Matthieu LAURENT wrote:
>
> Philippe Kalaf wrote:
>> Matthieu LAURENT wrote:
>>
>>> Hi,
>>>
>>> I know gstreamer a little bit (already done a local audio/video player
>>> for our target) and I am studying the possibility to use farsight for
>>> our streaming solution.
>>> Could you explain me how the synchronisation between audio and video is
>>> done?
>>> In fact, there 2 independant pipes
>>> udp | rtp | buffer-jitter | depayloader-audio | audio-decoder | audio-sink
>>> and
>>> udp | rtp | buffer-jitter | depayloader-video | video-decoder | video-sink
>>>
>>> Imagine the video-pipe goes to state rebuffering. How the audio-pipe
>>> knows it has to stop playing audio? How will it knows it can restart
>>> playing (once enough video has been bufferized)?
>>>
>> We have only recently started working on actual synchronization.
>> Previously, Gstreamer couldn't do proper sink synchronization for live
>> sources. Recent patches to CVS have made it possible. In the future we
>> will be able to use this to sync A/V using RTCP. Currently, there is no
>> such feature.
>>
>> Best Regards,
>> Philiipe
>>
> Hi,
>
> Could you tell me what you have been thinking about as a begin of solution?
> Is the solution based on RTCP (for example Sending Report)?
> Do you know when you will have a beta version? I can help you to test it
> maybe.
> Will your solution work for VoIP application only or for streaming too?
>
> For your information, I have also asked the same question to the gstreamer
> mailing list, and I have been answered that I should place my 2 pipes in one
> pipe.
> I don't think this will enable to have a correct control of the 2 buffer
> jitters.
>
> Regards
>
> Matthieu
Hi,
I did not get any answer to my last email. I hope you can help and answer.
Regards
Matthieu
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