community/{gstreamer, *gst*}: security upgrade to 1.22.7
-
community/gst-plugins-ugly: fails test for armv7 and armhf -> reported to upstream
Details
INFO: autodetecting backend as ninja
INFO: calculating backend command to run: /usr/bin/ninja -C /builds/chereskata/aports/community/gst-plugins-ugly/src/gst-plugins-ugly-1.22.7/output
ninja: nothing to do
ninja: entering directory '/builds/chereskata/aports/community/gst-plugins-ugly/src/gst-plugins-ugly-1.22.7/output'
ninja: nothing to do
1/3 elements_amrnbenc OK 0.20s
2/3 elements_x264enc FAIL 0.23s exit status 6
>>> CK_DEFAULT_TIMEOUT=20 GST_PLUGIN_LOADING_WHITELIST=gstreamer:gst-plugins-base:gst-plugins-good:gst-plugins-ugly@/builds/chereskata/aports/community/gst-plugins-ugly/src/gst-plugins-ugly-1.22.7/output MALLOC_PERTURB_=186 GST_PLUGIN_SYSTEM_PATH_1_0='' GST_REGISTRY=/builds/chereskata/aports/community/gst-plugins-ugly/src/gst-plugins-ugly-1.22.7/output/tests/check/elements_x264enc.registry GST_PLUGIN_SCANNER_1_0=/usr/libexec/gstreamer-1.0/gst-plugin-scanner GST_PLUGIN_PATH_1_0=/builds/chereskata/aports/community/gst-plugins-ugly/src/gst-plugins-ugly-1.22.7/output:/usr/lib/gstreamer-1.0:/usr/lib/gstreamer-1.0 /builds/chereskata/aports/community/gst-plugins-ugly/src/gst-plugins-ugly-1.22.7/output/tests/check/elements_x264enc
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Running suite(s): x264enc
0%: Checks: 6, Failures: 0, Errors: 6
../tests/check/elements/x264enc.c:253:E:general:test_video_baseline:0: (after this point) Received signal 4 (Illegal instruction)
../tests/check/elements/x264enc.c:253:E:general:test_video_main:0: (after this point) Received signal 4 (Illegal instruction)
../tests/check/elements/x264enc.c:253:E:general:test_video_high:0: (after this point) Received signal 4 (Illegal instruction)
../tests/check/elements/x264enc.c:253:E:general:test_video_high10:0: (after this point) Received signal 4 (Illegal instruction)
../tests/check/elements/x264enc.c:253:E:general:test_video_high422:0: (after this point) Received signal 4 (Illegal instruction)
../tests/check/elements/x264enc.c:253:E:general:test_video_high444:0: (after this point) Received signal 4 (Illegal instruction)
Check suite x264enc ran in 0.029s (tests failed: 6)
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3/3 generic_states OK 0.23s
Summary of Failures:
2/3 elements_x264enc FAIL 0.23s exit status 6
Ok: 2
Expected Fail: 0
Fail: 1
Unexpected Pass: 0
Skipped: 0
Timeout: 0 3.18 is at 1.22.5 -> !55372 (merged)
https://gitlab.freedesktop.org/gstreamer/gstreamer/-/compare/1.22.6...1.22.7
Changelog
1.22.7
The seventh 1.22 bug-fix release (1.22.7) was released on 13 November
2023.
This release only contains bugfixes and security fixes and it should be
safe to update from 1.22.x.
Highlighted bugfixes in 1.22.7
- Security fixes for the MXF demuxer and AV1 codec parser
- glfilter: Memory leak fix for OpenGL filter elements
- d3d11videosink: Fix toggling between fullscreen and maximized, and
window switching in fullscreen mode
- DASH / HLS adaptive streaming fixes
- Decklink card device provider device name string handling fixes
- interaudiosrc: handle non-interleaved audio properly
- openh264: Fail gracefully if openh264 encoder/decoder creation fails
- rtspsrc: improved whitespace handling in response headers by certain
cameras
- v4l2codecs: avoid wrap-around after 1000000 frames; tiled formats
handling fixes
- video-scaler, audio-resampler: downgraded “Can’t find exact taps”
debug log messages
- wasapi2: Don’t use global volume control object
- Rust plugins: various improvements in aws, fmp4mux, hlssink3,
livesync, ndisrc, rtpav1depay, rsfilesink, s3sink, sccparse
- WebRTC: various webrtchttp, webrtcsrc, and webrtcsink improvements
and fixes
- Cerbero build tools: recognise Windows 11; restrict parallelism of
gst-plugins-rs build on small systems
- Packages: ca-certificates update; fix gio module loading and TLS
support on macOS
gstreamer
- debugutils: provide gst_debug_bin_to_dot_data() implementation even
if debug system is disabled
gst-plugins-base
- audioaggregator, audiomixer: Make access to the pad list thread-safe
while mixing
- basetextoverlay: Fix overlay never rendering again if width reaches
1px
- glfiter: Protect GstGLContext access
- glfilter: Only add parent meta if inbuf != outbuf
- macOS: fix huge memory leak with glfilter-based elements
- rtspconnection: Ignore trailing whitespace in rtsp headers
- video-scaler, audio-resampler: downgrade ‘can’t find exact taps’ to
debug
gst-plugins-good
- adaptivedemux2: Do not submit_transfer when cancelled
- adaptivedemux2: Call GTasks’s return functions for blocking tasks
- flacenc: Correctly handle up to 255 cue entries
- flvmux: set the src segment position as running time
- imagesequencesrc: fix deadlock when feeding the same image in a loop
- pngenc: output one frame only in snapshot mode and mark output
frames as I-frames
- qmlglsrc: sync on the streaming thread
- souphttpsrc: Chain up to finalize to fix memory leak
- wavparse: fix buffer leak with adtl tag
- v4l2codecs: Avoid QBUF/DQBUF struct timeval .tv_usec wrap-around at
frame 1000000
- v4l2codecs: Fix tiled formats stride conversion
gst-plugins-bad
- audiobuffersplit: disable max-silence-time if set to 0
- libde265: Do not decode the non 4:2:0 8 bits format
- codecparsers: av1: Clip max tile rows and cols values
- codecs: h265: Do not free slice header before using it
- d3d11converter: Fix 10/12bits planar output
- d3d11decoder: Fix crash on negotiate() when decoder is not
configured
- d3d11videosink: Fix toggling between fullscreen and maximized
- d3d11videosink: Fix window switching in case of fullscreen mode
- d3d11screencapturesrc: Fix mouse cursor blending
- decklink: Fix broken COM string conversion
- decklink: Decklink Device Provider wrongly parses SDK strings
- gstwayland: Don’t depend on wayland-protocols
- interaudiosrc: Add audio meta to buffers containing non-interleaved
samples
- kmssink: Add TIDSS auto-detection
- mfvideoencoder: Fix typo in template caps
- mxfdemux: Store GstMXFDemuxEssenceTrack in their own fixed
allocation
- nvcodec: fix bounds for auto-select GPU enumeration
- openh264: Fail gracefully if openh264 encoder/decoder creation fails
- tsmux: More cleanups
- tsmux: Fill padding packets with stuffing bytes
- v4l2codecs: Fix tiled formats stride conversion
- v4l2videodec: Correctly free caps to avoid memory leak
- wasapi2: Don’t use global volume control object
- wasapi2device: Ignore activation failed device
gst-plugins-ugly
- No changes
gst-plugins-rs
- aws: s3sink: Fix handling of special characters in key
- aws: s3: Properly percent-decode GstS3Url
- fmp4mux: Don’t overflow negative composition offset calculation
- fmp4mux: specify the fragment duration unit
- hlssink3: Use Path API for getting file name
- hlssink3: Use sprintf for segment name formatting
- hlssink3: Remove unused deps
- hlssink3: Don’t remove old files if max-files is zero
- hlssink3: Don’t remove uri from playlist if playlist-length is zero
- hlssink3: Various cleanup
- livesync: log new pending segments
- livesync: display jitter when waiting on clock
- livesync: Rename activatemode methods to *_activatemode
- livesync: Simplify start_src_task and src_loop
- livesync: Improve audio duration fixups
- livesync: Log a category error when we are missing the segment
- livesync: Clean up state handling
- livesync: Replace an if-let with match
- livesync: Move a notify closer to the interesting state change
- livesync: Move num_in counting to the src task
- livesync: Simplify num_duplicate counting
- livesync: Handle flags and late buffer patching after queueing
- livesync: Separate out_buffer duplicate status from GAP flag
- livesync: Use fallback_duration for audio repeat buffers as well
- livesync: example: Add identities single-segment=1
- livesync: Split fallback_duration into in_ and out_duration
- livesync: Keep existing buffer duration in some cases
- livesync: Remove the stop from outgoing segments
- ndisrc: Assume input with more than 8 raw audio channels is
unpositioned
- rtpav1depay: Skip unexpected leading fragments
- rtpav1depay: Don’t push stale temporal delimiters downstream
- rsfilesink: set sync=false
- s3sink: set sync=false
- sccparse: Fix leading spaces between the tab and caption data
- webrtchttp: Respect HTTP redirects
- webrtcsrc: use @watch instead of @to-owned
- webrtcsrc: add turn-servers property
- webrtc: Fix paths in README
- webrtcsink: don’t miss ice candidates
gst-libav
- No changes
gst-rtsp-server
- rtspclientsink: Don’t leak previous server_ip
gstreamer-vaapi
- No changes
gstreamer-sharp
- No changes
gst-omx
- No changes
gst-python
- No changes
gst-editing-services
- No changes
gst-validate + gst-integration-testsuites
- gst-validate: Fix compatibility with Python 3.12
gst-examples
- No changes
Development build environment
- No changes
Cerbero build tool and packaging changes in 1.22.7
- Add Windows 11 to the supported versions list
- ca-certificates: Update to version from 2023-08-22
- cargo: Restrict parallelism if a small system is detected (for
gst-plugins-rs build)
- Fix venv setup on Python 3.11+
- Fix unlinking of Android NDK directories if install fails midway
- glib: Work around AppleClang + -Werror test build failure
- glib: Re-add gio module loading patch for macOS, remove unused patch
files
Contributors to 1.22.7
Albert Sjölund, Arun Raghavan, Balló György, Benjamin Gaignard, Detlev
Casanova, Doug Nazar, Eric, Florian Zwoch, François Laignel, Guillaume
Desmottes, He Junyan, Hou Qi, James Oliver, Jan Alexander Steffens
(heftig), Jan Schmidt, Johan Adam Nilsson, Jordan Yelloz, Kalev Lember,
L. E. Segovia, Lieven Paulissen, Maksym Khomenko, Marek Vasut, Matthias
Fuchs, Michiel Westerbeek, Nicolas Dufresne, Philippe Normand, Piotr
Brzeziński, Rahul T R, Sean DuBois, Sebastian Dröge, Seungha Yang,
Stéphane Cerveau, Thibault Saunier, Tim-Philipp Müller,
… and many others who have contributed bug reports, translations, sent
suggestions or helped testing. Thank you all!
List of merge requests and issues fixed in 1.22.7
- List of Merge Requests applied in 1.22.7
- List of Issues fixed in 1.22.7Edited by Krassy Boykinov