diff mbox series

[03/10] gstreamer1.0-plugins-good: Upgrade 1.28.0 -> 1.28.2

Message ID 20260408141719.1086823-4-pkj@axis.com
State New
Headers show
Series Update the GStreamer recipes to 1.28.2 | expand

Commit Message

Peter Kjellerstedt April 8, 2026, 2:17 p.m. UTC 
Changes since 1.28.0
* b2a3b2e: Back to development after 1.28.0
* 99c68cb: meson: Fix libxml2 not building due to wrong option type
* 8388e74: qtdemux: Improve debug output around seek event handling
* db3c0c6: qtdemux: Rename last mention of last_stop to position
* 19e3c57: qtdemux: Set the segment position to the start on EOS in
  reverse playback mode
* 5c681d2: v4l2: Add v4l2av1dec stateful decoder support
* 6438539: v4l2: update gst plugins cache
* d3c0283: vpxdec: Support downstream pools with alignment requirements
* 0edccca: qtdemux: Don't ignore flow return when pushing queued
  buffers downstream
* ff3edfc: qtdemux: Make sure to not output the same samples multiple
  times in reverse playback mode
* d415a2b: qtdemux: Push raw audio/video buffers downstream in reverse
  order if rate < 0
* e1c1979: wavpackparse: Parse 32 bit sample rate and channel masks
* 74d8469: wavpackparse: Print hexadecimal numbers with leading zeroes
  for easier reading
* d296187: wavpackparse: Sync flags with latest wavpack
* 4ad4418: wavpackparse: Include sample type (int / float / dsd) in the
  caps
* 6631df5: wavpackparse: Correctly parse and advertise depth vs. width
* f4d6909: wavpackdec: Set OPEN_NORMALIZE flag to normalize floating
  point samples into the [-1,1] range
* 7005bb5: wavpackdec: Allow up to 4096 channels and up to maximum
  sample rate
* 7514ac2: wavpackdec: Re-create wavpack decoder on caps changes
* 2cda74a: wavpackdec: Output 24 bit samples as actual 24 bit samples
* 69e76ae: wavpackdec: Allocate output buffer via the base class
* 293e350: wavpackdec: Output 18 and 20 bit as their corresponding
  formats
* 4d3bdcb: wavpackenc: Extend caps a bit
* c5143e8: wavpackenc: Map buffer readwrite for channel reordering
* 3db67a3: wavpackenc: Add support for S8/S16/S18/S20/S24/S32 and F32
  samples
* 4704dfe: wavpack: Update plugin docs cache
* d333aeb: qml6glsrc: Fix rendering of scene with clipped items
* ee8e500: qml6glsrc: Fix scraping of QQuickWindow content
* 2b09aec: rtpsource: Add locking for receive reports table
* eaadb4d: gst-plugins-good: fix author name: Kentaro Fukuchi
* 0f77771: gst: fix author name: add missing closing angle bracket
* e92f814: rtph263pay: fix author name: where not separated by ','
* 4506913: gst-plugins: fix author name: correct incomplete or wrong
  emails
* d125725: gst: also adapt author names in the gst_plugins_cache.json
  files
* f68c471: rtpptdemux/rtpssrcdemux: adapt klass "Demux" to "Demuxer"
* 75ae4d7: gstrtspsrc: Set new mki in the encoder upon crypto update
* 09635fe: rtspsrc: Memory leak in gst_rtspsrc_close() when
  GST_RTSP_EEOF error occurs
* 835da19: modules: Remove NEWS from git which is generated from full
  release notes
* 201b14e: modules: remove RELEASE from git, will be generated from
  template on dist
* 7694a7d: modules: remove subproject README.md from git
* 6376a84: modules: dist common files from monorepo root
* 291e479: meson: Deprecate `system = 'ios'` in cross files, use
  subsystem
* 88febbc: osxaudio: Stop using HAVE_IOS, use TARGET_OS_* macros
  instead
* c3b73e9: qtdemux: Fix out-of-bounds read when parsing PlayReady DRM
  UUIDs
* b4558a4: rtpqdm2depay: error out if anyone tries to use this element
* 1171ae8: wavparse: Remove pointless duplicated GST_ROUND_UP_2()
* 3564405: wavparse: Use unsigned integers for data sizes
* c73a1f4: wavparse: Use GST_ROUND_UP_2() in two more places instead of
  a manual implementation
* 8822ee3: wavparse: Define maximum chunk size in a single place
* 081484e: wavparse: Avoid integer overflow and out-of-bounds read when
  parsing adtl chunks
* dcb37e2: Release 1.28.1
* 32113a6: Back to development after 1.28.1
* 5ee8c64: rtptwcc: fix feedback packet count wrapping at 255
* 216d38a: all: GThreadFunc return type fixes
* 180a877: rtph264depay: fix invalid memory access in
  gst_rtp_h264_finish_fragmentation_unit
* d824117: Qt6GLVideoItem: caps update fixed
* d6ed0a0: qtdemux: fix invalid WebVTT timestamps
* b8436bf: wavparse: Avoid overflow in length when setting
  ignore-length=true
* 586ff9c: wavparse: Fix parsing of RF64 wave files
* b08a64e: rgvolume: don't apply dBSPL reference level compensation for
  LUFS values
* 9ca0bd6: hlsdemux2: fix seekable range for live HLS streams
* 100a0e6: qtdemux: Don't immediately push segment after moov in push
  mode for fmp4
* 95919fa: wavenc: Skip writing empty LIST INFO chunk
* e717c43: gst-plugins-good: update translations
* 1ac03ff: qtdemux: fix handling of in-between fragments without tfdt
* 457b197: qtdemux: Preserve Metas and Flags when doing row alignment
* 27a9cc0: qtdemux: Avoid integer overflows when handling transform
  matrices
* 1279ec9: qtdemux: Don't store 64 bit integers in 32 bit integers to
  avoid overflows
* 0871bb2: qtdemux: Check that big enough stco/stsz are available when
  parsing sample tables
* 160bba0: qtdemux: Error out instead of trying to handle a truncated
  stts box
* b73c493: audioinvert: fix float truncation in transform_float
* bcc8c6e: qmlglsink: Fix for caps tracking on multiple setCaps calls
* 5ebb94c: qt6: Avoid parsing caps on every buffer
* 49bab9c: qt5: Avoid parsing caps on every buffer
* c4f56c0: rtspsrc: Discard early data in ONVIF mode
* 86e640a: rtspsrc: Fix const-correctness issue around strchr() usage
* 48cefc4: flvmux: fix race condition on caps get and check
* dbd4cb4: qtdemux: Avoid division by zero if 0 audio channels are
  signalled
* 10fd1ab: qtdemux: Validate chnl defined layout before using it to
  index the layouts array
* 3441881: qtdemux: Avoid out-of-bounds reads and writes of 64 item
  audio channel positions array
* dc7ab66: qtdemux: Fix bit pattern check for omitted audio channels
  map
* bad6721: qtdemux: Add various integer overflow and bounds checks to
  uncompressed video handling
* 8aed48f: flvdemux: Avoid assertions on corrupted streams
* 35a905a: wavparse: Fix integer overflow when checking available
  buffer size for reading cues
* 0d819ce: wavparse: Use prepend+reverse instead of append when
  building the cues list
* 6db6dd0: matroskademux: Add missing parenthesis when calculating bz2
  buffer sizes
* 43421c2: Release 1.28.2

Signed-off-by: Peter Kjellerstedt <peter.kjellerstedt@axis.com>
---
 ...ugins-good_1.28.0.bb => gstreamer1.0-plugins-good_1.28.2.bb} | 2 +-
 1 file changed, 1 insertion(+), 1 deletion(-)
 rename meta/recipes-multimedia/gstreamer/{gstreamer1.0-plugins-good_1.28.0.bb => gstreamer1.0-plugins-good_1.28.2.bb} (97%)
diff mbox series

Patch

diff --git a/meta/recipes-multimedia/gstreamer/gstreamer1.0-plugins-good_1.28.0.bb b/meta/recipes-multimedia/gstreamer/gstreamer1.0-plugins-good_1.28.2.bb
similarity index 97%
rename from meta/recipes-multimedia/gstreamer/gstreamer1.0-plugins-good_1.28.0.bb
rename to meta/recipes-multimedia/gstreamer/gstreamer1.0-plugins-good_1.28.2.bb
index 116e328adf..366eb8189c 100644
--- a/meta/recipes-multimedia/gstreamer/gstreamer1.0-plugins-good_1.28.0.bb
+++ b/meta/recipes-multimedia/gstreamer/gstreamer1.0-plugins-good_1.28.2.bb
@@ -6,7 +6,7 @@  BUGTRACKER = "https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/issues
 
 SRC_URI = "https://gstreamer.freedesktop.org/src/gst-plugins-good/gst-plugins-good-${PV}.tar.xz"
 
-SRC_URI[sha256sum] = "d97700f346fdf9ef5461c035e23ed1ce916ca7a31d6ddad987f774774361db77"
+SRC_URI[sha256sum] = "1ace2d8ec74f632d82eab5006753a27fe0c2402db4ca94d63271e494b62f50bf"
 
 S = "${UNPACKDIR}/gst-plugins-good-${PV}"