Streaming audio and video RTP
Patrick Cusack
patrickcusack at mac.com
Thu Apr 30 03:49:13 UTC 2020
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> 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 <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 <mailto:gstreamer-devel at lists.freedesktop.org>
>>>> https://lists.freedesktop.org/mailman/listinfo/gstreamer-devel <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 <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 <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/20200429/d49441c4/attachment.htm>
More information about the gstreamer-devel
mailing list