<html><head><meta http-equiv="Content-Type" content="text/html charset=windows-1252"></head><body style="word-wrap: break-word; -webkit-nbsp-mode: space; -webkit-line-break: after-white-space;">On Sep 5, 2014, at 10:33 AM, Matthew Jordan <<a href="mailto:mjordan@digium.com">mjordan@digium.com</a>> wrote:<br><div><br class="Apple-interchange-newline"><blockquote type="cite"><div dir="ltr"><br><div class="gmail_extra"><br><br><div class="gmail_quote">On Fri, Aug 29, 2014 at 2:52 AM, Ben Merrills <span dir="ltr"><<a href="mailto:b.merrills@mersontech.co.uk" target="_blank">b.merrills@mersontech.co.uk</a>></span> wrote:<br><blockquote class="gmail_quote" style="margin: 0px 0px 0px 0.8ex; border-left-width: 1px; border-left-color: rgb(204, 204, 204); border-left-style: solid; padding-left: 1ex; position: static; z-index: auto;">Great news!<br>
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Will this go any way towards helping get some general support for UniMRCP via ARI? I know when I first brought this up a few months ago, my main motivation was to try and see if there was a path for getting speech functionality exposed into ARI.<br>
<span class="HOEnZb"></span><br clear="all"></blockquote></div><br></div><div class="gmail_extra">So, in my spare time, I have been playing around with this.<br><br></div><div class="gmail_extra">Putting an ARI wrapper around the res_speech API is pretty easy. I haven't really decided on what the resources would look like, but right now my working theory would be that it would look similar to the playbacks resource - that is, you can start speech detection via some operation on the channels resource, which hands you back a handle to a speech detection control resource. You can then use a speech detection resource to manipulate what is happening.<br><br></div><div class="gmail_extra">That's the theory anyway.<br><br></div><div class="gmail_extra">I've taken a glance at the Asterisk modules that were written by various folks that the UniMRCP project distributes. Unfortunately, it doesn't look like they made use of the res_speech API. As such, they can't just be used directly behind ARI - some substantial rework would have to be done.<br><br></div><div class="gmail_extra">The first step would still be to get ARI able to wrap res_speech. Once that's in place, if Arsen and others are interested, we can discuss how to tweak the UniMRCP Asterisk modules such that they fit into the overall speech architecture in Asterisk.<br><br></div></div></blockquote><div><br></div>Is it really required to use res_speech? If so, can we change the interfaces that ARI presents?</div><div><br></div><div>Over the last few years we’ve evaluated res_speech vs. the various UniMRCP applications (SynthAndRecog primarily). We’ve always come to the conclusion that the res_speech API either couldn’t give us what we needed, or was not as performant. SynthAndRecog isn’t perfect, but it does a couple of crucial things, perhaps most importantly is the combined lifecycle of TTS + ASR so that you can “barge” into a TTS playback before it is finished.</div><div><br></div><div>Would now be the appropriate time to talk about the ideal application interface for speech in Asterisk? Is it safe to assume that ARI’s speech APIs can break with the past (both res_speech and unimrcp-asterisk)?</div><div><br></div><div>/BAK/</div><div><br><blockquote type="cite"><div dir="ltr"><div class="gmail_extra">Matt<br></div><div class="gmail_extra"><br>-- <br><div dir="ltr"><div>Matthew Jordan<br></div><div>Digium, Inc. | Engineering Manager</div><div>445 Jan Davis Drive NW - Huntsville, AL 35806 - USA</div><div>Check us out at: <a href="http://digium.com/" target="_blank">http://digium.com</a> & <a href="http://asterisk.org/" target="_blank">http://asterisk.org</a></div></div>
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