GStreamer issueshttps://gitlab.freedesktop.org/groups/gstreamer/-/issues2021-09-13T13:35:49Zhttps://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/issues/23mpegtsmux: add CBR mode2021-09-13T13:35:49ZBugzilla Migration Usermpegtsmux: add CBR mode## Submitted by Bob Forsman
**[Link to original bug (#626867)](https://bugzilla.gnome.org/show_bug.cgi?id=626867)**
## Description
The streams generated by mpegtsmux are variable bit rate. Many of the applications that require MPEG...## Submitted by Bob Forsman
**[Link to original bug (#626867)](https://bugzilla.gnome.org/show_bug.cgi?id=626867)**
## Description
The streams generated by mpegtsmux are variable bit rate. Many of the applications that require MPEG transport streams require constant bit rate streams that adhere to the T-STD buffer model (ISO 13818 part 1 section 2.4.2)https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/issues/24[dshowvideosink] add I420 support2021-09-13T13:35:49ZBugzilla Migration User[dshowvideosink] add I420 support## Submitted by Thomas Löwe
**[Link to original bug (#628258)](https://bugzilla.gnome.org/show_bug.cgi?id=628258)**
## Description
Long time ago i've made a patch to support i420 in dshowvideosink.
After the commit from 07/2010...## Submitted by Thomas Löwe
**[Link to original bug (#628258)](https://bugzilla.gnome.org/show_bug.cgi?id=628258)**
## Description
Long time ago i've made a patch to support i420 in dshowvideosink.
After the commit from 07/2010 "Improvements contributed from the Moovida projet" this isn't longer working in the evr mode (vmr still works fine).
Could somebody please look at this or implement i420 format for dshow?
Thanks,
Thomashttps://gitlab.freedesktop.org/gstreamer/gstreamer/-/issues/2641Add support for DivX XSUB subtitles2023-06-06T09:53:18ZBugzilla Migration UserAdd support for DivX XSUB subtitles## Submitted by Bastien Nocera `@hadess`
**[Link to original bug (#628429)](https://bugzilla.gnome.org/show_bug.cgi?id=628429)**
## Description
Created attachment 169171
test.divx
$ gst-launch playbin2 'uri=file:///home/hades...## Submitted by Bastien Nocera `@hadess`
**[Link to original bug (#628429)](https://bugzilla.gnome.org/show_bug.cgi?id=628429)**
## Description
Created attachment 169171
test.divx
$ gst-launch playbin2 'uri=file:///home/hadess/Desktop/test.avi'
Setting pipeline to PAUSED ...
Pipeline is PREROLLING ...
WARNING: from element /GstPlayBin2:playbin20/GstURIDecodeBin:uridecodebin0: No decoder available for type 'video/x-avi-unknown, fourcc=(fourcc)DXSB'.
Additional debug info:
gsturidecodebin.c(712): unknown_type_cb (): /GstPlayBin2:playbin20/GstURIDecodeBin:uridecodebin0
Pipeline is PREROLLED ...
Setting pipeline to PLAYING ...
New clock: GstPulseSinkClock
The file was created with a "normal" avi file, a corresponding srt file, and AviAddXSUBs[1]. This is the "supported" way to embed subtitles in DIVX files for use on the PS3, and other "DivX certified" players.
(If the given file isn't good enough, feel free to ask me for another test one)
[1]: http://www.calcitapp.com/AVIAddXSubs.php (Windows app, works in WINE)
**Attachment 169171**, "test.divx":
[test.avi](/uploads/fad603faa6545fa5111fb0a5cc1ed958/test.avi)https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/issues/25Add support for DivX XSUB subtitles2023-06-06T09:46:31ZBugzilla Migration UserAdd support for DivX XSUB subtitles## Submitted by Bastien Nocera `@hadess`
**[Link to original bug (#628429)](https://bugzilla.gnome.org/show_bug.cgi?id=628429)**
## Description
Created attachment 169171
test.divx
$ gst-launch playbin2 'uri=file:///home/hades...## Submitted by Bastien Nocera `@hadess`
**[Link to original bug (#628429)](https://bugzilla.gnome.org/show_bug.cgi?id=628429)**
## Description
Created attachment 169171
test.divx
$ gst-launch playbin2 'uri=file:///home/hadess/Desktop/test.avi'
Setting pipeline to PAUSED ...
Pipeline is PREROLLING ...
WARNING: from element /GstPlayBin2:playbin20/GstURIDecodeBin:uridecodebin0: No decoder available for type 'video/x-avi-unknown, fourcc=(fourcc)DXSB'.
Additional debug info:
gsturidecodebin.c(712): unknown_type_cb (): /GstPlayBin2:playbin20/GstURIDecodeBin:uridecodebin0
Pipeline is PREROLLED ...
Setting pipeline to PLAYING ...
New clock: GstPulseSinkClock
The file was created with a "normal" avi file, a corresponding srt file, and AviAddXSUBs[1]. This is the "supported" way to embed subtitles in DIVX files for use on the PS3, and other "DivX certified" players.
(If the given file isn't good enough, feel free to ask me for another test one)
[1]: http://www.calcitapp.com/AVIAddXSubs.php (Windows app, works in WINE)
**Attachment 169171**, "test.divx":
[test.avi](/uploads/e9e4c5c80b100195810f734c05f938de/test.avi)https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/issues/33souphttpsrc: Don't do manual proxy configuration from environment variable bu...2021-09-24T13:29:53ZBugzilla Migration Usersouphttpsrc: Don't do manual proxy configuration from environment variable but depend on libsoup doing the right thing## Submitted by Justin Delegard
**[Link to original bug (#628746)](https://bugzilla.gnome.org/show_bug.cgi?id=628746)**
## Description
The gstreamer soup extension sets the proxy value from the environment, however in the event of a...## Submitted by Justin Delegard
**[Link to original bug (#628746)](https://bugzilla.gnome.org/show_bug.cgi?id=628746)**
## Description
The gstreamer soup extension sets the proxy value from the environment, however in the event of a pac config, it sets an inappropriate value which results in a name resolution failure in libsoup when trying to use the badly-configured proxy value.
One of the effects of this is that rhythmbox cannot stream last.fm through a proxy. It always whines about a proxy name resolution failure.
If you comment out every call to gst_soup_http_src_set_proxy inside gst-plugins-good/ext/soup/gstsouphttpsrc.c it streams correctly. libsoup handles environment and configured proxies itself, so I think that attempting to set libsoup's proxy from the env in gstreamer is redundant.https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/issues/26assrender: two cases where it doesn't use embedded fonts2021-09-24T14:31:49ZBugzilla Migration Userassrender: two cases where it doesn't use embedded fonts## Submitted by Hammered
**[Link to original bug (#629451)](https://bugzilla.gnome.org/show_bug.cgi?id=629451)**
## Description
I found two cases where assrender doesn't use the embedded fonts to render the text.
First case:
...## Submitted by Hammered
**[Link to original bug (#629451)](https://bugzilla.gnome.org/show_bug.cgi?id=629451)**
## Description
I found two cases where assrender doesn't use the embedded fonts to render the text.
First case:
1. Create a sample matroska file with mkvmerge that has a video track, an ASS track and an SRT track in that specific order. Also embed the fonts that the ASS track uses and make sure that these fonts don't exist in the system folder.
2. Launch the file in Totem. Notice that the subs will be rendered properly and the embedded fonts will be used correctly.
3. Change the Subtitles(through Totem's menu) to the SRT track. Allow it to render a line of text and then change it back to the ASS track. Observe now that the rendered subs aren't using the embedded fonts.
Second case;
1. The same as above number 1 but the tracks are in this order: video, SRT, ASS
2. Launch the file in Totem and change to the ASS track. Notice that the subs don't use the embedded fonts.
I use Debian Squeeze(testing) amd64.
Version: 1.4.3https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/issues/34rtpmp4apay: add "config-interval" property to multiplex aac audio config data...2021-09-24T13:29:54ZBugzilla Migration Userrtpmp4apay: add "config-interval" property to multiplex aac audio config data into resulting data stream## Submitted by wul..@..il.com
**[Link to original bug (#629545)](https://bugzilla.gnome.org/show_bug.cgi?id=629545)**
## Description
Freature Request: rtpmp4apay element GStreamer
Up till now with the help of rtpmp4apay one is...## Submitted by wul..@..il.com
**[Link to original bug (#629545)](https://bugzilla.gnome.org/show_bug.cgi?id=629545)**
## Description
Freature Request: rtpmp4apay element GStreamer
Up till now with the help of rtpmp4apay one is able to stream aac audio data compliant to RFC 3016. But it's still necessary to tell the backend
the config parameter (StreamMuxConfig element) by means of SDP for example in order to listen to the music at the receiver side. This request
is about adding the feature of multiplexing aac audio config data into the resulting data stream.https://gitlab.freedesktop.org/gstreamer/gst-plugins-base/-/issues/38baseaudiosink: Newly added sinks don't get notified about the bin's latency2021-09-24T13:20:11ZBugzilla Migration Userbaseaudiosink: Newly added sinks don't get notified about the bin's latency## Submitted by Håvard Graff (hgr)
**[Link to original bug (#630441)](https://bugzilla.gnome.org/show_bug.cgi?id=630441)**
## Description
Created attachment 170939
patch
1. make sure we have us_latency.
2. remember to subtr...## Submitted by Håvard Graff (hgr)
**[Link to original bug (#630441)](https://bugzilla.gnome.org/show_bug.cgi?id=630441)**
## Description
Created attachment 170939
patch
1. make sure we have us_latency.
2. remember to subtract render_delay, as this is added on to the latency
~~**Patch 170939**~~, "patch":
[0024-baseaudiosink-fixes-to-sync_latency.patch](/uploads/7d540974f65cce04d7d142402b710488/0024-baseaudiosink-fixes-to-sync_latency.patch)https://gitlab.freedesktop.org/gstreamer/gst-plugins-base/-/issues/39codec-utils: add support for HE-AAC level/profile properly2019-03-15T21:27:35ZBugzilla Migration Usercodec-utils: add support for HE-AAC level/profile properly## Submitted by Tim Müller `@tpm`
**[Link to original bug (#631460)](https://bugzilla.gnome.org/show_bug.cgi?id=631460)**
## Description
As a first step, the base-profile vs. profile thing could be fixed up for the backwards-compati...## Submitted by Tim Müller `@tpm`
**[Link to original bug (#631460)](https://bugzilla.gnome.org/show_bug.cgi?id=631460)**
## Description
As a first step, the base-profile vs. profile thing could be fixed up for the backwards-compatible cases (where base-profile != profile).https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/issues/35rtspsrc: pausing does not flush buffers2021-09-24T13:29:55ZBugzilla Migration Userrtspsrc: pausing does not flush buffers## Submitted by gud..@..il.com
**[Link to original bug (#631648)](https://bugzilla.gnome.org/show_bug.cgi?id=631648)**
## Description
Created attachment 171927
log of pausing a rtsp (mp4a) session
Here is how I reproduced on ...## Submitted by gud..@..il.com
**[Link to original bug (#631648)](https://bugzilla.gnome.org/show_bug.cgi?id=631648)**
## Description
Created attachment 171927
log of pausing a rtsp (mp4a) session
Here is how I reproduced on gstreamer-0.10.29 / gst-plugins-good-0.10.22 / gst-plugins-bad-0.10.18
1. launching in cmdline: totem-gstreamer --gst-debug=3,faad:5
2. start play rtsp://rx-mep-ptest01.real.com/ptest/7/5/1/0/0/PF000202_200111000000157.M4A
3. play for couple seconds and pause
4. wait 5 seconds and resume
5. I am expecting faad incoming buffer timestamps are 5 seconds later than before pausing, but there are still some timestamps looks like generated before pausing, aka the buffers before pausing are not flushed.
Example see my attached log:
faad runs to 9second before pausing, after pausing about 5 seconds and resume, faad is still getting two buffers timestamped at 9seconds.
**Attachment 171927**, "log of pausing a rtsp (mp4a) session":
[gstrtspfaad.log](/uploads/0974025f748895680a88643ff29992bc/gstrtspfaad.log)https://gitlab.freedesktop.org/gstreamer/www/-/issues/3GStreamer: artwork2018-11-03T10:30:52ZBugzilla Migration UserGStreamer: artwork## Submitted by John
**[Link to original bug (#631907)](https://bugzilla.gnome.org/show_bug.cgi?id=631907)**
## Description
Created attachment 172118
Basic elements to create gStreamer flows
On the 'Artwork' page:
If you...## Submitted by John
**[Link to original bug (#631907)](https://bugzilla.gnome.org/show_bug.cgi?id=631907)**
## Description
Created attachment 172118
Basic elements to create gStreamer flows
On the 'Artwork' page:
If you have additional artwork, send it to us...
No e-mail link. In fact, no e-mail link on the entire site, as far
as I can detect. I understand that, but you'll have to forgive me
using the bugzilla interface to send this file.
I suggest:
1) adding some way to communicate
2) I liked the 'block flow diagrams' as used in some documents
to explain the flow of the signals in gstreamer apps. I couldn't
find any reference as to how they were done, so I created
the basic elements in Inkscape so they can be duplicated.
Attached is the 'template' file.
When used with grid snapping on, it is fairly easy to create
complex flows.
If there is an easier way to do this, do inform me.
John
**Attachment 172118**, "Basic elements to create gStreamer flows":
![gstreamer_art.svg](/uploads/c677bcbf6ca0d619f728df9a1fe3780a/gstreamer_art.svg)https://gitlab.freedesktop.org/gstreamer/gstreamer/-/issues/16bin: set_state to PLAYING of non-toplevel bin might stop at PAUSED2023-05-15T18:28:47ZBugzilla Migration Userbin: set_state to PLAYING of non-toplevel bin might stop at PAUSED## Submitted by Mark Nauwelaerts `@mnauw`
**[Link to original bug (#632782)](https://bugzilla.gnome.org/show_bug.cgi?id=632782)**
## Description
... particularly if NO_PREROLL element present in bin.
Since NO_PREROLL "eats" any...## Submitted by Mark Nauwelaerts `@mnauw`
**[Link to original bug (#632782)](https://bugzilla.gnome.org/show_bug.cgi?id=632782)**
## Description
... particularly if NO_PREROLL element present in bin.
Since NO_PREROLL "eats" any ASYNC, occurrence of the former tends to trigger a "fake async-done" to counter the latter, see e.g. gst_bin_add_func when adding a NO_PREROLL element, or decodebin2 change_state function forcing async_done upon NO_PREROLL state change of bin.
In any case, bin_handle_async_done is triggered, which (silently) ignores further state change to PLAYING assuming that (some) parent will take care of it later on. While that is the case in "normal" state changes, it need not be so in more "advanced/dynamic" custom bin scenarios (decodebin2, etc etc), depending on whether parents already reached PLAYING earlier or other ASYNC_START are pending somewhere else. Specific cases are in e.g. [bug 628214](https://bugzilla.gnome.org/show_bug.cgi?id=628214) or [bug 632656](https://bugzilla.gnome.org/show_bug.cgi?id=632656).
As such, this may be intended behaviour, but it might need some tweaking or some warning/documentation indicating that a good old _set_state (..., PLAYING) needs some caution as it may "fail" silently and would have to be replaced by _set_state (..., PAUSED) followed by _set_state (..., PLAYING).https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/issues/36[flacenc] does not support changing the number of channels on the fly2021-09-24T13:29:55ZBugzilla Migration User[flacenc] does not support changing the number of channels on the fly## Submitted by Julien Isorce `@cap`
**[Link to original bug (#632798)](https://bugzilla.gnome.org/show_bug.cgi?id=632798)**
## Description
Created attachment 172941
aac file that contains 2 audio channels then 6 then 2 then 6 (re...## Submitted by Julien Isorce `@cap`
**[Link to original bug (#632798)](https://bugzilla.gnome.org/show_bug.cgi?id=632798)**
## Description
Created attachment 172941
aac file that contains 2 audio channels then 6 then 2 then 6 (recorded with faac outputformat=1)
** steps to reproduce:
gst-launch-0.10 filesrc location=res.aac ! aacparse ! faad ! flacenc ! fakesink
** Actual result:
"WARNING: flac already initialized -- fixme allow this"
** Excepted result:
Does the 'fixme allow this' mean that it would be possible to call 'gst_flac_enc_sink_setcaps' if encoder is already initialized ? flac specs ?
**Attachment 172941**, "aac file that contains 2 audio channels then 6 then 2 then 6 (recorded with faac outputformat=1)":
[res.aac](/uploads/307793f0af9adcab9da1fd42fae88db4/res.aac)https://gitlab.freedesktop.org/gstreamer/gst-plugins-base/-/issues/40streamsynchronizer: regression: hang when pausing near EOS2019-03-15T21:27:35ZBugzilla Migration Userstreamsynchronizer: regression: hang when pausing near EOS## Submitted by Mark Nauwelaerts `@mnauw`
**[Link to original bug (#633700)](https://bugzilla.gnome.org/show_bug.cgi?id=633700)**
## Description
That is, play (audio and video) until about eos, pause and then change to playing again...## Submitted by Mark Nauwelaerts `@mnauw`
**[Link to original bug (#633700)](https://bugzilla.gnome.org/show_bug.cgi?id=633700)**
## Description
That is, play (audio and video) until about eos, pause and then change to playing again, then pipeline might fail to preroll (and as such never reach playing again). Specifically, if one the streams has already reached eos, whereas the other has not, then streamsynchronizer will not forward eos. So, one of the sinks will preroll (with regular data), but the other never will (as it needs the eos that streamsynchronizer will not send since the other stream is blocked downstream in pause, assuming here any limited downstream queues do not suffice to compensate).
Not quite sure what to do about this; there are already some ugly streamsynchronizer hacks that send an empty buffer to trigger/force preroll (which probably won't be digested well by e.g. xvimagesink).
So, it seems that if we want to do the tricky/advanced stuff as in streamsynchronizer without breaking the basic stuff (and keep it more or less clean), we need some "force-preroll-event" (or something like that). The latter might also help in some other cases (e.g. handling empty edit in qtdemux). Or maybe that is too much 0.11 futuristic and there are other tricks or different ways to go about what streamsynchronizer/playsink/etc try to achieve ?
### Blocking
* [Bug 636313](https://bugzilla.gnome.org/show_bug.cgi?id=636313)https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/issues/27rsndvd: avoid dvd playback freezing briefly after un-pausing while disc spins...2021-09-24T14:31:49ZBugzilla Migration Userrsndvd: avoid dvd playback freezing briefly after un-pausing while disc spins up again, add more buffering in playbin## Submitted by Mehmet Giritli
**[Link to original bug (#633894)](https://bugzilla.gnome.org/show_bug.cgi?id=633894)**
## Description
If you are playing a dvd with totem via the resin-dvd plugin and you pause the movie for a while, ...## Submitted by Mehmet Giritli
**[Link to original bug (#633894)](https://bugzilla.gnome.org/show_bug.cgi?id=633894)**
## Description
If you are playing a dvd with totem via the resin-dvd plugin and you pause the movie for a while, you will get a brief freeze shortly after you resume the movie until the disc starts to spin-up again. I think it is a simple cache issue. This doesn't happen with windows media player for instance and perhaps this can be easily improved.https://gitlab.freedesktop.org/gstreamer/gst-plugins-base/-/issues/41xvimagesink: event handling fixes2021-09-24T13:20:12ZBugzilla Migration Userxvimagesink: event handling fixes## Submitted by Farkas Levente `@lfarkas`
**[Link to original bug (#634431)](https://bugzilla.gnome.org/show_bug.cgi?id=634431)**
## Description
(by default) on linux if we use xvimagesink as an overlay component it's does not propa...## Submitted by Farkas Levente `@lfarkas`
**[Link to original bug (#634431)](https://bugzilla.gnome.org/show_bug.cgi?id=634431)**
## Description
(by default) on linux if we use xvimagesink as an overlay component it's does not propagate mouse move, enter and leave events (but automatically redraw itself as needed). if we set xvimagesink's "handle-events" properties to false then we got mouse move, but still can't get mouse enter and mouse leave. but in this case (ie. "handle-events" false) xvimagesink don't handle expose events either (even if "handle-expose" still set to true).
in gstreamer-java we capture all XEvents and propagate the expose and all mouse events by ourself, but imho it's a bug in the current xvimagesink implementation.
it'd be nice to clearly document what is the
handle-events
handle-expose
means and be able to somehow get our expected result. which is:
- got mouse move, enter, leave
- and also automatically redraw itself in case it's needed (ie: ExposureMask | VisibilityChangeMask | StructureNotifyMask | FocusChangeMask)
ps. anyway on windows's videosinks (all) everything working as expected.https://gitlab.freedesktop.org/gstreamer/gstreamer/-/issues/17RFC: operation hints for GstElement2023-05-15T18:28:47ZBugzilla Migration UserRFC: operation hints for GstElement## Submitted by Stefan Kost `@ensonic`
**[Link to original bug (#634687)](https://bugzilla.gnome.org/show_bug.cgi?id=634687)**
## Description
[Bug 564749](https://bugzilla.gnome.org/show_bug.cgi?id=564749) mostly blocks on the GstTa...## Submitted by Stefan Kost `@ensonic`
**[Link to original bug (#634687)](https://bugzilla.gnome.org/show_bug.cgi?id=634687)**
## Description
[Bug 564749](https://bugzilla.gnome.org/show_bug.cgi?id=564749) mostly blocks on the GstTagReaderIface. An alternative would be to have a set of hint-flags on GstElement. Those flags can be set on the pipeline, individual bins and even elements. Bins could propagate the hints to new children and hint-changes to existing children.
The purpose of the hints is to specify the intended use case a bit more to give elements a chances for optimization. Right now gstreamer elements prepare to support everything.
Some use cases
1) quickly play something from start to end
- no need for metadata
- no seeking/no playback rate changes
- no mucking with the pipeline at all until eos
2) quick metadata reading
- no prerolling
- no seektable building
3) no caps changes (video in fullscreen)
- quicker pad_alloc or even no need to check for caps changes in basetransform
No proposal for a set of hints yet :/
### Blocking
* [Bug 532307](https://bugzilla.gnome.org/show_bug.cgi?id=532307)
* [Bug 564749](https://bugzilla.gnome.org/show_bug.cgi?id=564749)https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/issues/28opencv: Add cvcalchist element for calculating histogram2021-09-24T14:31:49ZBugzilla Migration Useropencv: Add cvcalchist element for calculating histogram## Submitted by Josh Doe
**[Link to original bug (#635646)](https://bugzilla.gnome.org/show_bug.cgi?id=635646)**
## Description
Created attachment 175134
Patch to add cvcalchist
Added the element cvcalchist which wraps the Op...## Submitted by Josh Doe
**[Link to original bug (#635646)](https://bugzilla.gnome.org/show_bug.cgi?id=635646)**
## Description
Created attachment 175134
Patch to add cvcalchist
Added the element cvcalchist which wraps the OpenCV function cvCalcHist. Only
8- and 16-bit monochrome video is supported at this time. By default the
element will not alter the data, but if "draw-histogram" is true, then a crude
histogram will be drawn over the image.
I've included a test program under tests/examples/opencv that draws a text histogram to the console using the "histogram" message.
Please review the patch, as I'm sure there are some issues. I wasn't sure of a nice way to handle RGB histograms, especially in terms of how to package the data in a message, so I left that alone for the time being. Also note I'm developed this under Windows, but hopefully I've modified the Makefiles appropriately.
~~**Patch 175134**~~, "Patch to add cvcalchist":
[0001-opencv-Add-cvcalchist-element-for-calculating-histog.patch](/uploads/dfab7ae226854d65adc86ca01167b97a/0001-opencv-Add-cvcalchist-element-for-calculating-histog.patch)https://gitlab.freedesktop.org/gstreamer/gst-devtools/-/issues/1gst-introspection tool [review]2023-11-09T18:16:33ZBugzilla Migration Usergst-introspection tool [review]## Submitted by Luis de Bethencourt
**[Link to original bug (#635860)](https://bugzilla.gnome.org/show_bug.cgi?id=635860)**
## Description
gst-gengui is an utility for testing and controlling live GStreamer pipelines and elements. ...## Submitted by Luis de Bethencourt
**[Link to original bug (#635860)](https://bugzilla.gnome.org/show_bug.cgi?id=635860)**
## Description
gst-gengui is an utility for testing and controlling live GStreamer pipelines and elements.
It will inspect the specified pipeline to create the GTK GUI automagically, based on the value type of properties.
A list of properties for each element will be displayed. Then you can search for/adjust any property value and see the outcome immediately.
Originally developed by Florent Thiery, I've fixed a few bugs and made it non-dependant of external code.
The code is available for review at:
https://github.com/luisbg/gst-introspectionhttps://gitlab.freedesktop.org/gstreamer/gst-editing-services/-/issues/2PiTiVi core does not handle playing audio clips properly at non-native speeds.2021-09-24T12:20:46ZBugzilla Migration UserPiTiVi core does not handle playing audio clips properly at non-native speeds.## Submitted by Brandon Lewis
**[Link to original bug (#636235)](https://bugzilla.gnome.org/show_bug.cgi?id=636235)**
## Description
In theory the only thing required to cause a clip to play back at a speed greater than or less than...## Submitted by Brandon Lewis
**[Link to original bug (#636235)](https://bugzilla.gnome.org/show_bug.cgi?id=636235)**
## Description
In theory the only thing required to cause a clip to play back at a speed greater than or less than it's native rate is to set the media duration to something other than the duration.
To test this I created a pitivi project file (attached) with a single clip and then hand-edited the media duration of the clip such that it should play at 50% speed. While the video portion of the clip does seem to slow down, the audio continues to play back at its normal rate. There are plenty of other issues besides, but the playback issue mystifies me because a simple test script (also attached) is actually able to adjust playback speed using this method.
### Blocking
* [Bug 593828](https://bugzilla.gnome.org/show_bug.cgi?id=593828)