Uses to reduce latency of a pipeline

[Jack]

Hello,

I would like to know if you have recommandations on how to reduce
latency on a running pipeline using gstwebrtc ?

For instance, a pipeline 1 like (using Python) :

webrtcbin name=sendrecv bundle-policy=max-bundle
filesrc location=my_stereo_sound.wav ! wavparse ! volume name="vol0"
volume=0.5 ! opusenc frame-size=10 ! rtpopuspay pt=96 !
application/x-rtp,media=audio,encoding-name=OPUS,payload=96,clock-rate=48000
! sendrecv.

and a pipeline 2 like (always with Python) :

webrtcbin name=sendrecv bundle-policy=max-bundle
filesrc location=my_stereo_sound.wav ! wavparse ! deinterleave name=d
d.src_0 ! queue ! volume name="vol0" volume=0.5 ! i.sink_0 d.src_1 !
queue ! volume name="vol1" volume=0.5 ! i.sink_1 interleave name=i !
audiopanorama ! opusenc frame-size=10 ! rtpopuspay pt=96 !
application/x-rtp,media=audio,encoding-name=OPUS,payload=96,clock-rate=48000
! sendrecv.

If I change dynamically the value on "vol0" on the first pipeline (with
set_property()), I can hear the modification 1 second later (for me it
is acceptable). But if I do the same on the pipeline 2, the modification
is heard 2 seconds later ! Huge difference ! It is normal regarding the
"complexity" of the second pipeline (compare to the first) ? What do you
advise to reduce latency on the second pipeline ?
++

Jack

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[Nicolas Dufresne-5]

Le lundi 28 janvier 2019 à 13:04 +0100, Jack a écrit :
> Hello,
>
> I would like to know if you have recommandations on how to reduce
> latency on a running pipeline using gstwebrtc ?
>
> For instance, a pipeline 1 like (using Python) :
>
> webrtcbin name=sendrecv bundle-policy=max-bundle

You can configured the jitterbuffer latency using child proxy
properties. But default it's 200ms (rtpbin latency property is in ms),
this example would be 20ms:

  webrtcbin rtpbin::latency=20

> filesrc location=my_stereo_sound.wav ! wavparse ! volume name="vol0"
> volume=0.5 ! opusenc frame-size=10 ! rtpopuspay pt=96 !
> application/x-rtp,media=audio,encoding-name=OPUS,payload=96,clock-rate=48000
> ! sendrecv.
>
> and a pipeline 2 like (always with Python) :
>
> webrtcbin name=sendrecv bundle-policy=max-bundle
> filesrc location=my_stereo_sound.wav ! wavparse ! deinterleave name=d
> d.src_0 ! queue ! volume name="vol0" volume=0.5 ! i.sink_0 d.src_1 !
> queue ! volume name="vol1" volume=0.5 ! i.sink_1 interleave name=i !
> audiopanorama ! opusenc frame-size=10 ! rtpopuspay pt=96 !
> application/x-rtp,media=audio,encoding-name=OPUS,payload=96,clock-rate=48000
> ! sendrecv.
>
> If I change dynamically the value on "vol0" on the first pipeline (with
> set_property()), I can hear the modification 1 second later (for me it
> is acceptable). But if I do the same on the pipeline 2, the modification
> is heard 2 seconds later ! Huge difference ! It is normal regarding the
> "complexity" of the second pipeline (compare to the first) ? What do you
> advise to reduce latency on the second pipeline ?
> ++
>
> Jack
>
> _______________________________________________
> gstreamer-devel mailing list
> [hidden email]
> https://lists.freedesktop.org/mailman/listinfo/gstreamer-devel

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[Jack]

Hello Nicolas,

Le 29/01/2019 à 17:46, Nicolas Dufresne a écrit :

> Le lundi 28 janvier 2019 à 13:04 +0100, Jack a écrit :
>> Hello,
>>
>> I would like to know if you have recommandations on how to reduce
>> latency on a running pipeline using gstwebrtc ?
>>
>> For instance, a pipeline 1 like (using Python) :
>>
>> webrtcbin name=sendrecv bundle-policy=max-bundle
>
> You can configured the jitterbuffer latency using child proxy
> properties. But default it's 200ms (rtpbin latency property is in ms),
> this example would be 20ms:
>
>   webrtcbin rtpbin::latency=20

If you can develop a little bit more your answer or give me a url with
an example or explanation I will be fine :)

Why using your solution on the first pipeline, which is working fine,
will solve my problem on the second pipeline ?
Thanx.
++

Jack

>
>> filesrc location=my_stereo_sound.wav ! wavparse ! volume name="vol0"
>> volume=0.5 ! opusenc frame-size=10 ! rtpopuspay pt=96 !
>> application/x-rtp,media=audio,encoding-name=OPUS,payload=96,clock-rate=48000
>> ! sendrecv.
>>
>> and a pipeline 2 like (always with Python) :
>>
>> webrtcbin name=sendrecv bundle-policy=max-bundle
>> filesrc location=my_stereo_sound.wav ! wavparse ! deinterleave name=d
>> d.src_0 ! queue ! volume name="vol0" volume=0.5 ! i.sink_0 d.src_1 !
>> queue ! volume name="vol1" volume=0.5 ! i.sink_1 interleave name=i !
>> audiopanorama ! opusenc frame-size=10 ! rtpopuspay pt=96 !
>> application/x-rtp,media=audio,encoding-name=OPUS,payload=96,clock-rate=48000
>> ! sendrecv.
>>
>> If I change dynamically the value on "vol0" on the first pipeline (with
>> set_property()), I can hear the modification 1 second later (for me it
>> is acceptable). But if I do the same on the pipeline 2, the modification
>> is heard 2 seconds later ! Huge difference ! It is normal regarding the
>> "complexity" of the second pipeline (compare to the first) ? What do you
>> advise to reduce latency on the second pipeline ?
>> ++
>>
>> Jack
>>
>> _______________________________________________
>> gstreamer-devel mailing list
>> [hidden email]
>> https://lists.freedesktop.org/mailman/listinfo/gstreamer-devel
>>
>> _______________________________________________
>> gstreamer-devel mailing list
>> [hidden email]
>> https://lists.freedesktop.org/mailman/listinfo/gstreamer-devel

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[Nicolas Dufresne-5]

Le mercredi 30 janvier 2019 à 00:10 +0100, Jack a écrit :

> Hello Nicolas,
>
> Le 29/01/2019 à 17:46, Nicolas Dufresne a écrit :
> > Le lundi 28 janvier 2019 à 13:04 +0100, Jack a écrit :
> > > Hello,
> > >
> > > I would like to know if you have recommandations on how to reduce
> > > latency on a running pipeline using gstwebrtc ?
> > >
> > > For instance, a pipeline 1 like (using Python) :
> > >
> > > webrtcbin name=sendrecv bundle-policy=max-bundle
> >
> > You can configured the jitterbuffer latency using child proxy
> > properties. But default it's 200ms (rtpbin latency property is in ms),
> > this example would be 20ms:
> >
> >   webrtcbin rtpbin::latency=20
>
> If you can develop a little bit more your answer or give me a url with
> an example or explanation I will be fine :)
>
> Why using your solution on the first pipeline, which is working fine,
> will solve my problem on the second pipeline ?
> Thanx.
> ++

It felt you you where asking general question from the subject,
configuring the jitterbuffer latency is a general way to reduce
webrtcbin (or rtpbin) latency.

>
> Jack
>
>
>
> > > filesrc location=my_stereo_sound.wav ! wavparse ! volume name="vol0"
> > > volume=0.5 ! opusenc frame-size=10 ! rtpopuspay pt=96 !
> > > application/x-rtp,media=audio,encoding-name=OPUS,payload=96,clock-rate=48000
> > > ! sendrecv.
> > >
> > > and a pipeline 2 like (always with Python) :
> > >
> > > webrtcbin name=sendrecv bundle-policy=max-bundle
> > > filesrc location=my_stereo_sound.wav ! wavparse ! deinterleave name=d
> > > d.src_0 ! queue ! volume name="vol0" volume=0.5 ! i.sink_0 d.src_1 !
> > > queue ! volume name="vol1" volume=0.5 ! i.sink_1 interleave name=i !
> > > audiopanorama ! opusenc frame-size=10 ! rtpopuspay pt=96 !
> > > application/x-rtp,media=audio,encoding-name=OPUS,payload=96,clock-rate=48000
> > > ! sendrecv.

In this case, the problem is likely related to having a non-live source
sent to webrtcbin. I'd first ask to give a try with a live source to
confirm (audiotestsrc is-live=1 as an example), then I might elaborate.

> > >
> > > If I change dynamically the value on "vol0" on the first pipeline (with
> > > set_property()), I can hear the modification 1 second later (for me it
> > > is acceptable). But if I do the same on the pipeline 2, the modification
> > > is heard 2 seconds later ! Huge difference ! It is normal regarding the
> > > "complexity" of the second pipeline (compare to the first) ? What do you
> > > advise to reduce latency on the second pipeline ?
> > > ++
> > >
> > > Jack
> > >
> > > _______________________________________________
> > > gstreamer-devel mailing list
> > > [hidden email]
> > > https://lists.freedesktop.org/mailman/listinfo/gstreamer-devel
> > >
> > > _______________________________________________
> > > gstreamer-devel mailing list
> > > [hidden email]
> > > https://lists.freedesktop.org/mailman/listinfo/gstreamer-devel
>
> _______________________________________________
> gstreamer-devel mailing list
> [hidden email]
> https://lists.freedesktop.org/mailman/listinfo/gstreamer-devel

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[Jack]

Le 30/01/2019 à 04:08, Nicolas Dufresne a écrit :

> Le mercredi 30 janvier 2019 à 00:10 +0100, Jack a écrit :
>> Hello Nicolas,
>>
>> Le 29/01/2019 à 17:46, Nicolas Dufresne a écrit :
>>> Le lundi 28 janvier 2019 à 13:04 +0100, Jack a écrit :
>>>> Hello,
>>>>
>>>> I would like to know if you have recommandations on how to reduce
>>>> latency on a running pipeline using gstwebrtc ?
>>>>
>>>> For instance, a pipeline 1 like (using Python) :
>>>>
>>>> webrtcbin name=sendrecv bundle-policy=max-bundle
>>>
>>> You can configured the jitterbuffer latency using child proxy
>>> properties. But default it's 200ms (rtpbin latency property is in ms),
>>> this example would be 20ms:
>>>
>>>   webrtcbin rtpbin::latency=20
>>
>> If you can develop a little bit more your answer or give me a url with
>> an example or explanation I will be fine :)
>>
>> Why using your solution on the first pipeline, which is working fine,
>> will solve my problem on the second pipeline ?
>> Thanx.
>> ++
>
> It felt you you where asking general question from the subject,
> configuring the jitterbuffer latency is a general way to reduce
> webrtcbin (or rtpbin) latency.

Yes, it is always good to know general behavior of gstreamer. I am quite
a beginner with this toolkit so... And if your solution can help to
reduce latency, I am very interested for sure ! And if you can develop a
little bit I will be very happy ! ;)

My original question was more about : why there is so much difference in
latency between my pipeline 1 and my pipeline 2, is it because I use
deinterleave/interleave/queue ? And how I could reduce it in the second
pipeline (if it is doable).
++

Jack

>
>>
>> Jack
>>
>>
>>
>>>> filesrc location=my_stereo_sound.wav ! wavparse ! volume name="vol0"
>>>> volume=0.5 ! opusenc frame-size=10 ! rtpopuspay pt=96 !
>>>> application/x-rtp,media=audio,encoding-name=OPUS,payload=96,clock-rate=48000
>>>> ! sendrecv.
>>>>
>>>> and a pipeline 2 like (always with Python) :
>>>>
>>>> webrtcbin name=sendrecv bundle-policy=max-bundle
>>>> filesrc location=my_stereo_sound.wav ! wavparse ! deinterleave name=d
>>>> d.src_0 ! queue ! volume name="vol0" volume=0.5 ! i.sink_0 d.src_1 !
>>>> queue ! volume name="vol1" volume=0.5 ! i.sink_1 interleave name=i !
>>>> audiopanorama ! opusenc frame-size=10 ! rtpopuspay pt=96 !
>>>> application/x-rtp,media=audio,encoding-name=OPUS,payload=96,clock-rate=48000
>>>> ! sendrecv.
>
> In this case, the problem is likely related to having a non-live source
> sent to webrtcbin. I'd first ask to give a try with a live source to
> confirm (audiotestsrc is-live=1 as an example), then I might elaborate.
>
>>>>
>>>> If I change dynamically the value on "vol0" on the first pipeline (with
>>>> set_property()), I can hear the modification 1 second later (for me it
>>>> is acceptable). But if I do the same on the pipeline 2, the modification
>>>> is heard 2 seconds later ! Huge difference ! It is normal regarding the
>>>> "complexity" of the second pipeline (compare to the first) ? What do you
>>>> advise to reduce latency on the second pipeline ?
>>>> ++
>>>>
>>>> Jack
>>>>
>>>> _______________________________________________
>>>> gstreamer-devel mailing list
>>>> [hidden email]
>>>> https://lists.freedesktop.org/mailman/listinfo/gstreamer-devel
>>>>
>>>> _______________________________________________
>>>> gstreamer-devel mailing list
>>>> [hidden email]
>>>> https://lists.freedesktop.org/mailman/listinfo/gstreamer-devel
>>
>> _______________________________________________
>> gstreamer-devel mailing list
>> [hidden email]
>> https://lists.freedesktop.org/mailman/listinfo/gstreamer-devel
>
> _______________________________________________
> gstreamer-devel mailing list
> [hidden email]
> https://lists.freedesktop.org/mailman/listinfo/gstreamer-devel
>

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