Opus via RTP

Mon Sep 12 22:30:11 UTC 2016

Now I have made it working with MPEGTSMUX to filesink, how can I put it
into single UDPSINK?

The MPEGTS to filesink working is like this:
OUT_FILE="stream.ts"

LISTEN_AUDIO_PORT=1236
LISTEN_VIDEO_PORT=1234

ACAPS="\"application/x-rtp, media=(string)audio, clock-rate=(int)48000,
encoding-name=(string)OPUS,payload=111\""
VCAPS="\"application/x-rtp, media=(string)video, clock-rate=(int)90000,
encoding-name=(string)VP8,framerate=30/1,payload=100\""

GST_PARAMS="-e -m -v --gst-debug=**:4 "

gst-launch-1.0 $GST_PARAMS \
 udpsrc port=$LISTEN_VIDEO_PORT caps=$VCAPS ! rtpjitterbuffer latency=3000
do-retransmission=true ! queue ! rtpvp8depay ! vp8dec ! videoscale
sharpen=1 method=2 ! videoconvert \
! queue ! x264enc bframes=0 bitrate=102400 subme=5 ! queue \
! muxer.sink_1 \
udpsrc port=$LISTEN_AUDIO_PORT caps=$ACAPS ! rtpjitterbuffer latency=3000
do-retransmission=true ! queue ! rtpopusdepay ! opusdec ! audioconvert \
! queue ! voaacenc bitrate=64000 ! queue ! muxer.sink_2 \
mpegtsmux name=muxer \
! filesink location=$OUT_FILE

On Mon, Sep 12, 2016 at 11:43 PM, Miha Nedok <mike at mike.si> wrote:

>
> Thanks guys I was missing a few elements, I tried with fakesink so long
> that it worked.
>
> But now I have a different question I have to mux and transport audio and
> video combined.
> I have to use MPEGTS, but when I use the MPEGTS muxer instead of mp4 an
> empty file is created and nothing happens,
> i thought that MPEGTS container can have x264 and AAC.
>
> My mp4mux pipeline is now like this:
> OUT_FILE="stream.mp4"
>
> LISTEN_AUDIO_PORT=1236
> LISTEN_VIDEO_PORT=1234
>
> ACAPS="\"application/x-rtp, media=(string)audio, clock-rate=(int)48000,
> encoding-name=(string)OPUS,payload=111\""
> VCAPS="\"application/x-rtp, media=(string)video, clock-rate=(int)90000,
> encoding-name=(string)VP8,framerate=30/1,payload=100\""
>
> GST_PARAMS="-e -m -v --gst-debug=**:4 "
>
> gst-launch-1.0 $GST_PARAMS \
>  udpsrc port=$LISTEN_VIDEO_PORT caps=$VCAPS ! rtpjitterbuffer latency=3000
> do-retransmission=true ! queue ! rtpvp8depay ! vp8dec ! videoscale
> sharpen=1 method=2 ! videoconvert \
> ! queue ! x264enc bitrate=102400 subme=5 ! queue \
> ! muxer.video_0 \
> udpsrc port=$LISTEN_AUDIO_PORT caps=$ACAPS ! rtpjitterbuffer latency=3000
> do-retransmission=true ! queue ! rtpopusdepay ! opusdec ! audioconvert \
> ! queue ! lamemp3enc bitrate=64000 ! queue ! muxer.audio_0 \
> mp4mux name=muxer streamable=true \
> ! filesink location=$OUT_FILE  \
>
>
>
> And if somebody could give me a hint how could i put this MP4 into a
> single udpsink?
>
>
>
>
>
> On Mon, Sep 12, 2016 at 6:16 PM, Olivier Crête <
> olivier.crete at collabora.com> wrote:
>
>> Hi,
>>
>> What version of GStreamer are you using? You may be using a very old
>> version. This  gst-launch-1.0 line works fine here on 1.8.x.
>>
>> Olivier
>>
>> On Sun, 2016-09-11 at 14:24 +0200, Miha Nedok wrote:
>>
>> I know that :)
>>
>> but even if i ad rtpopusdepay i always get the same result.
>>
>> gst-launch-1.0 -vvvvv udpsrc port=1236 caps="application/x-rtp,media=
>> (string)audio,clock-rate=48000,encoding-params=2,encoding-name=(string)OPUS,payload=111"
>>  ! rtpopusdepay ! opusdec ! audioconvert
>> Setting pipeline to PAUSED ...
>> Pipeline is live and does not need PREROLL ...
>> Setting pipeline to PLAYING ...
>> New clock: GstSystemClock
>> /GstPipeline:pipeline0/GstUDPSrc:udpsrc0.GstPad:src: caps =
>> "application/x-rtp\,\ media\=\(string\)audio\,\ clock-rate\=\(int\)48000\,\
>> encoding-params\=\(int\)2\,\ encoding-name\=\(string\)OPUS\,\
>> payload\=\(int\)111"
>> /GstPipeline:pipeline0/GstRTPOpusDepay:rtpopusdepay0.GstPad:src: caps =
>> "audio/x-opus\,\ channel-mapping-family\=\(int\)0"
>> /GstPipeline:pipeline0/GstOpusDec:opusdec0.GstPad:sink: caps =
>> "audio/x-opus\,\ channel-mapping-family\=\(int\)0"
>> /GstPipeline:pipeline0/GstRTPOpusDepay:rtpopusdepay0.GstPad:sink: caps =
>> "application/x-rtp\,\ media\=\(string\)audio\,\ clock-rate\=\(int\)48000\,\
>> encoding-params\=\(int\)2\,\ encoding-name\=\(string\)OPUS\,\
>> payload\=\(int\)111"
>> /GstPipeline:pipeline0/GstOpusDec:opusdec0.GstPad:src: caps =
>> "audio/x-raw\,\ format\=\(string\)S16LE\,\ layout\=\(string\)interleaved\,\
>> rate\=\(int\)48000\,\ channels\=\(int\)2\,\ channel-mask\=\(bitmask\)0x000
>> 0000000000003"
>> /GstPipeline:pipeline0/GstAudioConvert:audioconvert0.GstPad:src: caps =
>> "audio/x-raw\,\ format\=\(string\)S16LE\,\ layout\=\(string\)interleaved\,\
>> rate\=\(int\)48000\,\ channels\=\(int\)2\,\ channel-mask\=\(bitmask\)0x000
>> 0000000000003"
>> /GstPipeline:pipeline0/GstAudioConvert:audioconvert0.GstPad:sink: caps =
>> "audio/x-raw\,\ format\=\(string\)S16LE\,\ layout\=\(string\)interleaved\,\
>> rate\=\(int\)48000\,\ channels\=\(int\)2\,\ channel-mask\=\(bitmask\)0x000
>> 0000000000003"
>> ERROR: from element /GstPipeline:pipeline0/GstUDPSrc:udpsrc0: Internal
>> data flow error.
>> Additional debug info:
>> gstbasesrc.c(2948): gst_base_src_loop (): /GstPipeline:pipeline0/GstUDPS
>> rc:udpsrc0:
>> streaming task paused, reason not-linked (-1)
>> Execution ended after 0:00:00.020207760
>> Setting pipeline to PAUSED ...
>> Setting pipeline to READY ...
>> Setting pipeline to NULL ...
>> Freeing pipeline ...
>>
>> On Sun, Sep 11, 2016 at 2:11 PM, Sebastian Dröge <
>> sebastian at centricular.com> wrote:
>>
>> On So, 2016-09-11 at 04:17 -0700, MikeSI wrote:
>> > I cannot reeive opus via RTP. Even If i set payload it's always the
>> > same.
>> >
>> > gst-launch-1.0 -vvvvv udpsrc port=1236
>> > caps="application/x-rtp,media=(string)audio,clock-
>> > rate=48000,encoding-params=2,encoding-name=(string)OPUS"
>>
>> This is not a complete pipeline, you're missing at least the RTP
>> depayloader, possibly a decoder and converters, and a sink. E.g.
>>
>> gst-launch-1.0 -vvvvv udpsrc port=1236 caps="application/x-
>> rtp,media=(string)audio,clock-rate=48000,encoding-params=2,encoding-
>> name=(string)OPUS" ! rtpopusdepay ! opusdec ! fakesink
>>
>> --
>> Sebastian Dröge, Centricular Ltd · http://www.centricular.com
>> _______________________________________________
>> gstreamer-devel mailing list
>> gstreamer-devel at lists.freedesktop.org
>> https://lists.freedesktop.org/mailman/listinfo/gstreamer-devel
>>
>>
>> _______________________________________________
>> gstreamer-devel mailing listgstreamer-devel at lists.freedesktop.orghttps://lists.freedesktop.org/mailman/listinfo/gstreamer-devel
>>
>> --
>>
>> Olivier Crête olivier.crete at collabora.com
>>
>>
>> _______________________________________________
>> 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/20160913/f1c9cb07/attachment.html>