gst-plugins-bad issueshttps://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/issues2019-02-15T14:31:24Zhttps://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/issues/121wayland: add a westonsrc element2019-02-15T14:31:24ZBugzilla Migration Userwayland: add a westonsrc element## Submitted by Sebastian Wick
**[Link to original bug (#719853)](https://bugzilla.gnome.org/show_bug.cgi?id=719853)**
## Description
Created attachment 263519
patch
add a waylandsrc element to the wayland plugin
**Pat...## Submitted by Sebastian Wick
**[Link to original bug (#719853)](https://bugzilla.gnome.org/show_bug.cgi?id=719853)**
## Description
Created attachment 263519
patch
add a waylandsrc element to the wayland plugin
**Patch 263519**, "patch":
[0001-add-waylandsrc-to-the-wayland-plugin.patch](/uploads/b29e000ea31dc136eceb0e71b485d2a2/0001-add-waylandsrc-to-the-wayland-plugin.patch)https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/issues/122dvdspu: gst_segment_to_running_time: assertion 'segment->format == format' fa...2023-07-11T11:01:06ZBugzilla Migration Userdvdspu: gst_segment_to_running_time: assertion 'segment->format == format' failed## Submitted by Frank Ansari
**[Link to original bug (#719903)](https://bugzilla.gnome.org/show_bug.cgi?id=719903)**
## Description
What I want to achieve: DVD playback with totem media player.
I get this error message when I r...## Submitted by Frank Ansari
**[Link to original bug (#719903)](https://bugzilla.gnome.org/show_bug.cgi?id=719903)**
## Description
What I want to achieve: DVD playback with totem media player.
I get this error message when I run
gst-launch-1.0 playbin uri=dvd://
with "Desperate Housewives" (season 4, part 2, episodes 11-14).
gst_segment_to_running_time: assertion 'segment->format == format' failed
When I insert seasion 4, part 2, episodes 15-17 I get a different error message:
ERROR: from element /GstPlayBin:playbin0/GstPlaySink:playsink/GstBin:abin/GstAutoAudioSink:audiosink/GstPulseSink:audiosink-actual-sink-pulse: The stream is in the wrong format.
Additional debug info:
gstaudiobasesink.c(1972): gst_audio_base_sink_render (): /GstPlayBin:playbin0/GstPlaySink:playsink/GstBin:abin/GstAutoAudioSink:audiosink/GstPulseSink:audiosink-actual-sink-pulse:
sink not negotiated.
Both DVDs work fine using Fluendo DVD player.
What is going on?
I have a fresh Arch Linux instalation with these gstreamer compoments:
local/clutter-gst 2.0.8-1
GStreamer bindings for clutter
local/gnome-video-effects 0.4.0-2
A collection of GStreamer effects
local/gst-libav 1.2.1-1
Gstreamer libav Plugin
local/gst-plugins-bad 1.2.1-1
GStreamer Multimedia Framework Bad Plugins
local/gst-plugins-base 1.2.1-1
GStreamer Multimedia Framework Base Plugins
local/gst-plugins-base-libs 1.2.1-1
GStreamer Multimedia Framework Base Plugin libraries
local/gst-plugins-good 1.2.1-1
GStreamer Multimedia Framework Good Plugins
local/gst-plugins-ugly 1.2.1-2
GStreamer Multimedia Framework Ugly Plugins
local/gstreamer 1.2.1-1
GStreamer Multimedia Framework
local/gstreamer0.10 0.10.36-2
GStreamer Multimedia Framework
local/gstreamer0.10-bad 0.10.23-5
GStreamer Multimedia Framework Bad Plugin libraries (gst-plugins-bad)
local/gstreamer0.10-bad-plugins 0.10.23-5 (gstreamer0.10-plugins)
GStreamer Multimedia Framework Bad Plugins (gst-plugins-bad)
local/gstreamer0.10-base 0.10.36-1
GStreamer Multimedia Framework Base plugin libraries
local/gstreamer0.10-base-plugins 0.10.36-1 (gstreamer0.10-plugins)
GStreamer Multimedia Framework Base Plugins (gst-plugins-base)
local/gstreamer0.10-ffmpeg 0.10.13-1 (gstreamer0.10-plugins)
Gstreamer FFMpeg Plugin
local/gstreamer0.10-good 0.10.31-3
GStreamer Multimedia Framework Good plugin libraries
local/gstreamer0.10-ugly 0.10.19-7
GStreamer Multimedia Framework Ugly plugin libraries
local/gstreamer0.10-ugly-plugins 0.10.19-7 (gstreamer0.10-plugins)
GStreamer Multimedia Framework Ugly Plugins (gst-plugins-ugly)
local/gstreamer0.10-vaapi 0.5.7-1
GStreamer Multimedia Framework VA Plugins
local/phonon-gstreamer 4.7.0-2
Phonon GStreamer backend
local/totem 3.10.1-1 (gnome)
GNOME3 movie player based on GStreamer
Version: 1.2.1
### Depends on
* [Bug 695606](https://bugzilla.gnome.org/show_bug.cgi?id=695606)https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/issues/123game-music-emu files with multiple songs only play first song2021-09-24T14:32:18ZBugzilla Migration Usergame-music-emu files with multiple songs only play first song## Submitted by mih..@..il.com
**[Link to original bug (#720687)](https://bugzilla.gnome.org/show_bug.cgi?id=720687)**
## Description
Hi guys! The GBS format supports multiple songs (from the Game Boy) in one .gbs file. See [1] for ...## Submitted by mih..@..il.com
**[Link to original bug (#720687)](https://bugzilla.gnome.org/show_bug.cgi?id=720687)**
## Description
Hi guys! The GBS format supports multiple songs (from the Game Boy) in one .gbs file. See [1] for an example file. I've checked the git code [2], and even though the game-music-emu library supports playback and querying info about multiple songs in one file (via gme_track_count() and gme_start_track()), track 0 (first track of file) is hardcoded, thus ignoring any other tracks in the same file.
I would write the patch myself, but I don't know the GStreamer API well enough, I'm only familiar with the game-music-emu API. I've seen an article [3] about streamids, so I know this should be possible. If someone can hint to where I can look in the code, maybe a similar plugin that supports multiple tracks, or the API headers, I can make a patch.
[1] http://www.zophar.net/music/gbs/castlevania-2-belmont-s-revenge.html
[2] http://cgit.freedesktop.org/gstreamer/gst-plugins-bad/tree/ext/gme/gstgme.c
[3] http://blogs.gnome.org/uraeus/2012/11/25/improved-handling-of-files-with-multiple-tracks-in-gstreamer/https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/issues/124Unable to play animal_planet.mpg video stream2021-09-24T14:32:19ZBugzilla Migration UserUnable to play animal_planet.mpg video stream## Submitted by Jan ONDREJ (SAL)
**[Link to original bug (#721621)](https://bugzilla.gnome.org/show_bug.cgi?id=721621)**
## Description
This stream can't be played with gstreamer:
http://work.salstar.sk/public/gstreamer/animal_...## Submitted by Jan ONDREJ (SAL)
**[Link to original bug (#721621)](https://bugzilla.gnome.org/show_bug.cgi?id=721621)**
## Description
This stream can't be played with gstreamer:
http://work.salstar.sk/public/gstreamer/animal_planet.mpg
Response from Tim@IRC:
<__tim:#gstreamer> the primary bug is that it doesn't play at all, the
secondary issue is the detection of that lpcm or whatever stream
Steps to reproduce:
gst-launch-1.0 playbin uri=http://work.salstar.sk/public/gstreamer/animal_planet.mpg
Version: 1.2.2https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/issues/125waylandsink: fix surface width/height when caps have changed2021-09-24T14:32:19ZBugzilla Migration Userwaylandsink: fix surface width/height when caps have changed## Submitted by Benjamin Gaignard
**[Link to original bug (#722343)](https://bugzilla.gnome.org/show_bug.cgi?id=722343)**
## Description
Created attachment 266474
waylandsink: fix surface width/height when caps have changed
I...## Submitted by Benjamin Gaignard
**[Link to original bug (#722343)](https://bugzilla.gnome.org/show_bug.cgi?id=722343)**
## Description
Created attachment 266474
waylandsink: fix surface width/height when caps have changed
If a player (like gst-play) only change the state between two files
waylandsink need to be able to resize the surface output according
to the new width/height values.
I have tested this patch with a command line similar to this one:
gst-play --audiosink fakesink --videosink waylandsink FILE1 FILE2
~~**Patch 266474**~~, "waylandsink: fix surface width/height when caps have changed":
[0001-waylandsink-fix-surface-width-height-when-caps-have-.patch](/uploads/33e502321aa973a55da4db2e0b3a5796/0001-waylandsink-fix-surface-width-height-when-caps-have-.patch)
Version: 1.xhttps://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/issues/126tq: tee element with embedded queue elements on srcpads2021-09-24T14:32:20ZBugzilla Migration Usertq: tee element with embedded queue elements on srcpads## Submitted by Andrey Utkin
**[Link to original bug (#722511)](https://bugzilla.gnome.org/show_bug.cgi?id=722511)**
## Description
Created attachment 266633
patch
This was discussed in gstreamer-devel maillist a bit, and i w...## Submitted by Andrey Utkin
**[Link to original bug (#722511)](https://bugzilla.gnome.org/show_bug.cgi?id=722511)**
## Description
Created attachment 266633
patch
This was discussed in gstreamer-devel maillist a bit, and i was encouraged to post this as a patch to -plugins-bad.
Besides inclusion into upstream, what interests me is code review, so i figure out if used things are operated correctly. Any comments appreciated.
The code currently lacks management of inside tee and queues properties. I haven't decided yet how exactly they would be passed through, possibly as two properties "tee-props" and "queue-props" as strings representing a set of key-value pairs.
P. S. How xml doc is generated?
**Patch 266633**, "patch":
[0001-Introduce-tq-tee-element-with-embedded-queue-element.patch](/uploads/97957fa1cc2a4633096230a9d03b18c5/0001-Introduce-tq-tee-element-with-embedded-queue-element.patch)https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/issues/127New umxparse element2021-09-24T14:32:21ZBugzilla Migration UserNew umxparse element## Submitted by Carlos Rafael Giani
**[Link to original bug (#722618)](https://bugzilla.gnome.org/show_bug.cgi?id=722618)**
## Description
Created attachment 266761
Adds umxparse element
Unreal Engine 1 based games such as Un...## Submitted by Carlos Rafael Giani
**[Link to original bug (#722618)](https://bugzilla.gnome.org/show_bug.cgi?id=722618)**
## Description
Created attachment 266761
Adds umxparse element
Unreal Engine 1 based games such as Unreal 1, Unreal Tournament 1, and Deus Ex 1 use module music (like MOD,S3M,XM,IT..) contained within Unreal packages. Such files are called UMX files. These Unreal packages contain only one module song, and nothing more.
This new element parses the UMX data contents, extracts the module, and sends it downstream. Module players such as modplug can then play the data as if it came from an ordinary .s3m or .xm file.
**Patch 266761**, "Adds umxparse element":
[0001-umxparse-add-new-parser-element-for-module-music-in-.patch](/uploads/5b9d01e0c02b0ffbe7917b64626f6625/0001-umxparse-add-new-parser-element-for-module-music-in-.patch)https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/issues/128intervideosrc/intervideosink: cannot change channel while in playing state2022-11-17T14:54:23ZBugzilla Migration Userintervideosrc/intervideosink: cannot change channel while in playing state## Submitted by lou..@..il.com
**[Link to original bug (#722877)](https://bugzilla.gnome.org/show_bug.cgi?id=722877)**
## Description
Once the intervideosrc and intervideosink elements are in the playing state, changing the channel ...## Submitted by lou..@..il.com
**[Link to original bug (#722877)](https://bugzilla.gnome.org/show_bug.cgi?id=722877)**
## Description
Once the intervideosrc and intervideosink elements are in the playing state, changing the channel property does not have any effect on their behavior. The elements must be moved to null and then back to playing for the channel behavior to change.
Version: 1.2.0https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/issues/129tsdemux: "No program activated before EOS" with .ts files generated by a Xcru...2021-09-24T14:32:22ZBugzilla Migration Usertsdemux: "No program activated before EOS" with .ts files generated by a Xcruiser PVR## Submitted by Bill
**[Link to original bug (#723142)](https://bugzilla.gnome.org/show_bug.cgi?id=723142)**
## Description
I'm just trying out the new version of Pitivi that appeared in the fedora repos today (0.92). I tried import...## Submitted by Bill
**[Link to original bug (#723142)](https://bugzilla.gnome.org/show_bug.cgi?id=723142)**
## Description
I'm just trying out the new version of Pitivi that appeared in the fedora repos today (0.92). I tried importing a .ts file. I believe this is encoded using 4cc mpeg & audio mp2 (At least that is what Avidemux claims it is - handbrake just says its mpeg2 video). So avidemux & handbrake recognise it but Pitivi does not. Pitivi displays a music icon and then crashes when I try to import it. Current workaround is to transcode the file to a different format using avidemux or handbrake. Then I can import it ok into Pitivi.
The source of the file is a direct recording from an Xcruiser satellite receiver onto an external USB disk. As far as I know the satellite receiver doesn't transcode the recording just basically stores whatever was transmitted by the satellite.
So is there any way to get Pitivi to directly import this type of file. Maybe I need to install some other package or plugin?https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/issues/130mpeg4videoparse: push caps before segment when first data is a GAP2021-09-24T14:32:23ZBugzilla Migration Usermpeg4videoparse: push caps before segment when first data is a GAP## Submitted by Thiago Sousa Santos `@thiagossantos`
**[Link to original bug (#723646)](https://bugzilla.gnome.org/show_bug.cgi?id=723646)**
## Description
While the attached patch fixes it, I'd rather fix this in the baseclass
if...## Submitted by Thiago Sousa Santos `@thiagossantos`
**[Link to original bug (#723646)](https://bugzilla.gnome.org/show_bug.cgi?id=723646)**
## Description
While the attached patch fixes it, I'd rather fix this in the baseclass
if possible because it would not require fixing it in all parsers.
Unfortunately I found no sane way of doing it with the current baseclass
methods. And if we have to add a new method, better just use the
event handler just like this patch? Any ideas other than doing
a check for a src pad caps before pushing a segment?
Another option is to store the GAP as a pending event while caps
aren't decided, but then a GAP can be very long and disrupt playback/preroll
if not pushed. So not a good one.https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/issues/131new fakevideodec element2019-12-12T23:52:22ZBugzilla Migration Usernew fakevideodec element## Submitted by Julien Isorce `@cap`
**[Link to original bug (#723778)](https://bugzilla.gnome.org/show_bug.cgi?id=723778)**
## Description
I have started a fakevideodec element here http://cgit.collabora.com/git/user/julien/gst-plu...## Submitted by Julien Isorce `@cap`
**[Link to original bug (#723778)](https://bugzilla.gnome.org/show_bug.cgi?id=723778)**
## Description
I have started a fakevideodec element here http://cgit.collabora.com/git/user/julien/gst-plugins-good.git/commit/?h=fakevideodec&id=7a4c1ce6e3ecaa35ffbd74b7e37968e3adc70485
It's useful when you have a new embedded platform and you want to know what would be the performance if you had a decoder that use 0% CPU. (hardware decoder except the zero-copy part)
videotestsrc is not really usable on embedded like RPI, I mean it uses so much CPU so not really useful to identify what would be the best FPS.
Also fpsdisplaysink uses textoverlay for the visual fps information. When you want a visual information only (and not a console info)
Also fakevideodec is compatible with playbin as long as you increase its rank.
For now fakevideodec just draw a kind of snake on 1 line (to make it use the CPU the less possible)
And the snake moves from left to right in 1 sec if no drop. So that when there are frame dropping the snake freez and you see it jumps to other positions. At every new frame it clears the line and draw the next position base on the framerate and the resolution.
I made it quite quickly so it may not being exactly correct right now.
Also even if it currently visually gives an idea of the FPS, it still does not allow to determine visually what is the precise FPS (can't say 24 or 25 FPS)
It could be improved to draw a kind of minimal meter and then draw a spot to indicate what is the current framerate. There are several ideas.
Do no hesitate to put a comment :)https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/issues/132adaptivedemux: Handle users _eos event2021-09-24T14:32:25ZBugzilla Migration Useradaptivedemux: Handle users _eos event## Submitted by Thibault Saunier `@thiblahute`
**[Link to original bug (#723868)](https://bugzilla.gnome.org/show_bug.cgi?id=723868)**
## Description
Currently if the user sends an EOS to a pipeline with hlsdemux in it, it will basi...## Submitted by Thibault Saunier `@thiblahute`
**[Link to original bug (#723868)](https://bugzilla.gnome.org/show_bug.cgi?id=723868)**
## Description
Currently if the user sends an EOS to a pipeline with hlsdemux in it, it will basically be ignored, we should manage to detect when an EOS as been sent by the user itself, and properly handle it.https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/issues/133h264parse: interlaced detection unreliable on rtsp streams (french TV, free.f...2021-09-24T14:32:25ZBugzilla Migration Userh264parse: interlaced detection unreliable on rtsp streams (french TV, free.fr multiposte)## Submitted by Yannick
**[Link to original bug (#724714)](https://bugzilla.gnome.org/show_bug.cgi?id=724714)**
## Description
Hi,
I'm using totem on Fedora 20 to watch french TV. This service is provided by my ISP using rtsp. ...## Submitted by Yannick
**[Link to original bug (#724714)](https://bugzilla.gnome.org/show_bug.cgi?id=724714)**
## Description
Hi,
I'm using totem on Fedora 20 to watch french TV. This service is provided by my ISP using rtsp.
The auto-deinterlace feature is unreliable : sometimes it works, sometimes it fails. IMHO those streams are always interlaced.
Here is an exemple :
http://sevmek.free.fr/gstreamer/rtspvideo2.gdphttps://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/issues/134hlsdemux: Invalid timestamps generated for live streams and otherwise broken ...2023-05-16T16:34:21ZBugzilla Migration Userhlsdemux: Invalid timestamps generated for live streams and otherwise broken for live streams## Submitted by Sebastian Dröge `@slomo`
**[Link to original bug (#724983)](https://bugzilla.gnome.org/show_bug.cgi?id=724983)**
## Description
a) Currently hlsdemux generates timestamps by summing all the fragment durations in the ...## Submitted by Sebastian Dröge `@slomo`
**[Link to original bug (#724983)](https://bugzilla.gnome.org/show_bug.cgi?id=724983)**
## Description
a) Currently hlsdemux generates timestamps by summing all the fragment durations in the playlist. Only that this will lead to invalid timestamps because the playlists are like a ringbuffer, only having a few fragments and not all since the beginning of time. We could keep all fragments around forever or try to assign a timestamp to the beginning of the playlist by looking at the sequence numbers of the fragments, but this is going to fail once we e.g. switch from one bitrate to another. Also it will fail if after a playlist update we have no old fragments around... we can't know how much gap there was in between.
b) If we pause the pipeline, we will lack behind... and can easily get to a point where all fragments have disappeared already. We need to resync then, and probably always resync whenever we unpause.
c) Due to us and the sender having different clocks, we will run out of sync at some point. Either being in the future for the sender (downloading fragments that don't exist yet) or too far in the past (downloading fragments that disappeared already). We need some resync logic for that. Probably rather unlikely case as it requires a lot of clock drift and everybody using HLS for such long-term live streaming doesn't deserve something else ;)
a) and b) definitely need to be fixed, c) would be bonus if someone is bored :)https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/issues/135Add new multiappsrc or dynappsrc element with multiple output streams2021-09-24T14:32:26ZBugzilla Migration UserAdd new multiappsrc or dynappsrc element with multiple output streams## Submitted by Justin Kim `@joykim`
**[Link to original bug (#725187)](https://bugzilla.gnome.org/show_bug.cgi?id=725187)**
## Description
if Audio/Video streams are separately given, playbin isn't able to handle two sources.
Thi...## Submitted by Justin Kim `@joykim`
**[Link to original bug (#725187)](https://bugzilla.gnome.org/show_bug.cgi?id=725187)**
## Description
if Audio/Video streams are separately given, playbin isn't able to handle two sources.
This is a suggestion about how to deploy multiple appsrcs and to use with playbin.
The basic idea started from here; http://lists.freedesktop.org/archives/gstreamer-devel/2014-January/045771.html
Simple documentation is included in codes.Seungha Yangseungha@centricular.comSeungha Yangseungha@centricular.comhttps://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/issues/136dvdspu: handle video/x-raw(ANY) if downstream supports the GstVideoOverlayCom...2023-12-20T15:04:25ZBugzilla Migration Userdvdspu: handle video/x-raw(ANY) if downstream supports the GstVideoOverlayCompositionMeta API## Submitted by Matthieu Bouron
**[Link to original bug (#725900)](https://bugzilla.gnome.org/show_bug.cgi?id=725900)**
## Description
This patch makes the dvdspu element handle video/x-raw(ANY) if downstream
supports the GstVideo...## Submitted by Matthieu Bouron
**[Link to original bug (#725900)](https://bugzilla.gnome.org/show_bug.cgi?id=725900)**
## Description
This patch makes the dvdspu element handle video/x-raw(ANY) if downstream
supports the GstVideoOverlayCompositionMeta API since in this case it only
places the correct meta on outgoing buffers.
This patch in based on the on-going work on the GstVideoOverlayCompositionMeta API support in the dvdspu element. See: https://bugzilla.gnome.org/show_bug.cgi?id=685282.
It also depends on https://bugzilla.gnome.org/show_bug.cgi?id=725893.
Development branch can be found here: http://cgit.collabora.com/git/user/mateo/gst-plugins-bad.git/log/?h=dvdspu
### Depends on
* [Bug 685282](https://bugzilla.gnome.org/show_bug.cgi?id=685282)
* [Bug 725893](https://bugzilla.gnome.org/show_bug.cgi?id=725893)https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/issues/137atdec: Fails to decode multichannel streams2021-09-24T14:32:27ZBugzilla Migration Useratdec: Fails to decode multichannel streams## Submitted by Andoni Alastruey `@ylatuya`
**[Link to original bug (#727757)](https://bugzilla.gnome.org/show_bug.cgi?id=727757)**
## Description
Fails to decode http://mirror.bigbuckbunny.de/peach/bigbuckbunny_movies/big_buck_bunn...## Submitted by Andoni Alastruey `@ylatuya`
**[Link to original bug (#727757)](https://bugzilla.gnome.org/show_bug.cgi?id=727757)**
## Description
Fails to decode http://mirror.bigbuckbunny.de/peach/bigbuckbunny_movies/big_buck_bunny_720p_h264.mov
Marked as a regression since it's the default decoder in OS X
iMac:Commands$ LC_ALL=C GST_DEBUG=*atdec*:5 ./gst-launch-1.0 playbin uri=file:///Users/fluendo/Downloads/big_buck_bunny_720p_h264.mov
Setting pipeline to PAUSED ...
Pipeline is PREROLLING ...
Redistribute latency...
0:00:00.189682000 54717 0x101079320 DEBUG atdec atdec.c:181:gst_atdec_start:`<atdec0>` start
0:00:00.189928000 54717 0x101079320 DEBUG atdec atdec.c:271:gst_atdec_set_format:`<atdec0>` set_format
0:00:00.193547000 54717 0x101079320 ERROR default audio-channels.c:355:gst_audio_channel_positions_from_mask: Invalid channel mask 0x0000000000000003 for 6 channels
0:00:00.193624000 54717 0x101079320 ERROR default audio-info.c:286:gst_audio_info_from_caps: Invalid channel mask 0x0000000000000003 for 6 channels
2014-04-07 16:36:49.133 gst-launch-1.0[54717:1403] 16:36:49.133 ERROR: [0x102195000] >aqme> MESubmixGraph.h:218: SetMixerChannelLayout: (0x8d5bd060): scope 2, element 0, tag=0x650002: error -10851
0:00:00.199654000 54717 0x101079320 WARN atdec atdec.c:331:gst_atdec_set_format:`<atdec0>` error: AudioQueueSetOfflineRenderFormat returned error: 560226676
ERROR: from element /GstPlayBin:playbin0/GstURIDecodeBin:uridecodebin0/GstDecodeBin:decodebin0/GstATDec:atdec0: The stream is in the wrong format.
Additional debug info:
atdec.c(331): gst_atdec_set_format (): /GstPlayBin:playbin0/GstURIDecodeBin:uridecodebin0/GstDecodeBin:decodebin0/GstATDec:atdec0:
AudioQueueSetOfflineRenderFormat returned error: 560226676
ERROR: pipeline doesn't want to preroll.
Setting pipeline to NULL ...
Freeing pipeline ...https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/issues/138adpcmdec: WAV IMA ADPCM files don't get decoded correctly2021-09-24T14:32:27ZBugzilla Migration Useradpcmdec: WAV IMA ADPCM files don't get decoded correctly## Submitted by tia..@..il.com
**[Link to original bug (#727975)](https://bugzilla.gnome.org/show_bug.cgi?id=727975)**
## Description
Created attachment 274009
Audio file that can reproduce the problem
The problem has been de...## Submitted by tia..@..il.com
**[Link to original bug (#727975)](https://bugzilla.gnome.org/show_bug.cgi?id=727975)**
## Description
Created attachment 274009
Audio file that can reproduce the problem
The problem has been described in detail on the mailing list:
http://lists.freedesktop.org/archives/gstreamer-devel/2014-April/047322.html
Basically if i try to play a IMA ADPCM WAV file, the playback gets a lot of gaps on it and only the half of the duration of the file is actually played.
Also, querying the duration of the audio in time gets in result half of the actual duration.
I'm attaching a example file that can be used to reproduce the problem, just try to play it with adpcmdec and it is clear that it is not working properly:
gst-launch-1.0 filesrc location=test-adpcm.wav ! wavparse ! adpcmdec ! audioconvert ! pulsesink
The file seems to be ok and valid, and plays correctly using "play".
file test-adpcm.wav
test-adpcm.wav: RIFF (little-endian) data, WAVE audio, IMA ADPCM, mono 8000 Hz
**Attachment 274009**, "Audio file that can reproduce the problem":
[test-adpcm.wav](/uploads/fe314095c9c03e0b2016beea49663313/test-adpcm.wav)
Version: 1.2.1https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/issues/139h264parse: wait for resolution to be known before pushing2021-09-24T14:32:28ZBugzilla Migration Userh264parse: wait for resolution to be known before pushing## Submitted by Vincent Penquerc'h `@vincent`
**[Link to original bug (#728324)](https://bugzilla.gnome.org/show_bug.cgi?id=728324)**
## Description
Created attachment 274433
wait to get resolution before pushing buffers
This...## Submitted by Vincent Penquerc'h `@vincent`
**[Link to original bug (#728324)](https://bugzilla.gnome.org/show_bug.cgi?id=728324)**
## Description
Created attachment 274433
wait to get resolution before pushing buffers
This is something done a few years back for 0.10, and which may be useful to upstream.
**Patch 274433**, "wait to get resolution before pushing buffers":
[0002-h264parse-wait-to-get-resolution-before-pushing-buff.patch](/uploads/ef800ae195c0948eb9c265febf32a7b4/0002-h264parse-wait-to-get-resolution-before-pushing-buff.patch)https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/issues/140jpegparse: Does not passthrough timestamps depending on downstream elements2022-05-20T11:58:04ZBugzilla Migration Userjpegparse: Does not passthrough timestamps depending on downstream elements## Submitted by Nicola `@drakkan`
**[Link to original bug (#728356)](https://bugzilla.gnome.org/show_bug.cgi?id=728356)**
## Description
please test this pipeline:
gst-launch-0.10 -v souphttpsrc location=http://213.251.201.196/...## Submitted by Nicola `@drakkan`
**[Link to original bug (#728356)](https://bugzilla.gnome.org/show_bug.cgi?id=728356)**
## Description
please test this pipeline:
gst-launch-0.10 -v souphttpsrc location=http://213.251.201.196/anony/mjpg.cgi do-timestamp=true ! jpegparse ! videorate ! image/jpeg,framerate=3/1 ! fakesink silent=false
you'll see:
/GstPipeline:pipeline0/GstFakeSink:fakesink0: last-message = "chain ******* (fakesink0:sink) (8060 bytes, timestamp: 0:00:00.333333333, duration: 0:00:00.333333333, offset: 1, offset_end: 2, flags: 1 ro ) 0x7f33d0025a30"
/GstPipeline:pipeline0/GstFakeSink:fakesink0: last-message = "chain ******* (fakesink0:sink) (8042 bytes, timestamp: 0:00:00.666666666, duration: 0:00:00.333333334, offset: 2, offset_end: 3, flags: 1 ro ) 0x7f33d0025ad0"
/GstPipeline:pipeline0/GstFakeSink:fakesink0: last-message = "chain ******* (fakesink0:sink) (8072 bytes, timestamp: 0:00:01.000000000, duration: 0:00:00.333333333, offset: 3, offset_end: 4, flags: 1 ro ) 0x7f33d0008f30
now the same in 1.0 (today git):
gst-launch-1.0 -vm souphttpsrc location=http://213.251.201.196/anony/mjpg.cgi do-timestamp=true ! jpegparse ! videorate ! image/jpeg,framerate=3/1 ! fakesink silent=false
no buffer arrive to fakesink the logs show these messages:
0:00:04.502834927 4274 0x1678850 WARN videorate gstvideorate.c:1109:gst_video_rate_transform_ip:`<videorate0>` Got buffer with GST_CLOCK_TIME_NONE timestamp, discarding it
0:00:04.510393982 4274 0x1678850 WARN videorate gstvideorate.c:1109:gst_video_rate_transform_ip:`<videorate0>` Got buffer with GST_CLOCK_TIME_NONE timestamp, discarding it