trying to create a video chat app.

[Peter Maersk-Moller]

Hi Nicholas.

Unless you mux (multiplex) audio and video into a single multiplexed stream
or use rtspserver with two rtp streams and the rtspserver telling clients
how audio and video is synchronized or you use a more complicated setup
rtpsession/rtpbin and RTCP and NTP, you will not have audio and video
synchronized. Why not use a multiplexer like mpegtsmux and have audio/video
encoded and muxed and sent in a single pipeline and then in another
pipeline receiving multiplexed stream, demuxing it, decoding it and then
played?

regards
Peter

On Tue, Jun 23, 2015 at 8:28 AM, Lab work <480437 at gmail.com> wrote:

> Hi
>
> Thanks for the replay.
> Your answer left me a bit confused.
>
> >Finally, each streams should be sent to it's down socket in RTP, you may
> multiplex them but this is a lot more work.
>
> Do you mean that I should not mux them and do something like:
>
> pipeline 1:
>
> gst-launch-1.0 v4l2src !  video/x-raw,width=640,height=480 !  timeoverlay
> !  tee name="local" !  queue !  autovideosink local. !  queue ! x264enc
> tune=zerolatency byte-stream=true bitrate=500 threads=1 ! h264parse
> config-interval=1  ! rtph264pay !  udpsink host=192.168.100.3 port= 5000
> udpsrc port=5000 caps=\"application/x-rtp,payload=96,encoding-name=H264\" !
> queue ! rtph264depay ! h264parse ! decodebin ! autovideosink
>
> pipeline 2 :
>
> gst-launch-1.0 pulsesrc ! audioconvert ! audioresample ! speexenc !
> rtpspeexpay ! udpsink host=192.168.100.3 port=4444 udpsrc port=4444
> caps="application/x-rtp, media=(string)audio, clock-rate=(int)16000,
> encoding-name=(string)SPEEX, encoding-params=(string)1, payload=(int)110" !
> rtpjitterbuffer ! rtpspeexdepay ! speexdec ! audioconvert ! audioresample !
> autoaudiosink
>
>
> And playing similar pipelines on other PC. I have tried this type of
> approach and the video part was working but audio part was creating problem.
> when I run the second pipeline there is no audio at either end while they
> work fine independently. Can you please explain the reason. And me out of
> this problem.
> Thank you.
>
> On Mon, Jun 22, 2015 at 7:32 PM, Nicolas Dufresne <
> nicolas.dufresne at collabora.com> wrote:
>
>> Le lundi 22 juin 2015 à 17:00 +0530, Lab work a écrit :
>> > gst-launch-1.0  oggmux name="muxer"  v4l2src ! video/x-raw,
>> > framerate=30/1, width=640, height=480 ! videoconvert ! x264enc !
>> > multiqueue ! muxer.  videotestsrc ! video/x-raw, framerate=30/1,
>> > width=640, height=480 ! videoconvert ! x264enc ! multiqueue ! muxer.
>> > autoaudiosrc ! audioconvert ! speexenc ! queue ! muxer.  udpsink
>> > host=127.0.0.1 port=5000
>>
>> So you want to stream in RTP over UDP. First you need to drop this
>> unlinked oggmux. OGG is not RTP. Also it does not work with variable
>> framerate (which most Logitech camera produces). Second, pick a preset
>> on x264enc that is suited for live (like tune=zerolatency). Finally,
>> each streams should be sent to it's down socket in RTP, you may multi
>> -plex them but this is a lot more work.
>>
>> Nicolas
>> _______________________________________________
>> gstreamer-devel mailing list
>> gstreamer-devel at lists.freedesktop.org
>> http://lists.freedesktop.org/mailman/listinfo/gstreamer-devel
>>
>>
>
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>
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