Best Way to Synchronize Realtime Audio streams with Video Streams?

[Marc Leeman]

Indeed, only need the RTP stack to do so.

WebRTC used RTP as transport layer.

On Mon, 13 Jun 2022 at 13:15, Yu You via gstreamer-devel <
gstreamer-devel at lists.freedesktop.org> wrote:

> Without a working example, I still believe the A/V sync could be done with
> the element "rtpbin" with multiple udpsrc elements.
>
> Regards,
>
> Yu
>
> On Mon, 13 Jun 2022 at 09:22, Florian Echtler via gstreamer-devel <
> gstreamer-devel at lists.freedesktop.org> wrote:
>
>> Hello Abrar,
>>
>> having tried the exact same thing without much success some while ago, I
>> would
>> recommend to just use WebRTC (i.e. the webrtcbin element) for
>> synchronized A/V
>> streaming. But it obviously depends on your usecase whether that is
>> feasible.
>>
>> Best, Florian
>>
>> On 13.06.22 01:45, Abrar Shahriar via gstreamer-devel wrote:
>> > Hi,
>> >
>> > I will synchronize microphone data from a sender via UDP src sink with
>> video
>> > data coming from a camera, but not sure what the best way is.
>> >
>> > Right now, I take timestamps (PTP) at mic and video src plugin src pads
>> and add
>> > it to RTP header metadata.
>> >
>> > My receiver app receives data via udpsrc, has a pipeline for video and
>> one for
>> > audio. It can read the timestamps at sink pad of RTP depay elements.
>> >
>> > What can I do to make sure audio is within 30ms of video, for say
>> proper lipsync
>> > and such.
>> >
>> > Also will want to adjust audio video offset at runtime to compensate
>> for display
>> > latency, etc.
>> >
>> > Thanks,
>> > Abrar
>> >
>> > シャハリアル
>> >
>> > ーーーーーーーーー
>> >
>> > アブラル・ザヒン・シャハリアル(Abrar Zahin Shahriar)
>> >
>> > ハイパーダイン株式会社
>> > 〒108－0014 東京都港区芝5丁目9－12 3階
>> >
>>
>>
>> --
>> SENT FROM MY DEC VT50 TERMINAL
>>
>

-- 
g. Marc
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