GStreamer 1.22 Release Notes GStreamer 1.22.0 was originally released on 23 January 2023. The latest bug-fix release in the now old-stable 1.22 series is 1.22.12 and was released on 29 April 2024. See https://gstreamer.freedesktop.org/releases/1.22/ for the latest version of this document. The GStreamer 1.22 stable series has since been superseded by the GStreamer 1.24 stable release series. Last updated: Monday 29 April 2024, 20:00 UTC (log) Introduction The GStreamer team is proud to announce a new major feature release in the stable 1.x API series of your favourite cross-platform multimedia framework! As always, this release is again packed with many new features, bug fixes and other improvements. Highlights - AV1 video codec support improvements - New HLS, DASH and Microsoft Smooth Streaming adaptive streaming clients - Qt6 support for rendering video inside a QML scene - Minimal builds optimised for binary size, including only the individual elements needed - Playbin3, Decodebin3, UriDecodebin3, Parsebin enhancements and stabilisation - WebRTC simulcast support and support for Google Congestion Control - WebRTC-based media server ingestion/egress (WHIP/WHEP) support - New easy to use batteries-included WebRTC sender plugin - Easy RTP sender timestamp reconstruction for RTP and RTSP - ONVIF timed metadata support - New fragmented MP4 muxer and non-fragmented MP4 muxer - New plugins for Amazon AWS storage and audio transcription services - New gtk4paintablesink and gtkwaylandsink renderers - New videocolorscale element that can convert and scale in one go for better performance - High bit-depth video improvements - Touchscreen event support in navigation API - Rust plugins now shipped in macOS and Windows/MSVC binary packages - H.264/H.265 timestamp correction elements for PTS/DTS reconstruction before muxers - Improved design for DMA buffer sharing and modifier handling for hardware-accelerated video decoders/encoders/filters and capturing/rendering on Linux - Video4Linux2 hardware accelerated decoder improvements - CUDA integration and Direct3D11 integration and plugin improvements - New H.264 / AVC, H.265 / HEVC and AV1 hardware-accelerated video encoders for AMD GPUs using the Advanced Media Framework (AMF) SDK - applemedia: H.265 / HEVC video encoding + decoding support - androidmedia: H.265 / HEVC video encoding support - New “force-live” property for audiomixer, compositor, glvideomixer, d3d11compositor etc. - Lots of new plugins, features, performance improvements and bug fixes Major new features and changes AV1 video codec support improvements AV1 is a royalty free next-generation video codec by AOMedia and a free alternative to H.265/HEVC. While supported in earlier versions of GStreamer already, this release saw a lot of improvements across the board: - Support for hardware encoding and decoding via VAAPI/VA, AMF, D3D11, NVCODEC, QSV and Intel MediaSDK. Hardware codecs for AV1 are slowly becoming available in embedded systems and desktop GPUs (AMD, Intel, NVIDIA), and these can now be used via GStreamer. - New AV1 RTP payloader and depayloader elements. - New encoder settings in the AOM reference encoder-based av1enc element. - Various improvements in the AV1 parser and in the MP4/Matroska/WebM muxers/demuxers. - dav1d and rav1e based software decoder/encoder elements shipped as part of the binaries. - AV1 parser improvements and various bugfixes all over the place. Touchscreen event support in Navigation API The Navigation API supports the sending of key press events and mouse events through a GStreamer pipeline. Typically these will be picked up by a video sink on which these events happen and then the event is transmitted into the pipeline so it can be handled by elements inside the pipeline if it wasn’t handled by the application. This has traditionally been used for DVD menu support, but can also be used to forward such inputs to source elements that render a web page using a browser engine such as WebKit or Chromium. This API has now gained support for touchscreen events, and this has been implemented in various plugins such as the GTK, Qt, XV, and x11 video sinks as well as the wpevideosrc element. GStreamer CUDA integration - New gst-cuda library - integration with D3D11 and NVIDIA dGPU NVMM elements - new cudaconvertscale element GStreamer Direct3D11 integration - New gst-d3d11 public library - gst-d3d11 library is not integrated with GStreamer documentation system yet. Please refer to the examples - d3d11screencapture: Add Windows Graphics Capture API based capture mode, including Win32 application window capturing - d3d11videosink and d3d11convert can support flip/rotation and crop meta - d3d11videosink: New emit-present property and present signal so that applications can overlay an image on Direct3D11 swapchain’s backbuffer via Direct3D/Direct2D APIs. See also C++ and Rust examples - d3d11compositor supports YUV blending/composing without intermediate RGB(A) conversion to improve performance - Direct3D11 video decoders are promoted to GST_RANK_PRIMARY or higher, except for the MPEG2 decoder H.264/H.265 timestamp correction elements - Muxers are often picky and need proper PTS/DTS timestamps set on the input buffers, but that can be a problem if the encoded input media stream comes from a source that doesn’t provide proper signalling of DTS, such as is often the case for RTP, RTSP and WebRTC streams or Matroska container files. Theoretically parsers should be able to fix this up, but it would probably require fairly invasive changes in the parsers, so two new elements h264timestamper and h265timestamper bridge the gap in the meantime and can reconstruct missing PTS/DTS. Easy sender timestamp reconstruction for RTP and RTSP - it was always possible to reconstruct and retrieve the original RTP sender timestamps in GStreamer, but required a fair bit of understanding of the internal mechanisms and the right property configuration and clock setup. - rtspsrc and rtpjitterbuffer gained a new “add-reference-timestamp-meta” property that if set puts the original absolute reconstructed sender timestamps on the output buffers via a meta. This is particularly useful if the sender is synced to an NTP clock or PTP clock. The original sender timestamps are either based on the RTCP NTP times, NTP RTP header extensions (RFC6051) or RFC7273-style clock signalling. Qt6 support - new qml6glsink element for Qt6 similar to the existing Qt5 element. Matching source and overlay elements will hopefully follow in the near future. OpenGL + Video library enhancements - Support for new video formats (NV12_4L4, NV12_16L32S, NV12_8L128, NV12_10BE_8L128) and dmabuf import in more formats (Y410, Y212_LE, Y212_BE, Y210, NV21, NV61) - Improved support for tiled formats with arbitrary tile dimensions, as needed by certain hardware decoders/encoders - glvideomixer: New “crop-left,”crop-right, “crop-top” and “crop-bottom” pad properties for cropping inputs - OpenGL support for gst_video_sample_convert(): - Used for video snapshotting and thumbnailing, to convert buffers retrieved from appsinks or sink “last-sample” properties in JPG/PNG thumbnails. - This function can now take samples and buffers backed by GL textures as input and will automatically plug a gldownload element in that case. High bit-depth support (10, 12, 16 bits per component value) improvements - compositor can now handle any supported input format and also mix high-bitdepth (10-16 bit) formats (naively) - videoflip has gained support for higher bit depth formats. - vp9enc, vp9dec now support 12-bit formats and also 10-bit 4:4:4 WebRTC - Allow insertion of bandwidth estimation elements e.g. for Google Congestion Control (GCC) support - Initial support for sending or receiving simulcast streams - Support for asynchronous host resolution for STUN/TURN servers - GstWebRTCICE was split into base classes and implementation to make it possible to plug custom ICE implementations - webrtcsink: batteries-included WebRTC sender (Rust) - whipsink: WebRTC HTTP ingest (WHIP) to a MediaServer (Rust) - whepsrc: WebRTC HTTP egress (WHEP) from a MediaServer (Rust) - Many other improvements and bug fixes New HLS, DASH and MSS adaptive streaming clients A new set of “adaptive demuxers” to support HLS, DASH and MSS adaptive streaming protocols has been added. They provide improved performance, new features and better stream compatibility compared to the previous elements. These new elements require a “streams-aware” pipeline such as playbin3, uridecodebin3 or urisourcebin. The previous elements’ design prevented implementing several use-cases and fixing long-standing issues. The new elements were re-designed from scratch to tackle those: - Scheduling Only 3 threads are present, regardless of the number of streams selected. One in charge of downloading fragments and manifests, one in charge of outputting parsed data downstream, and one in charge of scheduling. This improves performance, resource usage and latency. - Better download control The elements now directly control the scheduling and download of manifests and fragments using libsoup directly instead of depending on external elements for downloading. - Stream selection, only the selected streams are downloaded. This improves bandwith usage. Switching stream is done in such a way to ensure there are no gaps, meaning the new stream will be switched to only once enough data for it has been downloaded. - Internal parsing, the downloaded streams are parsed internally. This allows the element to fully respect the various specifications and offer accurate buffering, seeking and playback. This is especially important for HLS streams which require parsing for proper positioning of streams. - Buffering and adaptive rate switching, the new elements handle buffering internally which allows them to have a more accurate visibility of which bandwith variant to switch to. Playbin3, Decodebin3, UriDecodebin3, Parsebin improvements The “new” playback elements introduced in 1.18 (playbin3 and its various components) have been refactored to allow more use-cases and improve performance. They are no longer considered experimental, so applications using the legacy playback elements (playbin and (uri)decodebin) can migrate to the new components to benefit from these improvements. - Gapless The “gapless” feature allows files and streams to be fetched, buffered and decoded in order to provide a “gapless” output. This feature has been refactored extensively in the new components: - A single (uri)decodebin3 (and therefore a single set of decoders) is used. This improves memory and cpu usage, since on identical codecs a single decoder will be used. - The “next” stream to play will be pre-rolled “just-in-time” thanks to the buffering improvements in urisourcebin (see below) - This feature is now handled at the uridecodebin3 level. Applications that wish to have a “gapless” stream and process it (instead of just outputting it, for example for transcoding, retransmission, …) can now use uridecodebin3 directly. Note that a streamsynchronizer element is required in that case. - Buffering improvements The urisourcebin element is in charge of fetching and (optionally) buffering/downloading the stream. It has been extended and improved: - When the parse-streams property is used (by default in uridecodebin3 and playbin3), compatible streams will be demuxed and parsed (via parsebin) and buffering will be done on the elementary streams. This provides a more accurate handling of buffering. Previously buffering was done on a best-effort basis and was mostly wrong (i.e. downloading more than needed). - Applications can use urisourcebin with this property as a convenient way of getting elementary streams from a given URI. - Elements can handle buffering themselves (such as the new adaptive demuxers) by answering the GST_QUERY_BUFFERING query. In that case urisourcebin will not handle it. - Stream Selection Efficient stream selection was previously only possible within decodebin3. The downside is that this meant that upstream elements had to provide all the streams from which to chose from, which is inefficient. With the addition of the GST_QUERY_SELECTABLE query, this can now be handled by elements upstream (i.e. sources) - Elements that can handle stream selection internally (such as the new adaptive demuxer elements) answer that query, and handle the stream selection events themselves. - In this case, decodebin3 will always process all streams that are provided to it. - Instant URI switching This new feature allows switching URIs “instantly” in playbin3 (and uridecodebin3) without having to change states. This mimics switching channels on a television. - If compatible, decoders will be re-used, providing lower latency/cpu/memory than by switching states. - This is enabled by setting the instant-uri property to true, setting the URI to switch to immediately, and then disabling the instant-uri property again afterwards. - playbin3, decodebin3, uridecodebin3, parsebin, and urisourcebin are no longer experimental - They were originally marked as ‘technology preview’ but have since seen extensive usage in production settings, so are considered ready for general use now. Fraunhofer AAC audio encoder HE-AAC and AAC-LD profile support - fdkaacenc: - Support for encoding to HE-AACv1 and HE-AACv2 profile - Support for encoding to AAC Low Delay (LD) profile - Advanced bitrate control options via new “rate-control”, “vbr-preset”, “peak-bitrate”, and “afterburner” properties RTP rapid synchronization support in the RTP stack (RFC6051) RTP provides several mechanisms how streams can be synchronized relative to each other, and how absolute sender times for RTP packets can be obtained. One of these mechanisms is via RTCP, which has the disadvantage that the synchronization information is only distributed out-of-band and usually some time after the start. GStreamer’s RTP stack, specifically the rtpbin, rtpsession and rtpjitterbuffer elements, now also have support for retrieving and sending the same synchronization information in-band via RTP header extensions according to RFC6051 (Rapid Synchronisation of RTP Flows). Only 64-bit timestamps are supported currently. This provides per packet synchronization information from the very beginning of a stream and allows accurate inter-stream, and (depending on setup) inter-device, synchronization at the receiver side. ONVIF XML Timed Metadata support The ONVIF standard implemented by various security cameras also specifies a format for timed metadata that is transmitted together with the audio/video streams, usually over RTSP. Support for this timed metadata is implemented in the MP4 demuxer now as well as the new fragmented MP4 muxer and the new non-fragmented MP4 muxer from the GStreamer Rust plugins. Additionally, the new onvif plugin ‒ which is part of the GStreamer Rust plugins ‒ provides general elements for handling the metadata and e.g. overlaying certain parts of it over a video stream. As part of this support for absolute UTC times was also implemented according to the requirements of the ONVIF standards in the corresponding elements. MP3 gapless playback support While MP3 can probably considered a legacy format at this point, a new feature was added with this release. When playing back plain MP3 files, i.e. outside a container format, switches between files can now be completely gapless if the required metadata is provided inside the file. There is no standardized metadata for this, but the LAME MP3 encoder writes metadata that can be parsed by the mpegaudioparse element now and forwarded to decoders for ensuring removal of padding samples at the front and end of MP3 files. “force-live” property for audio + video aggregators This is a quality of life fix for playout and streaming applications where it is common to have audio and video mixer elements that should operate in live mode from the start and produce output continuously. Often one would start a pipeline without any inputs hooked up to these mixers in the beginning, and up until now there was no way to easily force these elements into live mode from the start. One would have to add an initial live video or audio test source as dummy input to achieve this. The new “force-live” property makes these audio and video aggregators start in live mode without the need for any dummy inputs, which is useful for scenarios where inputs are only added after starting the pipeline. This new property should usually be used in connection with the “min-upstream-latency” property, i.e. you should always set a non-0 minimum upstream latency then. This is now supported in all GstAudioAggregator and GstVideoAggregator subclasses such as audiomixer, audiointerleave, compositor, glvideomixer, d3d11compositor, etc. New elements and plugins - new cudaconvertscale element that can convert and scale in one pass - new gtkwaylandsink element based on gtksink, but similar to waylandsink and uses Wayland APIs directly instead of rendering with Gtk/Cairo primitives. This approach is only compatible with Gtk3, and like gtksink this element only supports Gtk3. - new h264timestamper and h265timestamper elements to reconstruct missing pts/dts from inputs that might not provide them such as e.g. RTP/RTSP/WebRTC inputs (see above) - mfaacdec, mfmp3dec: Windows MediaFoundation AAC and MP3 decoders - new msdkav1enc AV1 video encoder element - new nvcudah264enc, nvcudah265enc, nvd3d11h264enc, and nvd3d11h265enc NVIDIA GPU encoder elements to support zero-copy encoding, via CUDA and Direct3D11 APIs, respectively - new nvautogpuh264enc and nvautogpuh265enc NVIDIA GPU encoder elements: The auto GPU elements will automatically select a target GPU instance in case multiple NVIDIA desktop GPUs are present, also taking into account the input memory. On Windows CUDA or Direct3D11 mode will be determined by the elements automatically as well. Those new elements are useful if target GPU and/or API mode (either CUDA or Direct3D11 in case of Windows) is undeterminable from the encoder point of view at the time when pipeline is configured, and therefore lazy target GPU and/or API selection are required in order to avoid unnecessary memory copy operations. - new nvav1dec AV1 NVIDIA desktop GPU decoder element - new qml6glsink element to render video with Qt6 - qsv: New Intel OneVPL/MediaSDK (a.k.a Intel Quick Sync) based decoder and encoder elements, with gst-d3d11 (on Windows) and gst-va (on Linux) integration - Support multi-GPU environment, for example, concurrent video encoding using Intel iGPU and dGPU in a single pipeline - H.264 / H.265 / VP9 and JPEG decoders - H.264 / H.265 / VP9 / AV1 / JPEG encoders with dynamic encoding bitrate update - New plugin does not require external SDK for building on Windows - vulkanoverlaycompositor: new vulkan overlay compositor element to overlay upstream GstVideoOverlayCompositonMeta onto the video stream. - vulkanshaderspv: performs operations with SPIRV shaders in Vulkan - win32ipcvideosink, win32ipcvideosrc: new shared memory videosrc/sink elements for Windows - wicjpegdec, wicpngdec: Windows Imaging Component (WIC) based JPEG and PNG decoder elements. - Many exciting new Rust elements, see Rust section below New element features and additions - audioconvert: Dithering now uses a slightly slower, less biased PRNG which results in better quality output. Also dithering can now be enabled via the new “dithering-threshold” property for target bit depths of more than 20 bits. - av1enc: Add “keyframe-max-dist” property for controlling max distance between keyframes, as well as “enc-pass”, “keyframe-mode”, “lag-in-frames” and “usage-profile” properties. - cccombiner: new “output-padding” property - decklink: Add support for 4k DCI, 8k/UHD2 and 8k DCI modes - dvbsubenc: Support for >SD resolutions is working correctly now. - fdkaacenc: Add HE-AAC / HE-AACv2 profile support - glvideomixer: New “crop-left,”crop-right, “crop-top” and “crop-bottom” pad properties for cropping inputs - gssink: new ‘content-type’ property. Useful when one wants to upload a video as video/mp4 instead of ’video/quicktime` for example. - jpegparse: Rewritten using the common parser library - msdk: - new msdkav1enc AV1 video encoder element - msdk decoders: Add support for Scaler Format Converter (SFC) on supported Intel platforms for hardware accelerated conversion and scaling - msdk encoders: support import of dmabuf, va memory and D3D11 memory - msdk encoders: add properties for low delay bitrate control and max frame sizes for I/P frames - msdkh264enc, msdkh265enc: more properties to control intra refresh - note that on systems with multi GPUs the Windows D3D11 integration might only work reliably if the Intel GPU is the primary GPU - mxfdemux: Add support for Canon XF-HEVC - openaptx: Support the freeaptx library - qroverlay: - new “qrcode-case-sensitive” property allows encoding case sensitive strings like wifi SSIDs or passwords. - added the ability to pick up data to render from an upstream-provided custom GstQROverlay meta - qtdemux: Add support for ONVIF XML Timed MetaData and AVC-Intra video - rfbsrc now supports the uri handler interface, so applications can use RFB/VNC sources in uridecodebin(3) and playbin, with e.g. rfb://:password@10.1.2.3:5903?shared=1 - rtponviftimestamp: Add support for using reference timestamps - rtpvp9depay now has the same keyframe-related properties as rtpvp8depay and rtph264depay: “request-keyframe” and “wait-for-keyframe” - rtspsrc: Various RTSP servers are using invalid URL operations for constructing the control URL. Until GStreamer 1.16 these worked correctly because GStreamer was just appending strings itself to construct the control URL, but starting version 1.18 the correct URL operations were used. With GStreamer 1.22, rtspsrc now first tries with the correct control URL and if that fails it will retry with the wrongly constructed control URL to restore support for such servers. - rtspsrc and rtpjitterbuffer gained a new “add-reference-timestamp-meta” property that makes them put the unmodified original sender timestamp on output buffers for NTP or PTP clock synced senders - srtsrc, srtsink: new “auto-reconnect” property to make it possible to disable automatic reconnects (in caller mode) and make the elements post an error immediately instead; also stats improvements - srtsrc: new “keep-listening” property to avoid EOS on disconnect and keep the source running while it waits for a new connection. - videocodectestsink: added YUV 4:2:2 support - wasapi2src: Add support for process loopback capture - wpesrc: Add support for modifiers in key/touch/pointer events Plugin and library moves - The xingmux plugin has been moved from gst-plugins-ugly into gst-plugins-good. - The various Windows directshow plugins in gst-plugins-bad have been unified into a single directshow plugin. Plugin removals - The dxgiscreencapsrc element has been removed, use d3d11screencapturesrc instead Miscellaneous API additions - GST_AUDIO_FORMAT_INFO_IS_VALID_RAW() and GST_VIDEO_FORMAT_INFO_IS_VALID_RAW() can be used to check if a GstAudioFormatInfo or GstVideoFormatInfo has been initialised to a valid raw format. - Video SEI meta: new GstVideoSEIUserDataUnregisteredMeta to carry H.264 and H.265 metadata from SEI User Data Unregistered messages. - vulkan: Expose gst_vulkan_result_to_string() Miscellaneous performance, latency and memory optimisations - liborc 0.4.33 adds support for aarch64 (64-bit ARM) architecture (not enabled by default on Windows yet though) and improvements for 32-bit ARM and should greatly enhance performance for certain operations that use ORC. - as always there have been plenty of performance, latency and memory optimisations all over the place. Miscellaneous other changes and enhancements - the audio/video decoder base classes will not consider decoding errors a hard error by default anymore but will continue trying to decode. Previously more than 10 consecutive errors were considered a hard error but this caused various partially broken streams to fail. The threshold is configurable via the “max-errors” property. - compatibility of the GStreamer PTP clock implementation with different PTP server implementations was improved, and synchronization is achieved successfully in various scenarios that failed before. Tracing framework and debugging improvements New tracers - buffer-lateness: Records lateness of buffers and the reported latency for each pad in a CSV file. Comes with a script for visualisation. - pipeline-snapshot: Creates a .dot file of all pipelines in the application whenever requested via SIGUSR1 (on UNIX systems) - queue-levels: Records queue levels for each queue in a CSV file. Comes with a script for visualisation. Debug logging system improvements - new log macros GST_LOG_ID, GST_DEBUG_ID, GST_INFO_ID, GST_WARNING_ID, GST_ERROR_ID, and GST_TRACE_ID allow passing a string identifier instead of a GObject. This makes it easier to log non-gobject-based items and also has performance benefits. Tools - gst-play-1.0 gained a --no-position command line option to suppress position/duration queries, which can be useful to reduce debug log noise. GStreamer FFMPEG wrapper - Fixed bitrate management and timestamp inaccuracies for video encoders - Fix synchronization issues and errors created by the (wrong) forwarding of upstream segment events by ffmpeg demuxers. - Clipping meta support for gapless mp3 playback GStreamer RTSP server - Add RFC5576 Source-specific media attribute to the SDP media for signalling the CNAME - Add support for adjusting request response on pipeline errors - Give the application the possibility to adjust the error code when responding to a request. For that purpose the pipeline’s bus messages are emitted to subscribers through a “handle-message” signal. The subscribers can then check those messages for errors and adjust the response error code by overriding the virtual method GstRTSPClientClass::adjust_error_code(). - Add gst_rtsp_context_set_token() method to make it possible to set the RTSPToken on some RTSPContext from bindings such as the Python bindings. - rtspclientsink gained a “publish-clock-mode” property to configure whether the pipeline clock should be published according to RFC7273 (RTP Clock Source Signalling), similar to the same API on GstRTSPMedia. GStreamer VA-API support - Development activity has shifted towards the new va plugin, with gstreamer-vaapi now basically in maintenance-only mode. Most of the below refers to the va plugin (not gstreamer-vaapi). - new gst-va library for GStreamer VA-API integration - vajpegdec: new JPEG decoder - vah264enc, vah265enc: new H.264/H.265 encoders - vah264lpenc, vah265lpenc: new low power mode encoders - vah265enc: Add extended formats support such as 10/12 bits, 4:2:2 and 4:4:4 - Support encoder reconfiguration - vacompositor: Add new compositor element using the VA-API VPP interface - vapostproc: - new “scale-method” property - Process HDR caps if supported - parse video orientation from tags - vaapipostproc: Enable the use of DMA-Buf import and export (gstreamer-vaapi) GStreamer Video4Linux2 support - Added support for Mediatek Stateless CODEC (VP8, H.264, VP9) - Stateless H.264 interlaced decoder support - Stateless H.265 decoder support - Stateful decoder support for driver resolution change events - Stateful decoding support fixes for NXP/Amphion driver - Support for hardware crop in v4l2src - Conformance test improvement for stateful decoders - Fixes for Raspberry Pi CODEC GStreamer OMX - There were no changes in this module GStreamer Editing Services and NLE - Handle compositors that are bins around the actual compositor implementation (like glvideomixers which wraps several elements) - Add a mode to disable timeline editing API so the user can be in full control of its layout (meaning that the user is responsible for ensuring its validity/coherency) - Add a new fade-in transition type - Add support for non-1/1 PAR source videos - Fix frame accuracy when working with very low framerate streams GStreamer validate - Clean up and stabilize API so we can now generate rust bindings - Enhance the appsrc-push action type allowing to find tune the buffers more in details - Add an action type to verify currently configured pad caps - Add a way to run checks from any thread after executing a ‘wait’ action. This is useful when waiting on a signal and want to check the value of a property right when it is emited for example. GStreamer Python Bindings - Add a Gst.init_python() function to be called from plugins which will initialise everything needed for the GStreamer Python bindings but not call Gst.init() again since this will have been called already. - Add support for the GstURIHandlerInterface that allows elements to advertise what URI protocols they support. GStreamer C# Bindings - Fix AppSrc and AppSink constructors - The C# bindings have yet to be updated to include new 1.22 API, which requires improvements in various places in the bindings / binding generator stack. See issue #1718 in GitLab for more information and to track progress. GStreamer Rust Bindings and Rust Plugins The GStreamer Rust bindings are released separately with a different release cadence that’s tied to gtk-rs, but the latest release has already been updated for the new GStreamer 1.22 API. Check the bindings release notes for details of the changes since 0.18, which was released around GStreamer 1.20. gst-plugins-rs, the module containing GStreamer plugins written in Rust, has also seen lots of activity with many new elements and plugins. A list of all Rust plugins and elements provided with the 0.9 release can be found in the repository. - 33% of GStreamer commits are now in Rust (bindings + plugins), and the Rust plugins module is also where most of the new plugins are added these days. - The Rust plugins are now shipped as part of the Windows MSVC + macOS binary packages. See below for the list of shipped plugins and the status of Rust support in cerbero. - The Rust plugins are also part of the documentation on the GStreamer website now. - Rust plugins can be used from any programming language. To the outside they look just like a plugin written in C or C++. New Rust plugins and elements - rtpav1pay / rtpav1depay: RTP (de)payloader for the AV1 video codec - gtk4paintablesink: a GTK4 video sink that provides a GdkPaintable for rendering a video in any place inside a GTK UI. Supports zero-copy rendering via OpenGL on Linux and macOS. - ndi: source, sink and device provider for NewTek NDI protocol - onvif: Various elements for parsing, RTP (de)payloading, overlaying of ONVIF timed metadata. - livesync: Element for converting a live stream into a continuous stream without gaps and timestamp jumps while preserving live latency requirements. - raptorq: Encoder/decoder elements for the RaptorQ FEC mechanism that can be used for RTP streams (RFC6330). WebRTC elements - webrtcsink: a WebRTC sink (batteries included WebRTC sender with specific signalling) - whipsink: WebRTC HTTP ingest (WHIP) to MediaServer - whepsrc: WebRTC HTTP egress (WHEP) from MediaServer - rtpgccbwe: RTP bandwidth estimator based on the Google Congestion Control algorithm (GCC), used by webrtcsink Amazon AWS services - awss3src / awss3sink: A source and sink element to talk to the Amazon S3 object storage system. - awss3hlssink: A sink element to store HLS streams on Amazon S3. - awstranscriber: an element wrapping the AWS Transcriber service. - awstranscribeparse: an element parsing the packets of the AWS Transcriber service. Video Effects (videofx) - roundedcorners: Element to make the corners of a video rounded via the alpha channel. - colordetect: A pass-through filter able to detect the dominant color(s) on incoming frames, using color-thief. - videocompare: Compare similarity of video frames. The element can use different hashing algorithms like Blockhash, DSSIM, and others. New MP4 muxer + Fragmented MP4 muxer - isofmp4mux, cmafmux, dashmp4mux, onviffmp4mux: New fragmented MP4/ISOBMFF/CMAF muxer for generating e.g. DASH/HLS media fragments. - isomp4mux, onvifmp4mux: New non-fragmented, normal MP4 muxer. Both plugins provides elements that replace the existing qtmux/mp4mux element from gst-plugins-good. While not feature-equivalent yet, the new codebase and using separate elements for the fragment and non-fragmented case allows for easier extensability in the future. Cerbero Rust support - Starting this release, cerbero has support for building and shipping Rust code on Linux, Windows (MSVC) and macOS. The Windows (MSVC) and macOS binaries also ship the GStreamer Rust plugins in this release. Only dynamic plugins are built and shipped currently. - Preliminary support for Android, iOS and Windows (MinGW) exists but more work is needed. Check the tracker issue for more details about future work. - The following plugins are included currently: audiofx, aws, cdg, claxon, closedcaption, dav1d, fallbackswitch, ffv1, fmp4, gif, hlssink3, hsv, json, livesync, lewton, mp4, ndi, onvif, rav1e, regex, reqwest, raptorq, png, rtp, textahead, textwrap, threadshare, togglerecord, tracers, uriplaylistbin, videofx, webrtc, webrtchttp. Build and Dependencies - meson 0.62 or newer is required - GLib >= 2.62 is now required (but GLib >= 2.64 is strongly recommended) - libnice >= 0.1.21 is now required and contains important fixes for GStreamer’s WebRTC stack. - liborc >= 0.4.33 is recommended for 64-bit ARM support and 32-bit ARM improvements - onnx: OnnxRT >= 1.13.1 is now required - openaptx: can now be built against libfreeaptx - opencv: allow building against any 4.x version - shout: libshout >= 2.4.3 is now required - gstreamer-vaapi’s Meson build options have been switched from a custom combo type (yes/no/auto) to the built-in Meson feature type (enabled/disabled/auto) - The GStreamer Rust plugins module gst-plugins-rs is now considered an essential part of the GStreamer plugin offering and packagers and distributors are strongly encouraged to package and ship those plugins alongside the existing plugin modules. - we now make use of Meson’s install tags feature which allows selective installation of installl components and might be useful for packagers. Monorepo build (gst-build) - new “orc-source” build option to allow build against a system-installed liborc instead of forcing the use of orc as a subproject. - GStreamer command line tools can now be linked to the gstreamer-full library if it’s built Cerbero Cerbero is a meta build system used to build GStreamer plus dependencies on platforms where dependencies are not readily available, such as Windows, Android, iOS, and macOS. General improvements - Rust support was added for all support configurations, controlled by the rust variant; see above for more details - All pkgconfig files are now reliably relocatable without requiring pkg-config --define-prefix. This also fixes statically linking with GStreamer plugins using the corresponding pkgconfig files. - New documentation on how to build a custom GStreamer repository using Cerbero, please see the README - HTTPS certificate checking is enabled for downloads on all platforms now - Fetching now automatically retries on error for robustness against transient errors - Support for building the new Qt6 plugin was added - pkgconfig files for various recipes were fixed - Several recipes were updated to newer versions - New plugins: adaptivedemux2 aes codectimestamper dav1d - New libraries: cuda webrtcnice macOS / iOS - Added support for running Cerbero on ARM64 macOS - GStreamer.framework and all libraries in it are now relocatable, which means they use LC_RPATH entries to find dependencies instead of using an absolute path. If you link to GStreamer using the pkgconfig files, no action is necessary. However, if you use the framework directly or link to the libraries inside the framework by hand, then you need to pass -Wl,-rpath, to the linker. - Apple bitcode support was dropped, since Apple has deprecated it - macOS installer now correctly advertises support for both x86_64 and arm64 - macOS framework now ships the gst-rtsp-server-1.0 library - Various fixes were made to make static linking to gstreamer libraries and plugins work correctly on macOS - When statically linking to the applemedia plugin using Xcode 13, you will need to pass -fno-objc-msgsend-selector-stubs which works around a backwards-incompatible change in Xcode 14. This is not required for the rest of GStreamer at present, but will be in the future. - macOS installer now shows the GStreamer logo correctly Windows - MSVC is now required by default on Windows, and the Visual Studio variant is enabled by default - To build with MinGW, use the mingw variant - Visual Studio props files were updated for newer Visual Studio versions - Visual Studio 2015 support was dropped - MSYS2 is now supported as the base instead of MSYS. Please see the README for more details. Some advantages include: - Faster build times, since parallel make works - Faster bootstrap, since some tools are provided by MSYS2 - Other speed-ups due to using MSYS2 tools instead of MSYS - Faster download by using powershell instead of hand-rolled Python code - Many recipes were ported from Autotools to Meson, speeding up the build - Universal Windows Platform is no longer supported, and binaries are no longer shipped for it - New documentation on how to force a specific Visual Studio installation in Cerbero, please see the README - New plugins: qsv wavpack directshow amfcodec wic win32ipc - New libraries: d3d11 Windows MSI installer - Universal Windows Platform prebuilt binaries are no longer available Linux - Various fixes for RHEL/CentOS 7 support - Added support for running on Linux ARM64 Android - Android support now requires Android API version 21 (Lollipop) - Support for Android Gradle plugin 7.2 Platform-specific changes and improvements Android - Android SDK 21 is required now as minimum SDK version - androidmedia: Add H.265 / HEVC video encoder mapping - Implement JNI_OnLoad() to register static plugins etc. automatically in case GStreamer is loaded from Java using System.loadLibrary(), which is also useful for the gst-full deployment scenario. Apple macOS and iOS - The GLib version shipped with the GStreamer binaries does not initialize an NSApp and does not run a NSRunLoop on the main thread anymore. This was a custom GLib patch and caused it to behave different from the GLib shipped by Homebrew or anybody else. The change was originally introduced because various macOS APIs require a NSRunLoop to run on the main thread to function correctly but as this change will never get merged into GLib and it was reverted for 1.22. Applications that relied on this behaviour should move to the new gst_macos_main() function, which also does not require the usage of a GMainLoop. See e.g. gst-play.c for an example for the usage of gst_macos_main(). - GStreamer.framework and all libraries in it are now relocatable, which means they use LC_RPATH entries to find dependencies instead of using an absolute path. If you link to GStreamer using the pkgconfig files, no action is necessary. However, if you use the framework directly or link to the libraries inside the framework by hand, then you need to pass -Wl,-rpath, to the linker. - avfvideosrc: Allow specifying crop coordinates during screen capture - vtenc, vtdec: H.265 / HEVC video encoding + decoding support - osxaudiosrc: Support a device as both input and output - osxaudiodeviceprovider now probes devices more than once to determine if the device can function as both an input AND and output device. Previously, if the device provider detected that a device had any output capabilities, it was treated solely as an Audio/Sink. This caused issues for devices that have both input and output capabilities (for example, USB interfaces for professional audio have both input and output channels). Such devicesare now listed as both an Audio/Sink as well as an Audio/Source. - osxaudio: support hidden devices on macOS - These are devices that will not be shown in the macOS UIs and that cannot be retrieved without having the specific UID of the hidden device. There are cases when you might want to have a hidden device, for example when having a virtual speaker that forwards the data to a virtual hidden input device from which you can then grab the audio. The blackhole project supports these hidden devices and this change provides a way that if the device id is a hidden device it will use it instead of checkinf the hardware list of devices to understand if the device is valid. Windows - win32ipcvideosink, win32ipcvideosrc: new shared memory videosrc/sink elements - wasapi2: Add support for process loopback capture for a specified PID (requires Windows 11/Windows Server 2022) - The Windows universal UWP build is currently non-functional and will need updating after the recent GLib upgrade. It is unclear if anyone is using these binaries, so if you are please make yourself known. - wicjpegdec, wicpngdec: Windows Imaging Component (WIC) based JPEG and PNG decoder elements. - mfaacdec, mfmp3dec: Windows MediaFoundation AAC and MP3 decoders - The uninstalled development environment supports PowerShell 7 now Linux - Improved design for DMA buffer sharing and modifier handling for hardware-accelerated video decoders/encoders/filters and capture/rendering on Linux and Linux-like system. - kmssink - new “fd” property which allows an application to provide their own opened DRM device fd handle to kmssink. That way an application can lease multiple fd’s from a DRM master to display on different CRTC outputs at the same time with multiple kmssink instances, for example. - new “skip-vsync” property to achieve full framerate with legacy emulation in drivers. - HDR10 infoframe support - va plugin and gstreamer-vaapi improvements (see above) - waylandsink: Add “rotate-method” property and “render-rectangle” property - new gtkwaylandsink element based on gtksink, but similar to waylandsink and uses Wayland APIs directly instead of rendering with Gtk/Cairo primitives. This approach is only compatible with Gtk3, and like gtksink this element only supports Gtk3. Documentation improvements - The GStreamer Rust plugins are now included and documented in the plugin documentation. Possibly Breaking Changes - the Opus audio RTP payloader and depayloader no longer accept the lower case encoding-format=multiopus but instead produce and accept only the upper case variant encoding-format=MULTIOPUS, since those should always be upper case in GStreamer (caps fields are always case sensitive). This should hopefully only affect applications where RTP caps are set manually and multi-channel audio (>= 3 channels) is used. - wpesrc: the URI handler protocols changed from wpe:// and web:// to web+http://, web+https://, and web+file:// which means URIs are RFC 3986 compliant and the source can simply strip the prefix from the protocol. - The Windows screen capture element dxgiscreencapsrc has been removed, please use d3d11screencapturesrc instead. - On Android the minimum supported Android API version is now version 21 and has been increased from 16. - On macOS, the GLib version shipped with the GStreamer binaries will no longer initialize an NSApp or run an NSRunLoop on the main thread. See macOS/iOS section above for details. - decklink: The decklink plugin is now using the 12.2.2 version of the SDK and will not work with drivers older than version 12. - On iOS Apple Bitcode support was removed from the binaries. This feature is deprecated since XCode 14 and not used on the App Store anymore. - The MP4/Matroska/WebM muxers now require the “stream-format” to be provided as part of the AV1 caps as only the original “obu-stream” format is supported in these containers and not the “annexb” format. Known Issues - The Windows UWP build in Cerbero needs fixing after the recent GLib upgrade (see above) - The C# bindings have not been updated to include new 1.22 API yet (see above) Statistics - 4072 commits - 2224 Merge Requests - 716 Issues - 200+ Contributors - ~33% of all commits and Merge Requests were in Rust modules - 4747 files changed - 469633 lines added - 209842 lines deleted - 259791 lines added (net) Contributors Ádám Balázs, Adam Doupe, Adrian Fiergolski, Adrian Perez de Castro, Alba Mendez, Aleix Conchillo Flaqué, Aleksandr Slobodeniuk, Alicia Boya García, Alireza Miryazdi, Andoni Morales Alastruey, Andrew Pritchard, Arun Raghavan, A. Wilcox, Bastian Krause, Bastien Nocera, Benjamin Gaignard, Bill Hofmann, Bo Elmgreen, Boyuan Zhang, Brad Hards, Branko Subasic, Bruce Liang, Bunio FH, byran77, Camilo Celis Guzman, Carlos Falgueras García, Carlos Rafael Giani, Célestin Marot, Christian Wick, Christopher Obbard, Christoph Reiter, Chris Wiggins, Chun-wei Fan, Colin Kinloch, Corentin Damman, Corentin Noël, Damian Hobson-Garcia, Daniel Almeida, Daniel Morin, Daniel Stone, Daniels Umanovskis, Danny Smith, David Svensson Fors, Devin Anderson, Diogo Goncalves, Dmitry Osipenko, Dongil Park, Doug Nazar, Edward Hervey, ekwange, Eli Schwartz, Elliot Chen, Enrique Ocaña González, Eric Knapp, Erwann Gouesbet, Evgeny Pavlov, Fabian Orccon, Fabrice Fontaine, Fan F He, F. Duncanh, Filip Hanes, Florian Zwoch, François Laignel, Fuga Kato, George Kiagiadakis, Guillaume Desmottes, Gu Yanjie, Haihao Xiang, Haihua Hu, Havard Graff, Heiko Becker, He Junyan, Henry Hoegelow, Hiero32, Hoonhee Lee, Hosang Lee, Hou Qi, Hugo Svirak, Ignacio Casal Quinteiro, Ignazio Pillai, Igor V. Kovalenko, Jacek Skiba, Jakub Adam, James Cowgill, James Hilliard, Jan Alexander Steffens (heftig), Jan Lorenz, Jan Schmidt, Jianhui Dai, jinsl00000, Johan Sternerup, Jonas Bonn, Jonas Danielsson, Jordan Petridis, Joseph Donofry, Jose Quaresma, Julian Bouzas, Junsoo Park, Justin Chadwell, Khem Raj, Krystian Wojtas, László Károlyi, Linus Svensson, Loïc Le Page, Ludvig Rappe, Marc Leeman, Marek Olejnik, Marek Vasut, Marijn Suijten, Mark Nauwelaerts, Martin Dørum, Martin Reboredo, Mart Raudsepp, Mathieu Duponchelle, Matt Crane, Matthew Waters, Matthias Clasen, Matthias Fuchs, Mengkejiergeli Ba, MGlolenstine, Michael Gruner, Michiel Konstapel, Mikhail Fludkov, Ming Qian, Mingyang Ma, Myles Inglis, Nicolas Dufresne, Nirbheek Chauhan, Olivier Crête, Pablo Marcos Oltra, Patricia Muscalu, Patrick Griffis, Paweł Stawicki, Peter Stensson, Philippe Normand, Philipp Zabel, Pierre Bourré, Piotr Brzeziński, Rabindra Harlalka, Rafael Caricio, Rafael Sobral, Rafał Dzięgiel, Raul Tambre, Robert Mader, Robert Rosengren, Rodrigo Bernardes, Rouven Czerwinski, Ruben Gonzalez, Sam Van Den Berge, Sanchayan Maity, Sangchul Lee, Sebastian Dröge, Sebastian Fricke, Sebastian Groß, Sebastian Mueller, Sebastian Wick, Sergei Kovalev, Seungha Yang, Seungmin Kim, sezanzeb, Sherrill Lin, Shingo Kitagawa, Stéphane Cerveau, Talha Khan, Taruntej Kanakamalla, Thibault Saunier, Tim Mooney, Tim-Philipp Müller, Tomasz Andrzejak, Tom Schuring, Tong Wu, toor, Tristan Matthews, Tulio Beloqui, U. Artie Eoff, Víctor Manuel Jáquez Leal, Vincent Cheah Beng Keat, Vivia Nikolaidou, Vivienne Watermeier, WANG Xuerui, Wojciech Kapsa, Wonchul Lee, Wu Tong, Xabier Rodriguez Calvar, Xavier Claessens, Yatin Mann, Yeongjin Jeong, Zebediah Figura, Zhao Zhili, Zhiyuaniu, مهدي شينون (Mehdi Chinoune), … and many others who have contributed bug reports, translations, sent suggestions or helped testing. Stable 1.22 branch After the 1.22.0 release there will be several 1.22.x bug-fix releases which will contain bug fixes which have been deemed suitable for a stable branch, but no new features or intrusive changes will be added to a bug-fix release usually. The 1.22.x bug-fix releases will be made from the git 1.22 branch, which will be a stable branch. 1.22.0 1.22.0 was originally released on 23 January 2023. 1.22.1 The first 1.22 bug-fix release (1.22.1) was released on 04 March 2023. This release only contains bugfixes and it should be safe to update from 1.22.0. Highlighted bugfixes in 1.22.1 - audio channel-mix: allow up to 64 channels (instead of up to 63 channels) - avfvideosrc: Don’t wait on main thread for permissions request - avvidenc: avoid generating inaccurate output timestamps, especially with variable framerate streams - AV1 video codec caps signalling improvements in various elements - codectimestamper: Fix timestamping on sequence update - d3d11overlaycompositor: fix texture width and height - d3d11videosink: Fix rendering on external handle - dashdemux2: fix seek operation taking a log time to finish for some streams - nvencoder: Fix B-frame encoding on Linux and min buffers in auto GPU mode - playbin3: fixing buffering for live pipelines - playbin: fix potential deadlock when stopping stream with subtitles visible - redenc: fix setting of extension ID for twcc - rtspsrc: improved compatibility with more broken RTSP servers - v4l2h264dec: Fix Raspberry Pi4 will not play video in application - vtdec: fix jittery playback of H.264 Level 4.1 movies in macOS - vtdec: Fix non-deterministic frame output after flushing seeks - vtenc: fix handling of interlaced ProRes on Apple M1 hardware - vtenc: don’t advertise ARGB/RGBA64 input caps on M1 Pro/Max with macOS <13 - wasapi2src: Fix loopback capture on Windows 10 Anniversary Update - tools: better handling of non-ASCII command line arguments on Windows - gst-libav: fix build against newer ffmpeg versions - gst-python: Use arch-specific install dir for gi overrides - cerbero: Fix setuptools site.py breakage in Python 3.11 - macOS packages: Fix broken binaries on macos < 11.0 - various bug fixes, memory leak fixes, and other stability and reliability improvements gstreamer - buffer: fix copy meta reference debug log formatting - bin: Don’t unlock unlocked mutex in gst_bin_remove_func() - pad: Don’t leak user_data in gst_pad_start_task() - aggregator: Always lock aggpad around update_time_level - inputselector: Avoid potential deadlock when shutting down, e.g. playbin with subtitles - multiqueue: Handle use-interleave latency live pipelines, fixing buffering for live pipelines in playbin3 - GstBaseSrc: fix transfer annotation for fixate() virtual method - GstBaseSrc, GstPushSrc: add nullable annotations to virtual methods - tools: Make sure UTF-8 encoded command line arguments on Windows gst-plugins-base - alsasink: Fix stall when going from PLAYING to NULL (stucked at PAUSED) with uac1 gadget - appsrc: Don’t chain up BaseSrc::negotiate() - audio: channel-mix: Fix channel count limit to be able to equal 64 - gldisplay: Mark gst_gl_display_create_context() other_context parameter as nullable - gldisplay: Remove unused code - gstglwindow_x11.c: Fix colormap leak - gl/cocoa: Return a strong ref to the parent GstGLContext - rtspconnection: Annotate RTSP message and RTSP events parameters correctly - sdp, typefind: Fix some annotations - sdp: gstmikey: gst_mikey_message_to_caps: extract ROC from first crypto session - subparse: Properly forward segment seqnum - uridecodebin: Set source element to READY before querying it - uritranscodebin: Fix unref of NULL - gst-play-1.0: Don’t force accurate seeking gst-plugins-good - adaptivedemux2: Fix buffering threshold initialization - dashdemux2: the seek operation takes a log time to finish for some streams - glvideomixer: Keep a reference to the underlying pad - qtdemux: Don’t emit GstSegment correcting start time when in MSE mode - qtdemux: Handle moov atom length=0 case by reading until the end - qtdemux, qtmux: Drop av1C version 0 parsing and implement version 1 parsing/writing - qtmux: Fix assertion on caps update - redenc: fix setting of extension ID for twcc - rtspsrc: Use the correct vfunc for the push-backchannel-sample action signal - rtpssrcdemux: set different stream-id on each src pad - udpsrc: GstSocketTimestampMessage only for SCM_TIMESTAMPNS - v4l2h264dec: Fix Raspberry Pi4 will not play video in application gst-plugins-bad - aom: Include stream-format and alignment in the AV1 caps - av1parser, h265parser: Fix some code defects - av1parser: Don’t consider unknown metadata OBUs a bitstream error - avfvideosrc: Don’t wait on main thread for permissions request - ccconverter: don’t debug a potentially freed filter caps - codectimestamper: Fix timestamping on sequence update - codecparsers: {h264, h265}bitwriter: Remove redundant condition checks - codecs: decoders: fail early if no input caps have been provided for all new decoder base classes - closedcaption: Don’t leak caps event - curlhttpsrc: Add curl anyauth option - d3d11overlaycompositor: fix texture width and height - d3d11videosink: Fix rendering on external handle - h265parse: Always set profile on src caps - msdkav1enc: fix the category for msdkav1enc debug - nvcodec: improve error reporting on plugin init - nvencoder: Fix b-frame encoding on Linux - nvencoder: Fix min buffers parameter of allocation query in auto GPU mode - nvvp9dec: Fix return value - qsvav1enc, amfav1enc: Set stream-format on caps - vtdec: Jittery playback of H.264 Level 4.1 movies in macOS (both x86_64 and arm64) - vtdec: Fix DPB size calculations not taking values from SPS into account - vtdec: Fix not waiting for async frames when flushing - vtenc: Disable ARGB/RGBA64 caps on M1 Pro/Max with macOS <13 - vtenc: Fix checking for certain CPU variants when running in VMs - vtenc: Disable HW acceleration for interlaced ProRes - va: Avoid the array index overflow when filling 8x8 scaling list. - va: Fix some code defects - vah265enc: Use helper to update properties. - vulkan: memory: Flush non coherent memory after write. - wasapi2src: Fix loopback capture on Windows 10 Anniversary Update - webrtcbin: small stats improvements - win32ipcutils: Add missing include - wpe: Logging fixes for the WebExtension gst-plugins-ugly - mpegpsdemux: Ignore DTS if PTS < DTS gst-libav - avauddec, avviddec: Free packet side data after usage - avviddec: change AV_CODEC_CAP_AUTO_THREADS->AV_CODEC_CAP_OTHER_THREADS to fix build against newer ffmpeg versions - Memory leak in ’ av_packet_add_side_data’ in /lib/x86_64-linux-gnu/libavcodec.so reading the file clock_odd_size_RLE_g1597902.avi - avvidenc: Don’t take ffmpeg timestamps verbatim but only use them to calculate DTS gst-rtsp-server - No changes gstreamer-vaapi - vaapi: Skip plugin pc file for shared plugins gstreamer-sharp - No changes gst-omx - No changes gst-python - gst-python: Use arch-specific install dir for gi overrides gst-editing-services - No changes gst-validate + gst-integration-testsuites - validate:scenario: sink refs when building - tests: Fix known issue definition location for unit tests and how we handle them in validate launcher - tests: mark elements_srtp.test_play test as flaky - Fix gstreamer-validate-1.0 dependency name - validate-scenario: fix g-i warning in annotation - validate: Fix gst_validate_execute_action annotation gst-examples - webrtc examples: Use webrtc.gstreamer.net - webrtc_sendrecv.py: Various fixes Development build environment - gst-env: Handle installing python modules to dist-packages - meson: Allow sysdeps to be forced as fallback subprojects - meson: Switch dav1d wrap to a tarball and update to dav1d 1.1.0 Cerbero build tool and packaging changes in 1.22.1 - macos: Fix broken binaries on macos < 11.0 - orc: Update pthread_jit_write_protect fix for macOS/iOS - dav1d: Update to 1.1.0 - libsrtp: update to v2.5.0 - rustup: Update to 1.25.2 - rust: Update to 1.67; cargo-c to 0.9.16 - cerbero: Don’t error out if bindir already exists - Fix setuptools site.py breakage in Python 3.11, bump gobject-introspection, bump windows image - cerbero: Retry if cargo update fails on macOS - gst-plugins-rs: Build glib/gio bindings with 2.74 API support Contributors to 1.22.1 Alessandro Bono, Arun Raghavan, Bart Van Severen, Carlos Falgueras García, Célestin Marot, David Svensson Fors, Edward Hervey, Enrique Ocaña González, Frank Dana, Guillaume Desmottes, He Junyan, James Hilliard, Jan Alexander Steffens (heftig), Jan Schmidt, Jordan Petridis, Mathieu Duponchelle, Matthew Waters, medithe, Mengkejiergeli Ba, Nicolas Beland, Nirbheek Chauhan, Patricia Muscalu, Pawel Stawicki, Philippe Normand, Piotr Brzeziński, Rajneesh Soni, Robert Rosengren, Sanchayan Maity, Sebastian Dröge, Seungha Yang, Simon Himmelbauer, Thibault Saunier, Tim-Philipp Müller, Tristan van Berkom, U. Artie Eoff, Víctor Manuel Jáquez Leal, Vivia Nikolaidou, Xuchen Yang, Yinhang Liu, … 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 1.22.1 - List of Merge Requests applied in 1.22.1 - List of Issues fixed in 1.22.1 1.22.2 The second 1.22 bug-fix release (1.22.2) was released on 11 April 2023. This release only contains bugfixes and it should be safe to update from 1.22.x. Highlighted bugfixes in 1.22.2 - avdec_h264: fix decoder deadlocks with FFmpeg 6.0 - rtspsrc: fix regression with URI protocols in OPTIONS requests for RTSP over TLS - rtspsrc: improved control url handling compatibility for broken servers - decklink: fix 10 bit RGB (r210) format auto detection for capture and fix playout if video caps are configured before audio caps - d3d11videosink: Fix tearing in case of fullscreen mode - playbin: fix deadlock when stopping stream with subtitles visible (even more) - typefinding: fix regression not detecting application/dash+xml in some corner cases - osxvideosink: fix broken aspect ratio and frame drawing region - decodebin3, parsebin: Improve elementary stream handling when decoders are not present and fix hang when removing a failing stream - urisourcebin: Propagate sticky events from parsebin, so that the STREAM_START event with the GstStream info is always available when pads get exposed - v4l2: Add support for YVU420M format; mark JPEG content as parsed - h264decoder, h265decoder: DPB bumping process and latency reporting fixes - Opus: Fix reading of extended channel config in MPEG-TS and fix missing sample rate when remuxing from RTP to Matroska - zxing: add support for building against zxing-c++ 2.0 - cerbero: Fix packaging of Rust plugins on Android; fix modern Gentoo distro detection - various bug fixes, memory leak fixes, and other stability and reliability improvements gstreamer - datetime: Return G_MAXFLOAT instead of G_MAXDOUBLE for no timezone offset - inputselector: Wake up streaming thread before PLAYING_TO_PAUSED transition - tools: fix potential crash when passing command-line options on Windows gst-plugins-base - alsasink: Fix for being stuck in stop_streaming_threads state - decodebin3: fix hang when removing a failing stream - gl: wayland: cleanup on close - parsebin: Improve elementary stream handling - playbin: fix deadlock when stopping stream with subtitles visible even more - sdp: Skip source-specific caps fields when creating an SDP media from caps - urisourcebin: Propagate sticky events from parsebin - urisourcebin: Activate pad before transferring sticky events - typefinding: fix failure to recognize application/dash+xml in some cases gst-plugins-good - osxvideosink: fix broken aspect ratio and frame drawing region - qtdemux: Fix seek adjustment with SNAP_AFTER flag - rtpopusdepay, matroskamux: Fix invalid rate while muxing Opus in Matroska - rtpmanager: twcc: Fix duplicate packet handling - rtsp: url: fix incorrect request URI scheme for TLS transport methods (regression) - rtspsrc: Consider “451: Parameter Not Understood” when handling broken control urls - rtspsrc: fix behavior change with URI protocols in OPTIONS requests - rtspsrc: Skip PTs with caps incompatible to the global caps - rtpjpegdepay: fix logic error when checking if an end of image (EOI) tag is present - v4l2: Add support for YVU420M format - v4l2: mark JPEG as parsed gst-plugins-bad - cea708overlay: fix HCR interpretation - d3d11bufferpool: Fix invalid access in debug print loop - d3d11compositor: Fix composition error on release_pad() - d3d11converter: Fix conversion backend selection - d3d11videosink: Fix tearing in case of fullscreen mode - d3d11bufferpool: Fix invalid access in debug print loop - d3d11window: fix memory leak - decklink: fix 10 bit RGB (r210) format auto detection - decklinkaudiosink: Fix playback when video caps is configured before audio - decklinkvideosrc: RGB 4:4:4 doesn’t work after GStreamer upgrade (regression) - decklinkvideosrc: unable to show HDMI stream that Blackmagic’s Media Express is able to see - debugqroverlay: fix string leak - gtkwaylandsink: Destroy GstWlWindow when parent GtkWindow is destroyed - gtkwaylandsink: Fix crash when rendering after the window is closed - ksvideo, directshow: Fix reference leaks in device providers - h264decoder: Fix DPB bumping process - h264decoder, h265decoder: Latency reporting related fixes - h264parse: Validate VUI framerate - jpegparse: reset parse state when the SOI is not the first marker - nvencoder: Fix CQP option setting - nvh264encoder: Fix template caps to include progressive mode as well - openjpegdec: allow multithread decoding only in subframe mode - tsdemux: Fix reading of extended Opus channel configuration - vulkan: fix validation layer issues - vulkanoverlaycompositor: fix potential use after free - vulkanswapper: correctly handle force-aspect-ratio=false - wasapi2: Fix potential crash on device activation failure - webrtc: Fix segfault traversing ice transports - webrtc: patch leak caused by early return - zxing: add support for zxing-c++ 2.0 gst-plugins-ugly - No changes gst-libav - avdec_h264 pipeline freeze with FFmpeg6 - avdeinterlace, avmux: fix element reference leak - avviddec: Drop decoder stream lock when calling send_packet gst-rtsp-server - rtsp-server: fix deadlock on shutdown with non-live pipeline if media isn’t playing/prerolled yet and eos-shutdown is enabled for the media 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 - No changes gst-examples - No changes Development build environment - git: prevent CRLF line ending conversion for patches to fix pango subproject patching issues on Windows Cerbero build tool and packaging changes in 1.22.2 - build: retry rust build on SIGBUS errors too - Fix packaging of rust plugins on Android - Modern Gentoo distro adaptation - sbc: update to 2.0 - speex: update to 1.2.1 Contributors to 1.22.2 Adrien De Coninck, Albert Sjölund, Alexande B, Antonio Rojas, Arun Raghavan, Bart Van Severen, Carlo Cabrera, Colin Kinloch, Edward Hervey, Guillaume Desmottes, Haihua Hu, He Junyan, Ilie Halip, Jordan Petridis, Josef Kolář, Lily Foster, Mathieu Duponchelle, Matt Feury, Matthew Waters, Maxim P. Dementyev, Michael Tretter, Nicolas Dufresne, Nirbheek Chauhan, Piotr Brzeziński, Robert Rosengren, Rouven Czerwinski, Sebastian Dröge, Seungha Yang, Shengqi Yu, Stéphane Cerveau, Talha Khan, Thibault Saunier, Tim-Philipp Müller, Víctor Manuel Jáquez Leal, Vivia Nikolaidou, Wang Chuan, Wojciech Kapsa, … 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 1.22.2 - List of Merge Requests applied in 1.22.2 - List of Issues fixed in 1.22.2 1.22.3 The third 1.22 bug-fix release (1.22.3) was released on 19 May 2023. This release only contains bugfixes and it should be safe to update from 1.22.x. Highlighted bugfixes in 1.22.3 - avdec: fix occasional video decoder deadlock on seeking with FFmpeg 6.0 - decodebin3: fix regression handling input streams without CAPS or TIME segment such as e.g. udpsrc or `pushfilesrc - bluez: a2dpsink: fix Bluetooth SIG Certification test failures - osxvideosink: fix deadlock upon closing output window - qtdemux: fix edit list handling regression and AV1 codec box parsing - qtmux: fix extraction of CEA608 closed caption data from S334-1A packets - rtspsrc: Fix handling of * control path - splitmux: timestamp handling improvements - v4l2videodec: Rework dynamic resolution change handling (needed for IMX6 mainline codec) - videoflip: fix regression with automatically rotating video based on tags - d3d11: many d3d11videosink and d3d11compositor fixes - webrtc, rtp: numerous data race fixes and stability fixes - various bug fixes, memory leak fixes, and other stability and reliability improvements gstreamer - tracing: Initialize tracing infrastructure even if the debug system is not compiled in - parse-launch: fix missing unref of looked-up child element - gstutils: Add category and object to most logging messages gst-plugins-base - allocators: Fix fdmem unit test with recent GLib versions - audiotestsrc: Initialize all samples in wave=ticks mode - decodebin3: Handle input streams without CAPS or TIME segment such as e.g. udpsrc or pushfilesrc - decodebin3: fix regression handling streams without caps - decodebin3: fix random hang when remove failing stream - uridecodebin3: Ensure atomic urisourcebin state change - glvideoflip: fix leaked caps - glcontext_wgl: fix missing unref - playsink: Fix volume leak gst-plugins-good - adaptivedemux2: fix critical when using an unsupported URI - dashdemux2: mpdclient: fix divide by 0 if segment has no duration - imagesequencesrc: Properly set default location - multifile: error out if no filename was set - osxvideosink: fix deadlock upon closing output window - rtpmanager: rtpsession: data race leading to critical warnings - rtpmanager: rtpsession: race conditions leading to critical warnings - rtspsrc: Fix handling of * control path - splitmuxsink: Catch invalid DTS to avoid running into problems later - splitmuxsrc: Make PTS contiguous by preference - qtdemux: emit no-more-pads after pruning old pads - Revert “qtdemux: fix conditions for end of segment in reverse playback” to fix edit list regression - qtdemux: Fix av1C parsing - qtmux: Fix extraction of CEA608 data from S334-1A packets - qtwindow: unref caps in destructor - v4l2: device provider: Fix GMainLoop leak - v4l2: videodec: Rework dynamic resolution change handling - v4l2: videodec: Prefer acquired caps over anything downstream - videoflip: fix setting of method property at construction time - videoflip 1.22.2 not rotating video when extracting frames gst-plugins-bad - a2dpsink: Fails many tests in Bluetooth SIG Certification - avdtputil: Use int instead of int range for fixed bitpool values - ccconverter: reintroduce frame count reset on cycle completion - ccconverter: integer overflow & crashing - codectimestamper: remove PC file generation from plugin’s own meson.build - cudamemory: Fix for semi planar YUV memory size decision - d3d11compositor: Reconfigure resource only when output caps is changed - d3d11compositor: Skip zero alpha input - d3d11convert: Fix for runtime property update - d3d11memory: Don’t clear wrapped texture memory - d3d11videosink: Fix for ignored initial render rectangle - d3d11videosink: fix race condition in window unprepare - d3d11videosink: Enhancement for initial window size decision - d3d11videosink: Don’t clear prepared buffer on unlock_stop() - dashdemux: mpdclient: fix divide by 0 if segment has no duration - dtlstransport: Keep strong ref of dtls encoder/decoder - GstPlay: avoid getting property of playbin2 if subtitle_sid is null - GstPlay: fix critical log when using playbin3 - h264decoder: Drop nonexisting picture silently without error - dtmf: element classification improvements - mfvideoenc: Allow only even resolution numbers - sctpenc: Fix potential shutdown deadlock - srtpdec: fix “srtp-key” check - tests: disable dtls test if openssl is not present - tsdemux: Set number of channels to 2 for dual mono Opus - va: Various fixes for defects found with MSVC - wasapi2: Allows process loopback capture on Windows 10 - webrtcdatachannel: Bind to parent webrtcbin using a weak reference - webrtcbin: Fix potential deadlock when closing before any data was sent - webrtc: Plug leaks of resolved ICE addresses - webrtc: do not tear down data channel before data is flushed gst-plugins-ugly - mpegpsdemux: Rework gap sending gst-libav - avviddec: Temporarily unlock stream lock while flushing buffers - Random freeze and deadlock in ffmpegviddec flush and get_buffer while seeking gst-rtsp-server - No changes gstreamer-vaapi - No changes gstreamer-sharp - No changes gst-omx - No changes gst-python - No changes gst-editing-services - ges: base-xml-formatter: Don’t pass non-GObject pointers to GST_DEBUG_OBJECT gst-validate + gst-integration-testsuites - No changes gst-examples - No changes Development build environment - No changes Cerbero build tool and packaging changes in 1.22.3 - glib: Ship Windows process spawning helpers - recipes: add recipe for libltc for timecodestamper element - Add support for RHEL9 and Rocky Linux Contributors to 1.22.3 Aleksandr Slobodeniuk, Antonio Kevo, Arun Raghavan, Carlos Rafael Giani, Daniel Moberg, Edward Hervey, Elliot Chen, François Laignel, Guillaume Desmottes, Haihua Hu, Jan Alexander Steffens (heftig), Jan Beich, Jan Schmidt, Johan Sternerup, John King, Jordan Petridis, Juan Navarro, Lily Foster, Martin Nordholts, Mathieu Duponchelle, Matthew Waters, Matthias Fuchs, Michael Olbrich, Mihail Ivanchev, Nick Steel, Nicolas Dufresne, Nirbheek Chauhan, Patricia Muscalu, Philippe Normand, Piotr Brzeziński, Sanchayan Maity, Sebastian Dröge, Seungha Yang, Stéphane Cerveau, Thibault Saunier, Tim-Philipp Müller, Xabier Rodriguez Calvar, … 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 1.22.3 - List of Merge Requests applied in 1.22.3 - List of Issues fixed in 1.22.3 1.22.4 The fourth 1.22 bug-fix release (1.22.4) was released on 20 June 2023. This release only contains bugfixes and security fixes and it should be safe to update from 1.22.x. Highlighted bugfixes in 1.22.4 - Security fixes for flacparse, dvdspu, and subparse - d3d11videosink: Fix error on pause and play - decklink: Correctly handle SDK strings on macOS and free strings after usage on Linux - filesink: Fix buffered mode writing of buffer lists and buffers with multiple memories - gldownload: handle passthrough without a critical - h265parse: Fix framerate handling regression - oggdemux: vp8 fixes - mp4mux, qtmux, qtdemux: Opus audio mapping fixes - pngdec: Fix wrong colours output from 16bit RGB images - ptp clock: Work around ptpd bug in default configuration - srtpdec: fix critical warnings on shutdown - v4l2src: fix support for bayer format - v4l2videoenc: support force-keyframe event in v4l2 encoder - vtenc: apply DTS offset to ensure DTS <= PTS - gst-python: allow more functions to be called before gst_init() - cerbero: fix vaapi variant; add qt6 build on windows; ensure errors on unguarded use of new APIs, require macOS 10.13 - packages: ship codecalpha, rtponvif, dvbsubenc, switchbin, videosignal plugins; fix pango crash on 32-bit windows - various bug fixes, memory leak fixes, and other stability and reliability improvements gstreamer - filesink: Fix buffered mode writing of buffer lists and buffers with multiple memories - basesink: Clear EOS flag on STREAM-START event - typefindhelper: downgrade bogus error level debug log message - ptp: Correctly parse clock ID from the commandline parameters in the helper - ptp: Work around bug in ptpd in default configuration gst-plugins-base - alsasink: Fix stall for transition from PAUSED to READY with USB speakerphone. - appsink: unref buffer in prev sample early so buffers from v4l2 can be released properly - basetextoverlay: Fix typo in “text-y” property description - gldownload: handle passthrough without a critical - glfilter: add parent meta to output buffer for input buffer - oggdemux: vp8: Push headers downstream and detect keyframe packets - opus: Fix potential crash when getting unexpected channel position - streamsynchronizer: reset eos on STREAM_START - subparse: Look for the closing > of a tag after the opening < - video: convertframe: Add D3D11 specific conversion path - videometa: Only validate the alignment only when it contains some info - video-blend: Fix linking error with C++ gst-plugins-good - flacparse: Avoid integer overflow in available data check for image tags - flvmux: use the correct timestamp to calculate wait times - isomp4: Fix (E)AC-3 channel count handling - jpegdec: fixes related to interlaced jpeg - pngdec: Fix wrong colours output from 16bit RGB images - qtmux, qtdemux: fix byte order for opus extension - rtspsrc: Do not try send dropped get/set parameter - qt5, qt6: Add more meson options and eliminate all automagic - qt: glrenderer: don’t attempt to use QWindow from non-Qt main thread - qml6glsink: Support building on win32 - v4l2src: fix support for bayer format - v4l2: Change to query only up to V4L2_CID_PRIVATE_BASE+V4L2_CID_MAX_CTRLS - v4l2videodec: treat MPEG-1 format as MPEG-2 - v4l2videoenc: support force keyframe event in v4l2 encoder - tests: rtpbin_buffer_list: fix possible unaligned write/read on 32-bit ARM gst-plugins-bad - asfmux: fix possible unaligned write on 32-bit ARM - d3d11videosink: Fix error on pause and play - d3dvideosink: Fix navigation event leak - decklink: Correctly handle SDK strings on macOS and free strings after usage on Linux - dvdspu: Make sure enough data is allocated for the available data - fdkaacdec: Support up to 5 rear channels - h265parse: Fix framerate handling - kmssink: Add STM32 LTDC and NXP i.MX8M Plus LCDIFv3 auto-detection - sdpdemux: ensure that only one srcpad is created per stream - srtpdec: fix critical warnings on shutdown - testsrcbin: Remove spurious caps unref - va: map the mbbrc to correct enum value in get_property() - vtenc: apply DTS offset to ensure DTS <= PTS - vtdec: time glitches on h264 playback - waylandsink: Emit “map” signal boarder surface is ready gst-plugins-ugly - No changes gst-libav - No changes gst-rtsp-server - No changes gstreamer-vaapi - vaapidecodebin: don’t load vaapipostproc if not available gstreamer-sharp - No changes gst-omx - No changes gst-python - python: More functions can be called before gst_init() gst-editing-services - ges: launcher: Never put sinks in a GstPipeline gst-validate + gst-integration-testsuites - No changes gst-examples - No changes Development build environment - No changes Cerbero build tool and packaging changes in 1.22.4 - Ship codecalpha, rtponvif, dvbsubenc, switchbin, videosignal plugins - pango: Fix crash on Windows 32bit build - qml6: Add support for building the qml6 plugin on Windows and bump meson to 1.1.1 - vaapi: update vaapi variant/recipe for meson option changes - packages: Put libass in the same category as assrender - cerbero: Don’t extract if already extracted in fetch - darwin: Ensure errors on unguarded use of new APIs, require macOS 10.13 Contributors to 1.22.4 Andoni Morales Alastruey, Arun Raghavan, Colin Kinloch, Daniel Morin, Edward Hervey, ekwange, Elliot Chen, François Laignel, Guillaume Desmottes, Haihua Hu, He Junyan, Hou Qi, Jan Alexander Steffens (heftig), Jochen Henneberg, Jordan Petridis, Kevin Song, Maksym Khomenko, Marek Vasut, Mathieu Duponchelle, Matthew Waters, Mengkejiergeli Ba, Michael Olbrich, Nicolas Beland, Nicolas Dufresne, Nirbheek Chauhan, Philippe Normand, Piotr Brzeziński, Sebastian Dröge, Seungha Yang, Thibault Saunier, Tim-Philipp Müller, Víctor Manuel Jáquez Leal, William Manley, Xavier Claessens, Yuri Fedoseev, … 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 1.22.4 - List of Merge Requests applied in 1.22.4 - List of Issues fixed in 1.22.4 1.22.5 The fifth 1.22 bug-fix release (1.22.5) was released on 20 July 2023. This release only contains bugfixes and security fixes and it should be safe to update from 1.22.x. Highlighted bugfixes in 1.22.5 - Security fixes for the RealMedia demuxer - vaapi decoders, postproc: Disable DMAbuf from caps negotiation to fix garbled video in some cases - decodebin3, playbin3, parsebin fixes, especially for stream reconfiguration - hlsdemux2: fix early seeking; don’t pass referer when updating playlists; webvtt fixes - gtk: Fix critical caused by pointer movement when stream is getting ready - qt6: Set sampler filtering method, fixes bad quality with qml6glsink and gstqt6d3d11 - v4l2src: handle resolution change when buffers are copied - videoflip: update orientation tag in auto mode - video timecode: Add support for framerates lower than 1fps and accept 119.88 (120/1.001) fps - webrtcsink: fixes for x264enc and NVIDIA encoders - cerbero: Pull ninja from system if possible, avoid spurious bootstrap of cmake - packages: Recipe updates for ffmpeg, libsoup, orc - various bug fixes, memory leak fixes, and other stability and reliability improvements gstreamer - taglist, plugins: fix compiler warnings with GLib >= 2.76 - tracerutils: allow casting parameter types - inputselector: fix playing variable is never set gst-plugins-base - appsink: add missing make_writable call - audioaggregator: Do not post message before being constructed - decodebin3: Prevent a critical warning when reassigning output slots - decodebin3: Fix slot input linking when the associated stream has changed - decodebin3: Remove spurious input locking during parsebin reconfiguration - urisourcebin: Set source element to READY before querying it - gl/viv-fb: meson build updates - plugins: fix compiler warnings with GLib >= 2.76 - subtitleoverlay: fix mutex error if sink caps is not video - video: timecode: Add support for framerates lower than 1fps - video: accept timecode of 119.88 (120/1.001) FPS - video: cannot attach time code meta when frame rate is 119.88 (120000/1001) - videodecoder: fix copying buffer metas gst-plugins-good - adaptivedemux2: Fix early seeking - hlsdemux2: Ensure processed webvtt ends with empty new line - hlsdemux2: Don’t set a referer when updating playlists - matroska: demux: Strip signal byte when encrypted - rtspsrc: Fix crash when is-live=false - gtk: Fix critical caused by pointer movement when stream is getting ready - qt6: Set sampler filtering method, fixes bad quality with qml6glsink and gstqt6d3d11 - qtdemux: opus: set entry as sampled - v4l2src: handle resolution change when buffers are copied - v4l2videodec: Fix handling of initial gaps - v4l2videodec: correctly register v4l2mpeg2dec - v4l2videoenc: replace custom QUERY_CAPS handling with getcaps callback - videoflip: update orientation tag in auto mode - videoflip: fix critical when tag list is not writable gst-plugins-bad - d3d11bufferpool: Fix heavy CPU usage in case of fixed-size pool - jpegparser: jpegdecoder: Don’t pollute bus and comply with spec - plugins: fix compiler warnings with GLib >= 2.76 - webrtcbin: Prevent critical warning when creating an additional data channel - webrtcstats: Properly report IceCandidate type gst-plugins-ugly - rmdemux: add some integer overflow checks gst-plugins-rs - fallbackswitch: Change the threshold for trailing buffers - fallbackswitch: Fix pad health calculation and notifies - fmp4mux: Fix draining in chunk mode if keyframes are too late - livesync: Wait for the end timestamp of the previous buffer before looking at queue - livesync: Improve EOS handling - togglerecord: Clip segment before calculating timestamp/duration - togglerecord: Error out if main stream buffer has no valid running time - webrtcsink: fix pipeline when input caps contain max-framerate - webrtcsink: Configure only 4 threads for x264enc - webrtcsink: Translate force-keyunit events to force-IDR action signal for NVIDIA encoders - webrtcsink: Set config-interval=-1 and aggregate-mode=zero-latency on rtph264pay and rtph265pay - webrtcsink: Set VP8/VP9 payloader based on payloader element factory name - webrtcink: Use correct property types for nvvideoconvert - webrtc/signalling: fix race condition in message ordering - videofx: Minimize dependencies of the image crate gst-libav - No changes gst-rtsp-server - No changes gstreamer-vaapi - vaapidecode,vaapipostproc: Disable DMAbuf from caps negotiation. gstreamer-sharp - No changes gst-omx - No changes gst-python - No changes gst-editing-services - ges: some fixes for 32-bit systems - ges, nle: Avoid setting state or sending query when constructing objects gst-validate + gst-integration-testsuites - No changes gst-examples - No changes Development build environment - No changes Cerbero build tool and packaging changes in 1.22.5 - Pull ninja from system if possible, avoid spurious bootstrap of cmake - ffmpeg: update to 5.0.3 - libsoup: update to 2.74.3 - orc: update to 0.4.34 Contributors to 1.22.5 Andoni Morales Alastruey, Bastien Nocera, Carlos Rafael Giani, David Craven, Doug Nazar, Edward Hervey, François Laignel, Guillaume Desmottes, He Junyan, Hou Qi, Jan Alexander Steffens (heftig), Jan Schmidt, Maksym Khomenko, Mathieu Duponchelle, Matthew Waters, Michael Olbrich, Michael Tretter, Nicolas Dufresne, Nirbheek Chauhan, Philippe Normand, Ruslan Khamidullin, Sebastian Dröge, Seungha Yang, Théo Maillart, Thibault Saunier, Tim-Philipp Müller, Víctor Manuel Jáquez Leal, Vivia Nikolaidou, Yatin Maan, … 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 1.22.5 - List of Merge Requests applied in 1.22.5 - List of Issues fixed in 1.22.5 1.22.6 The sixth 1.22 bug-fix release (1.22.6) was released on 20 September 2023. This release only contains bugfixes and security fixes and it should be safe to update from 1.22.x. Highlighted bugfixes in 1.22.6 - Security fixes for the MXF demuxer and H.265 video parser - Fix latency regression in H.264 hardware decoder base class - androidmedia: fix HEVC codec profile registration and fix coded_data handling - decodebin3: fix switching from a raw stream to an encoded stream - gst-inspect: prettier and more correct signal and action signals printing - rtmp2: Allow NULL flash version, omitting the field, for better RTMP server compatibility - rtspsrc: better compatibility with buggy RTSP servers that don’t set a clock-rate - rtpjitterbuffer: fix integer overflow that led to more packets being declared lost than have been lost - v4l2: fix video encoding regression on RPi and fix support for left and top padding - waylandsink: Crop surfaces to their display width height - cerbero: recognise Manjaro; add Rust support for MSVC ARM64; cmake detection fixes - various bug fixes, build fixes, memory leak fixes, and other stability and reliability improvements gstreamer - gst-inspect: prettier and more correct signal printing, and print action signals in g_signal_emit_by_name() format - gst-launch: Disable fault signal handlers on macOS gst-plugins-base - audio: Make sure to stop ringbuffer on error - decodebin3: avoid identity, sinkpad, parsebin leakage when reset input - decodebin3: Ensure the slot is unlinked before linking to decoder - sdp: fix wrong debug log error message for missing clock-rate in caps - sdp: Parse zero clock-rate as default gst-plugins-good - adaptivedemux2: fix memory leak - pulsedeviceprovider: fix incorrect usage of GST_ELEMENT_ERROR - qt: Unbreak build with qt-egl enabled but viv_fb missing - qt: Fix searching of qt5/qt6 tools with qmake in Meson - qtdemux: Fix premature EOS when some files are played in push mode - qtdemux: attach cbcs crypt info at the right moment - rtpjitterbuffer: Avoid integer overflow in max saveable packets calculation with negative offset - videoflip: fix concurrent access when modifying the tag list - v4l2: allocator: Don’t close foreign dmabuf - v4l2: bufferpool: Fix large encoded stream regression - v4l2: bufferpool: Problems when checking for truncated buffer - v4l2: Fix support for left and top padding - v4l2object: clear format lists if source change event is received gst-plugins-bad - androidmedia/enc: handle codec-data before popping GstVideoCodecFrames - androidmedia: fix hevc codec profile registration - androidmedia: Small fixes - androidmedia: Add more null checks (of env) to JNI utilities - applemedia: Fix pixel format for I420 and NV12 - audiolatency: Forward latency query and event upstream - av1parser: Fix segmentation params update - codecparsers: Fix MPEG-1 aspect ratio table - d3d11convert: Passthrough allocation query on same caps - h264decoder: Update latency dynamically - h265parser: Allow partially broken hvcC data - h265parser: Fix possible overflow using max_sub_layers_minus1 - hlssink2: Always use forward slash separator - mdns: Fix a crash on context error - mxfdemux: Fix integer overflow causing out of bounds writes when handling invalid uncompressed video and check channels for AES3 - nvencoder: Fix negotiation error when interlace-mode is unspecified - rtmp2: Allow NULL flash version, omitting the field - rtmp2sink: fix crash if message conversion failed - transcodebin: Fixes for upstream selectable support - va: Fix in error logs functions mismatches - waylandsink: Crop surfaces to their display width height - waylandsink: Fix cropping for video with non-square aspect ratio - webrtc: Fix docs for create-data-channel action signal - win32ipc: Fix pipe handle leak gst-plugins-ugly - No changes gst-plugins-rs - fallbackswitch: locking/deadlock fixes - onvifmetadataparse: Skip metadata frames with unrepresentable UTC time - transcriberbin: Configure audioresample in front of transcriber - webrtcsink: Propagate GstContext messages - webrtcsink: Add support for d3d11 memory and qsvh264enc - webrtcsink: fix TWCC extension adding - webrtcsink: don’t forget to setup encoders for discoveries - webrtcsink: NVIDIA V4L2 encoders always require NVMM memory - meson: Fix handling of optional deps, and don’t require Python 3.8 gst-libav - No changes gst-rtsp-server - No changes 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: Disable fault signal handlers on macOS gst-examples - No changes Development build environment - macos-bison: Update to 3.8.2 and add an ARM64 build - wrap: update libpsl to 0.21.2 Cerbero build tool and packaging changes in 1.22.6 - Add Rust support for MSVC ARM64 - Recognise PERL5LIB as a joinable Unix variable - Recognise Manjaro as an Arch derivative - Fix picking up cmake from build-tools Contributors to 1.22.6 Akihiro Sagawa, Alicia Boya García, Guillaume Desmottes, Haihua Hu, Hugues Fruchet, Ivan Molodetskikh, Jan Alexander Steffens (heftig), Jan Schmidt, L. E. Segovia, Mathieu Duponchelle, Matthew Waters, Ming Qian, Nicolas Dufresne, Nirbheek Chauhan, Olivier Blin, Olivier Crête, Philippe Normand, Piotr Brzeziński, Robert Ayrapetyan, Ryan Pavlik, Sebastian Dröge, Seungha Yang, Stéphane Cerveau, Stephan Seitz, Thomas Schneider, Tim-Philipp Müller, Víctor Manuel Jáquez Leal, Wang Chuan, Xabier Rodriguez Calvar, … 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 1.22.6 - 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 “Can’t find exact taps” debug log messages - wasapi2: Don’t use global volume control object - Rust plugins: various improvements in aws, fmp4mux, hlssink3, livesync, ndisrc, rtpav1depay, rsfilesink, s3sink, sccparse - WebRTC: various webrtchttp, webrtcsrc, and webrtcsink improvements and fixes - Cerbero build tools: recognise Windows 11; restrict parallelism of gst-plugins-rs build on small systems - Packages: ca-certificates update; fix gio module loading and TLS support on macOS gstreamer - debugutils: provide gst_debug_bin_to_dot_data() implementation even if debug system is disabled gst-plugins-base - audioaggregator, audiomixer: Make access to the pad list thread-safe while mixing - basetextoverlay: Fix overlay never rendering again if width reaches 1px - glfiter: Protect GstGLContext access - glfilter: Only add parent meta if inbuf != outbuf - macOS: fix huge memory leak with glfilter-based elements - rtspconnection: Ignore trailing whitespace in rtsp headers - video-scaler, audio-resampler: downgrade ‘can’t find exact taps’ to debug gst-plugins-good - adaptivedemux2: Do not submit_transfer when cancelled - adaptivedemux2: Call GTasks’s return functions for blocking tasks - flacenc: Correctly handle up to 255 cue entries - flvmux: set the src segment position as running time - imagesequencesrc: fix deadlock when feeding the same image in a loop - pngenc: output one frame only in snapshot mode and mark output frames as I-frames - qmlglsrc: sync on the streaming thread - souphttpsrc: Chain up to finalize to fix memory leak - wavparse: fix buffer leak with adtl tag - v4l2codecs: Avoid QBUF/DQBUF struct timeval .tv_usec wrap-around at frame 1000000 - v4l2codecs: Fix tiled formats stride conversion gst-plugins-bad - audiobuffersplit: disable max-silence-time if set to 0 - libde265: Do not decode the non 4:2:0 8 bits format - codecparsers: av1: Clip max tile rows and cols values - codecs: h265: Do not free slice header before using it - d3d11converter: Fix 10/12bits planar output - d3d11decoder: Fix crash on negotiate() when decoder is not configured - d3d11videosink: Fix toggling between fullscreen and maximized - d3d11videosink: Fix window switching in case of fullscreen mode - d3d11screencapturesrc: Fix mouse cursor blending - decklink: Fix broken COM string conversion - decklink: Decklink Device Provider wrongly parses SDK strings - gstwayland: Don’t depend on wayland-protocols - interaudiosrc: Add audio meta to buffers containing non-interleaved samples - kmssink: Add TIDSS auto-detection - mfvideoencoder: Fix typo in template caps - mxfdemux: Store GstMXFDemuxEssenceTrack in their own fixed allocation - nvcodec: fix bounds for auto-select GPU enumeration - openh264: Fail gracefully if openh264 encoder/decoder creation fails - tsmux: More cleanups - tsmux: Fill padding packets with stuffing bytes - v4l2codecs: Fix tiled formats stride conversion - v4l2videodec: Correctly free caps to avoid memory leak - wasapi2: Don’t use global volume control object - wasapi2device: Ignore activation failed device gst-plugins-ugly - No changes gst-plugins-rs - aws: s3sink: Fix handling of special characters in key - aws: s3: Properly percent-decode GstS3Url - fmp4mux: Don’t overflow negative composition offset calculation - fmp4mux: specify the fragment duration unit - hlssink3: Use Path API for getting file name - hlssink3: Use sprintf for segment name formatting - hlssink3: Remove unused deps - hlssink3: Don’t remove old files if max-files is zero - hlssink3: Don’t remove uri from playlist if playlist-length is zero - hlssink3: Various cleanup - livesync: log new pending segments - livesync: display jitter when waiting on clock - livesync: Rename activatemode methods to *_activatemode - livesync: Simplify start_src_task and src_loop - livesync: Improve audio duration fixups - livesync: Log a category error when we are missing the segment - livesync: Clean up state handling - livesync: Replace an if-let with match - livesync: Move a notify closer to the interesting state change - livesync: Move num_in counting to the src task - livesync: Simplify num_duplicate counting - livesync: Handle flags and late buffer patching after queueing - livesync: Separate out_buffer duplicate status from GAP flag - livesync: Use fallback_duration for audio repeat buffers as well - livesync: example: Add identities single-segment=1 - livesync: Split fallback_duration into in_ and out_duration - livesync: Keep existing buffer duration in some cases - livesync: Remove the stop from outgoing segments - ndisrc: Assume input with more than 8 raw audio channels is unpositioned - rtpav1depay: Skip unexpected leading fragments - rtpav1depay: Don’t push stale temporal delimiters downstream - rsfilesink: set sync=false - s3sink: set sync=false - sccparse: Fix leading spaces between the tab and caption data - webrtchttp: Respect HTTP redirects - webrtcsrc: use @watch instead of @to-owned - webrtcsrc: add turn-servers property - webrtc: Fix paths in README - webrtcsink: don’t miss ice candidates gst-libav - No changes gst-rtsp-server - rtspclientsink: Don’t leak previous server_ip gstreamer-vaapi - No changes gstreamer-sharp - No changes gst-omx - No changes gst-python - No changes gst-editing-services - No changes gst-validate + gst-integration-testsuites - gst-validate: Fix compatibility with Python 3.12 gst-examples - No changes Development build environment - No changes Cerbero build tool and packaging changes in 1.22.7 - Add Windows 11 to the supported versions list - ca-certificates: Update to version from 2023-08-22 - cargo: Restrict parallelism if a small system is detected (for gst-plugins-rs build) - Fix venv setup on Python 3.11+ - Fix unlinking of Android NDK directories if install fails midway - glib: Work around AppleClang + -Werror test build failure - glib: Re-add gio module loading patch for macOS, remove unused patch files Contributors to 1.22.7 Albert Sjölund, Arun Raghavan, Balló György, Benjamin Gaignard, Detlev Casanova, Doug Nazar, Eric, Florian Zwoch, François Laignel, Guillaume Desmottes, He Junyan, Hou Qi, James Oliver, Jan Alexander Steffens (heftig), Jan Schmidt, Johan Adam Nilsson, Jordan Yelloz, Kalev Lember, L. E. Segovia, Lieven Paulissen, Maksym Khomenko, Marek Vasut, Matthias Fuchs, Michiel Westerbeek, Nicolas Dufresne, Philippe Normand, Piotr Brzeziński, Rahul T R, Sean DuBois, Sebastian Dröge, Seungha Yang, Stéphane Cerveau, Thibault Saunier, Tim-Philipp Müller, … and many others who have contributed bug reports, translations, sent suggestions or helped testing. Thank you all! List of merge requests and issues fixed in 1.22.7 - List of Merge Requests applied in 1.22.7 - List of Issues fixed in 1.22.7 1.22.8 The eight 1.22 bug-fix release (1.22.8) was released on 18 December 2023. This release only contains bugfixes and security fixes and it should be safe to update from 1.22.x. Highlighted bugfixes in 1.22.8 - Security fixes for the AV1 video codec parser - avdec video decoder: fix another possible deadlock with FFmpeg 6.1 - qtdemux: reverse playback and seeking fixes for files with raw audio streams - v4l2: fix “newly allocated buffer … is not free” warning log flood - GstPlay + GstPlayer library fixes - dtls: Fix build failure on Windows when compiling against OpenSSL 3.2.0 - d3d11screencapturesrc: Fix wrong color with HDR enabled - Cerbero build tool: More python 3.12 string escape warning fixes; make sure to bundle build tools as well - various bug fixes, build fixes, memory leak fixes, and other stability and reliability improvements gstreamer - buffer: Unref memories before metas - pad: Recheck pads when linking after temporary unlock - baseparse: Fixes to buffers extracted from adapter gst-plugins-base - appsrc: Fix flow return when buffer is dropped - audioringbuffer: Don’t try to map MONO channel - encoding-target: Properly free when missing type field in parse_encoding_profile - pbutils: Don’t include default vp9 parameters in resulting codec mime string - videorate: Don’t forget last_ts on caps changes gst-plugins-good - dcaparse: keep upstream buffer meta - rtpklvdepay: Recover after invalid fragmented KLV unit - matroska-demux: fix accumulated base offset in segment seeks - qtdemux: fix bug report URL - qtdemux: Don’t overflow sample index - qtdemux: Fix reverse playback for pcm audio stream - qtdemux: Ignore raw audio streams when adjusting seek - qtdemux: Under-seeking to a key unit in certain (encoded by Adobe products) ProRes movies (macOS x86_64 & arm64, Windows x86_64, …) - rtpac3depay: should output audio/x-ac3 not audio/ac3 - rtp: Fix incorrect RTP channel order lookup by name - v4l2bufferpool: add lock as atomic operation for seek gst-plugins-bad - aesenc: Fix IV length addition to output buffer length - av1parser: Fix array sizes in scalability structure - camerabin: Fix source updates with user filters - codecparsers: av1: Clip max tile rows and cols values - dtlscertificate: Define WINSOCKAPI before including windows.h - d3d11: fix building with address sanitizer - d3d11screencapturesrc: Fix wrong color with HDR enabled - h264decoder: Fix GstVideoCodecFrame leak - ladspa: Make RDF parsing truly optional - rtponviftimestamp: Fix drop-out-of-segment=false mode - qsvdecoder: Fix stream format detection - webrtcsdp: Remove fingerprint validation that doesn’t make sense - GstPlay: Automatically flush the bus when disposing the signal adapter - GstPlayer: Without dispatcher emit signals directly instead of via the default main context gst-plugins-ugly - No changes gst-plugins-rs - threadshare: Fix a deadlock in used-socket notification - threadshare: Fix a typo while logging - webrtcsink: don’t panic on failure to request pad from webrtcbin - ndi: Remove wrong Clone impl on RecvInstance - ndi: Don’t mark private type as public - fallbacksrc: Fix timeout scheduling gst-libav - avviddec: Unlock stream lock while waiting for decoded frame. Fixes potential deadlock - avviddec: Calculate latency only for fixed framerate gst-rtsp-server - No changes 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 - No changes gst-examples - No changes Development build environment - No changes Cerbero build tool and packaging changes in 1.22.8 - cerbero: Fix some more python 3.12 string escape warnings - cerbero: Fix bundle-source not including build-tools recipes, fix CalledProcessError handling - pango: Add Perl interpreter consistency check Contributors to 1.22.8 Alessandro Bono, Alexander Slobodeniuk, Arun Raghavan, Benjamin Gaignard, Daniel Moberg, Dongyun Seo, Doug Nazar, Guillaume Desmottes, Hosang Lee, Jan Alexander Steffens (heftig), jeri.li, Jimmy Ohn, L. E. Segovia, Mathieu Duponchelle, Nicolas Dufresne, Nirbheek Chauhan, Olivier Crête, Philippe Normand, Piotr Brzeziński, Rabindra Harlalka, Robert Mader, Robin Gustavsson, Sebastian Dröge, Seungha Yang, Stefan Brüns, Tim-Philipp Müller, Xavier Claessens, … 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 1.22.8 - List of Merge Requests applied in 1.22.8 - List of Issues fixed in 1.22.8 1.22.9 The ninth 1.22 bug-fix release (1.22.9) was released on 24 January 2024. This release only contains bugfixes and security fixes and it should be safe to update from 1.22.x. Highlighted bugfixes in 1.22.9 - More Security fixes for the AV1 video codec parser - va: fixes for Mesa Gallium drivers in Mesa versions older than v23.2 - v4l2src: Consider framerate during caps selection - v4l2codec: decoder fixes - rtspsrc: multicast fixes - camerabin viewfinder fixes - various bug fixes, build fixes, memory leak fixes, and other stability and reliability improvements gstreamer - aggregator: fix use-after-free in queries processing - multiqueue: Ignore queue fullness for most events gst-plugins-base - audiobasesink: Don’t wait on gap events - audioconvert: change gst_audio_convert_get_unit_size() log levels - glcolorconvert: Correct transform_caps direction - gloverlay: Apply updated overlay coordinates correctly - videorate: keep pool if max_buffers is unlimited gst-plugins-good - rtpsession: Only warn once if configured latency needs to be known but isn’t yet - rtphdrext-clientaudiolevel: Fix level value being written by the extension - rtspsrc: set multicast-iface on udpsinks and fix RTCP sink TTL - v4l2object: clear old fds when initializing poll during opening v4l2 device - v4l2src: Consider framerate during caps selection - vpxdec: Use appropriate domain and code for decoding errors gst-plugins-bad - av1parser: Fix potential stack overflow during tile list parsing - camerabin: Correctly relink viewfinderbin_queue - GstPlay: Fix error details parsing - h264decoder: Handle malformed avc/avc3 packets - h264decoder: h265decoder: Align with wraparound fix - vp8decoder: vp9decoder: av1decoder: mpeg2decoder: Fix multiplication wraparound - vah264enc/vah264dec issues after recent upgrade to 1.22.8 from 1.22.7 - va: fixes for Mesa Gallium drivers in Mesa versions older than v23.2 - vp9parse: Fix critical warning during caps negotiation gst-plugins-ugly - No changes gst-plugins-rs - No changes gst-libav - No changes gst-rtsp-server - No changes 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 - No changes gst-examples - No changes Development build environment - No changes Cerbero build tool and packaging changes in 1.22.9 - No changes Contributors to 1.22.9 Alexander Slobodeniuk, Chao Guo, Damian Hobson-Garcia, Dan Searles, Guillaume Desmottes, Jan Schmidt, Marek Vasut, Mengkejiergeli Ba, Michael Tretter, Philippe Normand, Robert Mader, Sanchayan Maity, Sebastian Dröge, Seungha Yang, Tim-Philipp Müller, Víctor Manuel Jáquez Leal, Xavier Claessens, … 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 1.22.9 - List of Merge Requests applied in 1.22.9 - List of Issues fixed in 1.22.9 1.22.10 The tenth 1.22 bug-fix release (1.22.10) was released on 13 February 2024. This release only contains bugfixes and security fixes and it should be safe to update from 1.22.x. Highlighted bugfixes in 1.22.10 - gst-python: fix bindings overrides for Python >= 3.12 - glcolorconvert: fix wrong RGB to YUV matrix with bt709 - glvideoflip: fix “method” property setting at construction time - gtk4paintablesink: Always draw a black background behind the video frame, and other fixes - pad: keep segment event seqnums the same when applying a pad offset - basesink: Preroll on out of segment buffers when not dropping them - Prefer FFmpeg musepack decoder/demuxer, fixing musepack playback in decodebin3/playbin3 - livesync: add support for image formats such as JPEG or PNG - sdpdemux: Add SDP message (aka session) attributes to the caps too - textwrap: add support for gaps - macos: Fix gst_macos_main() terminating whole process, and set activation policy - webrtcbin: Improve SDP intersection for Opus - various bug fixes, build fixes, memory leak fixes, and other stability and reliability improvements gstreamer - pad: Copy over seqnum when creating a new segment event for applying pad offset - basesink: Preroll on out of segment buffers when not dropping them - macos: Fix gst_macos_main() terminating whole process before returning a value - macos: Set activation policy in gst_macos_main() gst-plugins-base - glcolorconvert: fix wrong RGB to YUV matrix with bt709 - glvideoflip: fix setting of method property at construction time - glvideoflip: “method”` property is broken if set during element construction - macos: Set activation policy in glimagesink - videoaggregator: fix bufferpool leak gst-plugins-good - taglib: Set cpp_std to c++17 to fix compilation with TagLib >= 2.0 - macos: Set activation policy in osxvideosink gst-plugins-bad - neon: Allow building against neon 0.33.x - sdpdemux: Add SDP message (aka session) attributes to the caps too - srtpenc: Fix potential leak - webrtcbin: Improve SDP intersection for Opus - wpe: Rename WPEView to WPEThreadedView to avoid clash with newer wpe version gst-plugins-ugly - No changes gst-plugins-rs - gtk4: Fix segfault running gst-inspect -a when GTK4 and GTK3 is installed - gtk4: Always draw a black background behind the video frame - livesync: add support for image formats - livesync: properly format jitter in debug logs - textwrap: add support for gaps - webrtc: only use close() to close websockets - webrtc: signallers: attempt to close the ws when an error occurs - webrtc/signalling: Fix potential hang and FD leak - webrtc/signalling: We get the address when accepting - meson: Fix build on Windows with MSVC - meson: pkg-config is required at build time - meson: Add nasm to PATH if meson can find it - meson: allow building plugins with GTK 4 examples - Update GStreamer bindings in Cargo.lock gst-libav - Prefer using FFmpeg musepack decoder/demuxer gst-rtsp-server - No changes gstreamer-vaapi - No changes gstreamer-sharp - No changes gst-omx - No changes gst-python - Some Python 3.12 fixes - python: Port from deprecated imp to importlib gst-editing-services - No changes gst-validate + gst-integration-testsuites - No changes gst-examples - No changes Development build environment - No changes Cerbero build tool and packaging changes in 1.22.10 - gst-plugins-bad: build soundtouch plugin on MSVC - cerbero: Fix GNU tar –checkpoint compatibility with macOS - cerbero: Fix bootstrap venv error after upgrading Python - gst-plugins-good: build taglib plugin on MSVC Contributors to 1.22.10 Alexander Slobodeniuk, Christian Curtis Veng, Edward Hervey, François Laignel, Guillaume Desmottes, Heiko Becker, Jan Schmidt, Jonas Kvinge, Jordan Petridis, L. E. Segovia, Lukas Geiger, Marvin Schmidt, Mathieu Duponchelle, Michael Tretter, Nirbheek Chauhan, Philippe Normand, Piotr Brzeziński, Ruben Gonzalez, Sebastian Dröge, 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 1.22.10 - List of Merge Requests applied in 1.22.10 - List of Issues fixed in 1.22.10 1.22.11 The eleventh 1.22 bug-fix release (1.22.11) was released on 19 March 2024. This release only contains bugfixes and security fixes and it should be safe to update from 1.22.x. The GStreamer 1.22 stable series has since been superseded by the GStreamer 1.24 stable release series. Highlighted bugfixes in 1.22.11 - Fix instant-EOS regression in audio sinks in some cases when volume is 0 - rtspsrc: server compatibility improvements and ONVIF trick mode fixes - libsoup linking improvements on non-Linux platforms - va: improvements for intel i965 driver - wasapi2: fix choppy audio and respect ringbuffer buffer/latency time - rtsp-server file descriptor leak fix - uridecodebin3 fixes - various bug fixes, build fixes, memory leak fixes, and other stability and reliability improvements gstreamer - uri: Sort uri protocol sources/sinks by feature name to break a feature rank tie - segment: Don’t use g_return_val_if_fail() in gst_segment_to_running_time_full() - identity: Don’t refuse seeks unless single-segment=true - ptp clock: Initialize expected DELAY_REQ seqnum to an invalid value gst-plugins-base - audiobasesink: Revert “Don’t wait on gap events” again, fixes instant-EOS in some cases - audioencoder: Avoid using temporarily mapped memory as base for input buffers - glimagesink: Fix the sink not always respecting preferred size on macOS - uridecodebin3: fix deadlock when switching input item - urisourcebin: Don’t acquire STATE_LOCK if shutting down - video: Fix NV12_16L32S video frame size gst-plugins-good - jpegdec: Fix progressive/interlaced detection - mpg123audiodec: Correctly handle the case of clipping all decoded samples - plugins: Fix GstFlowReturn/gboolean mixups - qtdemux: Do not set channel-mask to zero - rtpgstpay: Delay pushing of event packets until the next buffer - rtspsrc: remove ‘deprecated’ flag from the ‘push-backchannel-sample’ signal - rtspsrc: Don’t invoke close when stopping if we’ve started cleanup - rtspsrc: Reset combined flows after a seek before restarting - rtspsrc: Increase rank to PRIMARY for autoplug purposes - rtspsrc: Consider 503 Service Not Available when handling broken control urls - rtspsrc, rtponviftimestamp: ONVIF mode fixes - soup, adaptivedemux2: Backport various libsoup build fixes from main - v4l2src: fix cannot reuse current caps when fixate caps in negotiation gst-plugins-bad - asio: Fix {input,output}-channels property handling - asiosink: Fix channel selection - d3d11device: Fix adapter LUID comparison in wrapped device mode - d3d11window_win32: fix crash on RC unprepare() vs window_proc() - dvbsubenc: Fix bottom field size calculation - dvdspu: avoid null dereference - v4l2codecs: h264: Fix a memory leak on renegotiation - va: backport missing commits for i965 driver - vulkan/wayland: use xdg_wm_base when available - wasapi, wasapi2: Fix memory issues - wasapi2: Respect ringbuffer buffer/latency time - wasapi2: Fix choppy rendering gst-plugins-ugly - No changes gst-plugins-rs - No changes gst-libav - No changes gst-rtsp-server - rtsp-stream: clear sockets when leaving bin 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 - No changes gst-examples - No changes Development build environment - No changes Cerbero build tool and packaging changes in 1.22.11 - x264: fix linker path in pc file - taglib: Fix msvc x86 build race Contributors to 1.22.11 Alexander Slobodeniuk, Arnaud Vrac, Edward Hervey, Elizabeth Figura, Guillaume Desmottes, Haihua Hu, Jan Schmidt, Loïc Molinari, Mark Nauwelaerts, Mathieu Duponchelle, Mikhail Rudenko, Nirbheek Chauhan, Olivier Crête, Piotr Brzeziński, Robert Mader, Sebastian Dröge, Seungha Yang, Stéphane Cerveau, Tim-Philipp Müller, Víctor Manuel Jáquez Leal, … 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 1.22.11 - List of Merge Requests applied in 1.22.11 - List of Issues fixed in 1.22.11 1.22.12 The twelfth 1.22 bug-fix release (1.22.12) was released on 29 April 2024. This release only contains bugfixes and security fixes and it should be safe to update from 1.22.x. The GStreamer 1.22 stable series has since been superseded by the GStreamer 1.24 stable release series. Highlighted bugfixes in 1.22.12 - EXIF image tag parsing security fixes - glimagesink, gl/macos: race and reference count fixes - GstPlay, dvbsubenc, alphadecodebin, d3dvideosink fixes - rtpjitterbuffer extended timestamp handling fixes - v4l2: fix regression with tiled formats - ximagesink: fix regression on RPi/aarch64 - Thread-safety fixes - Python bindings fixes - cerbero build fixes with clang 15 on latest macOS/iOS - various bug fixes, build fixes, memory leak fixes, and other stability and reliability improvements gstreamer - basesrc: Clear submitted buffer lists consistently with buffers - inputselector: fix possible clock leak on shutdown - ptpclock: fix double free of domain data during deinit - gst-inspect-1.0: fix –exists for plugins with versions other than GStreamer’s version, like the Rust plugins gst-plugins-base - EXIF image tag parsing security fixes - glimagesink, gl/macos: a couple of race/reference count fixes - typefinding: Handle WavPack block sizes > 131072 - v4l2: fix error in calculating padding bottom for tile format - ximagesink: initialize mask for XISelectEvents gst-plugins-good - pulsedeviceprovider: Add compare_device_type_name function and missing lock - qtdemux: fix wrong full_range offset when parsing colr box - rtpjitterbuffer: Use an extended RTP timestamp for the clock-base - soup: fix thread name - tests: rtpred: fix out-of-bound writes in unit test - v4l2: silence valgrind warning gst-plugins-bad - alphadecodebin: Explicitly pass 64 bit integers as such through varargs - d3d11videosink: disconnect signals before releasing the window - dvbsubenc: fixed some memory leaks and a crash - GstPlay: Update video_snapshot to support playbin3 gst-plugins-ugly - No changes gst-plugins-rs - No changes gst-libav - No changes gst-rtsp-server - No changes gstreamer-vaapi - No changes gstreamer-sharp - No changes gst-omx - No changes gst-python - Don’t link to python for the gi overrides module gst-editing-services - No changes gst-devtools, gst-validate + gst-integration-testsuites - debug-viewer: Fix plugin loading machinery gst-examples - No changes Development build environment - No changes Cerbero build tool and packaging changes in 1.22.12 - glib: disable error for int-conversion introduced by default with clang 15 - cerbero: Fix shutil.rmtree hack to passthrough unknown kwargs Contributors to 1.22.12 Alexander Slobodeniuk, Arnaud Vrac, Elliot Chen, eri, F. Duncanh, Jan Schmidt, Jimmy Ohn, Matthew Waters, Nicolas Dufresne, Nirbheek Chauhan, Philippe Normand, Qian Hu (胡骞), Sebastian Dröge, Taruntej Kanakamalla, Thomas Goodwin, Tim Blechmann, 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 1.22.12 - List of Merge Requests applied in 1.22.12 - List of Issues fixed in 1.22.12 Schedule for 1.24 GStreamer 1.24 was released on 4 March 2024. ------------------------------------------------------------------------ These release notes have been prepared by Tim-Philipp Müller with contributions from Edward Hervey, Matthew Waters, Nicolas Dufresne, Nirbheek Chauhan, Olivier Crête, Sebastian Dröge, Seungha Yang, and Thibault Saunier. License: CC BY-SA 4.0