-This is GStreamer 1.2.0
-
-Changes since 1.0:
-
-New API:
- • GstContext negotiation / sharing / announcing for sharing a
- generic context between elements, e.g. a display handle
- • GL texture upload conversion meta for allowing different
- buffer types to be converted to an OpenGL texture
- • GstCapsFeatures as extension to GstCaps for allowing the
- negotiation of specific memory or meta requirements between
- elements
- • GstMemory flags for contiguous and non-mappable memory
- • The stream-start event has optional flags now, e.g. for signalling
- sparse streams
- • The stream-start even has an optional group-id field now to signal
- all streams that should be played together
- • Allocators library in gst-plugins-base, currently only with generic
- dmabuf memory support
- • insertbin library for easier handling of dynamically linked
- pipelines (in -bad for now)
- • EGL helper library (in -bad for now)
- • MPEG-TS data structure library (in -bad for now)
- • New GstVideoRegionOfInterestMeta to describe a region of interest on
- video frames.
- • GstVideoDecoder/Encoder has new ::flush() vfunc to replace the
- ill-defined ::reset() vfunc.
- • The URI query allows to query the redirected URI now.
-
-Major changes:
- • New tool: gst-play-1.0 in gst-plugins-base for basic playback
- testing on the command line.
- • New plugins:
- ∘ mssdemux for Microsoft Smooth Streaming
- ∘ dashdemux for DASH adaptive streaming protocol
- ∘ bluez for interaction with Bluetooth devices
- ∘ openjpeg for JPEG2000 decoding and encoding
- ∘ daala for experimental Daala decoding and encoding
- ∘ vpx plugin has experimental VP9 decoding and encoding support
- ∘ webp plugin for WebP decoding (encoding to be added later)
- ∘ Various others: yadif, srtp, sbc, fluidsynth, midiparse,
- mfc, ivtv, accuraterip and audiofxbad
-
- • Moved plugins:
- ∘ dtmf, vp8rtp, scaletempo and rtpmux plugins are in
- gst-plugins-good now
-
- • Video:
- ∘ Fix handling of interlaced video in converters such as videoscale
- and videoconvert (e.g. scale both fields independently)
- ∘ videoconvert will try harder to minimise quality losses when
- conversion is necessary
- ∘ The experimental GstSurfaceConverter, GstSurfaceMeta and
- GstVideoContext APIs from the (confusingly-named)
- libgstbasevideo-1.0 library in gst-plugins-bad have now been
- removed and been replaced by new APIs in GStreamer Core and
- gst-plugins-base (see above). Since that was all that was left in
- this library, the entire experimental libgstbasevideo-1.0 library
- has been removed from gst-plugins-bad
- ∘ Chroma subsampling and chroma siting conversion is better handled
- in videoconvert and the support for interlaced video was improved.
- ∘ New pinwheel and spoke patterns in videotestsrc
- ∘ videomixer can now accept different video formats on its sinkpads
- and converts to a common format during mixing
-
- • Audio:
- ∘ audioconvert will try harder to minimise quality losses when
- conversion is necessary
- ∘ adder now allows muting/unmuting of its input streams, and also
- per-input stream volume
- ∘ pulseaudio elements can switch between devices during playback now
- ∘ aacparse can convert between ADTS←→RAW
-
- • Platform specific changes:
- ∘ Caps, events, etc. are now printed in the GStreamer debug logs
- with their content instead of just the pointer address even on
- non-glibc platforms (e.g. Windows, OSX, Android).
- ∘ Network elements (UDP/TCP) now work better with platforms,
- where IPv6 sockets can't handle IPv4 (e.g. Windows)
- ∘ Linux/BSD: v4l2 had many improvements and cleanups
-
- • Other changes:
- ∘ gst-libav now uses libav 9
- ∘ Static linking of plugins is supported now (also in 1.0.7)
- ∘ rtspsrc: add support for NetClientClock: when the server suggests a
- GstNetTimeProvider in the SDP, set up a GstNetClientClock that
- slaves to the remote clock and suggest this clock in provide_clock.
- Simplifies synchronized playback of a resource from an RTSP server.
- gst-rtsp-server now supports adding this to the SDP and can provide
- a network clock
- ∘ RTP retransmission / NACK support and big RTP jitterbuffer improvements
- ∘ SRTP and DTLS support
- ∘ Changes to many elements and core to use the correct sticky event
- order and also not lose any important sticky events during flushing
- ∘ >1000 fixed bug reports, and many other bug fixes and other
- improvements everywhere that had no bug report
-
-Things to look out for:
- • Single header includes for all libraries, e.g. #include
- <gst/video/video.h> - this was needed for some bindings.
- • Stricter (correct) caps subset checking in some cases where this was
- not correct before. Caps will now always fail to be a compatible
- subset of another set of caps if the subset caps are missing some
- fields that the superset caps have. This might lead to not-negotiated
- errors if caps are incomplete now. However, it also prevents possible
- data corruption caused by piping data formatted in an
- incompatible/unexpected way into some elements. Check your h264 caps
- for stream-format and alignment fields and AAC caps for the
- stream-format field. This change will also be included in the next
- stable 1.0.8 release.
- • Stricter checking for missing events and correct sticky event order
- (stream-start, caps, segment) in some places; this is not enabled in
- stable releases by default, but you may get warnings when using git
- builds, development releases or when compiling with
- -UG_DISABLE_ASSERT in CFLAGS
- • x264enc now outputs data in byte-stream by default if downstream has
- ANY caps (e.g. appsink without caps set, filesink, udpsink,
- tcpserversink etc.)
- • The MPEG TS demuxer posts messages contain the PMT, PAT, etc. in a
- different format now. This new format uses the data structures from
- the new MPEGTS library
- • The GstContext API has changed between 1.1.4 and 1.1.90
-
+This is GStreamer 1.11.90.
projects / platform / upstream / gstreamer.git / blobdiff