# Changelog
All notable changes to this project will be documented in this file.

The format is based on [Keep a Changelog](http://keepachangelog.com/en/1.0.0/)
and this project adheres to [Semantic Versioning](http://semver.org/spec/v2.0.0.html),
specifically the [variant used by Rust](http://doc.crates.io/manifest.html#the-version-field).

## [0.15.3] - 2026-06-30
### Fixed
- fmp4mux: Fix draining in chunk mode after partial GOPs were drained.
- fmp4mux: Fix fragment / chunk decision for chunk-mode=keyframe.
- fmp4mux: Fix late GOP handling.
- fmp4mux: Fix delayed push when GOPs start after fragment end.
- fmp4mux: Split oversized GOPs at fragment boundary.
- onvifmeta2relatiometa: Fix inversion between width and height.
- rtspsrc2: Handle parse errors with TCP interleaved RTSP more gracefully.
- rtprecv: Fix buffer list split handling.
- webrtc: Set level in negotiated caps only if level asymmetry not allowed.
- webrtcsrc: Fix `.fixate()` causing panic during SDP renegotiation.
- webrtcsrc: Prevent duplicate src pad creation for existing streams.
- webrtcsrc: Set transceiver direction to Inactive for removed streams before
  renegotiation.
- webrtcsink: Allow renegotiation if caps is missing interlace-mode.
- webrtcsink: Allow change of interlace-mode in the input caps.
- webrtcsink: Fix negotiation for nvh264enc in the GLMemory case.
- webrtcsink: Handle payloader timestamp-offset prop type variants.
- webrtcsink: Read rav1enc bitrate as i32.

### Added
- gopbuffer: Add support for H.266/VVC.
- rtspsrc2: Add support for authentication, client auth with TLS, HTTP tunneling,
  keep alive, latency configuration, SRTP, SET_PARAMETER / GET_PARAMETER via
  signals, stream support, TLS validation flags, user-agent in tunnel requests,
  and URI handler updates.
- threadshare: Add ts-clocksync.
- webrtcsink: Add support for h264 profile constrained-high.

### Changed
- Various dependencies were updated.

## [0.15.2] - 2026-05-11
### Fixed
- fallbacksrc: Send select-streams to the actual stream collection source.
- gtk4paintablesink: Error out in NULL->READY if no default GDK display exists.
- quinnwt: Fix error when closing sessions.
- rspngenc: Convert strides if necessary.
- rtpbin2: Improve panic / error handling.
- rtprecv: Fix deadline calculation on re-ordered packets.
- rtprecv: Fix race condition on flush-start handling.
- rtprecv: Improve jitter calculation.
- speechmatics: Handle disconnects more gracefully.
- textaccumulate: Improve behaviour on flushing.
- threadshare: Some memory leaks were fixed, specifically around `ts-udpsink`
  sockets.
- ts-udpsink: Don't error out on packet send failure.
- ts-udpsrc: Handle various packet receive failures more gracefully.
- webrtcsink: Allow custom signaller to actually be set.
- whisper: Fix compilation on ARM.

### Changed
- Various dependencies were updated.
- Various new Rust 1.94 / 1.95 clippy warnings were fixed.
- burn-yoloxinference: Restrict widths/heights to a multiple of 32.
- rtprecv: Various performance improvements.

### Added
- fallbacksrc: Add property to disable dummy fallback source and to set
  fallback source from a bin.
- threadshare: Add leaky properties to queue-like elements.
- ts-udpsink: Add reuse property.
- webrtcsink: Add imx8 encoder support.

## [0.15.1] - 2026-02-28
### Fixed
- Don't convert push_event() false returns into flow errors.
- rtp: rtprecv: Send mandatory events on the RTCP srcpad before buffers.
- rtp: rtpav1pay: Insert sequence header if a keyframe is missing it.
- speechmaticstranscriber: Fix first buffer PTS race condition.

## [0.15.0] - 2026-02-21
### Fixed
- audiornnoise: Fix audio level value reporting.
- awspolly: Fix overflow budget calculation.
- awspolly: Fix various panics.
- awstranslate/awstranscriber2: Post error messages on connection errors.
- dav1ddec: Various fixes to allocation query handling.
- elevenlabssynthesizer: Improve gap handling in non-clip overflow modes.
- gifenc: Avoid unnecessary resets on flush/caps change.
- gtk4paintablesink: Fix odd-size subsampling workaround.
- isobmff: Fix writing of EAC3 stream metadata.
- isobmff: Make TAI timestamp support spec compliant.
- isobmff: Write more complete vpcC box for VP8/VP9.
- isobmff: fmp4: Don't overlap durations on fragment header buffers.
- ndisrc: Fix audio corruption related to stride padding.
- quinn: Various bugfixes and improvements.
- livesync: Various fixes related to reverse playback and segment handling.
- rtp: basepay: Consider size of header extensions in maximum payload size
  calculation.
- rtprecv: Various bugfixes and performance improvements.
- rtspsrc2: Fix `appsrc` max-time configuration.
- sccparse: Handle streams with more than one byte tuple.
- speech synthesizers: Various bugfixes and improvements.
- speechmaticstranscriber: Fix handling of diarization=none.
- st2038: Various fixes and support for more alignments.
- threadshare: Some deadlock fixes.
- transcriberbin: Fix latency reporting in various situations.
- webrtcsink: Fix deadlock when recalculating latency.
- webrtcsink: Fix deadlock between remote description and ICE handling.
- webrtcsink: Handle NVIDIA encoder interfaces changes between versions / platforms.
- webrtc: Only register data channels for internal use.
- webrtc: Various minor fixes and improvements.

- ... and many more fixes and minor improvements!

### Added
- analytics: Add YOLOX tensor decoder.
- audiornnoise: Forward input metadata.
- awss3: Support for S3-compatible URIs.
- awstranscriber2: Add property for setting `show_speaker_labels`.
- awstranslate: Expose property for turning brevity on.
- burn: Add burn-based YOLOX inference element.
- cea608: Non-relative positioning support.
- cea708mux: Expose `discarded-services` property on sink pads.
- debugseimetainserter: New plugin for testing SEI meta insertion.
- deepgram: New speech-to-text transcription element.
- demucs: Add audio source separation element based on demucs.
- fallbacksrc: Support for encoded inputs.
- gtk4paintablesink: Propose udmabuf allocator/pool if possible.
- gtk4paintablesink: New property for configuration window-resize behaviour.
- hlscmafsink: Support for I-frame-only playlists.
- elevenlabs: New voice cloner element.
- gif: New decoder element with support for animations and looping.
- icecastsink: New icecast sink element.
- inter: Various new properties to fine-tune behaviour.
- isobmff: Implement `gst::ChildProxy` for accessing sinkpads.
- isobmff: Support for raw audio (ISO 23003-5).
- isobmff: Support for bayer video formats.
- isobmff: Write (video) compressorname field.
- isobmff: mp4mux: Add (limited) support for caps changes.
- jsontovtt: Add property to enable per-cue line attributes.
- mccparse: Add support for non-caption VANC data.
- quinn: Support for QUIC/WebTransport connection/session sharing.
- quinn: Improve TLS certificate configuration.
- rtp: rtpmparobustdepay2: New depayloader.
- rtp: smpte291: New payloader for SMPTE291 ancillary data.
- speechmatics: Expose `mask-profanities` property.
- speechmatics: Support for configuring audio events.
- speechmatics: Add properties to control speaker identification.
- s302mparse: New parser for S302M audio.
- st2038: Add combiner/extractor elements for converting between ST2038
  streams and `GstAncillaryMeta`.
- speech synthesizers: New compression overflow mode.
- textaccumulate: New element for accumulating text.
- threadshare: ts-udpsink: Allow disabling IPv4 or IPv6 socket.
- tttocea608: Expose `speaker-prefix` property.
- validate: Add plugin with `check-last-frame-qrcode` validate action.
- webrtc: Add WHEP client and server signallers and whepclientsrc /
  whepserversink elements.
- webrtcsink: Add support for VA encoders.
- webrtcsink: Add support for renegotiation.
- webrtcsink: Negotiate H264 profile / level correctly.
- webrtcsink: Support for specifying customs HTTP headers to signalling
  server.
- webrtcsrc: Support forwarding encoded data.
- webrtcsrc: Support request source pads.
- whisper: New speech-to-text transcription element.

### Changed
- Updated to gtk-rs-core 0.22 APIs.
- Updated with final GStreamer 1.28.0 APIs and upcoming 1.30 APIs.
- Updated minimum supported Rust version to 1.92.

- awspolly: Port improvements from elevenlabssynthesizer.
- isobmff: fmp4 and mp4 plugins are merged into a single plugin.
- gtk4: Require at least GStreamer 1.18 instead of 1.16.
- mccparse: Outputs ST2038 now instead of CEA608/708.
- quinn: Rename quinnwtserversink to quinnsink and quinwtclientsrc to
  quinnsrc.
- threadshare: Relicense to MPL-2.0.

### Removed
- aws: Legacy rusotos3src / rusotos3sink element factories are dropped now.
- threadshare: ts-rtpjitterbuffer: All its use-cases are covered by `rtprecv`.
- webrtchttp: Not removed yet but deprecated in favour of the webrtc plugin.

## [0.14.5] - 2026-02-28
### Fixed
- audiornnoise: Fix audio level value reporting.
- awspolly: Fix various panics.
- dav1ddec: Various fix to allocation query handling.
- gtk4paintablesink: Fix odd-size subsampling workaround.
- isobmff: fmp4: Don't overlap durations on fragment header buffers.
- livesync: Ignore upstream latency when upstream is not live.
- rtp: basepay: Consider size of header extensions in maximum payload size
  calculation.
- rtp: rtprecv: Don't panic if no buffers of a bufferlist can be directly forwarded
- rtp: rtprecv: Send mandatory events on the RTCP srcpad before buffers.
- rtp: smpte291: Use video media type instead of application.
- rtp: smpte291: Use upper-case encoding-name.
- rtp: smpte291: Drop the current packet after processing if it was empty.
- speechmaticstranscriber: Fix handling of diarization=none.
- st2038combiner: Various minor fixes.
- transcriberbin: Fix latency reporting in various situations.
- webrtcsink: Don't upscale when mitigating low bitrate.
- webrtcsink: Handle NVIDIA encoder interfaces changes between versions / platforms.
- webrtc: Only register data channels for internal use.

### Added
- gtk4paintablesink: Propose udmabuf allocator/pool if possible.
- gtk4paintablesink: New property for configuration window-resize behaviour.
- st2038extractor: Add `always-add-st2038-pad` property.
- threadshare: ts-udpsink: Allow disabling IPv4 or IPv6 socket.

### Changed
- gtk4: Require at least GStreamer 1.18 instead of 1.16.

## [0.14.4] - 2025-12-02
### Fixed
- analyticssplitter / combiner: Update to latest batch meta API.
- audiornnoise: Copy input metadata to output.
- fmp4mux: Fix EAC-3 datarate calculation and substream writing.
- fmp4mux: Fix potential deadlock when requesting a pad.
- gtk4paintablesink: Add property to control the reconfigure-event behaviour.
- jsongstparse: Don't send error messages on flushing
- mccparse / sccparse: Don't send error messages on flushing
- mp4mux: Fix potential deadlock when requesting a pad.
- ndisrcdemux: Fix audio corruption when padding is used for non-interleaved audio.
- rtpsmtpe291: New RTP payloader / depayloader for SMPTE 291 ancillary data.
- st2038: Forward framerate in caps if available and set frame alignment.
- st2038: Fix checksum calculation.
- webrtcsink: Fix deadlock
- webrtcsink: Don't block tokio runtime thread when recalculating latency.
- webrtcsrc: Clean up EOS/session handling.

### Added
- fmp4mux: Implement childproxy interface.
- mp4mux: Implement childproxy interface.
- st2038extractor / combiner: New elements.
- webrtcsink: Add support for VA encoders.

### Changed
- Fix a couple of new clippy / compiler warnings.
- Update some dependencies.

## [0.14.3] - 2025-10-31
### Fixed
- fallbacksrc: Fix custom source re-use.
- gtk4paintablesink: Implement cropped imports for odd resolutions and padded
  input frames.
- livekitwebrtc: Fix deadlock.
- onvifmetadatapay: Copy metadata from source buffer.
- reqwest: Use the platform's native certificates again.
- rtpamrpay2: Set frame quality indicator flag correctly.
- rtpbasepay2: Re-use last PTS if no PTS is available.
- rtpbaseaudiopay2: Fix marker bit handling.
- rtp: Add L8/16/24 RTP payloaders / depayloaders.
- threadshare: Fix/add latency reporting.
- threadshare: Fix shutdown in error states.
- threadshare: Fix flush-start and query handling.
- ts-audiotestsrc: Act as a pseudo live source by default.
- tracers: Fix inverted append logic when writing log files.
- uriplaylistbin: Add details into upstream error message.

### Added
- ts-audiotestsrc: Support more audio formats.

### Changed
- Various dependency updates.
- Remove constraints on AWS SDK dependencies and tune optimizations.

## [0.14.2] - 2025-09-18
### Fixed
- analyticscombiner: Fix running with latest GStreamer git main.
- aws: Ensure pad tasks are stopped during PAUSED->READY state change.
- cea608overlay: Support non-system memory properly.
- fallbacksrc: Fix panic during retries in parallel with element shutdown.
- fallbacksrc: Don't restart source if element is being shut down.
- fallbacksrc: Fix deadlock when using a custom source element.
- fallbacksrc: Signal no-more-pads for streams-unaware parent elements.
- gtk4paintablesink: Also try importing dmabufs if not signalled via caps.
- gtk4paintablesink: Add support for 10/12/16 bit YUV formats in sysmem.
- rtpgccbwe: Avoid panic when min-bitrate > max-bitrate.
- rtpmp4gdepay2: Fix handling of specific caps.
- rtprecv: Fix deadlock race condition when receiving the first buffer.
- spotify: Fixes for spotify API changes that broke the element.
- ts-audiotestsrc: Fix deadlock when setting samples-per-buffer property.
- ts-inter: Fix latency handling.
- ts-intersink: Fix deadlock on shutdown.
- threadshare: Fix potential panics during state changes.
- transcriberbin: Fix various deadlocks.
- webrtcsink: Fix setting URI and CA file in signaller.

### Added
- intersink: Allow setting sync=false.
- intersrc: Allow setting bytes/time/buffers limits.
- ts-blocking-adapter: New element to translate between threadshare upstream
  elements and blocking downstream elements.

### Changed
- Various elements were switched from tokio-native-tls to rustls due to the
  former being unmaintained and buggy.
- Various dependency updates.

## [0.14.1] - 2025-08-10
### Fixed
- Various new clippy 1.89 warnings.
- awstranscriber2 / awstranslate: Handle multiple stream-start events.
- hlssmultivariantsink: Set correct master playlist version.
- mpegtslivesrc: Remove leftover debug message.
- rtpbasepay2: Fix `timestamp` property's range.
- rtpbin2: Add examples.
- rtprecv: Fix deadlock when receiving muxed RTCP on the RTP pad.
- threadshare: Fix flushing in various elements.
- transcriberbin: Fix latency handling.
- transcriberbin: Don't hold mutex while sending caps event.
- ts-audiotestsrc: Fix element to behave in a more expected way.
- ts-proxysrc / ts-proxysink: Fix race condition on shutdown.
- ts-rtpdtmfsrc: Fix various property ranges.
- ts-intersrc / ts-udpsrc / ts-tcpclientsrc: Fix latency handling.

### Added
- cea608overlay / cea708overlay: Add support for dmabuf, GL and other memory
  types.
- threadshare: Add `cur-level-XXX` properties to queue-like elements.

### Changed
- Update a couple of dependencies.
- pipeline-snapshot: Reduce WebSocket connection log level.
- webrtcsink: Move `videorate` before conversion to potentially improve
  performance.

## [0.14.0] - 2025-07-16
### Changed
- Update to gtk-rs-core 0.21 / gtk4-rs 0.10 / gstreamer-rs 0.24.
- Update various dependencies.
- Update MSRV to 1.83.
- gtk4: Update minimum GTK version to 4.6 but require 4.10 by default.

### Added
- analytics: New plugin with `relationmeta2onvifmeta` element that converts
  relation metadata to ONVIF metadata.
- analytics: New `analyticscombiner` and `analyticssplitter` elements that
  allow temporal batching of one or more streams.
- awspolly: New AWS Polly based text-to-speech element.
- cdpserviceinject: New element to inject CEA708 CDP service information.
- dav1ddec: Support decoding into downstream-provided buffer pools.
- elevenlabs: New speech synthesis elements based on ElevenLabs.
- fallbacksrc: Add multi-stream support.
- fmp4mux: Add `send-headers` and `split-at-running-time` action signals.
- fmp4mux: Add support for caps changes.
- fmp4mux: Add support for writing `edts` to handle audio priming.
- fmp4mux: Add support for serialized `split-now` event.
- fmp4mux: Add `send-force-keyunit`, `decode-time-offset` and
  `start-fragment-sequence-number` properties.
- fmp4mux: Write `prft` and `btrt` boxes.
- fmp4mux: Add support for AC-3/EAC-3.
- gtk4: Add colorimetry support.
- gtk4: Add support for YUV memory textures.
- hlscmafsink: Add `new-playlist` signal and `playlist-root-init` property.
- hlssink3: Support NTP timestamp metadata.
- janusvrwebrtcsrc: New Janus VideoRoom source element.
- memory-tracer: New tracer to track memory usage.
- mp4mux: Add support for edit lists and handle audio priming.
- mp4mux: Support ISO/IEC 23001-17 uncompressed video.
- mp4mux: Support HEIF output.
- mp4mux: Write `btrt` box.
- mp4mux: Add support for `taic` clock information box.
- ndisrc: Add new clocked timestamp mode that provides a `gst::Clock` to the
  pipeline.
- onvimetadataextractor: New element that extracts ONVIF metadata from buffers
  that can then be used by `rtponvifpay` for example.
- pcap-writer: New tracer to write streams of arbitrary pads as PCAP files.
- pipeline-snapshot: New tracer that allows to create snapshots of pipelines.
- quinn: Support multiple stream connections in `quinnquicsrc`.
- quinnquicmux/demux: New elements to support QUIC stream multiplexing and
  support for it in the other elements.
- quinnroqmux/demux: New elements to support RTP over QUIC (ROQ).
- quinnserversink/clientsrc: New elements for QUIC-based WebTransport.
- rtpamrpay2/depay2: New AMR NB/WB RTP payloader / depayloader.
- rtpbin2: Reduce number of threads and make better use of thread pools.
- skia: New skia-based compositor element.
- speechmatics: New transcriber element based on Speechmatics.
- spotifylyricssrc: New element to retrieve synchronized lyrics from Spotify.
- closedcaption: Set of new elements that handle ST2038 streams and allow
  extracting / inserting closed captions into them: `st2038ancdemux`,
  `st2038ancmux`, `st2038anctocc` and `cctost2038anc`.
- streamgrouper: New element that allows combining streams with different
  group-ids in their stream-start events to use the same one.
- transcriberbin: Various improvements and fixes.
- transcriberbin: Add support for speech synthesis.
- ts-intersrc/sink: New 1:N inter pipeline elements.
- ts-rtpdtmfsrc: New RTP DTMF source element.
- ts-proxysink: Add `event-types` property.
- ts-udpsrc: Add `buffer-size` and `loop` properties.
- ts-udpsrc/udpsink: Add `multicast-iface` property.
- vvdec: New VVC/H266 decoder element using VVdeC.
- webrtc: Add support for raw payload formats.
- webrtcsink: Add support for answering SDP offers.
- webrtcsink: Add generic data channel control mechanism and generic mechanism
  to forward metas over the control channel.
- webrtcsink: Add `define-encoder-bitrates` signal for customizing congestion
  control behaviour.
- webrtcsink: Add signal to configure congestion control mitigation modes.

### Fixed
- rtpgccbwe: Handle out-of-order packets better.

## [0.13.7] - 2025-07-16
### Fixed
- awss3sink: Write to S3 on output stream flush.
- cea608overlay: Reset output size on flush stop.
- cea708mux: Improve caption overflow handling.
- cea708mux: Correctly clip input buffers on the segment.
- cea708overlay: Reset output size on flush stop.
- dav1ddec: Use video decoder base class API for latency reporting and notify
  application about latency changes.
- gopbuffer: Push GOPs in the correct order on EOS.
- hlssink3: Use correct closed segment location when writing playlist.
- janusvrwebrtcsink: Various websocket handling improvements.
- mccparse: Use correct byte representation for "U".
- rtpbin2: Fix race condition when handling serialized queries.
- rtpbin2: Fix panic caused by race condition when retrieving clock-rate.
- rtpbin2: Send SSRC collision event in the correct direction.
- rtpbin2: Fix usage of min RTCP interval in senders.
- rtpbin2: Drop packets with an unknown payload type or clock-rate instead of
  panicking.
- rtpbin2: Improve detection of RTP inline, rtcp-mux RTCP packets.
- tttocea608: Disallow Pango markup as input.
- tttocea708: Handle caption overflows better.
- tttocea708: Disallow Pango markup as input.
- webrtcsink: Fix a couple of deadlocks.
- webrtcsink: Don't require an UUID for discovery to speed up startup on
  systems with low entropy.

## [0.13.6] - 2025-05-13
### Fixed
- buffer-lateness: Avoid integer overflows when logging.
- cdg: Fix division by zero in the typefinder.
- cea708mux: Improve support for overflowing input captions.
- dav1ddec: Use downstream buffer pool to copy into if videometa is not
  supported by downstream.
- dav1ddec: Fix handling of incomplete colorimetry information.
- dav1ddec: Drain decoder on caps changes.
- fmp4mux: Fix latency configuration for properties set during construction.
- fmp4mux: Write v0 tfdt box if decode time is small enough for improved
  compatibility.
- fmp4mux: Fix handling of multiple tags per taglist.
- fmp4mux: Parse language tags correctly as ISO 639-2T.
- fmp4mux: Fix tfdt value to be actually according to the spec.
- fmp4mux: Fix handling of negative DTS in composition time offset.
- gtk4paintablesink: Consider surface scale factor when proposing window
  dimensions to improve rendering quality with scale factors > 1.
- mp4mux: Don't write composition time offsets if they're all zero.
- mp4mux: Fix handling of multiple tags per taglist.
- mp4mux: Store language tags per stream and not globally.
- mp4mux: Parse language tags correctly as ISO 639-2T.
- mpegtslivesrc: Fix deadlock caused by pushing buffers downstream while
  keeping the state locked.
- mpegtslivesrc: Increase threshold for PCR / PTS discont detection.
- tttocea708: Fix origin-row handling for roll-up modes.
- webrtc/signalling: Don't error out on messages from unknown sessions.
- webrtcsink: Emit signals without holding mutexes, fix locking order and
  various deadlocks.

### Added
- dav1ddec: Add support for RGB encoded AV1.
- fmp4mux: Write lmsg compatibility brand into the last fragment.

### Changed
- Various updated dependencies.
- colordetect: Move to videofilter base class and allow working in passthrough mode.

## [0.13.5] - 2025-03-04
### Fixed
- cdg: Fix typefind errors on specific file sizes.
- cea608overlay: Ensure lines are rendered in order.
- cea608overlay: Clear output on each switch.
- cea608overlay / cea708overlay: Fix field lookup for S334-1A captions.
- cea608tocea708: Fix S334-1A field flag usage.
- closedcaption: Fix rollup mode not always using the correct base row.
- closedcaption: Only increase dtvcc packet sequence if there are services.
- fmp4mux: Fix state cleanup on flush.
- fmp4mux: Handle language/orientation tags as per-stream tags.
- hlssink3: Write playlist atomically.
- inter: Don't leak hashmap objects.
- mpegtslivesrc: Handle zero-byte adaptation fields correctly.
- mpegtslivesrc: Consider initial calibration of the clock.
- mpegtslivesrc: Ignore NIT programs from the PAT.
- onvifmetadatacombiner: Unset PTS/DTS of metadata.
- rtpbasepay / rtpbasedepay: Only forward buffers after a segment event.
- rtpac3depay2: Fix handling of non-fragmented payloads.
- togglerecord: Drop locks before sending queries to avoid deadlocks.
- tttocea708: Don't reset service writer for every incoming caption.
- whipserversrc: Handle concurrent POSTs.

### Added
- mpegtslivesrc: Take adaptation field discontinuity flag into account.
- uriplaylistbin: Add caching support

### Changed
- Updated various dependencies.

## [0.13.4] - 2024-12-24
### Fixed
- cea608overlay: Fix rendering when roll-up base row is at the top.
- cea708mux: Handle CEA608 data correctly and output padding by default.
- cea708mux: Clear leftover pending codes correctly.
- cea708overlay: Produce better CEA608 layouts.
- cea708overlay: Fix background/foreground types and enable black background by default.
- cea708overlay: Clear correctly on caption timeout.
- mpegtslivesrc: Various fixes related to stream discontinuities.
- tttocea708: Fix various conformance issues.
- togglerecord: Fix various deadlocks and simplify mutexes.
- webrtcsink: Fix various deadlocks.
- webrtcsink: Set caps-change-mode=delayed on encoder capsfilter.
- webrtcsink: Ignore more fields on caps changes.

### Added
- awss3putobjectsink: Add next-file support.
- tracers: Add signal to force writing log file to queue-levels and buffer-lateness tracers.
- webrtc: Handle some more Janus events.
- webrtcsink: Add support for openh264enc and nvh265enc.
- webrtcsrc: Add connect-to-first-producer property.

## [0.13.3] - 2024-11-02
### Fixed
- gtk4paintablesink: Don't check for a GL context when filtering dmabuf caps.
- gtk4paintablesink: Use a correctly typed None value when retrieving
  paintable property fails.
- mpegtslivesrc: Parse PAT/PMT to lock to a single program/PCR in case
  multiple are in the stream.
- rtp: Fix reference timestamp meta de-duplication in depayloaders.
- quinn: Specify a default crypto provider to avoid conflicts.
- transcriberbin: Fix linking of user-provided transcriber.
- webrtcsink: Allow pixel-aspect-ratio changes.
- webrtcsink: Fix naming of error dot files of discovery pipelines.
- webrtcsink: Fix session not in place errors.
- webrtc: janus: Do not block in end_session().

### Added
- awstranscriber: Post warning message with details when items are too late.
- transcriberbin: Support both latency and translate-latency properties.
- webrtc: janus: Add janus-state property.

### Changed
- gtk4paintablesink: Deprecated "wayland" feature and call it "waylandegl" as
  it has nothing to do with generic Wayland support.

## [0.13.2] - 2024-09-28
### Fixed
- cea608overlay: Avoid overflow when deciding which lines to retain.
- cea708mux: Actually push gap events downstream.
- cea708mux: Stop with EOS once all pads are EOS.
- cea708mux: Fix off-by-one when deciding if a buffer belongs to this or the
  next frame.
- mpegtslivesrc: Various timestamp tracking fixes.
- onvifmetadatapay: Set output caps earlier.
- transcriberbin: Fix passthrough state change.
- webrtcsink: Fix setting of RFC7273 attributes in the SDP.

### Added
- dav1ddec: Add properties for film grain synthesis and in-loop filters.
- mpegtslivesrc: Handle PCR discontinuities.
- rtpav1depay: Add wait-for-keyframe and request-keyframe properties.
- webrtcsrc: Expose msid property on source pads.

### Changed
- spotify: Reduce dependencies.

## [0.13.1] - 2024-08-27
### Fixed
- transcriberbin: Fix gst-inspect with missing elements.
- gtk4paintablesink: Move dmabuf cfg to the correct bracket level.
- webrtcsrc: Don't hold the state lock while removing sessions.
- rtpbasepay: Various fixes to payloader base class.
- webrtcsink: Fix various assertions when finalizing.
- webrtcsrc: Make sure to always call end_session() without state lock.
- mpegtslivesrc: Handle PCR discontinuities as errors.
- ndisrc: Calculate timestamps for metadata buffers too.
- Various new clippy warnings.
- webrtcsink: Fix segment format mismatch when using a remote offer.
- awstranscriber: Fix sanity check in transcribe loop.
- whepsrc: Fix incorrect default caps.

### Changed
- gtk4paintablesink: Enable `gtk::GraphicsOffload::black-background` when
  building with GTK 4.16 or newer.
- gstwebrtc-api: Always include index file in dist for convenience.
- rtpbasepay: Negotiate SSRC/PT with downstream via caps for backwards
  compatibility.
- hlssink3: Use more accurate fragment duration from splitmuxsink if
  available.

### Added
- gtk4paintablesink: Add `window-width` and `window-height` properties.
- gtk4paintablesink: Add custom widget for automatically updating window size.
- fmp4mux / mp4mux: Add image orientation tag support.
- webrtcsink: Add nvv4l2av1enc support.
- cmafmux: Add Opus support.

## [0.13.0] - 2024-07-16

### Added
- rtp: New RTP payloader and depayloader base classes, in addition to new
  payloader and depayloaders for: PCMA, PCMU, AC-3, AV1 (ported to the new
  base classes), MPEG-TS, VP8, VP9, MP4A, MP4G, JPEG, Opus, KLV.
- originalbuffer: New pair of elements that allows to save a buffer, perform
  transformations on it and then restore the original buffer but keeping any
  new analytics and other metadata on it.
- gopbuffer: New element for buffering an entire group-of-pictures.
- tttocea708: New element for converting timed text to CEA-708 closed captions.
- cea708mux: New element for muxing multiple CEA-708 services together.
- transcriberbin: Add support for generating CEA-708 closed captions and
  CEA-608-in-708.
- cea708overlay: New overlay element for CEA-708 and CEA-608 closed captions.
- dav1ddec: Signal colorimetry in the caps.
- webrtc: Add support for RFC7273 clock signalling and synchronization to
  webrtcsrc and webrtcsink.
- tracers: Add a new pad push durations tracer.
- transcriberbin: Add support for a secondary audio stream.
- quinn: New plugin with a QUIC source and sink element.
- rtpgccbwe: New mode based on linear regression instead of a kalman filter.
- rtp: New rtpsend and rtprecv elements that provide a new implementation of
  the rtpbin element with a separate send and receive side.
- rtpsrc2: Add support for new rtpsend / rtprecv elements instead of rtpbin.
- webrtcsrc: Add multi-producer support.
- livesync: Add sync property for enabling/disabling syncing of the output
  buffers to the clock.
- mpegtslivesrc: New element for receiving an MPEG-TS stream, e.g. over SRT or
  UDP, and exposing the remote PCR clock as a local GStreamer clock.
- gtk4paintablesink: Add support for rotations / flipping.
- gtk4paintablesink: Add support for RGBx formats in non-GL mode.

### Fixed
- livesync: Queue up to latency buffers instead of requiring a queue of the
  same size in front of livesync.
- livesync: Synchronize the first buffer to the clock too.
- livesync: Use correct duration for deciding whether a filler has to be
  inserted or not.
- audioloudnorm: Fix possible off-by-one in the limiter when handling the very
  last buffer.
- webrtcsink: Fix property types for rav1enc.

### Changed
- sccparse, mccparse: Port from nom to winnow.
- uriplaylistbin: Rely on uridecodebin3 gapless logic instead of
  re-implementing it.
- webrtc: Refactor of JavaScript API.
- janusvrwebrtcsink: New use-string-ids property to distinguish between
  integer and string room IDs, instead of always using strings and guessing
  what the server expects.
- janusvrwebrtcsink: Handle more events and expose some via signals.
- dav1ddec: Require dav1d 1.3.0.
- closedcaption: Drop libcaption C code and switch to a pure Rust
  implementation.

## [0.12.7] - 2024-06-19
### Fixed
- aws, spotifyaudiosrc, reqwesthttpsrc, webrtchttp: Fix race condition when unlocking
- rtp: Allow any payload type for the AV1 RTP payloader/depayloader
- rtp: Various fixes to the AV1 RTP payloader/depayloader to work correctly
  with Chrome and Pion
- meson: Various fixes to the meson-based build system around cargo
- webrtcsink: Use correct property names for configuring `av1enc`
- webrtcsink: Avoid lock poisoning when setting encoder properties

### Added
- ndi: Support for NDI SDK v6
- webrtcsink: Support for AV1 via `nvav1enc`, `av1enc` or `rav1enc`

### Changed
- Update to async-tungstenite 0.26

## [0.12.6] - 2024-05-23
### Fixed
- Various Rust 1.78 clippy warnings.
- gtk4paintablesink: Fix plugin description.

### Added
- fmp4mux / mp4mux: Add support for adding AV1 header OBUs into the MP4
  headers.
- fmp4mux / mp4mux: Take track language from the tags if provided.
- gtk4paintablesink: Add GST_GTK4_WINDOW_FULLSCREEN environment variable to
  create a fullscreen window for debugging purposes.
- gtk4paintablesink: Also create a window automatically when called from
  gst-play-1.0.
- webrtc: Add support for insecure TLS connections.
- webrtcsink: Add VP9 parser after the encoder.

### Changed
- webrtcsink: Improve error when no discovery pipeline runs.
- rtpgccbwe: Improve debug output in various places.

## [0.12.5] - 2024-04-29
### Fixed
- hrtfrender: Use a bitmask instead of an int in the caps for the channel-mask.
- rtpgccbwe: Don't log an error when pushing a buffer list fails while stopping.
- webrtcsink: Don't panic in bitrate handling with unsupported encoders.
- webrtcsink: Don't panic if unsupported input caps are used.
- webrtcsrc: Allow a `None` producer-id in `request-encoded-filter` signal.

### Added
- aws: New property to support path-style addressing.
- fmp4mux / mp4mux: Support FLAC instead (f)MP4.
- gtk4: Support directly importing dmabufs with GTK 4.14.
- gtk4: Add force-aspect-ratio property similar to other video sinks.

## [0.12.4] - 2024-04-08
### Fixed
- aws: Use fixed behaviour version to ensure that updates to the AWS SDK don't
  change any defaults configurations in unexpected ways.
- onvifmetadataparse: Fix possible deadlock on shutdown.
- webrtcsink: Set `perfect-timestamp=true` on audio encoders to work around
  bugs in Chrome's audio decoders.
- Various clippy warnings.

### Changed
- reqwest: Update to reqwest 0.12.
- webrtchttp: Update to reqwest 0.12.

## [0.12.3] - 2024-03-21
### Fixed
- gtk4paintablesink: Fix scaling of texture position.
- janusvrwebrtcsink: Handle 64 bit numerical room ids.
- janusvrwebrtcsink: Don't include deprecated audio/video fields in publish
  messages.
- janusvrwebrtcsink: Handle various other messages to avoid printing errors.
- livekitwebrtc: Fix shutdown behaviour.
- rtpgccbwe: Don't forward buffer lists with buffers from different SSRCs to
  avoid breaking assumptions in rtpsession.
- sccparse: Ignore invalid timecodes during seeking.
- webrtcsink: Don't try parsing audio caps as video caps.

### Changed
- webrtc: Allow resolution and framerate changes.
- webrtcsrc: Make producer-peer-id optional.

### Added
- livekitwebrtcsrc: Add new LiveKit source element.
- regex: Add support for configuring regex behaviour.
- spotifyaudiosrc: Document how to use with non-Facebook accounts.
- webrtcsrc: Add `do-retransmission` property.

## [0.12.2] - 2024-02-26
### Fixed
- rtpgccbwe: Don't reset PTS/DTS to `None` as otherwise `rtpsession` won't be
  able to generate valid RTCP.
- webrtcsink: Fix usage with 1.22.

### Added
- janusvrwebrtcsink: Add `secret-key` property.
- janusvrwebrtcsink: Allow for string room ids and add `string-ids` property.
- textwrap: Don't split on all whitespaces, especially not on non-breaking
  whitespace.

## [0.12.1] - 2024-02-13
### Added
- gtk4: Create a window for testing purposes when running in `gst-launch-1.0`
  or if `GST_GTK4_WINDOW=1` is set.
- webrtcsink: Add `msid` property.

## [0.12.0] - 2024-02-08
### Changed
- ndi: `ndisrc` passes received data downstream without an additional copy, if
  possible.
- webrtc: Cleanups to webrtcsrc/sink default signalling protocol, JavaScript
  implementation and server implementation.
- webrtc: `whipwebrtcsink` is renamed to `whipclientsink` and deprecate old
  `whipsink`.

### Fixed
- gtk4: Fix Windows build when using EGL.
- gtk4: Fix ARGB pre-multiplication with GTK 4.14. This requires building with
  the `gtk_v4_10` or even better `gtk_v4_14` feature.
- gtk4: Fix segfault if GTK3 is used in the same process.
- gtk4: Always draw background behind the video frame and not only when
  borders have to be added to avoid glitches.
- livekitwebrtcsink: Add high-quality layer for video streams.
- webrtc: Fix potential hang and fd leak in signalling server.
- webrtc: Fix closing of WebSockets.
- webrtchttp: Allow setting `None` for audio/video caps for WHEP.

### Added
- New `awss3putobjectsink` that works similar to `awss3sink` but with a
  different upload strategy.
- New `hlscmafsink` element for writing HLS streams with CMAF/ISOBMFF
  fragments.
- New `inter` plugin with `intersink` / `intersrc` elements that allow to
  connect different pipelines in the same process.
- New `janusvrwebrtcsink` element for the Janus VideoRoom API.
- New `rtspsrc2` element.
- New `whipserversrc` element.
- gtk4: New `background-color` property for setting the color of the
  background of the frame and the borders, if any.
- gtk4: New `scale-filter` property for defining how to scale the frames.
- livesync: Add support for image formats.
- ndi: Closed Caption support in `ndisrc` / `ndisink`.
- textwrap: Add support for gaps.
- tracers: Optionally only show late buffers in `buffer-lateness` tracer.
- webrtc: Add support for custom headers.
- webrtcsink: New `payloader-setup` signal to configure payloader elements.
- webrtcsrc: Support for navigation events.

## [0.11.3] - 2023-12-18
### Fixed
- ndi: Mark a private type as such and remove a wrong `Clone` impl of internal types.
- uriplaylistbin: Fix a minor clippy warning.
- fallbacksrc: Fix error during badly timed timeout scheduling.
- webrtcsink: Fail gracefully if webrtcbin pads can't be requested instead of
  panicking.
- threadshare: Fix deadlock in `ts-udpsrc` `notify::used-socket` signal
  emission.

### Changed
- Update to AWS SDK 1.0.
- Update to windows-sys 0.52.
- Update to async-tungstenite 0.24.
- Update to bitstream-io 2.0.
- tttocea608: De-duplicate some functions.
- gtk4: Use async-channel instead of deprecated GLib main context channel.

## [0.11.2] - 2023-11-11
### Fixed
- filesink / s3sink: Set `sync=false` to allow processing faster than
  real-time.
- hlssink3: Various minor bugfixes and cleanups.
- livesync: Various minor bugfixes and cleanups that should make the element
  work more reliable.
- s3sink: Fix handling of non-ASCII characters in URIs and keys.
- sccparse: Parse SCC files that are incorrectly created by CCExtractor.
- ndisrc: Assume > 8 channels are unpositioned.
- rtpav1depay: Skip unexpected leading fragments instead of repeatedly warning
  about the stream possibly being corrupted.
- rtpav1depay: Don't push stale temporal delimiters downstream but wait until
  a complete OBU is collected.
- whipwebrtcsink: Use correct URL during redirects.
- webrtcsink: Make sure to not miss any ICE candidates.
- webrtcsink: Fix deadlock when calling `set-local-description`.
- webrtcsrc: Fix reference cycles that prevented the element from being freed.
- webrtcsrc: Define signaller property as `CONSTRUCT_ONLY` to make it actually
  possible to set different signallers.
- webrtc: Update livekit signaller to livekit 0.2.
- meson: Various fixes to the meson-based build system.

### Added
- audiornnoise: Attach audio level meta to output buffers.
- hlssink3: Allow adding `EXT-X-PROGRAM-DATE-TIME` tag to the manifest.
- webrtcsrc: Add `turn-servers` property.

### Changed
- aws/webrtc: Update to AWS SDK 0.57/0.35.

## [0.11.1] - 2023-10-04
### Fixed
- fallbackswitch: Fix various deadlocks.
- webrtcsink: Gracefully fail if adding the TWCC RTP header extension fails.
- webrtcsink: Fix codec selection discovery.
- webrtcsink: Add support for D3D11 memory and qsvh264enc.
- onvifmetadataparse: Skip metadata frames with unrepresentable UTC times.
- gtk4paintablesink: Pre-multiply alpha when creating GL textures with alpha.
- gtk4paintablesink: Only support RGBA/RGB in the GL code path.
- webrtchttp: Respect HTTP redirects.
- fmp4mux: Specify unit of fragment-duration property.

### Changed
- threadshare: Port to polling 3.1.

## [0.11.0] - 2023-08-10
### Changed
- Updated MSRV to 1.70.
- Compatible with gtk-rs 0.18 and gstreamer-rs 0.21.
- awstranscriber: Move to HTTP2-based API via the aws-sdk-transcribestreaming
  crate instead of our own implementation around the WebSocket API.

### Added
- webrtcsink: Add AWS KVS signaller and corresponding aws-kvs-webrtcsink
  element.
- awstranscriber / transcriberbin: Add support for translations and outputting
  transcriptions from a single audio stream in multiple languages at once.
- gstwebrtc-api: JavaScript API for interacting with the default signalling
  protocol used by webrtcsink / webrtcsrc.
- cea608to708: New element for converting CEA608 to CEA708 closed captions.
- webrtcsink: Expose the signaller as property and allow implementing a
  custom signaller by connecting signal handlers to the default signaller.
- webrtcsink: Add support for pre-encoded streams.
- togglerecord: Add support for non-live input streams.
- webrtcsink: New whipwebrtcsink that implements WHIP around webrtcsink.
  The existing whipsink still exists but will sooner or later be deprecated.
- webrtcsink: Add LiveKit signaller and corresponding livekitwebrtcsink
  element.

## [0.10.11] - 2023-07-20
### Fixed
- fallbackswitch: Fix pad health calculation and notifies.
- fallbackswitch: Change the threshold for trailing buffers.
- webrtcsink: Fix pipeline when input caps contain a max-framerate field.
- webrtcsink: Set VP8/VP9 payloader properties based on payloader element
  factory name.
- webrtcsink: Set config-interval=-1 and aggregate-mode=zero-latency for
  H264/5 payloaders.
- webrtcsink: Translate force-keyunit events to custom force-IDR API of NVIDIA
  encoders.
- webrtcsink: Configure only 4 threads instead of 12 for x264enc for Chrome
  compatibility.
- fmp4mux: Fix draining in chunk mode if keyframes are after the desired
  fragment end.

## [0.10.10] - 2023-07-05
### Fixed
- livesync: Improve EOS handling to be in sync with `queue`'s behaviour.
- livesync: Wait for the end timestamp of the previous buffer before looking
  at queue to actually make use of the available latency.
- webrtcsink: Avoid panic on unprepare from an async tokio context.
- webrtc/signalling: Fix race condition in message ordering.
- webrtcsink: Use the correct property types when configuring `nvvideoconvert`.
- videofx: Minimize dependencies of the image crate.
- togglerecord: Fix segment clipping to actually work as intended.

### Added
- gtk4paintablesink: Support for WGL/EGL on Windows.
- gtk4paintablesink: Add Python example application to the repository.

## [0.10.9] - 2023-06-19
### Fixed
- mp4mux/fmp4mux: Fix byte order in Opus extension box.
- webrtcsrc: Add twcc extension to the codec-preferences when present.
- webrtcsink: Don't try using cudaconvert if it is not present.
- mccparse: Don't offset the first timecode to a zero PTS.
- Correctly use MPL as license specifier instead of MPL-2 for plugins that
  compile with GStreamer < 1.20.

### Added
- fallbackswitch: Add `stop-on-eos` property.

## [0.10.8] - 2023-06-07
### Fixed
- fmp4mux: Use updated start PTS when checking if a stream is filled instead
  of a stale one.
- fmp4mux: Fix various issues with stream gaps, especially in the beginning.
- fmp4mux: Fix waiting in live pipelines.
- uriplaylistbin: Prevent deadlocks during property notifications.
- webrtcsink: Fix panics during `twcc-stats` callback and related issues.
- awstranscriber: Handle stream disconts correctly.
- roundedcorners: Fix caps negotiation to not use I420 if a border radius is
  configured.
- whipsink: Use the correct pad template to request pads from the internal
  webrtcbin.
- fallbacksrc: Don't apply fallback audio caps to the main stream.
- webrtcsrc: Fix caps handling during transceiver creation.

### Changed
- rtpgccbwe: Improve packet handling.

## [0.10.7] - 2023-05-09
### Fixed
- ffv1dec: Drop rank until the implementation is feature-complete.
- spotifyaudiosrc: Check cached credentials before use and fix usage of
  credentials cache.
- tttocea608: Specify raw CEA608 field.
- gtk4paintablesink: Fix compilation on non-Linux UNIX systems.
- webrtcsrc: Don't set stun-server to the empty string if none was set.
- webrtcsink: Abort statistics collection before stopping the signaller.
- rtpgccbwe: Don't process empty lists.

### Changed
- ndi: Update to libloading 0.8.
- aws: Update to AWS SDK 0.55/0.27.
- webrtcsink: Order pads by serial number.
- Update to async-tungstenite 0.22.

### Added
- webrtcsink/webrtcsrc: Add `request-encoded-filter` signal to add support for
  inserting custom filters between encoder/payloader or depayloader/decoder.
  This allows interacting with the "insertable streams" API from Chrome.

## [0.10.6] - 2023-04-06
### Fixed
- webrtcsink: Fix max/min-bitrate property blurb/nick.
- uriplaylistbin: Add missing queues to example.
- tttocea608: Fix pushing of caps events that sometimes contained unfixed caps.
- tttocea608: Fix disappearing text after special character in non-popon mode.
- transcriberbin: Fix deadlock on construction.
- transcriberbin: Fix initial bin setup.
- fallbacksrc: Handle incompatible downstream caps without panicking.
- ndisrc: Fix copying of raw video frames with different NDI/GStreamer strides.
- livesync: Correctly assume zero upstream latency if latency query fails.

### Added
- webrtcsink: Add `ice-transport-policy` property that proxies the same
  `webrtcbin` property.

## [0.10.5] - 2023-03-19
### Fixed
- gtk4: Fix build with OpenGL support on macOS.
- threadshare: Fix symbol conflicts when statically linking the plugin.

## [0.10.4] - 2023-03-14
### Fixed
- fmp4mux: Return a running time from `AggregatorImpl::next_time()` to fix
  waiting in live pipelines.
- fmp4mux: Fix `hls_live` example to set properties on the right element.
- uriplaylistbin: Reset element when switching back to `NULL` state.
- livesync: Handle variable framerates correctly in fallback buffer duration
  calculation.
- meson: Fix GStreamer version feature detection.

### Added
- webrtc: New `webrtc` element.

## [0.10.3] - 2023-03-02
### Added
- tracers: `queue_levels` tracer now also supports printing the `appsrc` levels.
- webrtc: `webrtcsink` can use `nvvidconv` if `nvvideoconvert` does not exist
  on an NVIDIA platform.

### Fixed
- gtk4: Set the sync point on the video frame after mapping it as otherwise
  the frame might not be ready yet for further usage.
- livesync: Correctly calculate the fallback buffer duration from the video
  framerate.
- ndi: Handle caps changes correctly in `ndisinkcombiner`.

### Changed
- webrtc: Minor cleanup.

## [0.10.2] - 2023-02-23
### Fixed
- hlssink3: Allow signal handlers to return `None`
- gtk4: Make GL context sharing more reliable in pipelines with multiple
  `gtk4paintablesink`s
- gtk4: Attach channel receiver to the main context from the correct thread to
  make it possible to start the sink from a different thread than the main
  thread without having retrieved the paintable from the main thread before.
- fmp4mux/mp4mux: Ignore caps changes if only the framerate changes.

### Changed
- gtk4: Simplify and refactor GL context sharing. Apart from being more
  reliable this reduces GL resource usage.

## [0.10.1] - 2023-02-13
### Fixed
- rtpav1pay: Fix calculation of Leb128 size size to work correctly with
  streams from certain encoders.

## [0.10.0] - 2023-02-10
### Fixed
- audiornnoise: Use correct value range for the samples
- awss3sink: Treat stopping without EOS as an error for multipart upload
- awss3hlssink: Fix the name of the hlssink child element
- awss3hlssink: Fix deadlock on EOS
- dav1d: Various fixes to improve performance, to handle decoding errors more
  gracefully and to make sure all frames are output in the end
- fmp4mux: Various fixes to fragment splitting behaviour, output formatting
  and header generation
- gtk4: Various stability and rendering fixes
- meson: Various fixes and improvements to the meson-based build system
- ndi: provide non-Linux/macOS UNIX fallback for the soname
- ndisrc: Use default channel mask for audio output to allow >2 channels to
  work better
- rav1e: Correctly enable threading support
- rtpav1: Various fixes to the payloader and depayloader to handle streams
  more correctly and to handle errors more cleanly
- rtpav1depay: Set caps on the source pad
- spotify: fix "start a runtime from within a runtime" with static link
- textahead: fix previous buffers
- textwrap: Don't panic on empty buffers
- tttocea608: Don't fail if a GAP event contains no duration
- webrtchttp: whipsink: construct TURN URL correctly
- webrtcsink: fix panic on pre-bwe request error
- whipsink: Send ICE candidates together with the offer
- whipsink: Various cleanups and minor fixes

### Added
- audiornnoise: Add voice detection threshold property
- awss3hlssink: Add `stats` property
- awss3sink: Add properties to set Content-Type and Content-Disposition
- fmp4mux: add 'offset-to-zero' property
- fmp4mux/mp4mux: add support for muxing Opus, VP8, VP9 and AV1 streams
- fmp4mux/mp4mux: Make media/track timescales configurable
- fmp4mux: Add support for CMAF-style chunking, e.g. low-latency / LL HLS and DASH
- gtk4: Support for rendering GL textures on X11/EGL, X11/GLX, Wayland and macOS
- hlssink3: Allow generating i-frame-only playlist
- livesync: New element that allows maintaining a contiguous live stream
  without gaps from a potentially unstable source.
- mp4mux: New non-fragmented MP4 muxer element
- spotifyaudiosrc: Support configurable bitrate
- textahead: add settings to display previous buffers
- threadshare: Introduce new ts-audiotestsrc
- webrtcsink: Support nvv4l2vp9enc
- whepsource: Add a WebRTC WHEP source element

### Changed
- audiofx: Derive from AudioFilter where possible
- dav1ddec: Lower rank to primary to allow usage of hardware decoders with
  higher ranks
- fmp4mux: Only push `fragment_offset` if `write-mfra` is true to reduce memory usage
- webrtcsink: Make the `turn-server` property a `turn-servers` list
- webrtcsink: Move from async-std to tokio

[Unreleased]: https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/compare/0.15.3...HEAD
[0.15.3]: https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/compare/0.15.2...0.15.3
[0.15.2]: https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/compare/0.15.1...0.15.2
[0.15.1]: https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/compare/0.15.0...0.15.1
[0.15.0]: https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/compare/0.14.4...0.15.0
[0.14.5]: https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/compare/0.14.4...0.14.5
[0.14.4]: https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/compare/0.14.3...0.14.4
[0.14.3]: https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/compare/0.14.2...0.14.3
[0.14.2]: https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/compare/0.14.1...0.14.2
[0.14.1]: https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/compare/0.14.0...0.14.1
[0.14.0]: https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/compare/0.13.7...0.14.0
[0.13.7]: https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/compare/0.13.6...0.13.7
[0.13.6]: https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/compare/0.13.5...0.13.6
[0.13.5]: https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/compare/0.13.4...0.13.5
[0.13.4]: https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/compare/0.13.3...0.13.4
[0.13.3]: https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/compare/0.13.2...0.13.3
[0.13.2]: https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/compare/0.13.1...0.13.2
[0.13.1]: https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/compare/0.13.0...0.13.1
[0.13.0]: https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/compare/0.12.7...0.13.0
[0.12.7]: https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/compare/0.12.6...0.12.7
[0.12.6]: https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/compare/0.12.5...0.12.6
[0.12.5]: https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/compare/0.12.4...0.12.5
[0.12.4]: https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/compare/0.12.3...0.12.4
[0.12.3]: https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/compare/0.12.2...0.12.3
[0.12.2]: https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/compare/0.12.1...0.12.2
[0.12.1]: https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/compare/0.12.0...0.12.1
[0.12.0]: https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/compare/0.11.3...0.12.0
[0.11.3]: https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/compare/0.11.2...0.11.3
[0.11.2]: https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/compare/0.11.1...0.11.2
[0.11.1]: https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/compare/0.11.0...0.11.1
[0.11.0]: https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/compare/0.10.11...0.11.0
[0.10.11]: https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/compare/0.10.10...0.10.11
[0.10.10]: https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/compare/0.10.9...0.10.10
[0.10.9]: https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/compare/0.10.8...0.10.9
[0.10.8]: https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/compare/0.10.7...0.10.8
[0.10.7]: https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/compare/0.10.6...0.10.7
[0.10.6]: https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/compare/0.10.5...0.10.6
[0.10.5]: https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/compare/0.10.4...0.10.5
[0.10.4]: https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/compare/0.10.3...0.10.4
[0.10.3]: https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/compare/0.10.2...0.10.3
[0.10.2]: https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/compare/0.10.1...0.10.2
[0.10.1]: https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/compare/0.10.0...0.10.1
[0.10.0]: https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/compare/0.9.0...0.10.0
