Streaming audio and video RTP

[Patrick Cusack]

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> 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 <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 \
>> 
>> 
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