+GStreamer core
+
+- gst_task_resume(): This new API allows resuming a task if it was
+ paused, while leaving it in stopped state if it was stopped or not
+ started yet. This can be useful for callback-based driver workflows,
+ where you basically want to pause and resume the task when buffers
+ are notified while avoiding the race with a gst_task_stop() coming
+ from another thread.
+
+- info: add printf extensions GST_TIMEP_FORMAT and GST_STIMEP_FORMAT
+ for printing GstClockTime/GstClockTimeDiff pointers, which is much
+ more convenient to use in debug log statements than the usual
+ GST_TIME_FORMAT-followed-by-GST_TIME_ARGS dance. Also add an
+ explicit GST_STACK_TRACE_SHOW_NONE enum value.
+
+- gst_element_get_current_clock_time() and
+ gst_element_get_current_running_time(): new helper functions for
+ getting an element clock’s time, and the clock time minus base time,
+ respectively. Useful when adding additional input branches to
+ elements such as compositor, audiomixer, flvmux, interleave or
+ input-selector to determine initial pad offsets and such.
+
+- seeking: Add GST_SEEK_FLAG_TRICKMODE_FORWARD_PREDICTED to just skip
+ B-frames during trick mode, showing both keyframes + P-frame, and
+ add support for it in h264parse and h265parse.
+
+- elementfactory: add GST_ELEMENT_FACTORY_TYPE_HARDWARE to allow
+ elements to advertise that they are hardware-based or interact with
+ hardware. This has multiple applications:
+
+ - it makes it possible to easily differentiate hardware and
+ software based element implementations such as audio or video
+ encoders and decoders. This is useful in order to force the use
+ of software decoders for specific use cases, or to check if a
+ selected decoder is actually hardware-accelerated or not.
+ - elements interacting with hardware and their respective drivers
+ typically don’t know the actually supported capabilities until
+ the element is set into at least READY state and can open a
+ device handle and probe the hardware.
+
+- gst_uri_from_string_escaped(): identical to gst_uri_from_string()
+ except that the userinfo and fragment components of the URI will not
+ be unescaped while parsing. This is needed for correctly parsing
+ usernames or passwords with : in them .
+
+- paramspecs: new GstParamSpec flag GST_PARAM_CONDITIONALLY_AVAILABLE
+ to indicate that a property might not always exist.
+
+- gst_bin_iterate_all_by_element_factory_name() finds elements in a
+ bin by factory name
+
+- pad: gst_pad_get_single_internal_link() is a new convenience
+ function to return the single internal link of a pad, which is
+ useful e.g. to retrieve the output pad of a new multiqueue request
+ pad.
+
+- datetime: Add constructors to create datetimes with timestamps in
+ microseconds, gst_date_time_new_from_unix_epoch_local_time_usecs()
+ and gst_date_time_new_from_unix_epoch_utc_usecs().
+
+- gst_debug_log_get_lines() gets debug log lines formatted in the same
+ way the default log handler would print them
+
+- GstSystemClock: Add GST_CLOCK_TYPE_TAI as GStreamer abstraction for
+ CLOCK_TAI, to support transmission offloading features where network
+ packets are timestamped with the time they are deemed to be actually
+ transmitted. Useful in combination with the new AVTP plugin.
+
+- miscellaneous utility functions: gst_clear_uri(),
+ gst_structure_take().
+
+- harness: Added gst_harness_pull_until_eos()
+
+- GstBaseSrc:
+
+ - gst_base_src_new_segment() allows subclasses to update the
+ segment to be used at runtime from the ::create() function. This
+ deprecates gst_base_src_new_seamless_segment()
+ - gst_base_src_negotiate() allows subclasses to trigger format
+ renegotiation at runtime from inside the ::create() or ::alloc()
+ function
+
+- GstBaseSink: new stats property and gst_base_sink_get_stats() method
+ to retrieve various statistics such as average frame rate and
+ dropped/rendered buffers.
+
+- GstBaseTransform: gst_base_transform_reconfigure() is now public
+ API, useful for subclasses that need to completely re-implement the
+ ::submit_input_buffer() virtual method
+
+- GstAggregator:
+
+ - gst_aggregator_update_segment() allows subclasses to update the
+ output segment at runtime. Subclasses should use this function
+ rather than push a segment event onto the source pad directly.
+ - new sample selection API:
+ - subclasses should now call gst_aggregator_selected_samples()
+ from their ::aggregate() implementation to signal that they
+ have selected the next samples they will aggregate
+ - GstAggregator will then emit the samples-selected signal
+ where handlers can then look up samples per pad via
+ gst_aggregator_peek_next_sample().
+ - This is useful for example to atomically update input pad
+ properties in mixer subclasses such as compositor.
+ Applications can now update properties with precise control
+ of when these changes will take effect, and for which input
+ buffer(s).
+ - gst_aggregator_finish_buffer_list() allows subclasses to push
+ out a buffer list, improving efficiency in some cases.
+ - a ::negotiate() virtual method was added, for consistency with
+ other base classes and to allow subclasses to completely
+ override the negotiation behaviour.
+ - the new ::sink_event_pre_queue() and ::sink_query_pre_queue()
+ virtual methods allow subclasses to intercept or handle
+ serialized events and queries before they’re queued up
+ internally.
+
+GStreamer Plugins Base Libraries
+
+Audio library
+
+- audioaggregator, audiomixer: new output-buffer-duration-fraction
+ property which allows use cases such as keeping the buffers output
+ by compositor on one branch and audiomixer on another perfectly
+ aligned, by requiring the compositor to output a n/d frame rate, and
+ setting output-buffer-duration-fraction to d/n on the audiomixer.
+
+- GstAudioDecoder: new max-errors property so applications can
+ configure at what point the decoder should error out, or tell it to
+ just keep going
+
+- gst_audio_make_raw_caps() and gst_audio_formats_raw() are
+ bindings-friendly versions of the GST_AUDIO_CAPS_MAKE() C macro.
+
+- gst_audio_info_from_caps() now handles encoded audio formats as well
+
+PbUtils library
+
+- GstEncodingProfile:
+ - Do not restrict number of similar profiles in a container
+ - add GstValue serialization function
+- codec utils now support more H.264/H.265 profiles/levels and have
+ improved extension handling
+
+RTP library
+
+- rtpbasepayloader: Add scale-rtptime property for scaling RTP
+ timestamp according to the segment rate (equivalent to RTSP speed
+ parameter). This is useful for ONVIF trickmodes via RTSP.
+
+- rtpbasepayload: add experimental property for embedding twcc
+ sequencenumbers for Transport-Wide Congestion Control (gated behind
+ the GST_RTP_ENABLE_EXPERIMENTAL_TWCC_PROPERTY environment
+ variable) - more generic API for enabling this is expected to land
+ in the next development cycle.
+
+- rtcpbuffer: add RTPFB_TYPE_TWCC for Transport-Wide Congestion
+ Control
+
+- rtpbuffer: add
+ gst_rtp_buffer_get_extension_onebyte_header_from_bytes()``, so that one can parse theGBytes`
+ returned by gst_rtp_buffer_get_extension_bytes()
+
+- rtpbasedepayload: Add max-reorder property to make the
+ previously-hardcoded value when to consider a sender to have
+ restarted configurable. In some scenarios it’s particularly useful
+ to set max-reorder=0 to disable the behaviour that the depayloader
+ will drop packets: when max-reorder is set to 0 all
+ reordered/duplicate packets are considered coming from a restarted
+ sender.
+
+RTSP library
+
+- add gst_rtsp_url_get_request_uri_with_control() to create request
+ uri combined with control url
+
+- GstRTSPConnection: add the possibility to limit the Content-Length
+ for RTSP messages via
+ gst_rtsp_connection_set_content_length_limit(). The same
+ functionality is also exposed in gst-rtsp-server.
+
+SDP library
+
+- add support for parsing the extmap attribute from caps and storing
+ inside caps The extmap attribute allows mapping RTP extension header
+ IDs to well-known RTP extension header specifications. See RFC8285
+ for details.
+
+Tags library
+
+- update to latest iso-code and support more languages
+
+- add tags for acoustid id & acoustid fingerprint, plus MusicBrainz ID
+ handling fixes
+
+Video library
+
+- High Dynamic Range (HDR) video information representation and
+ signalling enhancements:
+
+ - New APIs for HDR video information representation and
+ signalling:
+ - GstVideoMasteringDisplayInfo: display color volume info as
+ per SMPTE ST 2086
+ - GstVideoContentLightLevel: content light level specified in
+ CEA-861.3, Appendix A.
+ - plus functions to serialise/deserialise and add them to or
+ parse them from caps
+ - gst_video_color_{matrix,primaries,transfer}_{to,from}_iso():
+ new utilility functions for conversion from/to ISO/IEC
+ 23001-8
+ - add ARIB STD-B67 transfer chracteristic function
+ - add SMPTE ST 2084 support and BT 2100 colorimetry
+ - define bt2020-10 transfer characteristics for clarity:
+ bt707, bt2020-10, and bt2020-12 transfer characteristics are
+ functionally identical but have their own unique values in
+ the specification.
+ - h264parse, h265parse: Parse mastering display info and content
+ light level from SEIs.
+ - matroskademux: parse HDR metadata
+ - matroskamux: Write MasteringMetadata and Max{CLL,FALL}. Enable
+ muxing with HDR meta data if upstream provided it
+ - avviddec: Extract HDR information if any and map bt2020-10, PQ
+ and HLG transfer functions
+
+- added bt601 transfer function (for completeness)
+
+- support for more pixel formats:
+
+ - Y412 (packed 12 bits 4:4:4:4)
+ - Y212 (packed 12 bits 4:2:2)
+ - P012 (semi-planar 4:2:0)
+ - P016_{LE,BE} (semi-planar 16 bits 4:2:0)
+ - Y444_16{LE,BE} (planar 16 bits 4:4:4)
+ - RGB10A2_LE (packed 10-bit RGB with 2-bit alpha channel)
+ - NV12_32L32 (NV12 with 32x32 tiles in linear order)
+ - NV12_4L4 (NV12 with 4x4 tiles in linear order)
+
+- GstVideoDecoder:
+
+ - new max-errors property so applications can configure at what
+ point the decoder should error out, or tell it to just keep
+ going
+
+ - new qos property to disable dropping frames because of QoS, and
+ post QoS messages on the bus when dropping frames. This is
+ useful for example in a scenario where the decoded video is
+ tee-ed off to go into a live sink that syncs to the clock in one
+ branch, and an encoding and save to file pipeline in the other
+ branch. In that case one wouldn’t want QoS events from the video
+ sink make the decoder drop frames because that would also leave
+ gaps in the encoding branch then.
+
+- GstVideoEncoder:
+
+ - gst_video_encoder_finish_subframe() is new API to push out
+ subframes (e.g. slices), so encoders can split the encoding into
+ subframes, which can be useful to reduce the overall end-to-end
+ latency as we no longer need to wait for the full frame to be
+ encoded to start decoding or sending out the data.
+ - new min-force-key-unit-interval property allows configuring the
+ minimum interval between force-key-unit requests and prevents a
+ big bitrate increase if a lot of key-units are requested in a
+ short period of time (as might happen in live streaming RTP
+ pipelines when packet loss is detected).
+ - various force-key-unit event handling fixes
+
+- GstVideoAggregator, compositor, glvideomixer: expose
+ max-last-buffer-repeat property on pads. This can be used to have a
+ compositor display either the background or a stream on a lower
+ zorder after a live input stream freezes for a certain amount of
+ time, for example because of network issues.
+
+- gst_video_format_info_component() is new API to find out which
+ components are packed into a given plane, which is useful to prevent
+ us from assuming a 1-1 mapping between planes and components.
+
+- gst_video_make_raw_caps() and gst_video_formats_raw() are
+ bindings-friendly versions of the GST_VIDEO_CAPS_MAKE() C macro.
+
+- video-blend: Add support for blending on top of 16 bit per component
+ formats, which makes sure we can support every currently supported
+ raw video format for blending subtitles or logos on top of video.
+
+- GST_VIDEO_BUFFER_IS_TOP_FIELD() and
+ GST_VIDEO_BUFFER_IS_BOTTOM_FIELD() convenience macros to check
+ whether the video buffer contains only the top field or bottom field
+ of an interlaced picture.
+
+- GstVideoMeta now includes an alignment field with the
+ GstVideoAlignment so buffer producers can explicitly specify the
+ exact geometry of the planes, allowing users to easily know the
+ padded size and height of each plane. Default values will be used if
+ this is not set.
+
+ Use gst_video_meta_set_alignment() to set the alignment and
+ gst_video_meta_get_plane_size() or gst_video_meta_get_plane_height()
+ to compute the plane sizes or plane heights based on the information
+ in the video meta.
+
+- gst_video_info_align_full() works like gst_video_info_align() but
+ also retrieves the plane sizes.
+
+MPEG-TS library
+
+- support for SCTE-35 sections
+
+- extend support for ATSC tables:
+
+ - System Time Table (STT)
+ - Master Guide Table (MGT)
+ - Rating Region Table (RRT)
+
+Miscellaneous performance, latency and memory optimisations