It'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. opusenc might be putting it in an ogg container format (or other). On 26 September 2014 22:47, jonathan augenstine <jaugenstine@gmail.com> wrote:
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@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@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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