From: Sebastian Dröge Date: Tue, 1 Nov 2016 16:10:45 +0000 (+0200) Subject: Release 1.10.0 X-Git-Tag: 1.19.3~499^2~312 X-Git-Url: http://review.tizen.org/git/?a=commitdiff_plain;h=50ffd5a8057d40e573b4b15b62d3a3321e168d89;p=platform%2Fupstream%2Fgstreamer.git Release 1.10.0 --- diff --git a/ChangeLog b/ChangeLog index e3d2e5f..f4ae5aa 100644 --- a/ChangeLog +++ b/ChangeLog @@ -1,9 +1,80 @@ +=== release 1.10.0 === + +2016-11-01 Sebastian Dröge + + * configure.ac: + releasing 1.10.0 + +2016-10-28 12:55:34 +0100 Tim-Philipp Müller + + * gst-libs/ext/libav: + libav: Update to ffmpeg n3.2 + +2016-10-27 10:44:20 +0100 Tim-Philipp Müller + + * meson.build: + meson: fix version + +2016-10-26 23:29:18 +0300 Sebastian Dröge + + * ext/libav/gstavauddec.c: + * ext/libav/gstavaudenc.c: + avaudenc/dec: Allow compilation against ffmpeg < 3.2 again + +2016-10-26 23:17:28 +0300 Sebastian Dröge + + * ext/libav/gstavauddec.c: + * ext/libav/gstavaudenc.c: + avaudenc/dec: Ignore S64BE/LE pseudo-codecs + +2016-10-26 23:10:57 +0300 Sebastian Dröge + + * gst-libs/ext/libav: + libav: Update to ffmpeg 3.2 release branch + Release 3.2.0 is planned tomorrow and we should keep track of the latest + major version for 1.10 as we did in the past too. + +2016-10-24 10:30:05 +0300 Sebastian Dröge + + * configure.ac: + configure: Fix shell syntax error + Assignments must not have spaces around the '=' + +2016-10-22 12:48:40 +0300 Sebastian Dröge + + * gst-libs/ext/libav: + libav: Update to ffmpeg n3.1.5 + +2016-10-15 22:20:40 +0530 Nirbheek Chauhan + + * meson.build: + meson: Don't set c_std to gnu99 + Use the default for each compiler on every platform instead. This + improves our compatibility with compilers that don't have gnu99 as + a c_std. + +2016-10-06 14:25:17 +0300 Sebastian Dröge + + * gst-libs/ext/libav: + libav: Update to ffmpeg n3.1.4 + +2016-09-30 11:35:41 -0300 Thibault Saunier + + * hooks/pre-commit.hook: + * meson.build: + meson: Setup pre-commit hooks when configuring + === release 1.9.90 === -2016-09-30 Sebastian Dröge +2016-09-30 13:03:42 +0300 Sebastian Dröge + * ChangeLog: + * NEWS: + * RELEASE: * configure.ac: - releasing 1.9.90 + * docs/plugins/inspect/plugin-libav.xml: + * gst-libav.doap: + Release 1.9.90 2016-09-29 12:01:59 +0300 Sebastian Dröge diff --git a/NEWS b/NEWS index 072b2df..547de7f 100644 --- a/NEWS +++ b/NEWS @@ -1 +1,1114 @@ -This is GStreamer 1.9.90 +# GStreamer 1.10 Release Notes + +**GStreamer 1.10.0 was released on 1st November 2016.** + +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 new features, bug fixes and other +improvements. + +See [https://gstreamer.freedesktop.org/releases/1.10/][latest] for the latest +version of this document. + +*Last updated: Tuesday 1 Nov 2016, 15:00 UTC [(log)][gitlog]* + +[latest]: https://gstreamer.freedesktop.org/releases/1.10/ +[gitlog]: https://cgit.freedesktop.org/gstreamer/www/log/src/htdocs/releases/1.10/release-notes-1.10.md + +## 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 new features, bug fixes and other +improvements. + +## Highlights + +- Several convenience APIs have been added to make developers' lives easier +- A new `GstStream` API provides applications a more meaningful view of the + structure of streams, simplifying the process of dealing with media in + complex container formats +- Experimental `decodebin3` and `playbin3` elements which bring a number of + improvements which were hard to implement within `decodebin` and `playbin` +- A new `parsebin` element to automatically unpack and parse a stream, stopping + just short of decoding +- Experimental new `meson`-based build system, bringing faster build and much + better Windows support (including for building with Visual Studio) +- A new `gst-docs` module has been created, and we are in the process of moving + our documentation to a markdown-based format for easier maintenance and + updates +- A new `gst-examples` module has been create, which contains example + GStreamer applications and is expected to grow with many more examples in + the future +- Various OpenGL and OpenGL|ES-related fixes and improvements for greater + efficiency on desktop and mobile platforms, and Vulkan support on Wayland was + also added +- Extensive improvements to the VAAPI plugins for improved robustness and + efficiency +- Lots of fixes and improvements across the board, spanning RTP/RTSP, V4L2, + Bluetooth, audio conversion, echo cancellation, and more! + +## Major new features and changes + +### Noteworthy new API, features and other changes + +#### Core API additions + +##### Receive property change notifications via bus messages + +New API was added to receive element property change notifications via +bus messages. So far, applications had to connect a callback to an element's +`notify::property-name` signal via the GObject API, which was inconvenient for +at least two reasons: one had to implement a signal callback function, and that +callback function would usually be called from one of the streaming threads, so +one had to marshal (send) any information gathered or pending requests to the +main application thread which was tedious and error-prone. + +Enter [`gst_element_add_property_notify_watch()`][notify-watch] and +[`gst_element_add_property_deep_notify_watch()`][deep-notify-watch] which will +watch for changes of a property on the specified element, either only for this +element or recursively for a whole bin or pipeline. Whenever such a +property change happens, a `GST_MESSAGE_PROPERTY_NOTIFY` message will be posted +on the pipeline bus with details of the element, the property and the new +property value, all of which can be retrieved later from the message in the +application via [`gst_message_parse_property_notify()`][parse-notify]. Unlike +the GstBus watch functions, this API does not rely on a running GLib main loop. + +The above can be used to be notified asynchronously of caps changes in the +pipeline, or volume changes on an audio sink element, for example. + +[notify-watch]: https://gstreamer.freedesktop.org/data/doc/gstreamer/head/gstreamer/html/GstElement.html#gst-element-add-property-notify-watch +[deep-notify-watch]: https://gstreamer.freedesktop.org/data/doc/gstreamer/head/gstreamer/html/GstElement.html#gst-element-add-property-deep-notify-watch +[parse-notify]: https://gstreamer.freedesktop.org/data/doc/gstreamer/head/gstreamer/html/GstMessage.html#gst-message-parse-property-notify + +##### GstBin "deep" element-added and element-removed signals + +GstBin has gained `"deep-element-added"` and `"deep-element-removed"` signals +which makes it easier for applications and higher-level plugins to track when +elements are added or removed from a complex pipeline with multiple sub-bins. + +`playbin` makes use of this to implement the new `"element-setup"` signal which +can be used to configure elements as they are added to `playbin`, just like the +existing `"source-setup"` signal which can be used to configure the source +element created. + +##### Error messages can contain additional structured details + +It is often useful to provide additional, structured information in error, +warning or info messages for applications (or higher-level elements) to make +intelligent decisions based on them. To allow this, error, warning and info +messages now have API for adding arbitrary additional information to them +using a `GstStructure`: +[`GST_ELEMENT_ERROR_WITH_DETAILS`][element-error-with-details] and +corresponding API for the other message types. + +This is now used e.g. by the new [`GST_ELEMENT_FLOW_ERROR`][element-flow-error] +API to include the actual flow error in the error message, and the +[souphttpsrc element][souphttpsrc-detailed-errors] to provide the HTTP +status code, and the URL (if any) to which a redirection has happened. + +[element-error-with-details]: https://gstreamer.freedesktop.org/data/doc/gstreamer/head/gstreamer/html/GstElement.html#GST-ELEMENT-ERROR-WITH-DETAILS:CAPS +[element-flow-error]: https://gstreamer.freedesktop.org/data/doc/gstreamer/head/gstreamer/html/GstElement.html#GST-ELEMENT-FLOW-ERROR:CAPS +[souphttpsrc-detailed-errors]: https://cgit.freedesktop.org/gstreamer/gst-plugins-good/tree/ext/soup/gstsouphttpsrc.c?id=60d30db912a1aedd743e66b9dcd2e21d71fbb24f#n1318 + +##### Redirect messages have official API now + +Sometimes, elements need to redirect the current stream URL and tell the +application to proceed with this new URL, possibly using a different +protocol too (thus changing the pipeline configuration). Until now, this was +informally implemented using `ELEMENT` messages on the bus. + +Now this has been formalized in the form of a new `GST_MESSAGE_REDIRECT` message. +A new redirect message can be created using [`gst_message_new_redirect()`][new-redirect]. +If needed, multiple redirect locations can be specified by calling +[`gst_message_add_redirect_entry()`][add-redirect] to add further redirect +entries, all with metadata, so the application can decide which is +most suitable (e.g. depending on the bitrate tags). + +[new-redirect]: https://gstreamer.freedesktop.org/data/doc/gstreamer/head/gstreamer/html/GstMessage.html#gst-message-new-redirect +[add-redirect]: https://gstreamer.freedesktop.org/data/doc/gstreamer/head/gstreamer/html/GstMessage.html#gst-message-add-redirect-entry + +##### New pad linking convenience functions that automatically create ghost pads + +New pad linking convenience functions were added: +[`gst_pad_link_maybe_ghosting()`][pad-maybe-ghost] and +[`gst_pad_link_maybe_ghosting_full()`][pad-maybe-ghost-full] which were +previously internal to GStreamer have now been exposed for general use. + +The existing pad link functions will refuse to link pads or elements at +different levels in the pipeline hierarchy, requiring the developer to +create ghost pads where necessary. These new utility functions will +automatically create ghostpads as needed when linking pads at different +levels of the hierarchy (e.g. from an element inside a bin to one that's at +the same level in the hierarchy as the bin, or in another bin). + +[pad-maybe-ghost]: https://gstreamer.freedesktop.org/data/doc/gstreamer/head/gstreamer/html/GstPad.html#gst-pad-link-maybe-ghosting +[pad-maybe-ghost-full]: https://gstreamer.freedesktop.org/data/doc/gstreamer/head/gstreamer/html/GstPad.html#gst-pad-link-maybe-ghosting-full + +##### Miscellaneous + +Pad probes: IDLE and BLOCK probes now work slightly differently in pull mode, +so that push and pull mode have opposite scenarios for idle and blocking probes. +In push mode, it will block with some data type and IDLE won't have any data. +In pull mode, it will block _before_ getting a buffer and will be IDLE once some +data has been obtained. ([commit][commit-pad-probes], [bug][bug-pad-probes]) + +[commit-pad-probes]: https://cgit.freedesktop.org/gstreamer/gstreamer/commit/gst/gstpad.c?id=368ee8a336d0c868d81fdace54b24431a8b48cbf +[bug-pad-probes]: https://bugzilla.gnome.org/show_bug.cgi?id=761211 + +[`gst_parse_launch_full()`][parse-launch-full] can now be made to return a +`GstBin` instead of a top-level pipeline by passing the new +`GST_PARSE_FLAG_PLACE_IN_BIN` flag. + +[parse-launch-full]: https://gstreamer.freedesktop.org/data/doc/gstreamer/head/gstreamer/html/gstreamer-GstParse.html#gst-parse-launch-full + +The default GStreamer debug log handler can now be removed before +calling `gst_init()`, so that it will never get installed and won't be active +during initialization. + +A new [`STREAM_GROUP_DONE` event][stream-group-done-event] was added. In some +ways it works similar to the `EOS` event in that it can be used to unblock +downstream elements which may be waiting for further data, such as for example +`input-selector`. Unlike `EOS`, further data flow may happen after the +`STREAM_GROUP_DONE` event though (and without the need to flush the pipeline). +This is used to unblock input-selector when switching between streams in +adaptive streaming scenarios (e.g. HLS). + +[stream-group-done-event]: https://gstreamer.freedesktop.org/data/doc/gstreamer/head/gstreamer/html/GstEvent.html#gst-event-new-stream-group-done + +The `gst-launch-1.0` command line tool will now print unescaped caps in verbose +mode (enabled by the -v switch). + +[`gst_element_call_async()`][call-async] has been added as convenience API for +plugin developers. It is useful for one-shot operations that need to be done +from a thread other than the current streaming thread. It is backed by a +thread-pool that is shared by all elements. + +[call-async]: https://gstreamer.freedesktop.org/data/doc/gstreamer/head/gstreamer/html/GstElement.html#gst-element-call-async + +Various race conditions have been fixed around the `GstPoll` API used by e.g. +`GstBus` and `GstBufferPool`. Some of these manifested themselves primarily +on Windows. + +`GstAdapter` can now keep track of discontinuities signalled via the `DISCONT` +buffer flag, and has gained [new API][new-adapter-api] to track PTS, DTS and +offset at the last discont. This is useful for plugins implementing advanced +trick mode scenarios. + +[new-adapter-api]: https://gstreamer.freedesktop.org/data/doc/gstreamer/head/gstreamer-libs/html/GstAdapter.html#gst-adapter-pts-at-discont + +`GstTestClock` gained a new [`"clock-type"` property][clock-type-prop]. + +[clock-type-prop]: https://gstreamer.freedesktop.org/data/doc/gstreamer/head/gstreamer-libs/html/GstTestClock.html#GstTestClock--clock-type + +#### GstStream API for stream announcement and stream selection + +New stream listing and stream selection API: new API has been added to +provide high-level abstractions for streams ([`GstStream`][stream-api]) +and collections of streams ([`GstStreamCollections`][stream-collection-api]). + +##### Stream listing + +A [`GstStream`][stream-api] contains all the information pertinent to a stream, +such as stream id, caps, tags, flags and stream type(s); it can represent a +single elementary stream (e.g. audio, video, subtitles, etc.) or a container +stream. This will depend on the context. In a decodebin3/playbin3 one +it will typically be elementary streams that can be selected and unselected. + +A [`GstStreamCollection`][stream-collection-api] represents a group of streams +and is used to announce or publish all available streams. A GstStreamCollection +is immutable - once created it won't change. If the available streams change, +e.g. because a new stream appeared or some streams disappeared, a new stream +collection will be published. This new stream collection may contain streams +from the previous collection if those streams persist, or completely new ones. +Stream collections do not yet list all theoretically available streams, +e.g. other available DVD angles or alternative resolutions/bitrate of the same +stream in case of adaptive streaming. + +New events and messages have been added to notify or update other elements and +the application about which streams are currently available and/or selected. +This way, we can easily and seamlessly let the application know whenever the +available streams change, as happens frequently with digital television streams +for example. The new system is also more flexible. For example, it is now also +possible for the application to select multiple streams of the same type +(e.g. in a transcoding/transmuxing scenario). + +A [`STREAM_COLLECTION` message][stream-collection-msg] is posted on the bus +to inform the parent bin (e.g. `playbin3`, `decodebin3`) and/or the application +about what streams are available, so you no longer have to hunt for this +information at different places. The available information includes number of +streams of each type, caps, tags etc. Bins and/or the application can intercept +the message synchronously to select and deselect streams before any data is +produced - for the case where elements such as the demuxers support the new +stream API, not necessarily in the parsebin compatibility fallback case. + +Similarly, there is also a [`STREAM_COLLECTION` event][stream-collection-event] +to inform downstream elements of the available streams. This event can be used +by elements to aggregate streams from multiple inputs into one single collection. + +The `STREAM_START` event was extended so that it can also contain a GstStream +object with all information about the current stream, see +[`gst_event_set_stream()`][event-set-stream] and +[`gst_event_parse_stream()`][event-parse-stream]. +[`gst_pad_get_stream()`][pad-get-stream] is a new utility function that can be +used to look up the GstStream from the `STREAM_START` sticky event on a pad. + +[stream-api]: https://gstreamer.freedesktop.org/data/doc/gstreamer/head/gstreamer/html/gstreamer-GstStream.html +[stream-collection-api]: https://gstreamer.freedesktop.org/data/doc/gstreamer/head/gstreamer/html/gstreamer-GstStreamCollection.html +[stream-collection-msg]: https://gstreamer.freedesktop.org/data/doc/gstreamer/head/gstreamer/html/GstMessage.html#gst-message-new-stream-collection +[stream-collection-event]: https://gstreamer.freedesktop.org/data/doc/gstreamer/head/gstreamer/html/GstEvent.html#gst-event-new-stream-collection +[event-set-stream]: https://gstreamer.freedesktop.org/data/doc/gstreamer/head/gstreamer/html/GstEvent.html#gst-event-set-stream +[event-parse-stream]: https://gstreamer.freedesktop.org/data/doc/gstreamer/head/gstreamer/html/GstEvent.html#gst-event-parse-stream +[pad-get-stream]: https://gstreamer.freedesktop.org/data/doc/gstreamer/head/gstreamer/html/GstPad.html#gst-pad-get-stream + +##### Stream selection + +Once the available streams have been published, streams can be selected via +their stream ID using the new `SELECT_STREAMS` event, which can be created +with [`gst_event_new_select_streams()`][event-select-streams]. The new API +supports selecting multiple streams per stream type. In the future, we may also +implement explicit deselection of streams that will never be used, so +elements can skip these and never expose them or output data for them in the +first place. + +The application is then notified of the currently selected streams via the +new `STREAMS_SELECTED` message on the pipeline bus, containing both the current +stream collection as well as the selected streams. This might be posted in +response to the application sending a `SELECT_STREAMS` event or when +`decodebin3` or `playbin3` decide on the streams to be initially selected without +application input. + +[event-select-streams]: https://gstreamer.freedesktop.org/data/doc/gstreamer/head/gstreamer/html/GstEvent.html#gst-event-new-select-streams + +##### Further reading + +See further below for some notes on the new elements supporting this new +stream API, namely: `decodebin3`, `playbin3` and `parsebin`. + +More information about the new API and the new elements can also be found here: + +- GStreamer [stream selection design docs][streams-design] +- Edward Hervey's talk ["The new streams API: Design and usage"][streams-talk] ([slides][streams-slides]) +- Edward Hervey's talk ["Decodebin3: Dealing with modern playback use cases"][db3-talk] ([slides][db3-slides]) + +[streams-design]: https://cgit.freedesktop.org/gstreamer/gstreamer/tree/docs/design/part-stream-selection.txt +[streams-talk]: https://gstconf.ubicast.tv/videos/the-new-gststream-api-design-and-usage/ +[streams-slides]: https://gstreamer.freedesktop.org/data/events/gstreamer-conference/2016/Edward%20Hervey%20-%20The%20New%20Streams%20API%20Design%20and%20Usage.pdf +[db3-talk]: https://gstconf.ubicast.tv/videos/decodebin3-or-dealing-with-modern-playback-use-cases/ +[db3-slides]: https://gstreamer.freedesktop.org/data/events/gstreamer-conference/2015/Edward%20Hervey%20-%20decodebin3.pdf + +#### Audio conversion and resampling API + +The audio conversion library received a completely new and rewritten audio +resampler, complementing the audio conversion routines moved into the audio +library in the [previous release][release-notes-1.8]. Integrating the resampler +with the other audio conversion library allows us to implement generic +conversion much more efficiently, as format conversion and resampling can now +be done in the same processing loop instead of having to do it in separate +steps (our element implementations do not make use of this yet though). + +The new audio resampler library is a combination of some of the best features +of other samplers such as ffmpeg, speex and SRC. It natively supports S16, S32, +F32 and F64 formats and uses optimized x86 and neon assembly for most of its +processing. It also has support for dynamically changing sample rates by incrementally +updating the filter tables using linear or cubic interpolation. According to +some benchmarks, it's one of the fastest and most accurate resamplers around. + +The `audioresample` plugin has been ported to the new audio library functions +to make use of the new resampler. + +[release-notes-1.8]: https://gstreamer.freedesktop.org/releases/1.8/ + +#### Support for SMPTE timecodes + +Support for SMPTE timecodes was added to the GStreamer video library. This +comes with an abstraction for timecodes, [`GstVideoTimeCode`][video-timecode] +and a [`GstMeta`][video-timecode-meta] that can be placed on video buffers for +carrying the timecode information for each frame. Additionally there is +various API for making handling of timecodes easy and to do various +calculations with them. + +A new plugin called [`timecode`][timecode-plugin] was added, that contains an +element called `timecodestamper` for putting the timecode meta on video frames +based on counting the frames and another element called `timecodewait` that +drops all video (and audio) until a specific timecode is reached. + +Additionally support was added to the Decklink plugin for including the +timecode information when sending video out or capturing it via SDI, the +`qtmux` element is able to write timecode information into the MOV container, +and the `timeoverlay` element can overlay timecodes on top of the video. + +More information can be found in the [talk about timecodes][timecode-talk] at +the GStreamer Conference 2016. + +[video-timecode]: https://gstreamer.freedesktop.org/data/doc/gstreamer/head/gst-plugins-base-libs/html/gst-plugins-base-libs-gstvideo.html#GstVideoTimeCode +[video-timecode-meta]: https://gstreamer.freedesktop.org/data/doc/gstreamer/head/gst-plugins-base-libs/html/gst-plugins-base-libs-gstvideometa.html#gst-buffer-add-video-time-code-meta +[timecode-plugin]: https://cgit.freedesktop.org/gstreamer/gst-plugins-bad/tree/gst/timecode +[timecode-talk]: https://gstconf.ubicast.tv/videos/smpte-timecodes-in-gstreamer/ + +#### GStreamer OpenMAX IL plugin + +The last gst-omx release, 1.2.0, was in July 2014. It was about time to get +a new one out with all the improvements that have happened in the meantime. +From now on, we will try to release gst-omx together with all other modules. + +This release features a lot of bugfixes, improved support for the Raspberry Pi +and in general improved support for zerocopy rendering via EGL and a few minor +new features. + +At this point, gst-omx is known to work best on the Raspberry Pi platform but +it is also known to work on various other platforms. Unfortunately, we are +not including configurations for any other platforms, so if you happen to use +gst-omx: please send us patches with your configuration and code changes! + +### New Elements + +#### decodebin3, playbin3, parsebin (experimental) + +This release features new decoding and playback elements as experimental +technology previews: `decodebin3` and `playbin3` will soon supersede the +existing `decodebin` and `playbin` elements. We skipped the number 2 because +it was already used back in the 0.10 days, which might cause confusion. +Experimental technology preview means that everything should work fine already, +but we can't guarantee there won't be minor behavioural changes in the +next cycle. In any case, please test and report any problems back. + +Before we go into detail about what these new elements improve, let's look at +the new [`parsebin`][parsebin] element. It works similarly to `decodebin` and +`decodebin3`, only that it stops one step short and does not plug any actual +decoder elements. It will only plug parsers, tag readers, demuxers and +depayloaders. Also note that parsebin does not contain any queueing element. + +[`decodebin3`'s][decodebin3] internal architecture is slightly different from +the existing `decodebin` element and fixes many long-standing issues with our +decoding engine. For one, data is now fed into the internal `multiqueue` element +*after* it has been parsed and timestamped, which means that the `multiqueue` +element now has more knowledge and is able to calculate the interleaving of the +various streams, thus minimizing memory requirements and doing away with magic +values for buffering limits that were conceived when videos were 240p or 360p. +Anyone who has tried to play back 4k video streams with decodebin2 +will have noticed the limitations of that approach. The improved timestamp +tracking also enables `multiqueue` to keep streams of the same type (audio, +video) aligned better, making sure switching between streams of the same type +is very fast. + +Another major improvement in `decodebin3` is that it will no longer decode +streams that are not being used. With the old `decodebin` and `playbin`, when +there were 8 audio streams we would always decode all 8 streams even +if 7 were not actually used. This caused a lot of CPU overhead, which was +particularly problematic on embedded devices. When switching between streams +`decodebin3` will try hard to re-use existing decoders. This is useful when +switching between multiple streams of the same type if they are encoded in the +same format. + +Re-using decoders is also useful when the available streams change on the fly, +as might happen with radio streams (chained Oggs), digital television +broadcasts, when adaptive streaming streams change bitrate, or when switching +gaplessly to the next title. In order to guarantee a seamless transition, the +old `decodebin2` would plug a second decoder for the new stream while finishing +up the old stream. With `decodebin3`, this is no longer needed - at least not +when the new and old format are the same. This will be particularly useful +on embedded systems where it is often not possible to run multiple decoders +at the same time, or when tearing down and setting up decoders is fairly +expensive. + +`decodebin3` also allows for multiple input streams, not just a single one. +This will be useful, in the future, for gapless playback, or for feeding +multiple external subtitle streams to decodebin/playbin. + +`playbin3` uses `decodebin3` internally, and will supercede `playbin`. +It was decided that it would be too risky to make the old `playbin` use the +new `decodebin3` in a backwards-compatible way. The new architecture +makes it awkward, if not impossible, to maintain perfect backwards compatibility +in some aspects, hence `playbin3` was born, and developers can migrate to the +new element and new API at their own pace. + +All of these new elements make use of the new `GstStream` API for listing and +selecting streams, as described above. `parsebin` provides backwards +compatibility for demuxers and parsers which do not advertise their streams +using the new API yet (which is most). + +The new elements are not entirely feature-complete yet: `playbin3` does not +support so-called decodersinks yet where the data is not decoded inside +GStreamer but passed directly for decoding to the sink. `decodebin3` is missing +the various `autoplug-*` signals to influence which decoders get autoplugged +in which order. We're looking to add back this functionality, but it will probably +be in a different way, with a single unified signal and using GstStream perhaps. + +For more information on these new elements, check out Edward Hervey's talk +[*decodebin3 - dealing with modern playback use cases*][db3-talk] + +[parsebin]: https://gstreamer.freedesktop.org/data/doc/gstreamer/head/gst-plugins-base-plugins/html/gst-plugins-base-plugins-parsebin.html +[decodebin3]: https://gstreamer.freedesktop.org/data/doc/gstreamer/head/gst-plugins-base-plugins/html/gst-plugins-base-plugins-decodebin3.html +[db3-talk]: https://gstconf.ubicast.tv/videos/decodebin3-or-dealing-with-modern-playback-use-cases/ + +#### LV2 ported from 0.10 and switched from slv2 to lilv2 + +The LV2 wrapper plugin has been ported to 1.0 and moved from using the +deprecated slv2 library to its replacement liblv2. We support sources and +filter elements. lv2 is short for *Linux Audio Developer's Simple Plugin API +(LADSPA) version 2* and is an open standard for audio plugins which includes +support for audio synthesis (generation), digital signal processing of digital +audio, and MIDI. The new lv2 plugin supersedes the existing LADSPA plugin. + +#### WebRTC DSP Plugin for echo-cancellation, gain control and noise suppression + +A set of new elements ([webrtcdsp][webrtcdsp], [webrtcechoprobe][webrtcechoprobe]) +based on the WebRTC DSP software stack can now be used to improve your audio +voice communication pipelines. They support echo cancellation, gain control, +noise suppression and more. For more details you may read +[Nicolas' blog post][webrtc-blog-post]. + +[webrtcdsp]: https://gstreamer.freedesktop.org/data/doc/gstreamer/head/gst-plugins-bad-plugins/html/gst-plugins-bad-plugins-webrtcdsp.html +[webrtcechoprobe]: https://gstreamer.freedesktop.org/data/doc/gstreamer/head/gst-plugins-bad-plugins/html/gst-plugins-bad-plugins-webrtcechoprobe.html +[webrtc-blog-post]: https://ndufresne.ca/2016/06/gstreamer-echo-canceller/ + +#### Fraunhofer FDK AAC encoder and decoder + +New encoder and decoder elements wrapping the Fraunhofer FDK AAC library have +been added (`fdkaacdec`, `fdkaacdec`). The Fraunhofer FDK AAC encoder is +generally considered to be a very high-quality AAC encoder, but unfortunately +it comes under a non-free license with the option to obtain a paid, commercial +license. + +### Noteworthy element features and additions + +#### Major RTP and RTSP improvements + +- The RTSP server and source element, as well as the RTP jitterbuffer now support + remote clock synchronization according to [RFC7273][https://tools.ietf.org/html/rfc7273]. +- Support for application and profile specific RTCP packets was added. +- The H265/HEVC payloader/depayloader is again in sync with the final RFC. +- Seeking stability of the RTSP source and server was improved a lot and + runs stably now, even when doing scrub-seeking. +- The RTSP server received various major bugfixes, including for regressions that + caused the IP/port address pool to not be considered, or NAT hole punching + to not work anymore. [Bugzilla #766612][https://bugzilla.gnome.org/show_bug.cgi?id=766612] +- Various other bugfixes that improve the stability of RTP and RTSP, including + many new unit / integration tests. + +#### Improvements to splitmuxsrc and splitmuxsink + +- The splitmux element received reliability and error handling improvements, + removing at least one deadlock case. `splitmuxsrc` now stops cleanly at the end + of the segment when handling seeks with a stop time. We fixed a bug with large + amounts of downstream buffering causing incorrect out-of-sequence playback. + +- `splitmuxsrc` now has a `"format-location"` signal to directly specify the list + of files to play from. + +- `splitmuxsink` can now optionally send force-keyunit events to upstream + elements to allow splitting files more accurately instead of having to wait + for upstream to provide a new keyframe by itself. + +#### OpenGL/GLES improvements + +##### iOS and macOS (OS/X) + +- We now create OpenGL|ES 3.x contexts on iOS by default with a fallback to + OpenGL|ES 2.x if that fails. +- Various zerocopy decoding fixes and enhancements with the + encoding/decoding/capturing elements. +- libdispatch is now used on all Apple platforms instead of GMainLoop, removing + the expensive poll()/pthread_*() overhead. + +##### New API + +- `GstGLFramebuffer` - for wrapping OpenGL frame buffer objects. It provides + facilities for attaching `GstGLMemory` objects to the necessary attachment + points, binding and unbinding and running a user-supplied function with the + framebuffer bound. +- `GstGLRenderbuffer` (a `GstGLBaseMemory` subclass) - for wrapping OpenGL + render buffer objects that are typically used for depth/stencil buffers or + for color buffers where we don't care about the output. +- `GstGLMemoryEGL` (a `GstGLMemory` subclass) - for combining `EGLImage`s with a GL + texture that replaces `GstEGLImageMemory` bringing the improvements made to the + other `GstGLMemory` implementations. This fixes a performance regression in + zerocopy decoding on the Raspberry Pi when used with an updated gst-omx. + +##### Miscellaneous improvements + +- `gltestsrc` is now usable on devices/platforms with OpenGL 3.x and OpenGL|ES + and has completed or gained support for new patterns in line with the + existing ones in `videotestsrc`. +- `gldeinterlace` is now available on devices/platforms with OpenGL|ES + implementations. +- The dispmanx backend (used on the Raspberry Pi) now supports the + `gst_video_overlay_set_window_handle()` and + `gst_video_overlay_set_render_rectangle()` functions. +- The `gltransformation` element now correctly transforms mouse coordinates (in + window space) to stream coordinates for both perspective and orthographic + projections. +- The `gltransformation` element now detects if the + `GstVideoAffineTransformationMeta` is supported downstream and will efficiently + pass its transformation downstream. This is a performance improvement as it + results in less processing being required. +- The wayland implementation now uses the multi-threaded safe event-loop API + allowing correct usage in applications that call wayland functions from + multiple threads. +- Support for native 90 degree rotations and horizontal/vertical flips + in `glimagesink`. + +#### Vulkan + +- The Vulkan elements now work under Wayland and have received numerous + bugfixes. + +#### QML elements + +- `qmlglsink` video sink now works on more platforms, notably, Windows, Wayland, + and Qt's eglfs (for embedded devices with an OpenGL implementation) including + the Raspberry Pi. +- New element `qmlglsrc` to record a QML scene into a GStreamer pipeline. + +#### KMS video sink + +- New element `kmssink` to render video using Direct Rendering Manager + (DRM) and Kernel Mode Setting (KMS) subsystems in the Linux + kernel. It is oriented to be used mostly in embedded systems. + +#### Wayland video sink + +- `waylandsink` now supports the wl_viewporter extension allowing + video scaling and cropping to be delegated to the Wayland + compositor. This extension is also been made optional, so that it can + also work on current compositors that don't support it. It also now has + support for the video meta, allowing zero-copy operations in more + cases. + +#### DVB improvements + +- `dvbsrc` now has better delivery-system autodetection and several + new parameter sanity-checks to improve its resilience to configuration + omissions and errors. Superfluous polling continues to be trimmed down, + and the debugging output has been made more consistent and precise. + Additionally, the channel-configuration parser now supports the new dvbv5 + format, enabling `dvbbasebin` to automatically playback content transmitted + on delivery systems that previously required manual description, like ISDB-T. + +#### DASH, HLS and adaptivedemux + +- HLS now has support for Alternate Rendition audio and video tracks. Full + support for Alternate Rendition subtitle tracks will be in an upcoming release. +- DASH received support for keyframe-only trick modes if the + `GST_SEEK_FLAG_TRICKMODE_KEY_UNITS` flag is given when seeking. It will + only download keyframes then, which should help with high-speed playback. + Changes to skip over multiple frames based on bandwidth and other metrics + will be added in the near future. +- Lots of reliability fixes around seek handling and bitrate switching. + +#### Bluetooth improvements + +- The `avdtpsrc` element now supports metadata such as track title, artist + name, and more, which devices can send via AVRCP. These are published as + tags on the pipeline. +- The `a2dpsink` element received some love and was cleaned up so that it + actually works after the initial GStreamer 1.0 port. + +#### GStreamer VAAPI + +- All the decoders have been split, one plugin feature per codec. So + far, the available ones, depending on the driver, are: + `vaapimpeg2dec`, `vaapih264dec`, `vaapih265dec`, `vaapivc1dec`, `vaapivp8dec`, + `vaapivp9dec` and `vaapijpegdec` (which already was split). +- Improvements when mapping VA surfaces into memory. It now differentiates + between negotiation caps and allocations caps, since the allocation + memory for surfaces may be bigger than one that is going to be + mapped. +- `vaapih265enc` now supports constant bitrate mode (CBR). +- Since several VA drivers are unmaintained, we decide to keep a whitelist + with the va drivers we actually test, which is mostly the i915 and to a lesser + degree gallium from the mesa project. Exporting the environment variable + `GST_VAAPI_ALL_DRIVERS` disables the whitelist. +- Plugin features are registered at run-time, according to their support by + the loaded VA driver. So only the decoders and encoder supported by the + system are registered. Since the driver can change, some dependencies are + tracked to invalidate the GStreamer registry and reload the plugin. +- `dmabuf` importation from upstream has been improved, gaining performance. +- `vaapipostproc` now can negotiate buffer transformations via caps. +- Decoders now can do I-frame only reverse playback. This decodes I-frames + only because the surface pool is smaller than the required by the GOP to show all the + frames. +- The upload of frames onto native GL textures has been optimized too, keeping + a cache of the internal structures for the offered textures by the sink. + +#### V4L2 changes + +- More pixels formats are now supported +- Decoder is now using `G_SELECTION` instead of the deprecated `G_CROP` +- Decoder now uses the `STOP` command to handle EOS +- Transform element can now scale the pixel aspect ratio +- Colorimetry support has been improved even more +- We now support the `OUTPUT_OVERLAY` type of video node in v4l2sink + +#### Miscellaneous + +- `multiqueue`'s input pads gained a new `"group-id"` property which + can be used to group input streams. Typically one will assign + different id numbers to audio, video and subtitle streams for + example. This way `multiqueue` can make sure streams of the same + type advance in lockstep if some of the streams are unlinked and the + `"sync-by-running-time"` property is set. This is used in + decodebin3/playbin3 to implement almost-instantaneous stream + switching. The grouping is required because different downstream + paths (audio, video, etc.) may have different buffering/latency + etc. so might be consuming data from multiqueue with a slightly + different phase, and if we track different stream groups separately + we minimize stream switching delays and buffering inside the + `multiqueue`. +- `alsasrc` now supports ALSA drivers without a position for each + channel, this is common in some professional or industrial hardware. +- `libvpx` based decoders (`vp8dec` and `vp9dec`) now create multiple threads on + computers with multiple CPUs automatically. +- `rfbsrc` - used for capturing from a VNC server - has seen a lot of + debugging. It now supports the latest version of the RFB + protocol and uses GIO everywhere. +- `tsdemux` can now read ATSC E-AC-3 streams. +- New `GstVideoDirection` video orientation interface for rotating, flipping + and mirroring video in 90° steps. It is implemented by the `videoflip` and + `glvideoflip` elements currently. +- It is now possible to give `appsrc` a duration in time, and there is now a + non-blocking try-pull API for `appsink` that returns NULL if nothing is + available right now. +- `x264enc` has support now for chroma-site and colorimetry settings +- A new JPEG2000 parser element was added, and the JPEG2000 caps were cleaned + up and gained more information needed in combination with RTP and various + container formats. +- Reverse playback support for `videorate` and `deinterlace` was implemented +- Various improvements everywhere for reverse playback and `KEY_UNITS` trick mode +- New cleaned up `rawaudioparse` and `rawvideoparse` elements that replace the + old `audioparse` and `videoparse` elements. There are compatibility element + factories registered with the old names to allow existing code to continue + to work. +- The Decklink plugin gained support for 10 bit video SMPTE timecodes, and + generally got many bugfixes for various issues. +- New API in `GstPlayer` for setting the multiview mode for stereoscopic + video, setting an HTTP/RTSP user agent and a time offset between audio and + video. In addition to that, there were various bugfixes and the new + gst-examples module contains Android, iOS, GTK+ and Qt example applications. +- `GstBin` has new API for suppressing various `GstElement` or `GstObject` + flags that would otherwise be affected by added/removed child elements. This + new API allows `GstBin` subclasses to handle for themselves if they + should be considered a sink or source element, for example. +- The `subparse` element can handle WebVTT streams now. +- A new `sdpsrc` element was added that can read an SDP from a file, or get it + as a string as property and then sets up an RTP pipeline accordingly. + +### Plugin moves + +No plugins were moved this cycle. We'll make up for it next cycle, promise! + +### Rewritten memory leak tracer + +GStreamer has had basic functionality to trace allocation and freeing of +both mini-objects (buffers, events, caps, etc.) and objects in the form of the +internal `GstAllocTrace` tracing system. This API was never exposed in the +1.x API series though. When requested, this would dump a list of objects and +mini-objects at exit time which had still not been freed at that point, +enabled with an environment variable. This subsystem has now been removed +in favour of a new implementation based on the recently-added tracing framework. + +Tracing hooks have been added to trace the creation and destruction of +GstObjects and mini-objects, and a new tracer plugin has been written using +those new hooks to track which objects are still live and which are not. If +GStreamer has been compiled against the libunwind library, the new leaks tracer +will remember where objects were allocated from as well. By default the leaks +tracer will simply output a warning if leaks have been detected on `gst_deinit()`. + +If the `GST_LEAKS_TRACER_SIG` environment variable is set, the leaks tracer +will also handle the following UNIX signals: + + - `SIGUSR1`: log alive objects + - `SIGUSR2`: create a checkpoint and print a list of objects created and + destroyed since the previous checkpoint. + +Unfortunately this will not work on Windows due to no signals, however. + +If the `GST_LEAKS_TRACER_STACK_TRACE` environment variable is set, the leaks +tracer will also log the creation stack trace of leaked objects. This may +significantly increase memory consumption however. + +New `MAY_BE_LEAKED` flags have been added to GstObject and GstMiniObject, so +that objects and mini-objects that are likely to stay around forever can be +flagged and blacklisted from the leak output. + +To give the new leak tracer a spin, simply call any GStreamer application such +as `gst-launch-1.0` or `gst-play-1.0` like this: + + GST_TRACERS=leaks gst-launch-1.0 videotestsrc num-buffers=10 ! fakesink + +If there are any leaks, a warning will be raised at the end. + +It is also possible to trace only certain types of objects or mini-objects: + + GST_TRACERS="leaks(GstEvent,GstMessage)" gst-launch-1.0 videotestsrc num-buffers=10 ! fakesink + +This dedicated leaks tracer is much much faster than valgrind since all code is +executed natively instead of being instrumented. This makes it very suitable +for use on slow machines or embedded devices. It is however limited to certain +types of leaks and won't catch memory leaks when the allocation has been made +via plain old `malloc()` or `g_malloc()` or other means. It will also not trace +non-GstObject GObjects. + +The goal is to enable leak tracing on GStreamer's Continuous-Integration and +testing system, both for the regular unit tests (make check) and media tests +(gst-validate), so that accidental leaks in common code paths can be detected +and fixed quickly. + +For more information about the new tracer, check out Guillaume Desmottes's +["Tracking Memory Leaks"][leaks-talk] talk or his [blog post][leaks-blog] about +the topic. + +[leaks-talk]: https://gstconf.ubicast.tv/videos/tracking-memory-leaks/ +[leaks-blog]: https://blog.desmottes.be/?post/2016/06/20/GStreamer-leaks-tracer + +### GES and NLE changes + +- Clip priorities are now handled by the layers, and the GESTimelineElement + priority property is now deprecated and unused +- Enhanced (de)interlacing support to always use the `deinterlace` element + and expose needed properties to users +- Allow reusing clips children after removing the clip from a layer +- We are now testing many more rendering formats in the gst-validate + test suite, and failures have been fixed. +- Also many bugs have been fixed in this cycle! + +### GStreamer validate changes + +This cycle has been focused on making GstValidate more than just a validating +tool, but also a tool to help developers debug their GStreamer issues. When +reporting issues, we try to gather as much information as possible and expose +it to end users in a useful way. For an example of such enhancements, check out +Thibault Saunier's [blog post](improving-debugging-gstreamer-validate) about +the new Not Negotiated Error reporting mechanism. + +Playbin3 support has been added so we can run validate tests with `playbin3` +instead of playbin. + +We are now able to properly communicate between `gst-validate-launcher` and +launched subprocesses with actual IPC between them. That has enabled the test +launcher to handle failing tests specifying the exact expected issue(s). + +[improving-debugging-gstreamer-validate]: https://blogs.s-osg.org/improving-debugging-gstreamer-validate/ + +### gst-libav changes + +gst-libav uses the recently released ffmpeg 3.2 now, which brings a lot of +improvements and bugfixes from the ffmpeg team in addition to various new +codec mappings on the GStreamer side and quite a few bugfixes to the GStreamer +integration to make it more robust. + +## Build and Dependencies + +### Experimental support for Meson as build system + +#### Overview + +We have have added support for building GStreamer using the +[Meson build system][meson]. This is currently experimental, but should work +fine at least on Linux using the gcc or clang toolchains and on Windows using +the MingW or MSVC toolchains. + +Autotools remains the primary build system for the time being, but we hope to +someday replace it and will steadily work towards that goal. + +More information about the background and implications of all this and where +we're hoping to go in future with this can be found in [Tim's mail][meson-mail] +to the gstreamer-devel mailing list. + +For more information on Meson check out [these videos][meson-videos] and also +the [Meson talk][meson-gstconf] at the GStreamer Conference. + +Immediate benefits for Linux users are faster builds and rebuilds. At the time +of writing the Meson build of GStreamer is used by default in GNOME's jhbuild +system. + +The Meson build currently still lacks many of the fine-grained configuration +options to enable/disable specific plugins. These will be added back in due +course. + +Note: The meson build files are not distributed in the source tarballs, you will +need to get GStreamer from git if you want try it out. + +[meson]: http://mesonbuild.com/ +[meson-mail]: https://lists.freedesktop.org/archives/gstreamer-devel/2016-September/060231.html +[meson-videos]: http://mesonbuild.com/videos.html +[meson-gstconf]: https://gstconf.ubicast.tv/videos/gstreamer-development-on-windows-ans-faster-builds-everywhere-with-meson/ + +#### Windows Visual Studio toolchain support + +Windows users might appreciate being able to build GStreamer using the MSVC +toolchain, which is not possible using autotools. This means that it will be +possible to debug GStreamer and applications in Visual Studio, for example. +We require VS2015 or newer for this at the moment. + +There are two ways to build GStreamer using the MSVC toolchain: + +1. Using the MSVC command-line tools (`cl.exe` etc.) via Meson's "ninja" backend. +2. Letting Meson's "vs2015" backend generate Visual Studio project files that + can be opened in Visual Studio and compiled from there. + +This is currently only for adventurous souls though. All the bits are in place, +but support for all of this has not been merged into GStreamer's cerbero build +tool yet at the time of writing. This will hopefully happen in the next cycle, +but for now this means that those wishing to compile GStreamer with MSVC will +have to get their hands dirty. + +There are also no binary SDK builds using the MSVC toolchain yet. + +For more information on GStreamer builds using Meson and the Windows toolchain +check out Nirbheek Chauhan's blog post ["Building and developing GStreamer using Visual Studio"][msvc-blog]. + +[msvc-blog]: http://blog.nirbheek.in/2016/07/building-and-developing-gstreamer-using.html + +### Dependencies + +#### gstreamer + +libunwind was added as an optional dependency. It is used only for debugging +and tracing purposes. + +The `opencv` plugin in gst-plugins-bad can now be built against OpenCV +version 3.1, previously only 2.3-2.5 were supported. + +#### gst-plugins-ugly + +- `mpeg2dec` now requires at least libmpeg2 0.5.1 (from 2008). + +#### gst-plugins-bad + +- `gltransformation` now requires at least graphene 1.4.0. + +- `lv2` now plugin requires at least lilv 0.16 instead of slv2. + +### Packaging notes + +Packagers please note that the `gst/gstconfig.h` public header file in the +GStreamer core library moved back from being an architecture dependent include +to being architecture independent, and thus it is no longer installed into +`$(libdir)/gstreamer-1.0/include/gst` but into the normal include directory +where it lives happily ever after with all the other public header files. The +reason for this is that we now check whether the target supports unaligned +memory access based on predefined compiler macros at compile time instead of +checking it at configure time. + +## Platform-specific improvements + +### Android + +#### New universal binaries for all supported ABIs + +We now provide a "universal" tarball to allow building apps against all the +architectures currently supported (x86, x86-64, armeabi, armeabi-v7a, +armeabi-v8a). This is needed for building with recent versions of the Android +NDK which defaults to building against all supported ABIs. Use [the Android +player example][android-player-example-build] as a reference for the required +changes. + +[android-player-example-build]: https://cgit.freedesktop.org/gstreamer/gst-examples/commit/playback/player/android?id=a5cdde9119f038a1eb365aca20faa9741a38e788 + +#### Miscellaneous + +- New `ahssrc` element that allows reading the hardware sensors, e.g. compass + or accelerometer. + +### macOS (OS/X) and iOS + +- Support for querying available devices on OS/X via the GstDeviceProvider + API was added. +- It is now possible to create OpenGL|ES 3.x contexts on iOS and use them in + combination with the VideoToolbox based decoder element. +- many OpenGL/GLES improvements, see OpenGL section above + +### Windows + +- gstconfig.h: Always use dllexport/import on Windows with MSVC +- Miscellaneous fixes to make libs and plugins compile with the MVSC toolchain +- MSVC toolchain support (see Meson section above for more details) + +## New Modules for Documentation, Examples, Meson Build + +Three new git modules have been added recently: + +### gst-docs + +This is a new module where we will maintain documentation in the markdown +format. + +It contains the former gstreamer.com SDK tutorials which have kindly been made +available by Fluendo under a Creative Commons license. The tutorials have been +reviewed and updated for GStreamer 1.x and will be available as part of the +[official GStreamer documentation][doc] going forward. The old gstreamer.com +site will then be shut down with redirects pointing to the updated tutorials. + +Some of the existing docbook XML-formatted documentation from the GStreamer +core module such as the *Application Development Manual* and the *Plugin +Writer's Guide* have been converted to markdown as well and will be maintained +in the gst-docs module in future. They will be removed from the GStreamer core +module in the next cycle. + +This is just the beginning. Our goal is to provide a more cohesive documentation +experience for our users going forward, and easier to create and maintain +documentation for developers. There is a lot more work to do, get in touch if +you want to help out. + +If you encounter any problems or spot any omissions or outdated content in the +new documentation, please [file a bug in bugzilla][doc-bug] to let us know. + +We will probably release gst-docs as a separate tarball for distributions to +package in the next cycle. + +[doc]: http://gstreamer.freedesktop.org/documentation/ +[doc-bug]: https://bugzilla.gnome.org/enter_bug.cgi?product=GStreamer&component=documentation + +### gst-examples + +A new [module][examples-git] has been added for examples. It does not contain +much yet, currently it only contains a small [http-launch][http-launch] utility +that serves a pipeline over http as well as various [GstPlayer playback frontends][puis] +for Android, iOS, Gtk+ and Qt. + +More examples will be added over time. The examples in this repository should +be more useful and more substantial than most of the examples we ship as part +of our other modules, and also written in a way that makes them good example +code. If you have ideas for examples, let us know. + +No decision has been made yet if this module will be released and/or packaged. +It probably makes sense to do so though. + +[examples-git]: https://cgit.freedesktop.org/gstreamer/gst-examples/tree/ +[http-launch]: https://cgit.freedesktop.org/gstreamer/gst-examples/tree/network/http-launch/ +[puis]: https://cgit.freedesktop.org/gstreamer/gst-examples/tree/playback/player + +### gst-build + +[gst-build][gst-build-git] is a new meta module to build GStreamer using the +new Meson build system. This module is not required to build GStreamer with +Meson, it is merely for convenience and aims to provide a development setup +similar to the existing `gst-uninstalled` setup. + +gst-build makes use of Meson's [subproject feature][meson-subprojects] and sets +up the various GStreamer modules as subprojects, so they can all be updated and +built in parallel. + +This module is still very new and highly experimental. It should work at least +on Linux and Windows (OS/X needs some build fixes). Let us know of any issues +you encounter by popping into the `#gstreamer` IRC channel or by +[filing a bug][gst-build-bug]. + +This module will probably not be released or packaged (does not really make sense). + +[gst-build-git]: https://cgit.freedesktop.org/gstreamer/gst-build/tree/ +[gst-build-bug]: https://bugzilla.gnome.org/enter_bug.cgi?product=GStreamer&component=gst-build +[meson-subprojects]: https://github.com/mesonbuild/meson/wiki/Subprojects + +## Contributors + +Aaron Boxer, Aleix Conchillo Flaqué, Alessandro Decina, Alexandru Băluț, Alex +Ashley, Alex-P. Natsios, Alistair Buxton, Allen Zhang, Andreas Naumann, Andrew +Eikum, Andy Devar, Anthony G. Basile, Arjen Veenhuizen, Arnaud Vrac, Artem +Martynovich, Arun Raghavan, Aurélien Zanelli, Barun Kumar Singh, Bernhard +Miller, Brad Lackey, Branko Subasic, Carlos Garcia Campos, Carlos Rafael +Giani, Christoffer Stengren, Daiki Ueno, Damian Ziobro, Danilo Cesar Lemes de +Paula, David Buchmann, Dimitrios Katsaros, Duncan Palmer, Edward Hervey, +Emmanuel Poitier, Enrico Jorns, Enrique Ocaña González, Fabrice Bellet, +Florian Zwoch, Florin Apostol, Francisco Velazquez, Frédéric Bertolus, Fredrik +Fornwall, Gaurav Gupta, George Kiagiadakis, Georg Lippitsch, Göran Jönsson, +Graham Leggett, Gregoire Gentil, Guillaume Desmottes, Gwang Yoon Hwang, Haakon +Sporsheim, Haihua Hu, Havard Graff, Heinrich Fink, Hoonhee Lee, Hyunjun Ko, +Iain Lane, Ian, Ian Jamison, Jagyum Koo, Jake Foytik, Jakub Adam, Jan +Alexander Steffens (heftig), Jan Schmidt, Javier Martinez Canillas, Jerome +Laheurte, Jesper Larsen, Jie Jiang, Jihae Yi, Jimmy Ohn, Jinwoo Ahn, Joakim +Johansson, Joan Pau Beltran, Jonas Holmberg, Jonathan Matthew, Jonathan Roy, +Josep Torra, Julien Isorce, Jun Ji, Jürgen Slowack, Justin Kim, Kazunori +Kobayashi, Kieran Bingham, Kipp Cannon, Koop Mast, Kouhei Sutou, Kseniia, Kyle +Schwarz, Kyungyong Kim, Linus Svensson, Luis de Bethencourt, Marcin Kolny, +Marcin Lewandowski, Marianna Smidth Buschle, Mario Sanchez Prada, Mark +Combellack, Mark Nauwelaerts, Martin Kelly, Matej Knopp, Mathieu Duponchelle, +Mats Lindestam, Matthew Gruenke, Matthew Waters, Michael Olbrich, Michal Lazo, +Miguel París Díaz, Mikhail Fludkov, Minjae Kim, Mohan R, Munez, Nicola Murino, +Nicolas Dufresne, Nicolas Huet, Nikita Bobkov, Nirbheek Chauhan, Olivier +Crête, Paolo Pettinato, Patricia Muscalu, Paulo Neves, Peng Liu, Peter +Seiderer, Philippe Normand, Philippe Renon, Philipp Zabel, Pierre Lamot, Piotr +Drąg, Prashant Gotarne, Raffaele Rossi, Ray Strode, Reynaldo H. Verdejo +Pinochet, Santiago Carot-Nemesio, Scott D Phillips, Sebastian Dröge, Sebastian +Rasmussen, Sergei Saveliev, Sergey Borovkov, Sergey Mamonov, Sergio Torres +Soldado, Seungha Yang, sezero, Song Bing, Sreerenj Balachandran, Stefan Sauer, +Stephen, Steven Hoving, Stian Selnes, Thiago Santos, Thibault Saunier, Thijs +Vermeir, Thomas Bluemel, Thomas Jones, Thomas Klausner, Thomas Scheuermann, +Tim-Philipp Müller, Ting-Wei Lan, Tom Schoonjans, Ursula Maplehurst, Vanessa +Chipirras Navalon, Víctor Manuel Jáquez Leal, Vincent Penquerc'h, Vineeth TM, +Vivia Nikolaidou, Vootele Vesterblom, Wang Xin-yu (王昕宇), William Manley, +Wim Taymans, Wonchul Lee, Xabier Rodriguez Calvar, Xavier Claessens, xlazom00, +Yann Jouanin, Zaheer Abbas Merali + +... and many others who have contributed bug reports, translations, sent +suggestions or helped testing. + +## Bugs fixed in 1.10 + +More than [750 bugs][bugs-fixed-in-1.10] have been fixed during +the development of 1.10. + +This list does not include issues that have been cherry-picked into the +stable 1.8 branch and fixed there as well, all fixes that ended up in the +1.8 branch are also included in 1.10. + +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. + +[bugs-fixed-in-1.10]: https://bugzilla.gnome.org/buglist.cgi?bug_status=RESOLVED&bug_status=VERIFIED&classification=Platform&limit=0&list_id=164074&order=bug_id&product=GStreamer&query_format=advanced&resolution=FIXED&target_milestone=1.8.1&target_milestone=1.8.2&target_milestone=1.8.3&target_milestone=1.8.4&target_milestone=1.9.1&target_milestone=1.9.2&target_milestone=1.9.90&target_milestone=1.10.0 + +## Stable 1.10 branch + +After the 1.10.0 release there will be several 1.10.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 1.10.x bug-fix releases will be made from the git 1.10 branch, +which is a stable branch. + +### 1.10.0 + +1.10.0 was released on 1st November 2016. + +## Known Issues + +- iOS builds with iOS 6 SDK and old C++ STL. You need to select iOS 6 instead + of 7 or 8 in your projects settings to be able to link applications. + [Bug #766366](https://bugzilla.gnome.org/show_bug.cgi?id=766366) +- Code signing for Apple platforms has some problems currently, requiring + manual work to get your application signed. [Bug #771860](https://bugzilla.gnome.org/show_bug.cgi?id=771860) +- Building applications with Android NDK r13 on Windows does not work. Other + platforms and earlier/later versions of the NDK are not affected. + [Bug #772842](https://bugzilla.gnome.org/show_bug.cgi?id=772842) +- The new leaks tracer may deadlock the application (or exhibit other undefined + behaviour) when `SIGUSR` handling is enabled via the `GST_LEAKS_TRACER_SIG` + environment variable. [Bug #770373](https://bugzilla.gnome.org/show_bug.cgi?id=770373) +- vp8enc crashes on 32 bit Windows, but was working fine in 1.6. 64 bit Windows is unaffected. + [Bug #763663](https://bugzilla.gnome.org/show_bug.cgi?id=763663) + +## Schedule for 1.12 + +Our next major feature release will be 1.12, and 1.11 will be the unstable +development version leading up to the stable 1.12 release. The development +of 1.11/1.12 will happen in the git master branch. + +The plan for the 1.12 development cycle is yet to be confirmed, but it is +expected that feature freeze will be around early/mid-January, +followed by several 1.11 pre-releases and the new 1.12 stable release +in March. + +1.12 will be backwards-compatible to the stable 1.10, 1.8, 1.6, 1.4, 1.2 and +1.0 release series. + +- - - + +*These release notes have been prepared by Olivier Crête, Sebastian Dröge, +Nicolas Dufresne, Edward Hervey, Víctor Manuel Jáquez Leal, Tim-Philipp +Müller, Reynaldo H. Verdejo Pinochet, Arun Raghavan, Thibault Saunier, +Jan Schmidt, Wim Taymans, Matthew Waters* + +*License: [CC BY-SA 4.0](http://creativecommons.org/licenses/by-sa/4.0/)* + diff --git a/RELEASE b/RELEASE index dc4005d..8ad93fc 100644 --- a/RELEASE +++ b/RELEASE @@ -1,14 +1,15 @@ -Release notes for GStreamer libav Plugins 1.9.90 +Release notes for GStreamer libav Plugins 1.10.0 -The GStreamer team is pleased to announce the first release candidate of the -stable 1.10 release series. The 1.10 release series is adding new features on -top of the 1.0, 1.2, 1.4, 1.6 and 1.8 series and is part of the API and -ABI-stable 1.x release series of the GStreamer multimedia framework. +The GStreamer team is pleased to announce the first release of the new stable +1.10 release series. The 1.10 release series is adding new features on top of +the 1.0, 1.2, 1.4, 1.6 and 1.8 series and is part of the API and ABI-stable 1.x +release series of the GStreamer multimedia framework. -Binaries for Android, iOS, Mac OS X and Windows will be provided in the next days. - +Binaries for Android, iOS, Mac OS X and Windows will be provided shortly after +the source release by the GStreamer project during the stable 1.10 release +series. This module contains plugins based on the ffmpeg project, including codecs. @@ -34,10 +35,6 @@ contains a set of less supported plugins that haven't passed the -Bugs fixed in this release - - * 771092 : avenc: Got Caught SIGSEGV when using avenc_xxx - ==== Download ==== You can find source releases of gst-libav in the download @@ -73,8 +70,8 @@ subscribe to the gstreamer-devel list. Contributors to this release - * Iain Lane - * Jan Schmidt + * Nirbheek Chauhan * Sebastian Dröge * Thibault Saunier + * Tim-Philipp Müller   \ No newline at end of file diff --git a/configure.ac b/configure.ac index 9cba611..013217c 100644 --- a/configure.ac +++ b/configure.ac @@ -3,7 +3,7 @@ AC_PREREQ(2.69) dnl initialize autoconf dnl when going to/from release please set the nano (fourth number) right ! dnl releases only do Wall, cvs and prerelease does Werror too -AC_INIT(GStreamer libav, 1.9.90, +AC_INIT(GStreamer libav, 1.10.0, http://bugzilla.gnome.org/enter_bug.cgi?product=GStreamer, gst-libav) @@ -40,11 +40,11 @@ GST_API_VERSION=1.0 AC_SUBST(GST_API_VERSION) AG_GST_LIBTOOL_PREPARE -AS_LIBTOOL(GST, 990, 0, 990) +AS_LIBTOOL(GST, 1000, 0, 1000) dnl *** required versions of GStreamer stuff *** -GST_REQ=1.9.90 -GST_PBREQ=1.9.90 +GST_REQ=1.10.0 +GST_PBREQ=1.10.0 ORC_REQ=0.4.16 ORC_CHECK([$ORC_REQ]) diff --git a/docs/plugins/gst-libav-plugins.args b/docs/plugins/gst-libav-plugins.args index 5058d34..6023fbc 100644 --- a/docs/plugins/gst-libav-plugins.args +++ b/docs/plugins/gst-libav-plugins.args @@ -33848,3 +33848,103 @@ 0 + +avmux_tta::maxdelay +gint +>= 0 +rw +maxdelay +Set the maximum demux-decode delay (in microseconds). +0 + + + +avmux_tta::preload +gint +>= 0 +rw +preload +Set the initial demux-decode delay (in microseconds). +0 + + + +avmux_ogv::maxdelay +gint +>= 0 +rw +maxdelay +Set the maximum demux-decode delay (in microseconds). +0 + + + +avmux_ogv::preload +gint +>= 0 +rw +preload +Set the initial demux-decode delay (in microseconds). +0 + + + +avmux_fifo::maxdelay +gint +>= 0 +rw +maxdelay +Set the maximum demux-decode delay (in microseconds). +0 + + + +avmux_fifo::preload +gint +>= 0 +rw +preload +Set the initial demux-decode delay (in microseconds). +0 + + + +avenc_truehd::bitrate +gint +>= 0 +rw +Bit Rate +Target Audio Bitrate. +128000 + + + +avenc_truehd::compliance +GstFFMpegCompliance + +rw +Compliance +Adherence of the encoder to the specifications. +Normal behavior + + + +avenc_mlp::bitrate +gint +>= 0 +rw +Bit Rate +Target Audio Bitrate. +128000 + + + +avenc_mlp::compliance +GstFFMpegCompliance + +rw +Compliance +Adherence of the encoder to the specifications. +Normal behavior + + diff --git a/docs/plugins/gst-libav-plugins.hierarchy b/docs/plugins/gst-libav-plugins.hierarchy index 1c01391..06653f9 100644 --- a/docs/plugins/gst-libav-plugins.hierarchy +++ b/docs/plugins/gst-libav-plugins.hierarchy @@ -155,6 +155,7 @@ GObject avenc_g722 avenc_g723_1 avenc_g726 + avenc_mlp avenc_mp2 avenc_mp2fixed avenc_nellymoser @@ -163,6 +164,7 @@ GObject avenc_s302m avenc_sonic avenc_sonicls + avenc_truehd avenc_tta avenc_wavpack avenc_wmav1 @@ -487,6 +489,7 @@ GObject avmux_dvd avmux_f4v avmux_ffm + avmux_fifo avmux_filmstrip avmux_flv avmux_gxf @@ -517,6 +520,7 @@ GObject avmux_nut avmux_oga avmux_ogg + avmux_ogv avmux_oma avmux_opus avmux_psp @@ -530,6 +534,7 @@ GObject avmux_spx avmux_svcd avmux_swf + avmux_tta avmux_uncodedframecrc avmux_vc1test avmux_vcd diff --git a/docs/plugins/gst-libav-plugins.interfaces b/docs/plugins/gst-libav-plugins.interfaces index 5461f21..12a161b 100644 --- a/docs/plugins/gst-libav-plugins.interfaces +++ b/docs/plugins/gst-libav-plugins.interfaces @@ -47,6 +47,7 @@ avenc_jpeg2000 GstPreset avenc_jpegls GstPreset avenc_ljpeg GstPreset avenc_mjpeg GstPreset +avenc_mlp GstPreset avenc_mp2 GstPreset avenc_mp2fixed GstPreset avenc_mpeg1video GstPreset @@ -86,6 +87,7 @@ avenc_sunrast GstPreset avenc_svq1 GstPreset avenc_targa GstPreset avenc_tiff GstPreset +avenc_truehd GstPreset avenc_tta GstPreset avenc_utvideo GstPreset avenc_v308 GstPreset @@ -126,6 +128,7 @@ avmux_dv GstTagSetter avmux_dvd GstTagSetter avmux_f4v GstTagSetter avmux_ffm GstTagSetter +avmux_fifo GstTagSetter avmux_filmstrip GstTagSetter avmux_flv GstTagSetter avmux_gxf GstTagSetter @@ -156,6 +159,7 @@ avmux_mxf_opatom GstTagSetter avmux_nut GstTagSetter avmux_oga GstTagSetter avmux_ogg GstTagSetter +avmux_ogv GstTagSetter avmux_oma GstTagSetter avmux_opus GstTagSetter avmux_psp GstTagSetter @@ -170,6 +174,7 @@ avmux_spdif GstTagSetter avmux_spx GstTagSetter avmux_svcd GstTagSetter avmux_swf GstTagSetter +avmux_tta GstTagSetter avmux_uncodedframecrc GstTagSetter avmux_vc1test GstTagSetter avmux_vcd GstTagSetter diff --git a/docs/plugins/inspect/plugin-libav.xml b/docs/plugins/inspect/plugin-libav.xml index 624c3ba..9e8adeb 100644 --- a/docs/plugins/inspect/plugin-libav.xml +++ b/docs/plugins/inspect/plugin-libav.xml @@ -3,7 +3,7 @@ All libav codecs and formats (local snapshot) ../../ext/libav/.libs/libgstlibav.so libgstlibav.so - 1.9.90 + 1.10.0 LGPL gst-libav libav @@ -3665,7 +3665,7 @@ avdec_magicyuv - libav MagicYUV Lossless Video decoder + libav MagicYUV video decoder Codec/Decoder/Video libav magicyuv decoder Wim Taymans <wim.taymans@gmail.com>, Ronald Bultje <rbultje@ronald.bitfreak.net>, Edward Hervey <bilboed@bilboed.com> @@ -8680,6 +8680,27 @@ + avenc_mlp + libav MLP (Meridian Lossless Packing) encoder + Codec/Encoder/Audio + libav mlp encoder + Wim Taymans <wim.taymans@gmail.com>, Ronald Bultje <rbultje@ronald.bitfreak.net> + + + sink + sink + always +
audio/x-raw, channel-mask=(bitmask)0x0000000000000000, channels=(int)1, rate=(int){ 44100, 48000, 88200, 96000, 176400, 192000 }, layout=(string)interleaved, format=(string)S16LE; audio/x-raw, channel-mask=(bitmask)0x0000000000000003, channels=(int)2, rate=(int){ 44100, 48000, 88200, 96000, 176400, 192000 }, layout=(string)interleaved, format=(string)S16LE; audio/x-raw, channel-mask=(bitmask)0x0000000000000103, channels=(int)3, rate=(int){ 44100, 48000, 88200, 96000, 176400, 192000 }, layout=(string)interleaved, format=(string)S16LE; audio/x-raw, channel-mask=(bitmask)0x0000000000000033, channels=(int)4, rate=(int){ 44100, 48000, 88200, 96000, 176400, 192000 }, layout=(string)interleaved, format=(string)S16LE; audio/x-raw, channel-mask=(bitmask)0x000000000000000b, channels=(int)3, rate=(int){ 44100, 48000, 88200, 96000, 176400, 192000 }, layout=(string)interleaved, format=(string)S16LE; audio/x-raw, channel-mask=(bitmask)0x0000000000000007, channels=(int)3, rate=(int){ 44100, 48000, 88200, 96000, 176400, 192000 }, layout=(string)interleaved, format=(string)S16LE; audio/x-raw, channel-mask=(bitmask)0x0000000000000107, channels=(int)4, rate=(int){ 44100, 48000, 88200, 96000, 176400, 192000 }, layout=(string)interleaved, format=(string)S16LE; audio/x-raw, channel-mask=(bitmask)0x0000000000000037, channels=(int)5, rate=(int){ 44100, 48000, 88200, 96000, 176400, 192000 }, layout=(string)interleaved, format=(string)S16LE; audio/x-raw, channel-mask=(bitmask)0x000000000000000f, channels=(int)4, rate=(int){ 44100, 48000, 88200, 96000, 176400, 192000 }, layout=(string)interleaved, format=(string)S16LE; audio/x-raw, channel-mask=(bitmask)0x000000000000010f, channels=(int)5, rate=(int){ 44100, 48000, 88200, 96000, 176400, 192000 }, layout=(string)interleaved, format=(string)S16LE; audio/x-raw, channel-mask=(bitmask)0x000000000000003f, channels=(int)6, rate=(int){ 44100, 48000, 88200, 96000, 176400, 192000 }, layout=(string)interleaved, format=(string)S16LE
+ + + src + source + always +
audio/x-mlp, channels=(int)[ 1, 2 ], rate=(int)[ 4000, 96000 ]
+ + + + avenc_mp2 libav MP2 (MPEG audio layer 2) encoder Codec/Encoder/Audio @@ -9415,6 +9436,27 @@ + avenc_truehd + libav TrueHD encoder + Codec/Encoder/Audio + libav truehd encoder + Wim Taymans <wim.taymans@gmail.com>, Ronald Bultje <rbultje@ronald.bitfreak.net> + + + sink + sink + always +
audio/x-raw, channel-mask=(bitmask)0x0000000000000003, channels=(int)2, rate=(int){ 44100, 48000, 88200, 96000, 176400, 192000 }, layout=(string)interleaved, format=(string)S16LE; audio/x-raw, channel-mask=(bitmask)0x0000000000000037, channels=(int)5, rate=(int){ 44100, 48000, 88200, 96000, 176400, 192000 }, layout=(string)interleaved, format=(string)S16LE; audio/x-raw, channel-mask=(bitmask)0x000000000000003f, channels=(int)6, rate=(int){ 44100, 48000, 88200, 96000, 176400, 192000 }, layout=(string)interleaved, format=(string)S16LE
+ + + src + source + always +
audio/x-true-hd, channels=(int)[ 1, 8 ], rate=(int)[ 4000, 96000 ]
+ + + + avenc_tta libav TTA (True Audio) encoder Codec/Encoder/Audio @@ -9446,7 +9488,7 @@ sink sink always -
video/x-raw, format=(string){ RGB, RGBA, Y42B, I420 }
+
video/x-raw, format=(string){ RGB, RGBA, Y42B, I420, Y444 }
src @@ -10177,6 +10219,15 @@ + avmux_fifo + libav FIFO queue pseudo-muxer muxer + Codec/Muxer + libav FIFO queue pseudo-muxer muxer + Wim Taymans <wim.taymans@chello.be>, Ronald Bultje <rbultje@ronald.bitfreak.net> + + + + avmux_filmstrip libav Adobe Filmstrip muxer Codec/Muxer @@ -10903,6 +10954,27 @@ + avmux_ogv + libav Ogg Video muxer + Codec/Muxer + libav Ogg Video muxer + Wim Taymans <wim.taymans@chello.be>, Ronald Bultje <rbultje@ronald.bitfreak.net> + + + video_%u + sink + request +
video/x-vp8
+ + + src + source + always +
application/x-gst-av-ogv
+ + + + avmux_oma libav Sony OpenMG audio muxer Codec/Muxer @@ -11194,6 +11266,27 @@ + avmux_tta + libav TTA (True Audio) muxer + Codec/Muxer + libav TTA (True Audio) muxer + Wim Taymans <wim.taymans@chello.be>, Ronald Bultje <rbultje@ronald.bitfreak.net> + + + audio_%u + sink + request +
audio/x-tta, channels=(int)[ 1, 2 ], rate=(int)[ 4000, 96000 ]
+ + + src + source + always +
audio/x-ttafile
+ + + + avmux_uncodedframecrc libav uncoded framecrc testing muxer Codec/Muxer diff --git a/gst-libav.doap b/gst-libav.doap index a7fd524..a9a20fe 100644 --- a/gst-libav.doap +++ b/gst-libav.doap @@ -34,6 +34,16 @@ colorspace conversion elements. + 1.10.0 + master + + 2016-11-01 + + + + + + 1.9.90 master