GStreamer issueshttps://gitlab.freedesktop.org/groups/gstreamer/-/issues2023-10-13T16:05:44Zhttps://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/issues/334Add support for resolving IPv4 address literals for DNS64 support2023-10-13T16:05:44ZBugzilla Migration UserAdd support for resolving IPv4 address literals for DNS64 support## Submitted by joe..@..il.com
**[Link to original bug (#776344)](https://bugzilla.gnome.org/show_bug.cgi?id=776344)**
## Description
The source of creating this bug can be found in the discussion here: http://gstreamer-devel.966125...## Submitted by joe..@..il.com
**[Link to original bug (#776344)](https://bugzilla.gnome.org/show_bug.cgi?id=776344)**
## Description
The source of creating this bug can be found in the discussion here: http://gstreamer-devel.966125.n4.nabble.com/NAT64-support-td4681208.html.
When on NAT64 networks, passing an IPv4 address literal does not get automatically resolved to its IPv6 equivalent so streaming via elements such as udpsink or udpsrc doesn't work. This is required for iOS as described here: https://developer.apple.com/library/content/documentation//NetworkingInternetWeb/Conceptual/NetworkingOverview/UnderstandingandPreparingfortheIPv6Transition/UnderstandingandPreparingfortheIPv6Transition.html.https://gitlab.freedesktop.org/gstreamer/gstreamer/-/issues/927Add support for User Defined Unregistered SEI2022-11-10T09:21:10ZBrad HardsAdd support for User Defined Unregistered SEIMISB uses User Defined Unregistered SEI messages for frame timestamp (ST 0604) and identifier (ST 2101). As far as I can tell, there is no way to expose that to a consumer (e.g. as a Meta).
For an example of this, see https://gitlab.fr...MISB uses User Defined Unregistered SEI messages for frame timestamp (ST 0604) and identifier (ST 2101). As far as I can tell, there is no way to expose that to a consumer (e.g. as a Meta).
For an example of this, see https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/uploads/addca4b4d4d9dca44456d0a5bcf6b8ed/klv_metadata_test_sync.tshttps://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/issues/7Add support for VDPAU (PureVideo)2021-09-13T13:35:48ZBugzilla Migration UserAdd support for VDPAU (PureVideo)## Submitted by Bastien Nocera `@hadess`
**[Link to original bug (#561225)](https://bugzilla.gnome.org/show_bug.cgi?id=561225)**
## Description
A way to accelerate display/decoding for VC-1, MPEG-1, MPEG-2, H264.
### Blocking
* ...## Submitted by Bastien Nocera `@hadess`
**[Link to original bug (#561225)](https://bugzilla.gnome.org/show_bug.cgi?id=561225)**
## Description
A way to accelerate display/decoding for VC-1, MPEG-1, MPEG-2, H264.
### Blocking
* [Bug 744698](https://bugzilla.gnome.org/show_bug.cgi?id=744698)https://gitlab.freedesktop.org/gstreamer/gstreamer/-/issues/1278Add support for windows application loopback audio capture to specify/exclude...2022-10-18T00:23:45ZDániel HorváthAdd support for windows application loopback audio capture to specify/exclude process idThis is a feature request.<br/>
Currently there is no way to capture/exclude application specific audio in windows.<br/>
Windows supports this, but wasapisrc and wasapi2src don't take the TargetProcessId and ProcessLoopbackMode parameter...This is a feature request.<br/>
Currently there is no way to capture/exclude application specific audio in windows.<br/>
Windows supports this, but wasapisrc and wasapi2src don't take the TargetProcessId and ProcessLoopbackMode parameters.<br/>
Related: [https://docs.microsoft.com/en-us/samples/microsoft/windows-classic-samples/applicationloopbackaudio-sample/](https://docs.microsoft.com/en-us/samples/microsoft/windows-classic-samples/applicationloopbackaudio-sample/)
[https://github.com/microsoft/windows-classic-samples/tree/main/Samples/ApplicationLoopback](https://github.com/microsoft/windows-classic-samples/tree/main/Samples/ApplicationLoopback)
Please, add support for this.https://gitlab.freedesktop.org/gstreamer/gst-docs/-/issues/1Add usage example to GstTagSetter2018-11-04T10:05:18ZBugzilla Migration UserAdd usage example to GstTagSetter## Submitted by Marcin Lewandowski
**[Link to original bug (#656562)](https://bugzilla.gnome.org/show_bug.cgi?id=656562)**
## Description
I think it would be useful to add the usage example to the GstTagSetter docs (http://gstreamer...## Submitted by Marcin Lewandowski
**[Link to original bug (#656562)](https://bugzilla.gnome.org/show_bug.cgi?id=656562)**
## Description
I think it would be useful to add the usage example to the GstTagSetter docs (http://gstreamer.freedesktop.org/data/doc/gstreamer/head/gstreamer/html/GstTagSetter.html)
I attach ready and working sample code.https://gitlab.freedesktop.org/gstreamer/gst-docs/-/issues/97Add versioned docs2023-05-24T12:16:38ZSebastian DrögeAdd versioned docsThe docs we currently publish are for the very latest git version, which regularly causes confusion for users.The docs we currently publish are for the very latest git version, which regularly causes confusion for users.https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/issues/505Add vpx (de)muxing capabilities to mpegtsmux and mpegtsdemux2023-06-16T11:27:21ZBugzilla Migration UserAdd vpx (de)muxing capabilities to mpegtsmux and mpegtsdemux## Submitted by Yann Jouanin
**[Link to original bug (#776948)](https://bugzilla.gnome.org/show_bug.cgi?id=776948)**
## Description
Created attachment 343024
Patch to add vpx support to both mpegtsmux and mpegtsdemux
It is cu...## Submitted by Yann Jouanin
**[Link to original bug (#776948)](https://bugzilla.gnome.org/show_bug.cgi?id=776948)**
## Description
Created attachment 343024
Patch to add vpx support to both mpegtsmux and mpegtsdemux
It is currently impossible to stream vpx codec with MPEG-TS muxer (demuxer).
This patch add this functionnality by creating a new stream type (private 0xe0) and a descriptor.
tested with vp8 and vp9
**Patch 343024**, "Patch to add vpx support to both mpegtsmux and mpegtsdemux":
[0001-mpegtsmux-add-support-for-vpx-streams.patch](/uploads/4fe0e0113d4a773d1c1709e02a342f76/0001-mpegtsmux-add-support-for-vpx-streams.patch)https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/issues/222Add Windows CI2022-09-16T09:06:44ZSebastian DrögeAdd Windows CICC @alatieraCC @alatieraJordan PetridіsJordan Petridіshttps://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/issues/29Add wrapper for GstAggregator2018-11-03T21:30:01ZSebastian DrögeAdd wrapper for GstAggregatorhttps://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/issues/10Add wrapper for GstBaseTransform2022-06-23T17:19:05ZSebastian DrögeAdd wrapper for GstBaseTransformSee https://gstreamer.freedesktop.org/data/doc/gstreamer/head/gstreamer-libs/html/GstBaseTransform.html
Might make sense to do this first: https://github.com/sdroege/gstreamer-rs/issues/7
First step would be to come up with an API de...See https://gstreamer.freedesktop.org/data/doc/gstreamer/head/gstreamer-libs/html/GstBaseTransform.html
Might make sense to do this first: https://github.com/sdroege/gstreamer-rs/issues/7
First step would be to come up with an API design (mostly the main trait) in the style of `Source`, `Sink` and `Demuxer`. That is:
- all functions return a value and don't call into other elements. So return value would be some kind of enum
- the general idea of the C base class would be kept (i.e. you have some kind of `transform` function, `transform_caps`, etc)
- implementors of the trait do not have to worry about threading
As part of this should also be a simple transform element. Maybe just a minimal "volume" element, or mono to stereo converter (the latter would at least make use of caps transformations).https://gitlab.freedesktop.org/gstreamer/meson-ports/libvpx/-/issues/3ads2gas.py contains Python 3.9 syntax, does not work with Python 3.82024-03-28T04:55:45ZSeungmin Kimads2gas.py contains Python 3.9 syntax, does not work with Python 3.8```
Configuring vpx_config.c using configuration
Program build/make/rtcd.py found: YES (/tmp/gstreamer/subprojects/vpx/build/make/rtcd.py)
Library m found: YES
Program meson/transform_config_asm.py found: YES (/tmp/gstreamer/subprojec...```
Configuring vpx_config.c using configuration
Program build/make/rtcd.py found: YES (/tmp/gstreamer/subprojects/vpx/build/make/rtcd.py)
Library m found: YES
Program meson/transform_config_asm.py found: YES (/tmp/gstreamer/subprojects/vpx/meson/transform_config_asm.py)
Program meson/stdinout_wrapper.py found: YES (/tmp/gstreamer/subprojects/vpx/meson/stdinout_wrapper.py)
Configuring vpx_config.adstmp with command
Running command: /tmp/gstreamer/subprojects/vpx/meson/transform_config_asm.py /tmp/gstreamer/builddir/subprojec
x_config.h /tmp/gstreamer/builddir/subprojects/vpx/vpx_config.adstmp
--- stdout ---
--- stderr ---
Configuring vpx_config.asm with command
Running command: /tmp/gstreamer/subprojects/vpx/meson/stdinout_wrapper.py --input /tmp/gstreamer/builddir/subpr
x/vpx_config.adstmp --output /tmp/gstreamer/builddir/subprojects/vpx/vpx_config.asm /tmp/gstreamer/subprojects/
/make/ads2gas.py
--- stdout ---
--- stderr ---
Traceback (most recent call last):
File "/tmp/gstreamer/subprojects/vpx/build/make/ads2gas.py", line 26, in <module>
proc_stack: list[str] = []
TypeError: 'type' object is not subscriptable
Traceback (most recent call last):
File "/tmp/gstreamer/subprojects/vpx/meson/stdinout_wrapper.py", line 19, in <module>
subprocess.run([args.executable] + args.args, input=args.input.read(), stdout=args.output, check=True)
File "/usr/lib/python3.8/subprocess.py", line 516, in run
raise CalledProcessError(retcode, process.args,
subprocess.CalledProcessError: Command '['/tmp/gstreamer/subprojects/vpx/build/make/ads2gas.py']' returned non-
status 1.
subprojects/vpx/meson.build:1690:14: ERROR: Command `/tmp/gstreamer/subprojects/vpx/meson/stdinout_wrapper.py -
mp/gstreamer/builddir/subprojects/vpx/vpx_config.adstmp --output /tmp/gstreamer/builddir/subprojects/vpx/vpx_co
/tmp/gstreamer/subprojects/vpx/build/make/ads2gas.py` failed with status 1.
```
CC @amyspark @nirbheekamysparkamysparkhttps://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/issues/1747AESdec wav file2023-07-13T16:57:15ZDan SirbuAESdec wav fileI'm trying to use AES enc/dec with wav. I have run the following pipeline to encrypt a wav file:
gst-launch-1.0 filesrc location=./convo2_long.raw ! rawaudioparse format=mulaw sample-rate=8000 num-channels=1 use-sink-caps =false ! mulaw...I'm trying to use AES enc/dec with wav. I have run the following pipeline to encrypt a wav file:
gst-launch-1.0 filesrc location=./convo2_long.raw ! rawaudioparse format=mulaw sample-rate=8000 num-channels=1 use-sink-caps =false ! mulawdec ! audioconvert ! capsfilter caps=audio/x-raw,format=S16LE ! wavenc ! aesenc key=1f9423681beb9a79215820f6bda73d0f iv=e9aa8e834d8d70 b7e0d254ff670dd718 per-buffer-padding=false ! filesink location=./convo2_long_enc.wav
And based on the 'log_aesenc' logs I think it is doing what is supposed to.
Then, I tried to decode the wav file using:
gst-launch-1.0 filesrc location=./convo2_long_enc.wav ! aesdec key=1f9423681beb9a79215820f6bda73d0f iv=e9aa8e834d8d70b7e0d25 4ff670dd718 per-buffer-padding=false ! filesink location=./convo2_long_dec.wav
but it fails with:
ERROR: from element /GstPipeline:pipeline0/GstAesDec:aesdec0: Cipher finalization failed. Additional debug info: ../ext/aes/gstaesdec.c(416): gst_aes_dec_sink_event (): /GstPipeline:pipeline0/GstAesDec:aesdec0: Error while finalizing the stream
I do not get any extra details in the log file.
GST_DEBUG="2,wavenc:7,aesenc:7,aesdec:7" GST_DEBUG_FILE="./log.txt"[log_aesdec.txt](/gstreamer/gst-build/uploads/972a382ed909fc7e5f1c96a40d66a23f/log_aesdec.txt)
Do I do something wrong or it is a bug ?[log_aesenc.txt](/uploads/ed5d67c653ad42cfa195b7137c1cc2ce/log_aesenc.txt)[log_aesdec.txt](/uploads/410c7ed6c1cd6d54b4c95102e5417c06/log_aesdec.txt)https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/issues/1660aesenc/aesdec unit tests fail2021-09-15T13:19:39ZU. Artie Eoffaesenc/aesdec unit tests failIf openssl development packages aren't installed, then aesenc/aesdec elements are not compiled. However, the associated unit tests are still compiled and fail. Patch forthcoming.If openssl development packages aren't installed, then aesenc/aesdec elements are not compiled. However, the associated unit tests are still compiled and fail. Patch forthcoming.https://gitlab.freedesktop.org/gstreamer/gstreamer/-/issues/1264AES-GCM to support RFC3711 in the Gstreamer2022-06-20T07:00:43ZChandrasekhar JharaplaAES-GCM to support RFC3711 in the GstreamerI am unable to encode and decode the stream with libsrtp
gst-launch-1.0 udpsrc port=5200 caps='application/x-srtp, encoding-name=JPEG,
ssrc=(uint)1356955624,\
srtp-key=(buffer)012345678901234567890123456789012345678901234567890123456789...I am unable to encode and decode the stream with libsrtp
gst-launch-1.0 udpsrc port=5200 caps='application/x-srtp, encoding-name=JPEG,
ssrc=(uint)1356955624,\
srtp-key=(buffer)012345678901234567890123456789012345678901234567890123456789,\
srtp-cipher=(string)aes-128-gcm, \
srtcp-cipher=(string)aes-128-gcm, roc=(uint)0'\
! srtpdec ! rtpjpegdepay ! jpegdec ! autovideosink
with the above pipe line I am not encode the data as it is getting encoder errorhttps://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/issues/1723AES-GCM to support RFC3711 in the Gstreamer2022-06-07T06:19:37ZChandrasekhar JharaplaAES-GCM to support RFC3711 in the GstreamerI am unable to encode and decode the stream with libsrtp
gst-launch-1.0 udpsrc port=5200 caps='application/x-srtp, encoding-name=JPEG,
ssrc=(uint)1356955624,\
srtp-key=(buffer)012345678901234567890123456789012345678901234567890123456789...I am unable to encode and decode the stream with libsrtp
gst-launch-1.0 udpsrc port=5200 caps='application/x-srtp, encoding-name=JPEG,
ssrc=(uint)1356955624,\
srtp-key=(buffer)012345678901234567890123456789012345678901234567890123456789,\
srtp-cipher=(string)aes-128-gcm, \
srtcp-cipher=(string)aes-128-gcm, roc=(uint)0'\
! srtpdec ! rtpjpegdepay ! jpegdec ! autovideosink
with the above pipe line I am not encode the data as it is getting encoder errorhttps://gitlab.freedesktop.org/gstreamer/gstreamer/-/issues/1668After call gst_element_factory_make(), return NULL2022-12-22T07:27:39ZLevey HermansAfter call gst_element_factory_make(), return NULLHi:
I wrote a very simple piece of code below:
![1671617858638](/uploads/88f5f0f94dd0fb10fe8abe807e9b18d5/1671617858638.png)
but it went wrong.
after called this function: gst_element_factory_make(), it's return is NULL.
Please help t...Hi:
I wrote a very simple piece of code below:
![1671617858638](/uploads/88f5f0f94dd0fb10fe8abe807e9b18d5/1671617858638.png)
but it went wrong.
after called this function: gst_element_factory_make(), it's return is NULL.
Please help to confirm the problem,Thk!https://gitlab.freedesktop.org/gstreamer/gstreamer/-/issues/1060After openh264enc all the VideoMeta is lost2022-03-02T14:38:14Zzack xueAfter openh264enc all the VideoMeta is lostit seems that the base video encode don't use gst_buffer_copy_into GST_BUFFER_COPY_METADATA?it seems that the base video encode don't use gst_buffer_copy_into GST_BUFFER_COPY_METADATA?https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/issues/1706After openh264enc all the VideoMeta is lost2022-03-02T06:27:46Zzack xueAfter openh264enc all the VideoMeta is lostit seems that the base video encode don't use gst_buffer_copy_into GST_BUFFER_COPY_METADATA?it seems that the base video encode don't use gst_buffer_copy_into GST_BUFFER_COPY_METADATA?https://gitlab.freedesktop.org/gstreamer/gstreamer/-/issues/603aggregator API is inherently racy2020-10-14T13:06:56ZVivia Nikolaidouaggregator API is inherently racyThe following discussion from https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/711 should be addressed:
- [ ] @meh started a [discussion](https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_reques...The following discussion from https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/711 should be addressed:
- [ ] @meh started a [discussion](https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/711#note_612623): (+1 comment)
> Hm, what if the pad gets flush stopped and receives a new buffer in the interval? In that case, we may end up not muxing the "best" pad. The alternative solution is to keep a reference to the "best" buffer alongside the best pad, and call drop_buffer() on it once actually processed.
Basically:
- @slomo [said](https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/711#note_612657):
> The only thing you can do is to pop() and then use up the buffer or queue it up locally. Everything else is racy :) So peek() and drop() are useless API and we just noticed that the aggregator API is bad 🤪https://gitlab.freedesktop.org/gstreamer/gstreamer/-/issues/888aggregator: does not forward navigation events2022-03-18T06:59:25ZFrederic Turmelaggregator: does not forward navigation eventsThis was found running this pipeline using the wpesrc plugins. (Master)
Navigation events are forwarded without the glvideomixer with this pipeline:
gst-launch-1.0 -v wpesrc location="https://gstreamer.freedesktop.org" ! queue ! glimage...This was found running this pipeline using the wpesrc plugins. (Master)
Navigation events are forwarded without the glvideomixer with this pipeline:
gst-launch-1.0 -v wpesrc location="https://gstreamer.freedesktop.org" ! queue ! glimagesink"
When including glvideomixer the navigation events are not forwarded
Example: gst-launch-1.0 glvideomixer name=m sink_0::zorder=1 ! glimagesink sync=false async=false wpesrc location="http://www.google.com" draw-background=1 ! video/x-raw\(memory:GLMemory\),height=720,width=1280 ! m.sink_0