diff options
| author | Sebastian Dröge <sebastian@centricular.com> | 2017-01-12 16:14:46 +0200 |
|---|---|---|
| committer | Sebastian Dröge <sebastian@centricular.com> | 2017-01-12 16:14:46 +0200 |
| commit | c860590a8cdccda4a73b29d3d2a57e2a60ee6bd3 (patch) | |
| tree | 5a05e323edfd21587729f053c11ef3ad5656d1b8 | |
| parent | fb7833245de53bd6e409f5faf228be7899ce933f (diff) | |
Release 1.11.1
| -rw-r--r-- | ChangeLog | 242 | ||||
| -rw-r--r-- | NEWS | 1115 | ||||
| -rw-r--r-- | RELEASE | 46 | ||||
| -rw-r--r-- | configure.ac | 12 | ||||
| -rw-r--r-- | gst-rtsp-server.doap | 10 | ||||
| -rw-r--r-- | win32/common/libgstrtspserver.def | 2 |
6 files changed, 292 insertions, 1135 deletions
@@ -1,9 +1,247 @@ +=== release 1.11.1 === + +2017-01-12 Sebastian Dröge <slomo@coaxion.net> + + * configure.ac: + releasing 1.11.1 + +2017-01-10 08:34:50 +0100 Patricia Muscalu <patricia@axis.com> + + * gst/rtsp-server/rtsp-stream.c: + rtsp-stream: corrected if-statement in _get_server_port() + This bug was accidentally introduced while fixing a segfault + in _get_server_port() function. + https://bugzilla.gnome.org/show_bug.cgi?id=776345 + +2017-01-09 14:12:05 +0100 Patricia Muscalu <patricia@axis.com> + + * gst/rtsp-server/rtsp-stream.c: + * tests/check/gst/stream.c: + rtsp-stream: fixed segmenation fault in _get_server_port() + Calling function gst_rtsp_stream_get_server_port() results in + segmenation fault in the RTP/RTSP/TCP case. + Port that the server will use to receive RTCP makes only + sense in the UDP case, however the function should handle + the TCP case in a nicer way. + https://bugzilla.gnome.org/show_bug.cgi?id=776345 + +2017-01-09 12:22:40 +0300 Aleksandr Slobodeniuk <alenuke@yandex.ru> + + * gst/rtsp-server/rtsp-media-factory.c: + dosc: Fix a little typo + https://bugzilla.gnome.org/show_bug.cgi?id=777037 + +2017-01-04 16:20:54 +0100 Guillaume Desmottes <guillaume.desmottes@collabora.co.uk> + + * pkgconfig/Makefile.am: + * pkgconfig/gstreamer-rtsp-server-uninstalled.pc.in: + * pkgconfig/meson.build: + meson: generate pkg-config -uninstalled pc files + Generating those files is useful for users building the GStreamer stack + using meson and having to link it to another project which is still + using the autotools. + https://bugzilla.gnome.org/show_bug.cgi?id=776810 + +2017-01-04 16:11:08 +0100 Guillaume Desmottes <guillaume.desmottes@collabora.co.uk> + + * pkgconfig/gstreamer-rtsp-server-uninstalled.pc.in: + pkgconfig: fix -uninstalled pc file + pcfiledir was never defined so the paths were wrong. + https://bugzilla.gnome.org/show_bug.cgi?id=776867 + +2016-12-21 13:41:50 +0100 Patricia Muscalu <patricia@axis.com> + + * gst/rtsp-server/rtsp-stream.c: + * tests/check/gst/rtspserver.c: + rtsp-stream: Fixed TCP transport case + Make sure that the appsink element is actually added to + the bin before trying to link it with the elements in it. + https://bugzilla.gnome.org/show_bug.cgi?id=776343 + +2016-12-16 17:26:04 +0000 Tim-Philipp Müller <tim@centricular.com> + + * .gitignore: + * Makefile.am: + * configure.ac: + * gst-rtsp.spec.in: + Remove generated .spec file + Likely extremely bitrotten, and we should not ship this anyway. + +2016-12-03 08:21:02 +0100 Edward Hervey <bilboed@bilboed.com> + + * common: + Automatic update of common submodule + From f980fd9 to 39ac2f5 + +2016-12-02 15:40:09 +0100 Edward Hervey <edward@centricular.com> + + * gst/rtsp-server/rtsp-media.c: + media: Fix pt map caps + Since decryption is handled within rtpbin, all outcoming stream + caps will be application/x-rtp (i.e. regular rtp) + Fixes RECORD with SRTP streams + +2016-12-02 15:38:04 +0100 Edward Hervey <edward@centricular.com> + + * gst/rtsp-server/rtsp-media-factory.c: + media-factory: Create media objects with the proper transport mode + The function called immediately afterwards (collect_streams()) will + need it to work properly + +2016-12-02 14:36:50 +0200 Sebastian Dröge <sebastian@centricular.com> + + * gst/rtsp-server/rtsp-auth.c: + rtsp-auth: Don't remove digest-auth nonces that already/still have a client connected + +2016-12-01 18:04:34 +0200 Sebastian Dröge <sebastian@centricular.com> + + * gst/rtsp-server/rtsp-media-factory.c: + rtsp-media-factory: Don't create a pipeline for the media pipeline string + We're going to put a pipeline into a pipeline otherwise, which is not + exactly ideal. + +2016-10-25 15:41:28 +0300 Kseniia Vasilchuk <vasilchukkseniia@gmail.com> + + * gst/rtsp-server/rtsp-media.c: + media: Fix race condition around finish_unprepare() if called multiple time + https://bugzilla.gnome.org/show_bug.cgi?id=755329 + +2016-11-30 14:06:36 +1100 Jan Schmidt <jan@centricular.com> + + * gst/rtsp-sink/gstrtspclientsink.c: + rtspclientsink: Don't leave stale pointer after unref + Fix a warning on shutdown - don't keep a pointer to an + alread-unreffed object. + +2016-11-26 11:24:50 +0000 Tim-Philipp Müller <tim@centricular.com> + + * .gitmodules: + common: use https protocol for common submodule + https://bugzilla.gnome.org/show_bug.cgi?id=775110 + +2016-11-21 23:29:56 +1100 Matthew Waters <matthew@centricular.com> + + * gst/rtsp-server/rtsp-stream.c: + stream: block the output of rtpbin instead of the source pipeline + 85c52e194bcb81928b96614be0ae47d59eccb1ce introduced a more correct + detection of the srtp rollover counter to add to the SDP. + Unfortunately, it was incomplete for live pipelines where the logic + blocks the source bin before creating the SDP and thus would never have + the necessary informaiton to create a correct SDP with srtp encryption. + Move the pad blocks to rtpbin's output pads instead so that the + necessary information can be created before we need the information for + the SDP. + https://bugzilla.gnome.org/show_bug.cgi?id=770239 + +2016-11-21 16:02:39 +0100 Dag Gullberg <dagg@axis.com> + + * gst/rtsp-server/rtsp-client.c: + rtsp-client: add IDLE timeout, before session exists + The RTSP server will not timeout an idle RTSP connection + (note this is different from doing timeout on a RTSP + session). + At least for Apache this is a problem when running RTSP over + HTTPS since it uses one of the threads (there is a rather + limited number) that are available for handling requests. + https://bugzilla.gnome.org/show_bug.cgi?id=771830 + +2016-11-23 09:45:08 +0000 Tim-Philipp Müller <tim@centricular.com> + + * .gitignore: + .gitignore more + +2016-11-21 13:05:50 +0100 Göran Jönsson <goranjn@axis.com> + + * gst/rtsp-server/rtsp-stream.c: + rtsp-stream: Set close-socket FALSE on UDP src:es + With this RTSP server can use the sockets independent on the udpsrc + state. + When the udp src is finalized it will unref socket and when g_socket + is finalized the socket will be closed. + https://bugzilla.gnome.org/show_bug.cgi?id=765673 + +2016-11-18 17:47:13 +0200 Sebastian Dröge <sebastian@centricular.com> + + * gst/rtsp-sink/gstrtspclientsink.c: + rtspclientsink: Move to new helper function to parse authentication responses + https://bugzilla.gnome.org/show_bug.cgi?id=774416 + +2016-11-16 08:42:24 +0200 Sebastian Dröge <sebastian@centricular.com> + + * examples/Makefile.am: + * examples/test-auth-digest.c: + * gst/rtsp-server/rtsp-auth.c: + * gst/rtsp-server/rtsp-auth.h: + * win32/common/libgstrtspserver.def: + rtsp-auth: Add support for Digest authentication + https://bugzilla.gnome.org/show_bug.cgi?id=774416 + +2016-11-17 09:41:53 -0800 Scott D Phillips <scott.d.phillips@intel.com> + + * Makefile.am: + * gst/rtsp-server/meson.build: + * meson.build: + * tests/check/meson.build: + * win32/MANIFEST: + * win32/common/libgstrtspserver.def: + Enable building with MSVC + https://bugzilla.gnome.org/show_bug.cgi?id=774640 + +2016-11-18 20:23:14 -0300 Thibault Saunier <thibault.saunier@osg.samsung.com> + + * meson.build: + meson: gstreamer gst_check_dep does not exist on windows + +2016-11-17 09:43:37 -0800 Scott D Phillips <scott.d.phillips@intel.com> + + * gst/rtsp-server/rtsp-client.c: + client: update do_send_message to match type GstRTSPClientSendFunc + This type mismatch fails building with MSVC + https://bugzilla.gnome.org/show_bug.cgi?id=774640 + +2016-11-11 14:42:08 +0200 Sebastian Dröge <sebastian@centricular.com> + + * gst/rtsp-server/rtsp-sdp.c: + rtsp-sdp: Fix indentation + +2016-11-10 05:16:00 +0000 Neha Arora <arora.neha@samsung.com> + + * gst/rtsp-server/rtsp-media.c: + rtsp-media: Only signal "new-state" if the state has actually changed + https://bugzilla.gnome.org/show_bug.cgi?id=774173 + +2016-08-24 11:39:13 +0200 Branko Subasic <branko@axis.com> + + * gst/rtsp-server/rtsp-client.c: + * gst/rtsp-server/rtsp-client.h: + client: emit signal in the beginning of each rtsp request + These signals let the application validate the requests, configure the + media/stream in a certain way and also generate error status code in + case of error or bad request. + https://bugzilla.gnome.org/show_bug.cgi?id=758062 + +2016-11-01 18:10:35 +0000 Tim-Philipp Müller <tim@centricular.com> + + * meson.build: + meson: update version + +=== release 1.11.0 === + +2016-11-01 18:53:15 +0200 Sebastian Dröge <sebastian@centricular.com> + + * configure.ac: + Back to development + === release 1.10.0 === -2016-11-01 Sebastian Dröge <slomo@coaxion.net> +2016-11-01 18:06:46 +0200 Sebastian Dröge <sebastian@centricular.com> + * ChangeLog: + * NEWS: + * RELEASE: * configure.ac: - releasing 1.10.0 + * gst-rtsp-server.doap: + Release 1.10.0 2016-10-28 18:38:01 +0100 Tim-Philipp Müller <tim@centricular.com> @@ -1,1114 +1 @@ -# 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/)* - +This is GStreamer 1.11.1. @@ -1,23 +1,32 @@ -Release notes for GStreamer RTSP Server Library 1.10.0 +Release notes for GStreamer RTSP Server Library 1.11.1 -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. +The GStreamer team is pleased to announce the first release of the unstable +1.11 release series. The 1.11 release series is adding new features on top of +the 1.0, 1.2, 1.4, 1.6, 1.8 and 1.10 series and is part of the API and ABI-stable 1.x release +series of the GStreamer multimedia framework. The unstable 1.11 release series +will lead to the stable 1.12 release series in the next weeks. Any newly added +API can still change until that point. -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. +Full release notes will be provided at some point during the 1.11 release +cycle, highlighting all the new features, bugfixes, performance optimizations +and other important changes. + + +Binaries for Android, iOS, Mac OS X and Windows will be provided in the next days. Bugs fixed in this release - * 771983 : Deadlock when closing session and backlog is full. - * 772478 : Missing video stream from SDP - * 773640 : rtspclient unit test failures + * 758062 : rtsp-client: emit new rtsp request signals in the beginning of each request + * 771830 : There is no time out in idle connection RTSP server + * 774173 : media: emit signal SIGNAL_NEW_STATE only when state change happens + * 774640 : gst-rtsp-server: Enable building with MSVC + * 776867 : pkgconfig: fix -uninstalled pc file + * 777037 : rtsp-factory: just fixing a little typo in comments + * 774416 : RTSP digest Authentification for gst-rtsp-server ==== Download ==== @@ -54,8 +63,19 @@ subscribe to the gstreamer-devel list. Contributors to this release + * Aleksandr Slobodeniuk + * Branko Subasic + * Dag Gullberg + * Edward Hervey + * Guillaume Desmottes * Göran Jönsson - * Nikita Bobkov + * Jan Schmidt + * Kseniia Vasilchuk + * Matthew Waters + * Neha Arora + * Patricia Muscalu + * Scott D Phillips + * Sebastian Dröge + * Thibault Saunier * Tim-Philipp Müller - * Xavier Claessens
\ No newline at end of file diff --git a/configure.ac b/configure.ac index 8167ff9..012531b 100644 --- a/configure.ac +++ b/configure.ac @@ -2,7 +2,7 @@ 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 RTSP Server Library], [1.11.0.1], +AC_INIT([GStreamer RTSP Server Library], [1.11.1], [http://bugzilla.gnome.org/enter_bug.cgi?product=GStreamer], [gst-rtsp-server]) AG_GST_INIT @@ -53,13 +53,13 @@ dnl 1.2.5 => 205 dnl 1.10.9 (who knows) => 1009 dnl dnl sets GST_LT_LDFLAGS -AS_LIBTOOL(GST, 1100, 0, 1100) +AS_LIBTOOL(GST, 1101, 0, 1101) dnl *** required versions of GStreamer stuff *** -GST_REQ=1.11.0.1 -GSTPB_REQ=1.11.0.1 -GSTPG_REQ=1.11.0.1 -GSTPD_REQ=1.11.0.1 +GST_REQ=1.11.1 +GSTPB_REQ=1.11.1 +GSTPG_REQ=1.11.1 +GSTPD_REQ=1.11.1 dnl *** autotools stuff **** diff --git a/gst-rtsp-server.doap b/gst-rtsp-server.doap index 27bb2ba..33fe0f8 100644 --- a/gst-rtsp-server.doap +++ b/gst-rtsp-server.doap @@ -32,6 +32,16 @@ RTSP server library based on GStreamer <release> <Version> + <revision>1.11.1</revision> + <branch>master</branch> + <name></name> + <created>2017-01-12</created> + <file-release rdf:resource="http://gstreamer.freedesktop.org/src/gst-rtsp-server/gst-rtsp-server-1.11.1.tar.xz" /> + </Version> + </release> + + <release> + <Version> <revision>1.10.0</revision> <branch>master</branch> <name></name> diff --git a/win32/common/libgstrtspserver.def b/win32/common/libgstrtspserver.def index 18318e3..93f12fa 100644 --- a/win32/common/libgstrtspserver.def +++ b/win32/common/libgstrtspserver.def @@ -197,6 +197,7 @@ EXPORTS gst_rtsp_session_get_sessionid gst_rtsp_session_get_timeout gst_rtsp_session_get_type + gst_rtsp_session_is_expired gst_rtsp_session_is_expired_usec gst_rtsp_session_manage_media gst_rtsp_session_media_alloc_channels @@ -212,6 +213,7 @@ EXPORTS gst_rtsp_session_media_set_state gst_rtsp_session_media_set_transport gst_rtsp_session_new + gst_rtsp_session_next_timeout gst_rtsp_session_next_timeout_usec gst_rtsp_session_pool_cleanup gst_rtsp_session_pool_create |