X-Git-Url: http://review.tizen.org/git/?a=blobdiff_plain;f=NEWS;h=e6f8c3dbe78494f900e2431b6f08520931987c1f;hb=5470f6df00595f4ab44871e0e633bf15006abc5c;hp=74fb1eaeb8e6e2bbfa050e7aac1495f4954e77f4;hpb=7854a6597868eeb91bca7518ad73adcb3107f56d;p=platform%2Fupstream%2Fgstreamer.git diff --git a/NEWS b/NEWS index 74fb1ea..e6f8c3d 100644 --- a/NEWS +++ b/NEWS @@ -1,734 +1,1274 @@ -# GStreamer 1.12 Release Notes -GStreamer 1.12.0 was originally released on 4th May 2017. -The GStreamer team is proud to announce a new major feature release in the -stable 1.x API series of your favourite cross-platform multimedia framework! +GSTREAMER 1.16 RELEASE NOTES -As always, this release is again packed with new features, bug fixes and other -improvements. -See [https://gstreamer.freedesktop.org/releases/1.12/][latest] for the latest +GStreamer 1.16 has not been released yet. It is scheduled for release in +March 2019. + +1.15.x is the unstable development version that is being developed in +the git master branch and which will eventually result in 1.16. + +1.16 will be backwards-compatible to the stable 1.14, 1.12, 1.10, 1.8, +1.6, 1.4, 1.2 and 1.0 release series. + +See https://gstreamer.freedesktop.org/releases/1.16/ for the latest version of this document. -*Last updated: Thursday 4 May 2017, 11:00 UTC [(log)][gitlog]* +_Last updated: Wednesday 27 January 2019, 00:30 UTC (log)_ + + +Introduction + +The GStreamer team is proud to announce a new major feature release in +the stable 1.x API series of your favourite cross-platform multimedia +framework! + +As always, this release is again packed with many new features, bug +fixes and other improvements. + + +Highlights + +- GStreamer WebRTC stack gained support for data channels for + peer-to-peer communication based on SCTP, BUNDLE support, as well as + support for multiple TURN servers. + +- AV1 video codec support for Matroska and QuickTime/MP4 containers + and more configuration options and supported input formats for the + AOMedia AV1 encoder + +- Support for Closed Captions and other Ancillary Data in video + +- Support for planar (non-interleaved) raw audio + +- GstVideoAggregator, compositor and OpenGL mixer elements are now in + -base -[latest]: https://gstreamer.freedesktop.org/releases/1.12/ -[gitlog]: https://cgit.freedesktop.org/gstreamer/www/log/src/htdocs/releases/1.12/release-notes-1.12.md +- New alternate fields interlace mode where each buffer carries a + single field -## Introduction +- WebM and Matroska ContentEncryption support in the Matroska demuxer -The GStreamer team is proud to announce a new major feature release in the -stable 1.x API series of your favourite cross-platform multimedia framework! +- new WebKit WPE-based web browser source element -As always, this release is again packed with new features, bug fixes and other -improvements. +- Video4Linux: HEVC encoding and decoding, JPEG encoding, and improved + dmabuf import/export -## Highlights +- Hardware-accelerated Nvidia video decoder gained support for VP8/VP9 + decoding, whilst the encoder gained support for H.265/HEVC encoding. -- new `msdk` plugin for Intel's Media SDK for hardware-accelerated video - encoding and decoding on Intel graphics hardware on Windows or Linux. +- Many improvements to the Intel Media SDK based hardware-accelerated + video decoder and encoder plugin (msdk): dmabuf import/export for + zero-copy integration with other components; VP9 decoding; 10-bit + HEVC encoding; video post-processing (vpp) support including + deinterlacing; and the video decoder now handles dynamic resolution + changes. -- `x264enc` can now use multiple x264 library versions compiled for different - bit depths at runtime, to transparently provide support for multiple bit - depths. +- The ASS/SSA subtitle overlay renderer can now handle multiple + subtitles that overlap in time and will show them on screen + simultaneously -- `videoscale` and `videoconvert` now support multi-threaded scaling and - conversion, which is particularly useful with higher resolution video. +- The Meson build is now feature-complete (*) and it is now the + recommended build system on all platforms. The Autotools build is + scheduled to be removed in the next cycle. -- `h264parse` will now automatically insert AU delimiters if needed when - outputting byte-stream format, which improves standard compliance and - is needed in particular for HLS playback on iOS/macOS. +- The GStreamer Rust bindings and Rust plugins module are now + officially part of upstream GStreamer. -- `rtpbin` has acquired bundle support for incoming streams +- Many performance improvements -## Major new features and changes -### Noteworthy new API +Major new features and changes -- The video library gained support for a number of new video formats: +Noteworthy new API - - `GBR_12LE`, `GBR_12BE`, `GBRA_12LE`, `GBRA_12BE` (planar 4:4:4 RGB/RGBA, 12 bits per channel) - - `GBRA_10LE`, `GBRA_10BE` (planar 4:4:4:4 RGBA, 10 bits per channel) - - `GBRA` (planar 4:4:4:4 ARGB, 8 bits per channel) - - `I420_12BE`, `I420_12LE` (planar 4:2:0 YUV, 12 bits per channel) - - `I422_12BE`,`I422_12LE` (planar 4:2:2 YUV, 12 bits per channel) - - `Y444_12BE`, `Y444_12LE` (planar 4:4:4 YUV, 12 bits per channel) - - `VYUY` (another packed 4:2:2 YUV format) +- GstAggregator has a new "min-upstream-latency" property that forces + a minimum aggregate latency for the input branches of an aggregator. + This is useful for dynamic pipelines where branches with a higher + latency might be added later after the pipeline is already up and + running and where a change in the latency would be disruptive. This + only applies to the case where at least one of the input branches is + live though, it won’t force the aggregator into live mode in the + absence of any live inputs. -- The high-level `GstPlayer` API was extended with functions for taking video - snapshots and enabling accurate seeking. It can optionally also use the - still-experimental `playbin3` element now. +- GstBaseSink gained a "processing-deadline" property and + setter/getter API to configure a processing deadline for live + pipelines. The processing deadline is the acceptable amount of time + to process the media in a live pipeline before it reaches the sink. + This is on top of the systemic latency that is normally reported by + the latency query. This defaults to 20ms and should make pipelines + such as v4l2src ! xvimagesink not claim that all frames are late in + the QoS events. Ideally, this should replace the "max-lateness" + property for most applications. -### New Elements +- RTCP Extended Reports (XR) parsing according to RFC 3611: + Loss/Duplicate RLE, Packet Receipt Times, Receiver Reference Time, + Delay since the last Receiver (DLRR), Statistics Summary, and VoIP + Metrics reports. This only provides the ability to parse such + packets, generation of XR packets is not supported yet and XR + packets are not automatically parsed by rtpbin / rtpsession but must + be actively handled by the application. -- msdk: new plugin for Intel's Media SDK for hardware-accelerated video encoding - and decoding on Intel graphics hardware on Windows or Linux. This includes - an H.264 encoder/decoder (`msdkh264dec`, `msdkh264enc`), - an H.265 encoder/decoder (`msdkh265dec`, `msdkh265enc`), - an MJPEG encoder/encoder (`msdkmjpegdec`, `msdkmjpegenc`), - an MPEG-2 video encoder (`msdkmpeg2enc`) and a VP8 encoder (`msdkvp8enc`). +- a new mode for interlaced video was added where each buffer carries + a single field of interlaced video, with buffer flags indicating + whether the field is the top field or bottom field. Top and bottom + fields are expected to alternate in this mode. Caps for this + interlace mode must also carry a format:Interlaced caps feature to + ensure backwards compatibility. -- `iqa` is a new Image Quality Assessment plugin based on [DSSIM][dssim], - similar to the old (unported) videomeasure element. +- The video library has gained support for three new raw pixel + formats: -- The `faceoverlay` element, which allows you to overlay SVG graphics over - a detected face in a video stream, has been ported from 0.10. + - Y410: packed 4:4:4 YUV, 10 bits per channel + - Y210: packed 4:2:2 YUV, 10 bits per channel + - NV12_10LE40: fully-packed 10-bit variant of NV12_10LE32, + i.e. without the padding bits -- our `ffmpeg` wrapper plugin now exposes/maps the ffmpeg Opus audio decoder - (`avdec_opus`) as well as the GoPro CineForm HD / CFHD decoder (`avdec_cfhd`), - and also a parser/writer for the IVF format (`avdemux_ivf` and `avmux_ivf`). +- GstRTPSourceMeta is a new meta that can be used to transport + information about the origin of depayloaded or decoded RTP buffers, + e.g. when mixing audio from multiple sources into a single stream. A + new "source-info" property on the RTP depayloader base class + determines whether depayloaders should put this meta on outgoing + buffers. Similarly, the same property on RTP payloaders determines + whether they should use the information from this meta to construct + the CSRCs list on outgoing RTP buffers. -- `audiobuffersplit` is a new element that splits raw audio buffers into - equal-sized buffers +- gst_sdp_message_from_text() is a convenience constructor to parse + SDPs from a string which is particularly useful for language + bindings. -- `audiomixmatrix` is a new element that mixes N:M audio channels according to - a configured mix matrix. +Support for Planar (Non-Interleaved) Raw Audio + +Raw audio samples are usually passed around in interleaved form in +GStreamer, which means that if there are multiple audio channels the +samples for each channel are interleaved in memory, e.g. +|LEFT|RIGHT|LEFT|RIGHT|LEFT|RIGHT| for stereo audio. A non-interleaved +or planar arrangement in memory would look like +|LEFT|LEFT|LEFT|RIGHT|RIGHT|RIGHT| instead, possibly with +|LEFT|LEFT|LEFT| and |RIGHT|RIGHT|RIGHT| residing in separate memory +chunks or separated by some padding. + +GStreamer has always had signalling for non-interleaved audio since +version 1.0, but it was never actually properly implemented in any +elements. audioconvert would advertise support for it, but wasn’t +actually able to handle it correctly. + +With this release we now have full support for non-interleaved audio as +well, which means more efficient integration with external APIs that +handle audio this way, but also more efficient processing of certain +operations like interleaving multiple 1-channel streams into a +multi-channel stream which can be done without memory copies now. + +New API to support this has been added to the GStreamer Audio support +library: There is now a new GstAudioMeta which describes how data is +laid out inside the buffer, and buffers with non-interleaved audio must +always carry this meta. To access the non-interleaved audio samples you +must map such buffers with gst_audio_buffer_map() which works much like +gst_buffer_map() or gst_video_frame_map() in that it will populate a +little GstAudioBuffer helper structure passed to it with the number of +samples, the number of planes and pointers to the start of each plane in +memory. This function can also be used to map interleaved audio buffers +in which case there will be only one plane of interleaved samples. + +Of course support for this has also been implemented in the various +audio helper and conversion APIs, base classes, and in elements such as +audioconvert, audioresample, audiotestsrc, audiorate. + +Support for Closed Captions and Other Ancillary Data in Video + +The video support library has gained support for detecting and +extracting Ancillary Data from videos as per the SMPTE S291M +specification, including: + +- a VBI (Vertical Blanking Interval) parser that can detect and + extract Ancillary Data from Vertical Blanking Interval lines of + component signals. This is currently supported for videos in v210 + and UYVY format. + +- a new GstMeta for closed captions: GstVideoCaptionMeta. This + supports the two types of closed captions, CEA-608 and CEA-708, + along with the four different ways they can be transported (other + systems are a superset of those). + +- a VBI (Vertical Blanking Interval) encoder for writing ancillary + data to the Vertical Blanking Interval lines of component signals. + +The new closedcaption plugin in gst-plugins-bad then makes use of all +this new infrastructure and provides the following elements: + +- cccombiner: a closed caption combiner that takes a closed captions + stream and another stream and adds the closed captions as + GstVideoCaptionMeta to the buffers of the other stream. + +- ccextractor: a closed caption extractor which will take + GstVideoCaptionMeta from input buffers and output them as a separate + closed captions stream. + +- ccconverter: a closed caption converter that can convert between + different formats + +- line21decoder: extract line21 closed captions from SD video streams + +- cc708overlay: decodes CEA 608/708 captions and overlays them on + video + +Additionally, the following elements have also gained Closed Caption +support: + +- qtdemux and qtmux support CEA 608/708 Closed Caption tracks + +- mpegvideoparse extracts Closed Captions from MPEG-2 video streams + +- decklinkvideosink can output closed captions and decklinkvideosrc + can extract closed captions + +- playbin and playbin3 learned how to autoplug CEA 608/708 CC overlay + elements + +- the externally maintained ajavideosrc element for AJA capture cards + has support for extracting closed captions + +The rsclosedcaption plugin in the Rust plugins collection includes a +MacCaption (MCC) file parser and encoder. + +New Elements + +- overlaycomposition: New element that allows applications to draw + GstVideoOverlayCompositions on a stream. The element will emit the + "draw" signal for each video buffer, and the application then + generates an overlay for that frame (or not). This is much more + performant than e.g. cairooverlay for many use cases, e.g. because + pixel format conversions can be avoided or the blitting of the + overlay can be delegated to downstream elements (such as + gloverlaycompositor). It’s particularly useful for cases where only + a small section of the video frame should be drawn on. + +- gloverlaycompositor: New OpenGL-based compositor element that + flattens any overlays from GstVideoOverlayCompositionMetas into the + video stream. This element is also always part of glimagesink. + +- glalpha: New element that adds an alpha channel to a video stream. + The values of the alpha channel can either be set to a constant or + can be dynamically calculated via chroma keying. It is similar to + the existing alpha element but based on OpenGL. Calculations are + done in floating point so results may not be identical to the output + of the existing alpha element. + +- rtpfunnel funnels together RTP streams into a single session. Use + cases include multiplexing and bundle. webrtcbin uses it to + implement BUNDLE support. + +- testsrcbin is a source element that provides an audio and/or video + stream and also announces them using the recently-introduced + GstStream API. This is useful for testing elements such as playbin3 + or uridecodebin3 etc. + +- New closed caption elements: cccombiner, ccextractor, ccconverter, + line21decoder and cc708overlay (see above) + +- wpesrc: new source element acting as a Web Browser based on WebKit + WPE + +- Two new OpenCV-based elements: cameracalibrate and cameraundistort + that can communicate to figure out distortion correction parameters + for a camera and correct for the distortion. + +- New sctp plugin based on usrsctp with sctpenc and sctpdec elements. + These elements are used inside webrtcbin for implementing data + channels. + +New element features and additions + +- playbin3, playbin and playsink have gained a new "text-offset" + property to adjust the positioning of the selected subtitle stream + vis-a-vis the audio and video streams. This uses subtitleoverlay’s + new "subtitle-ts-offset" property. GstPlayer has gained matching API + for this, namely gst_player_get_text_video_offset(). + +- playbin3 buffering improvements: in network playback scenarios there + may be multiple inputs to decodebin3, and buffering will be done + before decodebin3 using queue2 or downloadbuffer elements inside + urisourcebin. Since this is before any parsers or demuxers there may + not be any bitrate information available for the various streams, so + it was difficult to configure the buffering there smartly within + global constraints. This was improved now: The queue2 elements + inside urisourcebin will now use the new bitrate query to figure out + a bitrate estimate for the stream if no bitrate was provided by + upstream, and urisourcebin will use the bitrates of the individual + queues to distribute the globally-set "buffer-size" budget in bytes + to the various queues. urisourcebin also gained "low-watermark" and + "high-watermark" properties which will be proxied to the internal + queues, as well as a read-only "statistics" property which allows + querying of the minimum/maximum/average byte and time levels of the + queues inside the urisourcebin in question. + +- splitmuxsink has gained a couple of new features: + + - new "async-finalize" mode: This mode is useful for muxers or + outputs that can take a long time to finalize a file. Instead of + blocking the whole upstream pipeline while the muxer is doing + its stuff, we can unlink it and spawn a new muxer + sink + combination to continue running normally. This requires us to + receive the muxer and sink (if needed) as factories via the new + "muxer-factory" and "sink-factory" properties, optionally + accompanied by their respective properties structures (set via + the new "muxer-properties" and "sink-properties" properties). + There are also new "muxer-added" and "sink-added" signals in + case custom code has to be called for them to configure them. + + - "split-at-running-time" action signal: When called by the user, + this action signal ends the current file (and starts a new one) + as soon as the given running time is reached. If called multiple + times, running times are queued up and processed in the order + they were given. + + - "split-after" action signal to finish outputting the current GOP + to the current file and then start a new file as soon as the GOP + is finished and a new GOP is opened (unlike the existing + "split-now" which immediately finishes the current file and + writes the current GOP into the next newly-started file). + + - "reset-muxer" property: when unset, the muxer is reset using + flush events instead of setting its state to NULL and back. This + means the muxer can keep state across resets, e.g. mpegtsmux + will keep the continuity counter continuous across segments as + required by hlssink2. + +- qtdemux gained PIFF track encryption box support in addition to the + already-existing PIFF sample encryption support, and also allows + applications to select which encryption system to use via a + "drm-preferred-decryption-system-id" context in case there are + multiple options. + +- qtmux: the "start-gap-threshold" property determines now whether an + edit list will be created to account for small gaps or offsets at + the beginning of a stream in case the start timestamps of tracks + don’t line up perfectly. Previously the threshold was hard-coded to + 1% of the (video) frame duration, now it is 0 by default (so edit + list will be created even for small differences), but fully + configurable. + +- rtpjitterbuffer has improved end-of-stream handling + +- rtpmp4vpay will be prefered over rtpmp4gpay for MPEG-4 video in + autoplugging scenarios now + +- rtspsrc now allows applications to send RTSP SET_PARAMETER and + GET_PARAMETER requests using action signals. + +- rtspsrc has a small (100ms) configurable teardown delay by default + to try and make sure an RTSP TEARDOWN request gets sent out when the + source element shuts down. This will block the downward PAUSED to + READY state change for a short time, but can be disabled where it’s + a problem. Some servers only allow a limited number of concurrent + clients, so if no proper TEARDOWN is sent new clients may have + problems connecting to the server for a while. + +- souphttpsrc behaves better with low bitrate streams now. Before it + would increase the read block size too quickly which could lead to + it not reading any data from the socket for a very long time with + low bitrate streams that are output live downstream. This could lead + to servers kicking off the client. + +- filesink: do internal buffering to avoid performance regression with + small writes since we bypass libc buffering by using writev() + instead of fwrite() + +- identity: add "eos-after" property and fix "error-after" property + when the element is reused + +- input-selector: lets context queries pass through, so that + e.g. upstream OpenGL elements can use contexts and displays + advertised by downstream elements + +- queue2: avoid ping-pong between 0% and 100% buffering messages if + upstream is pushing buffers larger than one of its limits, plus + performance optimisations + +- opusdec: new "phase-inversion" property to control phase inversion. + When enabled, this will slightly increase stereo quality, but + produces a stream that when downmixed to mono will suffer audio + distortions. + +- The x265enc HEVC encoder also exposes a "key-int-max" property to + configure the maximum allowed GOP size now. + +- decklinkvideosink has seen stability improvements for long-running + pipelines (potential crash due to overflow of leaked clock refcount) + and clock-slaving improvements when performing flushing seeks + (causing stalls in the output timeline), pausing and/or buffering. + +- srtpdec, srtpenc: add support for MKIs which allow multiple keys to + be used with a single SRTP stream + +- The srt Secure Reliable Transport plugin has integrated server and + client elements srt{client,server}{src,sink} into one (srtsrc and + srtsink), since SRT connection mode can be changed by uri + parameters. + +- h264parse and h265parse will handle SEI recovery point messages and + mark recovery points as keyframes as well (in addition to IDR + frames) + +- webrtcbin: "add-turn-server" action signal to pass multiple ICE + relays (TURN servers). + +- The removesilence element has received various new features and + properties, such as a "threshold" property, detecting silence only + after minimum silence time/buffers, a "silent" property to control + bus message notifications as well as a "squash" property. + +- AOMedia AV1 decoder gained support for 10/12bit decoding whilst the + AV1 encoder supports more image formats and subsamplings now and + acquired support for rate control and profile related configuration. + +- The Fraunhofer fdkaac plugin can now be built against the 2.0.0 + version API and has improved multichannel support + +- kmssink now supports unpadded 24-bit RGB and can configure mode + setting from video info, which enables display of multi-planar + formats such as I420 or NV12 with modesetting. It has also gained a + number of new properties: The "restore-crtc" property does what it + says on the tin and is enabled by default. "plane-properties" and + "connector-properties" can be used to pass custom properties to the + DRM. + +- waylandsink has a "fullscreen" property now. + +Plugin and library moves + +- The stereo element was moved from -bad into the existing audiofx + plugin in -good. If you get duplicate type registration warnings + when upgrading, check that you don’t have a stale stereoplugin lying + about somewhere. + +GstVideoAggregator, compositor, and OpenGL mixer elements moved from -bad to -base + +GstVideoAggregator is a new base class for raw video mixers and muxers +and is based on GstAggregator. It provides defined-latency mixing of raw +video inputs and ensures that the pipeline won’t stall even if one of +the input streams stops producing data. + +As part of the move to stabilise the API there were some last-minute API +changes and clean-ups, but those should mostly affect internal elements. +Most notably, the "ignore-eos" pad property was renamed to +"repeat-after-eos" and the conversion code was moved to a +GstVideoAggregatorConvertPad subclass to avoid code duplication, make +things less awkward for subclasses like the OpenGL-based video mixer, +and make the API more consistent with the audio aggregator API. + +It is used by the compositor element, which is a replacement for +‘videomixer’ which did not handle live inputs very well. compositor +should behave much better in that respect and generally behave as one +would expected in most scenarios. + +The compositor element has gained support for per-pad blending mode +operators (SOURCE, OVER, ADD) which determines what operator to use for +blending this pad over the previous ones. This can be used to implement +crossfading and the available operators can be extended in the future as +needed. + +A number of OpenGL-based video mixer elements (glvideomixer, glmixerbin, +glvideomixerelement, glstereomix, glmosaic) which are built on top of +GstVideoAggregator have also been moved from -bad to -base now. These +elements have been merged into the existing OpenGL plugin, so if you get +duplicate type registration warnings when upgrading, check that you +don’t have a stale openglmixers plugin lying about somewhere. + +Plugin removals + +The following plugins have been removed from gst-plugins-bad: + +- The experimental daala plugin has been removed, since it’s not so + useful now that all effort is focused on AV1 instead, and it had to + be enabled explicitly with --enable-experimental anyway. + +- The spc plugin has been removed. It has been replaced by the gme + plugin. + +- The acmmp3dec and acmenc plugins for Windows have been removed. ACM + is an ancient legacy API and there was no point in keeping the + plugins around for a licensed MP3 decoder now that the MP3 patents + have expired and we have a decoder in -good. We also didn’t ship + these in our cerbero-built Windows packages, so it’s unlikely that + they’ll be missed. + + +Miscellaneous API additions + +- GstBitwriter: new generic bit writer API to complement the existing + bit reader + +- gst_buffer_new_wrapped_bytes() creates a wrap buffer from a GBytes + +- gst_caps_set_features_simple() sets a caps feature on all the + structures of a GstCaps + +- New GST_QUERY_BITRATE query: This allows determining from downstream + what the expected bitrate of a stream may be which is useful in + queue2 for setting time based limits when upstream does not provide + timing information. tsdemux, qtdemux and matroskademux have basic + support for this query on their sink pads. + +- elements: there is a new “Hardware” class specifier. Elements + interacting with hardware devices should specify this classifier in + their element factory class metadata. This is useful to advertise as + one might need to put such elements into READY state to test if the + hardware is present in the system for example. + +- protection: Add a new definition for unspecified system protection, + GST_PROTECTION_UNSPECIFIED_SYSTEM_ID + +- take functions for various mini objects that didn’t have them yet: + gst_query_take(), gst_message_take(), gst_tag_list_take(), + gst_buffer_list_take(). Unlike the various _replace() functions + _take() does not increase the reference count but takes ownership of + the mini object passed. + +- clear functions for various mini object types and GstObject which + unrefs the object or mini object (if non-NULL) and sets the variable + pointed to to NULL: gst_clear_structure(), gst_clear_tag_list(), + gst_clear_query(), gst_clear_message(), gst_clear_event(), + gst_clear_caps(), gst_clear_buffer_list(), gst_clear_buffer(), + gst_clear_mini_object(), gst_clear_object() + +- miniobject: new API gst_mini_object_add_parent() and + gst_mini_object_remove_parent() to set parent pointers on mini + objects to ensure correct writability: Every container of + miniobjects now needs to store itself as parent in the child object, + and remove itself again later. A mini object is then only writable + if there is at most one parent, that parent is writable itself, and + the reference count of the mini object is 1. GstBuffer (for + memories), GstBufferList (for buffers), GstSample (for caps, buffer, + bufferlist), and GstVideoOverlayComposition were updated + accordingly. Without this it was possible to have e.g. a buffer list + with a refcount of 2 used in two places at once that both modify the + same buffer with refcount 1 at the same time wrongly thinking it is + writable even though it’s really not. + +- poll: add API to watch for POLLPRI and stop treating POLLPRI as a + read. This is useful to wait for video4linux events which are + signalled via POLLPRI. + +- sample: new API to update the contents of a GstSample and make it + writable: gst_sample_set_buffer(), gst_sample_set_caps(), + gst_sample_set_segment(), gst_sample_set_info(), plus + gst_sample_is_writable() and gst_sample_make_writable(). This makes + it possible to reuse a sample object and avoid unnecessary memory + allocations, for example in appsink. + +- ClockIDs now keep a weak reference to underlying clock to avoid + crashes in basesink in corner cases where a clock goes away while + the ClockID is still in use, plus some new API + (gst_clock_id_get_clock(), gst_clock_id_uses_clock()) to check the + clock a ClockID is linked to. + +- The GstCheck unit test library gained a + fail_unless_equals_clocktime() convenience macro as well as some new + GstHarness API for for proposing meta APIs from the allocation + query: gst_harness_add_propose_allocation_meta(). ASSERT_CRITICAL() + checks in unit tests are now skipped if GStreamer was compiled with + GST_DISABLE_GLIB_CHECKS. + +- gst_audio_buffer_truncate() convenience function to truncate a raw + audio buffer + + +Miscellaneous performance and memory optimisations + +As always there have been many performance and memory usage improvements +across all components and modules. Some of them (such as dmabuf +import/export) have already been mentioned elsewhere so won’t be +repeated here. + +The following list is only a small snapshot of some of the more +interesting optimisations that haven’t been mentioned in other contexts +yet: + +- The GstVideoEncoder and GstVideoDecoder base classes now release the + STREAM_LOCK when pushing out buffers, which means (multi-threaded) + encoders and decoders can now receive and continue to process input + buffers whilst waiting for downstream elements in the pipeline to + process the buffer that was pushed out. This increases throughput + and reduces processing latency, also and especially for + hardware-accelerated encoder/decoder elements. + +- GstQueueArray has seen a few API additions + (gst_queue_array_peek_nth(), gst_queue_array_set_clear_func(), + gst_queue_array_clear()) so that it can be used in other places like + GstAdapter instead of a GList, which reduces allocations and + improves performance. + +- appsink now reuses the sample object in pull_sample() if possible + +- rtpsession only starts the RTCP thread when it’s actually needed now + +- udpsrc uses a buffer pool now and the GstUdpSrc object structure was + optimised for better cache performance + +GstPlayer + +- API was added to fine-tune the synchronisation offset between + subtitles and video + + +Miscellaneous changes + +- As a result of moving to newer FFmpeg APIs, encoder and decoder + elements exposed by the GStreamer FFmpeg wrapper plugin (gst-libav) + may have seen possibly incompatible changes to property names and/or + types, and not all properties exposed might be functional. We are + still reviewing the new properties and aim to minimise breaking + changes at least for the most commonly-used properties, so please + report any issues you run into! + +OpenGL integration + +- The OpenGL mixer elements have been moved from -bad to + gst-plugins-base (see above) + +- The Mesa GBM backend now supports headless mode + +- gloverlaycompositor: New OpenGL-based compositor element that + flattens any overlays from GstVideoOverlayCompositionMetas into the + video stream. + +- glalpha: New element that adds an alpha channel to a video stream. + The values of the alpha channel can either be set to a constant or + can be dynamically calculated via chroma keying. It is similar to + the existing alpha element but based on OpenGL. Calculations are + done in floating point so results may not be identical to the output + of the existing alpha element. + +- glupload: Implement direct dmabuf uploader, the idea being that some + GPUs (like the Vivante series) can actually perform the YUV->RGB + conversion internally, so no custom conversion shaders are needed. + To make use of this feature, we need an additional uploader that can + import DMABUF FDs and also directly pass the pixel format, relying + on the GPU to do the conversion. + + +Tracing framework and debugging improvements + +- There is now a GDB PRETTY PRINTER FOR VARIOUS GSTREAMER TYPES: For + GstObject pointers the type and name is added, e.g. + 0x5555557e4110 [GstDecodeBin|decodebin0]. For GstMiniObject pointers + the object type is added, e.g. 0x7fffe001fc50 [GstBuffer]. For + GstClockTime and GstClockTimeDiff the time is also printed in human + readable form, e.g. 150116219955 [+0:02:30.116219955]. + +- GDB EXTENSION WITH TWO CUSTOM GDB COMMANDS gst-dot AND gst-print: + + - gst-dot creates dot files that a very close to what + GST_DEBUG_BIN_TO_DOT_FILE() produces, but object properties and + buffer contents such as codec-data in caps are not available. + + - gst-print produces high-level information about a GStreamer + object. This is currently limited to pads for GstElements and + events for the pads. The output may look like this: + + (gdb) gst-print pad.object.parent + GstMatroskaDemux (matroskademux0) { + SinkPad (sink, pull) { + } + SrcPad (video_0, push) { + events: + stream-start: + stream-id: 0463ccb080d00b8689bf569a435c4ff84f9ff753545318ae2328ea0763fd0bec/001:1274058367 + caps: video/x-theora + width: 1920 + height: 800 + pixel-aspect-ratio: 1/1 + framerate: 24/1 + streamheader: < 0x5555557c7d30 [GstBuffer], 0x5555557c7e40 [GstBuffer], 0x7fffe00141d0 [GstBuffer] > + segment: time + rate: 1 + tag: global + container-format: Matroska + } + SrcPad (audio_0, push) { + events: + stream-start: + stream-id: 0463ccb080d00b8689bf569a435c4ff84f9ff753545318ae2328ea0763fd0bec/002:1551204875 + caps: audio/mpeg + mpegversion: 4 + framed: true + stream-format: raw + codec_data: 0x7fffe0014500 [GstBuffer] + level: 2 + base-profile: lc + profile: lc + channels: 2 + rate: 44100 + segment: time + rate: 1 + tag: global + container-format: Matroska + tag: stream + audio-codec: MPEG-4 AAC audio + language-code: en + } + } -- The `timecodewait` element got renamed to `avwait` and can operate in - different modes now. +- gst_structure_to_string() now serialises the actual value of + pointers when serialising GstStructures instead of claiming they’re + NULL. This makes debug logging in various places less confusing, + because it’s clear now that structure fields actually hold valid + objects. Such object pointer values will never be deserialised + however. -- The `opencv` video processing plugin has gained a new `dewarp` element that - dewarps fisheye images. -- `ttml` is a new plugin for parsing and rendering subtitles in Timed Text - Markup Language (TTML) format. For the time being these elements will not - be autoplugged during media playback however, unless the `GST_TTML_AUTOPLUG=1` - environment variable is set. Only the EBU-TT-D profile is supported at this - point. +Tools -[dssim]: https://github.com/pornel/dssim +- gst-inspect-1.0 has coloured output now and will automatically use a + pager if the output does not fit on a page. This only works in a + UNIX environment and if the output is not piped, and on Windows 10 + build 16257 or newer. If you don’t like the colours you can disable + them by setting the GST_INSPECT_NO_COLORS=1 environment variable or + passing the --no-color command line option. -### New element features and additions -- `x264enc` can now use multiple x264 library versions compiled for different - bit depths at runtime, to transparently provide support for multiple bit - depths. A new configure parameter `--with-x264-libraries` has been added to - specify additional paths to look for additional x264 libraries to load. - Background is that the libx264 library is always compile for one specific - bit depth and the `x264enc` element would simply support the depth supported - by the underlying library. Now we can support multiple depths. +GStreamer RTSP server -- `x264enc` also picks up the interlacing mode automatically from the input - caps now and passed interlacing/TFF information correctly to the library. +- Improved backlog handling when using TCP interleaved for data + transport. Before there was a fixed maximum size for backlog + messages, which was prone to deadlocks and made it difficult to + control memory usage with the watch backlog. The RTSP server now + limits queued TCP data messages to one per stream, moving queuing of + the data into the pipeline and leaving the RTSP connection + responsive to RTSP messages in both directions, preventing all those + problems. -- `videoscale` and `videoconvert` now support multi-threaded scaling and - conversion, which is particularly useful with higher resolution video. - This has to be enabled explicitly via the `"n-threads"` property. +- Initial ULP Forward Error Correction support in rtspclientsink and + for RECORD mode in the server. -- `videorate`'s new `"rate"` property lets you set a speed factor - on the output stream +- API to explicitly enable retransmission requests (RTX) -- `splitmuxsink`'s buffer collection and scheduling was rewritten to make - processing and splitting deterministic; before it was possible for a buffer - to end up in a different file chunk in different runs. `splitmuxsink` also - gained a new `"format-location-full"` signal that works just like the existing - `"format-location"` signal only that it is also passed the primary stream's - first buffer as argument, so that it is possible to construct the file name - based on metadata such as the buffer timestamp or any GstMeta attached to - the buffer. The new `"max-size-timecode"` property allows for timecode-based - splitting. `splitmuxsink` will now also automatically start a new file if the - input caps change in an incompatible way. +- Lots of multicast-related fixes -- `fakesink` has a new `"drop-out-of-segment"` property to not drop - out-of-segment buffers, which is useful for debugging purposes. +- rtsp-auth: Add support for parsing .htdigest files -- `identity` gained a `"ts-offset"` property. -- both `fakesink` and `identity` now also print what kind of metas are attached - to buffers when printing buffer details via the `"last-message"` property - used by `gst-launch-1.0 -v`. +GStreamer VAAPI -- multiqueue: made `"min-interleave-time"` a configurable property. +- this section will be filled in in due course -- video nerds will be thrilled to know that `videotestsrc`'s snow is now - deterministic. `videotestsrc` also gained some new properties to make the - ball pattern based on system time, and invert colours each second - (`"animation-mode"`, `"motion"`, and `"flip"` properties). - -- `oggdemux` reverse playback should work again now. You're welcome. - -- `playbin3` and `urisourcebin` now have buffering enabled by default, and - buffering message aggregation was fixed. - -- `tcpclientsrc` now has a `"timeout"` property - -- `appsink` has gained support for buffer lists. For backwards compatibility - reasons users need to enable this explicitly with `gst_app_sink_set_buffer_list_support()`, - however. Once activated, a pulled `GstSample` can contain either a buffer - list or a single buffer. - -- `splitmuxsrc` reverse playback was fixed and handling of sparse streams, such - as subtitle tracks or metadata tracks, was improved. - -- `matroskamux` has acquired support for muxing G722 audio; it also marks all - buffers as keyframes now when streaming only audio, so that `tcpserversink` - will behave properly with audio-only streams. - -- `qtmux` gained support for ProRes 4444 XQ, HEVC/H.265 and CineForm (GoPro) formats, - and generally writes more video stream-related metadata into the track headers. - It is also allows configuration of the maximum interleave size in bytes and - time now. For fragmented mp4 we always write the `tfdt` atom now as required - by the DASH spec. - -- `qtdemux` supports FLAC, xvid, mp2, S16L and CineForm (GoPro) tracks now, and - generally tries harder to extract more video-related information from track - headers, such as colorimetry or interlacing details. It also received a - couple of fixes for the scenario where upstream operates in TIME format and - feeds chunks to qtdemux (e.g. DASH or MSE). - -- `audioecho` has two new properties to apply a delay only to certain channels - to create a surround effect, rather than an echo on all channels. This is - useful when upmixing from stereo, for example. The `"surround-delay"` property - enables this, and the `"surround-mask"` property controls which channels - are considered surround sound channels in this case. - -- `webrtcdsp` gained various new properties for gain control and also exposes - voice activity detection now, in which case it will post `"voice-activity"` - messages on the bus whenever the voice detection status changes. - -- The `decklink` capture elements for Blackmagic Decklink cards have seen a - number of improvements: - - - `decklinkvideosrc` will post a warning message on "no signal" and an info - message when the signal lock has been (re)acquired. There is also a new - read-only `"signal"` property that can be used to query the signal lock - status. The `GAP` flag will be set on buffers that are captured without - a signal lock. The new `drop-no-signal-frames` will make `decklinkvideosrc` - drop all buffers that have been captured without an input signal. The - `"skip-first-time"` property will make the source drop the first few - buffers, which is handy since some devices will at first output buffers - with the wrong resolution before they manage to figure out the right input - format and decide on the actual output caps. - - - `decklinkaudiosrc` supports more than just 2 audio channels now. - - - The capture sources no longer use the "hardware" timestamps which turn - out to be useless and instead just use the pipeline clock directly. - -- `srtpdec` now also has a readonly `"stats"` property, just like `srtpenc`. - -- `rtpbin` gained RTP bundle support, as used by e.g. WebRTC. The first - rtpsession will have a `rtpssrcdemux` element inside splitting the streams - based on their SSRC and potentially dispatch to a different rtpsession. - Because retransmission SSRCs need to be merged with the corresponding media - stream the `::on-bundled-ssrc` signal is emitted on `rtpbin` so that the - application can find out to which session the SSRC belongs. - -- `rtprtxqueue` gained two new properties exposing retransmission - statistics (`"requests"` and `"fulfilled-requests"`) - -- `kmssink` will now use the preferred mode for the monitor and render to the - base plane if nothing else has set a mode yet. This can also be done forcibly - in any case via the new `"force-modesetting"` property. Furthermore, `kmssink` - now allows only the supported connector resolutions as input caps in order to - avoid scaling or positioning of the input stream, as `kmssink` can't know - whether scaling or positioning would be more appropriate for the use case at - hand. - -- `waylandsink` can now take DMAbuf buffers as input in the presence - of a compatible Wayland compositor. This enables zero-copy transfer - from a decoder or source that outputs DMAbuf. - -- `udpsrc` can be bound to more than one interface when joining a - multicast group, this is done by giving a comma separate list of - interfaces such as multicast-iface="eth0,eth1". - -### Plugin moves - -- `dataurisrc` moved from gst-plugins-bad to core - -- The `rawparse` plugin containing the `rawaudioparse` and `rawvideoparse` - elements moved from gst-plugins-bad to gst-plugins-base. These elements - supersede the old `videoparse` and `audioparse` elements. They work the - same, with just some minor API changes. The old legacy elements still - exist in gst-plugins-bad, but may be removed at some point in the future. - -- `timecodestamper` is an element that attaches time codes to video buffers - in form of `GstVideoTimeCodeMeta`s. It had a `"clock-source"` property - which has now been removed because it was fairly useless in practice. It - gained some new properties however: the `"first-timecode"` property can - be used to set the inital timecode; alternatively `"first-timecode-to-now"` - can be set, and then the current system time at the time the first buffer - arrives is used as base time for the time codes. - - -### Plugin removals - -- The `mad` mp1/mp2/mp3 decoder plugin was removed from gst-plugins-ugly, - as libmad is GPL licensed, has been unmaintained for a very long time, and - there are better alternatives available. Use the `mpg123audiodec` element - from the `mpg123` plugin in gst-plugins-ugly instead, or `avdec_mp3` from - the `gst-libav` module which wraps the ffmpeg library. We expect that we - will be able to move mp3 decoding to gst-plugins-good in the next cycle - seeing that most patents around mp3 have expired recently or are about to - expire. - -- The `mimic` plugin was removed from gst-plugins-bad. It contained a decoder - and encoder for a video codec used by MSN messenger many many years ago (in - a galaxy far far away). The underlying library is unmaintained and no one - really needs to use this codec any more. Recorded videos can still be played - back with the MIMIC decoder in gst-libav. - -## Miscellaneous API additions - -- Request pad name templates passed to `gst_element_request_pad()` may now - contain multiple specifiers, such as e.g. `src_%u_%u`. - -- [`gst_buffer_iterate_meta_filtered()`][buffer-iterate-meta-filtered] is a - variant of `gst_buffer_iterate_meta()` that only returns metas of the - requested type and skips all other metas. - -- [`gst_pad_task_get_state()`][pad-task-get-state] gets the current state of - a task in a thread-safe way. - -- [`gst_uri_get_media_fragment_table()`][uri-get-fragment-table] provides the - media fragments of an URI as a table of key=value pairs. - -- [`gst_print()`][print], [`gst_println()`][println], [`gst_printerr()`][printerr], - and [`gst_printerrln()`][printerrln] can be used to print to stdout or stderr. - These functions are similar to `g_print()` and `g_printerr()` but they also - support all the additional format specifiers provided by the GStreamer - logging system, such as e.g. `GST_PTR_FORMAT`. - -- a `GstParamSpecArray` has been added, for elements who want to have array - type properties, such as the `audiomixmatrix` element for example. There are - also two new functions to set and get properties of this type from bindings: - - gst_util_set_object_array() - - gst_util_get_object_array() - -- various helper functions have been added to make it easier to set or get - GstStructure fields containing caps-style array or list fields from language - bindings (which usually support GValueArray but don't know about the GStreamer - specific fundamental types): - - [`gst_structure_get_array()`][get-array] - - [`gst_structure_set_array()`][set-array] - - [`gst_structure_get_list()`][get-list] - - [`gst_structure_set_list()`][set-list] - -- a new ['dynamic type' registry factory type][dynamic-type] was added to - register dynamically loadable GType types. This is useful for automatically - loading enum/flags types that are used in caps, such as for example the - `GstVideoMultiviewFlagsSet` type used in multiview video caps. - -- there is a new [`GstProxyControlBinding`][proxy-control-binding] for use - with GstController. This allows proxying the control interface from one - property on one GstObject to another property (of the same type) in another - GstObject. So e.g. in parent-child relationship, one may need to call - `gst_object_sync_values()` on the child and have a binding (set elsewhere) - on the parent update the value. This is used in `glvideomixer` and `glsinkbin` - for example, where `sync_values()` on the child pad or element will call - `sync_values()` on the exposed bin pad or element. - - Note that this doesn't solve GObject property forwarding, that must - be taken care of by the implementation manually or using GBinding. - -- `gst_base_parse_drain()` has been made public for subclasses to use. - -- `gst_base_sink_set_drop_out_of_segment()' can be used by subclasses to - prevent GstBaseSink from dropping buffers that fall outside of the segment. - -- [`gst_calculate_linear_regression()`][calc-lin-regression] is a new utility - function to calculate a linear regression. - -- [`gst_debug_get_stack_trace`][get-stack-trace] is an easy way to retrieve a - stack trace, which can be useful in tracer plugins. - -- allocators: the dmabuf allocator is now sub-classable, and there is a new - `GST_CAPS_FEATURE_MEMORY_DMABUF` define. - -- video decoder subclasses can use the newly-added function - `gst_video_decoder_allocate_output_frame_with_params()` to - pass a `GstBufferPoolAcquireParams` to the buffer pool for - each buffer allocation. - -- the video time code API has gained a dedicated [`GstVideoTimeCodeInterval`][timecode-interval] - type plus related API, including functions to add intervals to timecodes. - -- There is a new `libgstbadallocators-1.0` library in gst-plugins-bad, which - may go away again in future releases once the `GstPhysMemoryAllocator` - interface API has been validated by more users and was moved to - `libgstallocators-1.0` from gst-plugins-base. - -[timecode-interval]: https://gstreamer.freedesktop.org/data/doc/gstreamer/head/gst-plugins-base-libs/html/gst-plugins-base-libs-gstvideo.html#gst-video-time-code-interval-new -[buffer-iterate-meta-filtered]: https://gstreamer.freedesktop.org/data/doc/gstreamer/head/gstreamer/html/GstBuffer.html#gst-buffer-iterate-meta-filtered -[pad-task-get-state]: https://gstreamer.freedesktop.org/data/doc/gstreamer/head/gstreamer/html/GstPad.html#gst-pad-task-get-state -[uri-get-fragment-table]: https://gstreamer.freedesktop.org/data/doc/gstreamer/head/gstreamer/html/gstreamer-GstUri.html#gst-uri-get-media-fragment-table -[print]: https://gstreamer.freedesktop.org/data/doc/gstreamer/head/gstreamer/html/gstreamer-GstInfo.html#gst-print -[println]: https://gstreamer.freedesktop.org/data/doc/gstreamer/head/gstreamer/html/gstreamer-GstInfo.html#gst-println -[printerr]: https://gstreamer.freedesktop.org/data/doc/gstreamer/head/gstreamer/html/gstreamer-GstInfo.html#gst-printerr -[printerrln]: https://gstreamer.freedesktop.org/data/doc/gstreamer/head/gstreamer/html/gstreamer-GstInfo.html#gst-printerrln -[get-array]: https://gstreamer.freedesktop.org/data/doc/gstreamer/head/gstreamer/html/GstStructure.html#gst-structure-get-array -[set-array]: https://gstreamer.freedesktop.org/data/doc/gstreamer/head/gstreamer/html/GstStructure.html#gst-structure-set-array -[get-list]: https://gstreamer.freedesktop.org/data/doc/gstreamer/head/gstreamer/html/GstStructure.html#gst-structure-get-list -[set-list]: https://gstreamer.freedesktop.org/data/doc/gstreamer/head/gstreamer/html/GstStructure.html#gst-structure-set-list -[dynamic-type]: https://gstreamer.freedesktop.org/data/doc/gstreamer/head/gstreamer/html/GstDynamicTypeFactory.html -[proxy-control-binding]: https://gstreamer.freedesktop.org/data/doc/gstreamer/head/gstreamer-libs/html/gstreamer-libs-GstProxyControlBinding.html -[calc-lin-regression]: https://gstreamer.freedesktop.org/data/doc/gstreamer/head/gstreamer/html/gstreamer-GstUtils.html#gst-calculate-linear-regression -[get-stack-trace]: https://gstreamer.freedesktop.org/data/doc/gstreamer/head/gstreamer/html/gstreamer-GstUtils.html#gst-debug-get-stack-trace - -### GstPlayer -New API has been added to: +GStreamer OMX - - get the number of audio/video/subtitle streams: - - `gst_player_media_info_get_number_of_streams()` - - `gst_player_media_info_get_number_of_video_streams()` - - `gst_player_media_info_get_number_of_audio_streams()` - - `gst_player_media_info_get_number_of_subtitle_streams()` +- Add support of NV16 format to video encoders input. - - enable accurate seeking: `gst_player_config_set_seek_accurate()` - and `gst_player_config_get_seek_accurate()` +- Video decoders now handle the ALLOCATION query to tell upstream + about the number of buffers they require. Video encoders will also + use this query to adjust their number of allocated buffers + preventing starvation when using dynamic buffer mode. - - get a snapshot image of the video in RGBx, BGRx, JPEG, PNG or - native format: [`gst_player_get_video_snapshot()`][snapshot] - - - selecting use of a specific video sink element - ([`gst_player_video_overlay_video_renderer_new_with_sink()`][renderer-with-vsink]) - - - If the environment variable `GST_PLAYER_USE_PLAYBIN3` is set, GstPlayer will - use the still-experimental `playbin3` element and the `GstStreams` API for - playback. +- The OMX_PERFORMANCE debug category has been renamed to OMX_API_TRACE + and can now be used to track a widder variety of interactions + between OMX and GStreamer. -[snapshot]: https://gstreamer.freedesktop.org/data/doc/gstreamer/head/gst-plugins-bad-libs/html/gst-plugins-bad-libs-gstplayer.html#gst-player-get-video-snapshot -[renderer-with-vsink]: https://gstreamer.freedesktop.org/data/doc/gstreamer/head/gst-plugins-bad-libs/html/gst-plugins-bad-libs-gstplayer-videooverlayvideorenderer.html#gst-player-video-overlay-video-renderer-new-with-sink +- Video encoders will now detect frame rate only changes and will + inform OMX about it rather than doing a full format reset. -## Miscellaneous changes +- Various Zynq UltraScale+ specific improvements: + - Video encoders are now able to import dmabuf from upstream. + - Support for HEVC range extension profiles and more AVC profiles. + - We can now request video encoders to generate an IDR using the + force key unit event. -- video caps for interlaced video may contain an optional `"field-order"` field - now in the case of `interlaced-mode=interleaved` to signal that the field - order is always the same throughout the stream. This is useful to signal to - muxers such as mp4mux. The new field is parsed from/to `GstVideoInfo` of course. -- video decoder and video encoder base classes try harder to proxy - interlacing, colorimetry and chroma-site related fields in caps properly. +GStreamer Editing Services and NLE -- The buffer stored in the `PROTECTION` events is now left unchanged. This is a - change of behaviour since 1.8, especially for the mssdemux element which used to - decode the base64 parsed data wrapped in the protection events emitted by the - demuxer. +- this section will be filled in in due course -- `PROTECTION` events can now be injected into the pipeline from the application; - source elements deriving from GstBaseSrc will forward those downstream now. -- The DASH demuxer is now correctly parsing the MSPR-2.0 ContentProtection nodes - and emits Protection events accordingly. Applications relying on those events - might need to decode the base64 data stored in the event buffer before using it. +GStreamer validate -- The registry can now also be disabled by setting the environment variable - `GST_REGISTRY_DISABLE=yes`, with similar effect as the `GST_DISABLE_REGISTRY` - compile time switch. +- this section will be filled in in due course -- Seeking performance with gstreamer-vaapi based decoders was improved. It would - recreate the decoder and surfaces on every seek which can be quite slow. -- more robust handling of input caps changes in videoaggregator-based elements - such as `compositor`. +GStreamer Python Bindings -- Lots of adaptive streaming-related fixes across the board (DASH, MSS, HLS). Also: +- add binding for gst_pad_set_caps() - - `mssdemux`, the Microsoft Smooth Streaming demuxer, has seen various - fixes for live streams, duration reporting and seeking. +- pygobject dependency requirement was bumped to >= 3.8 - - The DASH manifest parser now extracts MS PlayReady ContentProtection objects - from manifests and sends them downstream as `PROTECTION` events. It also - supports multiple Period elements in external xml now. +- new audiotestsrc, audioplot, and mixer plugin examples, and a + dynamic pipeline example -- gst-libav was updated to ffmpeg 3.3 but should still work with any 3.x - version. -- GstEncodingProfile has been generally enhanced so it can, for - example, be used to get possible profiles for a given file - extension. It is now possible to define profiles based on element - factory names or using a path to a `.gep` file containing a - serialized profile. +GStreamer C# Bindings -- `audioconvert` can now do endianness conversion in-place. All other - conversions still require a copy, but e.g. sign conversion and a few others - could also be implemented in-place now. +- bindings for the GstWebRTC library -- The new, experimental `playbin3` and `urisourcebin` elements got many - bugfixes and improvements and should generally be closer to a full - replacement of the old elements. -- `interleave` now supports > 64 channels. +GStreamer Rust Bindings -### OpenGL integration +The GStreamer Rust bindings are now officially part of the GStreamer +project and are also maintained in the GStreamer GitLab. -- As usual the GStreamer OpenGL integration library has seen numerous - fixes and performance improvements all over the place, and is hopefully - ready now to become API stable and be moved to gst-plugins-base during the - 1.14 release cycle. +The releases will generally not be synchronized with the releases of +other GStreamer parts due to dependencies on other projects. -- The GStreamer OpenGL integration layer has also gained support for the - Vivante EGL FB windowing system, which improves performance on platforms - such as Freescale iMX.6 for those who are stuck with the proprietary driver. - The `qmlglsink` element also supports this now if Qt is used with eglfs or - wayland backend, and it works in conjunction with [gstreamer-imx][gstreamer-imx] - of course. +Also unlike the other GStreamer libraries, the bindings will not commit +to full API stability but instead will follow the approach that is +generally taken by Rust projects, e.g.: -- various `qmlglsrc` improvements +1) 0.12.X will be completely API compatible with all other 0.12.Y + versions. +2) 0.12.X+1 will contain bugfixes and compatible new feature additions. +3) 0.13.0 will _not_ be backwards compatible with 0.12.X but projects + will be able to stay at 0.12.X without any problems as long as they + don’t need newer features. -[gstreamer-imx]: https://github.com/Freescale/gstreamer-imx +The current stable release is 0.12.2 and the next release series will be +0.13, probably around March 2019. -## Tracing framework and debugging improvements +At this point the bindings cover most of GStreamer core (except for most +notably GstAllocator and GstMemory), and most parts of the app, audio, +base, check, editing-services, gl, net. pbutils, player, rtsp, +rtsp-server, sdp, video and webrtc libraries. -- New tracing hooks have been added to track GstMiniObject and GstObject - ref/unref operations. +Also included is support for creating subclasses of the following types +and writing GStreamer plugins: -- The memory leaks tracer can optionally use this to retrieve stack traces if - enabled with e.g. `GST_TRACERS=leaks(filters="GstEvent,GstMessage",stack-traces-flags=full)` +- gst::Element +- gst::Bin and gst::Pipeline +- gst::URIHandler and gst::ChildProxy +- gst::Pad, gst::GhostPad +- gst_base::Aggregator and gst_base::AggregatorPad +- gst_base::BaseSrc and gst_base::BaseSink +- gst_base::BaseTransform -- The `GST_DEBUG_FILE` environment variable, which can be used to write the - debug log output to a file instead of printing it to stderr, can now contain - a name pattern, which is useful for automated testing and continuous - integration systems. The following format specifiers are supported: +Changes to 0.12.X since 0.12.0 - - `%p`: will be replaced with the PID - - `%r`: will be replaced with a random number, which is useful for instance - when running two processes with the same PID but in different containers. +Fixed -## Tools +- PTP clock constructor actually creates a PTP instead of NTP clock -- `gst-inspect-1.0` can now list elements by type with the new `--types` - command-line option, e.g. `gst-inspect-1.0 --types=Audio/Encoder` will - show a list of audio encoders. +Added -- `gst-launch-1.0` and `gst_parse_launch()` have gained a new operator (`:`) - that allows linking all pads between two elements. This is useful in cases - where the exact number of pads or type of pads is not known beforehand, such - as in the `uridecodebin : encodebin` scenario, for example. In this case, - multiple links will be created if the encodebin has multiple profiles - compatible with the output of uridecodebin. +- Bindings for GStreamer Editing Services +- Bindings for GStreamer Check testing library +- Bindings for the encoding profile API (encodebin) -- `gst-device-monitor-1.0` now shows a `gst-launch-1.0` snippet for each - device that shows how to make use of it in a `gst-launch-1.0` pipeline string. +- VideoFrame, VideoInfo, AudioInfo, StructureRef implements Send and + Sync now +- VideoFrame has a function to get the raw FFI pointer +- From impls from the Error/Success enums to the combined enums like + FlowReturn +- Bin-to-dot file functions were added to the Bin trait +- gst_base::Adapter implements SendUnique now +- More complete bindings for the gst_video::VideoOverlay interface, + especially + gst_video::is_video_overlay_prepare_window_handle_message() -## GStreamer RTSP server +Changed -- The RTSP server now also supports Digest authentication in addition to Basic - authentication. +- All references were updated from GitHub to freedesktop.org GitLab +- Fix various links in the README.md +- Link to the correct location for the documentation +- Remove GitLab badge as that only works with gitlab.com currently -- The `GstRTSPClient` class has gained a `pre-*-request` signal and virtual - method for each client request type, emitted in the beginning of each rtsp - request. These signals or virtual methods let the application validate the - requests, configure the media/stream in a certain way and also generate error - status codes in case of an error or a bad request. +Changes in git master for 0.13 -## GStreamer VAAPI +Fixed -- GstVaapiDisplay now inherits from GstObject, thus the VA display logging - messages are better and tracing the context sharing is more readable. +- gst::tag::Album is the album tag now instead of artist sortname -- When uploading raw images into a VA surfaces now VADeriveImages are tried - fist, improving the upload performance, if it is possible. +Added -- The decoders and the post-processor now can push dmabuf-based buffers to - downstream under certain conditions. For example: +- Subclassing infrastructure was moved directly into the bindings, + making the gst-plugin crate deprecated. This involves many API + changes but generally cleans up code and makes it more flexible. + Take a look at the gst-plugins-rs crate for various examples. - `GST_GL_PLATFORM=egl gst-play-1.0 video-sample.mkv --videosink=glimagesink` +- Bindings for CapsFeatures and Meta +- Bindings for + ParentBufferMeta,VideoMetaandVideoOverlayCompositionMeta` +- Bindings for VideoOverlayComposition and VideoOverlayRectangle +- Bindings for VideoTimeCode -- Refactored the wrapping of VA surface into gstreamer memory, adding lock - when mapping and unmapping, and many other fixes. +- UniqueFlowCombiner and UniqueAdapter wrappers that make use of the + Rust compile-time mutability checks and expose more API in a safe + way, and as a side-effect implement Sync and Send now -- Now `vaapidecodebin` loads `vaapipostproc` dynamically. It is possible to - avoid it usage with the environment variable `GST_VAAPI_DISABLE_VPP=1`. +- More complete bindings for Allocation Query +- pbutils functions for codec descriptions +- TagList::iter() for iterating over all tags while getting a single + value per tag. The old ::iter_tag_list() function was renamed to + ::iter_generic() and still provides access to each value for a tag +- Bus::iter() and Bus::iter_timed() iterators around the corresponding + ::pop\*() functions -- Regarding encoders: they have primary rank again, since they can discover, - in run-time, the color formats they can use for upstream raw buffers and - caps renegotiation is now possible. Also the encoders push encoding info - downstream via tags. +- serde serialization of Value can also handle Buffer now -- About specific encoders: added constant bit-rate encoding mode for VP8 and - H265 encoder handles P010_10LE color format. +- Extensive comments to all examples with explanations +- Transmuxing example showing how to use typefind, multiqueue and + dynamic pads +- basic-tutorial-12 was ported and added -- Regarding decoders, flush operation has been improved, now the internal VA - encoder is not recreated at each flush. Also there are several improvements - in the handling of H264 and H265 streams. +Changed -- VAAPI plugins try to create their on GstGL context (when available) if they - cannot find it in the pipeline, to figure out what type of VA Display they - should create. +- Rust 1.31 is the minimum supported Rust version now +- Update to latest gir code generator and glib bindings -- Regarding `vaapisink` for X11, if the backend reports that it is unable to - render correctly the current color format, an internal VA post-processor, is - instantiated (if available) and converts the color format. +- Functions returning e.g. gst::FlowReturn or other “combined” enums + were changed to return split enums like + Result to allow usage of the + standard Rust error handling. -## GStreamer Editing Services and NLE +- MiniObject subclasses are now newtype wrappers around the underlying + GstRc wrapper. This does not change the API in any breaking + way for the current usages, but allows MiniObjects to also be + implemented in other crates and makes sure rustdoc places the + documentation in the right places. -- Enhanced auto transition behaviour +- BinExt extension trait was renamed to GstBinExt to prevent conflicts + with gtk::Bin if both are imported -- Fix some races in `nlecomposition` +- Buffer::from_slice() can’t possible return None -- Allow building with msvc +- Various clippy warnings -- Added a UNIX manpage for `ges-launch` -- API changes: - - Added ges_deinit (allowing the leak tracer to work properly) - - Added ges_layer_get_clips_in_interval - - Finally hide internal symbols that should never have been exposed +GStreamer Rust Plugins -## GStreamer validate +Like the GStreamer Rust bindings, the Rust plugins are now officially +part of the GStreamer project and are also maintained in the GStreamer +GitLab. -- Port `gst-validate-launcher` to python 3 +In the 0.3.x versions this contained infrastructure for writing +GStreamer plugins in Rust, and a set of plugins. -- `gst-validate-launcher` now checks if blacklisted bugs have been fixed on - bugzilla and errors out if it is the case +In git master that infrastructure was moved to the GLib and GStreamer +bindings directly, together with many other improvements that were made +possible by this, so the gst-plugins-rs repository only contains +GStreamer elements now. -- Allow building with msvc +Elements included are: -- Add ability for the launcher to run GStreamer unit tests +- Tutorials plugin: identity, rgb2gray and sinesrc with extensive + comments -- Added a way to activate the leaks tracer on our tests and fix leaks +- rsaudioecho, a port of the audiofx element -- Make the http server multithreaded +- rsfilesrc, rsfilesink -- New testsuite for running various test scenarios on the DASH-IF test vectors +- rsflvdemux, a FLV demuxer. Not feature-equivalent with flvdemux yet -## Build and Dependencies +- threadshare plugin: ts-appsrc, ts-proxysrc/sink, ts-queue, ts-udpsrc + and ts-tcpclientsrc elements that use a fixed number of threads and + share them between instances. For more background about these + elements see Sebastian’s talk “When adding more threads adds more + problems - Thread-sharing between elements in GStreamer” at the + GStreamer Conference 2017. -- Meson build files are now disted in tarballs, for jhbuild and so distro - packagers can start using it. Note that the Meson-based build system is not - 100% feature-equivalent with the autotools-based one yet. +- rshttpsrc, a HTTP source around the hyper/reqwest Rust libraries. + Not feature-equivalent with souphttpsrc yet. -- Some plugin filenames have been changed to match the plugin names: for example - the file name of the `encoding` plugin in gst-plugins-base containing the - `encodebin` element was `libgstencodebin.so` and has been changed to - `libgstencodebin.so`. This affects only a handful of plugins across modules. +- togglerecord, an element that allows to start/stop recording at any + time and keeps all audio/video streams in sync. - **Developers who install GStreamer from source and just do `make install`** - **after updating the source code, without doing `make uninstall` first, will** - **have to manually remove the old installed plugin files from the installation** - **prefix, or they will get 'Cannot register existing type' critical warnings.** +- mccparse and mccenc, parsers and encoders for the MCC closed caption + file format. -- Most of the docbook-based documentation (FAQ, Application Development Manual, - Plugin Writer's Guide, design documents) has been converted to markdown and - moved into a new gst-docs module. The gtk-doc library API references and - the plugins documentation are still built as part of the source modules though. +Changes to 0.3.X since 0.3.0 -- GStreamer core now optionally uses libunwind and libdw to generate backtraces. - This is useful for tracer plugins used during debugging and development. +- All references were updated from GitHub to freedesktop.org GitLab +- Fix various links in the README.md +- Link to the correct location for the documentation -- There is a new `libgstbadallocators-1.0` library in gst-plugins-bad (which - may go away again in future releases once the `GstPhysMemoryAllocator` - interface API has been validated by more users). +Changes in git master for 0.4 -- `gst-omx` and `gstreamer-vaapi` modules can now also be built using the - Meson build system. +- togglerecord: Switch to parking_lot crate for mutexes/condition + variables for lower overhead +- Merge threadshare plugin here +- New closedcaption plugin with mccparse and mccenc elements +- New identity element for the tutorials plugin -- The `qtkitvideosrc` element for macOS was removed. The API is deprecated - since 10.9 and it wasn't shipped in the binaries since a few releases. +- Register plugins statically in tests instead of relying on the + plugin loader to find the shared library in a specific place -## Platform-specific improvements +- Update to the latest API changes in the GLib and GStreamer bindings +- Update to the latest versions of all crates -### Android -- androidmedia: add support for VP9 video decoding/encoding and Opus audio - decoding (where supported) +Build and Dependencies -### OS/X and iOS +- The MESON BUILD SYSTEM BUILD IS NOW FEATURE-COMPLETE (*) and it is + now the recommended build system on all platforms and also used by + Cerbero to build GStreamer on all platforms. The Autotools build is + scheduled to be removed in the next cycle. Developers who currently + use gst-uninstalled should move to gst-build. The build option + naming has been cleaned up and made consistent and there are now + feature options to enable/disable plugins and various other features + on a case-by-case basis. (*) with the exception of plugin docs which + will be handled differently in future -- `avfvideosrc`, which represents an iPhone camera or, on a Mac, a screencapture - session, so far allowed you to select an input device by device index only. - New API adds the ability to select the position (front or back facing) and - device-type (wide angle, telephoto, etc.). Furthermore, you can now also - specify the orientation (portrait, landscape, etc.) of the videostream. +- Symbol export in libraries is now controlled via explicit exports + using symbol visibility or export defines where supported, to ensure + consistency across all platforms. This also allows libraries to have + exports that vary based on detected platform features and configure + options as is the case with the GStreamer OpenGL integration library + for example. A few symbols that had been exported by accident in + earlier versions may no longer be exported. These symbols will not + have had declarations in any public header files then though and + would not have been usable. -### Windows +- The GStreamer FFmpeg wrapper plugin (gst-libav) now depends on + FFmpeg 4.x and uses the new FFmpeg 4.x API and stopped relying on + ancient API that was removed with the FFmpeg 4.x release. This means + that it is no longer possible to build this module against an older + system-provided FFmpeg 3.x version. Use the internal FFmpeg 4.x copy + instead if you build using autotools, or use gst-libav 1.14.x + instead which targets the FFmpeg 3.x API and _should_ work fine in + combination with a newer GStreamer. It’s difficult for us to support + both old and new FFmpeg APIs at the same time, apologies for any + inconvenience caused. -- `dx9screencapsrc` can now optionally also capture the cursor. +- Hardware-accelerated Nvidia video encoder/decoder plugins nvdec and + nvenc can be built against CUDA Toolkit versions 9 and 10.0 now. The + dynlink interface has been dropped since it’s deprecated in 10.0. -## Contributors +- The (optional) OpenCV requirement has been bumped to >= 3.0.0 and + the plugin can also be built against OpenCV 4.x now. -Aleix Conchillo Flaque, Alejandro G. Castro, Aleksandr Slobodeniuk, Alexandru -Băluț, Alex Ashley, Andre McCurdy, Andrew, Anton Eliasson, Antonio Ospite, -Arnaud Vrac, Arun Raghavan, Aurélien Zanelli, Axel Menzel, Benjamin Otte, -Branko Subasic, Brendan Shanks, Carl Karsten, Carlos Rafael Giani, ChangBok -Chae, Chris Bass, Christian Schaller, christophecvr, Claudio Saavedra, -Corentin Noël, Dag Gullberg, Daniel Garbanzo, Daniel Shahaf, David Evans, -David Schleef, David Warman, Dominique Leuenberger, Dongil Park, Douglas -Bagnall, Edgard Lima, Edward Hervey, Emeric Grange, Enrico Jorns, Enrique -Ocaña González, Evan Nemerson, Fabian Orccon, Fabien Dessenne, Fabrice Bellet, -Florent Thiéry, Florian Zwoch, Francisco Velazquez, Frédéric Dalleau, Garima -Gaur, Gaurav Gupta, George Kiagiadakis, Georg Lippitsch, Göran Jönsson, Graham -Leggett, Guillaume Desmottes, Gurkirpal Singh, Haihua Hu, Hanno Boeck, Havard -Graff, Heekyoung Seo, hoonhee.lee, Hyunjun Ko, Imre Eörs, Iñaki García -Etxebarria, Jagadish, Jagyum Koo, Jan Alexander Steffens (heftig), Jan -Schmidt, Jean-Christophe Trotin, Jochen Henneberg, Jonas Holmberg, Joris -Valette, Josep Torra, Juan Pablo Ugarte, Julien Isorce, Jürgen Sachs, Koop -Mast, Kseniia Vasilchuk, Lars Wendler, leigh123linux@googlemail.com, Luis de -Bethencourt, Lyon Wang, Marcin Kolny, Marinus Schraal, Mark Nauwelaerts, -Mathieu Duponchelle, Matthew Waters, Matt Staples, Michael Dutka, Michael -Olbrich, Michael Smith, Michael Tretter, Miguel París Díaz, namanyadav12, Neha -Arora, Nick Kallen, Nicola Murino, Nicolas Dechesne, Nicolas Dufresne, Nicolas -Huet, Nirbheek Chauhan, Ole André Vadla Ravnås, Olivier Crête, Patricia -Muscalu, Peter Korsgaard, Peter Seiderer, Petr Kulhavy, Philippe Normand, -Philippe Renon, Philipp Zabel, Rahul Bedarkar, Reynaldo H. Verdejo Pinochet, -Ricardo Ribalda Delgado, Rico Tzschichholz, Руслан Ижбулатов, Samuel Maroy, -Santiago Carot-Nemesio, Scott D Phillips, Sean DuBois, Sebastian Dröge, Sergey -Borovkov, Seungha Yang, shakin chou, Song Bing, Søren Juul, Sreerenj -Balachandran, Stefan Kost, Stefan Sauer, Stepan Salenikovich, Stian Selnes, -Stuart Weaver, suhas2go, Thiago Santos, Thibault Saunier, Thomas Bluemel, -Thomas Petazzoni, Tim-Philipp Müller, Ting-Wei Lan, Tobias Mueller, Todor -Tomov, Tomasz Zajac, Ulf Olsson, Ursula Maplehurst, Víctor Manuel Jáquez Leal, -Victor Toso, Vincent Penquerc'h, Vineeth TM, Vinod Kesti, Vitor Massaru Iha, -Vivia Nikolaidou, WeiChungChang, William Manley, Wim Taymans, Wojciech -Przybyl, Wonchul Lee, Xavier Claessens, Yasushi SHOJI - -... and many others who have contributed bug reports, translations, sent +- New sctp plugin based on usrsctp (for WebRTC data channels) + +Cerbero + +Cerbero is a meta build system used to build GStreamer plus dependencies +on platforms where dependencies are not readily available, such as +Windows, Android, iOS and macOS. + +Cerbero has seen a number of improvements: + +- Cerbero has been ported to Python 3 and requires Python 3.5 or newer + now + +- Source tarballs are now protected by checksums in the recipes to + guard against download errors and malicious takeover of projects or + websites. In addition, downloads are only allowed via secure + transports now and plain HTTP, FTP and git:// transports are not + allowed anymore. + +- There is now a new fetch-bootstrap command which downloads sources + required for bootstrapping, with an optional --build-tools-only + argument to match the bootstrap --build-tools-only command. + +- The bootstrap, build, package and bundle-source commands gained a + new --offline switch that ensures that only sources from the cache + are used and never downloaded via the network. This is useful in + combination with the fetch and fetch-bootstrap commands that acquire + sources ahead of time before any build steps are executed. This + allows more control over the sources used and when sources are + updated, and is particularly useful for build environments that + don’t have network access. + +- bootstrap --assume-yes will automatically say ‘yes’ to any + interactive prompts during the bootstrap stage, such as those from + apt-get or yum. + +- bootstrap --system-only will only bootstrap the system without build + tools. + +- Manifest support: The build manifest can be used in continuous + integration (CI) systems to fixate the Git revision of certain + projects so that all builds of a pipeline are on the same reference. + This is used in GStreamer’s gitlab CI for example. It can also be + used in order to re-produce a specific build. To set a manifest, you + can set manifest = 'my_manifest.xml' in your configuration file, or + use the --manifest command line option. The command line option will + take precendence over anything specific in the configuration file. + +- The new build-deps command can be used to build only the + dependencies of a recipe, without the recipe itself. + +- new --list-variants command to list available variants + +- variants can now be set on the command line via the -v option as a + comma-separated list. This overrides any variants set in any + configuration files. + +- new qt5, intelmsdk and nvidia variants for enabling Qt5 and hardware + codec support. See the Enabling Optional Features with Variants + section in the Cerbero documentation for more details how to enable + and use these variants. + +- A new -t / --timestamp command line switch makes commands print + timestamps + + +Platform-specific changes and improvements + +Android + +- toolchain: update compiler to clang and NDKr18. NDK r18 removed the + armv5 target and only has Android platforms that target at least + armv7 so the armv5 target is not useful anymore. + +- The way that GIO modules are named has changed due to upstream GLib + natively adding support for loading static GIO modules. This means + that any GStreamer application using gnutls for SSL/TLS on the + Android or iOS platforms (or any other setup using static libraries) + will fail to link looking for the g_io_module_gnutls_load_static() + function. The new function name is now + g_io_gnutls_load(gpointer data). data can be NULL for a static + library. Look at this commit for the necessary change in the + examples. + +- various build issues on Android have been fixed. + +macOS and iOS + +- various build issues on iOS have been fixed. + +- the minimum required iOS version is now 9.0. The difference in + adoption between 8.0 and 9.0 is 0.1% and the bump to 9.0 fixes some + build issues. + +- The way that GIO modules are named has changed due to upstream GLib + natively adding support for loading static GIO modules. This means + that any GStreamer application using gnutls for SSL/TLS on the + Android or iOS platforms (or any other setup using static libraries) + will fail to link looking for the g_io_module_gnutls_load_static() + function. The new function name is now + g_io_gnutls_load(gpointer data). data can be NULL for a static + library. Look at this commit for the necessary change in the + examples. + +Windows + +- The webrtcdsp element is shipped again as part of the Windows binary + packages, the build system issue has been resolved. + +- ‘Inconsistent DLL linkage’ warnings when building with MSVC have + been fixed + +- Hardware-accelerated Nvidia video encoder/decoder plugins nvdec and + nvenc build on Windows now, also with MSVC and using Meson. + +- The ksvideosrc camera capture plugin supports 16-bit grayscale video + now + +- The wasapisrc audio capture element implements loopback recording + from another output device or sink + +- wasapisink recover from low buffer levels in shared mode and some + exclusive mode fixes + +- dshowsrc now implements the GstDeviceMonitor interface + + +Contributors + +Aleix Conchillo Flaqué, Alessandro Decina, Alexandru Băluț, Alex Ashley, +Alexey Chernov, Alicia Boya García, Amit Pandya, Andoni Morales +Alastruey, Andreas Frisch, Andre McCurdy, Andy Green, Anthony Violo, +Antoine Jacoutot, Antonio Ospite, Arun Raghavan, Aurelien Jarno, +Aurélien Zanelli, ayaka, Bananahemic, Bastian Köcher, Branko Subasic, +Brendan Shanks, Carlos Rafael Giani, Christoph Reiter, Corentin Noël, +Daeseok Youn, Daniel Drake, Daniel Klamt, Dardo D Kleiner, David Ing, +David Svensson Fors, Devarsh Thakkar, Dimitrios Katsaros, Edward Hervey, +Emilio Pozuelo Monfort, Enrique Ocaña González, Ezequiel Garcia, Fabien +Dessenne, Fabrizio Gennari, Florent Thiéry, Francisco Velazquez, +Freyr666, Garima Gaur, Gary Bisson, George Kiagiadakis, Georg Lippitsch, +Georg Ottinger, Geunsik Lim, Göran Jönsson, Guillaume Desmottes, H1Gdev, +Haihao Xiang, Haihua Hu, Harshad Khedkar, Havard Graff, He Junyan, +Hoonhee Lee, Hosang Lee, Hyunjun Ko, Ingo Randolf, Iñigo Huguet, James +Stevenson, Jan Alexander Steffens, Jan Schmidt, Jerome Laheurte, Jimmy +Ohn, Joakim Johansson, Jochen Henneberg, Johan Bjäreholt, John-Mark +Bell, John Nikolaides, Jonathan Karlsson, Jonny Lamb, Jordan Petridis, +Josep Torra, Joshua M. Doe, Jos van Egmond, Juan Navarro, Jun Xie, +Junyan He, Justin Kim, Kai Kang, Kim Tae Soo, Kirill Marinushkin, Kyrylo +Polezhaiev, Lars Petter Endresen, Linus Svensson, Louis-Francis +Ratté-Boulianne, Luis de Bethencourt, Luz Paz, Lyon Wang, Maciej Wolny, +Marc-André Lureau, Marc Leeman, Marcos Kintschner, Marian Mihailescu, +Marinus Schraal, Mark Nauwelaerts, Marouen Ghodhbane, Martin Kelly, +Matej Knopp, Mathieu Duponchelle, Matteo Valdina, Matthew Waters, +Matthias Fend, memeka, Michael Drake, Michael Gruner, Michael Olbrich, +Michael Tretter, Miguel Paris, Mike Wey, Mikhail Fludkov, Naveen +Cherukuri, Nicola Murino, Nicolas Dufresne, Niels De Graef, Nirbheek +Chauhan, Norbert Wesp, Ognyan Tonchev, Olivier Crête, Omar Akkila, +Patricia Muscalu, Patrick Radizi, Patrik Nilsson, Paul Kocialkowski, Per +Forlin, Peter Körner, Peter Seiderer, Petr Kulhavy, Philippe Normand, +Philippe Renon, Philipp Zabel, Pierre Labastie, Roland Jon, Roman +Sivriver, Rosen Penev, Russel Winder, Sam Gigliotti, Sean-Der, Sebastian +Dröge, Seungha Yang, Sjoerd Simons, Snir Sheriber, Song Bing, Soon, +Thean Siew, Sreerenj Balachandran, Stefan Ringel, Stephane Cerveau, +Stian Selnes, Suhas Nayak, Takeshi Sato, Thiago Santos, Thibault +Saunier, Thomas Bluemel, Tianhao Liu, Tim-Philipp Müller, Tomasz +Andrzejak, Tomislav Tustonić, U. Artie Eoff, Ulf Olsson, Varunkumar +Allagadapa, Víctor Guzmán, Víctor Manuel Jáquez Leal, Vincenzo Bono, +Vineeth T M, Vivia Nikolaidou, Wang Fei, wangzq, Whoopie, Wim Taymans, +Wind Yuan, Wonchul Lee, Xabier Rodriguez Calvar, Xavier Claessens, +Haihao Xiang, Yacine Bandou, Yeongjin Jeong, Yuji Kuwabara, Zeeshan Ali, + +… and many others who have contributed bug reports, translations, sent suggestions or helped testing. -## Bugs fixed in 1.12 -More than [635 bugs][bugs-fixed-in-1.12] have been fixed during -the development of 1.12. +Bugs fixed in 1.16 + +- this section will be filled in in due course + +More than XXX bugs have been fixed during the development of 1.16. This list does not include issues that have been cherry-picked into the -stable 1.10 branch and fixed there as well, all fixes that ended up in the -1.10 branch are also included in 1.12. +stable 1.16 branch and fixed there as well, all fixes that ended up in +the 1.16 branch are also included in 1.16. + +This list also does not include issues that have been fixed without a +bug report in bugzilla, so the actual number of fixes is much higher. + -This list also does not include issues that have been fixed without a bug -report in bugzilla, so the actual number of fixes is much higher. +Stable 1.16 branch -[bugs-fixed-in-1.12]: https://bugzilla.gnome.org/buglist.cgi?bug_status=RESOLVED&bug_status=VERIFIED&classification=Platform&limit=0&list_id=213265&order=bug_id&product=GStreamer&query_format=advanced&resolution=FIXED&target_milestone=1.10.1&target_milestone=1.10.2&target_milestone=1.10.3&target_milestone=1.10.4&target_milestone=1.11.1&target_milestone=1.11.2&target_milestone=1.11.3&target_milestone=1.11.4&target_milestone=1.11.90&target_milestone=1.11.91&target_milestone=1.12.0 +After the 1.16.0 release there will be several 1.16.x bug-fix releases +which will contain bug fixes which have been deemed suitable for a +stable branch, but no new features or intrusive changes will be added to +a bug-fix release usually. The 1.16.x bug-fix releases will be made from +the git 1.16 branch, which is a stable branch. -## Stable 1.12 branch +1.16.0 -After the 1.12.0 release there will be several 1.12.x bug-fix releases which -will contain bug fixes which have been deemed suitable for a stable branch, -but no new features or intrusive changes will be added to a bug-fix release -usually. The 1.12.x bug-fix releases will be made from the git 1.12 branch, which -is a stable branch. +1.16.0 is scheduled to be released in March 2019. -### 1.12.0 -1.12.0 was released on 4th May 2017. +Known Issues -## Known Issues +- possibly breaking/incompatible changes to properties of wrapped + FFmpeg decoders and encoders (see above). -- The `webrtcdsp` element is currently not shipped as part of the Windows - binary packages due to a [build system issue][bug-770264]. +- The way that GIO modules are named has changed due to upstream GLib + natively adding support for loading static GIO modules. This means + that any GStreamer application using gnutls for SSL/TLS on the + Android or iOS platforms (or any other setup using static libraries) + will fail to link looking for the g_io_module_gnutls_load_static() + function. The new function name is now + g_io_gnutls_load(gpointer data). See Android/iOS sections above for + further details. -[bug-770264]: https://bugzilla.gnome.org/show_bug.cgi?id=770264 -## Schedule for 1.14 +Schedule for 1.18 -Our next major feature release will be 1.14, and 1.11 will be the unstable -development version leading up to the stable 1.12 release. The development -of 1.13/1.14 will happen in the git master branch. +Our next major feature release will be 1.18, and 1.17 will be the +unstable development version leading up to the stable 1.18 release. The +development of 1.17/1.18 will happen in the git master branch. -The plan for the 1.14 development cycle is yet to be confirmed, but it is -expected that feature freeze will be around September 2017 -followed by several 1.13 pre-releases and the new 1.14 stable release -in October. +The plan for the 1.18 development cycle is yet to be confirmed, but it +is expected that feature freeze will be around July 2019 followed by +several 1.17 pre-releases and the new 1.18 stable release in +August/September. -1.14 will be backwards-compatible to the stable 1.12, 1.10, 1.8, 1.6, 1.4, -1.2 and 1.0 release series. +1.18 will be backwards-compatible to the stable 1.16, 1.14, 1.12, 1.10, +1.8, 1.6, 1.4, 1.2 and 1.0 release series. -- - - +------------------------------------------------------------------------ -*These release notes have been prepared by Sebastian Dröge, Tim-Philipp Müller -and Víctor Manuel Jáquez Leal.* +_These release notes have been prepared by Tim-Philipp Müller with_ +_contributions from Sebastian Dröge and Guillaume Desmottes._ -*License: [CC BY-SA 4.0](http://creativecommons.org/licenses/by-sa/4.0/)* +_License: CC BY-SA 4.0_