+This is GStreamer Base Plugins 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+ removed and been replaced by new APIs in GStreamer Core and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
projects / platform / upstream / gst-plugins-base.git / commitdiff