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78703d3c-b907-432f-b066-88f7af9ca3af X-MS-Exchange-CrossTenant-OriginalAttributedTenantConnectingIp: TenantId=78703d3c-b907-432f-b066-88f7af9ca3af;Ip=[195.60.68.100];Helo=[mail.axis.com] X-MS-Exchange-CrossTenant-AuthSource: DB3PEPF0000885A.eurprd02.prod.outlook.com X-MS-Exchange-CrossTenant-AuthAs: Anonymous X-MS-Exchange-CrossTenant-FromEntityHeader: HybridOnPrem X-MS-Exchange-Transport-CrossTenantHeadersStamped: DB3PR02MB10403 List-Id: X-Webhook-Received: from 45-33-107-173.ip.linodeusercontent.com [45.33.107.173] by aws-us-west-2-korg-lkml-1.web.codeaurora.org with HTTPS for ; Wed, 08 Apr 2026 14:17:39 -0000 X-Groupsio-URL: https://lists.openembedded.org/g/openembedded-core/message/234842 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 --- ...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 --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}"