<div dir="ltr">Steve,<div><br></div><div>That was my thought, but the file would not stream.  I can review and test again, but trying to stream that file was unsuccessful.</div><div><br></div><div>Jonathan</div></div><div class="gmail_extra"><br><div class="gmail_quote">On Fri, Sep 26, 2014 at 3:33 PM, Steven Ayre <span dir="ltr">&lt;<a href="mailto:steveayre@gmail.com" target="_blank">steveayre@gmail.com</a>&gt;</span> wrote:<br><blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex"><div dir="ltr">It&#39;s not necessarily the same format. The codec is the same, but the container may be different. fs_encode gives you just the raw data without a container afaik. It should be playable with mod_native_file. <span style="font-family:arial,sans-serif;font-size:13px">opusenc might be putting it in an ogg container format (or other).</span></div><div class="HOEnZb"><div class="h5"><div class="gmail_extra"><br><div class="gmail_quote">On 26 September 2014 22:47, jonathan augenstine <span dir="ltr">&lt;<a href="mailto:jaugenstine@gmail.com" target="_blank">jaugenstine@gmail.com</a>&gt;</span> wrote:<br><blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex"><div dir="ltr">Brian,<div><br></div><div>It is a project requirement.  They want to have the audio files available in the format of the codec of the incoming calls.</div><span><font color="#888888"><div><br></div><div>Jonathan</div></font></span></div><div><div><div class="gmail_extra"><br><div class="gmail_quote">On Fri, Sep 26, 2014 at 2:20 PM, Brian West <span dir="ltr">&lt;<a href="mailto:brian@freeswitch.org" target="_blank">brian@freeswitch.org</a>&gt;</span> wrote:<br><blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex"><div dir="ltr">Why are you trying to create raw opus files in the first place?</div><div class="gmail_extra"><br><div class="gmail_quote"><div><div>On Fri, Sep 26, 2014 at 2:44 PM, jonathan augenstine <span dir="ltr">&lt;<a href="mailto:jaugenstine@gmail.com" target="_blank">jaugenstine@gmail.com</a>&gt;</span> wrote:<br></div></div><blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex"><div><div><div dir="ltr">I have been working with WebRTC and utilizing the opus codec.  The sound quality on calls is great.  I have been encountering some file issues however and want to get some input on how to best proceed.  When the call is has both ends in Opus, or if Freeswitch is transcoding the call has no issues.  The only issues I encounter are the following.<div><br></div><div>1. fs_encode fails to create a valid opus file</div><div>When fs_encode executes, you do see mod_opus load successfully.  However, the file that is created cannot be read by <a href="http://opus.org" target="_blank">opus.org</a> utilities, specifically opusinfo.</div><div><br></div><div>The command line I use is:</div><div>fs_encode -l mod_opus infile.wav outfile.opus</div><div><br></div><div>2. Streaming a file that is encoded in opus format fails</div><div>If I encode a file using <a href="http://opus.org" target="_blank">opus.org</a> tools, opusenc, I have a valid opus file and I am able to open it with other audio applications.  If I call into freeswitch and try to stream out that file, there is no error reported in the console logs, but the audio heard is garbled.</div><div><br></div><div>Basically the mod_opus appears to work but reading, writing files appears to have an issue.</div><div><br></div><div>My questions.</div><div>1. Is there anything I might have configured incorrectly?</div><div>2. What information can I grab to help troubleshoot this issue?</div><span><font color="#888888"><div><br></div><div>Jonathan</div><div><br></div></font></span></div>
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<p><font face="courier new, monospace"><b><i><font size="4">Brian West</font></i></b><br><span style="font-size:x-small"><a href="mailto:brian@freeswitch.org" target="_blank">brian@freeswitch.org</a></span></font></p>
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