Release 1.22.7

This commit is contained in:
Tim-Philipp Müller 2023-11-13 11:04:22 +00:00
parent 7dfaa57b6f
commit 4d13eddc8b
59 changed files with 2796 additions and 84 deletions

View file

@ -1,5 +1,5 @@
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,

View file

@ -2,13 +2,13 @@ GStreamer 1.22 Release Notes
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
@ -2366,6 +2366,222 @@ List of merge requests and issues fixed in 1.22.6
- 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 “Cant find exact taps”
debug log messages
- wasapi2: Dont 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 cant find exact taps to
debug
gst-plugins-good
- adaptivedemux2: Do not submit_transfer when cancelled
- adaptivedemux2: Call GTaskss 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: Dont 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: Dont 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: Dont 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: Dont remove old files if max-files is zero
- hlssink3: Dont 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: Dont 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: dont miss ice candidates
gst-libav
- No changes
gst-rtsp-server
- rtspclientsink: Dont 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

View file

@ -1,4 +1,4 @@
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!

View file

@ -53,6 +53,16 @@
</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>

View file

@ -1,5 +1,5 @@
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',

View file

@ -1,5 +1,5 @@
project('GStreamer manuals and tutorials', 'c',
version: '1.22.6.1',
version: '1.22.7',
meson_version : '>= 0.62')
hotdoc_p = find_program('hotdoc')

View file

@ -2,13 +2,13 @@ GStreamer 1.22 Release Notes
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
@ -2366,6 +2366,222 @@ List of merge requests and issues fixed in 1.22.6
- 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 “Cant find exact taps”
debug log messages
- wasapi2: Dont 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 cant find exact taps to
debug
gst-plugins-good
- adaptivedemux2: Do not submit_transfer when cancelled
- adaptivedemux2: Call GTaskss 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: Dont 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: Dont 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: Dont 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: Dont remove old files if max-files is zero
- hlssink3: Dont 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: Dont 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: dont miss ice candidates
gst-libav
- No changes
gst-rtsp-server
- rtspclientsink: Dont 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

View file

@ -1,4 +1,4 @@
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!

View file

@ -30,6 +30,16 @@ GStreamer library for creating audio and video editors
</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>

View file

@ -1,5 +1,5 @@
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' ])

View file

@ -1,4 +1,4 @@
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)

View file

@ -1 +1 @@
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')

View file

@ -2,13 +2,13 @@ GStreamer 1.22 Release Notes
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
@ -2366,6 +2366,222 @@ List of merge requests and issues fixed in 1.22.6
- 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 “Cant find exact taps”
debug log messages
- wasapi2: Dont 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 cant find exact taps to
debug
gst-plugins-good
- adaptivedemux2: Do not submit_transfer when cancelled
- adaptivedemux2: Call GTaskss 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: Dont 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: Dont 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: Dont 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: Dont remove old files if max-files is zero
- hlssink3: Dont 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: Dont 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: dont miss ice candidates
gst-libav
- No changes
gst-rtsp-server
- rtspclientsink: Dont 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

View file

@ -1,4 +1,4 @@
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!

View file

@ -32,6 +32,16 @@ colorspace conversion elements.
</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>

View file

@ -1,5 +1,5 @@
project('gst-libav', 'c',
version : '1.22.6.1',
version : '1.22.7',
meson_version : '>= 0.62',
default_options : [ 'warning_level=1',
'buildtype=debugoptimized' ])

View file

@ -2,13 +2,13 @@ GStreamer 1.22 Release Notes
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
@ -2366,6 +2366,222 @@ List of merge requests and issues fixed in 1.22.6
- 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 “Cant find exact taps”
debug log messages
- wasapi2: Dont 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 cant find exact taps to
debug
gst-plugins-good
- adaptivedemux2: Do not submit_transfer when cancelled
- adaptivedemux2: Call GTaskss 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: Dont 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: Dont 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: Dont 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: Dont remove old files if max-files is zero
- hlssink3: Dont 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: Dont 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: dont miss ice candidates
gst-libav
- No changes
gst-rtsp-server
- rtspclientsink: Dont 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

View file

@ -1,4 +1,4 @@
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!

View file

@ -31,6 +31,16 @@ a basic collection of elements
</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>

View file

@ -1,5 +1,5 @@
project('gst-omx', 'c',
version : '1.22.6.1',
version : '1.22.7',
meson_version : '>= 0.62',
default_options : [ 'warning_level=1',
'buildtype=debugoptimized' ])

View file

@ -2,13 +2,13 @@ GStreamer 1.22 Release Notes
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
@ -2366,6 +2366,222 @@ List of merge requests and issues fixed in 1.22.6
- 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 “Cant find exact taps”
debug log messages
- wasapi2: Dont 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 cant find exact taps to
debug
gst-plugins-good
- adaptivedemux2: Do not submit_transfer when cancelled
- adaptivedemux2: Call GTaskss 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: Dont 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: Dont 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: Dont 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: Dont remove old files if max-files is zero
- hlssink3: Dont 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: Dont 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: dont miss ice candidates
gst-libav
- No changes
gst-rtsp-server
- rtspclientsink: Dont 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

View file

@ -1,4 +1,4 @@
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!

View file

@ -33,6 +33,16 @@ real live maintainer, or some actual wide use.
</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>

View file

@ -1,5 +1,5 @@
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' ])

View file

@ -2,13 +2,13 @@ GStreamer 1.22 Release Notes
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
@ -2366,6 +2366,222 @@ List of merge requests and issues fixed in 1.22.6
- 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 “Cant find exact taps”
debug log messages
- wasapi2: Dont 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 cant find exact taps to
debug
gst-plugins-good
- adaptivedemux2: Do not submit_transfer when cancelled
- adaptivedemux2: Call GTaskss 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: Dont 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: Dont 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: Dont 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: Dont remove old files if max-files is zero
- hlssink3: Dont 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: Dont 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: dont miss ice candidates
gst-libav
- No changes
gst-rtsp-server
- rtspclientsink: Dont 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

View file

@ -1,4 +1,4 @@
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!

View file

@ -34,6 +34,16 @@ A wide range of video and audio decoders, encoders, and filters are included.
</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>

View file

@ -1,5 +1,5 @@
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' ])

View file

@ -2,13 +2,13 @@ GStreamer 1.22 Release Notes
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
@ -2366,6 +2366,222 @@ List of merge requests and issues fixed in 1.22.6
- 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 “Cant find exact taps”
debug log messages
- wasapi2: Dont 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 cant find exact taps to
debug
gst-plugins-good
- adaptivedemux2: Do not submit_transfer when cancelled
- adaptivedemux2: Call GTaskss 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: Dont 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: Dont 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: Dont 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: Dont remove old files if max-files is zero
- hlssink3: Dont 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: Dont 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: dont miss ice candidates
gst-libav
- No changes
gst-rtsp-server
- rtspclientsink: Dont 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

View file

@ -1,4 +1,4 @@
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!

View file

@ -7027,7 +7027,7 @@
"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",
@ -7039,7 +7039,7 @@
"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",
@ -21257,7 +21257,7 @@
"construct": false,
"construct-only": false,
"controllable": false,
"default": "GStreamer/1.22.6.1",
"default": "GStreamer/1.22.7",
"mutable": "null",
"readable": true,
"type": "gchararray",
@ -21816,7 +21816,7 @@
"construct": false,
"construct-only": false,
"controllable": false,
"default": "GStreamer 1.22.6.1",
"default": "GStreamer 1.22.7",
"mutable": "null",
"readable": true,
"type": "gchararray",
@ -23253,7 +23253,7 @@
"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",

View file

@ -32,6 +32,16 @@ the plug-in code, LGPL or LGPL-compatible for the supporting library).
</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>

View file

@ -1,5 +1,5 @@
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' ])

View file

@ -2,4 +2,4 @@
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

View file

@ -2,13 +2,13 @@ GStreamer 1.22 Release Notes
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
@ -2366,6 +2366,222 @@ List of merge requests and issues fixed in 1.22.6
- 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 “Cant find exact taps”
debug log messages
- wasapi2: Dont 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 cant find exact taps to
debug
gst-plugins-good
- adaptivedemux2: Do not submit_transfer when cancelled
- adaptivedemux2: Call GTaskss 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: Dont 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: Dont 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: Dont 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: Dont remove old files if max-files is zero
- hlssink3: Dont 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: Dont 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: dont miss ice candidates
gst-libav
- No changes
gst-rtsp-server
- rtspclientsink: Dont 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

View file

@ -1,4 +1,4 @@
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!

View file

@ -33,6 +33,16 @@ might be widely known to present patent problems.
</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>

View file

@ -1,5 +1,5 @@
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' ])

View file

@ -2,13 +2,13 @@ GStreamer 1.22 Release Notes
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
@ -2366,6 +2366,222 @@ List of merge requests and issues fixed in 1.22.6
- 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 “Cant find exact taps”
debug log messages
- wasapi2: Dont 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 cant find exact taps to
debug
gst-plugins-good
- adaptivedemux2: Do not submit_transfer when cancelled
- adaptivedemux2: Call GTaskss 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: Dont 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: Dont 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: Dont 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: Dont remove old files if max-files is zero
- hlssink3: Dont 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: Dont 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: dont miss ice candidates
gst-libav
- No changes
gst-rtsp-server
- rtspclientsink: Dont 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

View file

@ -1,4 +1,4 @@
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!

View file

@ -30,6 +30,16 @@ GStreamer Python Bindings is a set of overrides and Gst fundamental types handli
</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>

View file

@ -1,5 +1,5 @@
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',

View file

@ -2,13 +2,13 @@ GStreamer 1.22 Release Notes
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
@ -2366,6 +2366,222 @@ List of merge requests and issues fixed in 1.22.6
- 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 “Cant find exact taps”
debug log messages
- wasapi2: Dont 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 cant find exact taps to
debug
gst-plugins-good
- adaptivedemux2: Do not submit_transfer when cancelled
- adaptivedemux2: Call GTaskss 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: Dont 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: Dont 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: Dont 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: Dont remove old files if max-files is zero
- hlssink3: Dont 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: Dont 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: dont miss ice candidates
gst-libav
- No changes
gst-rtsp-server
- rtspclientsink: Dont 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

View file

@ -1,4 +1,4 @@
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!

View file

@ -333,7 +333,7 @@
"construct": false,
"construct-only": false,
"controllable": false,
"default": "GStreamer/1.22.6.1",
"default": "GStreamer/1.22.7",
"mutable": "null",
"readable": true,
"type": "gchararray",

View file

@ -30,6 +30,16 @@ RTSP server library based on GStreamer
</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>

View file

@ -1,5 +1,5 @@
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'])

View file

@ -1,4 +1,4 @@
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'

View file

@ -17,9 +17,9 @@ namespace Gst.PbUtils {
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
}
}

View file

@ -170,9 +170,9 @@ namespace Gst {
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
}
}

View file

@ -12127,10 +12127,10 @@
<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">
@ -21582,10 +21582,10 @@
<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">

View file

@ -2,13 +2,13 @@ GStreamer 1.22 Release Notes
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
@ -2366,6 +2366,222 @@ List of merge requests and issues fixed in 1.22.6
- 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 “Cant find exact taps”
debug log messages
- wasapi2: Dont 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 cant find exact taps to
debug
gst-plugins-good
- adaptivedemux2: Do not submit_transfer when cancelled
- adaptivedemux2: Call GTaskss 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: Dont 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: Dont 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: Dont 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: Dont remove old files if max-files is zero
- hlssink3: Dont 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: Dont 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: dont miss ice candidates
gst-libav
- No changes
gst-rtsp-server
- rtspclientsink: Dont 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

View file

@ -1,4 +1,4 @@
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!

View file

@ -25,6 +25,16 @@
</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>

View file

@ -1,5 +1,5 @@
project('gstreamer-vaapi', 'c',
version : '1.22.6.1',
version : '1.22.7',
meson_version : '>= 0.62',
default_options : [ 'warning_level=1',
'buildtype=debugoptimized' ])

View file

@ -2,13 +2,13 @@ GStreamer 1.22 Release Notes
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
@ -2366,6 +2366,222 @@ List of merge requests and issues fixed in 1.22.6
- 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 “Cant find exact taps”
debug log messages
- wasapi2: Dont 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 cant find exact taps to
debug
gst-plugins-good
- adaptivedemux2: Do not submit_transfer when cancelled
- adaptivedemux2: Call GTaskss 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: Dont 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: Dont 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: Dont 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: Dont remove old files if max-files is zero
- hlssink3: Dont 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: Dont 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: dont miss ice candidates
gst-libav
- No changes
gst-rtsp-server
- rtspclientsink: Dont 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

View file

@ -1,4 +1,4 @@
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!

View file

@ -38,6 +38,16 @@ hierarchy, and a set of media-agnostic core elements.
</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>

View file

@ -1,5 +1,5 @@
project('gstreamer', 'c',
version : '1.22.6.1',
version : '1.22.7',
meson_version : '>= 0.62',
default_options : [ 'warning_level=1',
'buildtype=debugoptimized' ])