-This is GStreamer Base Plug-ins 0.10.35, "Short Notice"
-
-Changes since 0.10.34:
-
- * work around GLib atomic ops API change
- * don't use G_CONST_RETURN in public headers
- * subparse: typefinding fixes for subtitles in non-UTF8 charsets
-
-Bugs fixed since 0.10.34:
-
- * 600043 : subparse: fails to recognise Cyrillic subtitles in windows-1251 encoding
-
-Changes since 0.10.33:
-
- * None: this release is identical to 0.10.33 and just done to keep core/base
- versions in sync
-
-Changes since 0.10.32:
-
- * audioringbuffer: make sure to not start if the may_start flag is FALSE
- * baseaudiosink: arrange for running clock when rendering eos
- * baseaudiosink: don't allow aligning behind the read-segment
- * baseaudiosink: start ringbuffer upon going to PLAYING and already EOS
- * riff: Add support for video/x-camstudio
- * rtcpbuffer: fix invalid read in validation of padding in rtcp packet
- * rtcpbuffer: Round to next 32bit word, not current 32bit word at end of SDES chunk
- * rtpbuffer: Off-by-one error when creating RTP header extensions with a two-byte header
- * rtsptransport: ensure valid int result when parsing ranges
- * tag: map the ID3v2 TENC frame to GST_TAG_ENCODED_BY
- * tag: add GST_TAG_CAPTURING_EXPOSURE_COMPENSATION incl. EXIF/XMP mappings
- * tag: add a new GstTagXmpWriter interface to select XMP schemas to be used
- * tagdemux: also push cached events downstream when operating in pull mode
- * video: add GST_VIDEO_BUFFER_PROGRESSIVE flag
- * video: add ARGB64 and AYUV64 (16 bits per channel) formats
- * video: add r210 (10 bits per channel) format
- * video: add gst_video_format_get_component_depth() and _new_template_caps()
- * video: fix creation of grayscale caps and height calculation for YUV9/YVU9
- * appsink: emit "new-buffer-list" signal for buffer lists if handled by app
- * audiorate: add "skip-to-first" property
- * decodebin2: don't use the same parser element multiple times in the same chain
- * decodebin2: improve detection of raw caps in expose-all-streams=false mode
- * discoverer: don't wait for subtitle streams to preroll; leak fixes
- * discoverer: use nominal bitrate if bitrate tag is unavailable
- * encodebin: add an audioconvert after the audio resampler
- * encodebin: fix refcounting issues and leaks related to request pads
- * encodebin: return a new reference of the pad for the "request-pad" signal
- * encodebin: set all elements to NULL and remove them from the bin when removing a source group
- * encodebin: tear down old profiles when setting new ones
- * multifdsink: disconnect inactive clients in the select loop too
- * oggmux: prefer headers from caps to determine stream type (for VP8)
- * oggmux: fix issue with ogg page numbering and discont flag handling
- * oggmux: ensure stream serial numbers are unique
- * oggmux: use running time for muxing instead of timestamps
- * oggparse: better detection of delta unit flag
- * playbin2, uridecodebin: add "source-setup" signal
- * playbin2: always prefer the custom set sink and also set it back to NULL in all cases
- * playbin2: check if an already existing sink supports the non-raw format too
- * playbin2: fix handling of non-raw custom sinks
- * playbin2: if a sink claims to support ANY caps assume that it only supports the usual raw formats
- * playbin2: only consider the audio/video sinks in autoplug_continue for the normal uridecodebin
- * playbin2: use gst_pad_accept_caps() instead of intersecting with the getcaps caps
- * playbin2: set sinks to READY before checking if it accept caps
- * textoverlay: add support for ARGB and other RGB alpha variants, and xBGR and RGBx
- * textoverlay: add support for vertical center alignment
- * textoverlay: converted AYUV to use 'A OVER B' alpha compositing
- * textoverlay: use a class wide mutex to work around pango reentrance issues
- * theoraenc: don't reset the video quality when setting the bitrate
- * theoraenc: allow adjustment of the speed level while running
- * theoraenc: set speed-level property defaults from libtheora's defaults
- * typefinding: MPEG-TS detection fixes
- * typefinding: detect HTTP live streaming m3u8 playlists
- * typefinding: detect windows icon files and DEGAS images (to avoid false positives)
- * typefinding: detect raw h.263
- * typefinding: add depth and endianness fields to DTS caps
- * uridecodebin: Add default handler for autoplug-select
- * uridecodebin: add https:// to protocols for which to enable buffering
- * uridecodebin: expose "autoplug-sort" signal
- * uridecodebin: post proper error message if decodebin2/typefind elements are missing
- * uridecodebin: Return NULL from the default autoplug-sort handler
- * videorate: fix "skip-to-first" timestamp setup
- * videoscale: add 16-bit-channel support (ARGB64, AYUV64), fix ARGB bilinear scaling
- * videotestsrc: add 16-bit-per-channel support (ARGB64, AYUV64)
- * vorbis: add support for using tremolo on android
- * vorbistag: Add support for METADATA_BLOCK_PICTURE tags
- * vorbistag: Write GST_TAG_IMAGE and GST_TAG_PREVIEW_IMAGE as METADATA_BLOCK_PICTURE
- * win32: fix DEFAULT_AUDIOSINK, should be direct*sound*sink
- * xvimagesink: don't paint the window black when going to NULL
-
-Bugs fixed since 0.10.32:
-
- * 618516 : [typefinding] need raw H.263 typefinder
- * 619778 : oggdemux: fails on zero-length pages with Patent_Absurdity_HD_3540kbit.ogv
- * 633837 : videoscale: invalid reads after conversion to orc linear scaling
- * 412678 : random segfaults or memory corruptions with multiple textoverlays (pango not reentrant)
- * 620364 : [typefinding] .ico file detected as AAC
- * 625129 : typefinding: file incorrectly detected as audio/x-dts
- * 626152 : [playbin2] add " source-setup " signal
- * 627268 : [tag] add GST_TAG_ENCODED_BY and map id3v2 TENC frame
- * 629196 : oggmux: re-tagging an Ogg Vorbis file may corrupt audio data
- * 632291 : discoverer: sparse tracks cause prerolling to hang till timeout
- * 632889 : [multifdsink] [PATCH] Disconnect inactive clients in the select loop too
- * 635669 : [vorbistag] Support METADATA_BLOCK_PICTURE for Vorbis cover art
- * 635784 : ringbuffer: make sure to not start if the may_start flag is FALSE
- * 635800 : xvimagesink flashes black when going from READY_TO_NULL
- * 636886 : baseaudiosink: no running clock when eos leads to hang in PLAYING
- * 639136 : [oggparse]code is not safe when using libogg fuctions
- * 639159 : [textoverloay] Add vertical center alignment option
- * 639237 : textoverlay: patch to use " A OVER B " alpha compositing
- * 639744 : [oggdemux] Removing dead code:
- * 640189 : oggmux: cleanup
- * 640211 : oggmux: ensure serialnos are unique
- * 640607 : appsink never sends " new-buffer-list " signal
- * 640709 : [typefindfunctions] h264 typefinder registered with MPEG_VIDEO_CAPS
- * 640804 : checks: encodebin test fails if theora or vorbis plugins are not available
- * 641706 : discoverer: Keep references on discoverer objects for callbacks
- * 641860 : discoverer: Use nominal bitrate if bitrate tag is unavailable
- * 641917 : [gdppay] Ensure buffer's medata is writeable before setting it
- * 641927 : [encodebin] refcount issue with the " request-pad " signal
- * 641952 : [videoscale] assertion on fixate_caps
- * 642174 : Playbin2 cannot work with non-raw custom sinks
- * 642232 : theoraenc sets Video quality to zero when explicitely setting the bitrate to 0
- * 642274 : [playbin2] arbitrary audio-sink is chosen even though explicitely having set a custom audio-sink bin
- * 642381 : potential memleak in decodebin2
- * 642466 : playbin2: after replacing a video sink with the pipeline in NULL state I still get the old one
- * 642720 : audiotestsrc: pipelines with multiple instances with wave=gaussian-noise, white-noise, or pink-noise are very slow
- * 642942 : adder: offset_end field of outgoing buffers is set to GST_BUFFER_OFFSET_NONE
- * 642949 : pbutils: encoding-target: chaining error object in loading target from file may cause crash if there is no error
- * 643775 : [oggmux] use running time instead of timestamps
- * 644416 : [encodebin] Cannot be reused
- * 644745 : [oggmux] Fails to mux Speex content, doesn't preroll
- * 644845 : [alsa] Comparison of unsigned int < 0 always false in gstalsamixer.c
- * 644996 : libsABI check doesn't depend only on architecture
- * 645167 : [xmp] Add a new XmpConfig interface
- * 645437 : encoding-profile: Fix syntax in Example: Creating a profile
- * 646570 : baseaudiosink: don't allow aligning behind the read-segment
- * 646572 : baseaudiosrc: protect against ringbuffer disappearing while in a query
- * 646573 : baseaudiosrc: Add src object lock around call to ringbuffer parse caps
- * 646575 : rtcpbuffer: Round to next 32bit word, not current 32bit word at end of SDES chunk
- * 646576 : rtcpbuffer: fix invalid read in validation of padding in rtcp packet
- * 646923 : video: Remove unused variable
- * 646924 : rtp: Remove unused variables
- * 646925 : encoding-profile: Remove unused variables
- * 646952 : Fix the strlol return type mismatch :
- * 647399 : Bad typo in ID3 tags: psychadelic - > psychedelic
- * 647721 : Remove excessive checking for video.c
- * 647781 : [playbin2] missing shutdown steps and inconsistent error behaviour
- * 647856 : [oggmux] Assumes that the first buffer can be used to detect the stream type
- * 647857 : [xvimagesink/ximagesink] Handle NULL caps in buffer_alloc()
- * 647942 : [pango] Use different Pango contexts for the different subclasses
- * 647943 : [pango] Class global pango mutex not always used
- * 648459 : tag: exif: register common tags from tag library
- * 648466 : Ogg to LPCM transcoding fails
- * 648548 : videoscale broken with orc 0.4.13
- * 642667 : [playbin2] autoplug-factories code does not do what it claims to do
- * 642732 : [playbin2] sinks set to READY after activating groups causes bad autoplug-continue decisions
- * 646744 : libgsttag: Minor issues building gst-plugins-base with MS compiler
- * 647294 : gst-plugins-base doesn't compile with GCC 4.6
-
-API additions since 0.10.32:
-
- * gst_tag_list_to_xmp_buffer_full()
- * gst_tag_xmp_list_schemas()
- * gst_tag_xmp_writer_add_all_schemas()
- * gst_tag_xmp_writer_add_schema()
- * gst_tag_xmp_writer_get_type()
- * gst_tag_xmp_writer_has_schema()
- * gst_tag_xmp_writer_remove_all_schemas()
- * gst_tag_xmp_writer_remove_schema()
- * gst_tag_xmp_writer_tag_list_to_xmp_buffer()
- * GST_TAG_CAPTURING_EXPOSURE_COMPENSATION
- * gst_video_format_get_component_depth()
- * gst_video_format_new_template_caps()
-
-Changes since 0.10.31:
-
- * GLib requirement is now >= 2.22, gobject-introspection >= 0.9.12
- * New encodebin element
- * New encoding profile and encoding targets API in pbutils
- * audioresample: corrected buffer duration calculation to account for nonzero initial timestamp
- * audioresample: provide as much valid output ts and offset as valid input
- * audioresample: push half a history length, instead of a full history length, at end-of-stream
- so that output segment and input segment have same duration
- * decodebin2: deprecate new-decoded-pad and removed-decoded-pad signals (use "pad-added" and "pad-removed" instead)
- * multifdsink: add first and last buffer's timestamp to the stats; only keep last valid timestamp
- * oggdemux: extract more tags (vorbis comment headers, Kate)
- * oggdemux: ignore header pages when looking for keyframe; set headers on caps
- * oggdemux: fix interpretation of Theora granule position and parsing of Theora size
- * oggparse: Set DELTA_UNIT on buffers
- * playbin2: delay stream-changed messages, fixing KATE subtitle recognition
- * textoverlay: make text, xpos, ypos, color, and silent properties controllable
- * typefinding: (E)AC-3 and ISO typefinder improvements; add yuv4mpeg typefinder
- * typefinding: add "stream-format" to h264 caps, and framed=false to DTS caps
- * typefinding: assume EBML files without doctype are matroska
- * videorate: fix behaviour for frame rate cap changes
- * vorbisdec: avoid using invalid timestamps; keep timestamps when no decoded output
- * ximagesink, xvimagesink: add read-only window-width and window-height properties
- * baseaudiopay: fix timestamps on buffer lists
- * baseaudiosink: protect against ringbuffer disappearing while in a query
- * basedepay: add support for buffer lists in the depayloader
- * basertppay: use RTP base time when invalid timestamps
- * rtpbuffer: relax arrangement for RTP bufferlists
- * rtpdepayloader: add support for getting events
- * rtppayload: copy applied rate to segment
- * sdp: add method to check for multicast addresses
- * sdp: only parse TTL for IP4 addresses
- * video: add 8-bit paletted RGB, YUV9, YVU9 and IYU1 video formats
- * video: return correct component width/height for A420
-
-Bugs fixed since 0.10.31:
-
- * 619778 : oggdemux: fails on zero-length pages with Patent_Absurdity_HD_3540kbit.ogv
- * 586570 : Add GAP Flag support to audioresample
- * 623413 : pbutils: Add/Fix some media descriptions
- * 627476 : New profile library and encoding plugin
- * 629349 : [oggdemux] extract stream tags for tagreadbin and transcoding
- * 632667 : [ximagesink] added read-only properties window-width and window-height
- * 634397 : [multifdsink] [PATCH] Add the timestamp of the first and last buffer to the stats
- * 634522 : gst-visualize-m.m imports but doesn't use File::Basename
- * 635231 : baseaudiosink: protect against ringbuffer disappearing while in a query
- * 636198 : decodebin2: " removed-decoded-pad " signal never fired
- * 636769 : [appsink] Flushing property is never reset
- * 636827 : Usage of gst_caps_interset where gst_caps_can_intersect was intended?
- * 637324 : oggdemux: unable to demux Ogg files with Skeleton in push mode
- * 637377 : timeoverlay: add missing break
- * 637519 : ogg: implement packet duration query for kate streams
- * 637586 : playbin2 fails to recognize subtitle caps from katedec
- * 637735 : [encoding-profile] automatic load/save support and registry
- * 637758 : [exiftag] Generates buffers with uninitialized data during taglist- > exif buffer serialization
- * 637822 : oggdemux: allocate buffers using gst_buffer_new_and_alloc
- * 637927 : oggdemux: set headers on caps
- * 638200 : [oggdemux] fails to playback video file
- * 638276 : oggstream: when the last keyframe position is not known, do not use -1
- * 638859 : textoverlay: make misc. properties controllable
- * 638901 : [encodebin] proper element documentation
- * 638903 : [encodebin] missing-plugin support
- * 638961 : Small configure bashism 0.10.31.2
- * 639039 : gobject-introspection: GstPbutils gir scanner fails to link with gold linker
- * 639121 : oggdemux: outdated comment for gst_ogg_demux_submit_buffer()
- * 639215 : examples: Allow building with newer GTK+
- * 639730 : discoverer: Validate timeouts before processing them
- * 639755 : discoverer: Clean up callbacks in dispose()
- * 639778 : discoverer: Drop new stream tags once preroll is done
- * 639790 : [gdp] Fix metadata g_warning
- * 639747 : Please export GST_TYPE_APP_STREAM_TYPE
- * 553244 : theoraparse doesn't work at all (throws criticals and asserts)
-
-
-API added since 0.10.31:
-
- * gst_app_stream_type_get_type()
- * gst_discoverer_info_get_seekable()
- * gst_encoding_audio_profile_get_type()
- * gst_encoding_audio_profile_new()
- * gst_encoding_container_profile_add_profile()
- * gst_encoding_container_profile_contains_profile()
- * gst_encoding_container_profile_get_profiles()
- * gst_encoding_container_profile_get_type()
- * gst_encoding_container_profile_new()
- * gst_encoding_list_all_targets()
- * gst_encoding_list_available_categories()
- * gst_encoding_profile_find()
- * gst_encoding_profile_get_description()
- * gst_encoding_profile_get_format()
- * gst_encoding_profile_get_input_caps()
- * gst_encoding_profile_get_name()
- * gst_encoding_profile_get_presence()
- * gst_encoding_profile_get_preset()
- * gst_encoding_profile_get_restriction()
- * gst_encoding_profile_get_type()
- * gst_encoding_profile_get_type_nick()
- * gst_encoding_profile_is_equal()
- * gst_encoding_profile_set_description()
- * gst_encoding_profile_set_format()
- * gst_encoding_profile_set_name()
- * gst_encoding_profile_set_presence()
- * gst_encoding_profile_set_preset()
- * gst_encoding_profile_set_restriction()
- * gst_encoding_target_add_profile()
- * gst_encoding_target_get_category()
- * gst_encoding_target_get_description()
- * gst_encoding_target_get_name()
- * gst_encoding_target_get_profile()
- * gst_encoding_target_get_profiles()
- * gst_encoding_target_get_type()
- * gst_encoding_target_load()
- * gst_encoding_target_load_from_file()
- * gst_encoding_target_new()
- * gst_encoding_target_save()
- * gst_encoding_target_save_to_file()
- * gst_encoding_video_profile_get_pass()
- * gst_encoding_video_profile_get_type()
- * gst_encoding_video_profile_get_variableframerate()
- * gst_encoding_video_profile_new()
- * gst_encoding_video_profile_set_pass()
- * gst_encoding_video_profile_set_variableframerate()
- * gst_base_rtp_depayload_push_list()
- * gst_rtsp_url_decode_path_components()
- * gst_sdp_address_is_multicast()
- * gst_video_parse_caps_palette()
-
-Changes since 0.10.30:
-
- * adder: Make sure FLUSH_STOP is always sent after a flushing seek
- * alsasrc, alsasink: add "card-name" property to get the card name in addition to the device name
- * appsrc: don't override buffer caps if appsrc caps are NULL; fix element classification
- * audioclock: add a function to invalidate the clock
- * audioconvert: optimise remaining conversion code paths with Orc as well
- * baseaudiosink,baseaudiosrc: post clock-provide and clock-lost messages when going from/to READY to/from PAUSED
- * baseaudiosink: subtract the render_delay from our latency
- * decodebin2: don't add non prerolled stream to topology
- * ffmpegcolorspace: add support for A420 and fix support for 8 bit paletted RGB and IYU1
- * gnomevfsrc: set GST_PARAM_MUTABLE_READY flag on the "handle" property
- * libvisual: add latency query; only drop frames that are really too old
- * multifdsink: gdp protocol is deprecated. People should use gdppay instead
- * oggdemux: fix seeking with negative rate with skeleton; fix wrong flowreturn handling
- * pbutils: AAC profile and level detection utility functions
- * pbutils: H.264 and MPEG-4 profile and level extraction utility functions
- * pbutils: new GstDiscoverer utility API for extracting metadata and tags
- * playbin2, decodebin2: declare stable, deprecate the old playbin/decodebin
- * playbin2, uridecodebin: add property to configure ring buffer size
- * rtcpbuffer: add function to manipulation the data in RTCP feedback packets
- * rtpbuffer: add functions to add RFC 5285 header extensions to GstBufferLists
- * rtpbuffer: add function to add RTP header extensions with a two bytes header
- * rtpbuffer: add function to append RFC 5285 one byte header extensions
- * rtpbuffer: add function to parse RFC 5285 header extensions
- * rtpbuffer: add function to read RFC 5285 header extensions from GstBufferLists
- * rtpbuffer: add function to transform a GstBuffer into a GstBufferList
- * rtsp: improve rtsp timeout calculation and handling
- * sdp: add methods to convert between uri and message
- * tags: try ISO-8859-1 as second fallback in case WINDOWS-1252 is not supported
- * tags: add many more photography/capture tags
- * tags: EXIF and XMP tag handling improvements
- * textoverlay: add support for NV12, NV21 and AYUV; configurable text color and position
- * theoradec: expose telemetry properties only if libtheora was compiled with --enable-telemetry
- * theoraenc: add support for two-pass encoding; allow change of bitrate and quality on-the-fly
- * tools: standalone gst-discoverer-0.10 tool for discovering media file properties
- * typefinding: detect avc1 ftyp as video/quicktime
- * typefinding: export 3gp profile in caps
- * typefinding: detect enhanced AC-3
- * typefinding: extend AAC typefinder to detect LOAS streams
- * typefinding: fix ADTS caps stream-format detail
- * typefinding: more reliable mpeg-ts typefinding
- * uridecodebin: Only enable progressive downloading if the upstream duration in bytes is known
- * video: add gst_video_convert_frame*() utility functions
- * videorate: fixate the pixel-aspect-ratio if necessary
- * videorate: mark duplicated frames with the GAP flag
- * videoscale: add support for adding black borders to keep the DAR if necessary ("add-borders" property)
- * videoscale: Fix caps fixating if the height is fixed but the width isn't
- * videoscale: only set the PAR if the caps already had a PAR
- * videoscale: refactor using more Orc code
- * videotestsrc: new patterns: solid-color, ball, bar and smpte100
- * videotestsrc: add "foreground-color" and "background-color" properties, deprecate "colorspec" property
- * videotestsrc: add support for UYVP format, fix NV21 rendering
- * volume: use Orc to optimise many code paths
- * vorbisdec: decode pending buffers upon EOS when doing reverse playback
- * xoverlay: add set_window_handle() with guintptr argument, deprecate set_xwindow_id() which doesn't work on some platforms
- * xoverlay: allow render rectangle coordinates to be negative
-
-Bugs fixed since 0.10.30:
-
- * 628028 : [uridecodebin] Don't enable progressive downloading for live streams
- * 623846 : typefinding: add support for " enhanced ac3 " (eac3)
- * 602437 : [playbin2] [gapless] Completely broken when switching between files with audio/video only
- * 612264 : Notification needed when the first buffer is pushed by the basertppayloader
- * 615471 : [videoscale] Interlaced handling makes output worse than no interlaced handling at all
- * 616392 : videotestsrc colorspec=0/1 does not affect color-matrix in caps
- * 617314 : pbutils: Add codec-specific utility functions for AAC, H.264, MPEG-4 video
- * 617506 : [videoscale] Add support for adding black borders if necessary to keep the DAR
- * 620291 : typefindfunctions: Export 3gp profile in caps
- * 623663 : [typefinding] mpeg-ts file detected as audio/mpeg
- * 623807 : [audioclock] Add gst_audio_clock_new_full() with GDestroyNotify for the user_data
- * 623837 : typefind: only associate .webm with WebM
- * 623918 : [typefind] Extend AAC typefinder to detect LOAS stream
- * 624598 : [adder] crash in orc_sse_set_mxcsr()
- * 624656 : [videoscale] UYVY scaling broken, introduces green lines
- * 624919 : [videotestsrc] add solid color pattern
- * 624920 : [textoverlay] configurable text color and position
- * 624949 : [playbin2] declare playbin2 stable
- * 625001 : [examples] Don't use GdkDraw/GdkGC
- * 625118 : [playbin2] Race condition with EOS events in gapless mode
- * 625944 : [pbutils] GstDiscoverer - API to discover metadata and stream information
- * 626125 : [alsa] Conditional jump or move depends on uninitialised value(s)
- * 626570 : [tag] Add resolution tags
- * 626581 : [playbin2] regression: occasional deadlocks in streamsynchronizer
- * 626621 : [playbin2] streamsynchronizer regressions
- * 626629 : [ffmpegcolorspace] doesn't handle palettes any longer
- * 626718 : playback: Delay usage of GstFactoryList
- * 627203 : [alsa] alsasrc and alsasink should expose card name via property
- * 627297 : [regression] build-failure
- * 627565 : [xoverlay][win64] gulong can't hold a HANDLE
- * 627768 : add NV12 support to textoverlay
- * 627780 : GstClockOverlay re-renders string even if it hasn't changed, resulting in very high CPU usage.
- * 627924 : riff: add support for 2vuy
- * 628009 : [volume] Float processing with orc broken
- * 628400 : [videorate] does not generate buffers to fill the duration of the last frame
- * 628500 : videotestsrc: add moving color bars pattern
- * 628747 : gst-plugins-base: unable to build because of compiler warning in libggsttag
- * 629157 : Move video frame conversion from playback plugin to libgstvideo
- * 629672 : gnomevfsrsrc: " handle " property should also have the GST_PARAM_MUTABLE_READY flag
- * 629848 : build problem with current gtk+: implicit declaration of function 'gdk_draw_rectangle', GtkStyle' has no member named 'black_gc'
- * 630303 : theoraenc: Make the bitrate/quality dynamically modifiable
- * 630353 : [appsrc] Avoid losing buffers' caps
- * 630440 : ringbuffer: use g_once for type-init
- * 630443 : baseaudiosink: Add getter and setter for drift tolerance
- * 630471 : [tag] ligatures " Œ " and " œ " are not supported in freeform strings
- * 630496 : seek example: add new #define to set seek bar graininess
- * 630802 : videotestsrc.c doesn't compile in Visual Studio 2008
- * 631128 : Add methods to manipulate RFC 5285 header extensions
- * 631312 : [streamsynchronizer] Advancing segments too much
- * 631633 : [oggdemux] fix seeking with negative rate with skeleton
- * 631703 : [oggdemux] sintel ogv delay when playing
- * 631756 : Fix build with glib 2.21.3
- * 631773 : [tags] Add new exif tags: sharpness, metering mode, file/capturing source
- * 631774 : [xvimagesink] sets non-simple caps on pad_alloced buffer
- * 632167 : [oggdemux] doesn't parse/push all headers in pull mode
- * 632653 : [seek] Don't use deprecated combo box API
- * 632656 : [uridecodebin] internal decodebin2 might fail to reach PLAYING in streaming case
- * 632789 : [PATCH] tests/icles/: adapted test-colorkey.c and test-xoverlay.c to deprecation of gtk_widget_hide_all
- * 632809 : [regression] build failure in 0.10.30.2 in tools/
- * 632988 : [discoverer] gst_caps_ref() critical for substreams of unknown streams
- * 633023 : [discoverer] Add versionized gst-discoverer tool
- * 633203 : Regression: streamsynchroniser + disabled deinterlacing = no DVD menu highlights/subtitles
- * 633311 : discoverer: use specific types in getters, rename some boolean getters
- * 633336 : [discoverer] Move documentation into the correct section
- * 633455 : [rtsp] don't let the rtsp connection timeout
- * 634014 : GTK+3 is a moving target, lets not compile against it by default.
- * 634584 : decodebin2 docs should mention that " new-decoded-pad " signal may be emitted after " no-more-pads "
- * 635067 : [*decodebin*] pad template leaked
- * 635392 : Missing information on exported packages from GIRs
- * 621349 : [theoraenc] Implement two-pass encoding
- * 628488 : [theoradec] add properties to enable telemetry overlay
- * 629746 : Enumerations have incorrect names of enum values (GEnumValue.value_name)
- * 626869 : The RTP depayloader is sometimes sending partial frames down the pipeline without the DISCONT bit set
-
-API added since 0.10.30:
-
- * gst_audio_clock_invalidate()
- * gst_audio_clock_new_full()
- * gst_base_audio_sink_get_drift_tolerance()
- * gst_base_audio_sink_set_drift_tolerance()
- * gst_x_overlay_got_window_handle()
- * gst_x_overlay_set_window_handle()
- * GstXOverlay::set_window_handle()
- * gst_codec_utils_aac_caps_set_level_and_profile()
- * gst_codec_utils_aac_get_level()
- * gst_codec_utils_aac_get_profile()
- * gst_codec_utils_aac_get_sample_rate_from_index()
- * gst_codec_utils_h264_caps_set_level_and_profile()
- * gst_codec_utils_h264_get_level()
- * gst_codec_utils_h264_get_profile()
- * gst_codec_utils_mpeg4video_caps_set_level_and_profile()
- * gst_codec_utils_mpeg4video_get_level()
- * gst_codec_utils_mpeg4video_get_profile()
- * gst_discoverer_audio_info_get_bitrate()
- * gst_discoverer_audio_info_get_channels()
- * gst_discoverer_audio_info_get_depth()
- * gst_discoverer_audio_info_get_max_bitrate()
- * gst_discoverer_audio_info_get_sample_rate()
- * gst_discoverer_audio_info_get_type()
- * gst_discoverer_container_info_get_streams()
- * gst_discoverer_container_info_get_type()
- * gst_discoverer_discover_uri()
- * gst_discoverer_discover_uri_async()
- * gst_discoverer_get_type()
- * gst_discoverer_info_copy()
- * gst_discoverer_info_get_audio_streams()
- * gst_discoverer_info_get_container_streams()
- * gst_discoverer_info_get_duration()
- * gst_discoverer_info_get_misc()
- * gst_discoverer_info_get_result()
- * gst_discoverer_info_get_stream_info()
- * gst_discoverer_info_get_stream_list()
- * gst_discoverer_info_get_streams()
- * gst_discoverer_info_get_tags()
- * gst_discoverer_info_get_type()
- * gst_discoverer_info_get_uri()
- * gst_discoverer_info_get_video_streams()
- * gst_discoverer_new()
- * gst_discoverer_result_get_type()
- * gst_discoverer_start()
- * gst_discoverer_stop()
- * gst_discoverer_stream_info_get_caps()
- * gst_discoverer_stream_info_get_misc()
- * gst_discoverer_stream_info_get_next()
- * gst_discoverer_stream_info_get_previous()
- * gst_discoverer_stream_info_get_stream_type_nick()
- * gst_discoverer_stream_info_get_tags()
- * gst_discoverer_stream_info_get_type()
- * gst_discoverer_stream_info_list_free()
- * gst_discoverer_video_info_get_bitrate()
- * gst_discoverer_video_info_get_depth()
- * gst_discoverer_video_info_get_framerate_denom()
- * gst_discoverer_video_info_get_framerate_num()
- * gst_discoverer_video_info_get_height()
- * gst_discoverer_video_info_get_max_bitrate()
- * gst_discoverer_video_info_get_par_denom()
- * gst_discoverer_video_info_get_par_num()
- * gst_discoverer_video_info_get_type()
- * gst_discoverer_video_info_get_width()
- * gst_discoverer_video_info_is_image()
- * gst_discoverer_video_info_is_interlaced()
- * GST_PLUGINS_BASE_VERSION_MAJOR
- * GST_PLUGINS_BASE_VERSION_MINOR
- * GST_PLUGINS_BASE_VERSION_MICRO
- * GST_PLUGINS_BASE_VERSION_NANO
- * GST_CHECK_PLUGINS_BASE_VERSION
- * gst_plugins_base_version()
- * gst_plugins_base_version_string()
- * gst_rtcp_packet_fb_get_fci()
- * gst_rtcp_packet_fb_get_fci_length()
- * gst_rtcp_packet_fb_set_fci_length()
- * gst_rtp_buffer_add_extension_onebyte_header()
- * gst_rtp_buffer_add_extension_twobytes_header()
- * gst_rtp_buffer_get_extension_onebyte_header()
- * gst_rtp_buffer_get_extension_twobytes_header()
- * gst_rtp_buffer_list_add_extension_onebyte_header()
- * gst_rtp_buffer_list_add_extension_twobytes_header()
- * gst_rtp_buffer_list_from_buffer()
- * gst_rtp_buffer_list_get_extension_onebyte_header()
- * gst_rtp_buffer_list_get_extension_twobytes_header()
- * gst_sdp_message_as_uri()
- * gst_sdp_message_parse_uri()
- * GST_TAG_CAPTURING_SOURCE
- * GST_TAG_CAPTURING_METERING_MODE
- * GST_TAG_CAPTURING_SHARPNESS
- * GST_TAG_IMAGE_HORIZONTAL_PPI
- * GST_TAG_IMAGE_VERTICAL_PPI
- * GST_TAG_CAPTURING_FLASH_FIRED
- * GST_TAG_CAPTURING_FLASH_MODE
- * GST_TAG_CAPTURING_EXPOSURE_PROGRAM
- * GST_TAG_CAPTURING_EXPOSURE_MODE
- * GST_TAG_CAPTURING_SCENE_CAPTURE_TYPE
- * GST_TAG_CAPTURING_GAIN_ADJUSTMENT
- * GST_TAG_CAPTURING_WHITE_BALANCE
- * GST_TAG_CAPTURING_CONTRAST
- * GST_TAG_CAPTURING_SATURATION
- * GST_TAG_CAPTURING_SHUTTER_SPEED
- * GST_TAG_CAPTURING_FOCAL_RATIO
- * GST_TAG_CAPTURING_FOCAL_LENGTH
- * GST_TAG_CAPTURING_DIGITAL_ZOOM_RATIO
- * GST_TAG_CAPTURING_ISO_SPEED
- * GST_VIDEO_FORMAT_UYVP
- * GST_VIDEO_FORMAT_A420
- * gst_video_convert_frame()
- * gst_video_convert_frame_async()
- * GstTextOverlay:xpos
- * GstTextOverlay:ypos
- * GstTextOverlay:color
- * GstVideoTestSrc:solid-color
- * GstVideoTestSrc::foreground-color
- * GstVideoTestSrc::background-color
-
-API deprecated since 0.10.30:
-
- * gst_x_overlay_set_xwindow_id()
- * gst_x_overlay_got_xwindow_id()
- * GstXOverlay::set_xwindow_id()
-
-Changes since 0.10.29:
-
- * Use Orc (Optimized Inner Loops Runtime Compiler) for SIMD and
- other optimisations, and remove liboil dependency. The main goal
- for this release was to make the transition from liboil to liborc.
- Performance improvements should not be expected and will be the
- focus of future versions. liborc is an optional dependency for
- the time being, to make it possible to test and develop the very
- latest GStreamer versions on systems that don't have orc yet.
- However, without orc slow unoptimised backup code will be used
- for many performance critical code paths. Distributors are urged
- to package and ship the latest version of liborc and compile
- GStreamer modules with --enable-orc. Please do not distribute
- GStreamer packages that are not orc enabled. More information on
- the orc integration can be found in the docs/design/ directory.
- * basertpaudiopayload: Set duration on buffers; add extra frame for non-complete frame lengths
- * riff: add mappings for On2 VP8 and VP6F (On2 VP6 Flash variant)
- * video: Add support for RGB/BGR with 15 and 16 bits, and Y800 and Y16
- * xmp/exif tags: add mappings for new tags (device, geo location, image orientation)
- * adder, audioconvert, audioresample, volume: convert from liboil to orc
- * adder: rework timestamping; only accept seek-types SEEK_NONE and SEEK_SET
- * decodebin2: add "expose-all-streams" property to not expose/decode all streams
- * decodebin2: use accumulator for autoplug-sort
- * ffmpegcolorspace: add YUY2/YVYU to all RGB formats conversions
- * ffmpegcolorspace: fix conversion of packed 4:2:2 YUV to RGB and 8 bit grayscale
- * ffmpegcolorspace: fix Y16 from/to GRAY8 conversion
- * ffmpegcolorspace: fix Y42B from/to YUY2/YVYU/UYVY conversion for odd widths
- * ffmpegcolorspace: Map "Y8 " and "GREY" to "Y800" and add it to the template caps
- * ffmpegcolorspace: negotiation speed-ups
- * oggdemux: implement seeking and duration estimates when operating in push mode (http etc.)
- * oggdemux: parse Skeleton index packets for better seeking in push mode
- * oggdemux: fix granulepos->key granule calculation for Dirac video
- * oggdemux: fix EOS flow aggregation: only EOS when all streams are EOS
- * oggmux: Start a new page for every CMML buffer
- * ogg: Implement Ogg VP8 mapping
- * playbin2: add "av-offset" property to adjust audio/video sync
- * playbin2: add flag for enabling/disabling automatic deinterlacing
- * playbin2: fix race when querying duration right after preroll, by forwarding duration
- query duration during group switch if no cached duration exists
- * playbin2: if a text sink is provided, let subtitle parsing be done by decodebin2 if required
- * playbin2: set the subtitle encoding on the decodebins again
- * playsink: also expose "convert-frame" action signal and "frame" property in playsink
- * playsink: reconfigure the video chain correctly when switching from a subtitle to a non-subtitle file
- * playsink: Don't fail if subtitles are used but only audio is available and no visualizations
- * typefinding: add WebM typefinder (was in -good before)
- * typefinding: add IVF and dts typefinders, improve AC-3 and jpeg typefinding
- * typefinding: detect ISO 14496-14 files as video/quicktime not audio/x-m4a
- * uridecodebin: add all qtdemux types to downloadable types
- * uridecodebin: add the 'expose-all-streams' property from decodebin2
- * uridecodebin: Allow video/webm for progressive downloading
- * videorate, videotestsrc: fixate color-matrix, chroma-site and interlaced fields if necessary
- * videoscale: Try to keep DAR when scaling
- * videoscale: Add support for Y444, Y42B and Y41B and more gray formats
- * videoscale: Fix resampling of ARGB scanlines
- * videoscale: Try harder to keep the DAR if possible
- * videoscale: Use passthrough mode if width and height are not changed
-
-Bugs fixed since 0.10.29:
-
- * 621428 : [playbin2] ghostpad with arbitrary getcaps func leads to not working srt subtitles
- * 371108 : videoscale sucks at basic mathematics when it comes to PAR
- * 512740 : unit test failures if compiling against installed core with installed plugins-base also present
- * 605100 : GNOME Goal: Remove deprecated glib symbols
- * 610866 : [playbin2] Don't fail if there are subtitles and audio but no video
- * 614872 : [tag] Add basic exif support
- * 614942 : playbin2: " text-sink " is supposed to handle raw subtitle data?
- * 615783 : reworked timestamping in adder
- * 616396 : [playbin2] might fail a duration query immediately following PAUSED state
- * 616422 : playsink might not handle reconfiguring after a text enabled file correctly
- * 616557 : [videorate] Add support for video/x-raw-gray
- * 617636 : [rtsp] uses unicode characters in date string
- * 617855 : [oggdemux] Fails to play LAC2010 videos
- * 617868 : [decodebin2] Option to not expose/decode all streams
- * 618324 : rtp payloader don't put the duration on their output buffers
- * 618392 : [avi-demux] Gstreamer does not support 1x1 or 1xN avi files
- * 618625 : lock priv mutex in appsrc when setting caps
- * 619090 : [uridecodebin] caps negociation fail
- * 619102 : [PATCH] WebM typefinder
- * 619310 : [videorate] negotiation issue, tries to set unfixed caps on pad
- * 619396 : gstreamer does not seek http streams of Ogg Vorbis and Flac audio files.
- * 620136 : Orc integration
- * 620140 : [gio] report out-of-space errors
- * 620211 : gst-plugins-base gets confused by dual-QT system
- * 620279 : [playsink] expose 'frame' property and move 'convert-frame' action from playbin2
- * 620342 : [baseaudiosink] Allocate and free the clock in state changes
- * 620412 : [video] Incomplete support for 15 and 16 bit RGB and BGR formats
- * 620441 : [video] Add support for Y800 and Y16 formats
- * 620500 : totem won't do progressive download for flv videos
- * 620720 : typefinding: Mark ISO 14496-14 files as video/quicktime
- * 620939 : [oggdemux] No support for Skeleton 4.0 streams
- * 621071 : [playbin2] no playback with fakesink instead of appsink as text-sink
- * 621161 : autoplug-sort default callback is run last - overwrite user supplied callback result
- * 621190 : video sink drops buffers if it's preceded by ffmpegcolorspace, videoscale and a capsfilter
- * 621509 : [xmptag] Uses uninitialized variable
- * 621572 : [videoscale] Adds horizontal green lines in bilinear mode
- * 622696 : ffmpegcolorspace: Speed up caps nego by using simpler caps
- * 622807 : [decodebin2] Doesn't recover properly after an error
- * 622944 : Require automake 1.9 or newer for $(builddir)
- * 623003 : Major problems with calls to gst_util_uint64_scale()
- * 623176 : riff: matroska file with FLV4 FOURCC fails to play
- * 623218 : oggdemux: Handle errors from _get_next_page in _do_seek.
- * 623233 : GstNetBuffer initialization isn't thread safe.
- * 623318 : [playbin2] If source setup fails the old uridecodebin is kept and breaks future playback
- * 623375 : [ffmpegcolorspace] Invalid memory accesses with odd widths/height during subsampling
- * 623384 : [ffmpegcolorspace] Doesn't write last pixel for odd widths in packed 4:2:2 YUV- > RGB conversion
- * 623418 : [ffmpegcolorspace] Fix packed 4:2:2 YUV to 8 bit grayscale conversion for odd widths
- * 623530 : [ffmpegcolorspace] Stripy pattern with videotestsrc ! ffmpegcolorspace ! ximagesink
- * 623583 : [playbin2] regression: DVD playback broken
- * 624266 : [playbin2] Internal uridecodebin are not properly removed in READY= > NULL
- * 547603 : [playbin2] add deinterlacing support
-
-API added since 0.10.29:
-
- * GST_VIDEO_FORMAT_v308
- * GST_VIDEO_FORMAT_Y800
- * GST_VIDEO_FORMAT_Y16
- * GST_VIDEO_FORMAT_RGB16
- * GST_VIDEO_FORMAT_BGR16
- * GST_VIDEO_FORMAT_RGB15
- * GST_VIDEO_FORMAT_BGR15
- * gst_tag_image_orientation_to_exif_value ()
- * gst_tag_image_orientation_from_exif_value ()
- * gst_tag_list_to_exif_buffer ()
- * gst_tag_list_to_exif_buffer_with_tiff_header ()
- * gst_tag_list_from_exif_buffer ()
- * gst_tag_list_from_exif_buffer_with_tiff_header ()
-
-Changes since 0.10.28:
-
- * video: add support for color-matrix and chroma-site fields in video caps and selected elements
- * video: Add support for 8-bit and 16-bit grayscale formats
- * typefinding: add AAC profile, level, channels and rate to ADTS caps
- * tags: add basic xmp metadata support
- * gio, gnomevfs: invert ranks of gio and gnomevfs elements: gio is prefered now, gnomevfs has been deprecated
- * riff: add mapping for On2 VP62 and VP7 and add some more MPEG4 fourccs
- * playsink: Don't fail if there are subtitles and audio but no video
- * oggdemux: map old FLAC mapping correctly
- * alsa: handle disappearing of sound device whilst in use more gracefully
- * playbin: Only unref the volume element on dispose and when a new audio sink is set
- * build: build plugin, example and libs directories in parallel if make -jN is used
- * uridecodebin/playbin2: we can handle avi in download mode too
- * rtsp: handle closed POST socket in tunneling, ignore unparsable ranges, allow for more ipv6 addresses
- * audiopayload: add property to control packet duration
-
-Bugs fixed since 0.10.28:
-
- * 615647 : xvimagesink could miss initial expose
- * 423086 : vorbisdec introduces timestamp discontinuity at the end
- * 601315 : [uridecodebin] No download buffering for AVI files
- * 609539 : xmp metadata support in tag library
- * 609801 : [volume] Use sample accurate property values if a controller is used
- * 610248 : don't poll geometry for every frame
- * 610249 : [xoverlay] add set_render_rectangle() methods
- * 610866 : [playbin2] Don't fail if there are subtitles and audio but no video
- * 611702 : playbin2: Warning: g_object_set: assertion `G_IS_OBJECT (object)' failed with custom text sink
- * 612223 : [base(audio)sink] hangs in _wait_eos
- * 612312 : typefind: Post AAC profile in caps
- * 612552 : Examples and tests don't compile with -DGSEAL_ENABLE
- * 612783 : Warning when compiling gstsubparse.c with MSVC
- * 612845 : [typefindfunctions] crash in strncasecmp() on windows
- * 612968 : Add On2 VP62 and VP7 support in riff-media
- * 613093 : Improper boundary condition handling in videoscale.
- * 613198 : ximagesink memory leak
- * 613248 : [audiopayload] add property to control packet duration
- * 613281 : [PATCH] autogen.sh: Don't call configure with --enable-plugin-docs
- * 613387 : [gio] compiler warning with GLib 2.18, breaks build with -Werror
- * 613403 : docs: gst_x_overlay_handle_events() lacks Since: tag
- * 613589 : typefind: Export AAC level in caps
- * 613591 : rtspsrc doesn't parse negative port numbers
- * 613690 : [xmp] refactoring to 1-n tag mappings
- * 613809 : [oggdemux] flac: file does not play locally
- * 614288 : Setting playbin volume has no effect the second time around
- * 614545 : gstalsasrc mixer task spins 100% CPU when USB sound card is removed
- * 614622 : Trying to compile and it blows up at seek
- * 614764 : Compile breaks on Mac OS 10.5.8 on new jsseek example
- * 615572 : Buffer Leak in audiorate during fill process
- * 615697 : Problems with Makefile
- * 615789 : [ximagesink] gst_ximagesink_xwindow_update_geometry: assertion `xwindow != NULL' failed
- * 616545 : [ffmpegcolorspace] Crashes when converting Y41B with some width/height combinations
-
-API added since 0.10.28:
-
- * gst_x_overlay_set_render_rectangle()
- * gst_tag_list_from_xmp_buffer()
- * gst_tag_list_to_xmp_buffer()
- * gst_video_format_is_gray()
- * gst_video_parse_caps_chroma_site()
- * gst_video_parse_caps_color_matrix()
- * GST_VIDEO_CAPS_GRAY8
- * GST_VIDEO_CAPS_GRAY16
- * GST_TYPE_RTSP_LOWER_TRANS
- * gst_rtsp_lower_trans_get_type()
-
-Changes since 0.10.27:
-
- * Ogg/Dirac fixes
- * build: really dist qtgv-xoverlay.h header file needed by overlay examples this time
- * rtspconnection: fix handling of x-server-ip-address
- * alsasrc fixes
-
-Bugs fixed since 0.10.27:
-
- * 610832 : qtgv-xoverlay.h header file missing in the tarball
- * 611900 : [oggdemux] Incorrect parsing of Dirac headers
-
-Changes since 0.10.26:
-
- * playbin2,decodebin2: lots of fixes for missing plugin installation
- * playbin2, playsink, subtitleoverlay: Set subtitle encoding properly
- * videorate: Improve upstream negotiation
- * oggdemux: use the chain begin_time instead of our counter
- * oggdemux: mark skeleton streams correctly
- * oggdemux: theora PAR of 0:N, N:0 or 0:0 is allowed and maps to 1:1
- * typefinding: detect stm module format
- * ffmpegcolorspace: add conversions from all ARGB formats to AYUV and back
- * theoradec: Fix chroma copying for 4:2:2
- * tcpclientsrc,tcpserversrc: Fix handling of closed sockets
- * examples,build: dist header file for the Qt graphics view example
- * playsink: Reset the sink's state to NULL before unreffing it unless it's the same instance again
- * rtspconnection: make sure not to dereference NULL username or password
- * appsrc: Update segment duration and post a duration message if the duration changes
- * vorbisdec: also support ivorbis tremor decoder
- * rtsp: fail gracefully on bad Content-Length headers
- * rtsp: ignore \n and \r as the first line
-
-Bugs fixed since 0.10.26:
-
- * 610449 : codec autodetection does not always work
- * 608025 : [videorate] fails at upstream negotiation
- * 608309 : [appsrc] Should request new data before the queue is empty
- * 608417 : rtspsrc problem with \n and \r as first line
- * 609063 : [vorbisdec] also support integer vorbis decoder (tremor) library implementation
- * 609314 : typefind: Typefind does not handle .stm module format
- * 609423 : [appsrc] gst_app_src_set_size() should update duration and post a duration message
- * 610005 : [oggdemux] regression: bad seek granularity
- * 610268 : [rtsp] NULL pointer reference in gstrtspconnection
- * 610310 : [playbin2] Subtitle encoding property has no effect
- * 610329 : [theoradec] doesn't copy all chroma lines for 4:2:2
- * 610379 : [playbin2] doesn't play if text flag is unset and media has text subtitles
- * 610386 : [tcpserversrc] Doesn't send EOS when socket is closed
- * 610672 : overlay examples are now inconsistent and broken
- * 610832 : missing header file in the tarball
- * 611225 : [oggdemux] doesn't preroll big_buck_bunny_427x240.indexed.ogg in push mode
- * 611227 : [oggdemux] no duration or seeking in local big_buck_bunny_427x240.indexed.ogg in pull mode
- * 604131 : Totem can no longer open Matroska files that hold ASS subtitles
-
-API added since 0.10.26:
-
- * appsrc::min-percent property
- * GST_RIFF_TAG_JUNQ
-
-Changes since 0.10.25:
-
- * playbin2: make about-to-finish signal work for raw sources (e.g. audio CDs)
- * playbin2: fix handling of the native audio/video flags
- * playbin2: add flag to enable decodebin buffering
- * playbin2: make subtitle error handling more robust and ignore late errors
- * playbin2: improve subtitle passthrough in uridecodebin
- * playbin2: new subtitleoverlay element for generic subtitle overlaying
- * playbin2: proxy notify::volume and notify::mute from the volume/mute
- elements (or audio sink)
- * playbin2: don't stop completely on initialization errors from subtitle
- elements; instead disable the subtitles and play the other
- parts of the stream
- * decodebin2: rewrite autoplugging and how groups of pads are exposed
- * uridecodebin: add use-buffering property that will perform buffering on
- parsed or demuxed media.
- * GstXOverlay: flesh out docs and add example for use with Gtk+ >= 2.18
- * libgsttag: add utility functions for ISO-639 language codes and tags
- * oggdemux: use internal granulepos<->timestamp mapper and make oggdemux
- more like a 'normal' demuxer that outputs timestamps
- * oggdemux: seeking improvements
- * subparse: add qttext support
- * ffmpegcolorspace: prefer transforming alpha formats to alpha formats
- and the other way around
- * libgstvideo: add functions to create/parse still frame events.
- * theoraenc: make the default quality property 48.
- * videotestsrc: add pattern with out-of-gamut colors
- * theora: port to 'new' theora 1.0 API; make misc. existing properties
- have no effect (quick, keyframe-mindistance, noise-sensitivity,
- sharpness, keyframe_threshold); those either never worked or
- aren't needed/provided/useful any longer with the newer API
- * typefinding: misc. performance improvements and fixes
- * baseaudiosink: make drift tolerance configurable
-
-Bugs fixed since 0.10.25:
-
- * 507131 : GStreamer does not play short ogg sounds
- * 583376 : [typefind] Detects MP3 as h264
- * 344013 : [oggdemux] use parsers to suck less
- * 598114 : build overwrites interfaces/interfaces-enumtypes.h with wrong enumtypes
- * 344706 : [playbin] problem changing subtitles and language
- * 350748 : [ffmpegcolorspace] ffmpeg colorspace should prefer RGBA over RGB
- * 499181 : audiorate inserting samples (due to rounding errors ?)
- * 524771 : Can't seek in YouTube videos
- * 537050 : [playbin2] QOS event problems
- * 542758 : [playbin2] Hangs in PLAYING forever if caps are not a subset of pad template caps
- * 549254 : [playbin/decodebin] Doesn't handle pads that are added much later than the other(s) correctly
- * 563828 : [decodebin2] Complains about loops in the graph when demuxer output requires another demuxer
- * 568014 : oggdemux/theoradec doesn't play last video frame
- * 570753 : [playbin] Support subtitle renderers additional to subtitle parsers
- * 574289 : [decodebin2] race in state change to PAUSED
- * 577326 : tcpclientsrc stops working if set to PLAYING, PAUSED and PLAYING again
- * 579394 : [playbin2] deadlock with wavpack files: type_found - > analyze_new_pad - > no_more_pads
- * 584441 : [playbin2] if suburi preroll fails with error, playback should continue
- * 584987 : [playbin2] [gapless] Fire a track-changed message on track change.
- * 585681 : Subtitle selector doesn't work
- * 585969 : [playbin2] [gapless] Position/Duration information mismatch on track change
- * 587704 : " GstDecodeBin2: This appears to be a text file " error when playing files from a samba share
- * 591625 : [alsasrc] odd timestamping on start
- * 591662 : [playbin2] can't handle both text subtitles and subpictures
- * 591677 : Easy codec installation is not working
- * 591706 : [playbin2] Support of files with subtitle subpicture streams
- * 594729 : theora: Convert to libtheora 1.0 API
- * 595123 : [playbin2] Should hide the difference between subtitles and subpictures
- * 595401 : gobject assertion and null access to volume instance in playbin
- * 595427 : avoid x event thread if not needed
- * 595849 : Fix Y41B strides in videotestsrc and gstvideo
- * 596159 : rtspsrc hangs when connecting over http tunneled rtsp
- * 596694 : [typefind] Detects quicktime as mp3
- * 596774 : Speed up subtitle display after seek/switch
- * 596981 : [audioresample] Compilation failure due to warning about use of %lu for guint64 variable
- * 597537 : [streamvolume.c]The cube root function is not defined in Microsoft's CRT
- * 597539 : [gststrpconnection.c] 'close' is not defined in Microsoft's CRT
- * 597786 : [tag] enhance gst_tag_freeform_string_to_utf8 to handle 16-bit Unicode
- * 598288 : [decodebin2] Plays a wav file but issues an error
- * 598533 : [decodebin2] Post element message with the stream topology on the bus
- * 598936 : DKS subtitle format
- * 599105 : [baseaudiosink] Remove pulsesink < 0.10.17 hack after gst-plugins-good release
- * 599154 : RtpAudioPayload can send out buffers that are not exact multiple of the frame size
- * 599266 : Requires restart after installing codecs
- * 599471 : uridecodebin: Store unused decodebin2 instances for further usage.
- * 599649 : Support for frame-based subtitles using playbin2 and subparse
- * 600027 : [playbin2,playsink] Should notify about volume/mute changes
- * 600370 : [subtitleoverlay] New element to overlay video with subtitles in every supported format
- * 600469 : gdpdepay: Clear adapter on flush and state change
- * 600479 : Deadlock when playing movie with subtitles
- * 600726 : [queue2] implement buffering-left argument to buffer messages
- * 600787 : playbin2 has a problem with Ogg stream with " info "
- * 600945 : silence buffers at start reusing pulsesrc
- * 600948 : [uridecodebin] Improve all raw caps detection on pads
- * 601104 : [cddabasesrc] always plays first track if device is specified
- * 601627 : theoradec breaks timestamps
- * 601772 : gst-rtsp-server crashing : bug fixed
- * 601809 : seek example doesn't work with csw
- * 601942 : Add a still-frame event to libgstvideo
- * 602000 : [playbin2] [gapless] Does state change PLAYING- > PAUSED- > PLAYING while it should stay in PLAYING
- * 602225 : Can't play another movie after using subtitles
- * 602790 : New oggdemux parsers break theora/vorbis playback
- * 602834 : [ffmpegcolorspace] does un-necessary conversion from RGB to ARGB
- * 602924 : Text subtitle rendering regression
- * 602954 : [oggdemux] can't get first chain on ogg/theora stream
- * 603345 : [playbin2] textoverlay refcount issues in git
- * 603357 : [subparse] support for QTtext
- * 605100 : GNOME Goal: Remove deprecated glib symbols
- * 605219 : Freezes nearly always when switching Audio CDs
- * 605960 : new examples require GTK 2.18
- * 606050 : Implement ptime support
- * 606163 : textoverlay: Ignore zero framerate
- * 606687 : playbin2: can't see video after setting native flags
- * 606744 : Totem fails to play video file: " Can't display both text subtitles and subpictures. "
- * 606926 : Vorbis: Implement Proper Channel Orderings for 6.1 and 7.1 Configurations
- * 607116 : [playbin2] no 'about-to-finish' signal with audio CDs
- * 607226 : Disallow setting the playbin uri property in state > = PAUSED
- * 607381 : GST_FRAMES_TO_CLOCK_TIME() GST_CLOCK_TIME_TO_FRAMES() should round result
- * 607403 : rtpaudiopayload: ptime is in milli-seconds, convert to nanosecs
- * 607569 : Playing a chained ogg stream from HTTP pauses or freezes between songs
- * 607652 : segfault with an ogg annodex file
- * 607848 : typefind wrong classifies mp4 file as mp3
- * 607870 : [oggdemux] OGM parsing broken
- * 607926 : [oggdemux] regression with certain chained ogg stream
- * 607929 : [oggdemux] regression: headers pushed twice at the beginnign of each stream
- * 608167 : [decodebin2] Doesn't push out full topology
- * 608179 : caps filter appearing after adder results in deadlock
- * 608446 : [playbin2] post an error message if no URI is set
- * 608484 : [playbin2] problem with redirect and reset to READY
- * 608699 : [oggdemux] memory leak while demuxing
- * 609252 : [theoradec] Doesn't handle unknown pixel aspect ratio properly
- * 596078 : Playbin2 takes ref of audio-/video-sink parameter
- * 596183 : decodebin2: Rewrite autoplugging and how groups of pads are handled
- * 601480 : [playback] Update factory lists not only after going back to NULL
- * 596313 : gstv4lelement.c:168: error: ‘client’ may be used uninitialized in this function
- * 606949 : [playbin2] verify type of volume property before using it
-
-API added since 0.10.25:
-
- * gst_rtcp_sdes_name_to_type()
- * gst_rtcp_sdes_type_to_name()
- * gst_tag_get_language_name()
- * gst_tag_get_language_codes()
- * gst_tag_get_language_code_iso_639_1()
- * gst_tag_get_language_code_iso_639_2B()
- * gst_tag_get_language_code_iso_639_2T()
- * gst_video_event_new_still_frame()
- * gst_video_event_parse_still_frame()
-
-Changes since 0.10.24:
-
- * Add per-stream volume controls
- * Theora 1.0 and Y444 and Y42B format support
- * Improve audio capture timing
- * GObject introspection support
- * Improve audio output startup
- * RTSP improvements
- * Use pango-cairo instead of pangoft2
- * Allow cdda://(device#)?track URI scheme in cddabasesrc
- * Support interlaced content in videoscale and ffmpegcolorspacee
- * Many other bug fixes and improvements
-
-Bugs fixed since 0.10.24:
-
- * 595401 : gobject assertion and null access to volume instance in playbin
- * 563828 : [decodebin2] Complains about loops in the graph when demuxer output requires another demuxer
- * 591677 : Easy codec installation is not working
- * 588523 : smarter sink selection in playbin2
- * 590146 : adder regressions
- * 321532 : [cddabasesrc] Support device setting in cdda:// URI
- * 340887 : add pangocairo textoverlay plugin.
- * 397419 : [oggdemux] ogm video with subtitles stuck on first frame
- * 556537 : [PATCH] typefind: more flexible MPEG4 start code recognition
- * 559049 : gstcheck.c:76:F:general:test_state_changes_* failure: GST_IS_CLOCK(clock) assertion fails
- * 567660 : [API] need a stream volume interface for sinks that do volume control
- * 567928 : Make videorate work with a live source
- * 571610 : [playbin] Scale of volume property is not documented
- * 583255 : [playbin2] deadlock when disabling visualisations
- * 586180 : RTSP improvements
- * 588717 : [oggmux] gst_caps_unref() warning if not linked downstream
- * 588761 : [videoscale] Needs special support for interlaced content
- * 588915 : audioresample's output offset counter's initialization could maybe be improved
- * 589095 : [appsrc] clarify documentation on caps and linkage
- * 589574 : [typefind] incorrect sdp file detection
- * 590243 : [videoscale] Claims to support MAX width/height
- * 590425 : Slaved alsasrc clock with slave-method=re-timestamp not usable for RTP audio
- * 590856 : [decodebin2] triggers assertion failure on NULL caps
- * 591207 : totem does display the following subtitle srt file.
- * 591357 : gst-plugins-base git won't build due to warning in gstrtspconnection.c
- * 591577 : [playbin2] Incorrect error message string
- * 591664 : [playbin2] after seeking, srt subtitles don't resync correctly
- * 591934 : timestamp drift in audioresample
- * 592544 : Remove regex.h check
- * 592657 : [appsink] Blocks after entering on pause state
- * 592864 : deadlocks from recent inputselector/streamselector change
- * 592884 : [playbin2] g_object_get increases refcount by 2 and therefore leaves memleak
- * 593035 : gdp doesn't preserve fields of the buffers put into the caps' streamheader
- * 593284 : basertppayloader takes time in instance init
- * 594020 : Totem don't play videos from ssh remote host
- * 594094 : Playback Error playing Midi file
- * 594136 : [alsasink] Regression from 0.10.23 -- element reuse doesn't work
- * 594165 : [theoraenc] Implement support for new formats
- * 594256 : improved slave-skew resynch mechanism
- * 594258 : missing break in rtcpbuffer
- * 594275 : Add cast to navigation to fix compiler warning
- * 594623 : Expose playsink as a fully-fledged element
- * 594732 : parse error
- * 594757 : build fails due to warning in gstbasertppayload.c
- * 594993 : [introspection] pkg-config file madness
- * 594994 : [streamvolume] Add get_type function to the documentation
- * 595454 : [cddabasesrc] uri format change breaks rhythmbox
- * 545807 : [baseaudiosink] audible crack when starting the pipeline
-
-API added since 0.10.24:
-
- * gst_rtsp_connection_create_from_fd()
- * gst_rtsp_connection_set_http_mode()
- * gst_rtsp_watch_write_data()
- * gst_rtsp_watch_send_message()
- * GstBaseRTPPayload::perfect-rtptime
- * GstBaseRTPAudioPayload::gst_base_rtp_audio_payload_flush()
- * GstVideoSinkClass::show_frame()
- * GstVideoSink:show-preroll-frame
- * GST_MIXER_TRACK_READONLY
- * GST_MIXER_TRACK_WRITEONLY
- * GstStreamVolume interface
-
-Changes since 0.10.23:
-
- * Recognise Kate subpicture subtitles
- * Support progressive download in playbin2
- * GIO improvements
- * Add buffer-list support in appsink
- * Add gaussian-noise mode to audiotestsrc
- * bump cdparanoia req to 0.10.2 and improve caching
- * Improve audio source base class
- * Add frame-by-frame stepping and examples
- * Extend stream-probing in decodebin2
- * Many RTSP improvements
- * support for PGS subpictures
- * adder improvements
- * Add Y444, v210, v216 formats
- * implement preset interface in vorbisenc, theoraenc, oggmux
- * Improve libvisual visualisation timestamp tracking
- * playbin2 enhancements: custom audiosink, subpictures, cdda
- * Improvements in textrender
- * Support raw YUV 4:2:2 and SIREN in RIFF
- * Add 4:2:2 and 4:4:4 support to theoradec
- * Many other bug-fixes and improvements
-
-Bugs fixed since 0.10.23:
-
- * 510417 : [gio] make non-experimental
- * 513373 : [PATCH] [gstvorbistag] Preserve cover art in Ogg/Vorbis tags
- * 529300 : [giosink] [PATCH] Allow overwrite
- * 531035 : [cdparanoia] Should depend on LGPL'd version of the libra...
- * 567997 : [patch] add allow-pull-scheduling property to audio sinks
- * 576552 : [subparse] post GST_TAG_SUBTITLE_CODEC tags
- * 577637 : [playbin2] expose temp-location property
- * 579692 : mp3_type_find is over-optimistic
- * 580318 : [tagdemux] drops tag events from upstream
- * 581460 : [baseaudiosrc] Reusing audio source leads to null timesta...
- * 581571 : ARGB and alignment added to textrender
- * 582021 : autogen: libtoolize must be called before aclocal
- * 582749 : uridecodebin caps property not implemented yet
- * 582819 : multifdsink: add num-fds property
- * 583867 : gdpdepay + identity cause failed assertions
- * 584020 : [playbin2] inadvertently resets configured audio/video sinks
- * 584686 : [playbin2] Need {audio,video,text}-tags-changed signals
- * 585197 : [subparse] fails to detect subrip subtitles with fewer th...
- * 585758 : Remove deprecated GTK+ symbols
- * 585970 : gst_audioringbuffer_get_type is not thread safe
- * 585994 : gst-rtsp-message doesn't support " Timestamp " filed
- * 586331 : [cdparanoia] expose cd cache size parameter
- * 586356 : [playbin2] use private copy of input-selector as long as ...
- * 586519 : white Gaussian noise would be useful in audiotestsrc
- * 587080 : rtsp fails to compile - doesn't see some ws2tcpip functions
- * 587278 : Support for GstBufferList in appsink
- * 587676 : Call tzset() before localtime_r(), in e.g. gst-plugins-ba...
- * 587695 : Patches to add stream-status messages audio elements
- * 587896 : " No stream given yet " error from giostreamsrc
- * 587980 : gstchannelmix.c: protect debug code with GST_DISABLE_GST_...
- * 588078 : [playbin2] Fails to go to READY again after an error
- * 588205 : Pipeline with giostreamsrc will not enter playing state
- * 588550 : build failure in git, missing gstinterfaces-0.10
- * 588551 : queue2: download buffering fixes
- * 588724 : [vorbisdec] empty encoder string causes GStreamer
- * 588746 : [audiotestsrc] Make sure tags are properly serialized in ...
- * 588747 : [adder] Serialize incoming in-band events (tags) in the d...
- * 588748 : [adder] Check dataflow consistency in unit tests
- * 589075 : [playbin2] changing volume doesn't work after stream rest...
- * 589581 : typefinder: recognise more Kate subtitle categories
- * 589622 : Cannot use both playbin and input-selector
- * 589663 : gstreamer asserts in gstaudiofilter
- * 589797 : alsasrc does not set GstAlsaSrc- > handle to NULL after snd...
- * 590470 : [typefinding] certain flac-in-ogg files not detected any ...
- * 536313 : [cdda] Remove sha1 copy once we depend on glib-2.16
- * 579642 : [oggdemux] handle broken ogg/vorbis files better
- * 582528 : playbin2 Audio CD playback broken since
- * 583318 : Assertion from within playbin2
- * 585079 : undefined references to gst_adapter_* functions in schro
- * 585708 : [adder] Wrong handling of flushing seeks
- * 588218 : Siren in .wav support
- * 586920 : rtsp: needs < netinet/in.h > on FreeBSD
-
-API added since 0.10.23:
-
- * GstNetAddress::gst_netaddress_to_string()
- * Add gst_rtsp_watch_queue_data()
- * playbin2: Add {audio,video,text}-tags-changed signals
- * Add gst_color_balance_get_balance_type()
- * Add gst_mixer_get_mixer_type()
-
-Changes since 0.10.22:
-
- * New navigation API to support DVD playback
- * playbin2 improvements
- * RTSP extensions to allow extra headers and options
- * Replace audioresampler with speexresample based code
- * Support interlacing flags in the gstvideo library
- * Support new RIFF formats
- * Improve typefinding
- * Support more frame formats in videoscale
- * Many other bug-fixes and improvements
-
-Bugs fixed since 0.10.22:
-
- * 577637 : [playbin2] expose temp-location property
- * 580120 : [playbin2] unit test fails
- * 478512 : [alsamixer] volume control slider not working
- * 574962 : rhythmbox crash in flac_type_find
- * 564139 : Documentation of TCP plugins
- * 577436 : xvimagesink should use xcontext- > depth and not count bits...
- * 350311 : [playbin2] support for subpicture subtitles
- * 378094 : Enable pango elements to handle UYVY
- * 543591 : Gnonlin can not play theora streams
- * 553295 : [riff] fuzzed AVI file causes segfault
- * 565105 : Gstreamer does not change from READY back to PAUSED in sa...
- * 565777 : [riff] unrecognised video fourcc 0x10000002 for mpeg2 in avi
- * 566661 : [typefind] Fall back to file extension using uri query
- * 567255 : [riff] doesn't detect codec_id 0x706d as AAC (amongst other)
- * 567636 : [pbutils] Missing plugins code shouldn't ask for the same...
- * 567740 : bogus warning in decodebin2?
- * 568482 : linking problems in gst-plugins-base
- * 569655 : [ffmpegcolorspace] Add UYVY422 to GRAY8 conversion function
- * 570142 : Documentation is broken for uridecodebin
- * 570356 : aac typefinder failure
- * 570768 : [ximagesink] wrong mouse pointer position if output windo...
- * 570832 : Add flags to enhance mixer interfaces
- * 571009 : [tagdemux] WMA file with id3v2 tag causes assertion to fail
- * 571147 : [ffmpegcolorspace/videotestsrc] Add support for packed/pl...
- * 572577 : [playbin2] deadlock on shutdown
- * 572872 : [ffmpegcolorspace] Add YVYU colorspace
- * 572993 : [subparse] broken libregex dependency on Windows
- * 573165 : Generate additional export files for gstreamer app plugin
- * 573528 : Wrong format modifier in gstgiobasesink.c
- * 573529 : In gstrtspconnection.c some functions are called with wro...
- * 574293 : [decodebin2] deadlock on shutdown
- * 574319 : Missing HAVE_PROCESS_H in win32/common/config.h
- * 574447 : gstadder.c: line 904: error C2036: 'gpointer' : unknown size
- * 574939 : [typefinding] flac typefinder mis-typefinds PDFs as flac ...
- * 575550 : srt subtitle file keeps playbin2 from playing
- * 575638 : kissfft copyright
- * 575649 : [oggdemux] duration query in time format returns true wit...
- * 576019 : On Windows queue2 can't write files longer than 2-4 GiB, ...
- * 576142 : [vorbisenc] Non-header output buffers have NULL caps
- * 576180 : [playbin2] Uses unref'd audiosink volume if using gconfau...
- * 576586 : [alsamixer] gnome-sound-properties freeze
- * 577054 : [videoscale] Not valgrind clean
- * 577709 : Review new navigation API
- * 577827 : [appsink] Have appsink new_buffer-callback return GstFlow...
- * 578583 : [PATCH] multifdsink doesn't handle sync-method=latest-key...
- * 578656 : Implement upstream GstForceKeyUnit events in theoraenc
- * 579129 : pkgconfig: appsrc/appsink can not be linked to uninstalled
- * 579130 : app: expose trivial type macros
- * 579192 : gst_rtcp_packet_get_type should not assert on packet content
- * 579203 : baseaudiosink: unparenting the ringbuffer in NULL causes ...
- * 579267 : [rtspconnection] g_async_queue_new_full() is GLib-2.16 AP...
- * 579463 : [cddabasesrc] [cdparanoiasrc] no longer emits discid
- * 579668 : audioresample fails to build with --disable-gst-debug
- * 579734 : [playbin] raw_decoding_mode seems to be set unconditionally
- * 579912 : [decodebin2] multiqueue is too small in time (interleave ...
- * 580470 : [audioresample] causes pipelines to go out of sync and be...
- * 580952 : [audioresample] bad quality/pops compared to plughw
- * 581727 : [playbin2] make playsink go to PAUSED async
- * 569682 : playbin2 leaks request pad from input selector
- * 580020 : [vorbisenc] causes buffers to be out of segment if new se...
- * 562794 : rtspsrc fails to create a socket on Win32 sometimes.
- * 567396 : playbin2: DECODE_BIN_LOCK occasionally called twice withi...
- * 567982 : " queued_bytes " field isn't updated while flushing the que...
- * 571299 : [appsink] Handoff callback API
- * 574443 : rtsp win32 - forgotten variable
- * 574516 : [typefind] add typefinder for photoshop .psd files
- * 574964 : gst_app_src_end_of_stream(), mutex on error return
- * 575256 : rtspsrc fails to resolve hostnames
- * 575588 : decodebin2 deadlock
- * 576187 : [playbin2] Stalls video sink when disabling subtitles in ...
- * 576188 : [playbin2] Reusing a playbin2 instance with visualization...
- * 576190 : [playbin2] Deadlock when reusing playbin2 after an error
- * 577288 : " Internal playbin error " when seeking to the end of files
- * 577610 : RTCP feedback messages support in GstRTCPPacket
- * 577794 : [playbin2] leaks elements set through properties
- * 578118 : [multifdsink] add option to not resend the streamheader w...
- * 578506 : Pipeline with alsasrc and alsasink cannot change state ba...
- * 578942 : Missing RTSP headers related to Windows Media extension.
- * 580271 : videorate: fails to clear discont flag on duplicated buffers
- * 580649 : uridecodebin: bug on documentation published in website
-
-API added since 0.10.22:
-
- * GstRTSP::gst_rtsp_options_as_text()
- * GstRTSPMessage::gst_rtsp_message_take_header()
- * GstRTSPRange::gst_rtsp_range_to_string()
- * New Navigation interface commands, queries and messages
- * gst_rtsp_channel_new()
- * gst_rtsp_channel_unref()
- * gst_rtsp_channel_attach()
- * gst_rtsp_channel_queue_message()
- * gst_rtsp_connection_accept()
- * GstAppSink::gst_app_sink_set_callbacks()
- * GST_VIDEO_FORMAT_YVYU,GST_VIDEO_BUFFER_TFF,GST_VIDEO_BUFFER_RFF,GST_VIDEO_BUFFER_ONEFIELD
- * GST_MIXER_FLAG_HAS_WHITELIST,GST_MIXER_FLAG_GROUPING,GST_MIXER_TRACK_NO_RECORD,GST_MIXER_TRACK_NO_MUTE,GST_MIXER_TRACK_WHITELIST
- * GstAppSrc::emit-signals
- * GstAppSrc::gst_app_src_set_emit_signals()
- * GstAppSrc::gst_app_src_get_emit_signals()
- * GstAppSrc::gst_app_src_set_callbacks()
- * RTSP::gst_rtsp_connection_get_url()
- * GstRTSPLowerTrans::GST_RTSP_LOWER_TRANS_HTTP
- * RTSP:gst_rtsp_connection_set_tunneled()
- * RTSP:gst_rtsp_connection_is_tunneled()
- * RTSP::gst_rtsp_connection_set_ip()
- * RTSP::gst_rtsp_connection_get_tunnelid()
- * RTSP::gst_rtsp_connection_do_tunnel()
- * RTSP::gst_rtsp_watch_reset()
-
-IMPORTANT NOTES
-
-1) Please note that decodebin2 and playbin2 API included in this release is
-still considered unstable and WILL change in future releases. At this stage,
-only developers or early adopters should consider using decodebin2 or playbin2
-API embodied in their signals and properties.
-
-Changes since 0.10.21:
-
- * Require gettext 0.17
- * Replace audioresample with speexresample from -bad
- * Support new formats in RIFF: uncompressed RGB, WMA lossless, VP6
- * Move libgstapp and elements from -bad
- * Support color-key setting and probing for Xv properties
- * Improve typefinding for various formats
- * Extend audio sinks for pull-mode operation
- * Support for more subtitle formats
- * More development on decode2bin and playbin2
- * RTP and SDP fixes
- * Many bug fixes and improvements
-
-Bugs fixed since 0.10.21:
-
- * 562163 : theoraenc likely ignoring segments
- * 562258 : rtspsrc element takes long time to error out if the addre...
- * 561789 : [volume] deadlocks with a controller attached
- * 554533 : [xvimagesink] allow setting colorkey if possible
- * 567511 : colorkey in xvimagesink gets reset when element is reused
- * 116051 : libresample doesn't handle > factor of 2 rate conversion
- * 346218 : [audioresample] doesn't do anti aliasing
- * 385061 : [audioresample?] investigate high CPU usage
- * 456788 : [subparse] can't handle UTF-16 charset encoded subtitle.
- * 525807 : [vorbisenc] vorbisenc has problems with a gnlsource that ...
- * 546955 : gstoggmux EOS handling issue
- * 549417 : [audioresample] unit test fails on 64bit linux
- * 549510 : audioresample doesn't negotiate ideal caps
- * 552237 : UTF-16 srt confuses gstreamer, misdetected as mp3
- * 552559 : Implementation of SLAVE_SKEW in baseaudiosrc
- * 552569 : audioresample producing strange sized buffers
- * 552801 : audioconvert can overflow with big audio buffers
- * 554879 : Add ability to specify format for date/time display in Gs...
- * 555257 : Doesn't display srt subtitles saved with BOM
- * 555319 : add FFV1 fourcc to riff-media
- * 555607 : subrip subtitles typefind too strict
- * 555699 : [PATCH] theoradec: prefer container's pixel aspect ratio ...
- * 556025 : build failure in tests/icles
- * 556066 : Last byte of FLAC image buffer chopped off
- * 557365 : subparse check fails
- * 558124 : [PLUGIN-MOVE] Move speexresample as audioresample2 to -base
- * 559111 : ALSA sink hangs on USB audio device unplug while playing
- * 559478 : does not play windows media streams correctly
- * 559567 : `gst_base_audio_sink_sync_latency' should call `gst_base_...
- * 561436 : videorate element add image/jpeg to caps template
- * 561734 : playbin2 additions
- * 561780 : Playbin2 should work without volume too
- * 561924 : oggdemux hangs when given corrupt input via non-seekable ...
- * 562270 : build without gdk fails
- * 563143 : ximagesink/xvimagesink : _alloc_buffer returns non-clean ...
- * 563174 : Implement gst_rtcp_packet_remove
- * 563508 : [rgvolume] Unit test fails with passthrough assertions
- * 563718 : Theora check out of date
- * 563904 : GNOME Goal: Clean up GLib and GTK+ includes
-
-API added since 0.10.21:
-
- * clockoverlay::time-format
- * GstRingBuffer:gst_ring_buffer_activate()
- * GstRingBuffer:gst_ring_buffer_is_active()
- * GstRingBuffer:gst_ring_buffer_convert()
- * Add GST_TYPE_BASE_AUDIO_(SRC|SINK)_SLAVE_METHOD to the public API
- * gst_netaddress_get_address_bytes()
- * gst_netaddress_set_address_bytes()
-
-Changes since 0.10.20:
-
- * Continue playbin2 development
- * Ogg improvements - CELT support, skeleton fixes
- * DVD subpicture support
- * Improved audio dithering random number generator
- * xvimagesink/ximagesink fixes
- * Vorbis encoding and decoding fixes
- * Recognise Kate subtitle streams
- * Many bug-fixes and enhancements
-
-Bugs fixed since 0.10.20:
-
- * 537380 : [gnomevfssrc] Doesn't handle short reads properly
- * 538656 : xvimagesink support for autofill/colorkey property
- * 540334 : Build fails without X in tests/examples/seek
- * 528299 : Multiple GstMixerTracks with the same label cause problem...
- * 530068 : Ogg Streams with Skeleton and Granulepos > 0 do not work(...
- * 537009 : playbin2 silly typo breaks signals
- * 537045 : decodebin2 sometimes emits 'drained' multiple times
- * 537599 : [oggdemux] skeleton streams not skipped in ogg
- * 537889 : [xvimagesink] colorbalance is bad
- * 538232 : vorbisenc/vorbisdec don't work with a live source
- * 538663 : gdppay memleak in gst_gdp_pay_reset
- * 540215 : decodebin does not insert a queue for raw data type
- * 540351 : [avidemux] Doesn't know about Duck DK4 ADPCM
- * 540497 : ffmpegcolorspace is returning wrong size
- * 541358 : cross mingw32 gcc: getaddrinfo is not in ws2_32.dll befor...
- * 544306 : rtspsrc debug=1 segfaults with some libc
- * 548898 : GStreamer-CRITICAL errors on seeking beyond stream borders
- * 548913 : vorbisenc being picky about rounding errors in timestamps
- * 549062 : Video devices aren't updated on subsequent probing.
- * 549814 : [typefind] add application/pdf typefinder
- * 550582 : [oggdemux] KATE streams not recognised
- * 550638 : [typefind] Recognize some jpeg2k file types
- * 550656 : recognize TrueSpeech in wavparse
- * 550729 : gst-plugins-base won't compile with " -pedantic " option
- * 552960 : tagdemux asserts and aborts on truncated files
- * 553244 : theoraparse doesn't work at all (throws criticals and ass...
-
-API added since 0.10.20:
-
- * Add "index" property to GstMixerTrack to differantiate between
- multiple mixer tracks with the same label.
-
-Changes since 0.10.19:
-
- * RTP improvements
- * Support digest auth for RTSP
- * Additional documentation
- * Support DSCP QoS in multifdsink
- * Add NV12/NV21 video buffer layouts
- * Video scaling now bilinear by default
- * Support more than 8 channels in audio conversions
- * Channel mapping fixes for audioconvert
- * Improve tmplayer and sami subtitle support
- * Support 1x1 pixel buffers for videoscale
- * Typefinding improvements for MPEG2, musepack
- * Ogg/Dirac mapping updated in oggmux
- * Fixes in ogg demuxing
- * audiosink synchronisation and slaving fixes
- * Support muting of the audio in playbin by selecting -1 as the audio stream
- * Work done on playbin2 and uridecodebin
- * Improvements in the experimental GIO plugin
- * decodebin fixes
- * Handle GAP buffers in some places
- * Various other leak and bug-fixes
-
-Bugs fixed since 0.10.20:
-
- * 526794 : [giosrc] totem doesn't work with some gvfs backends
- * 510417 : [PLUGIN-MOVE] Move gio to gst-plugins-base
- * 509125 : crash in CD Player: - playing CD - lowering/...
- * 517813 : [audioconvert] make gap aware
- * 302798 : [playbin] add mute property
- * 342294 : Setting playbin property current-audio=-1 also stops the ...
- * 398033 : [audioconvert] support more than 8 channels
- * 419351 : [avi/a52dec] AV synchronization problems
- * 467911 : [subparse] sami parser update
- * 469933 : multifdsink IPv6 and diffserv TOS/TC markup
- * 506659 : [textoverlay] rendering error when using non-standard widths
- * 512333 : [gstvorbistag] Retrieve Ogg/Vorbis cover art as image met...
- * 512382 : [playbin] race condition when pausing/playing multiple in...
- * 518037 : pbutils-enumtypes.c is not included in win32/vs6/libgstpb...
- * 521761 : gstaudioclock frozen the clock value until reaches latest...
- * 522401 : gdpdepay doesn't validate payload CRCs
- * 523993 : playbin2 blocks after a while when listening to a radio s...
- * 524724 : [PATCH] [baseaudiosrc] buffer-time and latency-time do no...
- * 525665 : Crash on Ogg/Vorbis with chain=NULL
- * 525915 : [streamheader] Unit test fails with " gst_adapter_peek: as...
- * 526173 : [typefinding] fails to detect mpeg video stream whereas m...
- * 529018 : gst_ogm_parse_stream_header creates fraction value with w...
- * 529500 : [videotestsrc] support for NV12 and NV21
- * 529546 : [Playbin] Memory leak in streaminfo handling
- * 530068 : Ogg Streams with Skeleton and Granulepos > 0 do not work(...
- * 530531 : [typefinding] bad read in mpeg_video_stream_type_find
- * 530719 : gst_video_calculate_display_ratio fails when playing Ogg ...
- * 530962 : [subparse] parses only every second line of TMPlayer subt...
- * 532454 : [NV12/NV21] videotestsrc and ffmpegcolorspace don't play ...
- * 533087 : GstRTSPTransport kept opaque in docs
- * 533817 : [audioconvert] Can't use default 7 channel layout / only ...
- * 534071 : Gdppay memleak
- * 534331 : race in decodebin when changing states while the internal...
- * 535356 : vorbisdec doesn't support 8 channels
- * 536475 : gdppay memleak and possible crash
- * 536521 : Refcounting errors in playbin
- * 536874 : Build failure on windows
- * 532166 : [ffmpegcolorspace] support NV12 format
- * 533617 : [audioconvert] Produces silence when converting 1/2 chann...
- * 536848 : [giosrc] Doesn't handle short reads properly
- * 536849 : [giosrc] Very slow doing any playback
- * 518082 : [alsamixer] playback volumes overwritten by capture volum...
- * 435633 : [PATCH] videorate not (fully) segment aware; causes frame...
- * 532364 : tcpclientsrc broken in 0.10.19
- * 533075 : gst_rtp_buffer_compare_seqnum doesn't do what it says
- * 533265 : [cddabasesrc] Sound Juicer cut a sector when ripping a track
-
-API additions since 0.10.20:
-
- * decodebin2::sink-caps property
- * giosrc::file property
- * giosink::file property
- * gst_base_audio_src_set_slave_method()
- * gst_base_audio_src_get_slave_method()
- * GstAudioClock::gst_audio_clock_reset()
- * GstBaseAudioSrc:actual-buffer-time property
- * GstBaseAudioSrc:actual-latency-time property
- * gst_audio_check_channel_positions()
- * add gst_tag_image_data_to_image_buffer()
- * add gst_tag_list_add_id3_image()
- * add GST_TAG_IMAGE_TYPE_NONE enum value
-
-Changes since 0.10.18:
-
- * Handle EAGAIN when polling sockets in rtspconnection
-
-Changes since 0.10.17:
-
- * Experimental GIO plugin
- * Continued playbin2 development
- * RTP fixes
- * Better network element support on Windows
- * Various other bug-fixes and improvements
-
-Bugs fixed since 0.10.17:
-
- * 509637 : [API] [basertpaudiopayload] add _set_samplebits_options()
- * 510229 : [gnomevfssrc] HTTPS support
- * 511478 : [rtpbuffer] add gst_rtp_buffer_set_extension_data function
- * 511810 : [RTSP] Uses MT-unsafe gmtime() function
- * 512899 : [alsa] gstalsasink.c:527: warning: 'snd_pcm_sw_params_set...
- * 513167 : Fix compiler warning due to disabled signals in mixertrac...
- * 514307 : [playbin] warning in nautilus, volume element can't be cr...
- * 514623 : Ogg Theora video slow
- * 514937 : Correct initialization of hints in is_multicast_address()
- * 515654 : xvimagesink doesn't build with --disable-xshm
- * 516246 : [alsasink] handle negative delay from snd_pcm_delay
- * 517420 : typefind: add h264 elementary stream discovery
- * 517991 : problems with configure file depending on GCC compiler
- * 518039 : libgstrtsp MSVC 6.0 compile error
- * 518162 : [subparse] handle italic text starting with " / " with Micr...
- * 518940 : [playbin2] make _get_*_tags() match vfuncs prototype in c...
- * 519906 : [API] add GstMixerOptions::get_values vfunc
- * 519916 : [API] add mixer-changed and options-list-changed messages
- * 520523 : [API] Unreviewed changes to ringbuffer API
- * 521743 : libgstnetbuffer.def exports not up to date
- * 522625 : [video] gst_video_format_parse_caps() broken for RGBA for...
- * 523054 : gstbasesrc crashes when called from typefind helpers
- * 511825 : [RTSP] compiler warning on FreeBSD
- * 520300 : [alsasrc] provide-clock=false messes up buffer durations
-
-API added since 0.10.17:
-
- * GstRTPBuffer:gst_rtp_buffer_set_extension_data()
- * add GST_VIDEO_FORMAT_Y41B and GST_VIDEO_FORMAT_Y42B.
- * add GstMixerOptions::get_values vfunc (#519906)
- * add gst_mixer_options_list_changed(), gst_mixer_mixer_changed() and
- gst_mixer_message_parse_options_list_changed(). Fixes #519916.
- * gst_base_rtp_audio_payload_set_samplebits_options()
- * GstNetBuffer::gst_netaddress_equal
-
-Changes since 0.10.16:
-
- * Work-around ABI breakage due to unfortunate use of the
- GST_DISABLE_DEPRECATED macro
- * Export 2 missing functions needed for bindings in the win32 build
- * Initialise the GstRingBuffer GType from a thread-safe context
-
-Bugs fixed since 0.10.16:
-
- * 511825 : [RTSP] compiler warning on FreeBSD
- * 513018 : crash in Volume Control: I typed my password at t...
- * 512334 : g_critical() when using GstAudioFilter & GST_DEBUG
-
-Changes since 0.10.15:
-
- * Handle newer Theora granule-pos semantics
- * Introducing first alpha version playbin2 - the upcoming successor to
- playbin
- * Fixes in playbin handling of stream-switching
- * New API for uniform handling of raw-video format buffers.
- * Improvements for RTSP/RTP handling
- * RIFF lib additions for VC-1 and AVC1 fourccs
- * Many other bug-fixes and improvements
-
-Bugs fixed since 0.10.15:
-
- * 506132 : Review of changes in video/video.h
- * 320984 : [oggdemux] cannot handle multiple chains
- * 373011 : [playbin] throws error when switching off subtitles
- * 436756 : Intermittent crashes in Pidgin in audioclock g_type_class...
- * 462740 : [streamselector] patch to improve default stream selection
- * 486840 : [alsamixer] use _all variants when setting the mixer
- * 497964 : theoraenc test fails
- * 498228 : gst-plugins-base-0.10.15 does not compile on FreeBSD (Gen...
- * 499697 : Provide better pkg-config files
- * 502497 : [subparse] SubRip subtitles starting from 0 not recognised
- * 503440 : The control sockets used by gstrtspconnection.c are never...
- * 503930 : [cdda] warning: 'eos' may be used uninitialized in this f...
- * 506928 : [alsamixer] add " PCM " as master fall back for cards that ...
- * 508138 : [decodebin] does not error out if pad activation fails
- * 509762 : missing file in win32/MANIFEST
- * 511274 : gst_rtp_buffer_get_extension_data is returning FALSE when...
- * 496731 : [PATCH] xvimagesink leaks memory if initialization fails
- * 496761 : [PATCH] RTSP message leaks memory when uninitialized
- * 500763 : SIGSEGV while playing ogg audio file
-
-API additions since 0.10.15:
-
- * New GstVideoFormat API and helper functions in libgstvideo
- * gst_base_audio_sink_set_provide_clock()
- * gst_base_audio_sink_get_provide_clock()
- * gst_base_audio_sink_set_slave_method()
- * gst_base_audio_sink_get_slave_method()
- * gst_base_audio_src_set_provide_clock()
- * gst_base_audio_src_get_provide_clock()
-
-Changes since 0.10.14:
-
- * RTP/RTSP/RTCP/SDP support improved
- * New FFT support library libgstfft, based on Kiss FFT
- * New formats supported in volume and audiotestsrc
- * Fixes in audiorate and videorate
- * Audio capture fixes
- * Playbin and decodebin fixes
- * New tagdemux base class for ID3/APE style tag readers
- * Fix a nasty crash in the X sinks on shutdown
- * New tags supported
- * Add support for multichannel WAV files.
- * Preserve channel layout information when up/down-mixing.
- * Many bug-fixes and improvements
-
-Bugs fixed since 0.10.14:
-
- * 475395 : decodebin2 leaks request-pads
- * 475451 : [decodebin2] leaks ghostpad
- * 378770 : [xvimagesink] race condition in event thread?
- * 407282 : [decodebin2] autoplug-sort signal has GList ** parameter
- * 430677 : [audioconvert] does not preserve channel positions when f...
- * 442654 : [volume] controller bypassed by default
- * 445529 : [volume] support for 24/32-bit audio/x-raw-int
- * 446766 : return code for gst_base_rtp_payload_audio_handle_event()
- * 451970 : Subparse requires HTML parser
- * 453650 : [audiobasesrc] two alsasrcs do not work in one pipeline
- * 459334 : [textoverlay] expose pango line alignment property
- * 459585 : [basertpdepayload] api without namespace
- * 460422 : [audiotestsrc] Add support for float and double output
- * 462805 : [alsa] compilation fails with gcc 4.2
- * 462979 : Add 'silent' property to GstTimeOverlay
- * 463215 : [audioconvert] compile errors
- * 464320 : [PATCH] gst-plugins-base-0.14 does not build for win32
- * 464666 : [playbin] QT trailer hangs in preroll with decodebin2
- * 464690 : Add connection-speed property to uridecodebin element
- * 465015 : [playbin] Not removed probes causes deadlocks in streamin...
- * 465028 : some warnings with mingw
- * 467667 : GST_FRAMES_TO_CLOCK_TIME() and GST_CLOCK_TIME_TO_FRAMES()...
- * 468129 : [basertpaudiopayload] event handler returns the wrong value
- * 468619 : New library gstfft: FFT library for integer and float typ...
- * 470456 : [API] add gst_missing_*_installer_detail_new()
- * 470766 : [ssaparse] line breaks in SSA subtitle parser
- * 471067 : Make the SDP code useable for generating SDP descriptions
- * 471194 : [rtpbuffer] RTP headers are wrong for win32
- * 473097 : [baseaudiosink] gstreamer-properties hangs when testing s...
- * 474384 : gstrtsp-enumtypes.c and .h needed for win32
- * 474880 : [xvimagesink] [ximagesink] leaking buffer caps reference
- * 475731 : rtspconnection is able to read incomplete messages
- * 483620 : All Rtp buffers are discarded -- gst_rtp_buffer_get_payl...
- * 484989 : memleak, not unrefed caps for gstbasertppayload.c
- * 489010 : Please change default channel order for WAVE_EXT-less .wa...
- * 491722 : [playbin] regression: crash with external subtitles
- * 492098 : [GstFFT] Broken scaling
- * 492114 : Build issues on Windows/MSVC
- * 492306 : compilation errors with MinGW
- * 492813 : Missing symbols in libgstrtp.def
- * 493986 : Build issues on Windows (missing symbols)
- * 494346 : pre-release vs6 patch
- * 496548 : Including malloc.h breaks macos build
- * 496724 : DSW file references non-existent DSP files
- * 464079 : audiotestsrc doesn't respond to conversion queries properly
- * 442065 : floatcast.h includes config.h and might break other apps
- * 466717 : gst_event_new_new_segment_full:assertion `start < = stop' ...
- * 485753 : Decodebin2 deadlocks when nulling pipeline during typefind
- * 464028 : Move connection-speed from playbin to playbasebin
-
-API added since 0.10.14:
-
- * GstTagDemux base class for simple tag demuxers
- * GstBaseAudioSrc::provide-clock property
- * gst_rtcp_ntp_to_unix()
- * gst_rtcp_unix_to_ntp()
- * gst_rtp_buffer_get_header_len()
- * gst_rtp_buffer_get_extension_data()
- * gst_rtp_buffer_compare_seqnum()
- * gst_rtp_buffer_ext_timestamp()
- * gst_rtcp_packet_sdes_copy_entry()
- * gst_install_plugins_supported()
- * gst_missing_*_installer_detail_new() convenience API
- * gst_rtsp_connection_poll()
- * GstTextOverlay::line-alignment property
-
-Changes since 0.10.13:
-
- * Audio dither and noise-shaping when reducing bit-depth
- * RTSP and SDP helper libraries added
- * Experimental buffering element "queue2" now supports pull-mode
- and file-based buffering.
- * Support for more 32-bit video pixel layouts
- * Various fixes and improvements
-
-Bugs fixed since 0.10.13:
-
- * 380625 : [x*imagesink] add 'handle-expose' property
- * 385527 : oggmux sometimes gets DELTA flag on output wrong near start
- * 402076 : videoscale 4-tap method broken for downscaling
- * 437169 : [xvimagesink] add property to disable Xv double-buffering
- * 441264 : queue2 support to do buffering on a file
- * 442553 : [v4lsrc] doesn't output segments in GST_FORMAT_TIME
- * 442557 : [videorate] doesn't handle latency queries
- * 442944 : Audiotestsrc can overflow on seeks
- * 444523 : [queue2] Pull mode support
- * 444630 : Compilation error with fsseko (from gstqueue2.c) -- unabl...
- * 445505 : [queue2] It does not work in pull mode with oggdemux
- * 446551 : [queue2] Buffering is not working properly if it is set t...
- * 446572 : [queue2] Division by zero
- * 446972 : warning when compiling gstoggdemux.c
- * 449156 : Regression in CVS for decodebin2
- * 450875 : Missing files in po/POTFILES.in
- * 451707 : [tag] UTF-8 in ID3v1 tag not correctly decoded
- * 451908 : [ffmpegcolorspace] regression: doesn't accept GST_VIDEO_C...
- * 454264 : Playbin fails to " play " image url after a movie url
- * 456656 : [API] Addition of audio buffer clipping function to gstaudio
- * 460978 : gst_audio_buffer_clip outputs warnings
- * 152864 : [PATCH] GstAlsaMixer doesn't support signals
- * 360246 : [audioconvert] Optionally apply dithering
- * 394061 : Add support for Subviewer subtitles
- * 420326 : Base payloader class has wrong property types and ranges
- * 451145 : [vorbisdec] errors out on 0-sized packets
- * 459204 : [PATCH] [playbin] gst_play_base_bin_get_streaminfo_value_...
-
-API added since 0.10.13:
-
- * RTSP and SDP libraries added
- * gst_rtsp_base64_decode_ip
- * Add buffer clipping function gst_audio_buffer_clip for raw audio
- buffers. Fixes #456656.
- * gst_mixer_get_mixer_flags
- * gst_mixer_message_parse_mute_toggled
- * gst_mixer_message_parse_record_toggled
- * gst_mixer_message_parse_volume_changed
- * gst_mixer_message_parse_option_changed
- * GstMixerMessageType
- * GstMixerFlags
-
-Changes since 0.10.12:
- * Many fixes and improvements
- * RTP and RTCP support improved
-
-Bugs fixed since 0.10.12:
-
- * 339838 : [audioconvert] support floats with non-native endianness
- * 393975 : closing x/xvimagesink window crashes gst-launch
- * 405072 : [API] add gst_tag_freeform_string_to_utf8()
- * 413799 : [subparse] add support for MPL2 format
- * 414645 : GstMixerTrack should make untranslated label available
- * 420079 : [audioconvert] Uses biased rounding which results in dist...
- * 420578 : [subparse] add more colour map in sami parser
- * 421834 : videorate breaks on dimension changes
- * 423051 : Vorbis tags of type double use locale-dependent formatting
- * 423055 : Verify ReplayGain vorbistag processing in libs/tag testsuite
- * 425455 : Decodebin2 leaks pads
- * 426250 : GstPlayBaseBin leaks streaminfo objects
- * 428187 : Rtp base depayloader class doesn't send new_segment after...
- * 431672 : gst_base_rtp_audio_payload_push() should take object of i...
- * 432362 : [ximagesink] doesn't build if XShm is not available
- * 432755 : [videorate] leaks buffer if flow != OK
- * 432984 : [baseaudiosrc] misleading warning message when dropping s...
- * 433888 : [theoradec] does not generate a perfect stream
- * 436562 : Theoradec doesn't work well with gnonlin
- * 438840 : [theoradec] does not compile with old version of libtheora
- * 440997 : [gstriff] Doesn't handle width!=depth files with audio/x-...
- * 441295 : audioconvert doesn't build on VS6
- * 442024 : regression in playbin buffering
- * 350299 : [playbin] " Internal data flow error " opening movie with s...
- * 410039 : totem crashed with SIGSEGV in new_decoded_pad_full()
- * 340842 : do latency calculation for live sources
- * 341078 : RB does not play beyond initially downloaded podcast file
- * 414496 : [id3demux, id3v2mux] Add support for GST_TAG_MUSICBRAINZ_...
-
-API additions since 0.10.12:
-
- * add gst_tag_freeform_string_to_utf8()
- * GstRTPBuffer::gst_rtp_buffer_default_clock_rate()
- * GstBaseAudioSink::slave-method property
- * add "min-ptime" property to RTP base audio payloader
- * gst_base_rtp_audio_payload_push()
- * gst_base_rtp_audio_payload_get_adapter()
- * GstMixerTrack::untranslated-label property
-
-Changes since 0.10.11:
-
- * New API for on-demand plugin installation
- * Xv thread-safety and configuration enhancements
- * decodebin2 improvements
- * Support more raw audio format conversions
- * Improvements in Ogg support
- * AudioFilter base class ported to 0.10
- * Fixes for subtitles
- * Latency/live-playback support for Alsa
- * Lots of bug fixes and improvements
-
-Bugs fixed since 0.10.11:
-
- * 398721 : No video in .ogm files with decodebin2
- * 339837 : [audioconvert] support for 64-bit float audio
- * 341524 : [decodebin] can't handle decoders with always src pads wi...
- * 352069 : Add de.po German translation
- * 363379 : [oggmux] doesn't detect EOS on all sinkpads
- * 378436 : [oggdemux] rhythmbox crash on fast clicking on rating in ...
- * 380342 : Totem does not play mp3 files when lyrics are present
- * 383195 : [cddabasesrc,basertpaudiopayload] compile errors with gcc...
- * 383198 : totem crashed to gst_xvimagesink_update_colorbalance
- * 384008 : [xvimagesink] accesses - > xwindow outside locks
- * 384060 : gst_xoverlay_set_xwindow_id() causing lockups with x(v)im...
- * 387138 : x input events processing in sinks with xoverlay interfac...
- * 390063 : Documentation typo
- * 390076 : add xv adaptor and port properties in xvimagesink element.
- * 391365 : [oggdemux] internal stream error on OggFlac
- * 392070 : [vorbis] GST_TAG_LOCATION not mapped
- * 392393 : [API] add libgstbaseutils library for missing plugins mes...
- * 396042 : mpeg4 video typefinder loops endlessly on quicktime redirect
- * 396835 : audioconvert/audioresample combination causing buffer of ...
- * 397673 : [patch] XIOError caught in x[v]imagesink.c
- * 397810 : [typefinding] .vob file: could not determine type of stream
- * 398110 : [theoraenc] GLib failed to allocate 3080991032 bytes on g...
- * 399340 : Crash in the oggdemux plugin when trying to play a specia...
- * 401029 : [playbin] rapidly changing visualisation freezes
- * 401072 : Move libgimme-codec helper functions to GStreamer
- * 402505 : visualisations don't work for some samplerates
- * 407811 : decodebin2 hang on HD clip
- * 409683 : Crash with Decodebin2
- * 410396 : not reading " DATE " tags from Flac files
- * 410963 : Fails to build with -z defs
- * 357503 : [suparse] wrong timing with microdvd subtitles
- * 393310 : [pango] localtime_r does not exist in MinGW
- * 397207 : Test failure w/ HP-UX 11.11 & native compiler
- * 399948 : [textoverlay] leaks upstream events if textpad unlinked
- * 403963 : GstAudioFilter base class broken
- * 404512 : [videoscale] floating point exception on 1x1 video
- * 405020 : [alsa] probing the device-name doesn't seem to work corre...
- * 408278 : [videorate] memory leak
- * 410772 : Crash copying a GstNetBuffer
- * 401118 : [visual] error if width not a multiple of 4
- * 405451 : [alsasink] deadlocks when disconnecting USB Sounddevice
-
-API additions since 0.10.11:
-
- * GstAudioFilter
- * GST_VIDEO_SINK_CAST()
- * gst_pb_utils_add_codec_description_to_tag_list()
- * gst_pb_utils_get_codec_description()
- * gst_pb_utils_get_source_description()
- * gst_pb_utils_get_sink_description()
- * gst_pb_utils_get_decoder_description()
- * gst_pb_utils_get_encoder_description()
- * gst_pb_utils_get_element_description()
- * gst_pb_utils_init()
- * gst_install_plugins_context_new()
- * gst_install_plugins_context_set_xid()
- * gst_install_plugins_context_free()
- * gst_install_plugins_async()
- * gst_install_plugins_sync()
- * gst_install_plugins_return_get_name()
- * gst_install_plugins_installation_in_progress()
- * gst_missing_uri_source_message_new()
- * gst_missing_uri_sink_message_new
- * gst_missing_element_message_new
- * gst_missing_decoder_message_new
- * gst_missing_encoder_message_new
- * gst_missing_plugin_message_get_installer_detail
- * gst_missing_plugin_message_get_description
- * gst_is_missing_plugin_message
-
-Bugs fixed since 0.10.10:
-
- * 360552 : [riff] [avi] extracts non-UTF8 metadata
- * 365501 : [x/xvimagesink] race condition when creating first image ...
- * 339366 : [playbin] hangs if suburi file type cannot be determined
- * 355914 : libvisual causes xvimagesink: assertion `GST_CAPS_REFCOU...
- * 363118 : gst_riff_create_video_caps() should also store variant in...
- * 363607 : xvimagesink xwindow_draw_border() slowness
- * 336301 : [playbin] can't handle RTSP source
- * 337026 : oggmux doesn't set EOS properly
- * 337031 : vorbisdec outputs too much data
- * 340049 : New BaseRTPAudioPayloader class to -base
- * 348264 : Theora encoding, Ogg muxing don't handle discontinuities
- * 354773 : xvimage assumes that XV_COLORKEY can be set in RGB888 format
- * 355917 : libvisual plugin is broken
- * 355935 : multifdsink doesn't allow setting maximums (soft, hard) i...
- * 357038 : [ffmpegcolorspace] RGBA handling broken
- * 357215 : [playbin] buffering notification not quite right yet
- * 357289 : [riff] riff parser can't detect aac audio stream
- * 357404 : [playbin] Linking can fail silently
- * 357531 : [subparse] problem if markup is not closed
- * 357577 : [playbin] regression: buffering still images broken
- * 357591 : Avoid compiler warning with uclibc and -Werror
- * 357613 : XvStopVideo in xvimagesink
- * 357800 : [libvisual] doesn't pass audio data to libvisual 0.4.0 co...
- * 359580 : tcpserversink and dataprotocol assert for multipart streams
- * 361095 : Fixes compiling with forte: warning clean up (part 3)
- * 361456 : [basertppayload] Memory leak
- * 361634 : sink- > ringbuffer NULL in BaseAudioSink's setcaps()
- * 361984 : [subparse] doesn't accept .srt file that doesn't start wi...
- * 366334 : [PATCH] Windows vs8 fixes
- * 368273 : Using the remove signal on multifdsink is not threadsafe
- * 368310 : include file gstbasertpaudiopayload.h not included for r...
- * 369482 : [typefind] MPEG system streams get recognized as mp3 files
- * 370092 : [PATCH] Decodebin v2 : Implementation
- * 377183 : regression: no eos when playing ogg vorbis files
- * 381219 : bad debugging code left in audiorate
- * 382223 : [decodebin] more delayed linking
- * 382269 : Typefind detects mpeg video clip as audio/mpeg
- * 335635 : Add an Ogg/Vorbis retagging element
- * 341681 : [textoverlay] flickering with continuously timestamped text
- * 342228 : [alsa] Recognize " Front " as a Master channel
- * 357330 : [subparse] some sami parser minor but enhanced patch
- * 357532 : [gsttag] vorbistag doesn't handle dates that include time...
- * 359237 : [typefinding] doesn't recognize XML files shorter than 25...
- * 362845 : [subparse] add support for tmplayer format
- * 357977 : [videorate] new segment start is not respected
- * 364812 : [PATCH] oggmux release pad does not remove pad
- * 364856 : pngenc stride problems
- * 372507 : Mac build fixes
-
-API added since 0.10.10:
-
- * playbin::queue-min-threshold property.
- * GstVideoOrientation interface
- * gst_base_rtp_depayload_push_ts
- * gst_base_rtp_depayload_push
- * Add dropped_buffers to multifdsink's get-stats GValueArray
- * gst_ring_buffer_commit_full
-
-Changes since 0.10.9:
-
- * New elements: gdppay, gdpdepay
-
-Bugs fixed since 0.10.9:
-
- * 343787 : The adder cannot handle when multiple elements tries to l...
- * 336075 : ALSA emu10k1 mixer tracks are wrongly classified as playb...
- * 349105 : crash with playbin and resizing screen
- * 342494 : [v4l] Query " device-name " even if device is not open
- * 342680 : [adder] seeking with multiple ogg files fails to work
- * 345188 : [alsa] can't handle more than 8 channels
- * 347091 : converting vorbis comments to GstTagLists is lossy
- * 348157 : Changed " Change Device " menu behaviour in gnome-volume-co...
- * 348916 : [typefind] add multipart/x-mixed-replace typefinder
- * 350157 : [riff] riff parser can't detect dts audio stream
- * 350655 : [oggdemux] should process seeking queries
- * 350900 : [adder] should not clamp floating point values
- * 351426 : API: add gst_tag_parse_extended_comment
- * 351502 : g_value_set_string leaks
- * 351742 : [vorbisenc] discontinuity detection too sensitive, might ...
- * 353658 : [videotestsrc] doesn't round strides correctly for YVYU
- * 354594 : multifdsink doesn't work reliably with sync-method = 'nex...
- * 351790 : [ogmparse] crash parsing video stream on x86-64
- * 140139 : [avidemux] can't play broken avi with ogg (not vorbis) au...
- * 347783 : [PLUGIN-MOVE] GDP elements should be moved
- * 347918 : Internal data flow error in udpsrc
- * 349656 : jitterbuffer in GstBaseRtp fails to handle rtp seqnum rol...
- * 350784 : element alsamixer doesn't respect asoundrc
- * 351308 : [netbuffer] build fails with gkt-doc critical warnings
- * 353234 : audiorate preserves DISCONT on buffers
- * 353912 : Add cmml caps to oggmux
-
-API added since 0.10.9:
-
- * gst_rtp_buffer_get_payload_subbuffer()
- * gst_tag_parse_extended_comment()
- * GstPlayBin::connection-speed
- * GstTheoraParse::synchronization-points
- * GST_AUDIO_CHANNEL_POSITION_NONE
-
-Changes since 0.10.8:
-
- * Parallel installability with 0.8.x series
- * Threadsafe design and API
- * Subtitle fixes
- * Support for images in tags
- * Playback improvements
- * Gnomevfssrc now supports burn:// uris
- * Videoscale now supports more RGBA formats
- * Multifdsink improvements
- * Testsuite can now generate coverage information
-
-Bugs fixed since 0.10.8:
-
- * 347296 : Problems with clocks on alsasrc hangs the application
- * 347295 : [vorbisdec] Pushes before being initialized
- * 329798 : [playbin] doesn't always give correct error message for m...
- * 342085 : [alsasink] doesn't set buffer-time correctly
- * 342789 : [audioresample] doesn't clear state when stopped, causing...
- * 343303 : [subparse] workaround for bad entities in sami parser
- * 343385 : [gnomevfs] add support for burn:// URIs
- * 343500 : [riff] gst_riff_parse_strf_vids() can't parse extra data.
- * 343699 : oggmux leaks
- * 344503 : [subparse] parse font face property in sami parser.
- * 345131 : [PATCH] videoscale support for 32-bit RGB-formats
- * 345206 : [textoverlay] crash with non-UTF8 input
- * 345225 : [theoradec] Clipping for exact seeking
- * 345641 : [API] [libgsttag] add enums for image tag type
- * 345879 : [riff] won't play a .wmv file with WMVA video stream
- * 346581 : [typefinding] recognise text/html
- * 347221 : [audioconvert] channel remapping does not work right
- * 347304 : Massive leaks with xvimagesink
- * 346527 : alsasrc get_range does not respect requested size
-
-Changes since 0.10.7:
-
- * alsasink probing fixes
- * xvimagesink error reporting fixes
- * subtitle fixes
- * adder fixes
- * vorbis multichannel fixes
- * multifdsink streamheader fixes
-
-Bugs fixed since 0.10.7:
-
- * 169936 : [subparse] support for SAMI subtitles
- * 315312 : Gstreamer Xv uses RGB instead of YUV.
- * 334002 : video4linux shouldn't depend on X in configure script
- * 336881 : [libvisual] additional support for libvisual-0.4
- * 337544 : [xvimagesink] Internal Error when image is too large
- * 339520 : [subparse] add " encoding " property
- * 340909 : [alsasink] can't enable spdif output
- * 341542 : some users have an assertion failed: (GST_VIDEO_SINK_WIDT...
- * 341562 : audioconvert doesn't list formats in order of preference
- * 341696 : audioconvert crashes if converting from a format with no ...
- * 341719 : bisection algorithm in ogg doesn't bisect in some cases
- * 341732 : [alsasink] doesn't query supported sample rates
- * 341873 : [alsasink] minor memory leak, uses unprotected static var...
- * 342143 : [subparse] sami parser needs to escape characters
- * 342181 : [alsa] add property probe interface to alsasink and alsasrc
- * 342268 : [playbin] add 'subtitle-encoding' property
- * 342345 : [riff] Elephant's Dream AVI does not play, JUNK chunk bef...
- * 342566 : Building without GTK+ fails
- * 343397 : H.264/AAC movie deadlocks with totem in gstreamer code, p...
- * 339935 : [adder] dead-locks when adding sink pads in PAUSED state
-
-Changes since 0.10.6:
-
- * typefind improvements
- * bug-fixes in textoverlay, audioconvert, videotestsrc,
- multifdsink and audio source/sink base classes
- * Ice-cast metadata support has moved from gnomevfssrc to the
- icydemux element in gst-plugins-good
- * audioresample now supports floating point samples
- * Adder element fixes.
- * Fixes for network playback and audio resampling in playbin
-
-Bugs fixed since 0.10.6:
-
- * 340060 : [adder] handle newsegment events properly
- * 340375 : [API 0.11] [patch] typefind to differentiate between mp4 ...
- * 339405 : [textoverlay] can't display '\n' character
- * 338657 : [patch] adder should send events from src-pad to all sink...
- * 338919 : [patch] alsasink should also query witdh capabilities fro...
- * 301759 : [audioresample] float audio support (for OSX audio sinks)
- * 331901 : [videotestsrc] framerate=0/1 gives assertion error
- * 333657 : Replacing icy demuxing in gnomevfssrc
- * 336339 : [audioresample] should support width != 16
- * 338718 : [patch] [audioconvert] correctly clip float samples > 1.0
- * 338778 : [patch] Bad audio with ASX files
- * 338991 : [patch] Videoscale doesn't pass on pixel-aspect ratio
- * 339574 : [patch] Race condition in multifdsink can lead to spuriou...
- * 339786 : [typefinding] wavpack typefinding doesn't always work
- * 340369 : [volume element] " volume " property range insufficient
- * 340379 : [playbin] doesn't insert audioresample, causes problems w...
- * 340392 : Problem with internal-decodebin
- * 341160 : [multifdsink] client_status enum has an uninitialized nick
- * 341182 : Accessing playbin's streaminfo property from high languag...
- * 341432 : [playbin] automatically get icecast metadata requiring ic...
- * 341542 : some users have an assertion failed: (GST_VIDEO_SINK_WIDT...
- * 341557 : Map GST_TAG_IMAGE < = > ID3v2 APIC tag
-
-API added since 0.10.6:
-
- * client-fd-removed signal added to multifdsink
- * stream-info-value-array property added to playbin
- * gst_video_calculate_display_ratio() in libgstvideo
-
-Changes since 0.10.5:
-
- * QoS in sinks and transform elements
- * Needs GStreamer 0.10.5 for new GstBaseSink::async_playback() vmethod
- * added theoraparse element
-
-Bugs fixed since 0.10.5:
-
- * 313136 : [playbin] hang while playing truncated ogg file
- * 172848 : [subparse] subtitles with special chars are displayed as ...
- * 305279 : [riff] uncompressed AVIs with 24bpp don't work
- * 320765 : [ffmpegcolorspace] make win32+msvc compliant, don't use _...
- * 323852 : Disable tests/icles on platforms that do not have X
- * 325653 : build errors compiling audioresample on win32(vs7)
- * 327357 : gst-plugins-base fails to compile with GCC 4.1
- * 334620 : [gnomevfssrc] fails to connect to icecast streaming servers
- * 334822 : [ffmpegcolorspace] YVU9 support
- * 335028 : [typefinding] ID3 v1 tag is not recognized with mp3-in-wa...
- * 335365 : inefficient use of GList in gst-plugins-base
- * 336190 : [gnomevfssink] should accept non-URI filenames as " location "
- * 336194 : [gnomevfssrc] some minor memory leaks
- * 336477 : plugins need better/univied descriptions
- * 336617 : Unable to recognise MPEG TS stream
- * 337548 : Memory leaks in basertpdepayload
- * 337945 : [oggdemux] segment stop position ignored
- * 338419 : Regression in the handling of files with multiple audio/s...
- * 338897 : Videoscale crashes as part of DVD to Ogg transcoding
- * 339013 : [videorate] Goes into an infinite loop
- * 339047 : [riff] handle H264 fourcc in addition to h264
- * 339212 : ISO file typefinding regression
- * 330748 : deadlock in base audio sink on playing- > paused state change
-
-Bugs fixed since 0.10.4:
-
- * 334216 : [gnomevfssrc] won't open some media on NFS mounts any longer
- * 334226 : typefindfunctions plugin crashes on PPC on registration
-
-Changes since 0.10.3:
-
- * (Experimental) QoS support
- * oggmuxer now creates 100% valid streams for Theora, Vorbis and Speex
- * documentation updates
- * better support for subtitles (seeking)
-
-Bugs fixed since 0.10.3:
-
- * 310202 : [subtitles] < i > < /i > tags and others should be supported i...
- * 312439 : XVideo output doesn't work on remote displays (probably r...
- * 321271 : audio output is truncated at EOS
- * 321650 : Can't decode this ogm file
- * 325732 : [oggdemux] problem when seeking to time less than 4s with...
- * 325972 : [typefinding] doesn't recognise this mp3
- * 326720 : [alsasink] doesn't support more than 2 channels anymore
- * 330711 : [ffmpegcolorspace] problems with palettized RGB (fencount...
- * 330789 : gstbaseaudiosink causes noise on seeking
- * 330888 : Fix build with gcc 2.95 (again)
- * 331295 : gnomevfssink doesn't respect umask when creating files
- * 331526 : 3GP type detection is too simple
- * 331678 : Decodebin is not reusable within a single pipeline (as in...
- * 331690 : playbin won't play my last.fm stream
- * 331763 : [alsamixer] unmute sets the volume to 100%
- * 331765 : [alsamixer] mixer applet slider doesn't want to move from...
- * 331903 : [videorate] doesnt handle input caps of framerate=0/1 sanely
- * 332778 : [ogmparse] " Already an existing pad " WARNING
- * 332964 : random crashes in mp3_type_find
- * 333254 : theora encoder does not set IN_CAPS flag properly
- * 333352 : [gnomevfssink] reports disk full as generic error
- * 333488 : Allow for palette < 256 colours in AVI files
- * 333510 : [PATCH] Fix gst_pad_new_from_template (gst_static_pad_tem...
- * 333545 : [riff] set depth on wma caps to make asfdemux and pitfdll...
- * 333663 : [patch] unref the result of gst_pad_get_parent
- * 333900 : [typefind] cannot play a particular mp3 file
- * 334112 : variable not initialized
- * 334129 : Disable frame dropping for now
- * 317038 : use default channel layout if none is specified in multic...
- * 319340 : [cdparanoia] uncorrected-error signal never fired
-
-API added since 0.10.3:
-
- * GstTextOverlay::halignment
- * GstTextOverlay::valignment
-
-Changes since 0.10.2:
-
- * typefind improvements
- * Ogg decoding and encoding fixes
- * Improved audio and video sink classes
- * Bug and leak fixes
- * Improved video scaling
- * On-the-fly visualisation switching
- * Subtitle support
-
-Bugs fixed since 0.10.2:
-
- * 330244 : gsttextoverlay.c:895: 'struct _GstCollectData' has no mem...
- * 324000 : [playbin] post error or message on unknown input
- * 153004 : [typefind] can't identify mp3 file with one single mpeg f...
- * 323874 : [playbin] leaks sinks and threads when using gconfaudiosink
- * 324626 : ffmpegcolorspace support for fourcc " UYVY "
- * 326447 : check that all elements in -base pass queries they can't ...
- * 328263 : Fix build with gcc 2.95
- * 328279 : [decodebin] timeout issue when pre-rolling
- * 329326 : Fix oggmux removing pads from collect pads
-
-Changes since 0.10.1:
-
- * ported gnomevfssink, cdparanoia
- * New library and base class: GstCddaBaseSrc
- * ported mixerutils.h
- * added 'sine-tab' waveform to audiotestsrc
- * added float audio to audiorate
-
-Bugs fixed since 0.10.1:
-
- * 324216 : [cdparanoia] missing patches from 0.8
- * 324696 : [videotestsrc] does not start counting the time from zero...
- * 324900 : Problem compiling gst-plugins-base with Forte
- * 325984 : [playbin] cannot handle sources that produce raw audio/video
- * 325990 : patch videotestsrc for using glib types
- * 326601 : GstRingBuffer crashes with alaw/mulaw caps
- * 327114 : [theoradec] should post tags on the bus
- * 327216 : vorbisdec segfaults on certain queries
-
-API added since 0.10.1:
-
- * added libgstcddabase
- * added mixerutils.h
-
-Changes since 0.10.0:
-
- * Parallel installability with 0.8.x series
- * Threadsafe design and API
- * removed gst-launch-ext
- * Ported: ogmparse
- * Fixes for: subparse, xvimagesink, audioresample, videorate, decodebin
-
-Bugs fixed since 0.10.0:
-
- * 322347 : GstBaseRtpDepayload timestamps are wring
- * 323900 : Basertpdepayloader lets NEWSEGMENT events through unfiltered
- * 323878 : missing < string.h > inclusion (for memset & FD_ZERO)
-
-API added since 0.10.0:
-
- * GstAlsaMixer::device
- * GstAlsaMixer::device-name
-
-Bugs fixed since 0.9.7:
-
- * 319172 : gstreamer-plugins-base-0.9.pc doesn't export linking flags
- * 323017 : While(1) loop with sleep(0) in basertpdepayload.c
-
-Changes since 0.9.6:
-
- * Parallel installability with 0.8.x series
- * Threadsafe design and API
- * ximagesink and xvimagesink updates and interactive test
- * added pango
- * rename net to netbuffer library
- * rtp element renaming
- * stream selector fixes
-
-Bugs fixed since 0.9.6:
-
- * 319618 : [decodebin] some ogg videos don't play
- * 320644 : RTP packetizer does't set the packet timestamps correctly
- * 322388 : xvimagesink force-aspect-ratio=True always displays squar...
- * 322704 : oggdemux typefind list leak
-
-Changes since 0.9.5:
-
- * Parallel installability with 0.8.x series
- * Threadsafe design and API
- * lots of leak fixes
- * flicker-free and rewritten X sinks
- * fractional framerates
- * removed sinesrc, replaced by audiotestsrc
-
-Bugs fixed since 0.9.5:
-
- * 316442 : playbin should use autoaudiosink/autovideosink by default
- * 318353 : [ffmpegcolorspace] forward-port fixes from 0.8 branch
- * 320200 : vorbisenc: min-bitrate and max-bitrate are 1/1000 bps rat...
- * 321164 : gstringbuffer stops working under load
- * 321426 : ximage plugin should be renamed to ximagesink
- * 321446 : sinesrc should be dropped in favour of audiotestsrc
- * 321451 : GstRtpBuffer: no way to create a sub buffer with only the...
- * 321816 : [API] xoverlay API to post prepare-xwindow-id message
- * 321894 : vorbisenc doesn't compile
- * 322117 : Rename libgsttagedit to libgsttag
-
-Changes since 0.9.4:
-
- * video caps now use a good range for framerate and w/h
- * oggdemux/oggmux improvements
- * playbin improvements
-
-Bugs fixed since 0.9.4:
-
- * 319110 : [PATCH] oggdemux chain finding is slow
- * 320058 : playbin of a jpeg over http does not work
- * 320923 : [volume] doesn't build on Solaris
- * 321011 : gstbasertpdepayload doesn't send the " new segment " event ...
-
-Changes since 0.9.3:
-
- * New element: audiotestsrc
- * typefind improvements
- * buffer-frames removed
-
-Changes since 0.9.2:
-
- * RTP base classes
-
-Bugs fixed since 0.9.2:
-
- * 313251 : ximagesink unused functions
- * 315159 : audioconvert lost 24 bit conversions in the rewrite
+# GStreamer 1.8 Release Notes
+**GStreamer 1.8.0 was released on 24 March 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/1.8/][latest] for the latest
+version of this document.
+
+*Last updated: Thursday 24 March 2016, 10:00 UTC [(log)][gitlog]*
+
+[latest]: https://gstreamer.freedesktop.org/releases/1.8/
+[gitlog]: https://cgit.freedesktop.org/gstreamer/www/log/src/htdocs/releases/1.8/release-notes-1.8.md
+
+## Highlights
+
+- **Hardware-accelerated zero-copy video decoding on Android**
+
+- **New video capture source for Android using the android.hardware.Camera API**
+
+- **Windows Media reverse playback** support (ASF/WMV/WMA)
+
+- **New tracing system** provides support for more sophisticated debugging tools
+
+- **New high-level GstPlayer playback convenience API**
+
+- **Initial support for the new [Vulkan][vulkan] API**, see
+ [Matthew Waters' blog post][vulkan-in-gstreamer] for more details
+
+- **Improved Opus audio codec support**: Support for more than two channels; MPEG-TS demuxer/muxer can now handle Opus;
+ [sample-accurate][opus-sample-accurate] encoding/decoding/transmuxing with
+ Ogg, Matroska, ISOBMFF (Quicktime/MP4), and MPEG-TS as container;
+ [new codec utility functions for Opus header and caps handling][opus-codec-utils]
+ in pbutils library. The Opus encoder/decoder elements were also moved to
+ gst-plugins-base (from -bad), and the opus RTP depayloader/payloader to -good.
+
+ [opus-sample-accurate]: https://gstreamer.freedesktop.org/data/doc/gstreamer/head/gst-plugins-base-libs/html/gst-plugins-base-libs-gstaudiometa.html#GstAudioClippingMeta
+ [opus-codec-utils]: https://gstreamer.freedesktop.org/data/doc/gstreamer/head/gst-plugins-base-libs/html/gst-plugins-base-libs-gstpbutilscodecutils.html
+
+- **GStreamer VAAPI module now released and maintained as part of the GStreamer project**
+
+ [vulkan]: https://www.khronos.org/vulkan
+ [vulkan-in-gstreamer]: http://ystreet00.blogspot.co.uk/2016/02/vulkan-in-gstreamer.html
+
+## Major new features and changes
+
+### Noteworthy new API, features and other changes
+
+- New GstVideoAffineTransformationMeta meta for adding a simple 4x4 affine
+ transformation matrix to video buffers
+
+- [g\_autoptr()](https://developer.gnome.org/glib/stable/glib-Miscellaneous-Macros.html#g-autoptr)
+ support for all types is exposed in GStreamer headers now, in combination
+ with a sufficiently-new GLib version (i.e. 2.44 or later). This is primarily
+ for the benefit of application developers who would like to make use of
+ this, the GStreamer codebase itself will not be using g_autoptr() for
+ the time being due to portability issues.
+
+- GstContexts are now automatically propagated to elements added to a bin
+ or pipeline, and elements now maintain a list of contexts set on them.
+ The list of contexts set on an element can now be queried using the new functions
+ [gst\_element\_get\_context()](https://gstreamer.freedesktop.org/data/doc/gstreamer/head/gstreamer/html/GstElement.html#gst-element-get-context)
+ and [gst\_element\_get\_contexts()](https://gstreamer.freedesktop.org/data/doc/gstreamer/head/gstreamer/html/GstElement.html#gst-element-get-contexts). GstContexts are used to share context-specific configuration objects
+ between elements and can also be used by applications to set context-specific
+ configuration objects on elements, e.g. for OpenGL or Hardware-accelerated
+ video decoding.
+
+- New [GST\_BUFFER\_DTS\_OR\_PTS()](https://gstreamer.freedesktop.org/data/doc/gstreamer/head/gstreamer/html/GstBuffer.html#GST-BUFFER-DTS-OR-PTS:CAPS)
+ convenience macro that returns the decode timestamp if one is set and
+ otherwise returns the presentation timestamp
+
+- New GstPadEventFullFunc that returns a GstFlowReturn instead of a gboolean.
+ This new API is mostly for internal use and was added to fix a race condition
+ where occasionally internal flow error messages were posted on the bus when
+ sticky events were propagated at just the wrong moment whilst the pipeline
+ was shutting down. This happened primarily when the pipeline was shut down
+ immediately after starting it up. GStreamer would not know that the reason
+ the events could not be propagated was because the pipeline was shutting down
+ and not some other problem, and now the flow error allows GStreamer to know
+ the reason for the failure (and that there's no reason to post an error
+ message). This is particularly useful for queue-like elements which may need
+ to asynchronously propagate a previous flow return from downstream.
+
+- Pipeline dumps in form of "dot files" now also show pad properties that
+ differ from their default value, the same as it does for elements. This is
+ useful for elements with pad subclasses that provide additional properties,
+ e.g. videomixer or compositor.
+
+- Pad probes are now guaranteed to be called in the order they were added
+ (before they were called in reverse order, but no particular order was
+ documented or guaranteed)
+
+- Plugins can now have dependencies on device nodes (not just regular files)
+ and also have a prefix filter. This is useful for plugins that expose
+ features (elements) based on available devices, such as the video4linux
+ plugin does with video decoders on certain embedded systems.
+
+- gst\_segment\_to\_position() has been deprecated and been replaced by the
+ better-named gst\_segment\_position\_from\_running\_time(). At the same time
+ gst\_segment\_position\_from\_stream\_time() was added, as well as \_full()
+ variants of both to deal with negative stream time.
+
+- GstController: the interpolation control source gained a new monotonic cubic
+ interpolation mode that, unlike the existing cubic mode, will never overshoot
+ the min/max y values set.
+
+- GstNetAddressMeta: can now be read from buffers in language bindings as well,
+ via the new gst\_buffer\_get\_net\_address\_meta() function
+
+- ID3 tag PRIV frames are now extraced into a new GST\_TAG\_PRIVATE\_DATA tag
+
+- gst-launch-1.0 and gst\_parse\_launch() now warn in the most common case if
+ a dynamic pad link could not be resolved, instead of just silently
+ waiting to see if a suitable pad appears later, which is often perceived
+ by users as hanging -- they are now notified when this happens and can check
+ their pipeline.
+
+- GstRTSPConnection now also parses custom RTSP message headers and retains
+ them for the application instead of just ignoring them
+
+- rtspsrc handling of authentication over tunneled connections (e.g. RTSP over HTTP)
+ was fixed
+
+- gst\_video\_convert\_sample() now crops if there is a crop meta on the input buffer
+
+- The debugging system printf functions are now exposed for general use, which
+ supports special printf format specifiers such as GST\_PTR\_FORMAT and
+ GST\_SEGMENT\_FORMAT to print GStreamer-related objects. This is handy for
+ systems that want to prepare some debug log information to be output at a
+ later point in time. The GStreamer-OpenGL subsystem is making use of these
+ new functions, which are [gst\_info\_vasprintf()][gst_info_vasprintf],
+ [gst\_info\_strdup\_vprintf()][gst_info_strdup_vprintf] and
+ [gst\_info\_strdup\_printf()][gst_info_strdup_printf].
+
+- videoparse: "strides", "offsets" and "framesize" properties have been added to
+ allow parsing raw data with strides and padding that do not match GStreamer
+ defaults.
+
+- GstPreset reads presets from the directories given in GST\_PRESET\_PATH now.
+ Presets are read from there after presets in the system path, but before
+ application and user paths.
+
+[gst_info_vasprintf]: https://gstreamer.freedesktop.org/data/doc/gstreamer/head/gstreamer/html/gstreamer-GstInfo.html#gst-info-vasprintf
+[gst_info_strdup_vprintf]: https://gstreamer.freedesktop.org/data/doc/gstreamer/head/gstreamer/html/gstreamer-GstInfo.html#gst-info-strdup-vprintf
+[gst_info_strdup_printf]: https://gstreamer.freedesktop.org/data/doc/gstreamer/head/gstreamer/html/gstreamer-GstInfo.html#gst-info-strdup-printf
+
+### New Elements
+
+- [netsim](): a new (resurrected) element to simulate network jitter and
+ packet dropping / duplication.
+
+- New VP9 RTP payloader/depayloader elements: rtpvp9pay/rtpvp9depay
+
+- New [videoframe_audiolevel]() element, a video frame synchronized audio level element
+
+- New spandsp-based tone generator source
+
+- New NVIDIA NVENC-based H.264 encoder for GPU-accelerated video encoding on
+ suitable NVIDIA hardware
+
+- [rtspclientsink](), a new RTSP RECORD sink element, was added to gst-rtsp-server
+
+- [alsamidisrc](), a new ALSA MIDI sequencer source element
+
+### Noteworthy element features and additions
+
+- *identity*: new ["drop-buffer-flags"](https://gstreamer.freedesktop.org/data/doc/gstreamer/head/gstreamer-plugins/html/gstreamer-plugins-identity.html#GstIdentity--drop-buffer-flags)
+ property to drop buffers based on buffer flags. This can be used to drop all
+ non-keyframe buffers, for example.
+
+- *multiqueue*: various fixes and improvements, in particular special handling
+ for sparse streams such as substitle streams, to make sure we don't overread
+ them any more. For sparse streams it can be normal that there's no buffer for
+ a long period of time, so having no buffer queued is perfectly normal. Before
+ we would often unnecessarily try to fill the subtitle stream queue, which
+ could lead to much more data being queued in multiqueue than necessary.
+
+- *multiqueue*/*queue*: When dealing with time limits, these elements now use the
+ new ["GST_BUFFER_DTS_OR_PTS"](https://gstreamer.freedesktop.org/data/doc/gstreamer/head/gstreamer/html/GstBuffer.html#GST-BUFFER-DTS-OR-PTS:CAPS)
+ and ["gst_segment_to_running_time_full()"](https://gstreamer.freedesktop.org/data/doc/gstreamer/head/gstreamer/html/GstSegment.html#gst-segment-to-running-time-full)
+ API, resulting in more accurate levels, especially when dealing with non-raw
+ streams (where reordering happens, and we want to use the increasing DTS as
+ opposed to the non-continuously increasing PTS) and out-of-segment input/output.
+ Previously all encoded buffers before the segment start, which can happen when
+ doing ACCURATE seeks, were not taken into account in the queue level calculation.
+
+- *multiqueue*: New ["use-interleave"](https://gstreamer.freedesktop.org/data/doc/gstreamer/head/gstreamer-plugins/html/gstreamer-plugins-multiqueue.html#GstMultiQueue--use-interleave)
+ property which allows the size of the queues to be optimized based on the input
+ streams interleave. This should only be used with input streams which are properly
+ timestamped. It will be used in the future decodebin3 element.
+
+- *queue2*: new ["avg-in-rate"](https://gstreamer.freedesktop.org/data/doc/gstreamer/head/gstreamer-plugins/html/gstreamer-plugins-queue2.html#GstQueue2--avg-in-rate)
+ property that returns the average input rate in bytes per second
+
+- audiotestsrc now supports all audio formats and is no longer artificially
+ limited with regard to the number of channels or sample rate
+
+- gst-libav (ffmpeg codec wrapper): map and enable JPEG2000 decoder
+
+- multisocketsink can, on request, send a custom GstNetworkMessage event
+ upstream whenever data is received from a client on a socket. Similarly,
+ socketsrc will, on request, pick up GstNetworkMessage events from downstream
+ and send any data contained within them via the socket. This allows for
+ simple bidirectional communication.
+
+- matroska muxer and demuxer now support the ProRes video format
+
+- Improved VP8/VP9 decoding performance on multi-core systems by enabling
+ multi-threaded decoding in the libvpx-based decoders on such systems
+
+- appsink has a new ["wait-on-eos"](https://gstreamer.freedesktop.org/data/doc/gstreamer/head/gst-plugins-base-plugins/html/gst-plugins-base-plugins-appsink.html#GstAppSink--wait-on-eos)
+ property, so in cases where it is uncertain if an appsink will have a consumer for
+ its buffers when it receives an EOS this can be set to FALSE to ensure that the
+ appsink will not hang.
+
+- rtph264pay and rtph265pay have a new "config-interval" mode -1 that will
+ re-send the setup data (SPS/PPS/VPS) before every keyframe to ensure
+ optimal coverage and the shortest possibly start-up time for a new client
+
+- mpegtsmux can now mux H.265/HEVC video as well
+
+- The MXF muxer was ported to 1.x and produces more standard conformant files now
+ that can be handled by more other software; The MXF demuxer got improved
+ support for seek tables (IndexTableSegments).
+
+### Plugin moves
+
+- The rtph265pay/depay RTP payloader/depayloader elements for H.265/HEVC video
+ from the rtph265 plugin in -bad have been moved into the existing rtp plugin
+ in gst-plugins-good.
+
+- The mpg123 plugin containing a libmpg123 based audio decoder element has
+ been moved from -bad to -ugly.
+
+- The Opus encoder/decoder elements have been moved to gst-plugins-base and
+ the RTP payloader to gst-plugins-good, both coming from gst-plugins-bad.
+
+### New tracing tools for developers
+
+A new tracing subsystem API has been added to GStreamer, which provides
+external tracers with the possibility to strategically hook into GStreamer
+internals and collect data that can be evaluated later. These tracers are a
+new type of plugin features, and GStreamer core ships with a few example
+tracers (latency, stats, rusage, log) to start with. Tracers can be loaded
+and configured at start-up via an environment variable (GST\_TRACER\_PLUGINS).
+
+Background: While GStreamer provides plenty of data on what's going on in a
+pipeline via its debug log, that data is not necessarily structured enough to
+be generally useful, and the overhead to enable logging output for all data
+required might be too high in many cases. The new tracing system allows tracers
+to just obtain the data needed at the right spot with as little overhead as
+possible, which will be particularly useful on embedded systems.
+
+Of course it has always been possible to do performance benchmarks and debug
+memory leaks, memory consumption and invalid memory access using standard
+operating system tools, but there are some things that are difficult to track
+with the standard tools, and the new tracing system helps with that. Examples
+are things such as latency handling, buffer flow, ownership transfer of
+events and buffers from element to element, caps negotiation, etc.
+
+For some background on the new tracing system, watch Stefan Sauer's
+GStreamer Conference talk ["A new tracing subsystem for GStreamer"][tracer-0]
+and for a more specific example how it can be useful have a look at
+Thiago Santos's lightning talk ["Analyzing caps negotiation using GstTracer"][tracer-1]
+and his ["GstTracer experiments"][tracer-2] blog post. There was also a Google
+Summer of Code project in 2015 that used tracing system for a graphical
+GStreamer debugging tool ["gst-debugger"][tracer-3].
+
+This is all still very much work in progress, but we hope this will provide the
+foundation for a whole suite of new debugging tools for GStreamer pipelines.
+
+[tracer-0]: https://gstconf.ubicast.tv/videos/a-new-tracing-subsystem-for-gstreamer/
+[tracer-1]: https://gstconf.ubicast.tv/videos/analyzing-caps-negotiation-using-gsttracer/
+[tracer-2]: http://blog.thiagoss.com/2015/07/23/gsttracer-experiments/
+[tracer-3]: https://git.gnome.org/browse/gst-debugger
+
+### GstPlayer: a new high-level API for cross-platform multimedia playback
+
+GStreamer has had reasonably high-level API for multimedia playback
+in the form of the playbin element for a long time. This allowed application
+developers to just configure a URI to play, and playbin would take care of
+everything else. This works well, but there is still way too much to do on
+the application-side to implement a fully-featured playback application, and
+too much general GStreamer pipeline API exposed, making it less accessible
+to application developers.
+
+Enter GstPlayer. GstPlayer's aim is to provide an even higher-level abstraction
+of a fully-featured playback API but specialised for its specific use case. It
+also provides easy integration with and examples for Gtk+, Qt, Android, OS/X,
+iOS and Windows. Watch Sebastian's [GstPlayer talk at the GStreamer Conference][gstplayer-talk]
+for more information, or check out the [GstPlayer API reference][gstplayer-api]
+and [GstPlayer examples][gstplayer-examples].
+
+[gstplayer-api]: https://gstreamer.freedesktop.org/data/doc/gstreamer/head/gst-plugins-bad-libs/html/player.html
+[gstplayer-talk]: https://gstconf.ubicast.tv/videos/gstplayer-a-simple-cross-platform-api-for-all-your-media-playback-needs-part-1/
+[gstplayer-examples]: https://github.com/sdroege/gst-player/
+
+### Adaptive streaming: DASH, HLS and MSS improvements
+
+- dashdemux now supports loading external xml nodes pointed from its MPD.
+
+- Content protection nodes parsing support for PlayReady WRM in mssdemux.
+
+- Reverse playback was improved to respect seek start and stop positions.
+
+- Adaptive demuxers (hlsdemux, dashdemux, mssdemux) now support the SNAP_AFTER
+ and SNAP_BEFORE seek flags which will jump to the nearest fragment boundary
+ when executing a seek, which means playback resumes more quickly after a seek.
+
+### Audio library improvements
+
+- audio conversion, quantization and channel up/downmixing functionality
+ has been moved from the audioconvert element into the audio library and
+ is now available as public API in form of [GstAudioConverter][audio-0],
+ [GstAudioQuantize][audio-1] and [GstAudioChannelMixer][audio-2].
+ Audio resampling will follow in future releases.
+
+- [gst\_audio\_channel\_get\_fallback\_mask()][audio-3] can be used
+ to retrieve a default channel mask for a given number of channels as last
+ resort if the layout is unknown
+
+- A new [GstAudioClippingMeta][audio-4] meta was added for specifying clipping
+ on encoded audio buffers
+
+- A new GstAudioVisualizer base class for audio visualisation elements;
+ most of the existing visualisers have been ported over to the new base class.
+ This new base class lives in the pbutils library rather than the audio library,
+ since we'd have had to make libgstaudio depend on libgstvideo otherwise,
+ which was deemed undesirable.
+
+[audio-0]: https://gstreamer.freedesktop.org/data/doc/gstreamer/head/gst-plugins-base-libs/html/gst-plugins-base-libs-GstAudioConverter.html
+[audio-1]: https://gstreamer.freedesktop.org/data/doc/gstreamer/head/gst-plugins-base-libs/html/gst-plugins-base-libs-GstAudioQuantize.html
+[audio-2]: https://gstreamer.freedesktop.org/data/doc/gstreamer/head/gst-plugins-base-libs/html/gst-plugins-base-libs-gstaudiochannels.html#gst-audio-channel-mix-new
+[audio-3]: https://gstreamer.freedesktop.org/data/doc/gstreamer/head/gst-plugins-base-libs/html/gst-plugins-base-libs-gstaudiochannels.html#gst-audio-channel-get-fallback-mask
+[audio-4]: https://gstreamer.freedesktop.org/data/doc/gstreamer/head/gst-plugins-base-libs/html/gst-plugins-base-libs-gstaudiometa.html#GstAudioClippingMeta
+
+### GStreamer OpenGL support improvements
+
+#### Better OpenGL Shader support
+
+[GstGLShader][shader] has been revamped to allow more OpenGL shader types
+by utilizing a new GstGLSLStage object. Each stage holds an OpenGL pipeline
+stage such as a vertex, fragment or a geometry shader that are all compiled
+separately into a program that is executed.
+
+The glshader element has also received a revamp as a result of the changes in
+the library. It does not take file locations for the vertex and fragment
+shaders anymore. Instead it takes the strings directly leaving the file
+management to the application.
+
+A new [example][liveshader-example] was added utilizing the new shader
+infrastructure showcasing live shader edits.
+
+[shader]: https://gstreamer.freedesktop.org/data/doc/gstreamer/head/gst-plugins-bad-libs/html/gst-plugins-bad-libs-gstglshader.html
+[liveshader-example]: https://cgit.freedesktop.org/gstreamer/gst-plugins-bad/tree/tests/examples/gtk/glliveshader.c
+
+#### OpenGL GLMemory rework
+
+[GstGLMemory] was extensively reworked to support the addition of multiple
+texture targets required for zero-copy integration with the Android
+MediaCodec elements. This work was also used to provide IOSurface based
+GLMemory on OS X for zero-copy with OS X's VideoToolbox decoder (vtdec) and
+AV Foundation video source (avfvideosrc). There are also patches in bugzilla
+for GstGLMemoryEGL specifically aimed at improving the decoding performance on
+the Raspberry Pi.
+
+[GstGLMemory]: https://gstreamer.freedesktop.org/data/doc/gstreamer/head/gst-plugins-bad-libs/html/gst-plugins-bad-libs-gstglmemory.html
+
+A texture-target field was added to video/x-raw(memory:GLMemory) caps to signal
+the texture target contained in the GLMemory. Its values can be 2D, rectangle
+or external-oes. glcolorconvert can convert between the different formats as
+required and different elements will accept or produce different targets. e.g.
+glimagesink can take and render external-oes textures directly as required for
+effecient zero-copy on android.
+
+A generic GL allocation framework was also implemented to support the generic
+allocation of OpenGL buffers and textures which is used extensively by
+GstGLBufferPool.
+
+#### OpenGL DMABuf import uploader
+
+There is now a DMABuf uploader available for automatic selection that will
+attempt to import the upstream provided DMABuf. The uploader will import into
+2D textures with the necesarry format. YUV to RGB conversion is still provided
+by glcolorconvert to avoid the laxer restrictions with external-oes textures.
+
+#### OpenGL queries
+
+Queries of various aspects of the OpenGL runtime such as timers, number of
+samples or the current timestamp are not possible. The GstGLQuery object uses a
+delayed debug system to delay the debug output to later to avoid expensive calls
+to the glGet\* family of functions directly after finishing a query. It is
+currently used to output the time taken to perform various operations of texture
+uploads and downloads in GstGLMemory.
+
+#### New OpenGL elements
+
+glcolorbalance has been created mirroring the videobalance elements.
+glcolorbalance provides the exact same interface as videobalance so can be used
+as a GPU accelerated replacement. glcolorbalance has been added to glsinkbin so
+usage with playsink/playbin will use it automatically instead of videobalance
+where possible.
+
+glvideoflip, which is the OpenGL equiavalant of videoflip, implements the exact
+same interface and functionality as videoflip.
+
+#### EGL implementation now selects OpenGL 3.x
+
+The EGL implementation can now select OpenGL 3.x contexts.
+
+#### OpenGL API removal
+
+The GstGLDownload library object was removed as it was not used by anything.
+Everything is performed by GstGLMemory or in the gldownloadelement.
+
+The GstGLUploadMeta library object was removed as it was not being used and we
+don't want to promote the use of GstVideoGLTextureUploadMeta.
+
+#### OpenGL: Other miscellaneous changes
+
+- The EGL implementation can now select OpenGL 3.x contexts. This brings
+ OpenGL 3.x to e.g. wayland and other EGL systems.
+
+- glstereomix/glstereosplit are now built and are usable on OpenGL ES systems
+
+- The UYVY/YUY2 to RGBA and RGBA to UYVY/YUY2 shaders were fixed removing the
+ sawtooth pattern and luma bleeding.
+
+- We now utilize the GL\_APPLE\_sync extension on iOS devices which improves
+ performance of OpenGL applications, especially with multiple OpenGL
+ contexts.
+
+- glcolorconvert now uses a bufferpool to avoid costly
+ glGenTextures/glDeleteTextures for every frame.
+
+- glvideomixer now has full glBlendFunc and glBlendEquation support per input.
+
+- gltransformation now support navigation events so your weird transformations
+ also work with DVD menus.
+
+- qmlglsink can now run on iOS, OS X and Android in addition to the already
+ supported Linux platform.
+
+- glimagesink now posts unhandled keyboard and mouse events (on backends that
+ support user input, current only X11) on the bus for the application.
+
+### Initial GStreamer Vulkan support
+
+Some new elements, vulkansink and vulkanupload have been implemented utilizing
+the new Vulkan API. The implementation is currently limited to X11 platforms
+(via xcb) and does not perform any scaling of the stream's contents to the size
+of the available output.
+
+A lot of infrasctructure work has been undertaken to support using Vulkan in
+GStreamer in the future. A number of GstMemory subclasses have been created for
+integrating Vulkan's GPU memory handling along with VkBuffer's and VkImage's
+that can be passed between elements. Some GStreamer refcounted wrappers for
+global objects such as VkInstance, VkDevice, VkQueue, etc have also been
+implemented along with GstContext integration for sharing these objects with the
+application.
+
+### GStreamer VAAPI support for hardware-accelerated video decoding and encoding on Intel (and other) platforms
+
+#### GStreamer VAAPI is now part of upstream GStreamer
+
+The GStreamer-VAAPI module which provides support for hardware-accelerated
+video decoding, encoding and post-processing on Intel graphics hardware
+on Linux has moved from its previous home at the [Intel Open Source Technology Center][iostc]
+to the upstream GStreamer repositories, where it will in future be maintained
+as part of the upstream GStreamer project and released in lockstep with the
+other GStreamer modules. The current maintainers will continue to spearhead
+the development at the new location:
+
+[http://cgit.freedesktop.org/gstreamer/gstreamer-vaapi/][gst-vaapi-git]
+
+[gst-vaapi-git]: http://cgit.freedesktop.org/gstreamer/gstreamer-vaapi/
+
+GStreamer-VAAPI relies heavily on certain GStreamer infrastructure API that
+is still in flux such as the OpenGL integration API or the codec parser
+libraries, and one of the goals of the move was to be able to leverage
+new developments early and provide tighter integration with the latest
+developments of those APIs and other graphics-related APIs provided by
+GStreamer, which should hopefully improve performance even further and in
+some cases might also provide better stability.
+
+Thanks to everyone involved in making this move happen!
+
+#### GStreamer VAAPI: Bug tracking
+
+Bugs had already been tracked on [GNOME bugzilla](bgo) but will be moved
+from the gstreamer-vaapi product into a new gstreamer-vaapi component of
+the GStreamer product in bugzilla. Please file new bugs against the new
+component in the GStreamer product from now on.
+
+#### GStreamer VAAPI: Pending patches
+
+The code base has been re-indented to the GStreamer code style, which
+affected some files more than others. This means that some of the patches
+in bugzilla might not apply any longer, so if you have any unmerged patches
+sitting in bugzilla please consider checking if they still apply cleany and
+refresh them if not. Sorry for any inconvenience this may cause.
+
+#### GStreamer VAAPI: New versioning scheme and supported GStreamer versions
+
+The version numbering has been changed to match the GStreamer version
+numbering to avoid confusion: there is a new gstreamer-vaapi 1.6.0 release
+and a 1.6 branch that is roughly equivalent to the previous 0.7.0 version.
+Future releases 1.7.x and 1.8.x will be made alongside GStreamer releases.
+
+While it was possible and supported by previous releases to build against
+a whole range of different GStreamer versions (such as 1.2, 1.4, 1.6 or 1.7/1.8),
+in the future there will only be one target branch, so that git master will
+track GStreamer git master, 1.8.x will target GStreamer 1.8, and
+1.6.x will target the 1.6 series.
+
+[iostc]: http://01.org
+[bgo]: http://bugzilla.gnome.og
+
+#### GStreamer VAAPI: Miscellaneous changes
+
+All GStreamer-VAAPI functionality is now provided solely by its GStreamer
+elements. There is no more public library exposing GstVaapi API, this API
+was only ever meant for private use by the elements. Parts of it may be
+resurrected again in future if needed, but for now it has all been made
+private.
+
+GStreamer-VAAPI now unconditionally uses the codecparser library in
+gst-plugins-bad instead of shipping its own internal copy. Similarly,
+it no longer ships its own codec parsers but relies on the upstream
+codec parser elements.
+
+The GStreamer-VAAPI encoder elements have been renamed from vaapiencode_foo
+to vaapifooenc, so encoders are now called vaapih264enc, vaapih265enc,
+vaapimpeg2enc, vaapijpegenc, and vaapivp8enc. With this change we now follow
+the standard names in GStreamer, and the plugin documentation is generated
+correctly.
+
+In the case of the decoders, only the jpeg decoder has been split from the
+general decoding element vaapidecode: vaapijpegdec. This is the first step to
+split per codec each decoding element. The vaapijpegdec has also been given
+marginal rank for the time being.
+
+#### GStreamer VAAPI: New features in 1.8: 10-bit H.265/HEVC decoding support
+
+Support for decoding 10-bit H.265/HEVC has been added. For the time being
+this only works in combination with vaapisink though, until support for the
+P010 video format used internally is added to GStreamer and to the
+vaGetImage()/vaPutimage() API in the vaapi-intel-driver.
+
+Several fixes for memory leaks, build errors, and in the internal
+video parsing.
+
+Finally, vaapisink now posts the unhandled keyboard and mouse events to the
+application.
+
+### GStreamer Video 4 Linux Support
+
+Colorimetry support has been enhanced even more. It will now properly select
+default values when not specified by the driver. The range of color formats
+supported by GStreamer has been greatly improved. Notably, support for
+multi-planar I420 has been added along with all the new and non-ambiguous RGB
+formats that got added in recent kernels.
+
+The device provider now exposes a variety of properties as found in the udev
+database.
+
+The video decoder is now able to negotiate the downstream format.
+
+Elements that are dynamically created from /dev/video\* now track changes on
+these devices to ensure the registry stay up to date.
+
+All this and various bug fixes that improve both stability and correctness.
+
+### GStreamer Editing Services
+
+Added APIs to handle asset proxying support. Proxy creation is not the
+responsibility of GES itself, but GES provides all the needed features
+for it to be cleanly handled at a higher level.
+
+Added support for changing playback rate. This means that now, whenever a
+user adds a 'pitch' element (as it is the only known element to change playback
+rate through properties), GES will handle everything internally. This change
+introduced a new media-duration-factor property in NleObject which will
+lead to tweaking of seek events so they have the proper playback range to be
+requested upstream.
+
+Construction of NLE objects has been reworked making copy/pasting fully
+functional and allowing users to set properties on effects right after
+creating them.
+
+Rework of the title source to add more flexibility in text positioning,
+and letting the user get feedback about rendered text positioning.
+
+Report nlecomposition structural issues (coming from user programing mistakes)
+into ERROR messages on the bus.
+
+Add GI/pythyon testsuite in GES itself, making sure the API is working as expected
+in python, and allowing writing tests faster.
+
+### GstValidate
+
+Added support to run tests inside gdb.
+
+Added a 'smart' reporting mode where we give as much information as possible about
+critical errors.
+
+Uses GstTracer now instead of a LD\_PRELOAD library.
+
+## Miscellaneous
+
+- encodebin now works with "encoder-muxers" such as wavenc
+
+- gst-play-1.0 acquired a new keyboard shortcut: '0' seeks back to the start
+
+- gst-play-1.0 supports two new command line switches: -v for verbose output
+ and --flags to configure the playbin flags to use.
+
+## Build and Dependencies
+
+- The GLib dependency requirement was bumped to 2.40
+
+- The -Bsymbolic configure check now works with clang as well
+
+- ffmpeg is now required as libav provider, incompatible changes were
+ introduced that make it no longer viable to support both FFmpeg and Libav
+ as libav providers. Most major distros have switched to FFmpeg or are in
+ the process of switching to it anyway, so we don't expect this to be a
+ problem, and there is still an internal copy of ffmpeg that can be used
+ as fallback if needed.
+
+- The internal ffmpeg snapshot is now FFMpeg 3.0, but it should be possible
+ to build against 2.8 as well for the time being.
+
+## Platform-specific improvements
+
+### Android
+
+- Zero-copy video decoding on Android using the hardware-accelerated decoders
+ has been implemented, and is fully integrated with the GStreamer OpenGL stack
+
+- ahcsrc, a new camera source element, has been merged and can be used to
+ capture video on android devices. It uses the android.hardware.Camera Java
+ API to capture from the system's cameras.
+
+- The OpenGL-based QML video sink can now also be used on Android
+
+- New tinyalsasink element, which is mainly useful for Android but can also
+ be used on other platforms.
+
+### OS/X and iOS
+
+- The system clock now uses mach\_absolute\_time() on OSX/iOS, which is
+ the preferred high-resolution monotonic clock to be used on Apple platforms
+
+- The OpenGL-based QML video sink can now also be used on OS X and iOS (with
+ some Qt build system massaging)
+
+- New IOSurface based memory implementation in avfvideosrc and vtdec on OS X
+ for zerocopy with OpenGL. The previously used OpenGL extension
+ GL_APPLE_ycbcr_422 is not compatible with GL 3.x core contexts.
+
+- New GstAppleCoreVideoMemory wrapping CVPixelBuffer's
+
+- avfvideosrc now supports renegotiation.
+
+### Windows
+
+- Various bugs with UDP and multicast were fixed on Windows, mostly related to
+ gst-rtsp-server.
+
+- A few bugs in directsoundsrc and directsoundsink were fixed that could cause
+ the element to lock up. Also the "mute" property on the sink was fixed, and
+ a new "device" property for device selection was added to the source.
+
+## Known Issues
+
+- Building GStreamer applications with the Android NDK r11 is currently not
+ supported due to incompatible changes in the NDK. This is expected to be
+ fixed for 1.8.1.
+ [Bugzilla #763999](https://bugzilla.gnome.org/show_bug.cgi?id=763999)
+
+- vp8enc crashes on 32 bit Windows, but was working fine in 1.6. 64 bit
+ Windows is unaffected.
+ [Bugzilla #763663](https://bugzilla.gnome.org/show_bug.cgi?id=763663)
+
+## Contributors
+
+Adam Miartus, Alban Bedel, Aleix Conchillo Flaqué, Aleksander Wabik,
+Alessandro Decina, Alex Ashley, Alex Dizengof, Alex Henrie, Alistair Buxton,
+Andreas Cadhalpun, Andreas Frisch, André Draszik, Anthony G. Basile,
+Antoine Jacoutot, Anton Bondarenko, Antonio Ospite, Arjen Veenhuizen,
+Arnaud Vrac, Arun Raghavan, Athanasios Oikonomou, Aurélien Zanelli, Ben Iofel,
+Bob Holcomb, Branko Subasic, Carlos Rafael Giani, Chris Bass, Csaba Toth,
+Daniel Kamil Kozar, Danilo Cesar Lemes de Paula, Dave Craig, David Fernandez,
+David Schleef, David Svensson Fors, David Waring, David Wu, Duncan Palmer,
+Edward Hervey, Egor Zaharov, Etienne Peron, Eunhae Choi, Evan Callaway,
+Evan Nemerson, Fabian Orccon, Florent Thiéry, Florin Apostol, Frédéric Wang,
+George Kiagiadakis, George Yunaev, Göran Jönsson, Graham Leggett,
+Guillaume Desmottes, Guillaume Marquebielle, Haihua Hu, Havard Graff,
+Heinrich Fink, Holger Kaelberer, HoonHee Lee, Hugues Fruchet, Hyunil Park,
+Hyunjun Ko, Ilya Konstantinov, James Stevenson, Jan Alexander Steffens (heftig),
+Jan Schmidt, Jason Litzinger, Jens Georg, Jimmy Ohn, Joan Pau Beltran,
+Joe Gorse, John Chang, John Slade, Jose Antonio Santos Cadenas, Josep Torra,
+Julian Bouzas, Julien Isorce, Julien Moutte, Justin Kim, Kazunori Kobayashi,
+Koop Mast, Lim Siew Hoon, Linus Svensson, Lubosz Sarnecki, Luis de Bethencourt,
+Lukasz Forynski, Manasa Athreya, Marcel Holtmann, Marcin Kolny, Marcus Prebble,
+Mark Nauwelaerts, Maroš Ondrášek, Martin Kelly, Matej Knopp, Mathias Hasselmann,
+Mathieu Duponchelle, Matt Crane, Matthew Marsh, Matthew Waters, Matthieu Bouron,
+Mersad Jelacic, Michael Olbrich, Miguel París Díaz, Mikhail Fludkov,
+Mischa Spiegelmock, Nicola Murino, Nicolas Dufresne, Nicolas Huet,
+Nirbheek Chauhan, Ognyan Tonchev, Olivier Crête, Pablo Anton, Pankaj Darak,
+Paolo Pettinato, Patricia Muscalu, Paul Arzelier, Pavel Bludov, Perry Hung,
+Peter Korsgaard, Peter Seiderer, Petr Viktorin, Philippe Normand,
+Philippe Renon, Philipp Zabel, Philip Van Hoof, Philip Withnall, Piotr Drąg,
+plamot, Polochon\_street, Prashant Gotarne, Rajat Verma, Ramiro Polla,
+Ravi Kiran K N, Reynaldo H. Verdejo Pinochet, Robert Swain, Romain Picard,
+Roman Nowicki, Ross Burton, Ryan Hendrickson, Santiago Carot-Nemesio,
+Scott D Phillips, Sebastian Dröge, Sebastian Rasmussen, Sergey Borovkov,
+Seungha Yang, Sjors Gielen, Song Bing, Sreerenj Balachandran, Srimanta Panda,
+Stavros Vagionitis, Stefan Sauer, Steven Hoving, Stian Selnes, Suhwang Kim,
+Thiago Santos, Thibault Saunier, Thijs Vermeir, Thomas Bluemel, Thomas Roos,
+Thomas Vander Stichele, Tim-Philipp Müller, Tim Sheridan, Ting-Wei Lan,
+Tom Deseyn, Vanessa Chipirrás Navalón, Víctor Manuel Jáquez Leal,
+Vincent Dehors, Vincent Penquerc'h, Vineeth T M, Vivia Nikolaidou,
+Wang Xin-yu (王昕宇), William Manley, Wim Taymans, Wonchul Lee, Xavi Artigas,
+Xavier Claessens, Youness Alaoui,
+
+... and many others who have contributed bug reports, translations, sent
+suggestions or helped testing.
+
+## Bugs fixed in 1.8
+
+More than [~700 bugs][bugs-fixed-in-1.8] have been fixed during
+the development of 1.8.
+
+This list does not include issues that have been cherry-picked into the
+stable 1.6 branch and fixed there as well, all fixes that ended up in the
+1.6 branch are also included in 1.8.
+
+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-1.8]: https://bugzilla.gnome.org/buglist.cgi?bug_status=RESOLVED&bug_status=VERIFIED&classification=Platform&limit=0&list_id=107311&order=bug_id&product=GStreamer&query_format=advanced&resolution=FIXED&target_milestone=1.6.1&target_milestone=1.6.2&target_milestone=1.6.3&target_milestone=1.7.0&target_milestone=1.7.1&target_milestone=1.7.2&target_milestone=1.7.3&target_milestone=1.7.4&target_milestone=1.7.90&target_milestone=1.7.91&target_milestone=1.7.92&target_milestone=1.7.x&target_milestone=1.8.0
+
+## Stable 1.8 branch
+
+After the 1.8.0 release there will be several 1.8.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 1.8.x bug-fix releases will be made from the git 1.8 branch, which
+is a stable branch.
+
+### 1.8.0
+
+1.8.0 was released on 24 March 2016.
+
+### 1.8.1
+
+The first 1.8 bug-fix release (1.8.1) is planned for April 2016.
+
+## Schedule for 1.10
+
+Our next major feature release will be 1.10, and 1.9 will be the unstable
+development version leading up to the stable 1.10 release. The development
+of 1.9/1.10 will happen in the git master branch.
+
+The plan for the 1.10 development cycle is yet to be confirmed, but it is
+expected that feature freeze will be around late July or early August,
+followed by several 1.9 pre-releases and the new 1.10 stable release
+in September.
+
+1.10 will be backwards-compatible to the stable 1.8, 1.6, 1.4, 1.2 and 1.0
+release series.
+
+- - -
+
+*These release notes have been prepared by Tim-Philipp Müller with
+contributions from Sebastian Dröge, Nicolas Dufresne, Edward Hervey, Víctor
+Manuel Jáquez Leal, Arun Raghavan, Thiago Santos, Thibault Saunier, Jan
+Schmidt and Matthew Waters.*
+
+*License: [CC BY-SA 4.0](http://creativecommons.org/licenses/by-sa/4.0/)*
projects / platform / upstream / gstreamer.git / blobdiff