November 2012 Archives by date

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Starting: Thu Nov 1 04:01:01 PDT 2012
Ending: Fri Nov 30 15:19:00 PST 2012
Messages: 631

• [0.10] gst-plugins-good: depay: correctly skip caps header size   Wim Taymans
• [0.10] gst-plugins-good: gstpay: add debug   Wim Taymans
• [0.10] gst-plugins-good: gstpay: send caps inline   Wim Taymans
• [0.10] gst-plugins-good: gstdepay: use the capsversion   Wim Taymans
• gst-plugins-good: depay: correctly skip caps header size   Wim Taymans
• gst-plugins-good: gstpay: add debug   Wim Taymans
• gst-plugins-good: gstpay: send caps inline   Wim Taymans
• gst-plugins-good: gstdepay: use the capsversion   Wim Taymans
• [0.10] gst-plugins-good: gstpay: set C flag on all buffers of the fragment   Wim Taymans
• [0.10] gst-plugins-good: gstdepay: check for correct fragment offset   Wim Taymans
• gst-plugins-good: gstpay: set C flag on all buffers of the fragment   Wim Taymans
• gst-plugins-good: gstdepay: check for correct fragment offset   Wim Taymans
• gst-plugins-bad: eglglessink: Post stream-status messages for our render thread   Sebastian Dröge
• [0.10] gst-plugins-bad: eglglessink: Post stream-status messages for our render thread   Sebastian Dröge
• gst-plugins-bad: androidmedia: Skip Aricent codecs for now   Sebastian Dröge
• [0.10] gst-plugins-bad: androidmedia: Skip Aricent codecs for now   Sebastian Dröge
• gst-plugins-good: videbox: fix border filling for gray formats   Tim Müller
• gst-plugins-good: videomixer: get height via GST_VIDEO_FRAME_HEIGHT, not _WIDTH   Tim Müller
• gst-plugins-base: audioconvert: Prefer output formats with the same depth or at least a higher depth   Sebastian Dröge
• gst-plugins-base: audioconvert: Also ignore the SIGNED flag when matching an output format   Sebastian Dröge
• gst-plugins-bad: opensles: Make sure to only ever create a single engine object   Sebastian Dröge
• gst-plugins-bad: opensles: Remove unused variable   Sebastian Dröge
• [0.10] gst-plugins-bad: opensles: Make sure to only ever create a single engine object   Sebastian Dröge
• [0.10] gst-plugins-bad: opensles: Remove unused variable   Sebastian Dröge
• gst-plugins-base: audioconvert: Always prefer the input format if possible   Sebastian Dröge
• [0.10] gst-plugins-good: gstpay: rewrite payloader   Wim Taymans
• [0.10] gst-plugins-good: rtpgstpay: add support for sending events   Wim Taymans
• [0.10] gst-plugins-good: gstdepay: add support for events   Wim Taymans
• [0.10] gst-plugins-bad: avfvideosrc: Allow changing the framerate through the caps   Olivier Crête
• gst-plugins-good: gstpay: rewrite payloader   Wim Taymans
• gst-plugins-good: rtpgstpay: add support for sending events   Wim Taymans
• gst-plugins-good: gstdepay: add support for events   Wim Taymans
• gst-plugins-good: gstdepay: fix small leak   Wim Taymans
• gst-plugins-good: gstpay: fix for 1.0 events   Wim Taymans
• gst-plugins-bad: rtpvp8: remove and move to -good   Tim Müller
• gst-plugins-good: rtpvp8: Add simple payloaders and depayloaders for VP8   Tim Müller
• gst-plugins-good: rtpvp8pay: Treat the frame header just like any other partition   Tim Müller
• gst-plugins-good: rtpvp8depay: Accept packets with only one byte of data   Tim Müller
• gst-plugins-good: rtpvp8: Fix unitialized variable   Tim Müller
• gst-plugins-good: rtpvp8: Reject unknown bitstream versions   Tim Müller
• gst-plugins-good: rtpvp8: fix bitstream parsing using the wrong kind of bitreader   Tim Müller
• gst-plugins-good: rtpvp8: Update the pay/depay to the ietf-draft-01 spec   Tim Müller
• gst-plugins-good: rtpvp8pay: Fix typo   Tim Müller
• gst-plugins-good: rtpvp8: port to 0.11   Tim Müller
• gst-plugins-good: rtpvp8: port some more to new memory API   Tim Müller
• gst-plugins-good: rtpvp8; fix compatibility with the third draft   Tim Müller
• gst-plugins-good: rtpvp8: update for buffer changes   Tim Müller
• gst-plugins-good: rtpvp8: update for GST_PLUGIN_DEFINE() API changes   Tim Müller
• gst-plugins-good: rtpvp8: replace gst_element_class_set_details_simple with gst_element_class_set_metadata   Tim Müller
• gst-plugins-good: rtpvp8: use gst_element_class_set_static_metadata()   Tim Müller
• gst-plugins-good: rtp: move VP8 payloader and depayloader from -bad   Tim Müller
• gst-plugins-good: rtp: fix tabs/space mess in Makefile.am   Tim Müller
• gst-plugins-good: rtpvp8: include config.h and minor style fixes   Tim Müller
• gst-plugins-good: videoconvert: Compare y offset with height, not width, when testing for overlap   Sebastian Dröge
• [1.0] gst-plugins-good: videoconvert: Compare y offset with height, not width, when testing for overlap   Sebastian Dröge
• [1.0] gst-plugins-good: videbox: fix border filling for gray formats   Sebastian Dröge
• [1.0] gst-plugins-good: videomixer: get height via GST_VIDEO_FRAME_HEIGHT, not _WIDTH   Sebastian Dröge
• gst-plugins-good: vp8dec: Short circuit gst_vp8_dec_handle_frame if keyframe is missing   Sebastian Dröge
• gst-plugins-good: vp8dec: Immediately return if opening the decoder failed   Sebastian Dröge
• [1.0] gst-plugins-good: vp8dec: Short circuit gst_vp8_dec_handle_frame if keyframe is missing   Sebastian Dröge
• [1.0] gst-plugins-good: vp8dec: Immediately return if opening the decoder failed   Sebastian Dröge
• gst-plugins-bad: opensles: Fix compilation with debugging enabled   Sebastian Dröge
• [0.10] gst-plugins-bad: opensles: Fix compilation with debugging enabled   Sebastian Dröge
• [0.10] gst-plugins-bad: opensles: Use GLib 2.24 thread API   Sebastian Dröge
• [0.10] gst-plugins-bad: opensles: Use GST_PLUGIN_DEFINE2 only when available   Sebastian Dröge
• [0.10] gst-plugins-bad: eglglessink: Use GST_PLUGIN_DEFINE2 if available and use some more #defines instead of hardcoding values   Sebastian Dröge
• gst-plugins-bad: eglglessink: Use more constants instead of hardcoding values   Sebastian Dröge
• [0.10] gst-rtsp-server: disable deprecation warnings   Wim Taymans
• gst-plugins-bad: scaletempo: Use a better plugin filename   Sebastian Dröge
• gst-plugins-base: rtspconnection: fix g-i annotations for out parameters   Tim Müller
• [1.0] gst-plugins-base: rtspconnection: fix g-i annotations for out parameters   Tim Müller
• [0.10] gst-plugins-base: ffmpegcolorspace: Add NV12/NV21 to I420 convertion   Olivier Crête
• gst-plugins-good: vrawdepay: don't access rtp buffer after unmap   Wim Taymans
• gst-plugins-base: rtspconnection: remove extra return and fix GError leak   Tim Müller
• gst-plugins-base: video-blend: fix memory leak when called with invalid parameters   Tim Müller
• gst-plugins-base: audiobasesink: use the same type as the internal type to return it   Tim Müller
• [1.0] gst-plugins-base: rtspconnection: remove extra return and fix GError leak   Tim Müller
• gst-plugins-bad: opencv: fix warnings and build against opencv 2.4.x   Tim Müller
• gnonlin: gnlcomposition: Fix leak   Edward Hervey
• gnonlin: events: Don't leak segments and carry-over seqnums   Edward Hervey
• gnonlin: composition: Ensure seeks always have flush/accurate   Edward Hervey
• gnonlin: composition: Only care about source pads that go away   Edward Hervey
• gnonlin: gnlghostpad: Control internal pad straight away   Edward Hervey
• gnonlin: gnloperation: Only release sink pads that need to be   Edward Hervey
• gnonlin: misc: Add more/better debugging   Edward Hervey
• gnonlin: check: Fix usage of probe   Edward Hervey
• gnonlin: check: Re-enable more tests   Edward Hervey
• [1.0] gst-plugins-good: vrawdepay: don't access rtp buffer after unmap   Tim Müller
• gstreamer: Fix FSF address   Tim Müller
• gst-plugins-base: Fix FSF address   Tim Müller
• gst-plugins-good: Fix FSF address   Tim Müller
• gst-plugins-ugly: Fix FSF address   Tim Müller
• gst-plugins-bad: Fix FSF address   Tim Müller
• gnonlin: Fix FSF address   Tim Müller
• gst-rtsp-server: Fix FSF address   Tim Müller
• common: Fix FSF address   Tim Müller
• gst-ffmpeg: Fix FSF address   Tim Müller
• gst-editing-services: Fix FSF address   Tim Müller
• insanity-gst: Fix FSF address   Tim Müller
• insanity: Fix FSF address   Tim Müller
• gst-python: overrides: move the Gst.Pad override before Gst.GhostPad   Alessandro Decina
• gst-python: overrides: move add(*args) override from Gst.Pipeline to Gst. Bin   Alessandro Decina
• [0.10] gst-plugins-bad: hlssink: Fix segment names on playlist   Tim Müller
• [0.10] gst-plugins-bad: hlssink: reset when going to NULL state   Tim Müller
• [0.10] gst-plugins-bad: hlssink: add a playlist-length property   Tim Müller
• [0.10] gst-plugins-bad: waylandsink: fail gracefully with an error message if we can't connect to wayland   Tim Müller
• gst-plugins-bad: hlssink: Fix segment names on playlist   Tim Müller
• gst-plugins-bad: hlssink: reset when going to NULL state   Tim Müller
• gst-plugins-bad: hlssink: add a playlist-length property   Tim Müller
• gst-plugins-base: rtsp: fix GstRTSPMessage g-i annotations for out parameters   Tim Müller
• gst-plugins-base: textoverlay: forward allocation queries on video sink pad instead of discarding them   Tim Müller
• [0.10] gst-plugins-good: jpegenc: error out gracefully if input buffer is too small   Tim Müller
• gst-plugins-ugly: dvdlpcmdec: Fix parsing of headers to set correct width to allow decode.   Michael Smith
• gst-plugins-base: rtp: add helpers for header extensions   Wim Taymans
• gst-plugins-base: rtp: fix ntp56 parsing   Wim Taymans
• gst-plugins-base: tests: add NTP64 and ntp56 header extension checks   Wim Taymans
• gst-plugins-base: win32: add new header extension methods   Wim Taymans
• gst-plugins-good: rtsp: the RTCP port number is inclusive   Wim Taymans
• gst-plugins-bad: h264parse: Actually add PPS data to the output buffer   Thibault Saunier
• gst-plugins-bad: h264parse: don' t rewind to the NAL length size in bytewriter if the format is bytestream   Thibault Saunier
• [1.0] gst-plugins-bad: h264parse: Actually add PPS data to the output buffer   Thibault Saunier
• [1.0] gst-plugins-bad: h264parse: don' t rewind to the NAL length size in bytewriter if the format is bytestream   Thibault Saunier
• gstreamer: elements: fix leading space in plugin description string   Tim Müller
• gstreamer: gst-inspect: fix alignment of rank, etc.   Tim Müller
• www: planet: update link in config.ini to fix atom feed url in html header   Tim Müller
• gstreamer: Document GST_DEBUG_FILE   Olivier Crête
• [0.10] gstreamer: Document GST_DEBUG_FILE   Olivier Crête
• gst-plugins-good: speexdec: Don' t unmap or finish_frame an invalid GstBuffer   Tim Müller
• [1.0] gst-plugins-good: rtsp: the RTCP port number is inclusive   Tim Müller
• [1.0] gst-plugins-good: speexdec: Don' t unmap or finish_frame an invalid GstBuffer   Tim Müller
• [1.0] gst-plugins-base: rtsp: fix GstRTSPMessage g-i annotations for out parameters   Tim Müller
• [1.0] gst-plugins-base: textoverlay: forward allocation queries on video sink pad instead of discarding them   Tim Müller
• gst-plugins-base: textoverlay: don't abort if we don' t know how to paint shaded background for a format   Tim Müller
• [1.0] gst-plugins-base: textoverlay: don't abort if we don' t know how to paint shaded background for a format   Tim Müller
• gst-plugins-base: typefind: isml is iso-fragmented video/quicktime   Thiago Sousa Santos
• [0.10] gst-plugins-base: typefind: isml is iso-fragmented video/quicktime   Thiago Sousa Santos
• gst-plugins-base: textoverlay: move background shading into separate function   Tim Müller
• gst-plugins-base: textoverlay: clamp shaded background box coordinates in one place   Tim Müller
• gst-plugins-base: textoverlay: draw shaded background for some more video formats   Tim Müller
• gst-plugins-base: textoverlay: fix up names of old gray formats   Tim Müller
• gst-plugins-good: videocrop: Add support for automatic cropping   Olivier Crête
• [0.10] gst-plugins-good: videocrop: Style fix   Olivier Crête
• [0.10] gst-plugins-good: videocrop: Add support for automatic cropping   Olivier Crête
• gst-plugins-base: video: don' t crash when blending onto video formats that unpack to 64 bits per pixel   Tim Müller
• gnonlin: check: Detect stray segments better   Edward Hervey
• gst-plugins-base: textoverlay: don't advertise 10-16-bit formats we can' t blend text onto yet   Tim Müller
• gst-plugins-base: textoverlay: we can do YVU9 as well   Tim Müller
• gst-plugins-good: Fix vp8rtp header names in Makefile   Christian Fredrik Kalager Schaller
• gst-plugins-base: textoverlay: also draw shaded backgrounds for RGB and BGR   Tim Müller
• gst-plugins-base: textoverlay: implement background shading for IYU1   Tim Müller
• common: gst-plugin-docs.m4: fix autogen.sh warnings   Tim Müller
• gstreamer: common: update for AG_GST_PLUGIN_DOCS python checks   Tim Müller
• gstreamer: configure: update courtesy of autoupdate   Tim Müller
• gst-plugins-base: configure: let AG_GST_PLUGIN_DOCS check for python   Tim Müller
• gst-plugins-base: configure.ac: update courtesy of autoupdate   Tim Müller
• gst-plugins-good: configure: let AG_GST_PLUGIN_DOCS check for python   Tim Müller
• gst-plugins-bad: mpegtsmux: fix DTS value being used for PTS   Tim Müller
• gst-plugins-good: configure.ac: update courtesy of autoupdate   Tim Müller
• gst-plugins-ugly: configure: let AG_GST_PLUGIN_DOCS check for python   Tim Müller
• gst-plugins-ugly: configure.ac: update courtesy of autoupdate   Tim Müller
• [1.0] gst-plugins-base: typefind: isml is iso-fragmented video/quicktime   Tim Müller
• [1.0] gst-plugins-base: textoverlay: move background shading into separate function   Tim Müller
• [1.0] gst-plugins-base: textoverlay: clamp shaded background box coordinates in one place   Tim Müller
• [1.0] gst-plugins-base: textoverlay: draw shaded background for some more video formats   Tim Müller
• [1.0] gst-plugins-base: textoverlay: fix up names of old gray formats   Tim Müller
• [1.0] gst-plugins-base: video: don' t crash when blending onto video formats that unpack to 64 bits per pixel   Tim Müller
• [1.0] gst-plugins-base: textoverlay: don' t advertise 10-16-bit formats we can't blend text onto yet   Tim Müller
• [1.0] gst-plugins-base: textoverlay: we can do YVU9 as well   Tim Müller
• [1.0] gst-plugins-base: textoverlay: also draw shaded backgrounds for RGB and BGR   Tim Müller
• [1.0] gst-plugins-ugly: dvdlpcmdec: Fix parsing of headers to set correct width to allow decode.   Tim Müller
• [1.0] gst-plugins-bad: mpegtsmux: fix DTS value being used for PTS   Tim Müller
• gst-plugins-bad: configure: let AG_GST_PLUGIN_DOCS check for python   Tim Müller
• gst-plugins-bad: configure.ac: update courtesy of autoupdate   Tim Müller
• gstreamer: bufferpool: lock before unlock in _get_config   Wim Taymans
• [1.0] gstreamer: bufferpool: lock before unlock in _get_config   Wim Taymans
• gst-plugins-bad: docs: Add missing plugin inspection files   Sebastian Dröge
• gst-plugins-good: udpsrc: Don' t error out if we get an ICMP destination-unreachable message when trying to read packets   Sebastian Dröge
• gst-plugins-good: udpsrc: Also clear GError   Sebastian Dröge
• gst-plugins-ugly: xingmux: Properly calculate the number of frames   Sebastian Dröge
• gst-plugins-bad: mpegts: Only use timestamp-based PCR/ PTS estimation when needed   Edward Hervey
• [1.0] gst-plugins-bad: mpegts: Only use timestamp-based PCR/ PTS estimation when needed   Edward Hervey
• gst-plugins-good: vp8dec: Don' t give up so easily if failed to decode a frame   Sebastian Dröge
• gst-plugins-base: videodecoder: Reset the error count to 0 after successfully decoding a frame   Sebastian Dröge
• gst-plugins-base: audiodecoder: Reset error count to 0 after successfully decoding a frame   Sebastian Dröge
• gstreamer: basesink: add simple rate control   Wim Taymans
• gstreamer: queue: don't fail in _sink_event for sticky events   Alessandro Decina
• gst-plugins-base: rtsp: fix g-i annotation for gst_rtsp_message_set_body(), take_body() and take_header()   Tim Müller
• gst-plugins-bad: tspacketizer: ensure that a null pointer isn't accessed   Josep Torra
• gst-plugins-bad: tsbase: add a guard with an atomic boolean when flushing   Josep Torra
• gst-plugins-good: videocrop: Add NV12/NV21 support   Olivier Crête
• [0.10] gst-plugins-good: videocrop: Add support for NV12/NV21   Olivier Crête
• gstreamer: queue: remove unused label. Fixes compiler warning.   Alessandro Decina
• gst-plugins-bad: tsbase: add a guard with an atomic boolean when flushing   Edward Hervey
• gst-plugins-good: multifilesink: don' t write stream headers with key-unit-event   Alessandro Decina
• gst-plugins-bad: hlssink: rework _reset a bit   Alessandro Decina
• gst-plugins-bad: hlssink: don't unref ->multifilesink.   Alessandro Decina
• gst-plugins-bad: hlssink: cut segments using the running-time   Alessandro Decina
• gst-plugins-bad: hlssink: port to 1.0   Alessandro Decina
• gst-plugins-bad: hlssink: remove vim . swp files that I just committed accidentally...   Alessandro Decina
• gst-plugins-base: tests: add test for video unpack and pack functions   Tim Müller
• gst-plugins-good: videocrop: add support for YV12   Tim Müller
• gst-plugins-bad: Revert "tsbase: add a guard with an atomic boolean when flushing"   Josep Torra
• gst-plugins-bad: Revert "tspacketizer: ensure that a null pointer isn' t accessed"   Josep Torra
• gst-plugins-base: video: Add Y444_10{LE,BE} video formats   Sebastian Dröge
• gst-plugins-base: video: Add GBR/GBR_10LE/GBR_10BE color formats   Sebastian Dröge
• gst-ffmpeg: avcodecmap: Add mappings for the GBR color formats   Sebastian Dröge
• gst-plugins-base: typefinding: fix block size calculation in wavpack typefinder   Tim Müller
• gst-plugins-base: typefinding: improve wavpack typefinder   Tim Müller
• gst-plugins-base: typefinding: improve AAC LOAS typefinding   Tim Müller
• gst-plugins-ugly: cdio: use gst_tag_freeform_string_to_utf8() for CD-TEXT strings   Tim Müller
• gstreamer: info: fix compiler warning when debugging disabled   Sebastian Dröge
• gst-plugins-good: vp8dec: Don't clear input state in reset()   Sebastian Dröge
• gst-plugins-good: vp8dec: Also destroy decoder in set_format() if it was created already   Sebastian Dröge
• gst-plugins-good: pngdec: Keep the input state in reset()   Sebastian Dröge
• [1.0] gst-plugins-good: pngdec: Keep the input state in reset()   Sebastian Dröge
• [1.0] gst-plugins-good: vp8dec: Also destroy decoder in set_format() if it was created already   Sebastian Dröge
• [1.0] gst-plugins-good: vp8dec: Don't clear input state in reset()   Sebastian Dröge
• [1.0] gst-plugins-good: vp8dec: Don' t give up so easily if failed to decode a frame   Sebastian Dröge
• [1.0] gst-plugins-good: udpsrc: Don' t error out if we get an ICMP destination-unreachable message when trying to read packets   Sebastian Dröge
• [1.0] gst-plugins-good: udpsrc: Also clear GError   Sebastian Dröge
• [1.0] gst-plugins-base: audioconvert: Prefer output formats with the same depth or at least a higher depth   Sebastian Dröge
• [1.0] gst-plugins-base: audioconvert: Also ignore the SIGNED flag when matching an output format   Sebastian Dröge
• [1.0] gst-plugins-base: audioconvert: Always prefer the input format if possible   Sebastian Dröge
• [1.0] gst-plugins-base: videodecoder: Reset the error count to 0 after successfully decoding a frame   Sebastian Dröge
• [1.0] gst-plugins-base: audiodecoder: Reset error count to 0 after successfully decoding a frame   Sebastian Dröge
• [1.0] gst-plugins-ugly: xingmux: Properly calculate the number of frames   Sebastian Dröge
• [1.0] gst-plugins-base: typefinding: improve wavpack typefinder   Sebastian Dröge
• [1.0] gst-plugins-base: typefinding: fix block size calculation in wavpack typefinder   Sebastian Dröge
• [1.0] gst-plugins-base: typefinding: improve AAC LOAS typefinding   Sebastian Dröge
• gstreamer: gst-inspect: Fix indention for printing typefinder features   Sebastian Dröge
• gst-plugins-good: vp8dec: Fix last commit   Sebastian Dröge
• [1.0] gst-plugins-good: vp8dec: Fix last commit   Sebastian Dröge
• gst-plugins-good: pngdec: Actually use the stop() vfunc implementation   Sebastian Dröge
• [1.0] gst-plugins-good: pngdec: Actually use the stop() vfunc implementation   Sebastian Dröge
• gst-plugins-base: rtpbuffer: protect against empty buffers   Wim Taymans
• gst-omx: omx: Update to new GLib thread API   Sebastian Dröge
• gst-plugins-bad: androidmedia: Update to new GLib thread API   Sebastian Dröge
• gstreamer: value: API: Add boxed type for GThread   Sebastian Dröge
• gst-plugins-good: rtpsource: protect against invalid RTP packets   Wim Taymans
• gst-plugins-base: audio: Use new GType for GThread instead of just G_TYPE_POINTER   Sebastian Dröge
• gst-plugins-good: pulse: Use new GType for GThread instead of just G_TYPE_POINTER   Sebastian Dröge
• gst-plugins-bad: eglglessink: Use new GType for GThread instead of just G_TYPE_POINTER   Sebastian Dröge
• gst-plugins-good: vp8enc: Don' t leak GstVideoCodecFrames that cause the creation of invisible frames   Sebastian Dröge
• [1.0] gst-plugins-good: vp8enc: Don' t leak GstVideoCodecFrames that cause the creation of invisible frames   Sebastian Dröge
• gst-plugins-base: pbutils: add description for Opus audio codec   Tim Müller
• gst-plugins-base: rtspconnection: improve docs   Wim Taymans
• [0.10] gst-plugins-bad: amcvideodec: Fix refcount problem with frames being dropped because of decoder bugs   Sebastian Dröge
• gst-plugins-bad: amcvideodec: Fix refcount problem with frames being dropped because of decoder bugs   Sebastian Dröge
• gst-omx: omx: Fix refcount problem with frames being dropped because of decoder bugs   Sebastian Dröge
• gst-plugins-bad: mpegtsbase: flush in FLUSH_STOP to avoid race conditions   Josep Torra
• gst-plugins-bad: mpegtsbase: fix double unref of seek event   Josep Torra
• gst-plugins-bad: mpegtspacketizer: flush observations too   Josep Torra
• gst-plugins-bad: tsdemux: reset the stream some more in the flushes   Josep Torra
• gst-rtsp-server: session: move session header code in session object   Wim Taymans
• gst-rtsp-server: server: rework maincontext handling in clients   Wim Taymans
• gst-rtsp-server: client: rename method   Wim Taymans
• gst-rtsp-server: server: use a threadpool for the mainloops   Wim Taymans
• gst-rtsp-server: server: add max-thread property   Wim Taymans
• gst-rtsp-server: auth: add locking   Wim Taymans
• gst-rtsp-server: media-factory: add locking   Wim Taymans
• gst-rtsp-server: mapping: add locking   Wim Taymans
• gst-rtsp-server: server: free old socket   Wim Taymans
• gst-rtsp-server: session: add locking   Wim Taymans
• gst-rtsp-server: session-media: add locking   Wim Taymans
• gst-rtsp-server: stream-transport: add method to handle RTP/RTCP   Wim Taymans
• gst-rtsp-server: stream-transport: add keep-alive method   Wim Taymans
• gst-rtsp-server: stream: add locking   Wim Taymans
• gst-rtsp-server: media: add lock to protect state changes   Wim Taymans
• gst-rtsp-server: media: add lock around message handler   Wim Taymans
• gst-rtsp-server: media: wait for concurrent _prepare   Wim Taymans
• gst-rtsp-server: server: set default max-threads property   Wim Taymans
• [0.10] gst-plugins-good: videoflip: Add NV12/NV21 support   Olivier Crête
• gst-plugins-good: videoflip: Add NV12/NV21 support   Olivier Crête
• gst-plugins-base: vorbisdec: Fix GType name conflict if tremor and libvorbis decoder are used in the same process   Sebastian Dröge
• [1.0] gst-plugins-base: vorbisdec: Fix GType name conflict if tremor and libvorbis decoder are used in the same process   Sebastian Dröge
• gst-plugins-base: Revert "vorbisdec: Fix GType name conflict if tremor and libvorbis decoder are used in the same process "   Sebastian Dröge
• [1.0] gst-plugins-base: Revert "vorbisdec: Fix GType name conflict if tremor and libvorbis decoder are used in the same process "   Sebastian Dröge
• gst-plugins-base: video-format: fix plane offsets for GBR formats   Wim Taymans
• gstreamer: structure: re-indent header file   Tim Müller
• gstreamer: basesrc: fix debug message   Tim Müller
• gstreamer: check: add GstTestClock as a deterministic clock for testing   Tim Müller
• gstreamer: check: add dependency on gstcheck header files for exports.sym   Tim Müller
• gstreamer: check: allow GstTestClock to handle clock notifications   Tim Müller
• gstreamer: testclock: minor cleanups, add since markers for gtk-doc   Tim Müller
• gst-plugins-bad: accurip: fix linker flags   Tim Müller
• gstreamer: tests: gst_adapter_prev_timestamp -> gst_adapter_prev_pts   Tim Müller
• gstreamer: testclock: port to new GLib threading API   Tim Müller
• gst-plugins-base: gst_adapter_prev_timestamp -> gst_adapter_prev_pts   Tim Müller
• gst-plugins-good: gst_adapter_prev_timestamp -> gst_adapter_prev_pts   Tim Müller
• gstreamer: task: documentation update   Sebastian Dröge
• gst-plugins-bad: hls: dist all header files needed to build   Tim Müller
• gst-plugins-bad: gst_adapter_prev_timestamp -> gst_adapter_prev_pts   Tim Müller
• gst-ffmpeg: gst_adapter_prev_timestamp -> gst_adapter_prev_pts   Tim Müller
• gst-plugins-bad: tsdemux: forward upstream time segments after flushes   Josep Torra
• [1.0] gst-plugins-bad: mpegtsbase: flush in FLUSH_STOP to avoid race conditions   Tim Müller
• [1.0] gst-plugins-bad: mpegtsbase: fix double unref of seek event   Tim Müller
• [1.0] gst-plugins-bad: mpegtspacketizer: flush observations too   Tim Müller
• [1.0] gst-plugins-bad: tsdemux: reset the stream some more in the flushes   Tim Müller
• [1.0] gst-plugins-bad: tsdemux: forward upstream time segments after flushes   Tim Müller
• gst-plugins-good: udpsrc: post error before stopping   Wim Taymans
• [fdomaster] gst-plugins-bad: mpeg4videoparse: Make sure pad template caps are not fixed from the begging   Thibault Saunier
• [1.0] gst-plugins-bad: mpeg4videoparse: Make sure pad template caps are not fixed from the begging   Thibault Saunier
• gst-plugins-bad: mpeg4videoparse: Make sure pad template caps are not fixed from the begging   Thibault Saunier
• gst-plugins-bad: mpeg4videoparse: Fix nitpick   Thibault Saunier
• [1.0] gst-plugins-bad: mpeg4videoparse: Fix nitpick   Thibault Saunier
• [0.10] gst-plugins-base: video: Add support for planar 10 bit YUV color formats   Sebastian Dröge
• [0.10] gst-plugins-base: ffmpegcolorspace: Add support for 10 bit YUV color formats   Sebastian Dröge
• [1.0] gst-plugins-good: rtpsource: protect against invalid RTP packets   Tim Müller
• [1.0] gst-plugins-good: udpsrc: post error before stopping   Tim Müller
• [1.0] gst-plugins-base: rtpbuffer: protect against empty buffers   Tim Müller
• [1.0] gst-plugins-base: pbutils: add description for Opus audio codec   Tim Müller
• [1.0] gst-plugins-base: rtspconnection: improve docs   Tim Müller
• [1.0] gst-plugins-ugly: cdio: use gst_tag_freeform_string_to_utf8() for CD-TEXT strings   Tim Müller
• gst-plugins-ugly: x264enc: Add support for Y444, Y42B and NV12   Sebastian Dröge
• gst-plugins-ugly: x264enc: Add support for 10-bit color formats   Sebastian Dröge
• gst-rtsp-server: address-pool: add object to manage multicast addresses   Wim Taymans
• gst-rtsp-server: tests: add addresspool unit test   Wim Taymans
• gst-rtsp-server: address-pool: small cleanups   Wim Taymans
• gst-rtsp-server: address-pool: add clear method   Wim Taymans
• gst-rtsp-server: rtsp: use AddressPool   Wim Taymans
• gst-rtsp-server: examples: add multicast example   Wim Taymans
• gst-rtsp-server: rtsp: make address-pool return an address object   Wim Taymans
• gst-rtsp-server: stream: share src and sink sockets   Wim Taymans
• gst-rtsp-server: client: set blocksize only on stream   Wim Taymans
• gst-rtsp-server: media: remove MTU property   Wim Taymans
• gst-rtsp-server: stream: add methods to deal with address pool   Wim Taymans
• gst-rtsp-server: media: configure address pool in new streams   Wim Taymans
• gst-rtsp-server: media: add signal for new streams   Wim Taymans
• gst-rtsp-server: rtsp: improve debug   Wim Taymans
• gst-rtsp-server: stream: use the address managed by the stream   Wim Taymans
• gst-rtsp-server: test: set shared   Wim Taymans
• gst-rtsp-server: examples: add another multicast example   Wim Taymans
• gst-rtsp-server: rtsp: cleanups   Wim Taymans
• gst-plugins-good: multifilesink: post messages in max-size mode as well   Tim Müller
• gst-plugins-good: rtspsrc: add more debug   Wim Taymans
• gst-plugins-good: rtspsrc: propose ports in multicast   Wim Taymans
• gst-plugins-good: rtspsrc: clear variables before retrying   Wim Taymans
• gst-plugins-good: rtspsrc: allow client to disable reconnection   Wim Taymans
• gst-plugins-ugly: xingmux: Fix unit test after change to correctly calculate number of frames   Sebastian Dröge
• gst-plugins-good: rtspsrc: create and add pads outside of lock   Wim Taymans
• gst-plugins-good: rtspsrc: fix check for active streams   Wim Taymans
• gst-plugins-good: rtspsrc: handle segment event   Wim Taymans
• [1.0] gst-plugins-good: rtspsrc: add more debug   Wim Taymans
• [1.0] gst-plugins-good: rtspsrc: propose ports in multicast   Wim Taymans
• [1.0] gst-plugins-good: rtspsrc: clear variables before retrying   Wim Taymans
• [1.0] gst-plugins-good: rtspsrc: create and add pads outside of lock   Wim Taymans
• [1.0] gst-plugins-good: rtspsrc: fix check for active streams   Wim Taymans
• [1.0] gst-plugins-good: rtspsrc: handle segment event   Wim Taymans
• [1.0] gst-plugins-ugly: xingmux: Fix unit test after change to correctly calculate number of frames   Tim Müller
• [1.0] gst-plugins-good: multifilesink: post messages in max-size mode as well   Tim Müller
• gst-editing-services: ges-track-effect: don't use deprecated API   Tim Müller
• gst-rtsp-server: rtsp-client: fix unused-but-set-variable compiler warning   Tim Müller
• gst-rtsp-server: rtsp-server: don't use deprecated API   Tim Müller
• gst-plugins-base: examples: don't use deprecated API   Tim Müller
• gstreamer: gstcompat.h: move more deprecated API into the deprecated section   Tim Müller
• gstreamer: inputselector: fix clock leak in wait_running_time   Tim Müller
• [1.0] gstreamer: inputselector: fix clock leak in wait_running_time   Tim Müller
• gstreamer: examples: don' t compile testrtpool example if pthreads are not available like on win32   Tim Müller
• [1.0] gstreamer: examples: don' t compile testrtpool example if pthreads are not available like on win32   Tim Müller
• [0.10] gst-ffmpeg: ffmpegcolorspace: Fix shifts, we' re having 10 bits per component, not 16   Sebastian Dröge
• [0.10] gst-plugins-base: ffmpegcolorspace: Fix shifts, we' re having 10 bits per component, not 16   Sebastian Dröge
• [0.10] gst-ffmpeg: libav: add decode support for prores   Sebastian Dröge
• gst-ffmpeg: av: Put declarations in header files and rename audio codec files   Sebastian Dröge
• [0.10] gst-ffmpeg: ffmpeg: Fix merge mistake   Sebastian Dröge
• common: gstdoc-scangobj: don't use deprecated API   Tim Müller
• gstreamer: Automatic update of common submodule   Tim Müller
• gst-plugins-base: Automatic update of common submodule   Tim Müller
• gst-plugins-good: Automatic update of common submodule   Tim Müller
• gst-plugins-bad: Automatic update of common submodule   Tim Müller
• gst-plugins-ugly: Automatic update of common submodule   Tim Müller
• gst-ffmpeg: Automatic update of common submodule   Tim Müller
• gst-omx: Automatic update of common submodule   Tim Müller
• gnonlin: Automatic update of common submodule   Tim Müller
• gst-plugins-gl: Automatic update of common submodule   Tim Müller
• gst-rtsp-server: Automatic update of common submodule   Tim Müller
• gst-editing-services: Automatic update of common submodule   Tim Müller
• gstreamer: Add SVG versions of .fig file   Christian Fredrik Kalager Schaller
• gstreamer: Merge branch 'master' of ssh://git.freedesktop.org/git/gstreamer /gstreamer   Christian Fredrik Kalager Schaller
• gstreamer: Remove two last .fig files from build, they are now replaced with .svg files. That said I don't think either . fig file was still being used anywhere. With this   Christian Fredrik Kalager Schaller
• gst-plugins-bad: codecparsers: h264: fix error code for invalid size parsed in SPS.   Tim Müller
• common: Remove check for xfig   Christian Fredrik Kalager Schaller
• gst-plugins-base: range: handle ranges starting with -   Wim Taymans
• gst-plugins-base: range: a single - is not allowed   Wim Taymans
• gst-plugins-base: check: add rtsp range checks   Wim Taymans
• gst-plugins-bad: codecparsers: vc1: fix PTYPE for interlaced frames.   Gwenole Beauchesne
• [0.10] gst-plugins-bad: codecparsers: vc1: fix PTYPE for interlaced frames.   Gwenole Beauchesne
• [1.0] gst-plugins-bad: codecparsers: vc1: fix PTYPE for interlaced frames.   Gwenole Beauchesne
• gst-ffmpeg: avauddec: Port to audio base classes   Sebastian Dröge
• gst-ffmpeg: avcodecmap: Y41B is YUV411P, not YUV410P   Sebastian Dröge
• [1.0] gst-ffmpeg: avcodecmap: Y41B is YUV411P, not YUV410P   Sebastian Dröge
• [0.10] gst-plugins-good: qtdemux: add support for prores   Sebastian Dröge
• [0.10] gst-plugins-good: qtdemux: support more different fourcc values for other ProRes variants.   Sebastian Dröge
• [0.10] gst-plugins-good: qtdemux: allow more streams. Bump this constant to 32, which should be   Sebastian Dröge
• [0.10] gst-plugins-good: qtdemux: read video format header fully ( so we can find 'pasp' atoms) for more fourccs.   Sebastian Dröge
• [0.10] gst-ffmpeg: avcodecmap: only restrict caps on encoders   Sebastian Dröge
• [0.10] gst-ffmpeg: ffmpegcodecmap: Y41B is YUV411P not YUV410P   Sebastian Dröge
• gst-plugins-base: rtsp: detect npt time parse errors   Wim Taymans
• gst-plugins-base: range: handle parse errors better   Wim Taymans
• gst-plugins-base: rtsp: parse SMPTE ranges   Wim Taymans
• gst-plugins-base: rtsp: parse UTC ranges   Wim Taymans
• gst-plugins-base: rtsp: fix format string   Wim Taymans
• gst-plugins-base: videodecoder: add getter for QoS proportion and earliest_time   Tim Müller
• [1.0] gst-plugins-base: videodecoder: add getter for QoS proportion and earliest_time   Tim Müller
• gst-plugins-base: pbutils: fix transfer annotation for gst_encoding_profile_set_restriction   Alessandro Decina
• gst-rtsp-server: rtsp-stream: plug socket leak   Wim Taymans
• gst-rtsp-server: client: wait until the TEARDOWN response is sent to close the connection   Wim Taymans
• gst-rtsp-server: rtsp-media: remove bus watch before finalizing   Wim Taymans
• gst-rtsp-server: media: unref source in finish_unprepare   Wim Taymans
• [1.0] gst-plugins-base: pbutils: fix transfer annotation for gst_encoding_profile_set_restriction   Tim Müller
• gst-plugins-bad: GstPhotography: Add new enum values to match android' s parameters   Tim Müller
• [0.10] gst-plugins-bad: GstPhotography: Add new enum values to match android's parameters   Tim Müller
• gst-ffmpeg: av: Use PROP_ instead of ARG_   Sebastian Dröge
• gst-ffmpeg: avvidenc: Use gst_caps_truncate()   Sebastian Dröge
• gst-ffmpeg: avaudenc: Port to audio encoder base class   Sebastian Dröge
• gst-ffmpeg: avaudenc: Use non-deprecated avcodec_encode_audio2() API   Sebastian Dröge
• gst-ffmpeg: avaudenc: Some minor cleanup   Sebastian Dröge
• gst-plugins-base: rtsp: avoid ABI break   Wim Taymans
• gst-plugins-base: check: update for larger struct   Wim Taymans
• gst-rtsp-server: server: fix small leak   Wim Taymans
• gst-rtsp-server: server: start and stop multiple times   Wim Taymans
• gst-rtsp-server: test: add test for server reuse   Wim Taymans
• gst-ffmpeg: avauddec: Port to non-deprecated avcodec_decode_audio4() API   Sebastian Dröge
• gst-ffmpeg: avaudenc: Implement flush vfunc properly   Sebastian Dröge
• gst-ffmpeg: avaudenc: Implement draining   Sebastian Dröge
• gst-ffmpeg: avvidenc: Don' t send any frames downstream when draining unless requested to do so   Sebastian Dröge
• gst-ffmpeg: avvidenc: Implement reset vfunc   Sebastian Dröge
• gst-plugins-base: videodecoder: Rename from get_qos_info() to get_qos_proportion()   Sebastian Dröge
• [1.0] gst-plugins-base: videodecoder: Rename from get_qos_info() to get_qos_proportion()   Sebastian Dröge
• gst-plugins-bad: d3dvideosink: fix redefinition of DECLARE_INTERFACE_IID   Tim Müller
• gst-plugins-base: videodecoder: Return the proportion directly   Sebastian Dröge
• [1.0] gst-plugins-base: videodecoder: Return the proportion directly   Sebastian Dröge
• gst-rtsp-server: factory-uri: take ref to factory   Wim Taymans
• gst-rtsp-server: stream: set udp sources to PLAYING   Wim Taymans
• gst-rtsp-server: factory-uri: add some debug   Wim Taymans
• gst-rtsp-server: factory: keep ref to factory while media active   Wim Taymans
• gstreamer: baseparse: forward stream-start event in push mode   Sebastian Dröge
• [1.0] gstreamer: baseparse: forward stream-start event in push mode   Sebastian Dröge
• gst-rtsp-server: test-uri: fix callback signature   Wim Taymans
• gst-rtsp-server: test-uri: remove mapping example   Wim Taymans
• gst-plugins-good: videoflip: Add gray 8/16 support   Sebastian Dröge
• gstreamer: gst: Don't set the locale in gst_init()   Sebastian Dröge
• gst-rtsp-server: test-auth: add example of how to remove sessions   Wim Taymans
• [0.10] gst-plugins-good: videoflip: Add gray 8/16 support   Sebastian Dröge
• gst-plugins-base: discoverer: Use switch/ case instead of lots of ifs for the event handling   Sebastian Dröge
• gst-plugins-base: discoverer: Add support for getting the stream-id   Sebastian Dröge
• gstreamer: value: Use the GLib GType for GThread if compiling against GLib 2.35.3 or newer   Sebastian Dröge
• gstreamer: Remove xfig from spec file   Christian Fredrik Kalager Schaller
• gstreamer: value: Make G-I happy by hiding gst_g_thread_get_type()   Sebastian Dröge
• gstreamer: basesink: reset START_TIME when needed   Wim Taymans
• gstreamer: basesink: add some debug   Wim Taymans
• gstreamer: tests: don't use deprecated thread API   Wim Taymans
• [1.0] gstreamer: basesink: reset START_TIME when needed   Tim Müller
• [1.0] gstreamer: basesink: add some debug   Tim Müller
• gst-plugins-bad: hlssink: add doc chunk with example command line   Tim Müller
• [1.0] gst-plugins-bad: d3dvideosink: fix redefinition of DECLARE_INTERFACE_IID   Tim Müller
• gst-plugins-bad: windows: Put the DirectX LDFLAGS in the correct place and reorder libraries   Sebastian Dröge
• gst-plugins-base: rtsprange: improve docs   Wim Taymans
• gst-plugins-base: streamsynchronizer: Make the element public   Sebastian Dröge
• gstreamer: libs: Add missing single include headers and use them in GIRs   Sebastian Dröge
• gst-plugins-base: libs: Add missing single include headers and use them in GIRs   Sebastian Dröge
• gst-plugins-base: libs: Fix last commit by using correct include paths and only include existing headers   Sebastian Dröge
• [1.0] gstreamer: Release 1.0.3   Tim Müller
• Tag 1.0.3: Release 1.0.3   Tim Müller
• [1.0] gst-plugins-base: Release 1.0.3   Tim Müller
• Tag 1.0.3: Release 1.0.3   Tim Müller
• [1.0] gst-plugins-good: Release 1.0.3   Tim Müller
• Tag 1.0.3: Release 1.0.3   Tim Müller
• [1.0] gst-plugins-ugly: Release 1.0.3   Tim Müller
• Tag 1.0.3: Release 1.0.3   Tim Müller
• [1.0] gst-plugins-bad: Release 1.0.3   Tim Müller
• Tag 1.0.3: Release 1.0.3   Tim Müller
• [1.0] gst-ffmpeg: Revert "Revert "ffmpeg: Fix make dist""   Tim Müller
• [1.0] gst-ffmpeg: Release 1.0.3   Tim Müller
• Tag 1.0.3: Release 1.0.3   Tim Müller
• www: releases: add 1.0.3 releases   Tim Müller
• www: news: announce 1.0.3 releases   Tim Müller
• www: new-release: modify release note template to say bug-fix or feature release   Tim Müller
• www: releases: remove double entry in release list   Tim Müller
• [1.0] gstreamer: Back to development (bug-fixing)   Tim Müller
• [1.0] gst-plugins-base: Back to development (bug-fixing)   Tim Müller
• [1.0] gst-plugins-good: Back to development (bug fixing)   Tim Müller
• [1.0] gst-plugins-ugly: Back to development (bug fixing)   Tim Müller
• [1.0] gst-plugins-bad: Back to development (bug fixing)   Tim Müller
• [1.0] gst-ffmpeg: Back to development (bug fixing)   Tim Müller
• gst-plugins-base: range: add g_return_if check   Wim Taymans
• gst-plugins-base: range: don't overwrite unit field   Wim Taymans
• gst-plugins-base: rtsprange: add method to convert ranges to GstClockTime   Wim Taymans
• gst-plugins-base: rtsprange: add string conversion for new formats   Wim Taymans
• gst-rtsp-server: examples: fix whitespace   Wim Taymans
• gst-rtsp-server: media: support more Range formats   Wim Taymans
• gst-rtsp-server: media: fix seeking   Wim Taymans
• gst-plugins-base: win32: update .def file for new rtsp API   Tim Müller
• gst-plugins-base: app: fix g-i annotation for gst_app_src_push_buffer()   Tim Müller
• [1.0] gst-plugins-base: app: fix g-i annotation for gst_app_src_push_buffer ()   Tim Müller
• [1.0] gst-plugins-base: audio: remove bogus Since marker from docs   Tim Müller
• gst-plugins-base: audio: remove bogus Since marker from docs   Tim Müller
• gst-python: overrides: implement Gst.Structure.__getitem__   Alessandro Decina
• gst-plugins-good: rtspsrc: pause the task instead of spinning   Wim Taymans
• gst-plugins-base: decodebin: Check if the element really accepts the caps after setting it to READY   Sebastian Dröge
• gst-plugins-base: decodebin: Set element to NULL state before removing it from the bin   Sebastian Dröge
• gst-plugins-bad: mpegtsmux: propagate flow returns   Tim Müller
• gst-plugins-base: Add new header files   Christian Fredrik Kalager Schaller
• gst-plugins-bad: Add latest ported plugins etc.   Christian Fredrik Kalager Schaller
• gstreamer: pad: Put all of the probe types in the enum so they work with bindings   Olivier Crête
• gstreamer: pad: don't use parenthesis for ORed pad probe flag enums   Tim Müller
• gstreamer: tests: fix clock unit test build failure after header changes   Tim Müller
• gstreamer: testclock: remove unnecessary include   Tim Müller
• gstreamer: pad: document more pad probe values   Tim Müller
• gst-plugins-base: docs: update audio multi-channel docs   Tim Müller
• gstreamer: event: fix annotation for gst_event_parse_stream_start   Alessandro Decina
• [1.0] gstreamer: pad: Put all of the probe types in the enum so they work with bindings   Tim Müller
• [1.0] gstreamer: pad: don't use parenthesis for ORed pad probe flag enums   Tim Müller
• [1.0] gstreamer: pad: document more pad probe values   Tim Müller
• [1.0] gstreamer: event: fix annotation for gst_event_parse_stream_start   Tim Müller
• gst-plugins-good: mulawdec: fix integer overrun   Tim Müller
• gst-plugins-good: mulaw: const-ify some arrays   Tim Müller
• [1.0] gst-plugins-good: rtspsrc: pause the task instead of spinning   Tim Müller
• [1.0] gst-plugins-good: mulawdec: fix integer overrun   Tim Müller
• [1.0] gst-plugins-good: mulaw: const-ify some arrays   Tim Müller
• gst-plugins-good: videobox: fix border filling for planar YUV formats   Tim Müller
• [1.0] gst-plugins-good: videobox: fix border filling for planar YUV formats   Tim Müller
• gst-plugins-good: qtdemux: avoid criticals if unknown fourcc has space at beginning or end   Tim Müller
• gst-plugins-good: v4l2: remove unused define   Tim Müller
• www: bugs: add more milestones   Tim Müller
• gst-plugins-base: streamsynchronizer: reduce debug log spam a bit   Tim Müller
• gstreamer: dataqueue: reduce debug log spam a bit   Tim Müller
• gst-editing-services: docs: Add make upload   Thibault Saunier
• common: gst-glib2.m4: we don't need to link against gthread-2.0 any more   Tim Müller
• gstreamer: docs: remove all mention of g_thread_init()   Tim Müller
• gstreamer: gstreamer-1.0.pc: move gmodule-no-export-2. 0 dependency to Requires.private   Tim Müller
• gstreamer: gst: don't require gthread-2.0   Tim Müller
• gstreamer: docs: clean up sections file for pad probe defines that moved into enum   Tim Müller
• gst-rtsp-server: configure: bump required version of -base   Wim Taymans
• gst-rtsp-server: MediaMapping -> MountPoints   Wim Taymans
• gst-rtsp-server: client: add generic close-after-send support   Wim Taymans
• gst-rtsp-server: client: fix compilation   Wim Taymans
• gst-rtsp-server: client: log more errors   Wim Taymans
• gst-plugins-good: law: fix accidental file permissions change   Tim Müller
• [0.10] gst-plugins-good: mulawdec: fix integer overrun   Tim Müller
• gst-rtsp-server: client: add locking   Wim Taymans
• gst-rtsp-server: client: remove reference to server   Wim Taymans
• gst-rtsp-server: client: small cleanup   Wim Taymans
• gst-rtsp-server: client: make the client usable without a socket   Wim Taymans
• gst-rtsp-server: client: call destroy without the lock   Wim Taymans
• gst-rtsp-server: client: fix compilation   Wim Taymans
• gst-rtsp-server: client: remove unused include   Wim Taymans
• gst-rtsp-server: client: small cleanup   Wim Taymans
• gst-plugins-base: streamsynchronizer: don' t send gap events with huge bogus durations when advancing EOS streams   Tim Müller
• gstreamer: add gst-editing-services to PKG_CONFIG_PATH   Luis de Bethencourt
• gst-plugins-base: videoscale: Fix unit test to ignore unsupported color formats   Sebastian Dröge
• gst-plugins-base: rtsp: add method to parse options list   Wim Taymans
• gst-plugins-good: rtspsrc: use new option parser function   Wim Taymans
• gst-rtsp-server: tests: add unit test for the client object   Wim Taymans
• gst-rtsp-server: client: only free connection when there is one   Wim Taymans
• [refs/remotes/origin/master] common: gst-glib2.m4: we don' t need to link against gthread-2.0 any more   Wim Taymans
• common: supp: add more suppressions   Wim Taymans
• gst-rtsp-server: client: use 454 when session can't be found   Wim Taymans
• gst-rtsp-server: test: add test for session in options request   Wim Taymans
• gst-rtsp-server: update common   Wim Taymans
• gst-rtsp-server: client: improve debug and fix leaks   Wim Taymans
• gst-rtsp-server: client: improve debug   Wim Taymans
• gst-rtsp-server: tests: add more tests   Wim Taymans
• [0.10] gstreamer: add gst-editing-services to PKG_CONFIG_PATH   Luis de Bethencourt
• gst-plugins-ugly: cdiocddasrc: detect whether drive produces samples in non-host endianness   Tim Müller
• gst-plugins-bad: mpegtsmux: crashes when trying to re-use the element   Tim Müller
• gst-rtsp-server: mounts: add g_return_if guards   Wim Taymans
• gst-rtsp-server: client: fix factory leak   Wim Taymans
• gst-rtsp-server: tests: add test for mountpoints   Wim Taymans
• gst-rtsp-server: server: remove unused include   Wim Taymans
• gst-rtsp-server: examples: fix include   Wim Taymans
• gst-rtsp-server: media-factory: require an url   Wim Taymans
• gst-rtsp-server: rtsp: use gst_object_unref on GstObjects   Wim Taymans
• gst-rtsp-server: media: unref pipeline in finalize to avoid leaking it   Wim Taymans
• gst-rtsp-server: stream: improve debug   Wim Taymans
• gst-rtsp-server: tests: add mediafactory test   Wim Taymans
• gst-plugins-bad: shm: Fix compilation of shm on OSX   Sebastian Dröge
• gst-plugins-bad: shm: Fix configure check and clean up Makefile   Sebastian Dröge
• [1.0] gst-plugins-bad: shm: Fix compilation of shm on OSX   Sebastian Dröge
• [1.0] gst-plugins-bad: shm: Fix configure check and clean up Makefile   Sebastian Dröge
• gst-rtsp-server: media: check if prepared for some methods   Wim Taymans
• gst-rtsp-server: tests: add media tests   Wim Taymans
• [1.0] gstreamer: docs: remove all mention of g_thread_init()   Tim Müller
• [1.0] gstreamer: gst-uninstalled: add gst-editing-services to PKG_CONFIG_PATH   Tim Müller
• [0.10] gnonlin: gnlcomposition: Make sure to pass flushes downstream if flushing seeks were received   Sebastian Dröge
• [0.10] gnonlin: gnlcomposition: Add support for reverse playback   Sebastian Dröge
• gnonlin: gnlcomposition: Make sure to pass flushes downstream if flushing seeks were received   Sebastian Dröge
• gnonlin: gnlcomposition: Add support for reverse playback   Sebastian Dröge
• gstreamer: configure.ac: Update libtool versioning   Edward Hervey
• gst-plugins-base: configure.ac: Update libtool versioning   Edward Hervey
• gstreamer: docs: add link to python porting doc and app dev manual to porting-to-1.0.txt   Tim Müller
• [1.0] gst-plugins-base: streamsynchronizer: reduce debug log spam a bit   Tim Müller
• [1.0] gst-plugins-base: streamsynchronizer: don' t send gap events with huge bogus durations when advancing EOS streams   Tim Müller
• [1.0] gst-plugins-bad: mpegtsmux: propagate flow returns   Tim Müller
• [1.0] gst-plugins-bad: mpegtsmux: crashes when trying to re-use the element   Tim Müller
• gst-rtsp-server: rtsp: make object details private   Wim Taymans
• gst-rtsp-server: client: TEARDOWN brings that state to Init again   Wim Taymans
• gst-plugins-base: test: add test for playbin in combination with appsink   Tim Müller
• gst-rtsp-server: Revert "client: TEARDOWN brings that state to Init again"   Wim Taymans
• gst-rtsp-server: media: require an element in media constructor   Wim Taymans
• gst-rtsp-server: media: avoid element leak   Wim Taymans
• gst-rtsp-server: small cleanup   Wim Taymans
• gst-rtsp-server: Add missing g_type_class_add_private in GstRTSPMediaFactoryURI   Alessandro Decina
• gst-rtsp-server: Fix compiler warning   Alessandro Decina
• gst-rtsp-server: client: never call gst_rtsp_media_unprepare, let gst_rtsp_media_finalize do it   Alessandro Decina
• gstreamer: bin: remove some casts   Wim Taymans
• gstreamer: message: add reset-time type string   Wim Taymans
• [0.10] gst-plugins-bad: wasapi: Integrate into the autotools build system   Sebastian Dröge
• gst-plugins-bad: wasapi: Integrate into the autotools build system   Sebastian Dröge
• gst-plugins-bad: configure: Add wasapi to the unported plugins   Sebastian Dröge
• [0.10] gst-plugins-bad: wasapi: make the build happy without the static plugins patch   Sebastian Dröge
• gst-plugins-bad: wasapi: make the build happy without the static plugins patch   Sebastian Dröge
• gst-plugins-base: win32: update .def for new API   Tim Müller
• gst-rtsp-server: media: use g_object_ref/unref for GObjects   Wim Taymans
• gst-rtsp-server: media: check state   Wim Taymans
• gst-rtsp-server: media: improve debug   Wim Taymans
• gst-rtsp-server: media: let the source unref the last media ref   Wim Taymans
• gst-rtsp-server: Revert "client: never call gst_rtsp_media_unprepare, let gst_rtsp_media_finalize do it"   Wim Taymans
• gst-rtsp-server: media: media has to be unprepared in finalize   Wim Taymans
• gst-rtsp-server: media: match prepare with unprepare   Wim Taymans
• [0.10] gst-plugins-bad: mxf: Add more support for RP2008 ( AVC in the MPEG mapping)   Sebastian Dröge
• gst-plugins-bad: mxf: Add more support for RP2008 (AVC in the MPEG mapping)   Sebastian Dröge
• gst-plugins-good: webmux: fix linking with shout2send element   Tim Müller
• gst-plugins-good: shout2send: accept audio/webm as well as video/webm   Tim Müller
• [1.0] gst-plugins-good: webmux: fix linking with shout2send element   Tim Müller
• [1.0] gst-plugins-good: shout2send: accept audio/webm as well as video/webm   Tim Müller
• [0.10] gst-plugins-bad: waylandsink: Port to 1.0 protocol   Tim Müller
• [0.10] gst-plugins-bad: configure: fix wayland check   Tim Müller
• gst-plugins-bad: configure: sprinkle some AC_LANG_PROGRAM in wasapi check   Tim Müller
• gst-plugins-bad: resindvd, frei0r: link to libgmodule-2. 0 in plugins that use g_module_* API   Tim Müller
• gst-plugins-ugly: dvdread: link to libgmodule-2. 0 since the g_module_* API is used   Tim Müller
• gst-plugins-good: v4l2src: link against -lrt for clock_gettime()   Tim Müller

Last message date: Fri Nov 30 15:19:00 PST 2012
Archived on: Fri Nov 30 15:19:01 PST 2012

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