[gst-devel] Audio + video with RTP

[Rémi BUISSON]

Hi,

One last question :

On the server side I execute the following command :
gst-launch -v gstrtpbin name=rtpbin \
    filesrc location=filesrc 
location=../../../partage/Videos/superman_originale.avi ! decodebin 
name=dec \
    dec. ! queue ! x264enc byte-stream=true bitrate=300 ! rtph264pay ! 
rtpbin.send_rtp_sink_0 \
    rtpbin.send_rtp_src_0 ! udpsink port=5000 host=127.0.0.1 ts-offset=0 
name=vrtpsink \
    rtpbin.send_rtcp_src_0 ! udpsink port=5001 host=127.0.0.1 sync=false 
async=false name=vrtcpsink \
    udpsrc port=5005 name=vrtpsrc ! rtpbin.recv_rtcp_sink_0 \
    dec. ! queue ! audioresample ! audioconvert ! alawenc ! rtppcmapay ! 
rtpbin.send_rtp_sink_1 \
    rtpbin.send_rtp_src_1 ! udpsink port=5002 host=127.0.0.1 ts-offset=0 
name=artpsink \
    rtpbin.send_rtcp_src_1 ! udpsink port=5003 host=127.0.0.1 sync=false 
async=false name=artcpsink \
    udpsrc port=5007 name=artpsrc ! rtpbin.recv_rtcp_sink_1

On the client side I execute the following command :
gst-launch -vvv playbin uri=file:///home/user/test/gstreamer/client.sdp

with client.sdp:
v=0
o=- 1188340656180883 1 IN IP4 127.0.0.1
s=Session streamed by GStreamer
i=server.sh
t=0 0
a=tool:GStreamer
a=type:broadcast
m=video 5000 RTP/AVP 96
c=IN IP4 127.0.0.1
a=rtpmap:96 H264/90000
m=audio 5002 RTP/AVP 8
c=IN IP4 127.0.0.1

It works perfectly with GStreamer, VLC and MPlayer (I didn't test more 
;-)) but the sever don't stop at the end of the stream whereas the 
client does ... is it normal ?

Rémi

Rémi BUISSON wrote:
> Thanks !!
>
> It is just what I wanted !
>
> ++
>
> Wim Taymans wrote:
>   
>> On Mon, 2008-08-25 at 16:42 +0200, Rémi BUISSON wrote:
>>   
>>     
>>> Hi,
>>>
>>> How can I send both audio and video with RTP ?
>>>
>>> I tried this but it doesn't seem to work :
>>>
>>> sender:
>>> gst-launch filesrc 
>>> location=../../../partage/Videos/superman_originale.avi ! decodebin 
>>> name=dec \ dec. ! queue ! x264enc ! rtph264pay ! rtpbin localport=5001 
>>> destinations=192.168.1.2:5000 sync=false \ dec. ! queue ! x264enc ! 
>>> rtph264pay ! rtpbin localport=5003 destinations=192.168.1.2:5002 sync=false
>>>
>>> receiver:
>>> gst-launch udpsrc port=5000 caps="application/x-rtp,media=(string)video" 
>>> ! rtph264depay ! decodebin ! xvimagesink (just for video for now)
>>>
>>>     
>>>       
>> That's old deprecated rtpbin, use the new gstrtpbin element instead.
>> Here are some examples of audio and video sending and receiving
>> pipelines: 
>>
>> http://webcvs.freedesktop.org/gstreamer/gst-plugins-good/tests/examples/rtp/
>>
>>
>>   
>>     
>>> without "\ dec. ! queue ! x264enc ! rtph264pay ! rtpbin localport=5003 
>>> destinations=192.168.1.2:5002 sync=false" it works.
>>>
>>> I thought audio could be encoded with h264 ... ?
>>>     
>>>       
>> It can't, H264 is a video only codec.
>>
>> Wim
>>   
>>     
>>> Thanks for your reply.
>>>
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