=== release 0.10.24 === 2009-08-05 Jan Schmidt * configure.ac: releasing 0.10.24, "Counting the days" 2009-08-05 00:38:40 +0100 Jan Schmidt * po/af.po: * po/az.po: * po/bg.po: * po/ca.po: * po/cs.po: * po/da.po: * po/de.po: * po/en_GB.po: * po/es.po: * po/fi.po: * po/fr.po: * po/hu.po: * po/id.po: * po/it.po: * po/ja.po: * po/lt.po: * po/lv.po: * po/nb.po: * po/nl.po: * po/or.po: * po/pl.po: * po/pt_BR.po: * po/ru.po: * po/sk.po: * po/sq.po: * po/sr.po: * po/sv.po: * po/tr.po: * po/uk.po: * po/vi.po: * po/zh_CN.po: Update .po files 2009-08-01 17:26:23 +0100 Tim-Philipp Müller * gst/typefind/gsttypefindfunctions.c: * tests/check/gst/typefindfunctions.c: typefinding: fix detection of fLaC id packet in broken flac-in-ogg There are flac-in-ogg files without the usual flac packet framing and these files just have a 4-byte fLaC ID packet as first packet. We need to recognise the type just from these four bytes if we want oggdemux to recognise these streams correctly. 2009-07-30 14:40:50 +0100 Jan Schmidt * ChangeLog: * configure.ac: * po/LINGUAS: * po/af.po: * po/az.po: * po/bg.po: * po/ca.po: * po/cs.po: * po/da.po: * po/de.po: * po/en_GB.po: * po/es.po: * po/fi.po: * po/fr.po: * po/hu.po: * po/id.po: * po/it.po: * po/ja.po: * po/lt.po: * po/lv.po: * po/nb.po: * po/nl.po: * po/or.po: * po/pl.po: * po/pt_BR.po: * po/ru.po: * po/sk.po: * po/sq.po: * po/sr.po: * po/sv.po: * po/tr.po: * po/uk.po: * po/vi.po: * po/zh_CN.po: 0.10.24.5 pre-release 2009-07-29 14:15:53 -0400 Olivier Crête * gst-libs/gst/audio/gstaudiofilter.c: audiofilter: Don't assert on slightly different caps Plugins should not assert on incompatible caps, caps negotiation will fail anyway. 2009-07-30 13:42:21 +0300 Stefan Kost * gst/adder/gstadder.c: adder: reset pending flush-stop flag in state_changed. (mostly) Fixes #590146. 2009-07-30 09:28:20 +0100 Tim-Philipp Müller * configure.ac: configure: bump Gtk+ requirement of GUI examples from 2.12 to 2.14 The gio mount example needs GtkMountOperation, which is new in 2.14. 2009-07-27 10:29:27 +0100 Balachandran C * ext/alsa/gstalsasrc.c: alsasrc: set alsasrc->handle back to NULL when closing device Fixes crashes in gst_alsa_find_device_name() when probing or reading the device-name property (e.g. when doing a dot-file dump). Fixes #589797. 2009-07-24 19:26:40 +0100 Tim-Philipp Müller * gst/playback/gststreamselector.c: playbin: rename GType of stream selector pad to avoid clash with input-selector from -bad Rename the GType of the pads of playbin's internal stream selector element so they don't use the same type name as input-selector's pads. Fixes #589622. 2009-07-24 13:39:55 +0100 Jan Schmidt * ChangeLog: * configure.ac: * po/af.po: * po/az.po: * po/bg.po: * po/ca.po: * po/cs.po: * po/da.po: * po/de.po: * po/en_GB.po: * po/es.po: * po/fi.po: * po/fr.po: * po/hu.po: * po/id.po: * po/it.po: * po/ja.po: * po/lt.po: * po/nb.po: * po/nl.po: * po/pl.po: * po/pt_BR.po: * po/ru.po: * po/sk.po: * po/sq.po: * po/sr.po: * po/sv.po: * po/tr.po: * po/uk.po: * po/vi.po: * po/zh_CN.po: 0.10.23.4 pre-release 2009-07-24 13:46:15 +0100 Jan Schmidt * tests/examples/v4l/.gitignore: ignores: Ignore v4l probing example binary 2009-07-24 09:35:38 +0100 Tim-Philipp Müller * gst/typefind/gsttypefindfunctions.c: typefind: recognise Kate spu subtitles as well Recognise spu-subtitles, SUB and K-SPU as valid categories for Kate subtitles as well. 2009-07-24 00:42:16 +0300 Stefan Kost * common: Automatic update of common submodule From fedaaee to 94f95e3 2009-07-22 14:21:43 +0100 Christian Schaller * gst-plugins-base.spec.in: Update spec file with latest changes 2009-07-20 17:28:20 +0100 Jan Schmidt * configure.ac: * po/af.po: * po/az.po: * po/bg.po: * po/ca.po: * po/cs.po: * po/da.po: * po/de.po: * po/en_GB.po: * po/es.po: * po/fi.po: * po/fr.po: * po/hu.po: * po/id.po: * po/it.po: * po/ja.po: * po/lt.po: * po/nb.po: * po/nl.po: * po/or.po: * po/pl.po: * po/pt_BR.po: * po/ru.po: * po/sk.po: * po/sq.po: * po/sr.po: * po/sv.po: * po/tr.po: * po/uk.po: * po/vi.po: * po/zh_CN.po: * win32/common/_stdint.h: * win32/common/audio-enumtypes.c: * win32/common/config.h: * win32/common/gstrtsp-enumtypes.c: * win32/common/interfaces-enumtypes.c: * win32/common/video-enumtypes.c: 0.10.23.3 pre-release 2009-07-20 12:51:30 +0200 Wim Taymans * gst/audiotestsrc/gstaudiotestsrc.c: audiotestsrc: call send_event directly We can't call gst_element_send_event() from a streaming thread as it gets the state lock. Instead call the send_event method directly until we have a nice API for this in basesrc. Fixes #588746 2009-07-03 04:42:24 -0400 Olivier Crête * gst-libs/gst/audio/gstaudiosink.c: audiosink: Add stream-status messages Fixes #587695 2009-07-03 04:41:05 -0400 Olivier Crête * gst-libs/gst/audio/gstaudiosrc.c: audiosrc: Add stream-status messages See #587695 2009-07-20 10:53:11 +0200 Edward Hervey * gst/adder/gstadder.c: gstadder: Don't forget to free pending events on flush/dispose. Fixes #588747 2009-07-12 10:08:12 +0200 Edward Hervey * tests/check/elements/adder.c: tests/adder: Add stream consistency checking. Fixes #588748 2009-07-12 10:07:34 +0200 Edward Hervey * gst/audiotestsrc/gstaudiotestsrc.c: audiotestsrc: Make sure tags are properly serialized. Fixes #588746 We do this by letting the basesrc base class handle the tags. 2009-07-13 09:28:54 +0200 Edward Hervey * gst/adder/gstadder.c: * gst/adder/gstadder.h: adder: Collect incoming tag events and send them after newsegment. Fixes #588747 2009-07-16 09:32:46 +0200 Edward Hervey * ext/vorbis/vorbisdec.c: vorbisdec: Check for empty tag strings. Fixes #588724 2009-07-14 17:03:35 +0200 Wim Taymans * gst/playback/gstqueue2.c: queue2: fix leak and improve buffering Keep track of the max requested position and compare this to the write position in the temp file to get the current amount of buffered data. Fix memleak of all incomming buffers. Fixes #588551 2009-07-15 17:40:14 +0100 Tim-Philipp Müller * gst/playback/Makefile.am: * gst/playback/gstinputselector.c: * gst/playback/gstinputselector.h: * gst/playback/gstplay-marshal.list: * gst/playback/gstplaybin2.c: playbin2: use private copy of input-selector We shouldn't really depend on elements from -bad for stream selection in playbin2, so use a private copy of input-selector until the selector plugin is ready to be moved to -base or -good. Fixes #586356. 2009-07-15 17:26:32 +0100 Tim-Philipp Müller * gst/playback/gstinputselector.c: * gst/playback/gstinputselector.h: playback: add private copy of the input-selector from gst-plugins-bad Not hooked up yet though. See #586356. 2009-07-14 19:00:36 +0200 Filippo Argiolas * tests/examples/v4l/Makefile.am: examples: fix v4l probe example build Fixes bug #588550. 2009-07-14 19:00:10 +0100 Jan Schmidt * ChangeLog: * configure.ac: * po/af.po: * po/az.po: * po/bg.po: * po/ca.po: * po/cs.po: * po/da.po: * po/de.po: * po/en_GB.po: * po/es.po: * po/fi.po: * po/fr.po: * po/hu.po: * po/id.po: * po/it.po: * po/ja.po: * po/lt.po: * po/nb.po: * po/nl.po: * po/or.po: * po/pl.po: * po/pt_BR.po: * po/ru.po: * po/sk.po: * po/sq.po: * po/sr.po: * po/sv.po: * po/tr.po: * po/uk.po: * po/vi.po: * po/zh_CN.po: 0.10.23.2 pre-release 2009-07-14 16:24:10 +0100 Jan Schmidt * po/LINGUAS: * po/tr.po: Add Turkish translations 2009-07-14 15:31:13 +0100 Jan Schmidt * tests/check/elements/adder.c: adder: One more attempt to fix the adder test Give up and discard and recreate the alsasrc after checking it can be opened, due to some strange crash inside alsa when we don't. 2009-07-14 15:06:41 +0100 Jan Schmidt * tests/check/elements/adder.c: adder: Perform get_state() in the unit test Wait for the alsasrc to return to NULL after setting it to PAUSED for testing, otherwise it leads to segfaults later on. 2009-07-14 14:39:32 +0100 Jan Schmidt * tests/check/elements/adder.c: adder: Don't fail when alsasrc is unavailable Make the liveadder test succeed silently when it can't be completed either because alsasrc is unavailable, or because the device is inaccessible. 2009-07-13 22:51:48 +0100 Tim-Philipp Müller * gst-libs/gst/pbutils/descriptions.c: * gst/typefind/gsttypefindfunctions.c: typefinding: use subtitle/x-kate for Kate subtitle streams and application/x-kate for the rest Differentiate subtitle streams and lyrics/cracktastic/complex streams via the category string in the headers. This seems like a useful distinction to make, and also seems more future-proof. See #525743. 2009-02-21 13:18:10 +0000 Vincent Penquerc'h * ext/ogg/gstoggmux.c: oggmux: add Kate caps to the list of accepted types See #525743. 2009-07-13 21:56:46 +0300 Stefan Kost * gst/playback/gsturidecodebin.c: uridecodebin: treat uri-schemas incasesensitive Treat uri-schemas incasesensitive. This is mandated in rfc2396 section 3.1. Fixes not showing buffering messages e.g. for HTTP://... 2009-07-13 21:54:47 +0300 Stefan Kost * gst-libs/gst/interfaces/navigation.c: navigation: simplify docs Make short-desc short - its used in the toc. Strip uneeded markup. 2009-07-13 18:31:15 +0100 Jan Schmidt * win32/common/libgstnetbuffer.def: * win32/common/libgstvideo.def: win32: Fix exports Remove methods from video base classes that have moved to -bad. Add gst_netaddress_to_string 2009-07-13 17:56:58 +0100 Jan Schmidt * tests/examples/gio/.gitignore: ignores: ignore the giosrc-mounting example binary 2009-07-13 17:54:40 +0100 Jan Schmidt * gst-libs/gst/interfaces/navigation.c: navigation: Add some partial documentation Add a general documentation blurb for the GstNavigation functionality. Still lacks some example code and detail on how to implement it. 2009-07-13 17:52:39 +0100 Tim-Philipp Müller * gst-libs/gst/pbutils/descriptions.c: pbutils: add description for Siren codec and make two descriptions non-translatable 2009-07-13 12:23:20 -0400 Olivier Crête * common: Automatic update of common submodule From 5845b63 to fedaaee 2009-07-13 18:21:49 +0200 Elliott Sales de Andrade * gst-libs/gst/riff/riff-ids.h: * gst-libs/gst/riff/riff-media.c: riff: add siren to the RIFF parser Add siren7 caps to the RIFF parser. 2009-07-13 14:55:59 +0200 Filippo Argiolas * configure.ac: * tests/examples/Makefile.am: * tests/examples/v4l/Makefile.am: * tests/examples/v4l/probe.c: v4lsrc: add a simple test case for device probing 2009-07-03 11:38:01 +0200 Filippo Argiolas * configure.ac: * sys/v4l/Makefile.am: * sys/v4l/gstv4lelement.c: v4lsrc: optional support for device probing with gudev Enumerate v4l devices using gudev if available. Fixes bug #583640. 2009-07-10 23:24:36 +0100 Stefan Kost * gst/adder/gstadder.c: adder: add since tags to docs 2009-07-10 21:29:51 +0100 Wim Taymans * tests/examples/seek/seek.c: seek: don't automatically start pipeline in DB Keep the pipeline paused when we detect download buffering. The user has to manually start the pipeline for now because we can't estimate when the buffering will finish or when we have underrun. 2009-07-10 21:01:39 +0100 Wim Taymans * gst/playback/gstqueue2.c: queue2: flush differently, avoiding deadlocks Don't flush the file by closing and opening it but instead use g_freopen. This avoids a deadlock in shutdown because we emit the temp-location property change with the wrong lock held. 2009-07-10 20:25:43 +0100 Wim Taymans * tests/examples/seek/seek.c: seek: add a checkbox for progressive download 2009-07-10 20:24:14 +0100 Wim Taymans * gst/playback/gsturidecodebin.c: uridecodebin: Fix template construction Fix the construction of the temporary filename construction as the application name can be NULL and we don't want a separator between the prgname and the template. 2009-07-10 20:04:33 +0100 Wim Taymans * gst/playback/gstplay-enum.c: * gst/playback/gstplay-enum.h: * gst/playback/gstplaybin2.c: playbin2: add support for progressive download Add a new playbin2 flag (initially disabled) to enable progressive download buffering in uridecodebin. 2009-07-10 19:59:30 +0100 Wim Taymans * gst/playback/gsturidecodebin.c: uridecodebin: add download property Add a download property that will attempt to configure queue2 into progressive download buffering. Make sure we only enable download buffering for quicktime and flv formats. 2009-07-10 19:49:46 +0100 Wim Taymans * gst/playback/gstqueue2.c: queue2: add temp-template property Add a new temp-template property so that queue2 can securely allocate a temporary filename. Deprecate the temp-location property for setting the location but still use it to notify the allocated temp file. 2009-07-10 20:06:28 +0100 Stefan Kost * gst/adder/gstadder.c: * gst/adder/gstadder.h: adder: add a caps-property to avoid to need to plug a capsfilter afterwards Adder can only handle one common format accross the pads. Thus one needed to add a capsfilter afterwards and manage the caps. Now one can simply set the caps on the property. 2009-07-10 18:59:05 +0100 Stefan Kost * tests/check/elements/adder.c: adder: skip live-seek text if we have no audiosrc, add new test The seek-test needs a real audiosrc. Also add a test that checks that adder is reusable. Finaly handle warnings as warnings to fix a assertion. 2009-07-10 19:16:35 +0200 Sebastian Dröge * ext/gio/gstgiosink.c: gio: Also post a "not-mounted" message from giosink 2009-07-10 17:15:48 +0200 Sebastian Dröge * tests/examples/gio/giosrc-mounting.c: gio: Remove workaround for playbin2 bug in the sample application The playbin2 bug was #588078. 2009-07-10 17:08:40 +0200 Sebastian Dröge * gst/playback/gstplaybin2.c: playbin2: Make it possible for READY->PAUSED to succeed after it failed the first time If READY->PAUSED failed in the source element we would've swapped the current and next group already. To allow READY->PAUSED to succeed after the first failure we have to swap the current and next group back again. This also ensure that we're again in the same state as before the failed state change and not at the next group. This was especially a problem for playbin2 pipelines that use the new mounting support in giosrc as the source would fail for READY->PAUSED the first time, the application mounts the location and then tries to go READY->PAUSED again (and this time it would succeed). Fixes bug #588078. 2009-07-10 11:42:51 +0200 Sebastian Dröge * configure.ac: * tests/examples/Makefile.am: * tests/examples/gio/Makefile.am: * tests/examples/gio/giosrc-mounting.c: gio: Add example application that shows how to handle the "not-mounted" message 2009-07-10 11:24:57 +0200 Sebastian Dröge * configure.ac: gio: Remove the experimental status from the GIO plugin Fixes bug #510417. 2009-07-10 11:24:05 +0200 Sebastian Dröge * ext/gio/gstgiosink.c: * ext/gio/gstgiosrc.c: gio: Add documentation for the new "not-mounted" and "file-exists" messages 2009-07-09 13:45:13 +0200 Sebastian Dröge * ext/gio/gstgiobasesrc.c: gio: Make sure that we have the correct stream position when starting 2009-07-08 17:24:19 +0200 Sebastian Dröge * ext/gio/gstgiobasesink.c: gio: Make sure to flush the output stream if it shouldn't be closed Otherwise there might still be unwritten data after the element has stopped. 2009-07-08 17:19:29 +0200 Sebastian Dröge * ext/gio/gstgiobasesink.c: * ext/gio/gstgiobasesink.h: * ext/gio/gstgiobasesrc.c: * ext/gio/gstgiobasesrc.h: * ext/gio/gstgiosink.c: * ext/gio/gstgiosrc.c: gio: Don't close the GIO streams for the giostream{src,sink} elements This makes it possible to do something useful with the streams after the element has stopped. Fixes bug #587896. 2009-07-08 17:19:05 +0200 Sebastian Dröge * tests/check/pipelines/gio.c: gio: Try to reuse the pipeline with the same stream objects 2009-07-08 17:02:54 +0200 Sebastian Dröge * ext/gio/gstgiobasesink.c: * ext/gio/gstgiobasesrc.c: gio: Improve the error message if a stream is already closed before usage 2009-07-08 16:55:41 +0200 Sebastian Dröge * ext/gio/gstgiosink.c: gio: Post a custom file-exists message on the bus if the file already exists An application can handle this message, remove the file in question and restart the pipeline again without showing an error. This fixes bug #529300. 2009-07-08 16:54:56 +0200 Sebastian Dröge * ext/gio/gstgiosrc.c: gio: Use OPEN_READ instead of NOT_FOUND if a location is not mounted 2009-07-08 16:50:56 +0200 Sebastian Dröge * ext/gio/gstgiosink.c: gio: Use OPEN_WRITE instead of OPEN_READ as error category in giosink 2009-07-08 15:52:35 +0200 Sebastian Dröge * ext/gio/gstgiosrc.c: gio: Post a custom "not-mounted" message on the bus This allows applications to mount the GFile if possible and restart the pipeline instead of simply giving an error. 2009-07-08 15:08:32 +0200 Philip Jägenstedt * gst/audioconvert/gstchannelmix.c: audioconvert: Fix compilation when debugging is disabled Fixes bug #587980. 2009-07-07 20:23:23 +0200 Sebastian Dröge * ext/gio/gstgiobasesink.c: * ext/gio/gstgiobasesink.h: * ext/gio/gstgiobasesrc.h: * ext/gio/gstgiosink.c: * ext/gio/gstgiosink.h: * ext/gio/gstgiostreamsink.c: * ext/gio/gstgiostreamsink.h: gio: Add vfunc for requesting the stream for the sinks too 2009-07-07 20:21:36 +0200 Sebastian Dröge * ext/gio/gstgiobasesink.c: * ext/gio/gstgiobasesink.h: * ext/gio/gstgiobasesrc.c: * ext/gio/gstgiosink.c: * ext/gio/gstgiosrc.c: * ext/gio/gstgiostreamsink.c: * ext/gio/gstgiostreamsrc.c: gio: Some more random cleanup 2009-07-07 20:20:58 +0200 Sebastian Dröge * ext/gio/gstgio.c: * ext/gio/gstgiobasesink.c: * ext/gio/gstgiobasesrc.c: * ext/gio/gstgiobasesrc.h: * ext/gio/gstgiosink.c: * ext/gio/gstgiosrc.c: * ext/gio/gstgiosrc.h: * ext/gio/gstgiostreamsink.c: * ext/gio/gstgiostreamsrc.c: * ext/gio/gstgiostreamsrc.h: gio: Update my mail address and copyright 2009-07-07 20:18:00 +0200 Sebastian Dröge * ext/gio/gstgiobasesrc.c: * ext/gio/gstgiobasesrc.h: * ext/gio/gstgiosrc.c: * ext/gio/gstgiostreamsrc.c: * ext/gio/gstgiostreamsrc.h: gio: General clean up and simplification The GInputStreams are now requested by a vfunc from the subclasses instead of relying that the subclass sets it until it's needed. This might also fix bug #587896. 2009-07-06 22:31:12 +0100 Stefan Kost * gst/adder/gstadder.c: adder: keep sending newsegments after seeking Adder sends with timestamps from 0 upwards. After seeking we need to send new-segments to get correct positions-queries. 2009-07-06 20:44:00 +0100 Stefan Kost * tests/check/elements/adder.c: adder: make test more robust Add audioconverts to the live-seeking test to make it negotiate. 2009-06-30 17:19:50 +0300 Stefan Kost * sys/xvimage/xvimagesink.c: xvimagesink: use core performance log category 2009-07-05 21:29:40 +0200 Edward Hervey * gst/adder/gstadder.c: adder: Call set_flushing(TRUE) for flushing seeks *when* the streaming is stopped. This ensures that collectpads' cookie is properly updated so that when the streaming threads will restart and be checking for the flushing status of all pads there will be no inconsistent state. 2009-07-05 18:01:38 +0200 Hans-Peter Nilsson * ext/pango/gstclockoverlay.c: pango: Call tzset() before localtime_r() POSIX and your local friendly ctime(3) manual entry says that localtime_r isn't required to set the state variables that define the current timezone. Indeed, glibc (at least 2.9) doesn't do this for subsequent calls. The effect is that if the system timezone is changed for a running program between two calls to gst_clock_overlay_render_time, it won't be noticed. For glibc, changing the timezone equals /etc/localtime being modified. Fixes bug #587676. 2009-07-01 17:33:14 -0700 David Schleef * ext/Makefile.am: build: remove spurious schroedinger reference 2009-07-01 10:25:43 -0700 David Schleef * configure.ac: * ext/Makefile.am: * ext/schroedinger/Makefile.am: * ext/schroedinger/gstschro.c: * ext/schroedinger/gstschrodec.c: * ext/schroedinger/gstschroenc.c: * ext/schroedinger/gstschroparse.c: * ext/schroedinger/gstschroutils.c: * ext/schroedinger/gstschroutils.h: * gst-libs/gst/video/Makefile.am: * gst-libs/gst/video/gstbasevideocodec.c: * gst-libs/gst/video/gstbasevideocodec.h: * gst-libs/gst/video/gstbasevideodecoder.c: * gst-libs/gst/video/gstbasevideodecoder.h: * gst-libs/gst/video/gstbasevideoencoder.c: * gst-libs/gst/video/gstbasevideoencoder.h: * gst-libs/gst/video/gstbasevideoparse.c: * gst-libs/gst/video/gstbasevideoparse.h: * gst-libs/gst/video/gstbasevideoutils.c: * gst-libs/gst/video/gstbasevideoutils.h: basevideo: send basevideo back to remedial school Move basevideo classes and schroedinger plugin to -bad. 2009-07-01 12:54:21 +0200 Wim Taymans * docs/libs/gst-plugins-base-libs-sections.txt: * gst-libs/gst/netbuffer/gstnetbuffer.h: netaddress: add constant for max len 2009-07-01 12:48:38 +0200 Wim Taymans * docs/libs/gst-plugins-base-libs-sections.txt: * gst-libs/gst/netbuffer/gstnetbuffer.c: * gst-libs/gst/netbuffer/gstnetbuffer.h: netbuffer: add gst_netaddress_to_string Add function to serialize a net address to a string. API: GstNetAddress::gst_netaddress_to_string() 2009-06-30 18:44:44 +0200 Wim Taymans * gst/playback/gsturidecodebin.c: uridecodebin: make fd:// uri use buffering too fd:// usually operate in push mode only and are thus suitable for buffering. 2009-06-30 14:46:38 +0300 Stefan Kost * gst/playback/gstplaybin2.c: * gst/volume/gstvolume.c: volume: include "1.0=100%" in property description 2009-06-30 14:45:51 +0300 Stefan Kost * gst/playback/gstplaysink.c: playsink: remove unused property defs 2009-06-29 17:11:50 +0300 Stefan Kost * gst-libs/gst/audio/multichannel.c: multichannel: rewrite the new doc comment a bit Its part of the audio lib. 2009-06-29 14:34:02 +0100 Jan Schmidt * gst/playback/gstplaysink.c: playsink: Avoid a segfault when the video sink fails to start Don't attempt to display the subpictures and segfault when the video sink failed to start (and hence the videochain is NULL). 2009-06-29 15:14:07 +0200 Wim Taymans * gst-libs/gst/audio/gstringbuffer.c: * gst-libs/gst/audio/gstringbuffer.h: ringbuffer: add vmethod to clear the ringbuffer Add a vmethod so that subclasses can be notified when they should clear the data in the ringbuffer. 2009-06-29 14:00:14 +0100 Jan Schmidt * gst-libs/gst/riff/riff-media.c: riff-media: Fix the fourcc caps property for VC-1/WMVA The caps property for carrying fourccs is 'format', not 'fourcc' 2009-06-29 12:20:52 +0200 Wim Taymans * gst-libs/gst/rtsp/gstrtspconnection.c: rtsp: include in.h for FreeBSD compat Fixes #586920 2009-06-29 12:20:20 +0200 Wim Taymans * win32/common/libgstapp.def: defs: add defs for new appsink buffer-list method 2009-06-29 12:14:43 +0200 Wim Taymans * gst-libs/gst/app/gstappsink.c: * gst-libs/gst/app/gstappsink.h: appsink: add docs and signals Add docs for the new callback. Add signals for the new buffer-list support. 2009-06-29 10:24:36 +0200 Branko Subasic * tests/check/elements/appsink.c: Added unit tests for buffer list support in appsink. 2009-06-17 11:12:08 +0200 Branko Subasic * gst-libs/gst/app/gstappsink.c: Added buffer list support. 2009-06-17 09:23:11 +0200 Branko Subasic * gst-libs/gst/app/gstappsink.h: Added buffer list support. 2009-06-29 09:36:27 +0200 Peter Kjellerstedt * gst-libs/gst/sdp/gstsdpmessage.c: sdp: Include winsock2.h after defining WINVER. Similar to bug #587080. 2009-06-29 09:31:40 +0200 Peter Kjellerstedt * gst-libs/gst/rtsp/gstrtspconnection.c: rtsp: Moved a comment. 2009-06-27 23:23:02 +0300 Stefan Kost * gst-libs/gst/audio/audio.c: * gst-libs/gst/audio/multichannel.c: docs: add basic section docs for multichannel and relocate the ones for audio Add section docs for multichannel, so that it has a short desc in the toc too. Move the section docs in adio up, so that the follow the copyright like elsewhere. 2009-06-26 21:11:45 +0300 Stefan Kost * sys/v4l/gstv4lelement.c: * sys/v4l/gstv4lsrc.c: v4l: open/close device in ready. Simillar change like in v4l2src. This allows probing feature in paused, where streaming is noit yet started. 2009-06-10 17:05:22 +0300 René Stadler * gst/playback/gstplaysink.c: playbin2: fix initial volume handling also when reusing the element This is a follow-up to commit 452988, making it work correctly when the audio chain is reused. 2009-06-26 21:48:58 +0400 Руслан Ижбулатов * gst-libs/gst/rtsp/gstrtspconnection.c: Define WINVER before including any win headers Fixes bug #587080. 2009-06-27 00:50:54 +0300 René Stadler * gst-libs/gst/riff/riff-read.c: riff: prevent crash if rounded up tag size exceeds data size When rounding up `tsize' exceeds the remaining buffer size, `size' underflows and an invalid read past the buffer data follows. 2009-06-26 15:17:21 +0200 Sebastian Dröge * gst-libs/gst/video/gstbasevideocodec.c: basevideocodec: By default don't allow caps changes on the srcpad This fixed playback of Dirac files with schrodec when upstream wants a different width/height, basevideocodec accepts this and then pushes buffers with new caps but content of the old caps. In the best case this will just result in wrong unit size and a failure in basestransform elements. 2009-06-26 14:11:21 +0100 Jan Schmidt * autogen.sh: autogen.sh: Use printf instead of 'echo -n'. Check for automake-1.1[01] Check for more automake command variants. Use printf instead of 'echo -n' for portability 2009-06-26 13:41:38 +0100 Jan Schmidt * common: Automatic update of common submodule From f810030 to 5845b63 2009-06-26 13:14:02 +0300 Stefan Kost * gst/playback/gstscreenshot.c: screenshot: don't leak message 2009-06-25 12:04:59 +0100 Tim-Philipp Müller * gst/typefind/gsttypefindfunctions.c: typefinding: lower the h264 typefinder's probability A NEARLY_CERTAIN is absolutely not warranted given the kind of things it checks for. Even a LIKELY is probably not entirely appropriate. 2009-06-24 15:13:56 +0100 Jan Schmidt * common: Automatic update of common submodule From f3bb51b to f810030 2009-06-24 09:48:41 +0100 Tim-Philipp Müller * gst-libs/gst/pbutils/descriptions.c: pbutils: add description for multipart So we get slightly nicer error messages when multipartdemux is missing. 2009-06-23 18:07:31 +0200 Wim Taymans * gst/adder/gstadder.c: adder: only unflush when we flushed before Ass suggested by Stefan Kost: Keep track of when the sinkpad was set to flushing and unflush the pad when an upstream flushing seek failed. 2009-06-23 15:10:37 +0100 Tim-Philipp Müller * gst/playback/gsturidecodebin.c: uridecodebin: fix leak when the source fails to change state 2009-06-23 12:40:56 +0200 Wim Taymans * gst/subparse/gstssaparse.c: ssaparse: avoid leaking all buffers 2009-06-22 22:18:03 +0300 Stefan Kost * tests/check/elements/adder.c: adder: test seek handling in adder This tests seeking on an adder that has a normal and a live source connected. Wheter the current behavior is the desired one needs to be discussed still (see #586033) 2009-06-22 16:17:10 +0300 Stefan Kost * sys/ximage/ximagesink.c: * sys/xvimage/xvimagesink.c: x(v)imagesink: pass the xwindow along to not look at the yet unset var. When we call this from xwindow_new, x(v)imagesink->xwindow is not yet set. 2009-06-22 11:40:33 +0300 Stefan Kost * sys/ximage/ximagesink.c: * sys/ximage/ximagesink.h: * sys/xvimage/xvimagesink.c: * sys/xvimage/xvimagesink.h: x(v)imagesink: catch tags and show title in own window Refactor the code that sets the window title. Catch tag-events and use title metadata for the window title. 2009-06-21 19:42:15 +0200 Sebastian Dröge * gst/audiotestsrc/gstaudiotestsrc.c: audiotestsrc: Name gaussian noise "gaussian-noise" instead of just "gaussian" Also make all the function arrays constant. 2009-06-21 12:27:37 +0200 Kipp Cannon * gst/audiotestsrc/gstaudiotestsrc.c: * gst/audiotestsrc/gstaudiotestsrc.h: audiotestsrc: Add support for generating gaussian white noise This patch adds support for stationary white Gaussian noise. The Box-Muller algorithm is used to generate pairs of independent normally-distributed random numbers. Fixes bug #586519. 2009-06-20 23:46:28 +0100 Jan Schmidt * gst/ffmpegcolorspace/imgconvert.c: * gst/ffmpegcolorspace/imgconvert_template.h: ffmpegcolorspace: Fix NV12 and NV21 transformations Fix some stride problems, fix the nv12 to nv21 direct transformation, and implement a direct conversion to yuv444 to save CPU. 2009-06-20 22:36:21 +0100 Jan Schmidt * gst/videotestsrc/videotestsrc.c: videotestsrc: Fix NV12 painting for odd strides/heights 2009-06-19 22:16:43 +0100 Tim-Philipp Müller * ext/cdparanoia/gstcdparanoiasrc.c: cdparanoia: run-time license is LGPL now that we require cdparanoia 0.10.2 cdparanoia has an LGPL v2.1 license since 0.10.1 and we now require 0.10.2. Finally fixes #531035. 2009-06-19 21:25:54 +0100 Tim-Philipp Müller * ext/cdparanoia/gstcdparanoiasrc.c: cdparanoia: try to guess a good cache size if it's set to -1 Try to guess from the paranoia-mode setting whether playback or ripping is wanted, and use a smaller cache size if we're likely to be doing playback, to avoid a long startup delay. Since this was the value used in older cdparanoia versions, it should be fine in any case. See #586331. 2009-06-19 11:27:40 +1000 Jonathan Matthew * configure.ac: * ext/cdparanoia/gstcdparanoiasrc.c: * ext/cdparanoia/gstcdparanoiasrc.h: cdparanoia: expose cache size setting This setting was added in cdparanoia 10.2. The default value is good for audio extraction, but lower values (previous versions of cdparanoia used 150) are better for realtime playback. Fixes #586331. 2009-06-19 17:43:03 +0100 Christian Schaller * gst-plugins-base.spec.in: Make build of schro plugin conditional 2009-06-19 15:52:34 +0200 Wim Taymans * docs/libs/gst-plugins-base-libs-sections.txt: * gst-libs/gst/rtp/gstbasertppayload.c: * gst-libs/gst/rtp/gstbasertppayload.h: * win32/common/libgstrtp.def: basertppayload: add support for bufferlists Based on patch from Ognyan Tonchev. See #585559 2009-06-19 15:33:04 +0200 Wim Taymans * gst-libs/gst/rtp/gstrtpbuffer.c: rtpbuffer: use new convenience functions New core convenience functions makes the list getters and setters trivial. Maybe even too trivial... 2009-06-18 19:07:22 +0200 Wim Taymans * win32/common/libgstrtp.def: defs: add new symbol to win32 defs file Based on patches by Ognyan Tonchev. See #585559 2009-06-18 19:04:52 +0200 Wim Taymans * docs/libs/gst-plugins-base-libs-sections.txt: * gst-libs/gst/rtp/gstrtpbuffer.c: rtp: cleanups, add _list_get_seq() too Clean up the docs a little. Add missing _list_get_seq method. Add new symbols to the docs 2009-06-18 18:47:49 +0200 Wim Taymans * gst-libs/gst/rtp/gstrtpbuffer.c: * win32/common/libgstrtp.def: rtp: cleanups Add Since tags to docs Move some code around Add win32 symbols 2009-06-18 17:46:01 +0200 Wim Taymans * gst-libs/gst/rtp/gstrtpbuffer.c: * gst-libs/gst/rtp/gstrtpbuffer.h: * tests/check/libs/rtp.c: rtp: add bufferlist support 2009-06-18 18:03:40 +0200 Wim Taymans * gst-libs/gst/rtp/gstrtpbuffer.c: rtp: pass data to macros instead of GstBuffer 2009-06-18 17:42:10 +0100 Jan Schmidt * win32/common/libgstrtsp.def: win32: Add gst_rtsp_watch_queue_data() to the exports Fix the tests by exporting the new symbol from the win32 dlls 2009-06-18 18:13:22 +0300 Stefan Kost * sys/xvimage/xvimagesink.c: xvimagesink: appname might be NULL Don't set title if appname is unknown. 2009-06-18 17:58:06 +0300 Stefan Kost * sys/xvimage/xvimagesink.c: xvimagesink: set window title from application name 2009-06-09 19:14:00 +0200 Peter Kjellerstedt * gst-libs/gst/rtsp/gstrtspurl.c: rtsp: Made the parsing of the RTSP URL scheme more generic. 2009-06-15 13:58:26 +0200 Peter Kjellerstedt * gst-libs/gst/rtsp/gstrtspconnection.c: * gst-libs/gst/rtsp/gstrtspconnection.h: rtsp: Added gst_rtsp_watch_queue_data(). gst_rtsp_watch_queue_data() is similar to gst_rtsp_watch_queue_message() but allows for queuing any data block for writing (much like gst_rtsp_connection_write() vs. gst_rtsp_connection_send().) API: gst_rtsp_watch_queue_data() 2009-06-09 16:37:09 +0200 Peter Kjellerstedt * gst-libs/gst/rtsp/gstrtspconnection.c: rtsp: Only extract the session ID from RTSP responses. 2009-06-09 19:06:57 +0200 Peter Kjellerstedt * gst-libs/gst/rtsp/gstrtspurl.c: rtsp: Added support for parsing IPv6 addresses in RTSP URLs. 2009-06-09 14:31:18 +0200 Peter Kjellerstedt * gst-libs/gst/rtsp/gstrtspconnection.c: rtsp: Use getaddrinfo() to support both IPv4 and IPv6. 2009-06-17 15:37:53 +0200 Peter Kjellerstedt * gst-libs/gst/rtsp/gstrtspconnection.c: rtsp: Improved base64 decoding in fill_bytes(). The base64 decoding in fill_bytes() expected the size of the read data to be evenly divisible by four (which is true for the base64 encoded data itself). This did not, however, take whitespace (especially line breaks) into account and would fail the decoding if any whitespace was present. 2009-06-17 14:00:23 +0200 Wim Taymans * gst-libs/gst/audio/gstbaseaudiosrc.c: audiosrc: fix get_offset When we need to jump to the most recently captured sample, jump to where the next sample will be written instead of to some old data. Fixes #581460 2009-06-17 13:18:18 +0200 Wim Taymans * gst-libs/gst/audio/gstbaseaudiosink.c: audiosink: free the ringbuffer when going to NULL Unparent and free the ringbuffer when going to NULL, like we do with the audiosrc element. We can do this now because we correctly manage the time jumping back to 0. 2009-06-17 13:17:30 +0200 Wim Taymans * gst-libs/gst/audio/gstaudiosink.c: * gst-libs/gst/audio/gstaudiosrc.c: audio: correctly handle short read/writes 2009-05-05 15:37:54 +0300 René Stadler * gst-libs/gst/audio/gstbaseaudiosrc.c: baseaudiosrc: add some extra logging for buffer timestamps 2009-06-17 11:22:51 +0200 Wim Taymans * gst/adder/gstadder.c: adder: more seeking fixes. When a seek failed upstream, make sure the adder sinkpad is set unflushing again so that streaming can continue. We only have a pending segment when we flushed. Set the flush_stop_pending flag inside the appropriate locks and before we attempt to perform the upstream seek. Add some more comments. Use the right lock to protect the flags in flush_stop. See #585708 2009-06-17 07:24:53 +0200 Sebastian Dröge * gst/playback/gstdecodebin2.c: decodebin2: Free iterator after removing all groups 2009-06-16 19:38:17 +0200 Sebastian Dröge * gst-libs/gst/video/gstvideofilter.c: videofilter: Add a default get_unit_size function This returns the correct values for all formats that are handled by GstVideoFormat and makes all the custom get_unit_size functions in many elements unnecessary. 2009-06-16 18:57:20 +0200 Wim Taymans * gst-libs/gst/rtsp/gstrtspdefs.c: * gst-libs/gst/rtsp/gstrtspdefs.h: rtsp: add Timestamp header field fixes #585994 2009-06-16 18:15:06 +0200 Wim Taymans * gst/playback/gstplaybin2.c: playbin2: set smarter target state on uridecodebin Set the target state of the newly added uridecodebins to somthing else that PAUSED so that we keep their state in sync with the playsink state. Fixes #585268 2009-06-16 18:13:53 +0200 Wim Taymans * gst/playback/gstplaysink.c: playsink: set the sink flag on the element 2009-06-16 18:09:43 +0200 Wim Taymans * gst/playback/gsturidecodebin.c: uridecodebin: add debug message 2009-06-16 14:05:04 +0100 Tim-Philipp Müller * gst-libs/gst/audio/gstaudiosink.c: * gst-libs/gst/audio/gstaudiosrc.c: audiosink, audiosrc: do the class_ref()s in the right class_init functions Spotted by Philip Jägenstedt. Hopefully fixes #585970 for real. 2009-06-15 15:39:09 +0100 Tim-Philipp Müller * gst-libs/gst/audio/gstaudiosink.c: * gst-libs/gst/audio/gstaudiosrc.c: audiosink,audiosrc: ref the audio ring buffer class and type in class_init Hack around thread-safety issues in GObject and our racy _get_type() functions (we could easily fix the _get_type() functions, but we still need to hack around the GObject class races until we require a newer GLib version, I think). 2009-06-15 12:57:39 +0200 Wim Taymans * gst-libs/gst/audio/gstbaseaudiosrc.c: audiosrc: return FALSE when receiving a SEEK event When receiving a seek event, return FALSE as we don't implement seeking. 2009-06-15 11:06:25 +0200 Sebastian Dröge * tests/examples/seek/seek.c: Don't use deprecated GTK API Fixes bug #585758. 2009-06-15 11:40:00 +0300 Stefan Kost * gst/adder/gstadder.c: adder: send flush_stop when seeking failed At least do the fix to sent the flush_stop when seeking failed to ensure we keep no pads flushing. before it was send when the seeking worked which is just plain wrong and was not the intention. 2009-06-12 15:17:14 +0200 Peter Kjellerstedt * gst-libs/gst/rtsp/gstrtspconnection.c: rtsp: Use a more consistent naming of GstRTSPRec variables. 2009-06-12 15:11:05 +0200 Peter Kjellerstedt * gst-libs/gst/rtsp/gstrtspconnection.c: * gst-libs/gst/rtsp/gstrtspconnection.h: rtsp: Call message_sent() callback for all sent messages. Previously the messages_sent() callback was only called for messages which had a CSeq, which excluded all data messages. Instead of using the CSeq as ID, use a simple index counter. 2009-06-14 22:13:41 +0100 Tim-Philipp Müller * ext/ogg/gstoggdemux.c: * ext/theora/theoradec.c: * ext/vorbis/vorbisdec.c: oggdemux: post/send tags with the container-format tag For this to work properly, theoradec and vorbisdec need to put tag events received from upstream into the pending_events list so they get pushed out after any newsegment event, not before. 2009-06-14 20:30:59 +0200 Sebastian Dröge * tests/examples/seek/scrubby.c: * tests/examples/seek/seek.c: * tests/old/examples/seek/cdplayer.c: Don't use deprecated GTK API Fixes bug #585758. 2009-06-12 16:31:00 +0200 Wim Taymans * gst/adder/gstadder.c: adder: send flush-stop earlier When no flush-stop has been sent by upstream, we have to send one ourselves to continue playback. Do this as soon as the collect function is called instead of after we possibly pushed segment events (that got then flushed out) 2009-06-12 13:55:33 +0200 Wim Taymans * tests/examples/seek/seek.c: seek: add shuttle controls 2009-06-12 13:55:02 +0200 Wim Taymans * tests/examples/seek/stepping2.c: example: fix compile 2009-06-12 13:52:25 +0200 Wim Taymans * tests/examples/seek/Makefile.am: examples: build the stepping2 example 2009-06-12 13:52:02 +0200 Wim Taymans * gst/playback/gstplaysink.c: playsink: update for new step API 2009-06-12 13:22:47 +0200 Wim Taymans * ext/ogg/gstoggdemux.c: oggdemux: do reverse seeks more accurate For reverse seeking with the accurate flag set, try to be more precise by seeking a little bit after the requested position. 2009-06-11 22:32:28 +0100 Tim-Philipp Müller * ext/ogg/gstogmparse.c: * gst/subparse/gstssaparse.c: * gst/subparse/gstssaparse.h: * gst/subparse/gstsubparse.c: * gst/subparse/gstsubparse.h: subparse, ogmparse: post tags with GST_TAG_SUBTITLE_CODEC Make subtitle parsers post a taglist with codec tags, so the application knows what kind of subtitle a subtitle stream is. Fixes #576552. 2009-06-11 19:12:51 +0200 Wim Taymans * gst-libs/gst/audio/gstringbuffer.c: ringbuffer: handle border cases in resampler 2009-06-11 13:28:20 +0100 Jan Schmidt * common: * docs/libs/Makefile.am: * docs/plugins/Makefile.am: docs: Update common. Use upload-doc.mak instead of upload.mak 2009-06-11 12:39:19 +0200 Wim Taymans * gst-libs/gst/rtp/gstbasertppayload.c: docs: fix typo 2009-06-11 12:17:16 +0200 Wim Taymans * gst-libs/gst/audio/gstbaseaudiosink.c: baseaudiosink: reset accum when dropping samples When we are resampling and we drop samples because we paused, reset the accum counter because it's now invalid. 2009-06-11 11:16:15 +0100 Jan Schmidt * docs/libs/gst-plugins-base-libs-sections.txt: * gst-libs/gst/interfaces/mixer.h: * gst-libs/gst/video/gstbasevideodecoder.h: docs: Fix a couple of warnings from the docs build. 2009-06-10 21:36:19 +0100 Tim-Philipp Müller * gst-libs/gst/audio/testchannels.c: Don't include config.h multiple times when build audio testchannel app. Fixes build problem on win32 (#585075). 2009-06-10 16:56:51 +0100 Jan Schmidt * gst/playback/gstplaybin2.c: * gst/playback/gsturidecodebin.c: playbin2/uridecodebin: Fix connection-speed propagation uridecodebin expects the passed connection-speed value in kbps, so we need to divide the value stored in bps by 1000. Also, lower the upper limit on the properties to the value that we can actually store in our internal guint (which is plenty high enough) 2009-06-10 14:37:36 +0100 Tim-Philipp Müller * gst/subparse/gstsubparse.c: * tests/check/elements/subparse.c: subparse: recognise more subrip timestamp variants Be even less restrictive in what we accept for .srt timestamps when typefinding and parsing subrip subtitles and add a unit test for the 'new' format. Fixes #585197. 2009-06-09 22:00:53 +0200 Wim Taymans * gst-libs/gst/rtsp/gstrtsptransport.h: rtsp: add some more docs 2009-06-09 18:24:55 +0200 Peter Kjellerstedt * gst-libs/gst/rtsp/gstrtspmessage.c: rtsp: Avoid a compiler warning. 2009-06-09 18:23:28 +0200 Peter Kjellerstedt * gst-libs/gst/rtsp/gstrtspdefs.h: rtsp: Updated documentation for GstRTSPResult. Moved GST_RTSP_ELAST to be last in the documentation to match the actual enum values. 2009-05-20 17:30:23 +0100 Tim-Philipp Müller * autogen.sh: autogen: remove -Wno-portability from here as it is in configure.ac now. 2009-06-09 16:28:20 +0200 Peter Kjellerstedt * gst-libs/gst/rtsp/gstrtspconnection.c: rtsp: Plug a memory leak. Free memory related to any partially read and/or written RTSP messages. 2009-06-09 12:09:15 +0200 Wim Taymans * gst-libs/gst/audio/gstbaseaudiosink.c: baseaudiosink: no need to cause discont when clipping Remove the discont-when-clipping hack now that basesink provides us with correctly clipped samples when stepping. 2009-06-08 17:26:59 +0200 Wim Taymans * gst-libs/gst/audio/gstbaseaudiosink.c: audiosink: don't align when we clip Don't align samples when they were clipped. Not entirely correct but better than nothing for now. 2009-06-08 16:41:58 +0200 Wim Taymans * tests/examples/seek/.gitignore: * tests/examples/seek/stepping2.c: examples: add stepping example in PLAYING Add stepping example in PLAYING, audio is a bit distorted because basesink does not provide good clipping info yet. 2009-06-08 10:25:00 +0200 Edward Hervey * gst-libs/gst/pbutils/descriptions.c: pbutils: Add description for hdv/aux-* formats. 2009-06-07 22:20:33 +0400 LRN * ext/schroedinger/Makefile.am: Added libgstbase to schro's LIBADD Fixes #585079 2009-06-06 02:15:05 +0100 Tim-Philipp Müller * gst-libs/gst/tag/gstid3tag.c: libgsttag: don't extract genres from empty ID3v1 tags If we don't have any other info, don't try to interpret the genre field. In particular we don't want to interpret a genre of 0 as 'Blues' if no other fields are set and the entire tag is just empty. 2009-06-05 18:13:25 +0200 Wim Taymans * gst/playback/gstdecodebin2.c: decodebin2: make sure varargs are of right type Explicitly cast the variables to g_object_set to their right types. 2009-06-05 16:49:58 +0200 Wim Taymans * gst/playback/gstdecodebin2.c: decodebin2: increase stream probing queues When we are probing for streams, we want to set the queue size in such a way that we can scan a maximum amount of data without consuming too much memory. Therefore, remove the time limit on the queue and only stop scanning after 2MB of data. See #584104. 2009-06-05 14:06:17 +0200 Peter Kjellerstedt * gst-libs/gst/rtsp/gstrtspconnection.c: rtsp: Fixed a typo. 2009-06-05 14:05:54 +0200 Peter Kjellerstedt * gst-libs/gst/rtsp/gstrtspconnection.c: rtsp: Remove an unused variable. 2009-06-05 13:59:14 +0200 Peter Kjellerstedt * gst-libs/gst/rtsp/gstrtspconnection.c: rtsp: Removed duplicate initialization of conn->writefd. 2009-06-05 13:55:08 +0200 Peter Kjellerstedt * gst-libs/gst/rtsp/gstrtspconnection.c: rtsp: Use #defined status codes. 2009-06-05 13:53:29 +0200 Peter Kjellerstedt * gst-libs/gst/rtsp/gstrtspconnection.c: rtsp: Correct gen_tunnel_reply(). Prevent gen_tunnel_reply() from generating an incomplete response in case an error response code is given. 2009-06-05 10:57:44 +0100 Tim-Philipp Müller * configure.ac: * win32/common/_stdint.h: * win32/common/config.h: * win32/common/video-enumtypes.c: configure: remove AC_C_INLINE which is not needed and causes problems with MSVC See #584835. Also update win32 files while we're at it. 2009-06-04 08:57:24 +0200 Sebastian Dröge * gst/playback/gstplaybin2.c: playbin2: API: Add {audio,video,text}-tags-changed signals Fixes bug #584686. 2009-06-03 20:42:39 +0100 Tim-Philipp Müller * ext/vorbis/vorbisdec.c: vorbisdec: don't put invalid bitrate values into the taglist Bitrates are stored as 32-bit signed integers in the vorbis identification headers, but seem to be read incorrectly, namely as unsigned 32-bit integers, into the vorbis structure members which are of type long, which makes our check for values <= 0 fail with files that put -1 in there for unset values. 2009-06-03 15:52:54 +0200 Wim Taymans * tests/examples/seek/.gitignore: ignore: add new stepping app to ignore 2009-06-03 15:31:27 +0200 Wim Taymans * tests/examples/seek/Makefile.am: * tests/examples/seek/stepping.c: examples: add stepping example. Add an example of using playbin2 and frame stepping to simulate variable rate playback based on a sine wave. 2009-06-03 12:45:08 +0200 Wim Taymans * gst/playback/gstplaybin2.c: * gst/playback/gstplaysink.h: playbin2: also set custom text and subp sinks Set the custom subpicture and text sinks along with the custom audio and video sinks when needed. Fix a little docs blurb too. 2009-06-02 12:10:39 +0200 Wim Taymans * gst-libs/gst/rtsp/gstrtspconnection.c: * gst-libs/gst/rtsp/gstrtspconnection.h: rtsp: add G_LIKELY because we can 2009-06-02 09:53:05 +0200 Edward Hervey * gst/typefind/gsttypefindfunctions.c: typefindfunctions: Fix caps for ogg typefinder. 2009-05-29 11:10:07 +0100 Tim-Philipp Müller * docs/libs/gst-plugins-base-libs-sections.txt: docs: remove some cruft from -sections.txt file 2009-06-01 11:31:49 +0200 Wim Taymans * gst/playback/gstplaysink.c: * tests/examples/seek/seek.c: add framestepping to playbin2 and seek 2009-06-01 09:59:22 +0200 Peter Kjellerstedt * gst-libs/gst/rtsp/gstrtspconnection.c: rtsp: Avoid compiler warnings with -Wextra. 2009-06-01 09:58:27 +0200 Peter Kjellerstedt * gst-libs/gst/rtsp/gstrtspconnection.h: rtsp: Include gst/gstconfig.h to make sure GST_PADDING is defined. 2009-06-01 09:43:04 +0200 Peter Kjellerstedt * gst-libs/gst/sdp/gstsdpmessage.c: sdp: Remove an unused variable. 2009-05-30 14:17:41 +0200 Sebastian Dröge * gst/ffmpegcolorspace/imgconvert.c: * gst/ffmpegcolorspace/imgconvert_template.h: ffmpegcolorspace: Add a lot more conversions from/to 16 bit grayscale 2009-05-29 00:09:15 +0100 Jan Schmidt * gst/playback/gstplaybin2.c: playbin2: Have playbin recognise PGS subpicture streams Recognise PGS subpicture streams and connect them to the SPU pad in playsink. Unfortunately this fails badly with negotiation errors if the SPU is not recent enough to support the stream. I'm not sure how to add format negotiation in yet. 2009-05-21 23:11:29 +0100 Jan Schmidt * gst/playback/gstdecodebin2.c: * gst/playback/gsturidecodebin.c: decodebin/uridecodebin: Recognise subpicture/x-pgs pads and output them. 2009-05-28 20:37:59 +0100 Tim-Philipp Müller * gst/playback/gstplaysink.c: playbin2: fix volume handling for audio sinks without "volume" property When using an audio sink without a "volume" property, volume control would only work for the first song. For the next song, we'd try to re-use the existing audio chain, but inadvertently set chain->volume to NULL instead of to the existing volume element. 2009-05-28 17:05:55 +0100 Tim-Philipp Müller * gst/playback/gstplaysink.c: playbin2: cosmetic change to avoid unnecessary line breaks Looks nicer and works around gst-indent silliness. 2009-05-28 17:21:35 +0200 Wim Taymans * gst/playback/gstplaysink.c: playbin2: don't lose the ref to the volume element Only release the ref to the volume element when it is controled by a sink. For software volume we never have to fear that it will change. 2009-05-28 15:21:42 +0100 Tim-Philipp Müller * gst/playback/gstplaybin2.c: * gst/playback/gstplaysink.c: playbin2: actually use configured audio/video sinks playbin2 inadvertently used autoaudiosink and autovideosink up to now, since it would overwrite the sinks configured via the "audio-sink" and "video-sink" properties with the stream-specific group sinks when configuring the outputs. Those are usually NULL however, so that would overwrite the configured sinks with NULL which makes playbin2 then default to the auto sinks. Fix this by keeping a reference to each configured sink in playbin2 and setting up the right sinks depending on whether there is a stream-specific sink or not. Fixes #584020. 2009-05-27 17:37:38 +0300 Stefan Kost * tests/examples/seek/seek.c: seek: add volume label and sync with sink volume Look at the volume and have the pulsemixer open at same time. Unfortunately playbin2 does not emit notify on volume right, so this polls for now. 2009-05-27 18:12:10 +0200 Wim Taymans * gst/playback/gstdecodebin2.c: decodebin2: remove leftover elements Remove all of the elements inside decodebin2 when goint to READY and NULL. Makes decodebin2 reusable. Fixes #583750 2009-05-27 15:36:51 +0200 Wim Taymans * gst/playback/gstplaysink.c: playbin2; release refs to volume/mute properties Release the refs to the volume and mute property elemens before setting the child elements to READY or NULL. Fixes #583318 2009-05-27 12:10:05 +0200 Wim Taymans * gst/gdp/gstgdppay.c: gdppay: set caps on outgoing buffers Set caps on outgoing buffers because NULL caps confuse basetransform. Fixes #583867 2009-05-27 11:08:37 +0200 Wim Taymans * gst-libs/gst/netbuffer/gstnetbuffer.c: netbuffer: also note the order of IP4 addresses IP4 addresses are also stored in network byte order. Make a note of this in the docs. 2009-05-26 22:43:34 +0200 Alessandro Decina * ext/theora/theoraparse.c: theoraparse: fix assertions in make_granulepos when using the new theora granulepos mapping. Fixes #583903. 2009-05-26 11:13:35 +0100 Tim-Philipp Müller * gst-libs/gst/rtsp/gstrtspconnection.c: Revert "rtspconnection: don't use GLib-2.16 API, we require only 2.14" This reverts commit 418760cf740332c12c3fd9cf3244af134fa9534b. We now require GLib 2.16. 2009-05-26 15:18:09 +0100 Jan Schmidt * common: Update common 2009-05-26 15:37:18 +0200 Wim Taymans * gst-libs/gst/netbuffer/gstnetbuffer.c: netbuffer: document that the port is network order Document the fact that we store the port number in network order in GstNetAddress and that the caller should byteswap appropriately. 2009-05-26 15:23:45 +0200 Sebastian Dröge * gst/videoscale/gstvideoscale.c: * gst/videoscale/vs_4tap.c: * gst/videoscale/vs_4tap.h: * gst/videoscale/vs_image.c: * gst/videoscale/vs_image.h: * gst/videoscale/vs_scanline.c: * gst/videoscale/vs_scanline.h: videoscale: Add support for 16 bit grayscale in native endianness 2009-05-26 14:58:28 +0200 Sebastian Dröge * gst/ffmpegcolorspace/avcodec.h: * gst/ffmpegcolorspace/gstffmpegcodecmap.c: * gst/ffmpegcolorspace/imgconvert.c: ffmpegcolorspace: Add support for 16 bit grayscale in little/big endian 2009-05-26 14:38:43 +0200 Sebastian Dröge * gst/videotestsrc/videotestsrc.c: * gst/videotestsrc/videotestsrc.h: videotestsrc: Add support for 16 bit grayscale in native endianness 2009-01-21 12:33:59 +0100 Andy Wingo add can-activate-pull property to baseaudiosink * gst-libs/gst/audio/gstbaseaudiosink.c: Add can-activate-pull property to baseaudiosink. 2009-05-26 13:14:07 +0200 Wim Taymans * ext/ogg/gstoggdemux.c: oggdemux: fix boundary case for seeking. When we have exactly 0 bytes left to search, make sure we stop instead of going into an infinite loop. 2009-05-26 11:11:03 +0200 Bastien Nocera * gst-libs/gst/cdda/Makefile.am: * gst-libs/gst/cdda/gstcddabasesrc.c: * gst-libs/gst/cdda/sha1.c: * gst-libs/gst/cdda/sha1.h: cddabasesrc: Remove copy of sha1 digest Remove our copy of sha1 digest now that we depend on glib 2.16. Fixes #536313 2009-05-25 17:54:01 +0100 Christian Schaller * gst-plugins-base.spec.in: Update spec file 2009-05-23 00:33:04 +0100 Tim-Philipp Müller * gst-libs/gst/video/gstbasevideodecoder.c: * gst-libs/gst/video/gstbasevideoparse.c: * gst-libs/gst/video/gstbasevideoutils.c: * gst-libs/gst/video/gstbasevideoutils.h: * win32/common/libgstvideo.def: video: don't expose internal gst_adapter_get_buffer() helper function If it's really needed it should go into GstAdapter in core. 2009-05-22 21:29:51 -0700 David Schleef * gst-libs/gst/video/gstbasevideodecoder.c: basevideo: Fix memleak 2009-05-22 21:27:58 -0700 David Schleef * ext/schroedinger/gstschrodec.c: * ext/schroedinger/gstschroparse.c: schro: Fix usage of adapter_masked_scan_uint32 Because *somebody* changed the API without telling me. 2009-05-22 21:25:06 -0700 David Schleef * ext/schroedinger/gstschro.c: schro: Change package name to GST_PACKAGE_NAME 2009-05-22 17:34:10 -0700 David Schleef * gst-libs/gst/video/gstbasevideoencoder.c: basevideo: Add preset interface to encoder 2009-05-22 17:31:14 -0700 David Schleef * gst/audioresample/gstaudioresample.c: Run liboil benchmark multiple times The statistics function requires multiple runs, otherwise it causes a divide by zero error. 2009-05-22 19:36:06 +0100 Tim-Philipp Müller * m4/gst-fionread.m4: m4: fix 'suspicious cache value' warning for gst-fionread.m4 .. here as well (should really be moved to common, but I'm too lazy). 2009-05-22 17:41:50 +0200 Wim Taymans * ext/vorbis/vorbisdec.c: vorbisdec: detect and report errors better Check the return values of a couple more libvorbis functions and post an error when something is wrong instead of continuing and crashing. 2009-05-22 15:49:14 +0300 Stefan Kost * gst/playback/gstplaysink.c: playbin2: fix initial volume and mute handling Use two flags to remember volume/mute changes at times when we don't have the audiochain yet (e.g. construction). Only set values when they were actualy changed. This makes pulseaudio's stream restore functional. 2009-05-22 10:19:51 +0100 Jan Schmidt * common: Automatic update of common submodule From d3a8fab to 888e0a2 2009-05-22 09:03:22 +0100 Jan Schmidt * win32/common/libgstvideo.def: win32: Remove gst_adapter_masked_scan_uint32 from the exports 2009-05-21 10:48:49 +0200 Wim Taymans * gst-libs/gst/audio/gstbaseaudiosink.c: audiosink: improve debug message 2009-05-19 18:10:55 -0700 Michael Smith * gst-libs/gst/tag/gstid3tag.c: gstid3tag: Don't extract a track number unless present. In ID3v1, a track number is present only if byte 125 is null AND byte 126 is non-null. If the track number is not present, don't add a track number tag with value 0. 2009-05-20 00:48:40 +0200 Wim Taymans * gst-libs/gst/video/gstbasevideoutils.c: * gst-libs/gst/video/gstbasevideoutils.h: videoutils: remove adapter methods Remove adapter methods now that they are in core. 2009-05-20 00:42:29 +0200 Wim Taymans * win32/common/libgstvideo.def: defs: add new symbols 2009-05-19 17:47:34 +0100 Tim-Philipp Müller * configure.ac: autogen: pass -Wno-portability to automake to suppress warnings GNU make is needed. 2009-05-19 02:28:20 +0100 Tim-Philipp Müller * docs/libs/.gitignore: gitignore: remove bogus *.sgml wildcard - these files are tracked in git 2009-05-19 18:41:58 +0200 Wim Taymans * gst/tcp/gsttcpclientsrc.c: tcpclientsrc: this is not a live source Don't mark us as a live source because we are not. 2009-05-19 18:41:02 +0300 Stefan Kost * gst/adder/gstadder.c: adder: only send flush_stop when seek failed This is still not the ultimate fix. Added some comment to explain the troubles. 2009-05-19 17:17:37 +0200 Wim Taymans * gst-libs/gst/audio/gstbaseaudiosink.c: audiosink: return the return value of wait_preroll Return the value that _wait_preroll() returned instead of always WRONG_STATE. 2009-05-19 16:45:56 +0300 Stefan Kost * gst/adder/gstadder.c: * gst/adder/gstadder.h: adder: send flush_stop to match flush_start Adder was relying that something else sends a flush stop. When using adder with a livesource it was not getting a flush_stop and thus all pads downstream where keept flushing. Mark a pending flush_stop and send it when we are working on the new segment back in the streaming thread. 2009-05-19 16:02:44 +0300 Stefan Kost * tests/examples/seek/seek.c: seek: ui improvements Repaint the window black on expose, as this looks nicer when resizing or using the expander. Also show time after slider, as this saves a whole line (nice on small displays). 2009-04-29 18:36:17 +0300 Stefan Kost * gst/playback/gstdecodebin.c: decodebin: use iterators instead of list The list api is deprecated. Use threadsafe iterators instead. 2009-05-19 15:35:54 +0200 Wim Taymans * gst/playback/gsturidecodebin.c: uridecodebin: configure caps on decodebin2 Implement the caps property by setting the configured caps on new decodebin2 objects. Fixes #582749 2009-05-19 15:34:38 +0200 Wim Taymans * gst/playback/gstdecodebin2.c: decodebin2: avoid some _caps_ref in some cases Only mess with the caps refcount when we configure different caps. 2009-05-19 15:27:12 +0200 Wim Taymans * gst/playback/gsturidecodebin.c: uridecodebin: fix potential caps leak Free the user-configured caps in finalize. 2009-05-19 15:20:27 +0200 Wim Taymans * gst/playback/gsturidecodebin.c: uridecodebin: add queue after cdda:// Add a queue2 after the raw output pads of certain sources such as those for uris like cdda:// No tuning of the queue is done yet as the defaults seem to work fine for me. Fixes #582528 2009-05-19 12:45:59 +0200 Wim Taymans * ext/ogg/gstoggdemux.c: oggdemux: don't loop when at EOS When we try to read the last page, don't try to read past the upper boundary, as this might cause endless loops. See #582942 2009-05-19 11:20:19 +0200 Edward Hervey * gst/audioresample/gstaudioresample.c: audioresample: Don't drain remaining buffers after a flush. If we were resetted (due to a flush), we can not drain the remaining buffers since they would be pushed before a valid new newsegment event. 2009-05-18 22:29:07 -0700 Michael Smith * ext/theora/theoradec.c: theoradec: for 4:2:2, use Y42B (planar) rather than a packed format. 2009-05-19 01:13:34 +0300 Stefan Kost * gst/adder/gstadder.c: adder: add more logging and return value checking 2009-05-19 01:11:45 +0300 Stefan Kost * gst/adder/gstadder.c: adder: handle the return value from iterator_fold 2009-05-19 01:03:44 +0300 Stefan Kost * gst/adder/gstadder.c: adder: use the pad in logging as objects Helps to differenciate between source and sinks pads. 2009-04-21 22:54:19 +0300 Stefan Kost * tests/examples/seek/seek.c: seek: use parser for mp3 and rename variable 2009-05-18 11:08:56 +0200 Wim Taymans * tests/examples/seek/seek.c: seek: add playbin2 options in expander Add the playbin2 stream selection options inside an expander to preserve some space on screen. 2009-02-10 15:29:10 -0800 David Schleef * gst/videotestsrc/videotestsrc.c: videotestsrc: Add support for v210 and v216 formats 2009-05-15 16:21:15 -0700 David Schleef * gst-libs/gst/video/gstbasevideocodec.c: * gst-libs/gst/video/gstbasevideodecoder.c: * gst-libs/gst/video/gstbasevideoencoder.c: * gst-libs/gst/video/gstbasevideoparse.c: video: remove // comments 2009-05-15 16:18:18 -0700 David Schleef * gst-libs/gst/video/video.c: * gst-libs/gst/video/video.h: video: Add Y444, v210, v216 formats 2009-05-15 16:12:37 -0700 David Schleef * configure.ac: * ext/Makefile.am: * ext/schroedinger/Makefile.am: * ext/schroedinger/gstschro.c: * ext/schroedinger/gstschrodec.c: * ext/schroedinger/gstschroenc.c: * ext/schroedinger/gstschroparse.c: * ext/schroedinger/gstschroutils.c: * ext/schroedinger/gstschroutils.h: schro: Move schro plugin from Schroedinger Previous history is in Schroedinger. Depends on, and is an example of using, GstBaseVideo* base classes. Code was reindented, and an #ifdef HAVE_ENCODER removed. 2009-05-15 10:23:08 -0700 David Schleef * gst-libs/gst/video/Makefile.am: * gst-libs/gst/video/gstbasevideocodec.c: * gst-libs/gst/video/gstbasevideocodec.h: * gst-libs/gst/video/gstbasevideodecoder.c: * gst-libs/gst/video/gstbasevideodecoder.h: * gst-libs/gst/video/gstbasevideoencoder.c: * gst-libs/gst/video/gstbasevideoencoder.h: * gst-libs/gst/video/gstbasevideoparse.c: * gst-libs/gst/video/gstbasevideoparse.h: * gst-libs/gst/video/gstbasevideoutils.c: * gst-libs/gst/video/gstbasevideoutils.h: video: Copy BaseVideo classes from Schroedinger 2009-05-15 23:05:45 +0200 Arnout Vandecappelle * gst/tcp/gstmultifdsink.c: multifdsink: add num-fds property multifdsink::num-fds 2009-05-15 20:36:29 +0100 Tim-Philipp Müller * gst-libs/gst/pbutils/descriptions.c: pbutils: add descriptions for 3GP, JPEG 2000 and Motion JPEG 2000 2009-05-14 11:44:27 +0200 Sebastian Dröge * ext/vorbis/vorbisenc.c: vorbisenc: Implement Preset interface 2009-05-14 11:43:07 +0200 Sebastian Dröge * ext/theora/theoraenc.c: theoraenc: Implement Preset interface 2009-05-14 11:41:13 +0200 Sebastian Dröge * ext/ogg/gstoggmux.c: oggmux: Implement Preset interface 2009-05-14 21:37:22 +0100 Jan Schmidt * gst/playback/gstplaysink.c: playbin2: Fix cdda:// playback Don't send async-start when the playsink has already been configured before changing state. 2009-05-14 01:31:48 +0100 Tim-Philipp Müller * configure.ac: configure: require core CVS for gst_adapter_prev_timestamp() which is used in the libvisual plugin. 2009-04-22 18:34:59 +0200 Wim Taymans * AUTHORS: AUTHORS: fix my email 2009-04-22 18:35:40 +0200 Wim Taymans * gst-libs/gst/audio/gstaudioclock.c: audioclock: make our internal time monotonic Make the internal time increase monotonically. 2009-05-13 19:27:54 +0200 Wim Taymans * ext/libvisual/visual.c: visual: remove next_ts variable We can remove the next_ts variable as we don't use it anymore. 2009-05-13 19:24:15 +0200 Wim Taymans * ext/libvisual/visual.c: visual: use new adapter timestamp code Use the new adapter timestamp tracking code to make things easier and produce vastly better output timestamps. 2009-05-13 01:35:07 +0100 Tim-Philipp Müller * po/Makevars: * po/af.po: * po/az.po: * po/bg.po: * po/ca.po: * po/cs.po: * po/da.po: * po/de.po: * po/en_GB.po: * po/es.po: * po/fi.po: * po/fr.po: * po/hu.po: * po/id.po: * po/it.po: * po/ja.po: * po/lt.po: * po/nb.po: * po/nl.po: * po/or.po: * po/pl.po: * po/pt_BR.po: * po/ru.po: * po/sk.po: * po/sq.po: * po/sr.po: * po/sv.po: * po/uk.po: * po/vi.po: * po/zh_CN.po: po: avoid conflicts of local *.po files with files in git Make it so that filenames and line numbers are only stored in the *.pot file (which is not in git), but not in the individual *.po files. This information is hardly useful for translators in our case, and it should avoid the constant conflicts of local *.po files with the ones in git which are caused by the source files changing and the line numbers being updated. This commit might cause one last merge conflict for you, which you can work around with "git checkout po/*.po" before merging or pulling. After that there should (hopefully) not be any more local modifications of these files (unless someone committed additions or changes to translated strings and the *.po files haven't been updated yet, that is). 2009-05-12 23:51:08 +0100 Tim-Philipp Müller * tests/check/elements/.gitignore: * tests/check/elements/audioresample.c: tests: fix audioresample unit test on big endian architectures Don't hardcode endianness=1234 in the filtercaps, it will cause pad link failures which will result in the test timing out. 2009-05-12 17:18:37 +0100 Tim-Philipp Müller * gst/audiotestsrc/gstaudiotestsrc.c: audiotestsrc: fix broken enum nick - it should have a hyphen The enum nick should be 'sine-table', not 'sine table'. Technically this is an API/ABI change I guess, but anyone who was using this and didn't report it deserves this. 2009-05-01 01:04:48 +0100 Tim-Philipp Müller * gst/audiotestsrc/gstaudiotestsrc.c: audiotestsrc: seek to the requested byte offset, not the expected byte offset 2009-05-01 01:03:06 +0100 Tim-Philipp Müller * gst/audiotestsrc/gstaudiotestsrc.c: * gst/audiotestsrc/gstaudiotestsrc.h: audiotestsrc: support more than just one channel 2009-05-12 15:52:41 +0200 Sebastian Dröge * gst-libs/gst/interfaces/propertyprobe.h: propertyprobe: Fix typo in the docs 2009-05-12 12:17:55 +0100 Christian Schaller * ext/ogg/gstoggmux.c: * ext/theora/theora.c: * ext/vorbis/vorbis.c: Add ranks to the Oggmuxer, Vorbis encoder and Theora encoder 2009-04-30 16:37:38 +0200 Wim Taymans * gst/videorate/gstvideorate.c: * gst/videorate/gstvideorate.h: videorate: handle invalid timestamps better Handle buffers with -1 timestamps better by keeping track of the en time of the previous buffer and assuming the -1 timestamp buffer goes right after the previous one. when we have two buffers that are equally good, output the oldest buffer once to minimize latency. don't try to calculate latency when the input framerate is unknown. 2009-04-28 11:37:14 +0200 Wim Taymans * ext/ogg/gstoggmux.c: oggmux: small debug statement in DISCONT 2009-04-28 11:24:19 +0200 Wim Taymans * ext/ogg/gstoggdemux.c: * ext/ogg/gstoggdemux.h: oggdemux: fix abuse of ogg API, handle broken oggs When we feed the ogg sync layer, we need to feed it contiguous data even if the sync layer did not consume all of it yet. This makes sure that it always finds the next page even for more corrupted files. Use a different read_offset for this purpose. since we now keep track of the sync layer, we don't have to reset after finding a start of a page. Add some more debug info for the error paths. Only reset the sync layer when we perform a seek operation. Avoid failure when the next chain has no bos pages but instead simply ignore it. when we receive unknown page serial numbers mid stream, don't fail but post a warning and hope that we get back on track later. Fixes #579642 2009-04-30 16:41:51 +0200 Wim Taymans * gst/playback/gstdecodebin2.c: decodebin2: make subpictures a raw output format Subpictures are a raw format, we want those pads exposed so that playbin2 can do the subpicture mixing. 2009-04-27 10:15:44 +0200 Wim Taymans * gst-libs/gst/rtp/gstbasertppayload.c: * gst-libs/gst/rtp/gstbasertppayload.h: rtpdepay: add some more comments 2009-04-17 10:54:31 +0200 Wim Taymans * gst-libs/gst/audio/gstaudioclock.c: audioclock: make sure values are ever increasing 2009-05-05 17:17:00 +0200 Wim Taymans * gst/playback/gstplaysink.c: playbin2: make fallback identity silent Set the signal-handoffs to FALSE and silent to TRUE for the fallback identity element so that it consumes less CPU. 2009-04-17 10:57:16 +0200 Wim Taymans * gst/playback/gstplaybin2.c: * gst/playback/gstplaysink.c: playbin2: handle custom audiosinks differently Keep track of the autoplugged custom sinks and configure them in the playsink element when we have collected all streams. Also make sure that we only select one custom sink. When unreffing the internal sink, we don't need to change the state to NULL. 2009-05-12 10:36:25 +0200 Wim Taymans * gst/playback/gstplaybin2.c: * gst/playback/gstplaysink.c: * gst/playback/gstplaysink.h: playbin2: unify custom sink get/set functions Use one function to set/get all of the different sink types. cleanup up the subpicture chain too. Allow setting a custom subpicture sink. 2009-05-11 18:29:34 +0200 Sebastian Dröge * gst-libs/gst/interfaces/tunernorm.h: interfaces: Seperate some more struct definitions from typedefs 2009-05-11 15:48:56 +0200 Sebastian Dröge * gst-libs/gst/interfaces/navigation.h: * gst-libs/gst/interfaces/videoorientation.h: * gst-libs/gst/interfaces/xoverlay.h: interfaces: Seperate some more struct definitions from typedefs 2009-05-10 17:28:53 +0200 Sebastian Dröge * win32/common/libgstinterfaces.def: Add new functions to win32 exports 2009-05-10 17:28:05 +0200 Sebastian Dröge * docs/libs/gst-plugins-base-libs-sections.txt: Add new functions to the docs 2009-05-10 17:25:58 +0200 Sebastian Dröge * gst-libs/gst/interfaces/mixer.c: * gst-libs/gst/interfaces/mixer.h: interfaces: API: Add gst_mixer_get_mixer_type() This is a convenience function that returns the mixer_type of the interface struct. 2009-05-10 17:25:31 +0200 Sebastian Dröge * gst-libs/gst/interfaces/colorbalance.c: interfaces: Add docs for gst_color_balance_get_balance_type() 2009-05-10 11:17:19 +0200 Marc-Andre Lureau * autogen.sh: Run libtoolize before aclocal This unbreaks the build in some cases. Fixes bug #582021 2009-05-07 17:38:40 +0200 Sebastian Dröge * ext/pango/gsttextrender.c: textrender: Correctly initialize the background for ARGB too 2009-05-07 16:59:32 +0200 Sebastian Dröge * ext/pango/gsttextrender.c: * ext/pango/gsttextrender.h: textrender: Use libgstvideo functions to create caps Also check if downstream wants ARGB always when we get new caps. 2009-05-07 16:52:02 +0200 Sebastian Dröge * ext/pango/gsttextrender.c: textrender: Don't always use ARGB if downstream supports it but take it's preference 2009-05-07 16:48:08 +0200 Kapil Agrawal * ext/pango/gsttextrender.c: * ext/pango/gsttextrender.h: textrender: Add support for ARGB and alignment properties Fixes bug #581571. 2009-05-07 16:42:20 +0200 Sebastian Dröge * ext/pango/gsttextrender.c: textrender: Add ; after GST_BOILERPLATE to fix indention 2009-05-07 15:10:30 +0200 Sebastian Dröge * gst-libs/gst/tag/gstvorbistag.c: vorbistag: Use text/uri-list as mimetype instead of ---> for URI lists 2009-05-07 14:59:36 +0200 Arnout Vandecappelle * gst/typefind/gsttypefindfunctions.c: typefindfunctions: made mp3_type_find less aggressive mp3_type_find could suggest already when only a single valid header was found, if it ran out of data before the end of the next frame. Therefore, ignore the last found frame if it was incomplete. Fixes bug #579692. 2009-05-07 14:48:29 +0200 John Millikin * gst-libs/gst/tag/gstvorbistag.c: vorbistag: Store cover art in vorbiscomments Fixes bug #513373. 2009-05-07 06:14:18 +0200 Sebastian Dröge * gst-libs/gst/interfaces/colorbalance.c: * gst-libs/gst/interfaces/colorbalance.h: interfaces: API: Add gst_color_balance_get_balance_type() This is a convenience function that returns the balance_type of the interface struct. 2009-05-06 17:59:13 +0200 Sebastian Dröge * gst-libs/gst/interfaces/colorbalance.h: * gst-libs/gst/interfaces/colorbalancechannel.h: * gst-libs/gst/interfaces/tuner.h: * gst-libs/gst/interfaces/tunerchannel.h: interfaces: Separate struct definitions from typedefs 2009-05-06 14:03:01 +0200 Sebastian Dröge * pkgconfig/gstreamer-app-uninstalled.pc.in: Fix libdir for uninstalled gstreamer-app library 2009-05-12 01:59:01 +0100 Tim-Philipp Müller * gst-libs/gst/pbutils/descriptions.c: pbutils: add description for APE tag caps 2009-05-12 01:35:27 +0100 Tim-Philipp Müller * configure.ac: configure: bump core requirement to last release as that's more likely to be true than that we need only 0.21.1. 2009-05-12 01:21:57 +0100 Tim-Philipp Müller * common: * configure.ac: configure: rename CVS -> git in a couple of places 2009-05-12 01:17:53 +0100 Tim-Philipp Müller * configure.ac: configure: bump GLib requirement to GLib >= 2.16 as per the New Regime (see wiki). 2009-05-01 00:09:15 +0100 Tim-Philipp Müller * gst-libs/gst/tag/gsttagdemux.c: tagdemux: cache events from upstream and re-send them once we have a source pad Makes sure tags don't get dropped when we have multiple tag demuxers in a row. Fixes #580318. 2009-05-07 14:07:44 -0700 Michael Smith * gst-libs/gst/riff/riff-media.c: riff: support UYVY raw 4:2:2 in riff. 2009-05-11 21:20:07 +0100 Jan Schmidt * configure.ac: Back to development -> 0.10.23.1 2009-04-27 22:42:55 -0700 Michael Smith * ext/theora/theoradec.c: theoradec: fix buffer overrun on 422 decode. 2009-04-27 21:39:01 -0700 Michael Smith * ext/theora/theoradec.c: theoradec: 444 support. 2009-04-27 21:30:04 -0700 Michael Smith * ext/theora/theoradec.c: theoradec: handle 422 images (as YUY2). 2009-04-27 21:01:51 -0700 Michael Smith * ext/theora/gsttheoradec.h: * ext/theora/theoradec.c: theoradec: rearrange code in preparation for 422 and 444 support.