project('gstreamer-full', 'c',
- version : '1.22.6.1',
+ version : '1.22.7',
meson_version : '>= 0.62.0',
default_options : ['buildtype=debugoptimized',
# Needed due to https://github.com/mesonbuild/meson/issues/1889,
GStreamer 1.22.0 was originally released on 23 January 2023.
-The latest bug-fix release in the stable 1.22 series is 1.22.5 and was
-released on 20 July 2023.
+The latest bug-fix release in the stable 1.22 series is 1.22.7 and was
+released on 13 November 2023.
See https://gstreamer.freedesktop.org/releases/1.22/ for the latest
version of this document.
-Last updated: Thursday 20 July 2023, 12:00 UTC (log)
+Last updated: Monday 13 November 2023, 10:00 UTC (log)
Introduction
- List of Merge Requests applied in 1.22.6
- List of Issues fixed in 1.22.6
+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 1.22.7
+
Schedule for 1.24
Our next major feature release will be 1.24, and 1.23 will be the
-This is GStreamer gst-devtools 1.22.6.
+This is GStreamer gst-devtools 1.22.7.
The GStreamer team is thrilled to announce a new major feature release
of your favourite cross-platform multimedia framework!
</GitRepository>
</repository>
+ <release>
+ <Version>
+ <revision>1.22.7</revision>
+ <branch>1.22</branch>
+ <name></name>
+ <created>2023-11-13</created>
+ <file-release rdf:resource="https://gstreamer.freedesktop.org/src/gst-devtools/gst-devtools-1.22.7.tar.xz" />
+ </Version>
+ </release>
+
<release>
<Version>
<revision>1.22.6</revision>
project('gst-devtools', 'c',
- version : '1.22.6.1',
+ version : '1.22.7',
meson_version : '>= 0.62',
default_options : [ 'warning_level=1',
'c_std=gnu99',
project('GStreamer manuals and tutorials', 'c',
- version: '1.22.6.1',
+ version: '1.22.7',
meson_version : '>= 0.62')
hotdoc_p = find_program('hotdoc')
GStreamer 1.22.0 was originally released on 23 January 2023.
-The latest bug-fix release in the stable 1.22 series is 1.22.5 and was
-released on 20 July 2023.
+The latest bug-fix release in the stable 1.22 series is 1.22.7 and was
+released on 13 November 2023.
See https://gstreamer.freedesktop.org/releases/1.22/ for the latest
version of this document.
-Last updated: Thursday 20 July 2023, 12:00 UTC (log)
+Last updated: Monday 13 November 2023, 10:00 UTC (log)
Introduction
- List of Merge Requests applied in 1.22.6
- List of Issues fixed in 1.22.6
+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 1.22.7
+
Schedule for 1.24
Our next major feature release will be 1.24, and 1.23 will be the
-This is GStreamer gst-editing-services 1.22.6.
+This is GStreamer gst-editing-services 1.22.7.
The GStreamer team is thrilled to announce a new major feature release
of your favourite cross-platform multimedia framework!
</GitRepository>
</repository>
+ <release>
+ <Version>
+ <revision>1.22.7</revision>
+ <branch>1.22</branch>
+ <name></name>
+ <created>2023-11-13</created>
+ <file-release rdf:resource="https://gstreamer.freedesktop.org/src/gst-editing-services/gst-editing-services-1.22.7.tar.xz" />
+ </Version>
+ </release>
+
<release>
<Version>
<revision>1.22.6</revision>
project('gst-editing-services', 'c',
- version : '1.22.6.1',
+ version : '1.22.7',
meson_version : '>= 0.62',
default_options : [ 'warning_level=1',
'buildtype=debugoptimized' ])
-project('gst-examples', 'c', version : '1.22.6.1', license : 'LGPL')
+project('gst-examples', 'c', version : '1.22.7', license : 'LGPL')
cc = meson.get_compiler('c')
m_dep = cc.find_library('m', required : false)
-project('gst-integration-testsuites', [], version: '1.22.6.1', meson_version : '>= 0.62', license: 'LGPL')
+project('gst-integration-testsuites', [], version: '1.22.7', meson_version : '>= 0.62', license: 'LGPL')
GStreamer 1.22.0 was originally released on 23 January 2023.
-The latest bug-fix release in the stable 1.22 series is 1.22.5 and was
-released on 20 July 2023.
+The latest bug-fix release in the stable 1.22 series is 1.22.7 and was
+released on 13 November 2023.
See https://gstreamer.freedesktop.org/releases/1.22/ for the latest
version of this document.
-Last updated: Thursday 20 July 2023, 12:00 UTC (log)
+Last updated: Monday 13 November 2023, 10:00 UTC (log)
Introduction
- List of Merge Requests applied in 1.22.6
- List of Issues fixed in 1.22.6
+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 1.22.7
+
Schedule for 1.24
Our next major feature release will be 1.24, and 1.23 will be the
-This is GStreamer gst-libav 1.22.6.
+This is GStreamer gst-libav 1.22.7.
The GStreamer team is thrilled to announce a new major feature release
of your favourite cross-platform multimedia framework!
</GitRepository>
</repository>
+ <release>
+ <Version>
+ <revision>1.22.7</revision>
+ <branch>1.22</branch>
+ <name></name>
+ <created>2023-11-13</created>
+ <file-release rdf:resource="https://gstreamer.freedesktop.org/src/gst-libav/gst-libav-1.22.7.tar.xz" />
+ </Version>
+ </release>
+
<release>
<Version>
<revision>1.22.6</revision>
project('gst-libav', 'c',
- version : '1.22.6.1',
+ version : '1.22.7',
meson_version : '>= 0.62',
default_options : [ 'warning_level=1',
'buildtype=debugoptimized' ])
GStreamer 1.22.0 was originally released on 23 January 2023.
-The latest bug-fix release in the stable 1.22 series is 1.22.5 and was
-released on 20 July 2023.
+The latest bug-fix release in the stable 1.22 series is 1.22.7 and was
+released on 13 November 2023.
See https://gstreamer.freedesktop.org/releases/1.22/ for the latest
version of this document.
-Last updated: Thursday 20 July 2023, 12:00 UTC (log)
+Last updated: Monday 13 November 2023, 10:00 UTC (log)
Introduction
- List of Merge Requests applied in 1.22.6
- List of Issues fixed in 1.22.6
+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 1.22.7
+
Schedule for 1.24
Our next major feature release will be 1.24, and 1.23 will be the
-This is GStreamer gst-omx 1.22.6.
+This is GStreamer gst-omx 1.22.7.
The GStreamer team is thrilled to announce a new major feature release
of your favourite cross-platform multimedia framework!
</GitRepository>
</repository>
+ <release>
+ <Version>
+ <revision>1.22.7</revision>
+ <branch>1.22</branch>
+ <name></name>
+ <created>2023-11-13</created>
+ <file-release rdf:resource="https://gstreamer.freedesktop.org/src/gst-omx/gst-omx-1.22.7.tar.xz" />
+ </Version>
+ </release>
+
<release>
<Version>
<revision>1.22.6</revision>
project('gst-omx', 'c',
- version : '1.22.6.1',
+ version : '1.22.7',
meson_version : '>= 0.62',
default_options : [ 'warning_level=1',
'buildtype=debugoptimized' ])
GStreamer 1.22.0 was originally released on 23 January 2023.
-The latest bug-fix release in the stable 1.22 series is 1.22.5 and was
-released on 20 July 2023.
+The latest bug-fix release in the stable 1.22 series is 1.22.7 and was
+released on 13 November 2023.
See https://gstreamer.freedesktop.org/releases/1.22/ for the latest
version of this document.
-Last updated: Thursday 20 July 2023, 12:00 UTC (log)
+Last updated: Monday 13 November 2023, 10:00 UTC (log)
Introduction
- List of Merge Requests applied in 1.22.6
- List of Issues fixed in 1.22.6
+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 1.22.7
+
Schedule for 1.24
Our next major feature release will be 1.24, and 1.23 will be the
-This is GStreamer gst-plugins-bad 1.22.6.
+This is GStreamer gst-plugins-bad 1.22.7.
The GStreamer team is thrilled to announce a new major feature release
of your favourite cross-platform multimedia framework!
</GitRepository>
</repository>
+ <release>
+ <Version>
+ <revision>1.22.7</revision>
+ <branch>1.22</branch>
+ <name></name>
+ <created>2023-11-13</created>
+ <file-release rdf:resource="https://gstreamer.freedesktop.org/src/gst-plugins-bad/gst-plugins-bad-1.22.7.tar.xz" />
+ </Version>
+ </release>
+
<release>
<Version>
<revision>1.22.6</revision>
project('gst-plugins-bad', 'c', 'cpp',
- version : '1.22.6.1',
+ version : '1.22.7',
meson_version : '>= 0.62',
default_options : [ 'warning_level=1',
'buildtype=debugoptimized' ])
GStreamer 1.22.0 was originally released on 23 January 2023.
-The latest bug-fix release in the stable 1.22 series is 1.22.5 and was
-released on 20 July 2023.
+The latest bug-fix release in the stable 1.22 series is 1.22.7 and was
+released on 13 November 2023.
See https://gstreamer.freedesktop.org/releases/1.22/ for the latest
version of this document.
-Last updated: Thursday 20 July 2023, 12:00 UTC (log)
+Last updated: Monday 13 November 2023, 10:00 UTC (log)
Introduction
- List of Merge Requests applied in 1.22.6
- List of Issues fixed in 1.22.6
+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 1.22.7
+
Schedule for 1.24
Our next major feature release will be 1.24, and 1.23 will be the
-This is GStreamer gst-plugins-base 1.22.6.
+This is GStreamer gst-plugins-base 1.22.7.
The GStreamer team is thrilled to announce a new major feature release
of your favourite cross-platform multimedia framework!
</GitRepository>
</repository>
+ <release>
+ <Version>
+ <revision>1.22.7</revision>
+ <branch>1.22</branch>
+ <name></name>
+ <created>2023-11-13</created>
+ <file-release rdf:resource="https://gstreamer.freedesktop.org/src/gst-plugins-base/gst-plugins-base-1.22.7.tar.xz" />
+ </Version>
+ </release>
+
<release>
<Version>
<revision>1.22.6</revision>
project('gst-plugins-base', 'c',
- version : '1.22.6.1',
+ version : '1.22.7',
meson_version : '>= 0.62',
default_options : [ 'warning_level=1',
'buildtype=debugoptimized' ])
GStreamer 1.22.0 was originally released on 23 January 2023.
-The latest bug-fix release in the stable 1.22 series is 1.22.5 and was
-released on 20 July 2023.
+The latest bug-fix release in the stable 1.22 series is 1.22.7 and was
+released on 13 November 2023.
See https://gstreamer.freedesktop.org/releases/1.22/ for the latest
version of this document.
-Last updated: Thursday 20 July 2023, 12:00 UTC (log)
+Last updated: Monday 13 November 2023, 10:00 UTC (log)
Introduction
- List of Merge Requests applied in 1.22.6
- List of Issues fixed in 1.22.6
+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 1.22.7
+
Schedule for 1.24
Our next major feature release will be 1.24, and 1.23 will be the
-This is GStreamer gst-plugins-good 1.22.6.
+This is GStreamer gst-plugins-good 1.22.7.
The GStreamer team is thrilled to announce a new major feature release
of your favourite cross-platform multimedia framework!
"construct": false,
"construct-only": false,
"controllable": false,
- "default": "GStreamer 1.22.6.1 FLV muxer",
+ "default": "GStreamer 1.22.7 FLV muxer",
"mutable": "null",
"readable": true,
"type": "gchararray",
"construct": false,
"construct-only": false,
"controllable": false,
- "default": "GStreamer 1.22.6.1 FLV muxer",
+ "default": "GStreamer 1.22.7 FLV muxer",
"mutable": "null",
"readable": true,
"type": "gchararray",
"construct": false,
"construct-only": false,
"controllable": false,
- "default": "GStreamer/1.22.6.1",
+ "default": "GStreamer/1.22.7",
"mutable": "null",
"readable": true,
"type": "gchararray",
"construct": false,
"construct-only": false,
"controllable": false,
- "default": "GStreamer 1.22.6.1",
+ "default": "GStreamer 1.22.7",
"mutable": "null",
"readable": true,
"type": "gchararray",
"construct": false,
"construct-only": false,
"controllable": false,
- "default": "GStreamer souphttpsrc 1.22.6.1 ",
+ "default": "GStreamer souphttpsrc 1.22.7 ",
"mutable": "null",
"readable": true,
"type": "gchararray",
</GitRepository>
</repository>
+ <release>
+ <Version>
+ <revision>1.22.7</revision>
+ <branch>1.22</branch>
+ <name></name>
+ <created>2023-11-13</created>
+ <file-release rdf:resource="https://gstreamer.freedesktop.org/src/gst-plugins-good/gst-plugins-good-1.22.7.tar.xz" />
+ </Version>
+ </release>
+
<release>
<Version>
<revision>1.22.6</revision>
project('gst-plugins-good', 'c',
- version : '1.22.6.1',
+ version : '1.22.7',
meson_version : '>= 0.62',
default_options : [ 'warning_level=1',
'buildtype=debugoptimized' ])
directory=gst-plugins-rs
url=https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs.git
push-url=git@gitlab.freedesktop.org:gstreamer/gst-plugins-rs.git
-revision=0.9
+revision=gstreamer-1.22.7
GStreamer 1.22.0 was originally released on 23 January 2023.
-The latest bug-fix release in the stable 1.22 series is 1.22.5 and was
-released on 20 July 2023.
+The latest bug-fix release in the stable 1.22 series is 1.22.7 and was
+released on 13 November 2023.
See https://gstreamer.freedesktop.org/releases/1.22/ for the latest
version of this document.
-Last updated: Thursday 20 July 2023, 12:00 UTC (log)
+Last updated: Monday 13 November 2023, 10:00 UTC (log)
Introduction
- List of Merge Requests applied in 1.22.6
- List of Issues fixed in 1.22.6
+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 1.22.7
+
Schedule for 1.24
Our next major feature release will be 1.24, and 1.23 will be the
-This is GStreamer gst-plugins-ugly 1.22.6.
+This is GStreamer gst-plugins-ugly 1.22.7.
The GStreamer team is thrilled to announce a new major feature release
of your favourite cross-platform multimedia framework!
</GitRepository>
</repository>
+ <release>
+ <Version>
+ <revision>1.22.7</revision>
+ <branch>1.22</branch>
+ <name></name>
+ <created>2023-11-13</created>
+ <file-release rdf:resource="https://gstreamer.freedesktop.org/src/gst-plugins-ugly/gst-plugins-ugly-1.22.7.tar.xz" />
+ </Version>
+ </release>
+
<release>
<Version>
<revision>1.22.6</revision>
project('gst-plugins-ugly', 'c',
- version : '1.22.6.1',
+ version : '1.22.7',
meson_version : '>= 0.62',
default_options : [ 'warning_level=1',
'buildtype=debugoptimized' ])
GStreamer 1.22.0 was originally released on 23 January 2023.
-The latest bug-fix release in the stable 1.22 series is 1.22.5 and was
-released on 20 July 2023.
+The latest bug-fix release in the stable 1.22 series is 1.22.7 and was
+released on 13 November 2023.
See https://gstreamer.freedesktop.org/releases/1.22/ for the latest
version of this document.
-Last updated: Thursday 20 July 2023, 12:00 UTC (log)
+Last updated: Monday 13 November 2023, 10:00 UTC (log)
Introduction
- List of Merge Requests applied in 1.22.6
- List of Issues fixed in 1.22.6
+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 1.22.7
+
Schedule for 1.24
Our next major feature release will be 1.24, and 1.23 will be the
-This is GStreamer gst-python 1.22.6.
+This is GStreamer gst-python 1.22.7.
The GStreamer team is thrilled to announce a new major feature release
of your favourite cross-platform multimedia framework!
</GitRepository>
</repository>
+ <release>
+ <Version>
+ <revision>1.22.7</revision>
+ <branch>1.22</branch>
+ <name></name>
+ <created>2023-11-13</created>
+ <file-release rdf:resource="https://gstreamer.freedesktop.org/src/gst-python/gst-python-1.22.7.tar.xz" />
+ </Version>
+ </release>
+
<release>
<Version>
<revision>1.22.6</revision>
project('gst-python', 'c',
- version : '1.22.6.1',
+ version : '1.22.7',
meson_version : '>= 0.62',
default_options : [ 'warning_level=1',
'c_std=gnu99',
GStreamer 1.22.0 was originally released on 23 January 2023.
-The latest bug-fix release in the stable 1.22 series is 1.22.5 and was
-released on 20 July 2023.
+The latest bug-fix release in the stable 1.22 series is 1.22.7 and was
+released on 13 November 2023.
See https://gstreamer.freedesktop.org/releases/1.22/ for the latest
version of this document.
-Last updated: Thursday 20 July 2023, 12:00 UTC (log)
+Last updated: Monday 13 November 2023, 10:00 UTC (log)
Introduction
- List of Merge Requests applied in 1.22.6
- List of Issues fixed in 1.22.6
+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 1.22.7
+
Schedule for 1.24
Our next major feature release will be 1.24, and 1.23 will be the
-This is GStreamer gst-rtsp-server 1.22.6.
+This is GStreamer gst-rtsp-server 1.22.7.
The GStreamer team is thrilled to announce a new major feature release
of your favourite cross-platform multimedia framework!
"construct": false,
"construct-only": false,
"controllable": false,
- "default": "GStreamer/1.22.6.1",
+ "default": "GStreamer/1.22.7",
"mutable": "null",
"readable": true,
"type": "gchararray",
</GitRepository>
</repository>
+ <release>
+ <Version>
+ <revision>1.22.7</revision>
+ <branch>1.22</branch>
+ <name></name>
+ <created>2023-11-13</created>
+ <file-release rdf:resource="https://gstreamer.freedesktop.org/src/gst-rtsp-server/gst-rtsp-server-1.22.7.tar.xz" />
+ </Version>
+ </release>
+
<release>
<Version>
<revision>1.22.6</revision>
project('gst-rtsp-server', 'c',
- version : '1.22.6.1',
+ version : '1.22.7',
meson_version : '>= 0.62',
default_options : ['warning_level=1', 'buildtype=debugoptimized'])
-project('gstreamer-sharp', ['cs', 'c'], version: '1.22.6.1',
+project('gstreamer-sharp', ['cs', 'c'], version: '1.22.7',
meson_version : '>= 0.62', license: 'LGPL')
if host_machine.system() == 'osx'
public const string ENCODING_CATEGORY_ONLINE_SERVICE = @"online-service";
public const string ENCODING_CATEGORY_STORAGE_EDITING = @"storage-editing";
public const int PLUGINS_BASE_VERSION_MAJOR = 1;
- public const int PLUGINS_BASE_VERSION_MICRO = 6;
+ public const int PLUGINS_BASE_VERSION_MICRO = 7;
public const int PLUGINS_BASE_VERSION_MINOR = 22;
- public const int PLUGINS_BASE_VERSION_NANO = 1;
+ public const int PLUGINS_BASE_VERSION_NANO = 0;
#endregion
}
}
public const int VALUE_LESS_THAN = -1;
public const int VALUE_UNORDERED = 2;
public const int VERSION_MAJOR = 1;
- public const int VERSION_MICRO = 6;
+ public const int VERSION_MICRO = 7;
public const int VERSION_MINOR = 22;
- public const int VERSION_NANO = 1;
+ public const int VERSION_NANO = 0;
#endregion
}
}
<constant value="1" ctype="gint" gtype="gint" name="VALUE_GREATER_THAN" />
<constant value="-1" ctype="gint" gtype="gint" name="VALUE_LESS_THAN" />
<constant value="2" ctype="gint" gtype="gint" name="VALUE_UNORDERED" />
- <constant value="1" ctype="gint" gtype="gint" name="VERSION_MAJOR" />
- <constant value="6" ctype="gint" gtype="gint" name="VERSION_MICRO" />
- <constant value="22" ctype="gint" gtype="gint" name="VERSION_MINOR" />
- <constant value="1" ctype="gint" gtype="gint" name="VERSION_NANO" />
+ <constant value="1" ctype="gint" gtype="gint" name="VERSION_MAJOR" />
+ <constant value="7" ctype="gint" gtype="gint" name="VERSION_MICRO" />
+ <constant value="22" ctype="gint" gtype="gint" name="VERSION_MINOR" />
+ <constant value="0" ctype="gint" gtype="gint" name="VERSION_NANO" />
</object>
<class name="Parse" cname="GstParse" disable_void_ctor="1">
<method name="ParseBinFromDescription" cname="gst_parse_bin_from_description" shared="true">
<constant value="file-extension" ctype="gchar*" gtype="gchar*" name="ENCODING_CATEGORY_FILE_EXTENSION" />
<constant value="online-service" ctype="gchar*" gtype="gchar*" name="ENCODING_CATEGORY_ONLINE_SERVICE" />
<constant value="storage-editing" ctype="gchar*" gtype="gchar*" name="ENCODING_CATEGORY_STORAGE_EDITING" />
- <constant value="1" ctype="gint" gtype="gint" name="PLUGINS_BASE_VERSION_MAJOR" />
- <constant value="6" ctype="gint" gtype="gint" name="PLUGINS_BASE_VERSION_MICRO" />
- <constant value="22" ctype="gint" gtype="gint" name="PLUGINS_BASE_VERSION_MINOR" />
- <constant value="1" ctype="gint" gtype="gint" name="PLUGINS_BASE_VERSION_NANO" />
+ <constant value="1" ctype="gint" gtype="gint" name="PLUGINS_BASE_VERSION_MAJOR" />
+ <constant value="7" ctype="gint" gtype="gint" name="PLUGINS_BASE_VERSION_MICRO" />
+ <constant value="22" ctype="gint" gtype="gint" name="PLUGINS_BASE_VERSION_MINOR" />
+ <constant value="0" ctype="gint" gtype="gint" name="PLUGINS_BASE_VERSION_NANO" />
</object>
</namespace>
<namespace name="Gst.Rtp" library="gstrtp-1.0-0.dll">
GStreamer 1.22.0 was originally released on 23 January 2023.
-The latest bug-fix release in the stable 1.22 series is 1.22.5 and was
-released on 20 July 2023.
+The latest bug-fix release in the stable 1.22 series is 1.22.7 and was
+released on 13 November 2023.
See https://gstreamer.freedesktop.org/releases/1.22/ for the latest
version of this document.
-Last updated: Thursday 20 July 2023, 12:00 UTC (log)
+Last updated: Monday 13 November 2023, 10:00 UTC (log)
Introduction
- List of Merge Requests applied in 1.22.6
- List of Issues fixed in 1.22.6
+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 1.22.7
+
Schedule for 1.24
Our next major feature release will be 1.24, and 1.23 will be the
-This is GStreamer gstreamer-vaapi 1.22.6.
+This is GStreamer gstreamer-vaapi 1.22.7.
The GStreamer team is thrilled to announce a new major feature release
of your favourite cross-platform multimedia framework!
</GitRepository>
</repository>
+ <release>
+ <Version>
+ <revision>1.22.7</revision>
+ <branch>1.22</branch>
+ <name></name>
+ <created>2023-11-13</created>
+ <file-release rdf:resource="https://gstreamer.freedesktop.org/src/gstreamer-vaapi/gstreamer-vaapi-1.22.7.tar.xz" />
+ </Version>
+ </release>
+
<release>
<Version>
<revision>1.22.6</revision>
project('gstreamer-vaapi', 'c',
- version : '1.22.6.1',
+ version : '1.22.7',
meson_version : '>= 0.62',
default_options : [ 'warning_level=1',
'buildtype=debugoptimized' ])
GStreamer 1.22.0 was originally released on 23 January 2023.
-The latest bug-fix release in the stable 1.22 series is 1.22.5 and was
-released on 20 July 2023.
+The latest bug-fix release in the stable 1.22 series is 1.22.7 and was
+released on 13 November 2023.
See https://gstreamer.freedesktop.org/releases/1.22/ for the latest
version of this document.
-Last updated: Thursday 20 July 2023, 12:00 UTC (log)
+Last updated: Monday 13 November 2023, 10:00 UTC (log)
Introduction
- List of Merge Requests applied in 1.22.6
- List of Issues fixed in 1.22.6
+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 1.22.7
+
Schedule for 1.24
Our next major feature release will be 1.24, and 1.23 will be the
-This is GStreamer core 1.22.6.
+This is GStreamer core 1.22.7.
The GStreamer team is thrilled to announce a new major feature release
of your favourite cross-platform multimedia framework!
</GitRepository>
</repository>
+ <release>
+ <Version>
+ <revision>1.22.7</revision>
+ <branch>1.22</branch>
+ <name></name>
+ <created>2023-11-13</created>
+ <file-release rdf:resource="https://gstreamer.freedesktop.org/src/gstreamer/gstreamer-1.22.7.tar.xz" />
+ </Version>
+ </release>
+
<release>
<Version>
<revision>1.22.6</revision>
project('gstreamer', 'c',
- version : '1.22.6.1',
+ version : '1.22.7',
meson_version : '>= 0.62',
default_options : [ 'warning_level=1',
'buildtype=debugoptimized' ])
projects / platform / upstream / gstreamer.git / commitdiff