"pausing" parts of a pipeline

[Sven Heyll]

Yes, but I would like leverage the plethora of plugins available for
gstreamer (audio filters, video, etc...)


2013/8/6 Federico Zamperini <fzamperini at tiscali.it>

> Take a look at pjsip (http://www.pjsip.org/), it is specifically designed
> for internet telephony, so maybe the functionality you are looking for are
> already implemented.
>
> Il 06/08/2013 12:47, Sven Heyll ha scritto:
>
>> Hi
>>
>> thanks for the answer. interaudiosink/src sounds interesting. It is one
>> approach I was thinking about implementing it by myself.
>> Yes, I would like to have a steady rtp stream between two telephone
>> calls with the option to start, pause and resume playbacks and
>> recordings via DTMF. Technically it seems unexpected hard to do with
>> gstreamer, though
>>
>> But isn't interaudiosink actually a hack? I haven't read the sources of
>> interaudiosink/src yet, but shouldn't elments have the ability to be
>> paused/seeked independent of the rest of the pipeline?
>>
>> wbr
>> Sven
>>
>>
>> 2013/8/6 Tim-Philipp Müller <t.i.m at zen.co.uk <mailto:t.i.m at zen.co.uk>>
>>
>>
>>     On Mon, 2013-08-05 at 16:55 +0200, Sven Heyll wrote:
>>
>>     Hi,
>>
>>      > this is what I would like to do:
>>      >
>>      >
>>      > RTP/UDP SRC ---> GstAdder ----> Some Audio Processing ---> RTP/UDP
>>      > SINK
>>      >
>>      >
>>      > Every now and then I'd like to dynamically add a playback:
>>      >
>>      >
>>      > RTP/UDP SRC ---> GstAdder ----> Some Audio Processing ---> RTP/UDP
>>      > SINK
>>      >                    /\
>>      >      urdidecode ---||
>>      >
>>      >
>>      > My question: How would I "pause" and "resume" the uridecoder in the
>>      > second figure.
>>
>>     Do you really want to pause and resume the uridecodebin part, or add
>> it,
>>     play, and then remove it, and then later add another one again?
>>
>>     You could do something with the inter elements (interaudiosink/src).
>>
>>     Cheers
>>       -Tim
>>
>>
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