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Opus via RTP

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MikeSI

Opus via RTP

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"

I always get not linked -1 and Internal data flow error.

Error:
/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"
ERROR: from element /GstPipeline:pipeline0/GstUDPSrc:udpsrc0: Internal data flow error.
Additional debug info:
gstbasesrc.c(2948): gst_base_src_loop (): /GstPipeline:pipeline0/GstUDPSrc:udpsrc0:
streaming task paused, reason not-linked (-1)

I have no problem with VP8 via RTP.

Sebastian Dröge-3

Re: Opus via RTP

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
[hidden email]
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MikeSI

Re: Opus via RTP

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

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/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"

ERROR: from element /GstPipeline:pipeline0/GstUDPSrc:udpsrc0: Internal data flow error.

Additional debug info:

gstbasesrc.c(2948): gst_base_src_loop (): /GstPipeline:pipeline0/GstUDPSrc:udpsrc0:

streaming task paused, reason not-linked (-1)

Execution ended after 0:00:00.016976512

Setting pipeline to PAUSED ...

Setting pipeline to READY ...

Setting pipeline to NULL ...

Freeing pipeline ...

[root@videodev : ~> 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

On Sun, Sep 11, 2016 at 2:12 PM, Sebastian Dröge-3 [via GStreamer-devel] <[hidden email]> 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
[hidden email]
https://lists.freedesktop.org/mailman/listinfo/gstreamer-devel

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MikeSI

Re: Opus via RTP

In reply to this post by Sebastian Dröge-3

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/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/GstOpusDec:opusdec0.GstPad:src: caps = "audio/x-raw\,\ format\=$string$S16LE\,\ layout\=$string$interleaved\,\ rate\=$int$48000\,\ channels\=$int$2\,\ channel-mask\=$bitmask$0x0000000000000003"

/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$0x0000000000000003"

/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$0x0000000000000003"

ERROR: from element /GstPipeline:pipeline0/GstUDPSrc:udpsrc0: Internal data flow error.

Additional debug info:

gstbasesrc.c(2948): gst_base_src_loop (): /GstPipeline:pipeline0/GstUDPSrc: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 <[hidden email]> 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
[hidden email]
https://lists.freedesktop.org/mailman/listinfo/gstreamer-devel

_______________________________________________
gstreamer-devel mailing list
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Olivier Crête-3

Re: Opus via RTP

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$0x0000000000000003"
/GstPipeline:pipeline0/GstAudioConvert:audioconvert0.GstPad:src: caps = "audio/x-raw\,\ format\=$string$S16LE\,\ layout\=$string$interleaved\,\ rate\=$int$48000\,\ channels\=$int$2\,\ channel-mask\=$bitmask$0x0000000000000003"
/GstPipeline:pipeline0/GstAudioConvert:audioconvert0.GstPad:sink: caps = "audio/x-raw\,\ format\=$string$S16LE\,\ layout\=$string$interleaved\,\ rate\=$int$48000\,\ channels\=$int$2\,\ channel-mask\=$bitmask$0x0000000000000003"
ERROR: from element /GstPipeline:pipeline0/GstUDPSrc:udpsrc0: Internal data flow error.
Additional debug info:
gstbasesrc.c(2948): gst_base_src_loop (): /GstPipeline:pipeline0/GstUDPSrc: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 <[hidden email]> 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
[hidden email]
https://lists.freedesktop.org/mailman/listinfo/gstreamer-devel
_______________________________________________
gstreamer-devel mailing list
[hidden email]
https://lists.freedesktop.org/mailman/listinfo/gstreamer-devel

-- 
Olivier Crête
[hidden email]

_______________________________________________
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[hidden email]
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MikeSI

Re: Opus via RTP

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 <[hidden email]> 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$0x0000000000000003"
/GstPipeline:pipeline0/GstAudioConvert:audioconvert0.GstPad:src: caps = "audio/x-raw\,\ format\=$string$S16LE\,\ layout\=$string$interleaved\,\ rate\=$int$48000\,\ channels\=$int$2\,\ channel-mask\=$bitmask$0x0000000000000003"
/GstPipeline:pipeline0/GstAudioConvert:audioconvert0.GstPad:sink: caps = "audio/x-raw\,\ format\=$string$S16LE\,\ layout\=$string$interleaved\,\ rate\=$int$48000\,\ channels\=$int$2\,\ channel-mask\=$bitmask$0x0000000000000003"
ERROR: from element /GstPipeline:pipeline0/GstUDPSrc:udpsrc0: Internal data flow error.
Additional debug info:
gstbasesrc.c(2948): gst_base_src_loop (): /GstPipeline:pipeline0/GstUDPSrc: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 <[hidden email]> 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
[hidden email]
https://lists.freedesktop.org/mailman/listinfo/gstreamer-devel

_______________________________________________ gstreamer-devel mailing list [hidden email] https://lists.freedesktop.org/mailman/listinfo/gstreamer-devel
--
Olivier Crête [hidden email]
_______________________________________________
gstreamer-devel mailing list
[hidden email]
https://lists.freedesktop.org/mailman/listinfo/gstreamer-devel

_______________________________________________
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[hidden email]
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MikeSI

Re: Opus via RTP

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 <[hidden email]> 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 <[hidden email]> 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$0x0000000000000003"
/GstPipeline:pipeline0/GstAudioConvert:audioconvert0.GstPad:src: caps = "audio/x-raw\,\ format\=$string$S16LE\,\ layout\=$string$interleaved\,\ rate\=$int$48000\,\ channels\=$int$2\,\ channel-mask\=$bitmask$0x0000000000000003"
/GstPipeline:pipeline0/GstAudioConvert:audioconvert0.GstPad:sink: caps = "audio/x-raw\,\ format\=$string$S16LE\,\ layout\=$string$interleaved\,\ rate\=$int$48000\,\ channels\=$int$2\,\ channel-mask\=$bitmask$0x0000000000000003"
ERROR: from element /GstPipeline:pipeline0/GstUDPSrc:udpsrc0: Internal data flow error.
Additional debug info:
gstbasesrc.c(2948): gst_base_src_loop (): /GstPipeline:pipeline0/GstUDPSrc: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 <[hidden email]> 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
[hidden email]
https://lists.freedesktop.org/mailman/listinfo/gstreamer-devel

_______________________________________________ gstreamer-devel mailing list [hidden email] https://lists.freedesktop.org/mailman/listinfo/gstreamer-devel
--
Olivier Crête [hidden email]
_______________________________________________
gstreamer-devel mailing list
[hidden email]
https://lists.freedesktop.org/mailman/listinfo/gstreamer-devel

_______________________________________________
gstreamer-devel mailing list
[hidden email]
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