GStreamer-devel

Recipe 'libpng' failed at the build step 'compile'

Classic

List

Threaded

9 messages Options

ran.attias

Recipe 'libpng' failed at the build step 'compile'

Hi all,

I'm straggling to go pass cross compilation fro androind.
I run on ubutu 16.04 with latest version:
https://gitlab.freedesktop.org/gstreamer/cerbero
Yes I'm getting next error:

===============================================

In file included from pngerror.c:19:
./pngpriv.h:920:4: error: ZLIB_VERNUM != PNG_ZLIB_VERNUM "-I (include
path) error: see the notes in pngpriv.h"
# error ZLIB_VERNUM != PNG_ZLIB_VERNUM \
^
1 error generated.
Makefile:1212: recipe for target 'pngerror.lo' failed
make[1]: *** [pngerror.lo] Error 1
libtool: compile:
/home/ra/temp_ran/cerbero/build/android-ndk-21/toolchains/llvm/prebuilt/linux-x86_64/bin/clang
-target x86_64-none-linux-android --sysroot
/home/ra/temp_ran/cerbero/build/android-ndk-21/platforms/android-21/arch-x86_64
-DHAVE_CONFIG_H -I. -DANDROID -DPIC -D__ANDROID_API__=21 -Wall -g -Os
-target x86_64-none-linux-android --sysroot
/home/ra/temp_ran/cerbero/build/android-ndk-21/platforms/android-21/arch-x86_64
-gcc-toolchain
/home/ra/temp_ran/cerbero/build/android-ndk-21/toolchains/x86_64-4.9/prebuilt/linux-x86_64
-isysroot /home/ra/temp_ran/cerbero/build/android-ndk-21/sysroot -isystem
/home/ra/temp_ran/cerbero/build/dist/android_universal/x86_64/include
-isystem /home/ra/temp_ran/cerbero/build/android-ndk-21/sysroot/usr/include
-isystem
/home/ra/temp_ran/cerbero/build/android-ndk-21/sysroot/usr/include/x86_64-linux-android
-fno-integrated-as -ffunction-sections -funwind-tables
-fstack-protector-strong -no-canonical-prefixes -fPIC
-Wno-invalid-command-line-argument -Wno-unused-command-line-argument
-DANDROID -DPIC -D__ANDROID_API__=21 -Wa,--noexecstack -Wall -g -Os -MT
pngget.lo -MD -MP -MF .deps/pngget.Tpo -c pngget.c -fPIC -DPIC -o
.libs/pngget.o
In file included from pngget.c:15:
./pngpriv.h:920:4: error: ZLIB_VERNUM != PNG_ZLIB_VERNUM "-I (include
path) error: see the notes in pngpriv.h"
# error ZLIB_VERNUM != PNG_ZLIB_VERNUM \
^
1 error generated.
Makefile:1212: recipe for target 'pngget.lo' failed
make[1]: *** [pngget.lo] Error 1
make[1]: Leaving directory
'/home/ra/temp_ran/cerbero/build/sources/android_universal/x86_64/libpng-1.6.37'
Makefile:820: recipe for target 'all' failed
make: *** [all] Error 2

Recipe 'libpng' failed at the build step 'compile'
Command Error: Running ['make', 'V=1', '-j4'] returned 2

============================

Can someone please advise?

Thanks

--
Sent from: http://gstreamer-devel.966125.n4.nabble.com/
_______________________________________________
gstreamer-devel mailing list
[hidden email]
https://lists.freedesktop.org/mailman/listinfo/gstreamer-devel

Patrick Cusack

Streaming audio and video RTP

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 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 port=5054 \

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

William

Re: Streaming audio and video RTP

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 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 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 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 port=5054 \
_______________________________________________
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

Patrick Cusack

Re: Streaming audio and video RTP

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 <[hidden email]> 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 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 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 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 port=5054 \
_______________________________________________
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

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

Patrick Cusack

Re: Streaming audio and video RTP

In reply to this post by William

This is a new iteration on the script. I can send video successfully, but the audio does not register on my receiver.

gst-launch-1.0 \

        rtpbin name=rtpbin \

        filesrc location=starwars.mov ! qtdemux name=demux \

        demux.audio_0 ! decodebin ! audioconvert ! opusenc bandwidth=superwideband bitrate=96000 \

                ! rtpopuspay ! "application/x-rtp, media=(string)audio, clock-rate=(int)48000, encoding-name=(string)OPUS,payload=111"  ! rtpbin.send_rtp_sink_0 \

        demux.video_0 ! decodebin ! videoconvert ! x264enc tune=zerolatency speed-preset=1 dct8x8=true quantizer=25 pass=qual \

                ! rtph264pay ! "application/x-rtp,payload=(int)103,clock-rate=(int)90000"  ! rtpbin.send_rtp_sink_1 \

        rtpbin.send_rtp_src_0 ! udpsink host=www.myurl.com port=5052 sync=true async=false \

        rtpbin.send_rtcp_src_0 ! udpsink host=www.myurl.com port=5053 sync=false async=false \

        rtpbin.send_rtp_src_1 ! udpsink host=www.myurl.com port=5054 sync=true async=false \

        rtpbin.send_rtcp_src_1 ! udpsink host=www.myurl.com port=5055 sync=false async=false

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

William

Re: Streaming audio and video RTP

In reply to this post by Patrick Cusack

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 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 \

! udpsink host=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 <[hidden email]> 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 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 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 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 port=5054 \
_______________________________________________
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
_______________________________________________
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

Patrick Cusack

Re: Streaming audio and video RTP

Hmm….again no audio comes through. I am wondering if my qtdemux is correct.

On Apr 29, 2020, at 12:29 PM, William Johnston <[hidden email]> 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 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 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 <[hidden email]> 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 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 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 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 port=5054 \
_______________________________________________
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
_______________________________________________
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

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

Patrick Cusack

Re: Streaming audio and video RTP

This works to stream picture and audio via rtp...

gst-launch-1.0  \

	filesrc location=starwars.mov \

	! qtdemux name=demux \

	demux.audio_0 \

	! queue max-size-time = 3000000000 \

	! decodebin \

	! audioconvert \

	! audioresample \

	! audio/x-raw,channels=2,rate=48000 \

	! opusenc bitrate=96000 \

	! rtpopuspay \

	! udpsink host=www.myurl.com port=5052 \

	demux.video_0 \

	! queue \

	! decodebin \

	! videoconvert \

	! videorate \

	! x264enc speed-preset=ultrafast tune=zerolatency byte-stream=true key-int-max=60 \

	! video/x-h264, profile=baseline \

	! queue \

	! rtph264pay \

	! udpsink host=www.myurl.com port=5054

On Apr 29, 2020, at 8:49 PM, Patrick Cusack <[hidden email]> 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 <[hidden email]> 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 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 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 <[hidden email]> 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 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 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 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 port=5054 \
_______________________________________________
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
_______________________________________________
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

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

William

Re: Streaming audio and video RTP

In reply to this post by Patrick Cusack

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 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 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 <[hidden email]> 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 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 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 <[hidden email]> 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 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 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 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 port=5054 \
_______________________________________________
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
_______________________________________________
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
_______________________________________________
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