[Freeswitch-users] Opus codec issue

[jonathan augenstine]

Brian,

It is a project requirement.  They want to have the audio files available
in the format of the codec of the incoming calls.

Jonathan

On Fri, Sep 26, 2014 at 2:20 PM, Brian West <brian at freeswitch.org> wrote:

> Why are you trying to create raw opus files in the first place?
>
> On Fri, Sep 26, 2014 at 2:44 PM, jonathan augenstine <
> jaugenstine at gmail.com> wrote:
>
>> 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.
>>
>> 1. fs_encode fails to create a valid opus file
>> When fs_encode executes, you do see mod_opus load successfully.  However,
>> the file that is created cannot be read by opus.org utilities,
>> specifically opusinfo.
>>
>> The command line I use is:
>> fs_encode -l mod_opus infile.wav outfile.opus
>>
>> 2. Streaming a file that is encoded in opus format fails
>> If I encode a file using opus.org 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.
>>
>> Basically the mod_opus appears to work but reading, writing files appears
>> to have an issue.
>>
>> My questions.
>> 1. Is there anything I might have configured incorrectly?
>> 2. What information can I grab to help troubleshoot this issue?
>>
>> Jonathan
>>
>>
>> _________________________________________________________________________
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>>
>> 
>> 
>>
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>
>
>
> --
>
> *Brian West*
> brian at freeswitch.org
>
>
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> _________________________________________________________________________
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>
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>
> 
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