GStreamer issueshttps://gitlab.freedesktop.org/groups/gstreamer/-/issues2019-07-06T04:55:01Zhttps://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/issues/1007webrtcbin: webrtc_sendrecv pipeline won't start on rpi3b+ when using omx264enc2019-07-06T04:55:01ZTyler Brookswebrtcbin: webrtc_sendrecv pipeline won't start on rpi3b+ when using omx264encI have built gstreamer/webrtcbn 1.14 on my rpi3b+. The 'sendrecv' demo from gstwebrtc-demo works just fine. When I modify it to capture video with v4l2src it also works fine. However, when I replace the encoder with the omx h264 encod...I have built gstreamer/webrtcbn 1.14 on my rpi3b+. The 'sendrecv' demo from gstwebrtc-demo works just fine. When I modify it to capture video with v4l2src it also works fine. However, when I replace the encoder with the omx h264 encoder the pipeline won't start. I never get the 'on-negotiation-needed' callback.
To be specific, this pipeline works (the original code in webrtc_sendrecv.c):
```
pipe1 =
gst_parse_launch ("webrtcbin bundle-policy=max-bundle name=sendrecv " STUN_SERVER
"videotestsrc is-live=true pattern=ball ! videoconvert ! queue ! vp8enc deadline=1 ! rtpvp8pay ! "
"queue ! " RTP_CAPS_VP8 "96 ! sendrecv. "
"audiotestsrc is-live=true wave=red-noise ! audioconvert ! audioresample ! queue ! opusenc ! rtpopuspay ! "
"queue ! " RTP_CAPS_OPUS "97 ! sendrecv. ",
&error);
```
This pipeline also works (capturing with v4l2src):
```
pipe1 =
gst_parse_launch ("webrtcbin bundle-policy=max-bundle name=sendrecv " STUN_SERVER
"v4l2src ! video/x-raw,width=640,height=480,framerate=20/1 ! videoflip video-direction=180 ! "
"queue ! vp8enc deadline=1 ! rtpvp8pay ! queue ! " RTP_CAPS_VP8 "96 ! sendrecv. "
, &error);
```
However, this pipeline does not work (encoding with omxh264enc):
```
pipe1 =
gst_parse_launch ("webrtcbin bundle-policy=max-bundle name=sendrecv " STUN_SERVER
"v4l2src ! video/x-raw,width=640,height=480,framerate=20/1 ! videoflip video-direction=180 ! "
"queue ! omxh264enc ! "queue ! rtph264pay ! " RTP_CAPS_H264 "127 ! sendrecv. "
, &error);
```
STUN_SERVER, RTP_CAPS_VP8, & RTP_CAPS_H264 look like this:
```
#define STUN_SERVER " stun-server=stun://stun.l.google.com:19302 "
#define RTP_CAPS_OPUS "application/x-rtp,media=audio,encoding-name=OPUS,payload="
#define RTP_CAPS_VP8 "application/x-rtp,media=video,encoding-name=VP8,payload="
#define RTP_CAPS_H264 "application/x-rtp,media=video,encoding-name=H264,payload="
```
It seems like the SDP never gets calculated. I never get the 'on-negotiation-needed' callback. The 'start_pipeline()' function does finish successfully and the pipeline object state is set to 'PLAYING'.
Any ideas as to what could be wrong?
Oh, this pipeline does write an h264 elemental stream to a file that I can playback with vlc... so I think my omx h264 encoder is working:
```
gst-launch-1.0 v4l2src ! video/x-raw,width=640,height=480,framerate=20/1 ! videoflip video-direction=180 ! queue ! omxh264enc ! filesink location=test.h264
```https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/issues/63gst_pad_link() hangs when called from "pad-added" handler of togglerecord2019-07-07T09:09:24ZNikola Hadžićgst_pad_link() hangs when called from "pad-added" handler of togglerecordThe issue occurs if I request a sink pad of toggle record like this
```
new_sink_pad = gst_element_get_request_pad(togglerecord, "sink_%u");
g_signal_connect(pause, "pad-added", G_CALLBACK(pad_added_cb), pad_to_link);
```
Now I want to ...The issue occurs if I request a sink pad of toggle record like this
```
new_sink_pad = gst_element_get_request_pad(togglerecord, "sink_%u");
g_signal_connect(pause, "pad-added", G_CALLBACK(pad_added_cb), pad_to_link);
```
Now I want to link newly created pad and `pad_to_link` in the "pad-added" handler function.
```
void pad_added_cb(GstElement *pause, GstPad *new_pad, GstPad *next_pad)
{
...
gst_pad_link(new_pad, next_pad);
...
}
```
But `gst_pad_link()` never returns.
Also backtrace is provided.
[bt.txt](/uploads/eea08244dba8502d7d512a0cd34863bd/bt.txt)https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/issues/69rtsp-server: RTP & RTCP packets in ONE udp port2023-10-17T17:48:59ZRenat Vakhitovrtsp-server: RTP & RTCP packets in ONE udp portHello everyone, sorry for my bad english, I am from Russia :)
At work it became necessary to use ONE udp port for RTP and RTCP packets. I found RFC about this: RFC5761. And as I understand through Google it may look like this: ...PAY -...Hello everyone, sorry for my bad english, I am from Russia :)
At work it became necessary to use ONE udp port for RTP and RTCP packets. I found RFC about this: RFC5761. And as I understand through Google it may look like this: ...PAY -> RTPBIN.send_rtp... RTPBIN.send_rtcp... -> FUNNEL -> UDPSINK.
Is this feature implemented in the gst-rtsp-server? Or need to add this feature yourself?
Best regards.https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/issues/1006Support for the DeckLink Quad HDMI Recorder2019-07-04T17:26:16ZStirling WestrupSupport for the DeckLink Quad HDMI RecorderThis card is currently not supported:
`[root@primary-1 ~]# gst-launch-1.0 decklinkvideosrc device-number=3 ! videoconvert ! videoscale ! video/x-raw,width=1920, height=1080 ! ximagesink display=:0`
`Setting pipeline to PAUSED ......This card is currently not supported:
`[root@primary-1 ~]# gst-launch-1.0 decklinkvideosrc device-number=3 ! videoconvert ! videoscale ! video/x-raw,width=1920, height=1080 ! ximagesink display=:0`
`Setting pipeline to PAUSED ...`
`Pipeline is live and does not need PREROLL ...`
`Setting pipeline to PLAYING ...`
`New clock: GstSystemClock`
`Redistribute latency...`
`WARNING: from element /GstPipeline:pipeline0/GstDecklinkVideoSrc:decklinkvideosrc0: No signal`
`Additional debug info:`
`gstdecklinkvideosrc.cpp(886): gst_decklink_video_src_create (): /GstPipeline:pipeline0/GstDecklinkVideoSrc:decklinkvideosrc0:`
`No input source was detected - video frames invalid`
`**`
`ERROR:gstdecklink.cpp:434:const GstDecklinkModeEnum gst_decklink_get_mode_enum_from_bmd(BMDDisplayMode): code should not be reached`
`Aborted`
One of our team members reported: "Went through gstreamer decklinkvideosrc code briefly. For my understanding, It looks like the Video display mode generated from this kind of device is not supported by the DeckLinkModeAPI used by decklinkvideosrc element."
So then we tried explicitly setting the mode:
`gst-launch-1.0 --gst-debug=decklinkvideosrc:6 decklinkvideosrc device-number=3 mode=1080p25 ! videoconvert ! videoscale ! video/x-raw,width=1920, height=1080 ! ximagesink display=:0`
We ended up with a black screen, which sounds like the DeckLink duplex bug that is already being worked on.
If there are any tests you would like us to do, or information to gather, please let me know.https://gitlab.freedesktop.org/gstreamer/gst-plugins-base/-/issues/633overlaycomposition: crash with access violation when using a pool of two buff...2019-07-02T21:07:22ZAaron Boxeroverlaycomposition: crash with access violation when using a pool of two buffers in "draw" callback(on windows 32 bit, btw)
create a pool of two buffers sized to the video window, and alternate between the two buffers
on the "draw" callback:
1. write to buffer 0 and attach buffer 0 via gst_video_overlay_rectangle_new_raw()
1. write t...(on windows 32 bit, btw)
create a pool of two buffers sized to the video window, and alternate between the two buffers
on the "draw" callback:
1. write to buffer 0 and attach buffer 0 via gst_video_overlay_rectangle_new_raw()
1. write to buffer 1 and attach buffer 1 via gst_video_overlay_rectangle_new_raw()
1. write to buffer 0 and attach buffer 0 via gst_video_overlay_rectangle_new_raw()
I am using a pool so I can sometimes make use of the "caching" feature, where an overlay buffer is cached when the pointer
doesn't change, but I can also change the buffer by cycling through the two different buffers, so i can refresh the display.
If I don't cycle between buffers, and always use buffer 0, then there is no crash.https://gitlab.freedesktop.org/gstreamer/gst-devtools/-/issues/45gst-validate build fails with "undefined reference to `g_enum_to_string'"2019-07-29T11:47:42ZStéphane Cerveauscerveau@igalia.comgst-validate build fails with "undefined reference to `g_enum_to_string'"On master branch, gst-build fails with undefined reference to `g_enum_to_string'On master branch, gst-build fails with undefined reference to `g_enum_to_string'https://gitlab.freedesktop.org/gstreamer/cerbero/-/issues/172master: some dlls not loading with mingw 32-bit cross compile for windows2020-01-20T15:34:01ZAaron Boxermaster: some dlls not loading with mingw 32-bit cross compile for windows(I don't see these warnings with the 1.16 cross-compile)
Error popup: `Bad Image: libgmp-10.dll is either not designed to run on WIndows or it contains an error ......`
and ...
```
0:00:00.039979300 7212 05CF1720 WARN ...(I don't see these warnings with the 1.16 cross-compile)
Error popup: `Bad Image: libgmp-10.dll is either not designed to run on WIndows or it contains an error ......`
and ...
```
0:00:00.039979300 7212 05CF1720 WARN GST_PLUGIN_LOADING gstplugin.c:806:_priv_gst_plugin_load_file_for_registry: module_open failed: 'C:\gstreamer\1.0\loki_x86\lib\gstreamer-1.0\libgstdecklink.dll': The specified module could not be found.
GStreamer-WARNING **: 12:13:30.626: Failed to load plugin 'C:\gstreamer\1.0\loki_x86\lib\gstreamer-1.0\libgstdecklink.dll': 'C:\gstreamer\1.0\loki_x86\lib\gstreamer-1.0\libgstdecklink.dll': The specified module could not be found.
0:02:06.615574300 7212 05CF1720 WARN GST_PLUGIN_LOADING gstplugin.c:806:_priv_gst_plugin_load_file_for_registry: module_open failed: 'C:\gstreamer\1.0\loki_x86\lib\gstreamer-1.0\libgstnice.dll': %1 is not a valid Win32 application.
GStreamer-WARNING **: 12:15:37.215: Failed to load plugin 'C:\gstreamer\1.0\loki_x86\lib\gstreamer-1.0\libgstnice.dll': 'C:\gstreamer\1.0\loki_x86\lib\gstreamer-1.0\libgstnice.dll': %1 is not a valid Win32 application.
0:02:06.619956800 7212 05CF1720 WARN GST_PLUGIN_LOADING gstplugin.c:806:_priv_gst_plugin_load_file_for_registry: module_open failed: 'C:\gstreamer\1.0\loki_x86\lib\gstreamer-1.0\libgstopenh264.dll': The specified module could not be found.
GStreamer-WARNING **: 12:15:37.220: Failed to load plugin 'C:\gstreamer\1.0\loki_x86\lib\gstreamer-1.0\libgstopenh264.dll': 'C:\gstreamer\1.0\loki_x86\lib\gstreamer-1.0\libgstopenh264.dll': The specified module could not be found.
0:02:07.670836700 7212 05CF1720 WARN GST_PLUGIN_LOADING gstplugin.c:806:_priv_gst_plugin_load_file_for_registry: module_open failed: 'C:\gstreamer\1.0\loki_x86\lib\gstreamer-1.0\libgstrtmp.dll': %1 is not a valid Win32 application.
(EatonRTSPTestHarness.exe:7212): GStreamer-WARNING **: 12:15:38.272: Failed to load plugin 'C:\gstreamer\1.0\loki_x86\lib\gstreamer-1.0\libgstrtmp.dll': 'C:\gstreamer\1.0\loki_x86\lib\gstreamer-1.0\libgstrtmp.dll': %1 is not a valid Win32 application.
0:02:07.692571200 7212 05CF1720 WARN GST_PLUGIN_LOADING gstplugin.c:806:_priv_gst_plugin_load_file_for_registry: module_open failed: 'C:\gstreamer\1.0\loki_x86\lib\gstreamer-1.0\libgstsoundtouch.dll': The specified module could not be found.
GStreamer-WARNING **: 12:15:38.293: Failed to load plugin 'C:\gstreamer\1.0\loki_x86\lib\gstreamer-1.0\libgstsoundtouch.dll': 'C:\gstreamer\1.0\loki_x86\lib\gstreamer-1.0\libgstsoundtouch.dll': The specified module could not be found.
0:02:07.707742600 7212 05CF1720 WARN GST_PLUGIN_LOADING gstplugin.c:806:_priv_gst_plugin_load_file_for_registry: module_open failed: 'C:\gstreamer\1.0\loki_x86\lib\gstreamer-1.0\libgstsrt.dll': The specified module could not be found.
GStreamer-WARNING **: 12:15:38.307: Failed to load plugin 'C:\gstreamer\1.0\loki_x86\lib\gstreamer-1.0\libgstsrt.dll': 'C:\gstreamer\1.0\loki_x86\lib\gstreamer-1.0\libgstsrt.dll': The specified module could not be found.
0:02:07.713073300 7212 05CF1720 WARN GST_PLUGIN_LOADING gstplugin.c:806:_priv_gst_plugin_load_file_for_registry: module_open failed: 'C:\gstreamer\1.0\loki_x86\lib\gstreamer-1.0\libgsttaglib.dll': The specified module could not be found.
GStreamer-WARNING **: 12:15:38.313: Failed to load plugin 'C:\gstreamer\1.0\loki_x86\lib\gstreamer-1.0\libgsttaglib.dll': 'C:\gstreamer\1.0\loki_x86\lib\gstreamer-1.0\libgsttaglib.dll': The specified module could not be found.
0:02:08.744093400 7212 05CF1720 WARN GST_PLUGIN_LOADING gstplugin.c:806:_priv_gst_plugin_load_file_for_registry: module_open failed: 'C:\gstreamer\1.0\loki_x86\lib\gstreamer-1.0\libgstwebrtc.dll': %1 is not a valid Win32 application.
GStreamer-WARNING **: 12:15:39.344: Failed to load plugin 'C:\gstreamer\1.0\loki_x86\lib\gstreamer-1.0\libgstwebrtc.dll': 'C:\gstreamer\1.0\loki_x86\lib\gstreamer-1.0\libgstwebrtc.dll': %1 is not a valid Win32 application.
0:02:08.747653900 7212 05CF1720 WARN GST_PLUGIN_LOADING gstplugin.c:806:_priv_gst_plugin_load_file_for_registry: module_open failed: 'C:\gstreamer\1.0\loki_x86\lib\gstreamer-1.0\libgstwebrtcdsp.dll': The specified module could not be found.
GStreamer-WARNING **: 12:15:39.346: Failed to load plugin 'C:\gstreamer\1.0\loki_x86\lib\gstreamer-1.0\libgstwebrtcdsp.dll': 'C:\gstreamer\1.0\loki_x86\lib\gstreamer-1.0\libgstwebrtcdsp.dll': The specified module could not be found.
0:02:08.751831800 7212 05CF1720 WARN GST_PLUGIN_LOADING gstplugin.c:806:_priv_gst_plugin_load_file_for_registry: module_open failed: 'C:\gstreamer\1.0\loki_x86\lib\gstreamer-1.0\libgstzbar.dll': The specified module could not be found.
GStreamer-WARNING **: 12:15:39.352: Failed to load plugin 'C:\gstreamer\1.0\loki_x86\lib\gstreamer-1.0\libgstzbar.dll': 'C:\gstreamer\1.0\loki_x86\lib\gstreamer-1.0\libgstzbar.dll': The specified module could not be found.
```https://gitlab.freedesktop.org/gstreamer/gstreamer-vaapi/-/issues/178VP9: totem crashes with videos of VP9 format on scrubbing2019-06-28T13:51:39ZUmang JainVP9: totem crashes with videos of VP9 format on scrubbingflatpak version of GNOME Videos (totem-3.32) crashes when I scrub the player's slider on a video of VP9 codec format.
I can reproduce this on a VP9 sample [here](https://base-n.de/webm/out9.webm):
Plus with all additional screencasts re...flatpak version of GNOME Videos (totem-3.32) crashes when I scrub the player's slider on a video of VP9 codec format.
I can reproduce this on a VP9 sample [here](https://base-n.de/webm/out9.webm):
Plus with all additional screencasts recording I do on Stock GNOME (shift + ctrl + Alt + r key combination).
The backtrace points to the crash in `gst_vp9_parser_parse_frame_header` : [backtrace](/uploads/673f709d0b2890eef5a76c5266abe941/backtrace)
Totem's flatpak seems to have GStreamer base version as `1.14.4`. It is coming via [freedesktop-sdk](https://gitlab.com/freedesktop-sdk/freedesktop-sdk/blob/18.08/elements/desktop/gstreamer.bst).
I wrote about this issue at length here too: https://gitlab.gnome.org/GNOME/totem/issues/335https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/issues/1005srtserversrc yields malformed rtp payloads in 1.162019-07-09T01:18:13ZJason Pereirasrtserversrc yields malformed rtp payloads in 1.16**Example Pipelines Gstreamer 1.16.0**
*Receiver / Server*
`gst-launch-1.0 srtsrc mode=listener uri=srt://:8888/ ! udpsink host=127.0.0.1 port=5060 `
*Sender / Client*
`gst-launch-1.0 videotestsrc ! video/x-raw, height=1080, width=1920...**Example Pipelines Gstreamer 1.16.0**
*Receiver / Server*
`gst-launch-1.0 srtsrc mode=listener uri=srt://:8888/ ! udpsink host=127.0.0.1 port=5060 `
*Sender / Client*
`gst-launch-1.0 videotestsrc ! video/x-raw, height=1080, width=1920 ! videoconvert ! vp9enc ! rtpvp9pay mtu=1300 ! srtsink mode=caller uri=srt://127.0.0.1:8888/`
**Example Pipelines Gstreamer 1.14.5**
*Receiver / Server*
`gst-launch-1.0 srtserversrc uri=srt://:8888/ ! udpsink host=127.0.0.1 port=5060 `
*Sender / Client*
`gst-launch-1.0 videotestsrc ! video/x-raw, height=1080, width=1920 ! videoconvert ! vp9enc ! rtpvp9pay mtu=1300 ! srtclientsink uri=srt://127.0.0.1:8888/`
**Results **(tcpdump -i lo udp port 5060 -s 0 -w test.pcap)
*Gstreamer 1.16.0* has oversized payloads and rtp stream has missing packets from the sequence, exactly 3 missing in every case, the assumption being they are somehow being combined prior to reaching the udpsink
```
"No.","Time","Source","Destination","Protocol","Length","Info"
"1","0.000000","127.0.0.1","127.0.0.1","RTP","3932","PT=DynamicRTP-Type-96, SSRC=0x680398C, Seq=21477, Time=1747653689"
"2","0.000079","127.0.0.1","127.0.0.1","RTP","3932","PT=DynamicRTP-Type-96, SSRC=0x680398C, Seq=21480, Time=1747653689"
"3","0.000098","127.0.0.1","127.0.0.1","RTP","3932","PT=DynamicRTP-Type-96, SSRC=0x680398C, Seq=21483, Time=1747653689"
"4","0.000116","127.0.0.1","127.0.0.1","RTP","3932","PT=DynamicRTP-Type-96, SSRC=0x680398C, Seq=21486, Time=1747653689"
"5","0.000167","127.0.0.1","127.0.0.1","RTP","189","PT=DynamicRTP-Type-96, SSRC=0x680398C, Seq=21489, Time=1747653689, Mark"
"6","1.105789","127.0.0.1","127.0.0.1","RTP","3932","PT=DynamicRTP-Type-96, SSRC=0x680398C, Seq=21490, Time=1747656688"
"7","1.105824","127.0.0.1","127.0.0.1","RTP","3932","PT=DynamicRTP-Type-96, SSRC=0x680398C, Seq=21493, Time=1747656688"
"8","1.105848","127.0.0.1","127.0.0.1","RTP","3932","PT=DynamicRTP-Type-96, SSRC=0x680398C, Seq=21496, Time=1747656688"
"9","1.105902","127.0.0.1","127.0.0.1","RTP","3932","PT=DynamicRTP-Type-96, SSRC=0x680398C, Seq=21499, Time=1747656688"
"10","1.106012","127.0.0.1","127.0.0.1","RTP","208","PT=DynamicRTP-Type-96, SSRC=0x680398C, Seq=21502, Time=1747656688, Mark"
```
*Gstreamer 1.14.5* entirely normal payload sizes, no packets missing from sequence
```
"No.","Time","Source","Destination","Protocol","Length","Info"
"1","0.000000","127.0.0.1","127.0.0.1","RTP","1342","PT=DynamicRTP-Type-96, SSRC=0xA5EE7B15, Seq=11967, Time=3514845691"
"2","0.000032","127.0.0.1","127.0.0.1","RTP","1334","PT=DynamicRTP-Type-96, SSRC=0xA5EE7B15, Seq=11968, Time=3514845691"
"3","0.000042","127.0.0.1","127.0.0.1","RTP","1334","PT=DynamicRTP-Type-96, SSRC=0xA5EE7B15, Seq=11969, Time=3514845691"
"4","0.000049","127.0.0.1","127.0.0.1","RTP","1334","PT=DynamicRTP-Type-96, SSRC=0xA5EE7B15, Seq=11970, Time=3514845691"
"5","0.000057","127.0.0.1","127.0.0.1","RTP","1334","PT=DynamicRTP-Type-96, SSRC=0xA5EE7B15, Seq=11971, Time=3514845691"
"6","0.000064","127.0.0.1","127.0.0.1","RTP","1334","PT=DynamicRTP-Type-96, SSRC=0xA5EE7B15, Seq=11972, Time=3514845691"
"7","0.000071","127.0.0.1","127.0.0.1","RTP","1334","PT=DynamicRTP-Type-96, SSRC=0xA5EE7B15, Seq=11973, Time=3514845691"
"8","0.000078","127.0.0.1","127.0.0.1","RTP","1334","PT=DynamicRTP-Type-96, SSRC=0xA5EE7B15, Seq=11974, Time=3514845691"
"9","0.000085","127.0.0.1","127.0.0.1","RTP","1334","PT=DynamicRTP-Type-96, SSRC=0xA5EE7B15, Seq=11975, Time=3514845691"
"10","0.000092","127.0.0.1","127.0.0.1","RTP","1334","PT=DynamicRTP-Type-96, SSRC=0xA5EE7B15, Seq=11976, Time=3514845691"
```
*Gstreamer 1.16.0* using `srtserversrc mode=listener blocksize=1500` on receiver pipeline, normal payload sizes, no packets missing from sequence
```
"No.","Time","Source","Destination","Protocol","Length","Info"
"1","0.000000","127.0.0.1","127.0.0.1","RTP","1332","PT=DynamicRTP-Type-96, SSRC=0x2CEB20BE, Seq=30783, Time=2615734653"
"2","0.000042","127.0.0.1","127.0.0.1","RTP","1332","PT=DynamicRTP-Type-96, SSRC=0x2CEB20BE, Seq=30784, Time=2615734653"
"3","0.000070","127.0.0.1","127.0.0.1","RTP","1332","PT=DynamicRTP-Type-96, SSRC=0x2CEB20BE, Seq=30785, Time=2615734653"
"4","0.000114","127.0.0.1","127.0.0.1","RTP","1332","PT=DynamicRTP-Type-96, SSRC=0x2CEB20BE, Seq=30786, Time=2615734653"
"5","0.000136","127.0.0.1","127.0.0.1","RTP","1332","PT=DynamicRTP-Type-96, SSRC=0x2CEB20BE, Seq=30787, Time=2615734653"
"6","0.000152","127.0.0.1","127.0.0.1","RTP","1332","PT=DynamicRTP-Type-96, SSRC=0x2CEB20BE, Seq=30788, Time=2615734653"
"7","0.000165","127.0.0.1","127.0.0.1","RTP","1332","PT=DynamicRTP-Type-96, SSRC=0x2CEB20BE, Seq=30789, Time=2615734653"
"8","0.000181","127.0.0.1","127.0.0.1","RTP","1332","PT=DynamicRTP-Type-96, SSRC=0x2CEB20BE, Seq=30790, Time=2615734653"
"9","0.000195","127.0.0.1","127.0.0.1","RTP","1332","PT=DynamicRTP-Type-96, SSRC=0x2CEB20BE, Seq=30791, Time=2615734653"
"10","0.000209","127.0.0.1","127.0.0.1","RTP","1332","PT=DynamicRTP-Type-96, SSRC=0x2CEB20BE, Seq=30792, Time=2615734653"
```
I have not validated if this behaviour is present when srt is used in combination with other rtp payload formats, I think the workaround above is also not idea and likely should be addressed in some other way, let me know thoughts and I am also happy to provide further information if required.https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/issues/1004avfvideosrc: incorrect retina support2021-09-24T14:37:31ZRoman Shpuntovavfvideosrc: incorrect retina supportI have UHD monitor with resolution: 3840x2160. When I change resolution to "Scaled" inside System Settings to 3360x1890 I have the next pipeline:
```
MacBookPro:~ roman$ gst-launch-1.0 avfvideosrc capture-screen=true ! glimagesink sync=...I have UHD monitor with resolution: 3840x2160. When I change resolution to "Scaled" inside System Settings to 3360x1890 I have the next pipeline:
```
MacBookPro:~ roman$ gst-launch-1.0 avfvideosrc capture-screen=true ! glimagesink sync=false -vv
Setting pipeline to PAUSED ...
Pipeline is live and does not need PREROLL ...
Got context from element 'sink': gst.gl.GLDisplay=context, gst.gl.GLDisplay=(GstGLDisplay)"\(GstGLDisplayCocoa\)\ gldisplaycocoa0";
Setting pipeline to PLAYING ...
New clock: GstSystemClock
/GstPipeline:pipeline0/GstAVFVideoSrc:avfvideosrc0.GstPad:src: caps = video/x-raw, width=(int)6720, height=(int)3780, format=(string)UYVY, framerate=(fraction)2147483647/1
/GstPipeline:pipeline0/GstGLImageSinkBin:glimagesinkbin0.GstGhostPad:sink.GstProxyPad:proxypad0: caps = video/x-raw, width=(int)6720, height=(int)3780, format=(string)UYVY, framerate=(fraction)2147483647/1
/GstPipeline:pipeline0/GstGLImageSinkBin:glimagesinkbin0/GstGLUploadElement:gluploadelement0.GstPad:src: caps = video/x-raw(memory:GLMemory), width=(int)6720, height=(int)3780, format=(string)UYVY, framerate=(fraction)2147483647/1, texture-target=(string)2D
/GstPipeline:pipeline0/GstGLImageSinkBin:glimagesinkbin0/GstGLColorConvertElement:glcolorconvertelement0.GstPad:src: caps = video/x-raw(memory:GLMemory), width=(int)6720, height=(int)3780, format=(string)RGBA, framerate=(fraction)2147483647/1, texture-target=(string)2D
/GstPipeline:pipeline0/GstGLImageSinkBin:glimagesinkbin0/GstGLColorBalance:glcolorbalance0.GstPad:src: caps = video/x-raw(memory:GLMemory), width=(int)6720, height=(int)3780, format=(string)RGBA, framerate=(fraction)2147483647/1, texture-target=(string)2D
/GstPipeline:pipeline0/GstGLImageSinkBin:glimagesinkbin0/GstGLImageSink:sink.GstPad:sink: caps = video/x-raw(memory:GLMemory), width=(int)6720, height=(int)3780, format=(string)RGBA, framerate=(fraction)2147483647/1, texture-target=(string)2D
/GstPipeline:pipeline0/GstGLImageSinkBin:glimagesinkbin0/GstGLColorBalance:glcolorbalance0.GstPad:sink: caps = video/x-raw(memory:GLMemory), width=(int)6720, height=(int)3780, format=(string)RGBA, framerate=(fraction)2147483647/1, texture-target=(string)2D
/GstPipeline:pipeline0/GstGLImageSinkBin:glimagesinkbin0/GstGLColorConvertElement:glcolorconvertelement0.GstPad:sink: caps = video/x-raw(memory:GLMemory), width=(int)6720, height=(int)3780, format=(string)UYVY, framerate=(fraction)2147483647/1, texture-target=(string)2D
/GstPipeline:pipeline0/GstGLImageSinkBin:glimagesinkbin0/GstGLUploadElement:gluploadelement0.GstPad:sink: caps = video/x-raw, width=(int)6720, height=(int)3780, format=(string)UYVY, framerate=(fraction)2147483647/1
/GstPipeline:pipeline0/GstGLImageSinkBin:glimagesinkbin0.GstGhostPad:sink: caps = video/x-raw, width=(int)6720, height=(int)3780, format=(string)UYVY, framerate=(fraction)2147483647/1
/GstPipeline:pipeline0/GstGLImageSinkBin:glimagesinkbin0/GstGLUploadElement:gluploadelement0.GstPad:sink: caps = video/x-raw, width=(int)6720, height=(int)3780, format=(string)UYVY, framerate=(fraction)2147483647/1
^Chandling interrupt.
Interrupt: Stopping pipeline ...
Execution ended after 0:00:09.593902000
Setting pipeline to PAUSED ...
Setting pipeline to READY ...
Setting pipeline to NULL ...
Freeing pipeline ...
```
In system info I see:
```
Resolution: 6720 x 3780
Interface Type: 3360 x 1890 @ 60 Hz
```
This is incorrect behaviour, because when I try to capture and encode this stream I have the next error:
```
MacBookPro:~ roman$ gst-launch-1.0 avfvideosrc capture-screen=true ! vtenc_h264_hw ! fakesink -vv
Setting pipeline to PAUSED ...
Pipeline is live and does not need PREROLL ...
Setting pipeline to PLAYING ...
New clock: GstSystemClock
/GstPipeline:pipeline0/GstAVFVideoSrc:avfvideosrc0.GstPad:src: caps = video/x-raw, width=(int)6720, height=(int)3780, framerate=(fraction)2147483647/1, format=(string)UYVY
0:00:00.616859000 79505 0x7fec1e818ed0 ERROR vtenc vtenc.c:861:gst_vtenc_create_session:<vtenc_h264_hw0> VTCompressionSessionCreate() returned: -12915
0:00:00.710160000 79505 0x7fec1e818ed0 ERROR vtenc vtenc.c:861:gst_vtenc_create_session:<vtenc_h264_hw0> VTCompressionSessionCreate() returned: -12915
0:00:00.790690000 79505 0x7fec1e818ed0 ERROR vtenc vtenc.c:861:gst_vtenc_create_session:<vtenc_h264_hw0> VTCompressionSessionCreate() returned: -12915
0:00:00.861438000 79505 0x7fec1e818ed0 ERROR vtenc vtenc.c:861:gst_vtenc_create_session:<vtenc_h264_hw0> VTCompressionSessionCreate() returned: -12915
ERROR: from element /GstPipeline:pipeline0/GstAVFVideoSrc:avfvideosrc0: Internal data stream error.
Additional debug info:
../libs/gst/base/gstbasesrc.c(3064): gst_base_src_loop (): /GstPipeline:pipeline0/GstAVFVideoSrc:avfvideosrc0:
streaming stopped, reason not-negotiated (-4)
Execution ended after 0:00:00.817796000
Setting pipeline to PAUSED ...
Setting pipeline to READY ...
0:00:00.938246000 79505 0x7fec1e818ed0 ERROR vtenc vtenc.c:861:gst_vtenc_create_session:<vtenc_h264_hw0> VTCompressionSessionCreate() returned: -12915
Setting pipeline to NULL ...
Freeing pipeline ...
```
The real resolution is can not be more than 3840x2160. When I set "default resolution for this monitor" in System Settings I have:
```
MacBookPro:~ roman$ gst-launch-1.0 avfvideosrc capture-screen=true ! vtenc_h264_hw ! fakesink -vv
Setting pipeline to PAUSED ...
Pipeline is live and does not need PREROLL ...
Setting pipeline to PLAYING ...
New clock: GstSystemClock
/GstPipeline:pipeline0/GstAVFVideoSrc:avfvideosrc0.GstPad:src: caps = video/x-raw, width=(int)3840, height=(int)2160, framerate=(fraction)2147483647/1, format=(string)UYVY
/GstPipeline:pipeline0/vtenc_h264_hw:vtenc_h264_hw0.GstPad:sink: caps = video/x-raw, width=(int)3840, height=(int)2160, framerate=(fraction)2147483647/1, format=(string)UYVY
/GstPipeline:pipeline0/vtenc_h264_hw:vtenc_h264_hw0.GstPad:src: caps = video/x-h264, width=(int)3840, height=(int)2160, framerate=(fraction)2147483647/1, stream-format=(string)avc, alignment=(string)au, codec_data=(buffer)014d003cffe10019274d003c898b601e0021f6024c6606000bb80002ee0bdef82801000428ee1f20, level=(string)6, profile=(string)main, pixel-aspect-ratio=(fraction)1/1, interlace-mode=(string)progressive, colorimetry=(string)bt2020, chroma-site=(string)mpeg2
/GstPipeline:pipeline0/GstFakeSink:fakesink0.GstPad:sink: caps = video/x-h264, width=(int)3840, height=(int)2160, framerate=(fraction)2147483647/1, stream-format=(string)avc, alignment=(string)au, codec_data=(buffer)014d003cffe10019274d003c898b601e0021f6024c6606000bb80002ee0bdef82801000428ee1f20, level=(string)6, profile=(string)main, pixel-aspect-ratio=(fraction)1/1, interlace-mode=(string)progressive, colorimetry=(string)bt2020, chroma-site=(string)mpeg2
Redistribute latency...
Redistribute latency...
^Chandling interrupt.
Interrupt: Stopping pipeline ...
Execution ended after 0:00:02.537316000
Setting pipeline to PAUSED ...
Setting pipeline to READY ...
Setting pipeline to NULL ...
Freeing pipeline ...
```
and
```
Resolution: 3840 x 2160 (2160p 4K UHD — Ultra High Definition)
Interface Type: 1920 x 1080 @ 60 Hz
```
When I set to maximum UHD resolution I have:
```
MacBookPro:~ roman$ gst-launch-1.0 avfvideosrc capture-screen=true ! vtenc_h264_hw ! fakesink -vv
Setting pipeline to PAUSED ...
Pipeline is live and does not need PREROLL ...
Setting pipeline to PLAYING ...
New clock: GstSystemClock
/GstPipeline:pipeline0/GstAVFVideoSrc:avfvideosrc0.GstPad:src: caps = video/x-raw, width=(int)3840, height=(int)2160, framerate=(fraction)2147483647/1, format=(string)UYVY
/GstPipeline:pipeline0/vtenc_h264_hw:vtenc_h264_hw0.GstPad:sink: caps = video/x-raw, width=(int)3840, height=(int)2160, framerate=(fraction)2147483647/1, format=(string)UYVY
/GstPipeline:pipeline0/vtenc_h264_hw:vtenc_h264_hw0.GstPad:src: caps = video/x-h264, width=(int)3840, height=(int)2160, framerate=(fraction)2147483647/1, stream-format=(string)avc, alignment=(string)au, codec_data=(buffer)014d003cffe10019274d003c898b601e0021f6024c6606000bb80002ee0bdef82801000428ee1f20, level=(string)6, profile=(string)main, pixel-aspect-ratio=(fraction)1/1, interlace-mode=(string)progressive, colorimetry=(string)bt2020, chroma-site=(string)mpeg2
/GstPipeline:pipeline0/GstFakeSink:fakesink0.GstPad:sink: caps = video/x-h264, width=(int)3840, height=(int)2160, framerate=(fraction)2147483647/1, stream-format=(string)avc, alignment=(string)au, codec_data=(buffer)014d003cffe10019274d003c898b601e0021f6024c6606000bb80002ee0bdef82801000428ee1f20, level=(string)6, profile=(string)main, pixel-aspect-ratio=(fraction)1/1, interlace-mode=(string)progressive, colorimetry=(string)bt2020, chroma-site=(string)mpeg2
Redistribute latency...
Redistribute latency...
```
and
```
Resolution: 3840 x 2160 (2160p 4K UHD — Ultra High Definition)
Interface Type: 3840 x 2160 @ 60 Hz
```
I think the problem inside of scale factor for intermediates resolutions. May be this Apple problem...https://gitlab.freedesktop.org/gstreamer/gstreamer-vaapi/-/issues/177vaapih265enc generate noise in some special picture with low power mode.2019-07-07T15:06:25Zwangfeifei.w.wang@intel.comvaapih265enc generate noise in some special picture with low power mode.With cmd:
gst-launch-1.0 videotestsrc num-buffers=100 ! video/x-raw,format=NV12,width=1920,height=1080 ! vaapih265enc tune=3 low-delay-b=true rate-control=cbr ! filesink location=out.h265
Check out.h265, there are some green noise in gr...With cmd:
gst-launch-1.0 videotestsrc num-buffers=100 ! video/x-raw,format=NV12,width=1920,height=1080 ! vaapih265enc tune=3 low-delay-b=true rate-control=cbr ! filesink location=out.h265
Check out.h265, there are some green noise in gray and snow part:
![Untitled](/uploads/10214b024de2642e66126678c47fcef7/Untitled.png)https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/issues/62Share runtime between S3 plugins2019-08-05T13:34:59ZMarcin KolnyShare runtime between S3 pluginsBoth `s3sink` and `s3src` elements create their own `tokio::runtime::Runtime` to execute asynchronous networking operations. We should be able to share the runtime between those two plugins, so when they're used in one pipeline, only one...Both `s3sink` and `s3src` elements create their own `tokio::runtime::Runtime` to execute asynchronous networking operations. We should be able to share the runtime between those two plugins, so when they're used in one pipeline, only one runtime is created.https://gitlab.freedesktop.org/gstreamer/gst-editing-services/-/issues/70Tests 'ges/layer' fails on arm* and x86'2021-09-24T12:17:14ZRasmus Thomsenoss@cogitri.devTests 'ges/layer' fails on arm* and x86'It fails with the following, not so helpful output:
```
FAIL: ges/layer
===============
Running suite(s): ges-layer
90%: Checks: 20, Failures: 0, Errors: 2
ges/layer.c:145:E:timeline-layer:test_layer_priorities:0: (after this point) Rec...It fails with the following, not so helpful output:
```
FAIL: ges/layer
===============
Running suite(s): ges-layer
90%: Checks: 20, Failures: 0, Errors: 2
ges/layer.c:145:E:timeline-layer:test_layer_priorities:0: (after this point) Received signal 11 (Segmentation fault)
ges/layer.c:1542:E:timeline-layer:test_layer_get_clips_in_interval:0: (after this point) Received signal 11 (Segmentation fault)
Check suite ges ran in 0.654s (tests failed: 2)
FAIL ges/layer (exit status: 2)
```
See https://cloud.drone.io/alpinelinux/aports/7891
OS: Alpine Linux Edge (mind you, this uses musl)https://gitlab.freedesktop.org/gstreamer/cerbero/-/issues/169Tests 'ges/layer' fails on arm* and x86'2019-06-26T19:43:34ZRasmus Thomsenoss@cogitri.devTests 'ges/layer' fails on arm* and x86'It fails with the following, not so helpful output:
```
FAIL: ges/layer
===============
Running suite(s): ges-layer
90%: Checks: 20, Failures: 0, Errors: 2
ges/layer.c:145:E:timeline-layer:test_layer_priorities:0: (after this point) Rec...It fails with the following, not so helpful output:
```
FAIL: ges/layer
===============
Running suite(s): ges-layer
90%: Checks: 20, Failures: 0, Errors: 2
ges/layer.c:145:E:timeline-layer:test_layer_priorities:0: (after this point) Received signal 11 (Segmentation fault)
ges/layer.c:1542:E:timeline-layer:test_layer_get_clips_in_interval:0: (after this point) Received signal 11 (Segmentation fault)
Check suite ges ran in 0.654s (tests failed: 2)
FAIL ges/layer (exit status: 2)
```
See https://cloud.drone.io/alpinelinux/aports/7891
OS: Alpine Linux Edge (mind you, this uses musl)https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/issues/1002siren: asan global buffer overflow2020-07-10T14:08:11ZFabrice Belletfabrice@bellet.infosiren: asan global buffer overflowHi!
While testing pidgin + sipe with a skype client on a network link where I artificially corrupt *a small proportion* of the UDP packets, I could produce this attached asan global-buffer-overflow error inside the huffman decoding fu...Hi!
While testing pidgin + sipe with a skype client on a network link where I artificially corrupt *a small proportion* of the UDP packets, I could produce this attached asan global-buffer-overflow error inside the huffman decoding function of the siren codec. This log seems to suggest that the side-effect of this corruption creates a value of ``absolute_region_power_index[i] + 24`` out of the bounds of the array ?
[pidgin.global-buffer-overflow.log.2.txt](/uploads/4d46e8a34e845b55bea8bff08c6fdfd9/pidgin.global-buffer-overflow.log.2.txt)https://gitlab.freedesktop.org/gstreamer/gstreamer-vaapi/-/issues/176vaapih265dec can not decode main-10 stream with variable SPS correctly2021-07-22T08:11:39ZHe Junyanvaapih265dec can not decode main-10 stream with variable SPS correctlygst-launch-1.0 filesrc location=Allegro_HEVC_Main10_MT51_BITDEPTHRND_00_1920x1080@60Hz_1.8.bin ! h265parse ! vaapih265dec !
videoconvert ! video/x-raw,format=P010_10LE ! checksumsink2 file-checksum=false qos=false frame-checksum=false ...gst-launch-1.0 filesrc location=Allegro_HEVC_Main10_MT51_BITDEPTHRND_00_1920x1080@60Hz_1.8.bin ! h265parse ! vaapih265dec !
videoconvert ! video/x-raw,format=P010_10LE ! checksumsink2 file-checksum=false qos=false frame-checksum=false plane-checksum=false dump-output=true dump-location=h26510.yuv
Decoding process generate errorshttps://gitlab.freedesktop.org/gstreamer/gstreamer-rs/-/issues/209Make gst::Seqnum, gst::MetaSeqnum similar to Gst::ClockTime and represent inv...2020-01-24T22:34:49ZMathieu DuponchelleMake gst::Seqnum, gst::MetaSeqnum similar to Gst::ClockTime and represent invalid values in the typesThe following discussion from gstreamer/gst-plugins-rs!90 should be addressed:
- [ ] @slomo started a [discussion](https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/merge_requests/90#note_126579): (+2 comments)
> I wonder if...The following discussion from gstreamer/gst-plugins-rs!90 should be addressed:
- [ ] @slomo started a [discussion](https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/merge_requests/90#note_126579): (+2 comments)
> I wonder if we should change the type definition of that so that seqnums (and group ids, etc) are never `0 == INVALID`, and instead use `Option`s in the code. Unrelated to your changes here of course, but seems worth doing.
>
> Basically changing `gst::Seqnum` from `u32` to a `NonZeroU32`, and changing the API that can take invalid seqnum (is there any at all?) to an `Option<gst::Seqnum>`.https://gitlab.freedesktop.org/gstreamer/gstreamer-vaapi/-/issues/175[gstreame-vaapi][vp9 10bit dec]the md5 does not match if the display width/he...2019-07-01T09:16:51Zwangzhanjun[gstreame-vaapi][vp9 10bit dec]the md5 does not match if the display width/heght is smaller than frame_hdr width/heght.System Environment
=======
* Platform: KBL
* Arch: x86_64
* Kernel: 4.18.0+
* libva commit 7071bb35a45150786b15c4c20c61cfc3696c73b2
* gmmlib commit 5a931e80d9bc1da7967927fc2faf02def8bd1234
* msdk: commit 11a171e6b4efd9675a845691e0...System Environment
=======
* Platform: KBL
* Arch: x86_64
* Kernel: 4.18.0+
* libva commit 7071bb35a45150786b15c4c20c61cfc3696c73b2
* gmmlib commit 5a931e80d9bc1da7967927fc2faf02def8bd1234
* msdk: commit 11a171e6b4efd9675a845691e0756e13eee6496f
* media_driver commit 0e54d0966e62fa6b7790de31818919633074423b
* gstreamer master branch 0257c7813b3d13081ff107070124ebc79dc7baf90
* repo: https://gitlab.freedesktop.org/gstreamer/gstreamer.git
* gstreamer-vaapi: a6dfb6e5bee0fd7b7bbeb654e5ef1da2abe73446
Reproduce Steps
==============
* take one case for example
* case info
* picture width: 128
* picture height: 220
* display width: 14883
* display heght: 128
* 1. build enc as above lists
* 2. gst-launch-1.0 filesrc location=./input.ivf ! ivfparse ! vaapivp9dec ! videoconvert ! video/x-raw,format=P010_10LE ! checksumsink2 file-checksum=true frame-checksum=true dump-output=true dump-location=./test_P010.yuv
* 3. the output md5 are different every times, and does not match with standardhttps://gitlab.freedesktop.org/gstreamer/cerbero/-/issues/168librtmp and openssl fail to build on native windows2019-07-24T21:17:24ZNirbheek Chauhannirbheek.chauhan@gmail.comlibrtmp and openssl fail to build on native windowsInitially reported at https://gitlab.freedesktop.org/gstreamer/cerbero/issues/164#note_180139
```
[(45/73) librtmp -> compile ]
Running command '['sh', '-c', 'make SYS=mingw prefix=$CERBERO_PREFIX CRYPTO=GNUTLS XLDFLAGS=\\"$LDFLAGS\\" X...Initially reported at https://gitlab.freedesktop.org/gstreamer/cerbero/issues/164#note_180139
```
[(45/73) librtmp -> compile ]
Running command '['sh', '-c', 'make SYS=mingw prefix=$CERBERO_PREFIX CRYPTO=GNUTLS XLDFLAGS=\\"$LDFLAGS\\" XCFLAGS=\\"$CFLAGS\\" CC=\\"$CC\\" LD=\\"$LD\\"']'
make: invalid option -- g
make: invalid option -- O
make: invalid option -- 2
make: invalid option -- D
make: invalid option -- D
make: invalid option -- _
```
openssl:
```
Running command '['sh', '-c', 'make V=1 AR=\\"$AR\\" RANLIB=\\"$RANLIB\\" CC=\\"$CC $CFLAGS\\" LD=\\"$LD $LDFLAGS\\" LDFLAG=\\"$LDFLAGS -fPIC\\" CFLAG=\\"$CFLAGS -fPIC -DOPENSSL_PIC\\"']'
make: invalid option -- g
make: invalid option -- O
make: invalid option -- 2
make: invalid option -- D
make: invalid option -- D
make: invalid option -- _
make: invalid option -- c
make: invalid option -- :
make: invalid option -- /
make: invalid option -- c
make: invalid option -- g
make: invalid option -- O
make: invalid option -- 2
make: invalid option -- D
make: invalid option -- D
make: invalid option -- _
make: invalid option -- D
make: invalid option -- O
make: invalid option -- P
make: invalid option -- E
make: invalid option -- N
make: invalid option -- _
make: invalid option -- P
```
This is probably fallout from https://gitlab.freedesktop.org/gstreamer/cerbero/merge_requests/195.Nirbheek Chauhannirbheek.chauhan@gmail.comNirbheek Chauhannirbheek.chauhan@gmail.comhttps://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/issues/1000dashdemux: Dash content with "isoff-on-demand" profile does not adapt during ...2021-10-20T07:42:17ZSanjay Guptadashdemux: Dash content with "isoff-on-demand" profile does not adapt during playbackHi,
During the playback of dash vod content with "isoff-on-demand" profile, it is not adapting to different bitrate representation in video adaptation set. It is adapting to different bitrates for live profile contents.
Checked with con...Hi,
During the playback of dash vod content with "isoff-on-demand" profile, it is not adapting to different bitrate representation in video adaptation set. It is adapting to different bitrates for live profile contents.
Checked with content:
https://dash.akamaized.net/dash264/TestCases/1a/sony/SNE_DASH_SD_CASE1A_REVISED.mpd
Gstreamer version used during test: 1.14.2