Streaming audio and video RTP

[William Johnston]

Careless of me, I linked it wrong. I linked the input of rtpbin to the 
input of udpsink.

I'll try again:

gst-launch-1.0 -e \
         rtpbin name=rb
         uridecodebin uri="file:///home/fedora/starwars.mov" \
         ! qtdemux name=demux  demux.audio_0 \
         ! queue \
         ! audioconvert \
         ! opusenc bandwidth=superwideband bitrate-type=vbr \
         ! rtpopuspay  \
         ! rtprtxqueue max-size-time=2000 max-size-packets=0 \
         ! udpsink host=www.playbacktc.com <http://www.playbacktc.com/> 
port=5052 \
         demux.video_0 \
         ! queue \
         ! videorate ! video/x-raw, framerate=30000/1001 \
         ! videoconvert \
         ! x264enc tune=zerolatency speed-preset=1 dct8x8=true 
quantizer=17 pass=qual \
         ! rtph264pay \
         ! rtprtxqueue max-size-time=2000 max-size-packets=0 \
         ! rb.send_rtp_sink_0 \
         rb
         ! udpsink host=www.playbacktc.com <http://www.playbacktc.com/> 
port=5054 \



On 4/28/2020 6:32 PM, Patrick Cusack wrote:
> Ok. Good to know. Unfortunately, that doesn’t work. I get the following:
>
> Setting pipeline to PAUSED ...
> Pipeline is PREROLLING ...
> DtsGetHWFeatures: Create File Failed
> DtsGetHWFeatures: Create File Failed
> Running DIL (3.22.0) Version
> DtsDeviceOpen: Opening HW in mode 0
> DtsDeviceOpen: Create File Failed
> Redistribute latency...
> WARNING: from element 
> /GstPipeline:pipeline0/GstURIDecodeBin:uridecodebin0: Delayed linking 
> failed.
> Additional debug info:
> ./grammar.y(506): gst_parse_no_more_pads (): 
> /GstPipeline:pipeline0/GstURIDecodeBin:uridecodebin0:
> failed delayed linking some pad of GstURIDecodeBin named uridecodebin0 
> to some pad of GstQTDemux named demux
> Redistribute latency…
>
> I checked the stats on my server and don’t see any audio or video 
> packets coming in. The goal is to stream a file (eventually a video 
> input like Decklink) to a server that receives rtp.
>
> I can send audio or video separately and I don’t have issues.
>
> Patrick
>
>> On Apr 28, 2020, at 11:49 AM, William Johnston <wgj at cast.uark.edu 
>> <mailto:wgj at cast.uark.edu>> wrote:
>>
>> You can only specify ports on element names. Try this:
>>
>> gst-launch-1.0 -e \
>>         uridecodebin uri="file:///home/fedora/starwars.mov" \
>>         ! qtdemux name=demux  demux.audio_0 \
>>         ! queue \
>>         ! audioconvert \
>>         ! opusenc bandwidth=superwideband bitrate-type=vbr \
>>         ! rtpopuspay  \
>>         ! rtprtxqueue max-size-time=2000 max-size-packets=0 \
>>         ! udpsink host=www.playbacktc.com 
>> <http://www.playbacktc.com/> port=5052 \
>>         demux.video_0 \
>>         ! queue \
>>         ! videorate ! video/x-raw, framerate=30000/1001 \
>>         ! videoconvert \
>>         ! x264enc tune=zerolatency speed-preset=1 dct8x8=true 
>> quantizer=17 pass=qual \
>>         ! rtph264pay \
>>         ! rtprtxqueue max-size-time=2000 max-size-packets=0 \
>>         ! rtpbin name=rb rb.send_rtp_sink_0 \
>>         ! udpsink host=www.playbacktc.com 
>> <http://www.playbacktc.com/> port=5054 \
>>
>>
>> On 4/28/2020 12:42 PM, Patrick Cusack wrote:
>>> I have a endpoint that expects audio and video over ports 5052 and 
>>> 5054 respectively. I am using the following script to send audio and 
>>> video. I am getting a 'WARNING: erroneous pipeline: syntax error’ 
>>> when I run the command.
>>> Also, does using simple rtp payloads into a udp sink bypass RTCP 
>>> feedback, ie if my server is NACKing on account of dropped packets, 
>>> does this hinder retransmission of rtp packets?
>>>
>>> gst-launch-1.0 -e \
>>>         uridecodebin uri="file:///home/fedora/starwars.mov" \
>>>         ! qtdemux name=demux  demux.audio_0 \
>>>         ! queue \
>>>         ! audioconvert \
>>>         ! opusenc bandwidth=superwideband bitrate-type=vbr \
>>>         ! rtpopuspay  \
>>>         ! rtprtxqueue max-size-time=2000 max-size-packets=0 \
>>>         ! udpsink host=www.playbacktc.com 
>>> <http://www.playbacktc.com/> port=5052 \
>>>         demux.video_0 \
>>>         ! queue \
>>>         ! videorate ! video/x-raw, framerate=30000/1001 \
>>>         ! videoconvert \
>>>         ! x264enc tune=zerolatency speed-preset=1 dct8x8=true 
>>> quantizer=17 pass=qual \
>>>         ! rtph264pay \
>>>         ! rtprtxqueue max-size-time=2000 max-size-packets=0 \
>>>         ! rtpbin.send_rtp_sink_0 \
>>>         ! udpsink host=www.playbacktc.com 
>>> <http://www.playbacktc.com/> port=5054 \
>>>
>>> _______________________________________________
>>> gstreamer-devel mailing list
>>> gstreamer-devel at lists.freedesktop.org
>>> https://lists.freedesktop.org/mailman/listinfo/gstreamer-devel
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>
>
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