1F497D">Is the decision to copy information from N-1 into N binary?&nbsp; In other words, is it ever the case that a partial copy occurs, or is it all or nothing
 decision?<o:p></o:p></span></p>
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<![if !supportLists]><span style="font-size:11.0pt;font-family:Symbol;color:#1F497D"><span style="mso-list:Ignore">&middot;<span style="font:7.0pt &quot;Times New Roman&quot;">&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;
</span></span></span><![endif]><span style="font-size:11.0pt;font-family:&quot;Calibri&quot;,&quot;sans-serif&quot;;color:#1F497D">Does packet N ever contain information from more than one previous packets (eg N-2), or is it always a single?<o:p></o:p></span></p>
<p class="MsoListParagraph" style="margin-left:23.0pt;text-indent:-.25in;mso-list:l0 level1 lfo1">
<![if !supportLists]><span style="font-size:11.0pt;font-family:Symbol;color:#1F497D"><span style="mso-list:Ignore">&middot;<span style="font:7.0pt &quot;Times New Roman&quot;">&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;
</span></span></span><![endif]><span style="font-size:11.0pt;font-family:&quot;Calibri&quot;,&quot;sans-serif&quot;;color:#1F497D">What is the limit of the redundancy provided by this feature?&nbsp; Is it simply that every packet contains the previous packet&#8217;s data, or does it go further
 than this by sending duplicate copies of each individual packet (which themselves contain the previous packet data).<o:p></o:p></span></p>
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<![if !supportLists]><span style="font-size:11.0pt;font-family:Symbol;color:#1F497D"><span style="mso-list:Ignore">&middot;<span style="font:7.0pt &quot;Times New Roman&quot;">&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;
</span></span></span><![endif]><span style="font-size:11.0pt;font-family:&quot;Calibri&quot;,&quot;sans-serif&quot;;color:#1F497D">Continuing on the above, how does one trigger that maximum mode.&nbsp; Perhaps one should set expect-loss to 100%, but that has some odd implications.&nbsp;
 If all packets will be dropped, the situation seems hopeless </span><span style="font-size:11.0pt;font-family:Wingdings;color:#1F497D">J</span><span style="font-size:11.0pt;font-family:&quot;Calibri&quot;,&quot;sans-serif&quot;;color:#1F497D">.<o:p></o:p></span></p>
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<![if !supportLists]><span style="font-size:11.0pt;font-family:Symbol;color:#1F497D"><span style="mso-list:Ignore">&middot;<span style="font:7.0pt &quot;Times New Roman&quot;">&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;
</span></span></span><![endif]><span style="font-size:11.0pt;font-family:&quot;Calibri&quot;,&quot;sans-serif&quot;;color:#1F497D">When setting these parameters, any advice for Ethernet frame sizes?&nbsp; I&#8217;m not much of a networking guy, but it seems that the risk of failure would
 climb dramatically if a packet spanned multiple frames.<o:p></o:p></span></p>
<p class="MsoNormal"><span style="font-size:11.0pt;font-family:&quot;Calibri&quot;,&quot;sans-serif&quot;;color:#1F497D"><o:p>&nbsp;</o:p></span></p>
<p class="MsoNormal"><span style="font-size:11.0pt;font-family:&quot;Calibri&quot;,&quot;sans-serif&quot;;color:#1F497D">Regards<o:p></o:p></span></p>
<p class="MsoNormal"><span style="font-size:11.0pt;font-family:&quot;Calibri&quot;,&quot;sans-serif&quot;;color:#1F497D"><o:p>&nbsp;</o:p></span></p>
<p class="MsoNormal"><span style="font-size:11.0pt;font-family:&quot;Calibri&quot;,&quot;sans-serif&quot;;color:#1F497D"><o:p>&nbsp;</o:p></span></p>
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<p class="MsoNormal"><b><span style="font-size:10.0pt;font-family:&quot;Tahoma&quot;,&quot;sans-serif&quot;">From:</span></b><span style="font-size:10.0pt;font-family:&quot;Tahoma&quot;,&quot;sans-serif&quot;"> Benjamin Schwartz [mailto:benjamin.m.schwartz@gmail.com]
<br>
<b>Sent:</b> Friday, November 30, 2012 8:41 AM<br>
<b>To:</b> Young, Milan<br>
<b>Cc:</b> opus@xiph.org<br>
<b>Subject:</b> Re: [opus] Opus for ASR - update and questions<o:p></o:p></span></p>
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<p class="MsoNormal"><o:p>&nbsp;</o:p></p>
<p class="MsoNormal">The Low Bit-Rate Redundancy (LBRR) feature works by sometimes adding another copy of packet N-1 onto packet N. &nbsp;This means that a traditional decoder that wishes to use the LBRR info must increase its jitter buffer depth, and hence latency,
 by one packet, so that if packet K is dropped, there is time to receive and decode packet K&#43;1. &nbsp;However, in the case of ASR I think this logic does not apply, and the latency penalty must only be paid in the rare case when the last packet before a data return
 event is dropped.<o:p></o:p></p>
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<p class="MsoNormal"><o:p>&nbsp;</o:p></p>
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<p class="MsoNormal">Expect-loss is not a simulation of loss. &nbsp;It is a way to tell the encoder &quot;I expect that my network will drop X% of packets&quot;. &nbsp;The higher the value of X, the more the encoder will spend bits to avoid depending on previous packets that may
 not have arrived. &nbsp;This means increasing the use of LBRR, decreasing the use of sensitive long-term filters, and many other changes in encoding strategy.<o:p></o:p></p>
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<p class="MsoNormal" style="margin-bottom:12.0pt"><o:p>&nbsp;</o:p></p>
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<p class="MsoNormal">On Wed, Nov 28, 2012 at 3:50 PM, Young, Milan &lt;<a href="mailto:Milan.Young@nuance.com" target="_blank">Milan.Young@nuance.com</a>&gt; wrote:<o:p></o:p></p>
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<p class="MsoNormal" style="mso-margin-top-alt:auto;mso-margin-bottom-alt:auto">For the last couple months, Nuance has performed extensive testing on how the Opus codec performs in the speech recognition task.&nbsp; I&#8217;m hoping to publish a full report in the coming
 months, but until then all I have is a teaser.&nbsp; Opus performed within about 1% of the WER (Word Error Rate) of unencoded audio.&nbsp; This is compared to about 5% for Speex, which was the previous codec of choice.&nbsp; Well done to you all!<o:p></o:p></p>
<p class="MsoNormal" style="mso-margin-top-alt:auto;mso-margin-bottom-alt:auto">&nbsp;<o:p></o:p></p>
<p class="MsoNormal" style="mso-margin-top-alt:auto;mso-margin-bottom-alt:auto">As Nuance considers migrating to Opus, we&#8217;d like to consider the topic of transport.&nbsp; Traditionally we&#8217;ve relied on TCP for reasons of reliability.&nbsp; Opus, with its packet redundancy
 features, offers an attractive real-time alternative that we will soon be testing.&nbsp; But in order to apply an apples-apples comparison we need to model both data rates and latency in real world scenarios.<o:p></o:p></p>
<p class="MsoNormal" style="mso-margin-top-alt:auto;mso-margin-bottom-alt:auto">&nbsp;<o:p></o:p></p>
<p class="MsoNormal" style="mso-margin-top-alt:auto;mso-margin-bottom-alt:auto">For UDP, I&#8217;m assuming that the redundancy feature adds no additional latency.&nbsp; Correct?&nbsp; On the data rate question, I see that the Opusenc tool provides an &#8220;expec-loss&#8221; parameter
 with the value expressed as a percentage.&nbsp; Could someone please describe how this is implemented?&nbsp; Are you simply removing some percentage of packets from the result, or is there a more complex model underpinning the exercise?<o:p></o:p></p>
<p class="MsoNormal" style="mso-margin-top-alt:auto;mso-margin-bottom-alt:auto">&nbsp;<o:p></o:p></p>
<p class="MsoNormal" style="mso-margin-top-alt:auto;mso-margin-bottom-alt:auto">Modeling TCP data rates and latency in similarly losssy scenarios seems much more difficult since dropped packets have cascading effects.&nbsp; Has anyone on this list considered this
 class of comparison?&nbsp; Any suggestions for modeling software that could aid my search?<o:p></o:p></p>
<p class="MsoNormal" style="mso-margin-top-alt:auto;mso-margin-bottom-alt:auto">&nbsp;<o:p></o:p></p>
<p class="MsoNormal" style="mso-margin-top-alt:auto;mso-margin-bottom-alt:auto">Thank you<o:p></o:p></p>
<p class="MsoNormal" style="mso-margin-top-alt:auto;mso-margin-bottom-alt:auto">&nbsp;<o:p></o:p></p>
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_______________________________________________<br>
opus mailing list<br>
<a href="mailto:opus@xiph.org">opus@xiph.org</a><br>
<a href="http://lists.xiph.org/mailman/listinfo/opus" target="_blank">http://lists.xiph.org/mailman/listinfo/opus</a><o:p></o:p></p>
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<p class="MsoNormal"><o:p>&nbsp;</o:p></p>
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