Streaming audio and video RTP

William Johnston wgj at cast.uark.edu
Thu Apr 30 15:37:06 UTC 2020 

Oh, brother. I made a syntax error. I need a period after that second 
rb.  But that's on the video side and not the audio side so that wont 
fix your audio problem.

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 4/29/2020 10:49 PM, Patrick Cusack wrote:
> Hmm….again no audio comes through. I am wondering if my qtdemux is 
> correct.
>
>> On Apr 29, 2020, at 12:29 PM, William Johnston <wgj at cast.uark.edu 
>> <mailto:wgj at cast.uark.edu>> wrote:
>>
>> 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
>>>> _______________________________________________
>>>> gstreamer-devel mailing list
>>>> gstreamer-devel at lists.freedesktop.org 
>>>> <mailto:gstreamer-devel at lists.freedesktop.org>
>>>> https://lists.freedesktop.org/mailman/listinfo/gstreamer-devel
>>>
>>>
>>> _______________________________________________
>>> gstreamer-devel mailing list
>>> gstreamer-devel at lists.freedesktop.org
>>> https://lists.freedesktop.org/mailman/listinfo/gstreamer-devel
>> _______________________________________________
>> gstreamer-devel mailing list
>> gstreamer-devel at lists.freedesktop.org 
>> <mailto:gstreamer-devel at lists.freedesktop.org>
>> https://lists.freedesktop.org/mailman/listinfo/gstreamer-devel
>
>
> _______________________________________________
> gstreamer-devel mailing list
> gstreamer-devel at lists.freedesktop.org
> https://lists.freedesktop.org/mailman/listinfo/gstreamer-devel
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <https://lists.freedesktop.org/archives/gstreamer-devel/attachments/20200430/eb9cecf7/attachment-0001.htm>

More information about the gstreamer-devel mailing list