=== release 0.10.18 === 2010-02-10 Tim-Philipp Müller * configure.ac: releasing 0.10.18, "Short Circuit" 2010-02-10 20:36:56 +0000 Robert Swain * gst/qtdemux/qtdemux.c: qtdemux: temporary safety check to avoid crashes with a certain file Add temporary check to avoid crashes with a certain file when seeking until the real cause of this is figured out. See #609405. 2010-02-05 18:05:39 +0100 Robert Swain * gst/qtdemux/qtdemux.c: * gst/qtdemux/qtdemux.h: qtdemux: skip unknown atoms when looking for moov Fixes bug #609107 2010-02-05 02:13:33 +0000 Tim-Philipp Müller * configure.ac: * win32/common/config.h: 0.10.17.3 pre-release 2010-02-04 19:10:36 +0000 Tim-Philipp Müller * po/bg.po: * po/hu.po: po: update translations 2010-02-04 14:46:56 +0100 Robert Swain * gst/qtdemux/qtdemux.c: * gst/qtdemux/qtdemux.h: qtdemux: Set the segment start time to the requested seek time for non-keyframe seeks 2010-02-04 12:00:03 +0100 Robert Swain * gst/qtdemux/qtdemux.c: qtdemux: Fix time returned for index at a byte offset The logic for searching forwards/backwards was swapped 2010-02-01 19:22:24 +0100 Mark Nauwelaerts * ext/speex/gstspeexdec.c: speexdec: initialize stereo decoding state 2010-01-28 18:58:08 +0100 Mark Nauwelaerts * gst/matroska/matroska-demux.c: matroskademux: improve stream synchronization In particular, do not make it send newsegment updates that sort-of contradict the indented playback segment (e.g. start time). 2010-01-28 18:53:18 +0100 Mark Nauwelaerts * gst/matroska/matroska-demux.c: matroskademux: fix bridging (time) gaps in streams As a side effect, avoid sending newsegment updates with start times that go back and forth, which leads to bogus downstream running_time. Also fixes seeking in bug #606744. 2010-01-28 18:49:57 +0100 Mark Nauwelaerts * gst/matroska/matroska-demux.c: matroskademux: fix stream synchronization .. by initializing streams starting at 0, as that is basically where we 'seek to' at the start and assume streams to start elsewhere. Also enables newsegment update events for subtitle streams. 2010-02-02 13:41:03 +0200 Stefan Kost * ext/jpeg/gstjpegdec.c: jpeg: don't directly access message, some message have args This caused bogus messages, such as reported in bug #607471. 2010-02-02 00:02:34 +0000 David Hoyt * ext/libpng/gstpngdec.c: png: fix compilation with libpng 1.4 png_set_gray_1_2_4_to_8() has been deprecated for a while and was finally removed in libpng 1.4.x. Use png_set_expand_gray_1_2_4_to_8() instead. Fixes #608629. 2010-02-01 16:46:36 +0100 Wim Taymans * gst/rtsp/gstrtspsrc.c: rtspsrc: free transports on errors See #608564 2010-02-01 09:18:53 +0000 Tim-Philipp Müller * sys/v4l2/v4l2_calls.c: v4l2: fix unportable printf format 2010-01-30 15:18:48 +0000 Tim-Philipp Müller * common: Automatic update of common submodule From 15d47a6 to 96dc793 2010-01-27 17:53:07 +0100 Robert Swain * gst/flv/gstflvmux.c: flvmux: index timestamps should be in seconds, not milliseconds 2010-01-27 15:24:52 +0100 Mark Nauwelaerts * ext/speex/gstspeexdec.c: speexdec: free some more when resetting Fixes #608255. 2010-01-27 15:24:24 +0100 Mark Nauwelaerts * gst/rtp/gstrtpspeexpay.c: rtpspeexpay: fix occasional buffer leak Fixes #608255. 2010-01-27 15:22:46 +0100 Mark Nauwelaerts * ext/speex/gstspeexenc.c: speexenc: prevent invalid arithmetic if not setup yet Fixes #608255. 2010-01-27 16:34:21 +0100 Sebastian Dröge * gst/videomixer/blend_mmx.h: videomixer: Fix assembly register constraints Fixes bug #608209. 2010-01-27 01:56:03 +0000 Tim-Philipp Müller * configure.ac: * win32/common/config.h: 0.10.17.2 pre-release 2010-01-27 01:52:59 +0000 Tim-Philipp Müller * po/LINGUAS: * po/af.po: * po/az.po: * po/bg.po: * po/ca.po: * po/cs.po: * po/da.po: * po/de.po: * po/el.po: * po/en_GB.po: * po/es.po: * po/eu.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/mt.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: * po/zh_HK.po: * po/zh_TW.po: po: update translations 2010-01-27 01:49:49 +0000 Tim-Philipp Müller * tests/check/elements/.gitignore: checks: ignore deinterlace check binary 2010-01-27 01:18:51 +0000 Tim-Philipp Müller * configure.ac: configure: purge all mention of CVS 2010-01-26 11:18:28 +0100 Wim Taymans * gst/avi/gstavidemux.c: avidemux: ignore streams that finished When we receive an UNEXPECTED from a stream, move to the next stream and only go EOS when all streams are EOS. When selecting a stream to push, ignore streams that went EOS. Fixes #607949 2010-01-25 17:23:43 +0200 Stefan Kost * sys/v4l2/v4l2src_calls.c: v4l2src: don't deref NULL Error out when the pool gets shutdown. 2010-01-25 17:21:13 +0200 Stefan Kost * ext/jpeg/gstjpegenc.c: * sys/v4l2/v4l2src_calls.c: * tests/check/Makefile.am: Revert "v4l2src: don't deref NULL" This reverts commit 3d9d34bd60faeb940b36d992a47168fc895036ba. 2010-01-25 14:16:22 +0200 Stefan Kost * ext/jpeg/gstjpegenc.c: * sys/v4l2/v4l2src_calls.c: * tests/check/Makefile.am: v4l2src: don't deref NULL Error out when the pool gets shutdown. 2010-01-23 15:32:48 -0800 Michael Smith * ext/jpeg/gstjpegenc.c: jpegenc: when creating an overflow buffer, copy timestamps. 2010-01-23 14:47:55 +0100 Edward Hervey * gst/qtdemux/qtdemux.c: qtdemux: dmb1 is a valid fourcc for Motion-JPEG 2010-01-23 14:20:02 +0100 Edward Hervey * gst/qtdemux/qtdemux.c: qtdeux: IV32 is also used for Indeo 3 video streams 2010-01-22 16:48:01 +0200 Stefan Kost * tests/icles/ximagesrc-test.c: build: no unused variables when disabling asserts 2010-01-21 23:17:40 -0300 Roland Krikava * gst/qtdemux/qtdemux.c: qtdemux: Avoid negative overflow on keyframe search Do not overflow negatively when searching a previous "keyframe" on audio streams. Could cause infinite loops on backwards playback Fixes #607718 2010-01-21 17:22:38 -0800 Peter van Hardenberg * ext/jpeg/gstjpegenc.c: * ext/jpeg/gstjpegenc.h: jpegenc: enlarge buffer if libjpeg tells us it's out of space. Fixes buffer overflow on some high-quality, low-resolution jpeg encodes. 2010-01-21 19:24:22 +0100 Alessandro Decina * gst/qtdemux/qtdemux.c: qtdemux: fix compiler warnings under OS X. 2010-01-21 17:57:36 +0100 Wim Taymans * gst/avi/gstavidemux.c: avidemux: don't parse NULL indexes for some streams we might fail to fetch the index offsets. Don't try to parse NULL indexes in those cases. 2010-01-18 21:15:51 -0500 Olivier Crête * gst/rtp/gstrtpg729pay.c: rtpg729pay: ptime should is in nanoseconds https://bugzilla.gnome.org/show_bug.cgi?id=607403 2010-01-20 15:11:15 -0300 Thiago Santos * gst/wavenc/gstwavenc.c: * gst/wavenc/gstwavenc.h: wavenc: Post warning if file isnt finished properly When the pipeline is shut down and the file isn't finished properly, wavenc should post a warning. Fixes #607440 2009-05-27 13:51:44 +0200 Arnout Vandecappelle * gst/matroska/matroska-mux.c: * gst/matroska/matroska-mux.h: matroskamux: make index size configurable. Added the 'min-index-interval' property to matroskamux, which determines how much time (nanoseconds) is left between keyframes stored in the index. Fixes #583985. 2010-01-20 16:28:31 +0100 Wim Taymans * gst/rtp/gstrtph264pay.c: rtph264pay: scale spspps_interval to milliseconds The spspps_interval is kept in seconds. Convert it to milliseconds before comparing it to another value in milliseconds. 2010-01-20 15:18:47 +0100 Mark Nauwelaerts * gst/qtdemux/qtdemux.c: qtdemux: always keep media segments within total duration ... as opposed to only doing so following a seek. 2010-01-20 15:44:40 +0100 Wim Taymans * gst/rtp/gstrtph264pay.c: rtph264pay: rename spspps-interval property Rename the spspps-interval property to config-interval because it is nicer. 2010-01-19 18:37:31 +0100 Wim Taymans * gst/avi/gstavidemux.c: avidemux: skip RIFF and index in push mode When we are in push mode, we can encounter RIFF and idx tags in the data chunk when we are dealing with ODML files. In these cases, simply skip the chunks and continue streaming instead of going EOS. 2010-01-20 11:27:23 +0100 Wim Taymans * gst/avi/gstavidemux.c: avidemux: more DISCONT handling Add some debug in the DISCONT handling code. When we receive a DISCONT in push mode, mark all streams as DISCONT. 2010-01-20 11:26:34 +0100 Wim Taymans * gst/avi/gstavidemux.c: avidemux: reset on flush events When we receive a flush event on the sinkpad, reset the EOS state and the flowreturn of all streams. Also mark the streams with a DISCONT. 2010-01-20 11:22:04 +0100 Wim Taymans * gst/avi/gstavidemux.c: * gst/avi/gstavidemux.h: avidemux: rename some variable Rename the seek_event variable to seg_event because it really contains the newsegment event that needs to be pushed. 2010-01-20 00:54:03 +0000 Tim-Philipp Müller * common: Automatic update of common submodule From 14cec89 to 15d47a6 2010-01-18 14:49:26 -0500 Olivier Crête * gst/rtp/gstrtph264pay.c: * gst/rtp/gstrtph264pay.h: rtph264pay: Don't set profile-level-id in out caps The profile-level-id represents restrictions on what can be sent, it does not describe the stream. So it should be reflected in the sink caps of the payloader, not the src caps. https://bugzilla.gnome.org/show_bug.cgi?id=607353 2010-01-18 14:41:10 -0500 Olivier Crête * gst/rtp/gstrtph264pay.c: rtph264pay: Don't ignore the return value from set_outcaps https://bugzilla.gnome.org/show_bug.cgi?id=607353 2010-01-18 17:43:41 +0100 Sebastian Dröge * gst/deinterlace/tvtime/greedyhmacros.h: * gst/deinterlace/tvtime/linear.c: * gst/deinterlace/tvtime/linearblend.c: * gst/deinterlace/tvtime/tomsmocomp.c: * gst/deinterlace/tvtime/weave.c: * gst/deinterlace/tvtime/weavebff.c: * gst/deinterlace/tvtime/weavetff.c: deinterlace: Fix license and copyright headers 2010-01-18 14:57:42 +0200 Stefan Kost * sys/v4l2/gstv4l2bufferpool.h: v4l2: move G_END_DECLS to the end 2010-01-18 14:55:38 +0200 Stefan Kost * sys/v4l2/gstv4l2bufferpool.c: * sys/v4l2/gstv4l2bufferpool.h: v4l2: fix bufferpool file names in header comment 2010-01-15 18:15:14 +0100 Wim Taymans * gst/avi/gstavidemux.c: avidemux: avoid some typecasting 2010-01-15 18:13:24 +0100 Wim Taymans * gst/avi/gstavidemux.c: avidemux: avoid some type checks 2010-01-15 18:09:15 +0100 Wim Taymans * gst/avi/gstavidemux.c: * gst/avi/gstavidemux.h: avidemux: fallback to avih duration when we have not yet parsed the indexes (in push mode, for example) use the duration as given in the avih header instead of -1. 2010-01-15 13:32:32 -0300 Thiago Santos * gst/qtdemux/qtdemux.c: qtdemux: g_free is NULL safe 2010-01-15 13:27:40 -0300 Thiago Santos * gst/qtdemux/qtdemux.c: qtdemux: use DEMUX errors, instead of DECODE qtdemux should use DEMUX errors, and not DECODE Conflicts: gst/qtdemux/qtdemux.c 2010-01-14 19:16:19 -0300 Thiago Santos * gst/qtdemux/qtdemux.c: qtdemux: Minor refactor Replace repeated code with a function call 2010-01-14 17:11:13 -0300 Thiago Santos * gst/qtdemux/qtdemux.c: * gst/qtdemux/qtdemux_fourcc.h: qtdemux: Handle another kind of redirect trak Some traks might contain a redirect rtsp uri inside hndl atom (which is a dref atom entry). This commit makes qtdemux post a message when it finds one of these traks and there are no other traks. Fixes #597497 2010-01-14 16:13:08 -0300 Thiago Santos * gst/qtdemux/qtdemux.c: * gst/qtdemux/qtdemux.h: qtdemux: Post error when reaching EOS without pads Post an error when EOS is reached and there are no src pads 2010-01-14 14:13:50 -0300 Thiago Santos * gst/qtdemux/qtdemux.c: qtdemux: Do not post empty redirect messages Some misinterpreted data could result in posting redirect messages with empty redirect strings. It is better not to post them. An example is the file on bug #597497 2010-01-14 18:19:25 +0100 Mark Nauwelaerts * gst/matroska/matroska-demux.c: matroskademux: polish last buffer end time usage That is, reset it upon seek, and note that (rarely) last pushed buffer time might precede segment start. 2010-01-13 16:48:46 +0200 Stefan Kost * gst/videomixer/blend_mmx.h: videomixer: use 'q' constraint instead of 'r' This avoids the "bad register name `%dil'" compilation errors on 32bit where because of 'r' gcc puts the value in a general purpose register and then tries to access the lower part as %dil/%sil which is not existing on 32bit. 'q' requests a-d registers 2010-01-13 16:44:58 +0200 Stefan Kost * gst/avi/gstavidemux.c: avi: add missing include for sscanf 2010-01-13 09:36:03 +0100 Sebastian Dröge * gst/equalizer/gstiirequalizer10bands.c: equalizer: Fix property description for the 3rd band of the 10band equalizer The frequency is actually 237 Hz, not 227 Hz. Fixes bug #606692. 2010-01-13 09:22:20 +0100 Kipp Cannon * gst/audiofx/audioamplify.c: audioamplify: Allow negative amplifications Fixes bug #606807. 2010-01-13 09:17:05 +0100 Sebastian Dröge * ext/taglib/gstapev2mux.cc: apev2mux: Don't call constructors directly, this leads to compiler errors with gcc 4.5 2010-01-12 17:39:05 +0100 Edward Hervey * gst/qtdemux/qtdemux.c: qtdemux: use G_GSIZE_FORMAT for platform independent gsize qualifier Fixes build on macosx 2010-01-11 19:02:34 +0100 Mark Nauwelaerts * gst/matroska/matroska-demux.c: matroskademux: refactor eos sending when pausing loop Also, prevent hanging if no pads yet on which to send eos by posting a message instead. 2010-01-11 17:50:35 +0100 Mark Nauwelaerts * gst/matroska/matroska-demux.c: matroskademux: standardize seek handling ... which implies fixing some corner cases. 2010-01-11 15:14:06 +0100 Mark Nauwelaerts * gst/matroska/matroska-mux.c: matroskamux: use more generic xiphN_streamheader_to_codecdata helper 2010-01-11 17:50:04 +0100 Mark Nauwelaerts * gst/matroska/matroska-mux.c: matroskamux: reflow audio and video setcaps and improve logging Also ensure width and height are available as they are mandatory in matroska specs. 2010-01-11 11:42:43 -0800 Michael Smith * gst/qtdemux/qtdemux.c: qtdemux: fix offset for type 2 mp4a sound sample descriptions. Allows us to correctly find the esds (and thus the codec data) for such mp4a files. 2010-01-11 15:45:49 -0300 Thiago Santos * gst/rtp/gstrtpmp4gdepay.c: * gst/rtp/gstrtpmp4gpay.c: rtpmp4g(de)pay: Only handle raw aac rtpmp4g(de)pay should only handle raw AAC streams 2010-01-11 18:59:43 +0100 Sebastian Dröge * gst/videomixer/videomixer.c: * gst/videomixer/videomixer.h: videomixer: Implement basic QoS This drops frames if they're too late anyway before blending and all that starts but QoS events are not forwarded upstream. In the future the QoS events should be transformed somehow and forwarded upstream. 2010-01-11 14:48:26 -0300 Thiago Santos * gst/rtp/gstrtpmp4adepay.c: * gst/rtp/gstrtpmp4apay.c: rtpmp4a(de)pay: Only accept raw aac rtpmp4a(de)pay should only handle raw aac to conform to the RFC 2010-01-11 18:35:47 +0100 Sebastian Dröge * gst/videomixer/blend.c: * gst/videomixer/blend_mmx.h: videomixer: Add MMX implementations for I420 and all non-alpha RGB formats 2010-01-04 10:24:45 +0100 Sebastian Dröge * gst/videomixer/Makefile.am: * gst/videomixer/blend.c: * gst/videomixer/blend.h: * gst/videomixer/blend_ayuv.c: * gst/videomixer/blend_bgra.c: * gst/videomixer/blend_i420.c: * gst/videomixer/blend_mmx.h: * gst/videomixer/blend_rgb.c: * gst/videomixer/videomixer.c: * gst/videomixer/videomixer.h: videomixer: Refactor processing functions This allows easier plugging of optimized processing functions in the future, like for SSE or AltiVec. 2010-01-11 13:26:32 -0300 Thiago Santos * gst/avi/gstavimux.c: * gst/matroska/matroska-mux.c: avimux: matroskamux: rename aac's stream-format to raw AAC's none stream-format has been renamed to raw, rename on avimux and matroskamux as well 2010-01-11 12:07:29 -0300 Thiago Santos * gst/matroska/matroska-mux.c: matroskamux: Only accept raw aac makes matroskamux reject aac streams that are not in raw format (stream-format=none) Fixes #598350 2010-01-11 12:08:55 -0300 Thiago Santos * gst/avi/gstavimux.c: avimux: Only accept raw aac makes avimux reject aac streams that are not in raw format (stream-format=none) Fixes #598350 2010-01-11 10:38:10 +0100 Robert Swain * gst/qtdemux/qtdemux.c: qtdemux: Oops. The gpointer cast is needed because of the const qualifiers on the data elements 2010-01-11 10:17:54 +0100 Robert Swain * gst/qtdemux/qtdemux.c: qtdemux: Debug -> info level for a message for benchmarking index parsing The extra message output at higher levels affects the accuracy of the benchmark. 2010-01-11 10:05:10 +0100 Robert Swain * gst/qtdemux/qtdemux.c: qtdemux: Don't check for NULL pointers or cast to gpointer as this is not needed 2010-01-08 13:55:05 +0100 Robert Swain * gst/qtdemux/qtdemux.c: qtdemux: Refactor stbl sub-atom freeing. Free when index has been completely parsed. 2010-01-08 14:32:06 +0100 Robert Swain * gst/qtdemux/qtdemux.c: qtdemux: Avoid whitespace commits due to inconsistent GNU indent behaviour 2010-01-11 00:10:34 +0000 Tim-Philipp Müller * gst/qtdemux/qtdemux.c: qtdemux: remove newline at end of debug statement 2010-01-08 19:26:21 +0100 Havard Graff * gst/udp/gstmultiudpsink.c: multiudpsink: Compiler warning fixes for Windows Just simple missing casts Fixes bug #606438. 2010-01-08 18:04:14 +0100 Mark Nauwelaerts * ext/flac/gstflacenc.c: flacenc: fix seekpoints property copy-and-paste documentation 2010-01-06 17:06:53 +0100 Mark Nauwelaerts * ext/flac/gstflacenc.c: * ext/flac/gstflacenc.h: flacenc: optionally add a seek table API: GstFlacEnc:seekpoints Fixes #351595. 2010-01-08 11:33:02 -0300 Thiago Santos * gst/avi/gstavidemux.c: avidemux: Use more glib and be safer Be safer on sscanf by limiting string format sizes. Remove useless parameter and use g_strndup. 2010-01-08 10:44:44 -0300 Thiago Santos * gst/avi/gstavidemux.c: avidemux: Simplifying code Greatly simplify the IDIT chunk handling by using sscanf instead of 'manually' parsing. Also replaces strncasecmp and is_alpha/is_digit with glib versions. 2010-01-08 10:18:30 -0300 Thiago Santos * gst/avi/gstavidemux.c: avidemux: it's feb for february Fix typo in last commit. 2010-01-08 09:17:22 -0300 Thiago Santos * gst/avi/gstavidemux.c: avidemux: Parse and post IDIT dates Parses and post date tags contained in IDIT chunks. Fixes #503582 2010-01-07 17:25:05 +0100 Sebastian Dröge * gst/audiofx/audiofirfilter.c: * gst/audiofx/audiofxbasefirfilter.c: * gst/audiofx/audiofxbasefirfilter.h: audiofxbasefirfilter: Add property for not draining the history on kernel changes Currently this only works if the kernel size doesn't change, in the future it will be possible to change the kernel size too without draining the complete history and without loosing anything. Partially based on a patch by Thiago Santos 2010-01-07 16:58:55 +0100 Wim Taymans * gst/rtp/gstrtph264pay.c: rtph264pay: remove weird memcmp code Use plain memcmp for comparing memory instead of the custom buggy one. Fixes #606198 2010-01-07 15:38:36 +0100 Edward Hervey * gst/level/gstlevel.c: level: fix typo in 'message' property description 2010-01-06 14:06:14 +0100 Mark Nauwelaerts * ext/flac/gstflacdec.c: flacdec: really use upstream timestamp if there is one See/fixes #603471. 2010-01-06 13:45:59 +0100 Wim Taymans * gst/rtp/gstrtpg729pay.c: rtpg728pay: remove unused adapter peek 2010-01-05 19:00:35 -0300 Thiago Santos * tests/check/elements/deinterlace.c: deinterlace: Improve passthrough tests Improve passthrough tests by forcing more specific interlaced/deinterlaced caps to be tested 2010-01-05 18:22:49 -0300 Thiago Santos * tests/check/elements/deinterlace.c: deinterlace: Adds some docs to the new tests Adds some docs explaining the utility functions of the check tests of deinterlace 2010-01-05 18:14:08 -0300 Thiago Santos * tests/check/elements/deinterlace.c: deinterlace: Adds tests for passthrough Adds tests for checking if the element really does passthrough in disabled mode and in auto (if the input is not interlaced) 2010-01-05 07:50:51 -0300 Thiago Santos * tests/check/Makefile.am: * tests/check/elements/deinterlace.c: deinterlace: Adds tests for caps acceptance Adds check unit tests for deinterlace for validating caps accepting and the expected caps output on the other pad 2010-01-04 13:43:00 -0300 Thiago Santos * tests/check/Makefile.am: * tests/check/elements/deinterlace.c: deinterlace: Adds basic check test Adds a basic check test for deinterlace element 2010-01-04 15:44:28 -0800 Michael Smith * gst/qtdemux/Makefile.am: * gst/qtdemux/qtdemux.c: qtdemux: Add support for wave-style audio in qt. Uses gstriff to parse the wave headers appropriately. Tested with MS-ADPCM content. 2009-12-31 17:09:03 -0500 Olivier Crête * tests/check/elements/rtp-payloading.c: tests: Add G.729 RTP payloader/depayloader test https://bugzilla.gnome.org/show_bug.cgi?id=606050 2009-12-31 16:52:30 -0500 Olivier Crête * gst/rtp/gstrtpg729pay.c: rtpg729pay: Simplify adapter usage https://bugzilla.gnome.org/show_bug.cgi?id=606050 2009-12-31 16:27:30 -0500 Olivier Crête * gst/rtp/gstrtpg729pay.c: rtpg729pay: Support ptime from caps https://bugzilla.gnome.org/show_bug.cgi?id=606050 2009-12-02 19:35:21 +0530 Olivier Crête * gst/rtp/README: rtp: Add maxptime to the README https://bugzilla.gnome.org/show_bug.cgi?id=606050 2010-01-05 19:03:06 +0100 Wim Taymans * gst/rtp/Makefile.am: * gst/rtp/gstrtp.c: * gst/rtp/gstrtpg723depay.c: * gst/rtp/gstrtpg723depay.h: rtpg723depay: add G723 depayloader 2010-01-05 19:02:39 +0100 Wim Taymans * gst/rtp/gstrtpg729depay.c: * gst/rtp/gstrtpg729depay.h: rtpg729depay: remove unused variable 2010-01-05 18:33:25 +0100 Wim Taymans * gst/rtp/gstrtpg723pay.c: * gst/rtp/gstrtpg723pay.h: rtpg723pay: rewrite payloader Handle all 3 packet sizes according to RFC 3551. Totally untested, we don't have a G723 encoder. Fixes #605882 2010-01-05 11:47:20 +0100 Wim Taymans * gst/qtdemux/qtdemux.c: qtdemux: fix chunk counter 2010-01-04 19:44:53 +0100 Wim Taymans * gst/qtdemux/qtdemux.c: qtdemux: more work at reducing loop overhead Try to avoid derefs when parsing the index. Save the state into the structures when we exit the loop instead of for each iteration. 2010-01-04 16:33:30 +0100 Wim Taymans * gst/qtdemux/qtdemux.c: qtdemux: cleanups and make duration more accurate Make the QtDemuxSample struct smaller by keeping the duration and the pts_offset as their 32 bit values. Make some macros to calculate PTS, DTS and duration of a sample. Deref the sample index less often by keeping a ref to the sample we're dealing with. 2010-01-04 13:41:18 +0100 Wim Taymans * gst/qtdemux/qtdemux.c: qtdemux: simplify logic to calculate duration Since we no longer store the timestamp and duration in nanoseconds, we can now simply store the duration as-is. 2010-01-01 16:42:57 +0100 Robert Swain * gst/qtdemux/qtdemux.c: qtdemux: Store timestamps in mov format in the index This allows faster building of the index upon seeks so that scaling of timestamps only occurs when actually needed. 2009-12-18 13:54:46 +0100 Wim Taymans * gst/qtdemux/qtdemux.c: qtdemux: make seeking in push mode work Move sample position checks into qtdemux_parse_samples where we can protect it with a lock. Refactor and make an qtdemux_ensure_index function. Rename qtdemux_do_push_seek to qtdemux_seek_offset in order to avoid confusion with gst_qtdemux_do_push_seek. 2009-12-18 12:44:27 +0100 Wim Taymans * gst/qtdemux/qtdemux.c: qtdemux: move error code out of normal flow 2009-11-24 16:27:26 +0100 Robert Swain * gst/qtdemux/qtdemux.c: * gst/qtdemux/qtdemux.h: qtdemux: Add push mode seek support for seeking to obtain the moov atom 2010-01-05 12:22:09 +0100 Wim Taymans * gst/rtsp/gstrtspsrc.c: rtspsrc: fix on-npt-stop signal warnings for RDT The RDT manager does not implement this signal so we need to check for it before trying to connect to it. 2010-01-05 09:47:00 +0000 Tim-Philipp Müller * sys/v4l2/gstv4l2src.c: v4l2src: fix memory leak in new uri handler code Don't leak a string everytime get_uri() is called and a device has been set. There's a limited number of devices, so just intern the string instead of doing more elaborate housekeeping and storing it in the instance struct or so. 2010-01-01 14:10:49 +0200 Stefan Kost * gst/avi/gstavimux.c: avimux: fix typo in warning message 2010-01-04 09:28:36 -0300 Robert Weidlich * ext/shout2/gstshout2.c: * ext/shout2/gstshout2.h: shout2send: Add 'public' property Adds a property to set 'public' flag on libshout, making the stream listed on the server's stream directory. Fixes #605269 2009-12-30 14:14:55 +0530 Arun Raghavan * gst/qtdemux/qtdemux.c: qtdemux: Add tags for average and maximum bitrate Fixes #599300. 2009-12-26 16:59:14 -0300 Thiago Santos * gst/audiofx/audiofxbasefirfilter.c: audiofxbasefirfilter: do not try to alloc really large buffers When nsamples_out is larger than nsamples_in, using unsigned ints lead to a overflow and the resulting value is wrong and way too large for allocating a buffer. Use signed integers and returning immediatelly when that happens. 2009-12-25 12:38:35 +0100 Wim Taymans * gst/videomixer/blend_ayuv.c: videomixer: optimize blend code some more Use more efficient formula that uses less multiplies. Reduce the amount of scalar code, use MMX to calculate the desired alpha value. Unroll and handle 2 pixels in one iteration for improved pairing. 2009-12-24 22:59:09 +0100 Wim Taymans * gst/videomixer/blend_ayuv.c: * gst/videomixer/blend_bgra.c: * gst/videomixer/blend_i420.c: * gst/videomixer/blend_rgb.c: videomixer: scale and clamp Scale and clamp to the max alpha values. 2009-12-24 22:50:31 +0100 Wim Taymans * gst/alpha/gstalpha.c: alpha: scale and clamp alpha to its full extend Convert the alpha value to 0->255 when setting and to 0->256 when using as a scaling factor. This makes sure we can reach the full opacity value of 0xff in all cases. 2009-12-24 22:23:01 +0100 Wim Taymans * gst/rtsp/gstrtspsrc.c: rtspsrc: fix some comments, remove property check Fix some comments, clarify some FIXMEs Remove the on-ntp-stop signal check now that the jitterbuffer is in -good and we know that it supports this signal. 2009-12-24 20:27:57 +0100 Wim Taymans * gst/videomixer/videomixer.c: videomixer: some trivial cleanups 2009-12-24 17:04:28 -0300 Thiago Santos * gst/rtsp/gstrtspsrc.c: rtspsrc: Parse all rtpinfo entries Do not forget to parse all rtp-info entries, instead of parsing the first one only. Fixes #605222 2009-12-22 12:44:50 -0300 Thiago Santos * gst/qtdemux/qtdemux.c: qtdemux: perf tag should map to GST_TAG_ARTIST 2009-12-24 17:03:02 +0100 Wim Taymans * gst/interleave/interleave.c: interleave: fix weird indentation 2009-12-24 17:01:54 +0100 Wim Taymans * gst/rtp/gstrtph263ppay.c: rtph263ppay: use faster _adapter_copy() whem possible 2009-12-24 17:01:15 +0100 Wim Taymans * tests/examples/audiofx/firfilter-example.c: tests: use right type when passing vararg value 2009-12-23 17:50:34 +0100 Mark Nauwelaerts * ext/flac/gstflacdec.c: * ext/flac/gstflacdec.h: flacdec: use a single decoder field for both push and pull mode 2009-12-23 17:03:32 +0100 Mark Nauwelaerts * ext/flac/gstflacdec.c: flacdec: fix possible hanging in pull mode seeking A seek in multi-sink pipeline typically leads to several seek events in a row, which could lead to sending several newsegments in a row without intermediate flushing. These would then accumulate, distort rendering times and as such lead to 'hanging'. 2009-12-23 19:39:05 +0100 Mark Nauwelaerts * gst/rtp/gstrtph264pay.c: rtph264pay: fix uninitialized variable 2009-12-23 13:09:54 +0100 Wim Taymans * gst/rtp/gstasteriskh263.c: * gst/rtp/gstrtpL16depay.c: * gst/rtp/gstrtpac3depay.c: * gst/rtp/gstrtpamrdepay.c: * gst/rtp/gstrtpamrpay.c: * gst/rtp/gstrtpbvpay.c: * gst/rtp/gstrtpdepay.c: * gst/rtp/gstrtpg729depay.c: * gst/rtp/gstrtpgsmdepay.c: * gst/rtp/gstrtpgsmpay.c: * gst/rtp/gstrtph263depay.c: * gst/rtp/gstrtph263pay.c: * gst/rtp/gstrtph263pdepay.c: * gst/rtp/gstrtph263ppay.c: * gst/rtp/gstrtpilbcpay.c: * gst/rtp/gstrtpjpegdepay.c: * gst/rtp/gstrtpmp1sdepay.c: * gst/rtp/gstrtpmp2tdepay.c: * gst/rtp/gstrtpmp4apay.c: * gst/rtp/gstrtpmp4gdepay.c: * gst/rtp/gstrtpmp4gpay.c: * gst/rtp/gstrtpmp4vpay.c: * gst/rtp/gstrtpmpadepay.c: * gst/rtp/gstrtpmpapay.c: * gst/rtp/gstrtpmpvdepay.c: * gst/rtp/gstrtppcmadepay.c: * gst/rtp/gstrtppcmudepay.c: * gst/rtp/gstrtppcmupay.c: * gst/rtp/gstrtpqdmdepay.c: * gst/rtp/gstrtpsirenpay.c: * gst/rtp/gstrtpsv3vdepay.c: * gst/rtp/gstrtptheorapay.c: * gst/rtp/gstrtpvorbispay.c: * gst/rtp/gstrtpvrawdepay.c: * gst/rtp/gstrtpvrawpay.c: rtp: use boilerplate 2009-12-23 00:38:05 +0100 Wim Taymans * gst/rtp/gstrtpL16pay.c: * gst/rtp/gstrtpL16pay.h: rtpL16pay: convert to baseaudiopayload Use GstRTPBaseAudioPayload as the base class. This saves a lot of code and fixes a bunch of problems that were already solved in the base class. Fixes #853367 2009-12-23 00:30:49 +0100 Wim Taymans * gst/rtp/gstrtppcmapay.c: rtppcmapay: the boilerplate macro sets parent_class 2009-12-22 22:27:21 +0100 Wim Taymans * gst/rtpmanager/rtpsession.c: * gst/rtpmanager/rtpsource.c: * gst/rtpmanager/rtpsource.h: rtpbin: avoid some structure copies Don't make copied in the getter and setter for SDES in the RTPSource. This avoids a couple of copies of the SDES structure when generating RTCP packets. 2009-08-31 18:42:25 +0200 Pascal Buhler * gst/rtpmanager/rtpsession.c: * gst/rtpmanager/rtpsource.c: * gst/rtpmanager/rtpsource.h: rtpmanager: improve SDES handling Store SDES internally as a struct to support multiple PRIV values. Include all values set in SDES struct when sending RTCP SDES. 2009-12-22 14:41:35 +0100 Wim Taymans * gst/rtp/gstrtph263depay.c: rtph263depay: add some fixmes 2009-12-22 14:35:13 +0100 Wim Taymans * gst/rtp/gstrtph263depay.c: rtph263depay: baseclass handles timestamps for us 2009-12-22 14:27:40 +0100 Wim Taymans * gst/rtp/gstrtph263depay.c: rtph263depay: reset start variable properly 2009-05-29 15:49:27 +0300 Marco Ballesio * gst/rtp/gstrtph263depay.c: * gst/rtp/gstrtph263depay.h: Drop the whole frame if a packet is lost. Fixes #582575 2009-12-21 20:39:53 +0100 Wim Taymans * gst/rtp/gstrtph264pay.c: * gst/rtp/gstrtph264pay.h: rtph264pay: add option to insert PPS/SPS in streams Add a new spspps-interval property to instruct the payloader to insert SPS and PPS at periodic intervals in the stream. Rework the SPS/PPS handling so that bytestream and AVC sample code both use the same code paths to handle sprop-parameter-sets. This also allows to have the AVC code to insert SPS/PPS like the bytestream code. Fixes #604913 2009-12-21 19:12:22 +0100 Mark Nauwelaerts * common: Automatic update of common submodule From 47cb23a to 14cec89 2009-12-21 12:01:53 -0300 Jonathan Conder * gst/qtdemux/qtdemux.c: * gst/qtdemux/qtdemux_fourcc.h: * gst/qtdemux/qtdemux_types.c: qtdemux: Adds new tags Adds some new tags mapping to qtdemux. Fixes #599759 2009-12-21 15:05:09 +0100 Wim Taymans * gst/rtpmanager/gstrtpbin.c: rtpbin: add property to remove pads automatically Add a property called autoremove to automatically remove the pads of sources that timed out. Fixes #554839 2009-12-21 14:55:16 +0100 Wim Taymans * gst/rtpmanager/gstrtpssrcdemux.c: ssrcdemux: fix comparison A NULL means no pad was found. 2009-11-08 11:49:14 +0100 Edward Hervey * sys/v4l2/gstv4l2src.c: v4l2src: Add GstURIHandler interface. Fixes #601143 This allows using v4l2://[] 2009-12-20 17:24:47 -0800 Michael Smith * gst/udp/gstmultiudpsink.c: multiudpsink: pass length parameter to g_convert 2009-12-18 12:44:50 +0100 Edward Hervey * gst/matroska/matroska-demux.c: matroska: Fix unitialized variable. Yes, it's stupid, but macosx compilers are even more stupid. 2009-12-17 16:01:25 +0100 Sebastian Dröge * gst/videomixer/blend_ayuv.c: videomixer: Fix assembly compilation on x86 Fixes bug #604814. 2009-12-17 17:37:03 +0100 Branko Čibej * gst/replaygain/rganalysis.c: rganalysis: fix timestamp rounding Use scaling function to round and avoid overflows. Fixes #604352 2009-12-17 17:27:42 +0100 Tiago Katcipis * gst/rtp/Makefile.am: * gst/rtp/gstrtp.c: * gst/rtp/gstrtpg723pay.c: * gst/rtp/gstrtpg723pay.h: rtp: add G723 payloader Fixes #597823 2009-12-17 16:22:56 +0100 Wim Taymans * gst/qtdemux/qtdemux.c: * gst/qtdemux/qtdemux_types.c: qtdemux: Fix ALAC codec_data parsing Fixes #604611 2009-12-16 17:28:30 -0300 Thiago Santos * gst/qtdemux/qtdemux.c: qtdemux: Remove cpp style coments Removes // comments and replace them with /* */ comments 2009-12-16 12:48:02 +0100 Mark Nauwelaerts * gst/matroska/matroska-demux.c: * gst/matroska/matroska-demux.h: matroskademux: also consider BlockNumber indicated in index when seeking 2009-12-16 12:43:27 +0100 Mark Nauwelaerts * gst/matroska/ebml-read.c: * gst/matroska/ebml-read.h: * gst/matroska/matroska-demux.c: * gst/matroska/matroska-demux.h: matroskademux: support push based mode Fixes #598610. 2009-12-16 12:44:36 +0100 Mark Nauwelaerts * gst/matroska/ebml-read.c: matroskademux: fix ebml read cache usage 2009-12-16 10:50:32 +0100 Sebastian Dröge * gst/videomixer/blend_ayuv.c: videomixer: Use movzbl instead of movzxb for moving one byte to a l register For some reason latest gcc/binutils accept movzxb here while movzbl would be correct and is the only thing accepted by older gcc/binutils. Fixes bug #604679. 2009-12-16 06:59:01 +0100 Sebastian Dröge * gst/videomixer/blend_ayuv.c: videomixer: src/dest are input and output of the AYUV blending MMX assembler 2009-12-15 18:18:54 +0100 Sebastian Dröge * gst/audiofx/audiowsincband.c: audiowsincband: Use the same upper length limit as audiowsinclimit 2009-12-12 17:00:50 +0100 Sebastian Dröge * gst/audiofx/audiowsincband.c: * gst/audiofx/audiowsinclimit.c: audiowsinc{limit,band}: Allow much larger filter lengths now 2009-12-11 12:27:32 +0100 Sebastian Dröge * gst/audiofx/audiofxbasefirfilter.c: audiofxbasefirfilter: Fix frequency response calculation 2009-12-08 14:57:02 +0100 Sebastian Dröge * gst/audiofx/audiofxbasefirfilter.c: audiofxbasefirfilter: Remove dead assignments 2009-12-06 16:58:51 +0100 Sebastian Dröge * gst/audiofx/audiofxbasefirfilter.c: audiofxbasefirfilter: Add special processing functions for Mono/Stereo This provides another 7% speedup for the time domain convolution and 1.5% speedup for the FFT convolution on Mono input. This optimization assumes that the compiler simplifies calculations and conditions on constant numbers and unrolls loops with a constant number of repeats. 2009-12-04 09:25:49 +0100 Sebastian Dröge * gst/audiofx/audiofxbasefirfilter.c: * gst/audiofx/audiofxbasefirfilter.h: audiofxbasefirfilter: Add a "low-latency" mode This will always use time-domain convolution, which lowers the latency. With FFT convolution it's always a multiple of the kernel length, with time domain convolution it's only the pre-latency of the filter kernel. 2009-12-04 09:00:22 +0100 Sebastian Dröge * gst/audiofx/audiofxbasefirfilter.c: audiofxbasefirfilter: Remove obsolete TODO comments 2009-12-03 20:12:01 +0100 Sebastian Dröge * gst/audiofx/audiofxbasefirfilter.c: audiofxbasefirfilter: Use samples everywhere instead of samples*channels sometimes 2009-12-03 17:27:13 +0100 Sebastian Dröge * gst/audiofx/Makefile.am: * gst/audiofx/audiofxbasefirfilter.c: * gst/audiofx/audiofxbasefirfilter.h: audiofxbasefirfilter: FFT convolution implementation This provides a great speedup, especially the relationship between kernel length and processing size is now logarithmic instead of linear. Below a kernel size of 32 it's a bit slower, afterwards it's much faster: 17 0.788000 -> 0.950000 33 1.208000 -> 1.146000 65 2.166000 -> 1.146000 ... 4097 107.444000 -> 1.508000 For sizes smaller 32 the normal time-domain convolution is chosen, for larger sizes the FFT convolution is automatically used. Fixes bug #594381. 2009-11-27 20:33:14 +0100 Sebastian Dröge * gst/audiofx/audiofxbasefirfilter.c: * gst/audiofx/audiofxbasefirfilter.h: audiofxbasefirfilter: Make most code parts independent of the processing functions and used convolution algorithm Only remaining part is the residue pushing, which will be fixed later. 2009-11-26 15:17:27 +0100 Sebastian Dröge * gst/audiofx/audiofxbasefirfilter.c: audiofxbasefirfilter: Optimize time-domain convolution Remove some redundant calculations, move comparisions out of inner loops, etc. This makes the convolution about 3 (!) times faster but processing time is of course still proportional to the filter size. 2009-11-26 10:45:37 +0100 Sebastian Dröge * gst/audiofx/audiofxbasefirfilter.c: audiofxbasefirfilter: Use _CAST macros in some places and do some calculations only once 2009-11-25 18:12:05 +0100 Sebastian Dröge * gst/audiofx/audiofxbasefirfilter.c: * gst/audiofx/audiofxbasefirfilter.h: audiofxbasefirfilter: Rewrite timestamp tracking It's much simpler now and doesn't introduce accumulating rounding errors. 2009-11-25 17:39:53 +0100 Sebastian Dröge * gst/audiofx/audiofxbasefirfilter.c: * gst/audiofx/audiofxbasefirfilter.h: audiofxbasefirfilter: Rename some variables and change comments 2009-11-24 20:06:25 +0100 Sebastian Dröge * gst/audiofx/audiofxbasefirfilter.c: * gst/audiofx/audiofxbasefirfilter.h: audiofxbasefirfilter: Add const qualifier to the source data array 2009-12-14 20:08:06 +0100 Sebastian Dröge * gst/videomixer/Makefile.am: * gst/videomixer/blend_ayuv.c: * gst/videomixer/videomixer.c: videomixer: Add MMX implementations of the AYUV blending and color filling functions This provides a 20% speedup for blending and 100% for color filling. The blending can probably be optimized even more. 2009-12-13 13:19:43 +0000 Tim-Philipp Müller * gst/id3demux/id3v2frames.c: id3demux: prefer two letter ISO 639-1 code for extended comment 2009-12-13 13:10:12 +0000 Tim-Philipp Müller * gst/qtdemux/qtdemux.c: qtdemux: fix up language code extraction some more Quicktime uses ISO 639-2 for language codes, but GST_TAG_LANGUAGE is supposed to hold a ISO 639-1 code, so convert as needed using the new API from -base. See #602126. 2009-12-13 12:45:22 +0000 Tim-Philipp Müller * gst/matroska/matroska-demux.c: * gst/matroska/matroska-mux.c: matroska: fix language code writing and extraction Matroska uses three-letter ISO 639-2B codes, but GST_TAG_LANGUAGE is supposed to contain two-letter ISO 639-1 codes, so use new language code mapping functions in -base to convert between those two as needed. Fixes #505823. 2009-12-07 20:54:07 +0000 Tim-Philipp Müller * gst/avi/gstavidemux.c: avidemux: minor debug message changes Fix up a few debug messages so that it's clearer what they mean. 2009-12-12 17:44:04 -0300 Thiago Santos * gst/qtdemux/qtdemux.c: Revert "qtdemux: Correctly parse classification tags" This reverts commit cd883aa60c1133196a6ae052884d15c295c37dde. Previous code was correct, 4 is due to table and language code, not only language code 2009-12-12 16:28:36 -0300 Thiago Santos * gst/qtdemux/qtdemux.c: qtdemux: Correctly parse classification tags In clsf atoms, the language code is 2 bytes long, not 4. 2009-12-12 16:55:13 +0100 Sebastian Dröge * gst/videomixer/videomixer.c: videomixer: Dequeue current buffer on FLUSH_STOP and don't unref NULL buffers ... NULL buffers shouldn't really happen anymore when popping the buffer from GstCollectPads but better check for this and print a warning. 2009-12-11 13:11:12 +0100 Sebastian Dröge * gst/videomixer/blend_i420.c: videomixer: Fix stupid mistake in last commit 2009-12-11 12:35:59 +0100 Sebastian Dröge * gst/videomixer/blend_i420.c: videomixer: Don't do floating point math in the inner processing loop for I420 blending 2009-12-10 18:43:44 +0100 Wim Taymans * gst/rtsp/gstrtspsrc.c: rtspsrc: handle NULL and empty transport strings When an RTSP extension returns NULL or an empty transport string, just ignore it and try to get the next possible transport. Fixes playback of RealMedia streams. 2009-12-10 18:42:51 +0100 Wim Taymans * gst/rtsp/gstrtspsrc.c: rtspsrc: install event function on internal RTCP pad Install a custom event function on the internal RTCP pad so that we can reply TRUE to a latency event. 2009-12-10 10:48:49 +0100 Sebastian Dröge * gst/videomixer/blend_ayuv.c: * gst/videomixer/blend_bgra.c: * gst/videomixer/blend_rgb.c: videomixer: Remove wrong comments, copied from the I420 blend function 2009-12-09 21:15:07 +0100 Sebastian Dröge * gst/videomixer/videomixer.c: videomixer: The queued duration is a signed integer ...and it will really be negative sometimes. 2009-12-09 21:03:57 +0100 Sebastian Dröge * gst/videomixer/videomixer.c: videomixer: Only pop buffers from collectpads after they're fully consumed This decreases latency and memory usage because new buffers are only accepted by collectpads if there's no queued buffer. 2009-12-09 20:42:44 +0100 Sebastian Dröge * gst/matroska/matroska-demux.c: * gst/matroska/matroska-demux.h: matroskademux: Clean up position/duration handling Also use the last end time for closing the segment, not the start time of the last buffer. 2009-12-09 16:50:02 +0100 Sebastian Dröge * gst/matroska/matroska-demux.c: matroskademux: Close the segment on EOS if the real duration is known 2009-12-09 16:46:18 +0100 Sebastian Dröge * gst/matroska/matroska-demux.c: matroskademux: Update duration if current buffer is already after the old duration 2009-12-09 16:43:41 +0100 Sebastian Dröge * gst/matroska/matroska-demux.c: matroskademux: Drop buffers that are after segment stop ...and if this happened for all streams go EOS. 2009-12-09 16:41:04 +0100 Sebastian Dröge * gst/matroska/matroska-demux.c: matroskademux: Fix position tracking and sending of filler segments 2009-12-09 16:15:09 +0100 Sebastian Dröge * gst/videomixer/videomixer.c: videomixer: Use gst_util_uint64_scale_int() for fps to seconds per frame calculations 2009-12-08 17:34:15 +0100 Sebastian Dröge * gst/matroska/matroska-demux.c: matroskademux: Keep the segment stop position for update newsegment events 2009-12-04 14:42:49 +0100 Sebastian Dröge * configure.ac: * ext/Makefile.am: * ext/ladspa/Makefile.am: * ext/ladspa/gstladspa.c: * ext/ladspa/gstladspa.h: * ext/ladspa/gstsignalprocessor.c: * ext/ladspa/gstsignalprocessor.h: * ext/ladspa/load.c: * ext/ladspa/search.c: * ext/ladspa/utils.h: ladspa: Remove the sources from gst-plugins-good It's disabled anyway and the latest version of it is in gst-plugins-bad. Fixes bug #603779. 2009-12-04 13:50:59 +0100 Wim Taymans * gst/avi/gstavidemux.c: avidemux: init current_entry in push mode Set the current_entry to 0 (instead of -1) in push mode so that we correctly calculate the current frame number and timestamp. Add some more debug info and fic the duration debug. 2009-12-04 11:14:03 +0000 Tim-Philipp Müller * gst/rtsp/gstrtspsrc.c: rtspsrc: fix major memory leak when playing back rtsp video streams Don't forget to unref QoS, navigation and latency events when dropping them. 2009-12-03 08:58:08 +0000 Tim-Philipp Müller * gst/matroska/matroska-demux.c: matroskademux: only send pending tags with newsegment events Send pending tags only from the streaming thread, just after we've sent the newsegment event, not with e.g. flush-start. This not only does the right thing, but also makes sure we're not trampling over variables set up in the streaming thread from the seeking thread in case someone tries to issue a seek just as the demuxer is parsing the headers. Fixes #601617. Spotted by Ognyan Tonchev. 2009-12-03 17:49:55 -0300 Thiago Santos * gst/qtdemux/qtdemux.c: qtdemux: fix debug message printf args Fixes debug message printf format to make it build in mac's gcc 2009-12-02 13:33:20 -0300 Thiago Santos * ext/shout2/gstshout2.c: shout2: Convert delay correctly Use GST_MSECOND to convert delay in msecs to nanosecs Fixes #603547 2009-12-01 19:24:02 +0100 Wim Taymans * ext/jpeg/gstjpegdec.c: jpegdec: reset segment info after flush Reset the segment info after a flush. We use the segment for handling QoS and if we don't reset the segment, QoS is basically disabled after a flushing seek. 2009-12-01 15:07:06 +0000 Tim-Philipp Müller * common: Automatic update of common submodule From 87bf428 to 47cb23a 2009-12-01 14:15:46 +0100 Sebastian Dröge * common: Automatic update of common submodule From da4c75c to 87bf428 2009-11-30 15:59:50 +0100 Aurelien Grimaud * gst/rtpmanager/rtpsession.c: rtpsession: avoid buffer ref/unref pairs for CSRCs We ref the buffer before pushing it downstream in order to get the CSRCs of it after pushing. This causes performance problems when downstream elements want to change the metadata because the buffer needs to be subbuffered. Instead, read and store the CSRCs of the buffer in an array before pushing it and process the array after pushing the buffer. This allows us to remove the ref/unref pair. Fixes #603376 2009-11-28 19:23:26 +0100 Wim Taymans * ext/shout2/gstshout2.c: * ext/shout2/gstshout2.h: shout2: use gstpoll for timeouts Use our own GstPoll based timeout instead of the shout sleep so that we can interrupt when doing a state change and shutting down. Fixes #602887 2009-11-28 12:25:06 +0100 Wim Taymans * tests/check/elements/rtpjitterbuffer.c: check: fix jitterbuffer check Make sure we set a base_time on the element. Fix the timeout to at least twice the jitterbuffer latency. Enable previously failing tests. Remove impossible checks. 2009-11-27 18:55:20 +0100 Edward Hervey * common: Automatic update of common submodule From 53a2485 to da4c75c 2009-11-26 16:14:30 +0100 Mark Nauwelaerts * gst/rtp/gstrtph264depay.c: * gst/rtp/gstrtph264depay.h: rtph264depay: optionally merge NALUs into Access Units ... which may be expected/desired by some downstream decoders (and spec-wise highly recommended for at least non-bytestream mode). 2009-11-26 17:29:03 +0100 Mark Nauwelaerts * gst/qtdemux/qtdemux.c: qtdemux: fix timestamp datatype 2009-11-25 10:38:23 -0600 Wim Taymans * gst/rtpmanager/gstrtpjitterbuffer.c: jitterbuffer: avoid using wrong clock-rate Check for a valid clock-rate before attempting to estimate the npt stop time. 2009-11-25 10:37:30 -0600 Wim Taymans * gst/rtpmanager/gstrtpbin.c: rtpbin: fix typo in comments 2009-11-25 16:05:10 +0200 Stefan Kost * tests/check/elements/rtpjitterbuffer.c: rtpjitterbuffertest: add one more test and file a bug now CHange the backwards test to always send first buffer first to have a define basetime. Add another test that sends buffers backwards to assert that only first sent buffer is keep and used as basetime. Disabled those tests still, as its not passing/failing consitently and file a bug for jitterbuffer. 2009-11-25 10:17:34 +0200 Stefan Kost * tests/check/elements/rtpjitterbuffer.c: jitterbuffertest: improve the test the tests are a bit more solid now but still not produce reliable results. Wonder if they are still flawky or if its a bug in jitterbuffer. 2009-11-24 11:13:06 -0800 Michael Smith * gst/udp/gstmultiudpsink.c: multiudpsink: return error message on windows too. 2009-11-24 10:58:49 -0800 Michael Smith * gst/udp/gstmultiudpsink.c: multiudpsink: first phase of fixing up error reporting for windows. 2009-10-30 03:13:54 -0300 Thiago Santos * gst/avi/gstavimux.c: avimux: also set the suggested buf size for audio We were only setting the suggested buf size for video, we can set it for audio as well. This and 195e14529d80ef318ce3a778c1995efb11f266cd fix an issue that prevented seeking on large avi files on WMP (non-recent versions). 2009-11-04 16:10:23 -0300 Thiago Santos * gst/avi/gstavimux.c: * gst/avi/gstavimux.h: avimux: fix indx duration for PCM audio GstBuffers for PCM audio usually contains more than 1 sample, we need to get the total number of samples to set the indx duration. 2009-11-04 16:04:10 -0300 Thiago Santos * gst/avi/gstavimux.c: avimux: Audio buffers should be picked earlier Adds a 0.5s advantage for audio buffers to being picked earlier for muxing. 2009-11-24 16:40:19 +0100 Robert Swain * gst/qtdemux/qtdemux.c: qtdemux: Fix push mode by making sure stbl information is available in next_entry_size () 2009-11-24 16:35:20 +0100 Robert Swain * gst/qtdemux/qtdemux.c: qtdemux: Fix order of arguments in log message 2009-11-24 15:51:21 +0200 Stefan Kost * ext/jpeg/gstjpegenc.c: jpegenc: fix spelling in comment 2009-11-23 17:58:17 +0100 Robert Swain * common: build system: Fix wrongly committed change to common/ 2009-11-10 10:26:07 +0100 Robert Swain * gst/qtdemux/qtdemux.c: qtdemux: Ease debugging by removing a goto for an error message 2009-11-14 15:52:09 +0100 Robert Swain * common: * gst/qtdemux/qtdemux.c: qtdemux: Parse per sample rather than all at once but build complete index when seeking 2009-11-04 17:31:15 +0100 Robert Swain * gst/qtdemux/qtdemux.c: qtdemux: Save atom data for later use so it doesn't get freed after initial parsing 2009-11-06 11:00:04 +0100 Robert Swain * gst/qtdemux/qtdemux.c: qtdemux: Parse from the previously parsed sample up to sample n 2009-11-04 17:04:22 +0100 Robert Swain * gst/qtdemux/qtdemux.c: qtdemux: Make qtdemux_parse_samples () parse up to n samples 2009-10-28 17:49:02 +0000 Robert Swain * gst/qtdemux/qtdemux.c: qtdemux: Separate off stbl sub-atom initialisation 2009-10-26 22:42:36 +0000 Robert Swain * gst/qtdemux/qtdemux.c: qtdemux: Move variables into context in preparation for refactorisation 2009-10-26 20:36:08 +0000 Robert Swain * gst/qtdemux/qtdemux.c: qtdemux: Fix bug where stps is never parsed due to logic error 2009-11-04 17:31:15 +0100 Robert Swain * gst/qtdemux/qtdemux.c: qtdemux: Port ctts from Gnode * to GstByteReader 2009-10-23 13:06:44 +0100 Robert Swain * gst/qtdemux/qtatomparser.h: * gst/qtdemux/qtdemux.c: * gst/qtdemux/qtdemux_dump.c: * gst/qtdemux/qtdemux_dump.h: * gst/qtdemux/qtdemux_types.h: qtdemux: Switch from QtAtomParser to GstByteReader 2009-11-23 12:53:50 +0100 Wim Taymans * gst/qtdemux/qtdemux.c: qtdemux: fix typo and grammar 2009-11-20 10:30:00 +0000 Tim-Philipp Müller * gst/deinterlace/gstdeinterlace.c: deinterlace: fix typo in mode enum description 2009-11-20 11:25:49 +0200 Stefan Kost * gst/rtpmanager/gstrtpbin.c: docs: more links and better short description Fix spelling of GstRtpSsrcDemux to get it linked. Add more links. Change the short description to be more meaningful. 2009-11-20 09:58:26 +0100 Sebastian Dröge * tests/check/elements/wavpackparse.c: wavpackparse: Fix unit test for recent position reporting changes 2009-11-19 16:09:38 +0100 Sebastian Dröge * ext/wavpack/gstwavpackparse.c: wavpackparse: After pushing a frame, update last_stop to the end of the frame This improves position reporting, especially because of the fact that WavPack frames are usually 0.5-1.0 seconds long. 2009-11-19 16:08:33 +0100 Sebastian Dröge * ext/wavpack/gstwavpackparse.c: wavpackparse: Allow pulling the last WavPack frame of a file Because of a >= instead of a >, that last frame of a WavPack file would never be parsed in pull mode. 2009-11-19 10:30:43 +0000 Tim-Philipp Müller * common: Automatic update of common submodule From 0702fe1 to 53a2485 2009-10-29 08:29:38 -0300 Thiago Santos * gst/qtdemux/qtdemux.c: * gst/qtdemux/qtdemux_fourcc.h: qtdemux: Add more fields to SVQ3 caps qtdemux only added the whole stsd atom as 'codec_data' in its output caps for SVQ3. This patch makes it add the SEQH (inside a SMI atom) and a gamma field (taken from the gama atom) if available. Fixes #587922 2009-11-18 17:55:42 +0100 Edward Hervey * gst/wavenc/gstwavenc.c: wavenc: Raise rank of muxer to PRIMARY 2009-11-18 17:54:16 +0100 Edward Hervey * gst/y4m/gsty4mencode.c: y4m: Raise rank of encoder to PRIMARY 2009-11-18 17:54:02 +0100 Edward Hervey * gst/law/alaw.c: * gst/law/mulaw.c: law: Raise rank of encoders to PRIMARY 2009-11-12 19:11:18 +0000 Bastien Nocera * gst/rtsp/gstrtspsrc.c: * gst/rtsp/gstrtspsrc.h: Add user-id and user-pw properties So that one doesn't need to modify the URL to have access to authenticated RTSP streams. fixes #601728 2009-11-18 12:22:10 +0100 Wim Taymans * ext/pulse/pulsesink.c: pulsesink: use acquired flag when checking valid state Use the acquired field of the ringbuffer in get_time to know when we are in an invalid state. We don't clear the rate flag when releasing the ringbuffer so this values is not usable. Avoids some error messages being posted because the pulseaudio connection is down. 2009-11-18 10:17:02 +0000 Tim-Philipp Müller * configure.ac: configure: bump core requirement to 0.10.25.1 as well Make implicit requirement explicit. 2009-11-18 12:53:44 +0100 Mark Nauwelaerts * gst/qtdemux/qtdemux.c: qtdemux: fix bogus memory chunk size check 2009-11-18 12:01:52 +0100 Wim Taymans * ext/pulse/pulsesink.c: pulsesink: implement some more callbacks Implement some more callbacks for debugging purposes. 2009-11-11 15:50:19 +0100 Wim Taymans * gst/rtpmanager/gstrtpjitterbuffer.c: jitterbuffer: release lock before emiting signals Release the jbuf lock before emiting the request-pt-map signal to avoid deadlocks. We also need to catch the shutdown case when locking again. Fixes #593354 2009-11-11 11:59:16 +0100 Wim Taymans * gst/rtp/Makefile.am: * gst/rtp/gstrtp.c: * gst/rtp/gstrtpbvdepay.c: * gst/rtp/gstrtpbvdepay.h: rtp: add BroadcomVoice depayloader 2009-11-11 11:38:36 +0100 Wim Taymans * gst/rtp/gstrtpbvpay.c: rtpbvpay: add rfc reference 2009-11-11 11:37:07 +0100 Wim Taymans * gst/rtp/Makefile.am: * gst/rtp/gstrtp.c: * gst/rtp/gstrtpbvpay.c: * gst/rtp/gstrtpbvpay.h: rtp: add BroadcomVoice payloader 2009-11-09 12:17:34 +0100 Jan Urbański * gst/flv/gstflvmux.c: flvmux: properly finish the ECMA array The ECMA array with the file index was missing a mandatory end marker. Fixes bug #601242. 2009-11-18 02:15:15 +0000 Jan Schmidt * gst/deinterlace/gstdeinterlace.c: Use new still-frame API from gst-plugins-base 2009-11-18 02:14:46 +0000 Jan Schmidt * configure.ac: Bump gst-plugins-base requirement to 0.10.25.1 2009-11-17 17:59:13 -0800 Michael Smith * gst/qtdemux/qtdemux.c: qtdemux: identify IMA adpcm in qt properly. 2009-11-18 01:27:37 +0000 Jan Schmidt * configure.ac: * win32/common/config.h: Back to development -> 0.10.17.1 2009-11-17 01:53:08 +0000 Jan Schmidt * gst-plugins-good.doap: Add release 0.10.17 to the doap file