GStreamer issueshttps://gitlab.freedesktop.org/groups/gstreamer/-/issues2023-02-08T20:34:08Zhttps://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/issues/308awss3hlssink : audio support info2023-02-08T20:34:08ZAadeITawss3hlssink : audio support infoHi, Does the `awss3hlssink` support all audio types?
gst-launch ```awss3hlssink```
```
Pad Templates:
SINK template: 'audio'
Availability: On request
Capabilities:
ANY
SINK template: 'video'
Availability: On requ...Hi, Does the `awss3hlssink` support all audio types?
gst-launch ```awss3hlssink```
```
Pad Templates:
SINK template: 'audio'
Availability: On request
Capabilities:
ANY
SINK template: 'video'
Availability: On request
Capabilities:
ANY
```
wile fail when i link audio/x-alaw mediatype to the audio pad of awss3hlssink ,Look forward to your replyhttps://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/issues/1755elements_dash_mpd test case is failed, can any one please check2023-02-07T00:41:57Zvenkatesh uelements_dash_mpd test case is failed, can any one please checkNumber of TestSuite Executed :15
Test Suite Name :elements_dash_mpd
Finding the execution status from log with the keyword : Checks:
Running suite(s): dash
noname.xml:1: parser error : Start tag expected, '<' not found
<?xml version="1.0...Number of TestSuite Executed :15
Test Suite Name :elements_dash_mpd
Finding the execution status from log with the keyword : Checks:
Running suite(s): dash
noname.xml:1: parser error : Start tag expected, '<' not found
<?xml version="1.0"?>
^
noname.xml:1: parser error : Opening and ending tag mismatch: MPD line 1 and NPD
sh:schema:mpd:2011" profiles="urn:mpeg:dash:profile:isoff-main:2011"> </NPD>
^
99%: Checks: 119, Failures: 1, Errors: 0
../gst-plugins-bad-1.18.5/tests/check/elements/dash_mpd.c:5874:F:simpleMPD:dash_mpdparser_xlink_period:0: 'g_list_length (period_list)' (1) is not equal to '4' (4)
Check suite dash ran in 3.403s (tests failed: 1)https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/issues/307I'm new to webrtcsink. Is it possible to have some general questions here?2023-02-06T19:51:34ZDaniel EI'm new to webrtcsink. Is it possible to have some general questions here?Sorry is to not open a lot of Issues here.Sorry is to not open a lot of Issues here.https://gitlab.freedesktop.org/gstreamer/gstreamer/-/issues/1774av1parse: handle unknown metadata OBU type not correctly2023-02-10T00:36:51ZJianfeng Liuav1parse: handle unknown metadata OBU type not correctlyIn our code we just drop the whole data: https://gitlab.freedesktop.org/gstreamer/gstreamer/-/blob/main/subprojects/gst-plugins-bad/gst-libs/gst/codecparsers/gstav1parser.c#L1783.
But libgav1 which google uses handles it as av1 spec des...In our code we just drop the whole data: https://gitlab.freedesktop.org/gstreamer/gstreamer/-/blob/main/subprojects/gst-plugins-bad/gst-libs/gst/codecparsers/gstav1parser.c#L1783.
But libgav1 which google uses handles it as av1 spec describes: https://chromium.googlesource.com/codecs/libgav1/+/refs/heads/main/src/obu_parser.cc#2472.
Sample video bilibili-av1-test.mp4 can be found at https://github.com/rockchip-linux/mpp/issues/351.https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/issues/306s3:awss3hlssink Bucket does not support setting subdirectory2023-02-06T16:38:16ZAadeITs3:awss3hlssink Bucket does not support setting subdirectoryHi, When I use awss3sink, the plug-in supports setting subdirectory ,such as bucket ```awss3sink bucket=test/123 key=test.mp4```, but awss3hlsink does not support subdirectory . Will awss3hlssink support setting subdirectory in the futur...Hi, When I use awss3sink, the plug-in supports setting subdirectory ,such as bucket ```awss3sink bucket=test/123 key=test.mp4```, but awss3hlsink does not support subdirectory . Will awss3hlssink support setting subdirectory in the future?
When i setting awss3hlssink
```
awss3hlssink net/aws/src/s3hlssink/imp.rs:278:gstaws::s3hlssink::imp::S3HlsSink::s3_request:<awss3hlssink0> Put object request for S3 key playlist.m3u8 of data length 118 failed with error FutureError(ServiceError(ServiceError { source: PutObjectError { kind: Unhandled(Unhandled { source: Error { code: Some("SignatureDoesNotMatch"), message: Some("The request signature we calculated does not match the signature you provided. Check your key and signing method."), request_id: Some("17412CD79CE95F12"), extras: {} } }), meta: Error { code: Some("SignatureDoesNotMatch"), message: Some("The request signature we calculated does not match the signature you provided. Check your key and signing method."), request_id: Some("17412CD79CE95F12"), extras: {} } }, raw: Response { inner: Response { status: 403, version: HTTP/1.1, headers: {"accept-ranges": "bytes", "content-length": "415", "content-security-policy": "block-all-mixed-content", "content-type": "application/xml", "server": "MinIO", "strict-transport-security": "max-age=31536000; includeSubDomains", "vary": "Origin", "vary": "Accept-Encoding", "x-amz-request-id": "17412CD79CE95F12", "x-content-type-options": "nosniff", "x-xss-protection": "1; mode=block", "date": "Mon, 06 Feb 2023 07:50:40 GMT"}, body: SdkBody { inner: Once(Some(b"<?xml version=\"1.0\" encoding=\"UTF-8\"?>\n<Error><Code>SignatureDoesNotMatch</Code><Message>The request signature we calculated does not match the signature you provided. Check your key and signing method.</Message><Key>playlist.m3u8</Key><BucketName>123%2F%2F22</BucketName><Resource>/123//22/playlist.m3u8</Resource><RequestId>17412CD79CE95F12</RequestId><HostId>740bdc4a-b5d1-4e61-bb37-137420b13dc6</HostId></Error>")), retryable: true } }, properties: SharedPropertyBag(Mutex { data: PropertyBag, poisoned: false, .. }) } }))
ERROR: from element /GstPipeline:pipeline0/GstAwsS3HlsSink:awss3hlssink0: Could not write to resource.
```https://gitlab.freedesktop.org/gstreamer/gst-plugins-base/-/issues/985Video scaling output size not expected2023-02-08T12:14:11ZBharati KhanijoVideo scaling output size not expectedOn running the videoscale command,
on multiple resolutions , eg. from 1920x1080 to 1280x720 , 960x540, 640x360 etc,
The final file sizes are expected to be sizeof(1280x720) > sizeof(960x540) > sizeof(640x360)
But when the below command ...On running the videoscale command,
on multiple resolutions , eg. from 1920x1080 to 1280x720 , 960x540, 640x360 etc,
The final file sizes are expected to be sizeof(1280x720) > sizeof(960x540) > sizeof(640x360)
But when the below command is run on nvidia-jetson, the file sizes are similar.
gst-launch-1.0 -q filesrc location={video_path} ! qtdemux ! queue ! h264parse ! nvv4l2decoder enable-max-performance=1 ! nvvidconv ! videoscale method=3 ! 'video/x-raw(memory:NVMM), width=(int){downscaled_wd}, height=(int){downscaled_ht}, format=(string)I420' ! nvv4l2h264enc ! 'video/x-h264, stream-format=(string)byte-stream' ! h264parse ! qtmux ! filesink location={new_path} -e >/dev/null 2>&1https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/issues/305rtpgccbwe: Panics with 'assertion failed: min <= max'2023-02-09T19:44:01ZDaniel Ertpgccbwe: Panics with 'assertion failed: min <= max'```
thread '<unnamed>' panicked at 'assertion failed: min <= max', /rustc/fc594f15669680fa70d255faec3ca3fb507c3405/library/core/src/num/f64.rs:1392:9
0:02:16.583324309 1327 0x7fc0a00af760 ERROR webrtcsink net/webrtc/src/webr...```
thread '<unnamed>' panicked at 'assertion failed: min <= max', /rustc/fc594f15669680fa70d255faec3ca3fb507c3405/library/core/src/num/f64.rs:1392:9
0:02:16.583324309 1327 0x7fc0a00af760 ERROR webrtcsink net/webrtc/src/webrtcsink/imp.rs:1683:gstrswebrtc::webrtcsink::imp::WebRTCSink::start_session::{{closure}}: session fcd97386-ff10-4ade-b6fe-564f8bd10b40 error: Panicked: assertion failed: min <= max, details: None
0:02:16.583464833 1327 0x7fc0a00af760 ERROR webrtcsink net/webrtc/src/webrtcsink/imp.rs:1683:gstrswebrtc::webrtcsink::imp::WebRTCSink::start_session::{{closure}}: session fcd97386-ff10-4ade-b6fe-564f8bd10b40 error: Panicked, details: None
0:02:16.584007540 1327 0x7fc0a00af760 ERROR webrtcsink net/webrtc/src/webrtcsink/imp.rs:1683:gstrswebrtc::webrtcsink::imp::WebRTCSink::start_session::{{closure}}: session fcd97386-ff10-4ade-b6fe-564f8bd10b40 error: Panicked, details: None
0:02:16.585135565 1327 0x7fc0a00af760 ERROR webrtcsink net/webrtc/src/webrtcsink/imp.rs:1683:gstrswebrtc::webrtcsink::imp::WebRTCSink::start_session::{{closure}}: session fcd97386-ff10-4ade-b6fe-564f8bd10b40 error: Panicked, details: None
thread '<unnamed>' panicked at 'called `Result::unwrap()` on an `Err` value: PoisonError { .. }', net/rtp/src/gcc/imp.rs:1062:56
thread '<unnamed>' panicked at 'assertion failed: min <= max', /rustc/fc594f15669680fa70d255faec3ca3fb507c3405/library/core/src/num/f64.rs:1392:9
0:02:17.380933862 1327 0x7fc0a00af760 ERROR webrtcsink net/webrtc/src/webrtcsink/imp.rs:1683:gstrswebrtc::webrtcsink::imp::WebRTCSink::start_session::{{closure}}: session 52ae7c50-0abb-460d-90fc-fc3d2f19a3ab error: Panicked: assertion failed: min <= max, details: None
0:02:17.381155195 1327 0x7fc0a00af760 ERROR webrtcsink net/webrtc/src/webrtcsink/imp.rs:1683:gstrswebrtc::webrtcsink::imp::WebRTCSink::start_session::{{closure}}: session 52ae7c50-0abb-460d-90fc-fc3d2f19a3ab error: Panicked, details: None
0:02:17.381642795 1327 0x7fc0a00af760 ERROR webrtcsink net/webrtc/src/webrtcsink/imp.rs:1683:gstrswebrtc::webrtcsink::imp::WebRTCSink::start_session::{{closure}}: session 52ae7c50-0abb-460d-90fc-fc3d2f19a3ab error: Panicked, details: None
0:02:17.382954925 1327 0x7fc0a00af760 ERROR webrtcsink net/webrtc/src/webrtcsink/imp.rs:1683:gstrswebrtc::webrtcsink::imp::WebRTCSink::start_session::{{closure}}: session 52ae7c50-0abb-460d-90fc-fc3d2f19a3ab error: Panicked, details: None
thread '<unnamed>' panicked at 'called `Result::unwrap()` on an `Err` value: PoisonError { .. }', net/rtp/src/gcc/imp.rs:1062:56
0:02:27.807139547 1327 0x560b6ff54090 ERROR webrtcnice nice.c:294:on_resolve_host: failed to resolve: Error resolving “82973f9a-a2ae-4e51-9c4e-51b4d5b1c4d0.local”: Temporary failure in name resolution
```
Is there any explanation, especially for the "PoisonError"?https://gitlab.freedesktop.org/gstreamer/gstreamer/-/issues/1770avtp_rvf.h header not found when building on Ubuntu 22.042023-02-06T14:14:01ZEnrique Ocaña Gonzálezavtp_rvf.h header not found when building on Ubuntu 22.04At some point between 16158df5b2 and 9fa5fbc25e I started to get build failures like this:
```
[1650/3989] Compiling C object subprojects/gst-plugins-bad/ext/avtp/libgstavtp.so.p/gstavtprvfpay.c.o
FAILED: subprojects/gst-plugins-bad/ext...At some point between 16158df5b2 and 9fa5fbc25e I started to get build failures like this:
```
[1650/3989] Compiling C object subprojects/gst-plugins-bad/ext/avtp/libgstavtp.so.p/gstavtprvfpay.c.o
FAILED: subprojects/gst-plugins-bad/ext/avtp/libgstavtp.so.p/gstavtprvfpay.c.o
cc -Isubprojects/gst-plugins-bad/ext/avtp/libgstavtp.so.p -Isubprojects/gst-plugins-bad/ext/avtp -I../subprojects/gst-plugins-bad/ext/avtp -Isubprojects/gst-plugins-bad -I../subprojects/gst-plugins-bad -Isubprojects/gst-plugins-base/gst-libs -I../subprojects/gst-plugins-base/gst-libs -Isubprojects/gstreamer/libs -I../subprojects/gstreamer/libs -Isubprojects/gstreamer -I../subprojects/gstreamer -Isubprojects/orc -I../subprojects/orc -I../subprojects/avtp/include -Isubprojects/gst-plugins-base/gst-libs/gst/audio -Isubprojects/gst-plugins-base/gst-libs/gst/tag -Isubprojects/gstreamer/libs/gst/base -Isubprojects/gstreamer/gst -Isubprojects/gst-plugins-base/gst-libs/gst/video -I/usr/include/glib-2.0 -I/usr/lib/x86_64-linux-gnu/glib-2.0/include -fdiagnostics-color=always -D_FILE_OFFSET_BITS=64 -Wall -Winvalid-pch -O2 -g -fvisibility=hidden -fno-strict-aliasing -DG_DISABLE_DEPRECATED -Wmissing-prototypes -Wold-style-definition -Wmissing-declarations -Wredundant-decls -Wwrite-strings -Wformat -Wformat-security -Winit-self -Wmissing-include-dirs -Waddress -Wno-multichar -Wvla -Wpointer-arith -fPIC -pthread -DHAVE_CONFIG_H -MD -MQ subprojects/gst-plugins-bad/ext/avtp/libgstavtp.so.p/gstavtprvfpay.c.o -MF subprojects/gst-plugins-bad/ext/avtp/libgstavtp.so.p/gstavtprvfpay.c.o.d -o subprojects/gst-plugins-bad/ext/avtp/libgstavtp.so.p/gstavtprvfpay.c.o -c ../subprojects/gst-plugins-bad/ext/avtp/gstavtprvfpay.c
../subprojects/gst-plugins-bad/ext/avtp/gstavtprvfpay.c:42:10: fatal error: avtp_rvf.h: No such file or directory
42 | #include <avtp_rvf.h>
| ^~~~~~~~~~~~
compilation terminated.
```
@tpm suggested me to create this issue and cc @adrianf0 or mention it in https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1335https://gitlab.freedesktop.org/gstreamer/meson-ports/ffmpeg/-/issues/34libavutil: compile-/link-errors in tests during meson-build by missing depend...2023-04-28T12:17:52ZEmanuel Schmidtlibavutil: compile-/link-errors in tests during meson-build by missing dependency to liblzoI came across this problem, as I used ffmpeg as a meson-subproject.
The compilation on `meson-4.4` fails with multiple linker-errors of the kind:
`/ffmpeg/build2/../libavutil/tests/lzo.c:56: undefined reference to 'lzo1x_999_compress'`
...I came across this problem, as I used ffmpeg as a meson-subproject.
The compilation on `meson-4.4` fails with multiple linker-errors of the kind:
`/ffmpeg/build2/../libavutil/tests/lzo.c:56: undefined reference to 'lzo1x_999_compress'`
To reproduce, execute the following steps:
* Start on a fresh environment, e.g. with: `docker run --rm -it --entrypoint /bin/bash ubuntu:22.04`
* Install the necessary tools and the liblzo-headers: `apt update && apt install -y git meson gcc liblzo2-dev`
* Clone and checkout the according repository and branch: `git clone --branch meson-4.4 --depth 1 https://gitlab.freedesktop.org/gstreamer/meson-ports/ffmpeg.git`
* Build the project: `cd ffmpeg && meson setup build && meson compile -C build`
The error-messages contain several other references, all related to liblzo/liblzo2.
I assume it could be fixed, by adding a dependency for this library to the build of the test within `libavutil/meson.build` or disable the test, if the specific dependency is not availableamysparkamysparkhttps://gitlab.freedesktop.org/gstreamer/gstreamer/-/issues/1768nvvp9dec: internal data flow problems causing issues with hlssink2 (warnings ...2023-02-28T19:55:56ZChristian Koopnvvp9dec: internal data flow problems causing issues with hlssink2 (warnings and criticals)* **GStreamer**: `1.20.5`
* **GPU**: NVIDIA GeForce 1070
* **OS**: Manjaro Linux (arch based)
---
I have a pipeline that basically turns any input video file into a desired HLS output (via `uridecodebin` and `hlssink2`).
I encountered...* **GStreamer**: `1.20.5`
* **GPU**: NVIDIA GeForce 1070
* **OS**: Manjaro Linux (arch based)
---
I have a pipeline that basically turns any input video file into a desired HLS output (via `uridecodebin` and `hlssink2`).
I encountered some issues with a specific video file that produces
a broken HLS manifest (and maybe segments too) in regard to the *targetDuration*.
The troubling bit is that when End-Of-Stream is reached, hlssink2 seems to rewrite the playlist file and fixed the broken durations (matching the generated segments?).
That is why we might need to kill/interrupt the process before it reaches the EOS – The duration is still not the expected `4` for the segments after the rewrite.
---
Video file (removed all streams except the video stream and only kept the first 20 seconds): [test.mkv](/uploads/80bf91443ceb0eeb492ee3163b70bb21/test.mkv)
My simple command: `gst-launch-1.0 filesrc location=./test.mkv ! matroskademux ! vp9parse ! nvvp9dec ! videoconvert ! autovideosink`
My command using `hlssink2` (kill/interrupt the process at ca. 70% progress): `gst-launch-1.0 filesrc location=./test.mkv ! matroskademux ! vp9parse ! nvvp9dec ! videoconvert ! videorate ! video/x-raw,framerate=24/1 ! x264enc key-int-max=24 ! hlssink2 target-duration=4`
**Using `vp9dec` instead of `nvvp9dec` gets rid of the warnings and issues (the output playlist.m3u8 properly generates with durations of 4 too).
---
The command using `hlssink2` prints the following:
```
Setting pipeline to PAUSED ...
Pipeline is PREROLLING ...
Got context from element 'nvvp9dec0': gst.cuda.context=context, gst.cuda.context=(GstCudaContext)"\(GstCudaContext\)\ cudacontext0", cuda-device-id=(int)0;
Got context from element 'nvvp9dec0': gst.gl.GLDisplay=context, gst.gl.GLDisplay=(GstGLDisplay)"\(GstGLDisplayX11\)\ gldisplayx11-0";
Redistribute latency...
Redistribute latency...
(gst-launch-1.0:353098): GStreamer-WARNING **: 13:37:52.316: ../gstreamer/subprojects/gstreamer/gst/gstpad.c:4677:gst_pad_push_data:<nvvp9dec0:src> Got data flow before segment event
(gst-launch-1.0:353098): GStreamer-WARNING **: 13:37:52.316: ../gstreamer/subprojects/gstreamer/gst/gstpad.c:4416:gst_pad_chain_data_unchecked:<videoconvert0:sink> Got data flow before segment event
(gst-launch-1.0:353098): GStreamer-WARNING **: 13:37:52.316: ../gstreamer/subprojects/gstreamer/gst/gstpad.c:4677:gst_pad_push_data:<videoconvert0:src> Got data flow before segment event
(gst-launch-1.0:353098): GStreamer-WARNING **: 13:37:52.316: ../gstreamer/subprojects/gstreamer/gst/gstpad.c:4416:gst_pad_chain_data_unchecked:<videorate0:sink> Got data flow before segment event
(gst-launch-1.0:353098): GStreamer-WARNING **: 13:37:52.319: ../gstreamer/subprojects/gstreamer/gst/gstpad.c:4677:gst_pad_push_data:<videorate0:src> Got data flow before segment event
(gst-launch-1.0:353098): GStreamer-WARNING **: 13:37:52.319: ../gstreamer/subprojects/gstreamer/gst/gstpad.c:4416:gst_pad_chain_data_unchecked:<capsfilter0:sink> Got data flow before segment event
(gst-launch-1.0:353098): GStreamer-WARNING **: 13:37:52.319: ../gstreamer/subprojects/gstreamer/gst/gstpad.c:4677:gst_pad_push_data:<capsfilter0:src> Got data flow before segment event
(gst-launch-1.0:353098): GStreamer-WARNING **: 13:37:52.319: ../gstreamer/subprojects/gstreamer/gst/gstpad.c:4416:gst_pad_chain_data_unchecked:<x264enc0:sink> Got data flow before segment event
(gst-launch-1.0:353098): GStreamer-WARNING **: 13:37:52.639: ../gstreamer/subprojects/gstreamer/gst/gstpad.c:4677:gst_pad_push_data:<x264enc0:src> Got data flow before segment event
(gst-launch-1.0:353098): GStreamer-WARNING **: 13:37:52.640: ../gstreamer/subprojects/gstreamer/gst/gstpad.c:4416:gst_pad_chain_data_unchecked:<hlssink2-0:video> Got data flow before segment event
(gst-launch-1.0:353098): GStreamer-WARNING **: 13:37:52.640: ../gstreamer/subprojects/gstreamer/gst/gstpad.c:4677:gst_pad_push_data:<video:proxypad1> Got data flow before segment event
(gst-launch-1.0:353098): GStreamer-WARNING **: 13:37:52.640: ../gstreamer/subprojects/gstreamer/gst/gstpad.c:4416:gst_pad_chain_data_unchecked:<splitmuxsink0:video> Got data flow before segment event
(gst-launch-1.0:353098): GStreamer-WARNING **: 13:37:52.640: ../gstreamer/subprojects/gstreamer/gst/gstpad.c:4677:gst_pad_push_data:<video:proxypad0> Got data flow before segment event
(gst-launch-1.0:353098): GStreamer-WARNING **: 13:37:52.640: ../gstreamer/subprojects/gstreamer/gst/gstpad.c:4416:gst_pad_chain_data_unchecked:<queue_video:sink> Got data flow before segment event
(gst-launch-1.0:353098): GStreamer-CRITICAL **: 13:37:52.640: gst_segment_to_running_time_full: assertion 'segment->format == format' failed
(gst-launch-1.0:353098): GStreamer-CRITICAL **: 13:37:52.640: gst_segment_to_running_time_full: assertion 'segment->format == format' failed
(gst-launch-1.0:353098): GStreamer-CRITICAL **: 13:37:52.640: gst_segment_to_running_time_full: assertion 'segment->format == format' failed
(gst-launch-1.0:353098): GStreamer-CRITICAL **: 13:37:52.640: gst_segment_to_running_time_full: assertion 'segment->format == format' failed
(gst-launch-1.0:353098): GStreamer-WARNING **: 13:37:52.640: ../gstreamer/subprojects/gstreamer/gst/gstpad.c:4677:gst_pad_push_data:<queue_video:src> Got data flow before segment event
Redistribute latency...
Pipeline is PREROLLED ...
Setting pipeline to PLAYING ...
New clock: GstSystemClock
(gst-launch-1.0:353098): GStreamer-WARNING **: 13:37:52.897: ../gstreamer/subprojects/gstreamer/gst/gstpad.c:4416:gst_pad_chain_data_unchecked:<mpegtsmux0:sink_65> Got data flow before segment event
(gst-launch-1.0:353098): GStreamer-CRITICAL **: 13:37:52.901: gst_segment_to_running_time: assertion 'segment->format == format' failed
```
---
The generated `playlist.m3u8`:
`#EXT-X-TARGETDURATION` should be 4 here and `#EXTINF` should be 4 too as I force the framerate and keyframes to align.
```m3u
#EXTM3U
#EXT-X-VERSION:3
#EXT-X-MEDIA-SEQUENCE:0
#EXT-X-TARGETDURATION:633437445
#EXTINF:9223371776,
segment00000.ts
#EXTINF:3.5416665077209473,
segment00001.ts
#EXTINF:4,
segment00002.ts
#EXT-X-ENDLIST
```https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/issues/1754[Ubuntu:18.04 Docker]: webrtc-unidirectional-h264.c:207:8: error: unknown typ...2023-05-30T16:03:20ZVijay Kamble[Ubuntu:18.04 Docker]: webrtc-unidirectional-h264.c:207:8: error: unknown type name 'GstWebRTCPriorityType'HellHello,
I am using `main` branch of the [GStreamer monorepo](https://gitlab.freedesktop.org/gstreamer/gstreamer/) and the webrtc/sendonly example in `subprojects/gst-examples` within Ubuntu 18.04 Docker.
Before compiling the webrtc/sen...Hello,
I am using `main` branch of the [GStreamer monorepo](https://gitlab.freedesktop.org/gstreamer/gstreamer/) and the webrtc/sendonly example in `subprojects/gst-examples` within Ubuntu 18.04 Docker.
Before compiling the webrtc/sendonly example, I installed below GStreamer Plugins -
`sudo apt-get install -y gstreamer1.0-tools gstreamer1.0-nice gstreamer1.0-plugins-bad gstreamer1.0-plugins-ugly gstreamer1.0-plugins-good libgstreamer1.0-dev git libglib2.0-dev libgstreamer-plugins-bad1.0-dev libsoup2.4-dev libjson-glib-dev`
But I am getting below error while compiling with Ubuntu 18.04 Docker -
"gcc" -O0 -ggdb -Wall -fno-omit-frame-pointer webrtc-unidirectional-h264.c -pthread -I/usr/include/gstreamer-1.0 -I/usr/include/libsoup-2.4 -I/usr/include/libxml2 -I/usr/include/json-glib-1.0 -I/usr/include/glib-2.0 -I/usr/lib/aarch64-linux-gnu/glib-2.0/include -lgstwebrtc-1.0 -lgstbase-1.0 -lgstreamer-1.0 -lgstsdp-1.0 -lsoup-2.4 -ljson-glib-1.0 -lgio-2.0 -lgobject-2.0 -lglib-2.0 -o webrtc-unidirectional-h264
**webrtc-unidirectional-h264.c:207:8: error: unknown type name 'GstWebRTCPriorityType'**
static GstWebRTCPriorityType
^~~~~~~~~~~~~~~~~~~~~
webrtc-unidirectional-h264.c: In function '_priority_from_string':
**webrtc-unidirectional-h264.c:211:40: error: 'GST_TYPE_WEBRTC_PRIORITY_TYPE' undeclared (first use in this function); did you mean 'GST_TYPE_WEBRTC_STATS_TYPE'?**
(GEnumClass *) g_type_class_ref (GST_TYPE_WEBRTC_PRIORITY_TYPE);
^~~~~~~~~~~~~~~~~~~~~~~~~~~~~
GST_TYPE_WEBRTC_STATS_TYPE
webrtc-unidirectional-h264.c:211:40: note: each undeclared identifier is reported only once for each function it appears in
webrtc-unidirectional-h264.c: In function 'create_receiver_entry':
**webrtc-unidirectional-h264.c:272:5: error: unknown type name 'GstWebRTCPriorityType'; did you mean 'GstWebRTCStatsType'?
GstWebRTCPriorityType priority;**
^~~~~~~~~~~~~~~~~~~~~
GstWebRTCStatsType
webrtc-unidirectional-h264.c:279:7: warning: implicit declaration of function 'gst_webrtc_rtp_sender_set_priority'; did you mean 'gst_webrtc_rtp_sender_set_transport'? [-Wimplicit-function-declaration]
gst_webrtc_rtp_sender_set_priority (sender, priority);
^~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~
gst_webrtc_rtp_sender_set_transport
webrtc-unidirectional-h264.c:287:5: error: unknown type name 'GstWebRTCPriorityType'; did you mean 'GstWebRTCStatsType'?
GstWebRTCPriorityType priority;
^~~~~~~~~~~~~~~~~~~~~
GstWebRTCStatsType
Makefile:8: recipe for target 'webrtc-unidirectional-h264' failed
make: *** [webrtc-unidirectional-h264] Error 1
I am trying the use case on Qualcomm SoC with arm64 architecture. The SW running on Qualcomm SoC is latest Ubuntu Version, which which we get the libsoup error mentioned in #1750 . Due to which, I have created Ubuntu 18.04 based Docker on device and building the example.
Could you please guide me, how I can fix the error..? Is it due to Docker environment.
Thanks and Regards!https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/issues/304awss3hlssink can't exit in gst_launch2023-02-06T07:54:55ZAadeITawss3hlssink can't exit in gst_launchI use "awss3hlssink" in "gst_launch". When I exit, the program will block in the step of setting it to null. Replacing "awss3hlssink" with "awss3sink" will not cause this problem.
error info
```
gst-launch-1.0 -e videotestsrc pattern=ba...I use "awss3hlssink" in "gst_launch". When I exit, the program will block in the step of setting it to null. Replacing "awss3hlssink" with "awss3sink" will not cause this problem.
error info
```
gst-launch-1.0 -e videotestsrc pattern=ball ! queue ! x264enc ! h264parse ! awss3hlssink endpoint-uri=http://192.168.100.23:9000 bucket=001
0:00:00.020849435 45760 0x5620fe2b72d0 ERROR GST_PLUGIN_LOADING gstpluginloader.c:442:gst_plugin_loader_try_helper: Spawning gst-plugin-scanner helper failed: Failed to close file descriptor for child process (Operation not permitted)
(gst-launch-1.0:45760): GStreamer-WARNING **: 09:21:46.280: External plugin loader failed. This most likely means that the plugin loader helper binary was not found or could not be run. You might need to set the GST_PLUGIN_SCANNER environment variable if your setup is unusual. This should normally not be required though.
Setting pipeline to PAUSED ...
Pipeline is PREROLLING ...
Redistribute latency...
Redistribute latency...
Pipeline is PREROLLED ...
Setting pipeline to PLAYING ...
New clock: GstSystemClock
Redistribute latency...
Redistribute latency...
Redistribute latency...
Redistribute latency...
Redistribute latency...
Redistribute latency...
Redistribute latency...
^Chandling interrupt.
Interrupt: Stopping pipeline ...
EOS on shutdown enabled -- Forcing EOS on the pipeline
Waiting for EOS...
Redistribute latency...
Redistribute latency...
Got EOS from element "pipeline0".
EOS received - stopping pipeline...
Execution ended after 0:00:08.960874711
Setting pipeline to NULL ...
^C
```
will block in ``Setting pipeline to NULL ...``
Normal info
```
gst-launch-1.0 -e videotestsrc pattern=ball ! queue ! x264enc ! h264parse ! queue ! isofmp4mux fragment-duration=2000000000 write-mfra=true ! awss3sink endpoint-uri=http://192.168.100.23:9000 bucket=123/22 key=tes8.mp4
0:00:00.019504724 48186 0x55f7d58670d0 ERROR GST_PLUGIN_LOADING gstpluginloader.c:442:gst_plugin_loader_try_helper: Spawning gst-plugin-scanner helper failed: Failed to close file descriptor for child process (Operation not permitted)
(gst-launch-1.0:48186): GStreamer-WARNING **: 09:24:17.257: External plugin loader failed. This most likely means that the plugin loader helper binary was not found or could not be run. You might need to set the GST_PLUGIN_SCANNER environment variable if your setup is unusual. This should normally not be required though.
Setting pipeline to PAUSED ...
Pipeline is PREROLLING ...
Redistribute latency...
Pipeline is PREROLLED ...
Setting pipeline to PLAYING ...
Redistribute latency...
New clock: GstSystemClock
^Chandling interrupt.
Interrupt: Stopping pipeline ...
EOS on shutdown enabled -- Forcing EOS on the pipeline
Waiting for EOS...
Got EOS from element "pipeline0".
EOS received - stopping pipeline...
Execution ended after 0:00:06.060543222
Setting pipeline to NULL ...
Freeing pipeline ...
```
Look forward to your replyhttps://gitlab.freedesktop.org/gstreamer/gstreamer/-/issues/1763plugin docs: Link texts in online documentation are missing2023-04-14T13:44:01ZRicardoplugin docs: Link texts in online documentation are missingWhen reading the online documentation ([example](https://gstreamer.freedesktop.org/documentation/videorate/index.html?gi-language=c)), the links for the properties have missing texts.
Example:
How the text is rendered to the user:
> T...When reading the online documentation ([example](https://gstreamer.freedesktop.org/documentation/videorate/index.html?gi-language=c)), the links for the properties have missing texts.
Example:
How the text is rendered to the user:
> The properties , , and can be read
How the source HTML looks like:
`The properties <a href="GstVideoRate:in"></a>, <a href="GstVideoRate:out"></a>, <a href="GstVideoRate:duplicate"></a>
and <a href="GstVideoRate:drop"></a> can be read` [...]
The link text should have the name of the element it is pointing to. In this case, it should be, for example:
`The properties <a href="GstVideoRate:in">in</a>, <a href="GstVideoRate:out">out</a>, <a href="GstVideoRate:duplicate">duplicate</a>
and <a href="GstVideoRate:drop">drop</a> can be read` [...]
Browser: Firefox 109.0 (64-bit).Thibault Sauniertsaunier@igalia.comThibault Sauniertsaunier@igalia.comhttps://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/issues/303Build failure due to missing but unneeded gstreamer dependencies for the sele...2023-10-05T16:05:49ZTimo GurrBuild failure due to missing but unneeded gstreamer dependencies for the selected optionsWhen trying to build only with basic/minimal functionality and the spotify plugin enabled:
```
-Dfile=enabled
-Dsodium=enabled
-Dthreadshare=enabled
-Dspotify=enabled
```
via:
`-Daudiofx=disabled -Daws=disabled -Dcdg=disabled -Dclaxon...When trying to build only with basic/minimal functionality and the spotify plugin enabled:
```
-Dfile=enabled
-Dsodium=enabled
-Dthreadshare=enabled
-Dspotify=enabled
```
via:
`-Daudiofx=disabled -Daws=disabled -Dcdg=disabled -Dclaxon=disabled -Dcsound=disabled -Ddoc=disabled -Dexamples=disabled -Dfallbackswitch=disabled -Dffv1=disabled -Dfile=enabled -Dflavors=disabled -Dfmp4=disabled -Dgif=disabled -Dhlssink3=disabled -Dhsv=disabled -Djson=disabled -Dlewton=disabled -Dlivesync=disabled -Dndi=disabled -Dmp4=disabled -Dpng=disabled -Draptorq=disabled -Drav1e=disabled -Dregex=disabled -Dreqwest=disabled -Drtp=disabled -Dsodium=enabled -Dsodium-source=system -Dtextahead=disabled -Dtextwrap=disabled -Dthreadshare=enabled -Dtogglerecord=disabled -Dtracers=disabled -Duriplaylistbin=disabled -Dvideofx=disabled -Dwebrtc=disabled -Dwebrtchttp=disabled -Dclosedcaption=disabled -Ddav1d=disabled -Dgtk4=disabled -Donvif=disabled -Dspotify=enabled -Dwebp=disabled`
build fails with:
```
Run-time dependency gstreamer-gl-1.0 found: NO (tried pkgconfig and cmake)
Looking for a fallback subproject for the dependency gstreamer-gl-1.0
../gst-plugins-rs-0.9.8/meson.build:77:2: ERROR: Neither a subproject directory nor a gst-plugins-base.wrap file was found
```
due to https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/blob/main/meson.build#L65
which is true, I don't have gst-plugins-base compiled with gl support as it's running on a headless machine (as in spotify plugin only required for mopidy-spotify). The dependency shouldn't be required and forced at all. This was working fine in gst-plugins-rs-0.9.4 which didn't offer the whole lot of options to disable unneeded stuff in a big scale like newer versions do, but the build worked fine without having e.g. gstreamer-gl-1.0 installed.
The package builds fine for me with the options stated above if I remove the dependency from meson.build:
```
# gstreamer-gl-1.0 is only required for gtk4
if option !gstreamer_plugins:gtk4 ; then
edo sed \
-e '/gstreamer-gl-1.0/d' \
-i meson.build
fi
```
Same goes for the gstreamer-webrtc-1.0 line / dependency pulling in gst-plugins-bad which is not needed if webrtc/webrtchttp are disabled.
Pulling in these dependencies should be made conditional in regards to the selected plugins actually needing those.https://gitlab.freedesktop.org/gstreamer/www/-/issues/43Link texts in online documentation are missing2023-02-01T11:41:36ZRicardoLink texts in online documentation are missingWhen reading the online documentation ([example](https://gstreamer.freedesktop.org/documentation/videorate/index.html?gi-language=c)), the links for the properties have missing texts.
Example:
How the text is rendered to the user:
> T...When reading the online documentation ([example](https://gstreamer.freedesktop.org/documentation/videorate/index.html?gi-language=c)), the links for the properties have missing texts.
Example:
How the text is rendered to the user:
> The properties , , and can be read
How the source HTML looks like:
`The properties <a href="GstVideoRate:in"></a>, <a href="GstVideoRate:out"></a>, <a href="GstVideoRate:duplicate"></a>
and <a href="GstVideoRate:drop"></a> can be read` [...]
The link text should have the name of the element it is pointing to. In this case, it should be, for example:
`The properties <a href="GstVideoRate:in">in</a>, <a href="GstVideoRate:out">out</a>, <a href="GstVideoRate:duplicate">duplicate</a>
and <a href="GstVideoRate:drop">drop</a> can be read` [...]
Browser: Firefox 109.0 (64-bit).https://gitlab.freedesktop.org/gstreamer/cerbero/-/issues/408macOS gst-1.22.0 Broken binaries (on macOS 10.15.7 Catalina)2023-02-25T12:01:52ZF. DuncanhmacOS gst-1.22.0 Broken binaries (on macOS 10.15.7 Catalina)The gitlab.freedesktop.org macOS binaries are broken since 1.21.2 (1.21.1 binaries work)
* In gstreamer#1752 I verified that 1.22.0 could be compiled from source on macOS, and behaved perfectly in my tests with the AirPlay server "uxpla...The gitlab.freedesktop.org macOS binaries are broken since 1.21.2 (1.21.1 binaries work)
* In gstreamer#1752 I verified that 1.22.0 could be compiled from source on macOS, and behaved perfectly in my tests with the AirPlay server "uxplay". This was tested and rebuilt with 1.22.0, 1.21.1, 1.21.2, 1.21.3 binaries, and with my own build of 1.22.0. This is in macOS 10.15.7 Catalina.
* 1.21.1 binaries work fine. something broke in 1.21.2, and something more broke in 1.21.3 which behaves identically to 1.22.0
* with 1.22.0 binaries I get
```
% rm -rf ~/.cache/gstreamer-1.0
% ./uxplay
UxPlay 1.62: An Open-Source AirPlay mirroring and audio-streaming server.
ERROR: Caught a segmentation fault while loading plugin file:
/Library/Frameworks/GStreamer.framework/Versions/1.0/lib/gstreamer-1.0/libgstgoom.dylib
Please either:
- remove it and restart.
- run with --gst-disable-segtrap --gst-disable-registry-fork and debug.
ERROR: Caught a segmentation fault while loading plugin file:
/Library/Frameworks/GStreamer.framework/Versions/1.0/lib/gstreamer-1.0/libgstgoom.dylib
Please either:
- remove it and restart.
- run with --gst-disable-segtrap --gst-disable-registry-fork and debug.
ERROR: Caught a segmentation fault while loading plugin file:
/Library/Frameworks/GStreamer.framework/Versions/1.0/lib/gstreamer-1.0/libgsta52dec.dylib
Please either:
- remove it and restart.
- run with --gst-disable-segtrap --gst-disable-registry-fork and debug.
ERROR: Caught a segmentation fault while loading plugin file:
/Library/Frameworks/GStreamer.framework/Versions/1.0/lib/gstreamer-1.0/libgsta52dec.dylib
Please either:
- remove it and restart.
- run with --gst-disable-segtrap --gst-disable-registry-fork and debug.
ERROR: Caught a segmentation fault while loading plugin file:
/Library/Frameworks/GStreamer.framework/Versions/1.0/lib/gstreamer-1.0/libgstdeinterlace.dylib
Please either:
- remove it and restart.
- run with --gst-disable-segtrap --gst-disable-registry-fork and debug.
ERROR: Caught a segmentation fault while loading plugin file:
/Library/Frameworks/GStreamer.framework/Versions/1.0/lib/gstreamer-1.0/libgstdeinterlace.dylib
Please either:
- remove it and restart.
- run with --gst-disable-segtrap --gst-disable-registry-fork and debug.
ERROR: Caught a segmentation fault while loading plugin file:
/Library/Frameworks/GStreamer.framework/Versions/1.0/lib/gstreamer-1.0/libgstdtsdec.dylib
Please either:
- remove it and restart.
- run with --gst-disable-segtrap --gst-disable-registry-fork and debug.
ERROR: Caught a segmentation fault while loading plugin file:
/Library/Frameworks/GStreamer.framework/Versions/1.0/lib/gstreamer-1.0/libgstdtsdec.dylib
Please either:
- remove it and restart.
- run with --gst-disable-segtrap --gst-disable-registry-fork and debug.
raop_rtp_mirror starting mirroring
Begin streaming to GStreamer video pipeline
```
* at this point a video screen (glimagesink is selected by autovideosink) should have opened, but nothing happened.
* now an audio stream starts:
```
raop_rtp starting audio
dyld: lazy symbol binding failed: Symbol not found: _pthread_jit_write_protect_supported_np
Referenced from: /Library/Frameworks/GStreamer.framework/Versions/1.0/lib/../lib/liborc-0.4.0.dylib
Expected in: /usr/lib/libSystem.B.dylib
dyld: Symbol not found: _pthread_jit_write_protect_supported_np
Referenced from: /Library/Frameworks/GStreamer.framework/Versions/1.0/lib/../lib/liborc-0.4.0.dylib
Expected in: /usr/lib/libSystem.B.dylib
zsh: abort ./uxplay
```
* with 1.21.1, things work fine:
* with 1.21.2, it's the same as 1.22.0, but the video window does open.
* 1.21.3 behaves identically to 1.22.0
* the native build of 1.22.0 from source I did works perfectly
what in the build process changed after 1.21.1?
```https://gitlab.freedesktop.org/gstreamer/gstreamer/-/issues/1759macOS gst-1.22.0 Broken binaries2023-01-31T12:19:15ZF. DuncanhmacOS gst-1.22.0 Broken binariesThe gitlab.freedesktop.org macOS binaries are broken since 1.21.1
* In #1752 I verified that 1.22.0 could be compiled from source on macOS, and behaved perfectly in my tests with the AirPlay server "uxplay". This was tested and rebuil...The gitlab.freedesktop.org macOS binaries are broken since 1.21.1
* In #1752 I verified that 1.22.0 could be compiled from source on macOS, and behaved perfectly in my tests with the AirPlay server "uxplay". This was tested and rebuilt with 1.22.0, 1.21.1, 1.21.2, 1.21.3 binaries, and with my own build of 1.22.0. This is in macOS 10.15.7 Catalina.
* 1.21.1 binaries work fine. something broke in 1.21.2, and something more broke in 1.21.3 which behaves identically to 1.22.0
* with 1.22.0 binaries I get
```
% rm -rf ~/.cache/gstreamer-1.0
% ./uxplay
UxPlay 1.62: An Open-Source AirPlay mirroring and audio-streaming server.
ERROR: Caught a segmentation fault while loading plugin file:
/Library/Frameworks/GStreamer.framework/Versions/1.0/lib/gstreamer-1.0/libgstgoom.dylib
Please either:
- remove it and restart.
- run with --gst-disable-segtrap --gst-disable-registry-fork and debug.
ERROR: Caught a segmentation fault while loading plugin file:
/Library/Frameworks/GStreamer.framework/Versions/1.0/lib/gstreamer-1.0/libgstgoom.dylib
Please either:
- remove it and restart.
- run with --gst-disable-segtrap --gst-disable-registry-fork and debug.
ERROR: Caught a segmentation fault while loading plugin file:
/Library/Frameworks/GStreamer.framework/Versions/1.0/lib/gstreamer-1.0/libgsta52dec.dylib
Please either:
- remove it and restart.
- run with --gst-disable-segtrap --gst-disable-registry-fork and debug.
ERROR: Caught a segmentation fault while loading plugin file:
/Library/Frameworks/GStreamer.framework/Versions/1.0/lib/gstreamer-1.0/libgsta52dec.dylib
Please either:
- remove it and restart.
- run with --gst-disable-segtrap --gst-disable-registry-fork and debug.
ERROR: Caught a segmentation fault while loading plugin file:
/Library/Frameworks/GStreamer.framework/Versions/1.0/lib/gstreamer-1.0/libgstdeinterlace.dylib
Please either:
- remove it and restart.
- run with --gst-disable-segtrap --gst-disable-registry-fork and debug.
ERROR: Caught a segmentation fault while loading plugin file:
/Library/Frameworks/GStreamer.framework/Versions/1.0/lib/gstreamer-1.0/libgstdeinterlace.dylib
Please either:
- remove it and restart.
- run with --gst-disable-segtrap --gst-disable-registry-fork and debug.
ERROR: Caught a segmentation fault while loading plugin file:
/Library/Frameworks/GStreamer.framework/Versions/1.0/lib/gstreamer-1.0/libgstdtsdec.dylib
Please either:
- remove it and restart.
- run with --gst-disable-segtrap --gst-disable-registry-fork and debug.
ERROR: Caught a segmentation fault while loading plugin file:
/Library/Frameworks/GStreamer.framework/Versions/1.0/lib/gstreamer-1.0/libgstdtsdec.dylib
Please either:
- remove it and restart.
- run with --gst-disable-segtrap --gst-disable-registry-fork and debug.
raop_rtp_mirror starting mirroring
Begin streaming to GStreamer video pipeline
```
* at this point a video screen (glimagesink is selected by autovideosink) should have opened, but nothing happened.
* now an audio stream starts:
```
raop_rtp starting audio
dyld: lazy symbol binding failed: Symbol not found: _pthread_jit_write_protect_supported_np
Referenced from: /Library/Frameworks/GStreamer.framework/Versions/1.0/lib/../lib/liborc-0.4.0.dylib
Expected in: /usr/lib/libSystem.B.dylib
dyld: Symbol not found: _pthread_jit_write_protect_supported_np
Referenced from: /Library/Frameworks/GStreamer.framework/Versions/1.0/lib/../lib/liborc-0.4.0.dylib
Expected in: /usr/lib/libSystem.B.dylib
zsh: abort ./uxplay
```
* with 1.21.1, things work fine:
* with 1.21.2, it's the same as 1.22.0, but the video window does open.
* 1.21.3 behaves identically to 1.22.0
* the native build from source I did works perfectly
```https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/issues/1753nvvp9dec: internal data flow problems causing issues with hlssink2 (warnings ...2023-02-03T08:34:40ZChristian Koopnvvp9dec: internal data flow problems causing issues with hlssink2 (warnings and criticals)* **GStreamer**: `1.20.5`
* **GPU**: NVIDIA GeForce 1070
* **OS**: Manjaro Linux (arch based)
---
I have a pipeline that basically turns any input video file into a desired HLS output (via `uridecodebin` and `hlssink2`).
I encountered...* **GStreamer**: `1.20.5`
* **GPU**: NVIDIA GeForce 1070
* **OS**: Manjaro Linux (arch based)
---
I have a pipeline that basically turns any input video file into a desired HLS output (via `uridecodebin` and `hlssink2`).
I encountered some issues with a specific video file that produces
a broken HLS manifest (and maybe segments too) in regard to the *targetDuration*.
The troubling bit is that when End-Of-Stream is reached, hlssink2 seems to rewrite the playlist file and fixed the broken durations (matching the generated segments?).
That is why we might need to kill/interrupt the process before it reaches the EOS – The duration is still not the expected `4` for the segments after the rewrite.
---
Video file (removed all streams except the video stream and only kept the first 20 seconds): [test.mkv](/uploads/399b18451722a32515d5abaa362c2d77/test.mkv)
My simple command: `gst-launch-1.0 filesrc location=./test.mkv ! matroskademux ! vp9parse ! nvvp9dec ! videoconvert ! autovideosink`
My command using `hlssink2` (kill/interrupt the process at ca. 70% progress): `gst-launch-1.0 filesrc location=./test.mkv ! matroskademux ! vp9parse ! nvvp9dec ! videoconvert ! videorate ! video/x-raw,framerate=24/1 ! x264enc key-int-max=24 ! hlssink2 target-duration=4`
**Using `vp9dec` instead of `nvvp9dec` gets rid of the warnings and issues (the output playlist.m3u8 properly generates with durations of 4 too).
---
The command using `hlssink2` prints the following:
```
Setting pipeline to PAUSED ...
Pipeline is PREROLLING ...
Got context from element 'nvvp9dec0': gst.cuda.context=context, gst.cuda.context=(GstCudaContext)"\(GstCudaContext\)\ cudacontext0", cuda-device-id=(int)0;
Got context from element 'nvvp9dec0': gst.gl.GLDisplay=context, gst.gl.GLDisplay=(GstGLDisplay)"\(GstGLDisplayX11\)\ gldisplayx11-0";
Redistribute latency...
Redistribute latency...
(gst-launch-1.0:353098): GStreamer-WARNING **: 13:37:52.316: ../gstreamer/subprojects/gstreamer/gst/gstpad.c:4677:gst_pad_push_data:<nvvp9dec0:src> Got data flow before segment event
(gst-launch-1.0:353098): GStreamer-WARNING **: 13:37:52.316: ../gstreamer/subprojects/gstreamer/gst/gstpad.c:4416:gst_pad_chain_data_unchecked:<videoconvert0:sink> Got data flow before segment event
(gst-launch-1.0:353098): GStreamer-WARNING **: 13:37:52.316: ../gstreamer/subprojects/gstreamer/gst/gstpad.c:4677:gst_pad_push_data:<videoconvert0:src> Got data flow before segment event
(gst-launch-1.0:353098): GStreamer-WARNING **: 13:37:52.316: ../gstreamer/subprojects/gstreamer/gst/gstpad.c:4416:gst_pad_chain_data_unchecked:<videorate0:sink> Got data flow before segment event
(gst-launch-1.0:353098): GStreamer-WARNING **: 13:37:52.319: ../gstreamer/subprojects/gstreamer/gst/gstpad.c:4677:gst_pad_push_data:<videorate0:src> Got data flow before segment event
(gst-launch-1.0:353098): GStreamer-WARNING **: 13:37:52.319: ../gstreamer/subprojects/gstreamer/gst/gstpad.c:4416:gst_pad_chain_data_unchecked:<capsfilter0:sink> Got data flow before segment event
(gst-launch-1.0:353098): GStreamer-WARNING **: 13:37:52.319: ../gstreamer/subprojects/gstreamer/gst/gstpad.c:4677:gst_pad_push_data:<capsfilter0:src> Got data flow before segment event
(gst-launch-1.0:353098): GStreamer-WARNING **: 13:37:52.319: ../gstreamer/subprojects/gstreamer/gst/gstpad.c:4416:gst_pad_chain_data_unchecked:<x264enc0:sink> Got data flow before segment event
(gst-launch-1.0:353098): GStreamer-WARNING **: 13:37:52.639: ../gstreamer/subprojects/gstreamer/gst/gstpad.c:4677:gst_pad_push_data:<x264enc0:src> Got data flow before segment event
(gst-launch-1.0:353098): GStreamer-WARNING **: 13:37:52.640: ../gstreamer/subprojects/gstreamer/gst/gstpad.c:4416:gst_pad_chain_data_unchecked:<hlssink2-0:video> Got data flow before segment event
(gst-launch-1.0:353098): GStreamer-WARNING **: 13:37:52.640: ../gstreamer/subprojects/gstreamer/gst/gstpad.c:4677:gst_pad_push_data:<video:proxypad1> Got data flow before segment event
(gst-launch-1.0:353098): GStreamer-WARNING **: 13:37:52.640: ../gstreamer/subprojects/gstreamer/gst/gstpad.c:4416:gst_pad_chain_data_unchecked:<splitmuxsink0:video> Got data flow before segment event
(gst-launch-1.0:353098): GStreamer-WARNING **: 13:37:52.640: ../gstreamer/subprojects/gstreamer/gst/gstpad.c:4677:gst_pad_push_data:<video:proxypad0> Got data flow before segment event
(gst-launch-1.0:353098): GStreamer-WARNING **: 13:37:52.640: ../gstreamer/subprojects/gstreamer/gst/gstpad.c:4416:gst_pad_chain_data_unchecked:<queue_video:sink> Got data flow before segment event
(gst-launch-1.0:353098): GStreamer-CRITICAL **: 13:37:52.640: gst_segment_to_running_time_full: assertion 'segment->format == format' failed
(gst-launch-1.0:353098): GStreamer-CRITICAL **: 13:37:52.640: gst_segment_to_running_time_full: assertion 'segment->format == format' failed
(gst-launch-1.0:353098): GStreamer-CRITICAL **: 13:37:52.640: gst_segment_to_running_time_full: assertion 'segment->format == format' failed
(gst-launch-1.0:353098): GStreamer-CRITICAL **: 13:37:52.640: gst_segment_to_running_time_full: assertion 'segment->format == format' failed
(gst-launch-1.0:353098): GStreamer-WARNING **: 13:37:52.640: ../gstreamer/subprojects/gstreamer/gst/gstpad.c:4677:gst_pad_push_data:<queue_video:src> Got data flow before segment event
Redistribute latency...
Pipeline is PREROLLED ...
Setting pipeline to PLAYING ...
New clock: GstSystemClock
(gst-launch-1.0:353098): GStreamer-WARNING **: 13:37:52.897: ../gstreamer/subprojects/gstreamer/gst/gstpad.c:4416:gst_pad_chain_data_unchecked:<mpegtsmux0:sink_65> Got data flow before segment event
(gst-launch-1.0:353098): GStreamer-CRITICAL **: 13:37:52.901: gst_segment_to_running_time: assertion 'segment->format == format' failed
```
---
The generated `playlist.m3u8`:
`#EXT-X-TARGETDURATION` should be 4 here and `#EXTINF` should be 4 too as I force the framerate and keyframes to align.
```m3u
#EXTM3U
#EXT-X-VERSION:3
#EXT-X-MEDIA-SEQUENCE:0
#EXT-X-TARGETDURATION:633437445
#EXTINF:9223371776,
segment00000.ts
#EXTINF:3.5416665077209473,
segment00001.ts
#EXTINF:4,
segment00002.ts
#EXT-X-ENDLIST
```https://gitlab.freedesktop.org/gstreamer/gstreamer/-/issues/1754Incorrect line endings in the Windows CI image checkout2023-11-23T18:10:59ZAndoni Morales AlastrueyIncorrect line endings in the Windows CI image checkoutI open this issue to write down what I found so far, as it will hit the Windows CI once meson is upgraded with support for `diff_files` in wrap. I tried the update in a branch and I had to roll back since all wraps with diff_files were f...I open this issue to write down what I found so far, as it will hit the Windows CI once meson is upgraded with support for `diff_files` in wrap. I tried the update in a branch and I had to roll back since all wraps with diff_files were failing to apply their patches.
After seeing the failures in the CI, I was able to reproduce the error locally and it's caused when git changes the line endings from LF to CRLF, for example when `core.autocrlf` is `auto` instead of `false`.
The first thing I did was checking that git is correctly configured in the Windows image:
1. ✅ It is installed with the `/NoAutoCrlf` flag
2. ✅ `git config --get core.autocrlf` returns `false` as expected.
I than tried looking at the line endings:
1. ❌ `(Get-Content subprojects/packagefiles/pango-1.50.12/0001-meson-Fix-pangoft2.pc-when-using-freetype-and-fontco.patch -Raw) -match "\r\n$"` returns `true`
That means that for some reason, the line endings are still not correct.
cc: @nirbheek @tpmhttps://gitlab.freedesktop.org/gstreamer/meson-ports/x264/-/issues/10video files played by totem (22.04ubuntu-repo or flatpak) > only audio, video...2023-01-28T12:32:54Zmrkapqavideo files played by totem (22.04ubuntu-repo or flatpak) > only audio, video blackHello ,
i have a computer where i installed Ubuntu 22.04 as secondary OS.
the graphics card is Polaris based AMD-GPU and have installed proprietary drivers for it.
issue is with the default video player for Ubuntu, which is called "vid...Hello ,
i have a computer where i installed Ubuntu 22.04 as secondary OS.
the graphics card is Polaris based AMD-GPU and have installed proprietary drivers for it.
issue is with the default video player for Ubuntu, which is called "videos" but is in fact "totem".
the player would not play any of the videos; after installing
"ubuntu-restricted-extras" , it would play video, but no output is shown, the video remains black, and audio can be heard.
tried installing "org.gnome.totem" via flatpak, but the issue remains, only audio, no video. several video files tried, same story.
it might be related to the amdgpu drivers, but other players function normally (vlc or mpv as an example).
please let me know a possible solution,
thanks.