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<p class="MsoNormal"><span lang="EN-US">Hi all,<o:p></o:p></span></p>
<p class="MsoNormal"><span lang="EN-US"><o:p> </o:p></span></p>
<p class="MsoNormal"><span lang="EN-US">I am developing a VoIP application and have used SPEEX for that purpose; I successfully managed to port the codec in fixed-point to an ADSP-21364 processor, and the codec works fine in narrowband, wideband and ultra-wideband
mode. Currently, I have chosen, in accordance with the speex manual, a 20 ms frame size in each of these modes: in other words, the input buffer for speex was set to respectively 160 – 320 – 640 samples. Now, I was wondering whether it is possible to reduce
this buffer size in order to reduce the latency of the codec. Is this actually possible, or should I leave it as is? In addition, I have another question regarding packet loss concealment. In the manual, I found the encoder can be optimized for a certain percentage
of packet loss by means of the <strong><span style="font-family:"Calibri","sans-serif"">SPEEX_SET_PLC_TUNING
</span></strong><strong><span style="font-family:"Calibri","sans-serif";font-weight:normal">parameter. What exactly does that optimization mean? Does it mean the speex decoder will use some kind of packet loss concealment algorithm to compensate for lost or
late packets? And if so, what kind of packet loss concealment scheme does speex use?
<o:p></o:p></span></strong></span></p>
<p class="MsoNormal"><strong><span lang="EN-US" style="font-family:"Calibri","sans-serif";font-weight:normal"><o:p> </o:p></span></strong></p>
<p class="MsoNormal"><strong><span lang="EN-US" style="font-family:"Calibri","sans-serif";font-weight:normal">Best regards,<o:p></o:p></span></strong></p>
<p class="MsoNormal"><strong><span lang="EN-US" style="font-family:"Calibri","sans-serif";font-weight:normal"><o:p> </o:p></span></strong></p>
<p class="MsoNormal"><strong><span lang="EN-US" style="font-family:"Calibri","sans-serif";font-weight:normal">Peter</span></strong><span lang="EN-US"><o:p></o:p></span></p>
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