diff options
Diffstat (limited to 'RELEASE')
-rw-r--r-- | RELEASE | 140 |
1 files changed, 40 insertions, 100 deletions
@@ -1,5 +1,5 @@ -Release notes for GStreamer Base Plug-ins 0.10.15 "No need to argue" +Release notes for GStreamer Base Plug-ins 0.10.16 "Scheduled Interruption" @@ -54,89 +54,47 @@ contains a set of less supported plug-ins that haven't passed the Features of this release - * 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 - * + * 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 in this release - * 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 + * 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 changed in this release - API additions: -* 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 +* 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() Download @@ -166,40 +124,22 @@ Applications Contributors to this release - * Stefan Kost - * Alexander Shopov - * Damien Lespiau - * Dan Williams - * Daniel Díaz + * Bastien Nocera * David Schleef - * Davyd Madeley - * Funda Wang - * Haakon Sporsheim - * Ilkka Tuohela - * Jakub Bogusz + * Edward Hervey * Jan Schmidt - * Jason Kivlighn - * Jens Granseuer - * Johan Dahlin - * Jorge González González - * Josep Torra Valles + * Jerone Young + * Joe Peterson * Julien MOUTTE - * Laurent Glayal + * Julien Moutte * Michael Smith - * Mogens Jaeger - * Ole André Vadla Ravnås - * Olivier Crete * Peter Kjellerstedt - * Renato Filho - * René Stadler + * Robin Stocker * Sebastian Dröge * Sebastien Moutte * Stefan Kost * Thijs Vermeir - * Thomas Vander Stichele * Tim-Philipp Müller * Tommi Myöhänen - * Vincent Torri * Wim Taymans - * Yang Hong
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