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24 KiB
24 KiB
Changelog
All notable changes to this project will be documented in this file.
The format is based on Keep a Changelog and this project adheres to Semantic Versioning, specifically the variant used by Rust.
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
orrav1enc
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 inrequest-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 otherwisertpsession
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 ifGST_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 towhipclientsink
and deprecate oldwhipsink
.
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 bettergtk_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 toawss3sink
but with a different upload strategy. - New
hlscmafsink
element for writing HLS streams with CMAF/ISOBMFF fragments. - New
inter
plugin withintersink
/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 samewebrtcbin
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 theappsrc
levels. - webrtc:
webrtcsink
can usenvvidconv
ifnvvideoconvert
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
gtk4paintablesinks
- 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
ifwrite-mfra
is true to reduce memory usage - webrtcsink: Make the
turn-server
property aturn-servers
list - webrtcsink: Move from async-std to tokio