[asterisk-dev] Avoid audio loss in ChanSpy (ref: ASTERISK-25866)

Jean Aunis jean.aunis at prescom.fr
Fri Apr 1 07:59:00 CDT 2016



Le 01/04/2016 14:00, Joshua Colp a écrit :
> Jean Aunis wrote:
>> Hello,
>>
>> As described in the issue ASTERISK-25866
>> <https://issues.asterisk.org/jira/browse/ASTERISK-25866>, it appears
>> that ChanSpy is randomly loosing audio frames, because it sets the flags
>> AST_AUDIOHOOK_TRIGGER_SYNC and AST_AUDIOHOOK_SMALL_QUEUE when creating
>> the audiohook: these option cause the audiohook's queue to be flushed of
>> "old" frames at very short intervals.
>>
>> I have submitted a patch on gerrit which does two things :
>> 1- do not set the flag AST_AUDIOHOOK_TRIGGER_SYNC if ChanSpy has been
>> called with the option "o": in this case we only listen to the audio
>> coming from this channel so this flag is useless
>> 2- create a new option "l" which prevents the flag
>> AST_AUDIOHOOK_SMALL_QUEUE from being set. I find it better than just
>> removing the flag in all cases, because this may introduce some delay in
>> the audio, which may not be acceptable for many ChanSpy users.
>>
>> What do you think of the idea ?
>
> I think this is fine, provided there is still some high ceiling value. 
> Allowing a queue to potentially grow out of control would be bad.
>
> I think if we were able to switch things to using a timer instead we 
> could actually get rid of this synchronization. It exists because it 
> has to bring together two directions and then provide the media.
>
> I also don't see your review on gerrit. Java (yay Java!) ran out of 
> memory somewhat overnight it seems and I've restarted gerrit this 
> morning. That may have caused the review to not actually get 
> submitted. If you do so again it should work fine and if not we can 
> help figure out why.
>
> Cheers,
>
I just pushed again, but it seems the branch was not created. I used the 
following command (taken from the wiki):

git review -t ASTERISK-25866

And here is the output :

remote: Processing changes: new: 1, refs: 1, done
remote:
remote: New Changes:
remote:   https://gerrit.asterisk.org/2520 app_chanspy: reduce audio 
loss on the spying channel.
remote:
To ssh://PrescomJA@gerrit.asterisk.org:29418/asterisk
  * [new branch]      HEAD -> refs/publish/master/ASTERISK-25866

Best regards

Jean Aunis
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