- Update to version 0.13.0:
* 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.
- Update to version 0.12.7:
* 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
- Update to version 0.12.6:
* 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.
- Update to version 0.12.5:
* 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.
- Update to version 0.12.4:
* 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.
- Update to version 0.12.3:
* 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.