2011-05-10 Tim-Philipp Müller <tim.muller@collabora.co.uk>

=== release 0.10.29 ===

2011-05-10 Tim-Philipp Müller <[email protected]>

* configure.ac:
releasing 0.10.29, "Soft Cheese Enthusiast"

2011-05-05 13:24:23 +0200 Edward Hervey <[email protected]>

* gst/isomp4/gstqtmux.c:
qtmux: Fix signed floating point values writing
You would end up on some architectures with 0 being written out
instead of the proper value.
https://bugzilla.gnome.org/show_bug.cgi?id=649449

2011-05-04 12:04:15 +0200 Mark Nauwelaerts <[email protected]>

* gst/matroska/matroska-mux.c:
matroskamux: avoid building index when streamable
... as it will not be written anyway.
Fixes #648937 (?).

2011-05-02 12:09:02 +0100 Tim-Philipp Müller <[email protected]>

* Makefile.am:
build: add old qtdemux/quicktime directories to CRUFT_DIRS and CRUFT_FILES

2011-05-01 00:04:03 -0400 Tom Janiszewski <[email protected]>

* gst/flv/gstflvmux.c:
flvmux: don't overwrite metadata tag with duration in streaming mode
A duration tag gets inserted only for streamable=false, so only
update/write the duration later if we actually inserted that tag,
otherwise we write garbage into other tags.
https://bugzilla.gnome.org/show_bug.cgi?id=649060

2011-04-30 18:16:36 +0100 Tim-Philipp Müller <[email protected]>

* configure.ac:
* docs/plugins/gst-plugins-good-plugins.hierarchy:
* docs/plugins/gst-plugins-good-plugins.interfaces:
* docs/plugins/gst-plugins-good-plugins.prerequisites:
* docs/plugins/inspect/plugin-1394.xml:
* docs/plugins/inspect/plugin-aasink.xml:
* docs/plugins/inspect/plugin-alaw.xml:
* docs/plugins/inspect/plugin-alpha.xml:
* docs/plugins/inspect/plugin-alphacolor.xml:
* docs/plugins/inspect/plugin-annodex.xml:
* docs/plugins/inspect/plugin-apetag.xml:
* docs/plugins/inspect/plugin-audiofx.xml:
* docs/plugins/inspect/plugin-audioparsers.xml:
* docs/plugins/inspect/plugin-auparse.xml:
* docs/plugins/inspect/plugin-autodetect.xml:
* docs/plugins/inspect/plugin-avi.xml:
* docs/plugins/inspect/plugin-cacasink.xml:
* docs/plugins/inspect/plugin-cairo.xml:
* docs/plugins/inspect/plugin-cutter.xml:
* docs/plugins/inspect/plugin-debug.xml:
* docs/plugins/inspect/plugin-deinterlace.xml:
* docs/plugins/inspect/plugin-dv.xml:
* docs/plugins/inspect/plugin-efence.xml:
* docs/plugins/inspect/plugin-effectv.xml:
* docs/plugins/inspect/plugin-equalizer.xml:
* docs/plugins/inspect/plugin-esdsink.xml:
* docs/plugins/inspect/plugin-flac.xml:
* docs/plugins/inspect/plugin-flv.xml:
* docs/plugins/inspect/plugin-flxdec.xml:
* docs/plugins/inspect/plugin-gconfelements.xml:
* docs/plugins/inspect/plugin-gdkpixbuf.xml:
* docs/plugins/inspect/plugin-goom.xml:
* docs/plugins/inspect/plugin-goom2k1.xml:
* docs/plugins/inspect/plugin-gstrtpmanager.xml:
* docs/plugins/inspect/plugin-halelements.xml:
* docs/plugins/inspect/plugin-icydemux.xml:
* docs/plugins/inspect/plugin-id3demux.xml:
* docs/plugins/inspect/plugin-imagefreeze.xml:
* docs/plugins/inspect/plugin-interleave.xml:
* docs/plugins/inspect/plugin-isomp4.xml:
* docs/plugins/inspect/plugin-jack.xml:
* docs/plugins/inspect/plugin-jpeg.xml:
* docs/plugins/inspect/plugin-level.xml:
* docs/plugins/inspect/plugin-matroska.xml:
* docs/plugins/inspect/plugin-monoscope.xml:
* docs/plugins/inspect/plugin-mulaw.xml:
* docs/plugins/inspect/plugin-multifile.xml:
* docs/plugins/inspect/plugin-multipart.xml:
* docs/plugins/inspect/plugin-navigationtest.xml:
* docs/plugins/inspect/plugin-oss4.xml:
* docs/plugins/inspect/plugin-ossaudio.xml:
* docs/plugins/inspect/plugin-png.xml:
* docs/plugins/inspect/plugin-pulseaudio.xml:
* docs/plugins/inspect/plugin-replaygain.xml:
* docs/plugins/inspect/plugin-rtp.xml:
* docs/plugins/inspect/plugin-rtsp.xml:
* docs/plugins/inspect/plugin-shapewipe.xml:
* docs/plugins/inspect/plugin-shout2send.xml:
* docs/plugins/inspect/plugin-smpte.xml:
* docs/plugins/inspect/plugin-soup.xml:
* docs/plugins/inspect/plugin-spectrum.xml:
* docs/plugins/inspect/plugin-speex.xml:
* docs/plugins/inspect/plugin-taglib.xml:
* docs/plugins/inspect/plugin-udp.xml:
* docs/plugins/inspect/plugin-video4linux2.xml:
* docs/plugins/inspect/plugin-videobox.xml:
* docs/plugins/inspect/plugin-videocrop.xml:
* docs/plugins/inspect/plugin-videofilter.xml:
* docs/plugins/inspect/plugin-videomixer.xml:
* docs/plugins/inspect/plugin-wavenc.xml:
* docs/plugins/inspect/plugin-wavpack.xml:
* docs/plugins/inspect/plugin-wavparse.xml:
* docs/plugins/inspect/plugin-ximagesrc.xml:
* docs/plugins/inspect/plugin-y4menc.xml:
* po/fr.po:
* win32/common/config.h:
0.10.28.4 pre-release

2011-04-30 17:46:36 +0100 Tim-Philipp Müller <[email protected]>

* Android.mk:
* configure.ac:
* docs/plugins/Makefile.am:
* docs/plugins/gst-plugins-good-plugins-docs.sgml:
* docs/plugins/inspect/plugin-isomp4.xml:
* docs/plugins/inspect/plugin-quicktime.xml:
* gst-plugins-good.spec.in:
* gst/isomp4/LEGAL:
* gst/isomp4/Makefile.am:
* gst/isomp4/atoms.c:
* gst/isomp4/atoms.h:
* gst/isomp4/atomsrecovery.c:
* gst/isomp4/atomsrecovery.h:
* gst/isomp4/descriptors.c:
* gst/isomp4/descriptors.h:
* gst/isomp4/fourcc.h:
* gst/isomp4/ftypcc.h:
* gst/isomp4/gstqtmoovrecover.c:
* gst/isomp4/gstqtmoovrecover.h:
* gst/isomp4/gstqtmux-doc.c:
* gst/isomp4/gstqtmux-doc.h:
* gst/isomp4/gstqtmux.c:
* gst/isomp4/gstqtmux.h:
* gst/isomp4/gstqtmuxmap.c:
* gst/isomp4/gstqtmuxmap.h:
* gst/isomp4/gstrtpxqtdepay.c:
* gst/isomp4/gstrtpxqtdepay.h:
* gst/isomp4/isomp4-plugin.c:
* gst/isomp4/properties.c:
* gst/isomp4/properties.h:
* gst/isomp4/qtatomparser.h:
* gst/isomp4/qtdemux.c:
* gst/isomp4/qtdemux.h:
* gst/isomp4/qtdemux.vcproj:
* gst/isomp4/qtdemux_dump.c:
* gst/isomp4/qtdemux_dump.h:
* gst/isomp4/qtdemux_fourcc.h:
* gst/isomp4/qtdemux_lang.c:
* gst/isomp4/qtdemux_lang.h:
* gst/isomp4/qtdemux_types.c:
* gst/isomp4/qtdemux_types.h:
* gst/isomp4/qtpalette.h:
* gst/quicktime/LEGAL:
* gst/quicktime/Makefile.am:
* gst/quicktime/atoms.c:
* gst/quicktime/atoms.h:
* gst/quicktime/atomsrecovery.c:
* gst/quicktime/atomsrecovery.h:
* gst/quicktime/descriptors.c:
* gst/quicktime/descriptors.h:
* gst/quicktime/fourcc.h:
* gst/quicktime/ftypcc.h:
* gst/quicktime/gstqtmoovrecover.c:
* gst/quicktime/gstqtmoovrecover.h:
* gst/quicktime/gstqtmux-doc.c:
* gst/quicktime/gstqtmux-doc.h:
* gst/quicktime/gstqtmux.c:
* gst/quicktime/gstqtmux.h:
* gst/quicktime/gstqtmuxmap.c:
* gst/quicktime/gstqtmuxmap.h:
* gst/quicktime/gstrtpxqtdepay.c:
* gst/quicktime/gstrtpxqtdepay.h:
* gst/quicktime/properties.c:
* gst/quicktime/properties.h:
* gst/quicktime/qtatomparser.h:
* gst/quicktime/qtdemux.c:
* gst/quicktime/qtdemux.h:
* gst/quicktime/qtdemux.vcproj:
* gst/quicktime/qtdemux_dump.c:
* gst/quicktime/qtdemux_dump.h:
* gst/quicktime/qtdemux_fourcc.h:
* gst/quicktime/qtdemux_lang.c:
* gst/quicktime/qtdemux_lang.h:
* gst/quicktime/qtdemux_types.c:
* gst/quicktime/qtdemux_types.h:
* gst/quicktime/qtpalette.h:
* gst/quicktime/quicktime.c:
* po/POTFILES.in:
quicktime: rename plugin to isomp4
https://bugzilla.gnome.org/show_bug.cgi?id=648004

2011-04-27 12:45:51 +0100 Tim-Philipp Müller <[email protected]>

* configure.ac:
* docs/plugins/gst-plugins-good-plugins.args:
* docs/plugins/gst-plugins-good-plugins.hierarchy:
* docs/plugins/gst-plugins-good-plugins.interfaces:
* docs/plugins/gst-plugins-good-plugins.prerequisites:
* docs/plugins/inspect/plugin-1394.xml:
* docs/plugins/inspect/plugin-aasink.xml:
* docs/plugins/inspect/plugin-alaw.xml:
* docs/plugins/inspect/plugin-alpha.xml:
* docs/plugins/inspect/plugin-alphacolor.xml:
* docs/plugins/inspect/plugin-annodex.xml:
* docs/plugins/inspect/plugin-apetag.xml:
* docs/plugins/inspect/plugin-audiofx.xml:
* docs/plugins/inspect/plugin-audioparsers.xml:
* docs/plugins/inspect/plugin-auparse.xml:
* docs/plugins/inspect/plugin-autodetect.xml:
* docs/plugins/inspect/plugin-avi.xml:
* docs/plugins/inspect/plugin-cacasink.xml:
* docs/plugins/inspect/plugin-cairo.xml:
* docs/plugins/inspect/plugin-cutter.xml:
* docs/plugins/inspect/plugin-debug.xml:
* docs/plugins/inspect/plugin-deinterlace.xml:
* docs/plugins/inspect/plugin-dv.xml:
* docs/plugins/inspect/plugin-efence.xml:
* docs/plugins/inspect/plugin-effectv.xml:
* docs/plugins/inspect/plugin-equalizer.xml:
* docs/plugins/inspect/plugin-esdsink.xml:
* docs/plugins/inspect/plugin-flac.xml:
* docs/plugins/inspect/plugin-flv.xml:
* docs/plugins/inspect/plugin-flxdec.xml:
* docs/plugins/inspect/plugin-gconfelements.xml:
* docs/plugins/inspect/plugin-gdkpixbuf.xml:
* docs/plugins/inspect/plugin-goom.xml:
* docs/plugins/inspect/plugin-goom2k1.xml:
* docs/plugins/inspect/plugin-gstrtpmanager.xml:
* docs/plugins/inspect/plugin-halelements.xml:
* docs/plugins/inspect/plugin-icydemux.xml:
* docs/plugins/inspect/plugin-id3demux.xml:
* docs/plugins/inspect/plugin-imagefreeze.xml:
* docs/plugins/inspect/plugin-interleave.xml:
* docs/plugins/inspect/plugin-jack.xml:
* docs/plugins/inspect/plugin-jpeg.xml:
* docs/plugins/inspect/plugin-level.xml:
* docs/plugins/inspect/plugin-matroska.xml:
* docs/plugins/inspect/plugin-mulaw.xml:
* docs/plugins/inspect/plugin-multifile.xml:
* docs/plugins/inspect/plugin-multipart.xml:
* docs/plugins/inspect/plugin-navigationtest.xml:
* docs/plugins/inspect/plugin-oss4.xml:
* docs/plugins/inspect/plugin-ossaudio.xml:
* docs/plugins/inspect/plugin-png.xml:
* docs/plugins/inspect/plugin-pulseaudio.xml:
* docs/plugins/inspect/plugin-quicktime.xml:
* docs/plugins/inspect/plugin-replaygain.xml:
* docs/plugins/inspect/plugin-rtp.xml:
* docs/plugins/inspect/plugin-rtsp.xml:
* docs/plugins/inspect/plugin-shapewipe.xml:
* docs/plugins/inspect/plugin-shout2send.xml:
* docs/plugins/inspect/plugin-smpte.xml:
* docs/plugins/inspect/plugin-soup.xml:
* docs/plugins/inspect/plugin-spectrum.xml:
* docs/plugins/inspect/plugin-speex.xml:
* docs/plugins/inspect/plugin-taglib.xml:
* docs/plugins/inspect/plugin-udp.xml:
* docs/plugins/inspect/plugin-video4linux2.xml:
* docs/plugins/inspect/plugin-videobox.xml:
* docs/plugins/inspect/plugin-videocrop.xml:
* docs/plugins/inspect/plugin-videofilter.xml:
* docs/plugins/inspect/plugin-videomixer.xml:
* docs/plugins/inspect/plugin-wavenc.xml:
* docs/plugins/inspect/plugin-wavpack.xml:
* docs/plugins/inspect/plugin-wavparse.xml:
* docs/plugins/inspect/plugin-ximagesrc.xml:
* docs/plugins/inspect/plugin-y4menc.xml:
* po/bg.po:
* po/ja.po:
* po/nl.po:
* po/ru.po:
* win32/common/config.h:
0.10.28.3 pre-release

2011-04-26 15:58:12 +0200 Wim Taymans <[email protected]>

* gst/rtp/gstrtpgstpay.c:
rtpgstpay: fix buffer leak

2011-04-25 10:04:52 +0200 Philip Jägenstedt <[email protected]>

* ext/jpeg/gstjpegdec.c:
jpegdec: documentation typo "jpegddec"
https://bugzilla.gnome.org/show_bug.cgi?id=648589

2011-04-24 16:45:07 -0700 David Schleef <[email protected]>

* gst/avi/gstavimux.c:
* gst/matroska/matroska-mux.c:
avimux,matroskamux: Add stream-format to h264 caps
Fixes #606662.

2011-02-20 12:13:49 -0800 David Schleef <[email protected]>

* ext/libpng/gstpngdec.c:
pngdec: Remove temporary code
Now that we depend on (what will be) -base-0.10.33.

2011-04-24 14:03:56 +0100 Tim-Philipp Müller <[email protected]>

* configure.ac:
configure: don't pass -Waddress to ObjC compiler on OSX when compiling osxvideosink
Temporary workaround until we fix this properly and check for
the ObjC warning/error flags instead of just passing CFLAGS to the
ObjC compiler.
https://bugzilla.gnome.org/show_bug.cgi?id=643939

2011-04-24 13:29:32 +0100 Tim-Philipp Müller <[email protected]>

* docs/plugins/inspect/plugin-quicktime.xml:
* gst-plugins-good.spec.in:
* gst/quicktime/Makefile.am:
quicktime: rename plugin filename from *qtdemux* to *quicktime*
https://bugzilla.gnome.org/show_bug.cgi?id=648004

2011-04-24 14:03:41 +0100 Tim-Philipp Müller <[email protected]>

* common:
Automatic update of common submodule
From c3cafe1 to 46dfcea

2011-04-21 23:30:26 +0100 Tim-Philipp Müller <[email protected]>

* docs/plugins/Makefile.am:
* docs/plugins/gst-plugins-good-plugins-docs.sgml:
* docs/plugins/gst-plugins-good-plugins-sections.txt:
* gst/quicktime/Makefile.am:
* gst/quicktime/gstqtmoovrecover.c:
* gst/quicktime/gstqtmux-doc.c:
* gst/quicktime/gstqtmux-doc.h:
docs: add various qtmux variants to documentation

2011-04-21 22:51:52 +0100 Tim-Philipp Müller <[email protected]>

* gst/quicktime/gstqtmux.c:
* gst/quicktime/gstqtmuxmap.c:
* gst/quicktime/gstqtmuxmap.h:
quicktime: register 3gppmux element in addition to the misnamed gppmux

2011-04-18 18:08:30 -0400 Olivier Crête <[email protected]>

* gst/rtpmanager/gstrtpsession.c:
* gst/rtpmanager/rtpsession.c:
* gst/rtpmanager/rtpsession.h:
rtpsession: Remove incomplete support for RTCP FIR
Remove bits that were meant to suppport RTCP FIR
https://bugzilla.gnome.org/show_bug.cgi?id=648160

2011-04-19 14:33:25 +0100 Tim-Philipp Müller <[email protected]>

* tests/check/Makefile.am:
* tests/check/generic/.gitignore:
* tests/check/generic/index.c:
tests: add generic set_index test

2011-04-19 14:33:42 +0100 Tim-Philipp Müller <[email protected]>

* gst/flv/gstflvdemux.c:
flvdemux: fix deadlock on setting index on flvdemux

2011-04-19 14:16:11 +0100 Tim-Philipp Müller <[email protected]>

* tests/check/elements/flacparse.c:
tests: add index-setting test for baseparse/flacparse
https://bugzilla.gnome.org/show_bug.cgi?id=646811

2011-04-18 11:29:15 +0200 Sebastian Dröge <[email protected]>

* tests/check/pipelines/wavpack.c:
wavpack: Remove bus GSource to prevent a valgrind warning

2011-04-18 11:14:32 +0200 Sebastian Dröge <[email protected]>

* tests/check/pipelines/wavenc.c:
wavenc: Remove bus GSource to prevent a valgrind warning

2011-04-18 11:11:53 +0200 Sebastian Dröge <[email protected]>

* tests/check/pipelines/tagschecking.c:
tagschecking: Remove bus GSource to prevent a valgrind warning

2011-04-18 11:10:01 +0200 Sebastian Dröge <[email protected]>

* tests/check/elements/imagefreeze.c:
imagefreeze: Remove bus GSource to prevent a valgrind warning

2011-04-17 01:29:01 +0100 Tim-Philipp Müller <[email protected]>

* gst/avi/gstavidemux.c:
avidemux: fix 'variable may be used uninitialized' warnings caused by -DG_DISABLE_ASSERT

2011-04-16 18:50:11 +0100 Tim-Philipp Müller <[email protected]>

* configure.ac:
* win32/common/config.h:
* win32/common/gstrtpbin-marshal.c:
* win32/common/gstrtpbin-marshal.h:
0.10.28.2 pre-release

2011-04-16 18:49:27 +0100 Tim-Philipp Müller <[email protected]>

* gst/deinterlace/tvtime-dist.c:
* gst/deinterlace/tvtime-dist.h:
* gst/videobox/gstvideoboxorc-dist.c:
* gst/videobox/gstvideoboxorc-dist.h:
* gst/videomixer/blendorc-dist.c:
* gst/videomixer/blendorc-dist.h:
gst: update disted orc backup code

2011-04-16 18:29:45 +0100 Tim-Philipp Müller <[email protected]>

* docs/plugins/gst-plugins-good-plugins.args:
* docs/plugins/gst-plugins-good-plugins.hierarchy:
* docs/plugins/gst-plugins-good-plugins.interfaces:
* docs/plugins/gst-plugins-good-plugins.prerequisites:
* docs/plugins/inspect/plugin-1394.xml:
* docs/plugins/inspect/plugin-aasink.xml:
* docs/plugins/inspect/plugin-alaw.xml:
* docs/plugins/inspect/plugin-alpha.xml:
* docs/plugins/inspect/plugin-alphacolor.xml:
* docs/plugins/inspect/plugin-annodex.xml:
* docs/plugins/inspect/plugin-apetag.xml:
* docs/plugins/inspect/plugin-audiofx.xml:
* docs/plugins/inspect/plugin-audioparsers.xml:
* docs/plugins/inspect/plugin-auparse.xml:
* docs/plugins/inspect/plugin-autodetect.xml:
* docs/plugins/inspect/plugin-avi.xml:
* docs/plugins/inspect/plugin-cacasink.xml:
* docs/plugins/inspect/plugin-cairo.xml:
* docs/plugins/inspect/plugin-cutter.xml:
* docs/plugins/inspect/plugin-debug.xml:
* docs/plugins/inspect/plugin-deinterlace.xml:
* docs/plugins/inspect/plugin-dv.xml:
* docs/plugins/inspect/plugin-efence.xml:
* docs/plugins/inspect/plugin-effectv.xml:
* docs/plugins/inspect/plugin-equalizer.xml:
* docs/plugins/inspect/plugin-esdsink.xml:
* docs/plugins/inspect/plugin-flac.xml:
* docs/plugins/inspect/plugin-flv.xml:
* docs/plugins/inspect/plugin-flxdec.xml:
* docs/plugins/inspect/plugin-gconfelements.xml:
* docs/plugins/inspect/plugin-gdkpixbuf.xml:
* docs/plugins/inspect/plugin-goom.xml:
* docs/plugins/inspect/plugin-goom2k1.xml:
* docs/plugins/inspect/plugin-gstrtpmanager.xml:
* docs/plugins/inspect/plugin-halelements.xml:
* docs/plugins/inspect/plugin-icydemux.xml:
* docs/plugins/inspect/plugin-id3demux.xml:
* docs/plugins/inspect/plugin-imagefreeze.xml:
* docs/plugins/inspect/plugin-interleave.xml:
* docs/plugins/inspect/plugin-jack.xml:
* docs/plugins/inspect/plugin-jpeg.xml:
* docs/plugins/inspect/plugin-level.xml:
* docs/plugins/inspect/plugin-matroska.xml:
* docs/plugins/inspect/plugin-monoscope.xml:
* docs/plugins/inspect/plugin-mulaw.xml:
* docs/plugins/inspect/plugin-multifile.xml:
* docs/plugins/inspect/plugin-multipart.xml:
* docs/plugins/inspect/plugin-navigationtest.xml:
* docs/plugins/inspect/plugin-oss4.xml:
* docs/plugins/inspect/plugin-ossaudio.xml:
* docs/plugins/inspect/plugin-png.xml:
* docs/plugins/inspect/plugin-pulseaudio.xml:
* docs/plugins/inspect/plugin-quicktime.xml:
* docs/plugins/inspect/plugin-replaygain.xml:
* docs/plugins/inspect/plugin-rtp.xml:
* docs/plugins/inspect/plugin-rtsp.xml:
* docs/plugins/inspect/plugin-shapewipe.xml:
* docs/plugins/inspect/plugin-shout2send.xml:
* docs/plugins/inspect/plugin-smpte.xml:
* docs/plugins/inspect/plugin-soup.xml:
* docs/plugins/inspect/plugin-spectrum.xml:
* docs/plugins/inspect/plugin-speex.xml:
* docs/plugins/inspect/plugin-udp.xml:
* docs/plugins/inspect/plugin-video4linux2.xml:
* docs/plugins/inspect/plugin-videobox.xml:
* docs/plugins/inspect/plugin-videocrop.xml:
* docs/plugins/inspect/plugin-videofilter.xml:
* docs/plugins/inspect/plugin-videomixer.xml:
* docs/plugins/inspect/plugin-wavenc.xml:
* docs/plugins/inspect/plugin-wavpack.xml:
* docs/plugins/inspect/plugin-wavparse.xml:
* docs/plugins/inspect/plugin-ximagesrc.xml:
* docs/plugins/inspect/plugin-y4menc.xml:
docs: update for pre-release

2011-04-16 18:27:54 +0100 Tim-Philipp Müller <[email protected]>

* po/bg.po:
* po/cs.po:
* po/de.po:
* po/es.po:
* po/id.po:
* po/sl.po:
po: update translations

2011-04-16 18:17:01 +0100 Tim-Philipp Müller <[email protected]>

* gst/quicktime/gstqtmux.c:
qtmux: refuse incomplete legacy h264 caps
Refuse h264 caps without stream-format and codec_data fields for
now, to avoid creating broken files. This might cause some pipelines
that worked previously to fail. However, the move from -bad to -good
is our only chance to fix this up, so make it strict for now. We can
always change it back to be less strict in future.
https://bugzilla.gnome.org/show_bug.cgi?id=647919

2011-04-16 18:16:11 +0100 Tim-Philipp Müller <[email protected]>

* sys/v4l2/gstv4l2sink.c:
v4l2sink: fix another unused-but-set-variable warning

2011-04-16 18:10:24 +0100 Tim-Philipp Müller <[email protected]>

* ext/pulse/pulsesink.c:
* ext/pulse/pulsesrc.c:
* ext/speex/gstspeexenc.c:
* gst/rtp/gstrtpgsmpay.c:
pulse, speexenc, rtpgsmpay: don't use g_assert() for error handling
Don't use g_assert() for error handling, even if they're highly unlikely.
Either we *know* that something can't happen, in which case we
should just not handle it, or we think something can happen, but it is
very very unlikely that it will ever happen, in which case we should
handle it like any other error instead of asserting.
g_assert() is best left for conditions we have control of, like checking
internal consistency of our code, not checking return values of external
code.
Fixes a bunch of warnings when compiling with -DG_DISABLE_ASSERT:
gstrtpgsmpay.c: In function 'gst_rtp_gsm_pay_handle_buffer':
gstrtpgsmpay.c:130:17: warning: variable 'rtpgsmpay' set but not used
gstspeexenc.c: In function 'gst_speex_enc_encode':
gstspeexenc.c:904:19: warning: variable 'written' set but not used
pulsesink.c: In function 'gst_pulsesink_change_state':
pulsesink.c:2725:9: warning: variable 'res' set but not used
pulsesrc.c: In function 'gst_pulsesrc_change_state':
pulsesrc.c:1253:7: warning: variable 'e' set but not used

2011-04-16 18:07:35 +0100 Tim-Philipp Müller <[email protected]>

* tests/examples/rtp/server-alsasrc-PCMA.c:
examples: fix some warnings in rtp example
Caused by -DG_DISABLE_ASSERT

2011-04-16 17:57:32 +0100 Tim-Philipp Müller <[email protected]>

* tests/examples/level/level-example.c:
examples: don't put code with side-effects into g_assert()
Otherwise things won't work too well when compiling with
-DG_DISABLE_ASSERT (as we do for pre-releases and releases).

2011-04-16 16:51:32 +0100 Tim-Philipp Müller <[email protected]>

* gst/deinterlace/tvtime/greedyh.c:
* gst/matroska/matroska-mux.c:
deinterlace, matroska: fix two variable-may-be-used-uninitialized compiler warnings
We use -DG_DISABLE_ASSERT for the pre-releases, which makes these
warnings pop up in cases that were previously covered by g_assert_not_reached()
and the like:
tvtime/greedyh.c:801:14: warning: 'scanline' may be used uninitialized in this function
matroska-mux.c:501:19: warning: 'context' may be used uninitialized in this function

2011-04-16 13:33:45 +0100 Tim-Philipp Müller <[email protected]>

* ext/jack/gstjackaudiosink.c:
* ext/jack/gstjackaudiosrc.c:
jack: fix unused-but-set-variable warnings with gcc-4.6

2011-04-16 13:23:50 +0100 Tim-Philipp Müller <[email protected]>

* tests/examples/cairo/cairo_overlay.c:
examples: fix 'control reaches end of non-void function' warning in cairo example

2011-04-15 15:47:24 +0200 Robert Swain <[email protected]>

* sys/v4l2/gstv4l2src.c:
v4l2src: Address unused but set variable
The v4l2object formats list was being obtained into a local variable and
then still used from the context. Make use of the local variable.

2011-04-15 15:17:34 +0200 Robert Swain <[email protected]>

* sys/oss4/oss4-mixer-slider.c:
* sys/oss4/oss4-mixer-switch.c:
* sys/oss4/oss4-property-probe.c:
* sys/oss4/oss4-source.c:
oss4: Address unused but set variables
GCC 4.6.x complains about such variable usage. Unused but set variables
were removed except that gst_oss4_mixer_slider_set_mute () now returns
the value from the call to gst_oss4_mixer_set_control_val ().

2011-04-15 15:14:13 +0200 Robert Swain <[email protected]>

* ext/jpeg/gstjpegenc.c:
* ext/pulse/pulsesink.c:
* ext/raw1394/gstdv1394src.c:
* ext/raw1394/gsthdv1394src.c:
jpegenc: pulsesink: raw1394: Address unused but set variables
GCC 4.6.x spits warnings about such usage of variables. The variables in
raw1394 were marked with G_GNUC_UNUSED as this seemed omre appropriate.
The others were removed.

2011-04-15 15:12:44 +0200 Robert Swain <[email protected]>

* gst/shapewipe/gstshapewipe.c:
* gst/y4m/gsty4mencode.c:
y4mencode: shapewipe: Address unused but set variables
GCC 4.6.x complains about such usage.

2011-04-15 15:11:35 +0200 Robert Swain <[email protected]>

* tests/check/elements/deinterlace.c:
* tests/check/elements/rtp-payloading.c:
* tests/check/pipelines/flacdec.c:
* tests/examples/level/level-example.c:
* tests/icles/videocrop-test.c:
* tests/icles/ximagesrc-test.c:
tests: Address unused but set variables
GCC 4.6.x spits warnings about such usage of variables.

2011-04-15 15:36:41 +0200 Robert Swain <[email protected]>

* gst/videomixer/blendorc.orc:
videomixer: Fix argb/rgba overlay orc code
Remove some redundant operations (convubw) and use the correct variable,
t2, in the orc_overlay_bgra function.

2011-04-15 15:33:35 +0200 Robert Swain <[email protected]>

* gst/videomixer/blend.c:
* gst/videomixer/gstcollectpads2.c:
* gst/videomixer/videomixer2.c:
videomixer: address unused but set variables
GCC 4.6.x spits warnings about variables that are set but unused. Such
variables have been removed in blend, collectpads2 and videomixer2.

2011-04-15 14:57:20 +0200 Robert Swain <[email protected]>

* gst/rtp/gstrtpamrdepay.c:
* gst/rtp/gstrtpbvdepay.c:
* gst/rtp/gstrtpbvpay.c:
* gst/rtp/gstrtpg722pay.c:
* gst/rtp/gstrtpgstdepay.c:
* gst/rtp/gstrtpgstpay.c:
* gst/rtp/gstrtpj2kpay.c:
* gst/rtp/gstrtpmp4gpay.c:
* gst/rtp/gstrtpmp4vpay.c:
* gst/rtp/gstrtpmpadepay.c:
* gst/rtp/gstrtpqcelpdepay.c:
* gst/rtpmanager/gstrtpjitterbuffer.c:
* gst/rtpmanager/gstrtpsession.c:
rtp, rtpmanager: Address unused but set variables
GCC 4.6.x spits warnings about variables that are unused but set. Such
variables have been removed where trivial but with comments left behind
for informational purposes in some cases.
gst_rtp_session_chain_recv_rtcp () was changed in commit 490113d4
to always return GST_FLOW_OK instead of the return value of
rtp_session_process_rtcp (), so we'll keep it that way.

2011-04-15 11:29:30 +0200 Robert Swain <[email protected]>

* gst/quicktime/descriptors.c:
* gst/quicktime/gstrtpxqtdepay.c:
* gst/quicktime/qtdemux.c:
quicktime: Remove unused but set variables
GCC 4.6.x spits warnings about such variable usage. Note that some
calculations are left as comments for informative purposes.

2011-04-15 11:23:38 +0200 Robert Swain <[email protected]>

* gst/matroska/matroska-demux.c:
* gst/matroska/matroska-parse.c:
matroska: Remove unused but set variables
GCC 4.6.x spits warnings about such variable usage.

2011-04-15 11:19:26 +0200 Robert Swain <[email protected]>

* gst/imagefreeze/gstimagefreeze.c:
imagefreeze: Remove unused but set duration variable
GCC 4.6.x spits warnings about such variable usage.

2011-04-15 11:18:19 +0200 Robert Swain <[email protected]>

* gst/flv/gstflvdemux.c:
flxdemux: Remove unused but set keyframe variables
The FIXMEs about the keyframe flag never being used are left for later
fixing, at which point the keyframe variables could be added back.

2011-04-15 11:16:42 +0200 Robert Swain <[email protected]>

* gst/effectv/gstedge.c:
edgetv: Remove unused but set height variable
GCC 4.6.x spits warnings about such variables.

2011-04-15 18:51:20 +0100 Tim-Philipp Müller <[email protected]>

* gst/audioparsers/gstflacparse.c:
flacparse: update for gst_base_parse_frame_init() API change

2011-02-01 15:57:01 -0500 Olivier Crête <[email protected]>

* gst/rtpmanager/rtpsession.c:
rtpsession: Use existing functions to parse RTCP FB packets
Use existing functions to get the FCI from FB packets.
https://bugzilla.gnome.org/show_bug.cgi?id=622553

2011-02-01 16:23:52 -0500 Olivier Crête <[email protected]>

* gst/rtpmanager/gstrtpbin-marshal.list:
* gst/rtpmanager/rtpsession.c:
rtpsession: marshal GstBuffer as a MiniObject instead of a pointer
https://bugzilla.gnome.org/show_bug.cgi?id=622553

2011-04-14 23:24:56 -0700 David Schleef <[email protected]>

* gst/matroska/matroska-demux.c:
matroskademux: Better calculation of framerate
https://bugzilla.gnome.org/show_bug.cgi?id=647833

2011-04-13 12:37:09 +0100 Tim-Philipp Müller <[email protected]>

* gst/quicktime/gstqtmux.c:
qtmux: default to dts-method=reorder and presentation-time=true
https://bugzilla.gnome.org/show_bug.cgi?id=636699

2011-04-15 12:47:52 +0200 Mark Nauwelaerts <[email protected]>

* tests/check/elements/qtmux.c:
tests: qtmux: test various dts-methods

2011-04-15 12:34:05 +0200 Mark Nauwelaerts <[email protected]>

* gst/quicktime/gstqtmux.c:
qtmux: fix corner case buffer handling for reorder method

2011-04-14 13:47:05 +0200 Sebastian Dröge <[email protected]>

* gst/flv/gstflvdemux.c:
flvdemux: Don't leak the SEEKING query

2011-04-14 13:43:06 +0200 Sebastian Dröge <[email protected]>

* gst/quicktime/gstqtmoovrecover.c:
* gst/quicktime/gstqtmoovrecover.h:
qtmoovrecover: Don't leak the static recursive mutex

2011-04-14 13:37:52 +0200 Sebastian Dröge <[email protected]>

* sys/v4l2/gstv4l2radio.c:
v4l2radio: Free videodev string before replacing it

2011-04-14 13:24:21 +0200 Sebastian Dröge <[email protected]>

* gst/matroska/matroska-parse.c:
matroskaparse: Allow webm and matroska caps and don't leak caps

2011-04-14 07:35:29 +0100 Christian Fredrik Kalager Schaller <[email protected]>

* gst-plugins-good.spec.in:
Add parser plugin

2011-03-24 14:34:24 -0700 David Schleef <[email protected]>

* sys/directsound/gstdirectsoundsink.c:
directsoundsink: Add conditionals on WAVE_FORMAT_DOLBY_AC3_SPDIF

2011-04-11 20:09:14 +0100 Tim-Philipp Müller <[email protected]>

* gst/debugutils/gstcapsdebug.c:
capsdebug: fix unused-but-set-variable warnings with gcc 4.6

2011-04-11 20:05:54 +0100 Tim-Philipp Müller <[email protected]>

* gst/avi/gstavidemux.c:
avidemux: fix unused-but-set-variable warning with gcc 4.6
Most likely a leftover from when the index parsing code was rewritten.

2011-04-11 19:54:00 +0100 Tim-Philipp Müller <[email protected]>

* gst/audioparsers/gstac3parse.c:
ac3parse: fix unused-but-set-variable warning with gcc 4.6

2011-04-11 19:50:07 +0100 Tim-Philipp Müller <[email protected]>

* gst/videofilter/gstvideobalance.c:
videobalance: fix handling of YUV images with 'odd' widths
Fixes unused-but-set-variable warnings with gcc 4.6.

2011-04-11 19:49:22 +0100 Tim-Philipp Müller <[email protected]>

* gst/videofilter/gstvideoflip.c:
videoflip: fix unused-but-set-variable warnings with gcc 4.6

2011-04-13 18:11:34 +0200 Sebastian Dröge <[email protected]>

* gst/audiofx/audiowsincband.c:
* gst/audiofx/audiowsinclimit.c:
audiowsinc{band,limit}: Fix check for divison by zero

2011-04-13 18:01:01 +0200 Sebastian Dröge <[email protected]>

* gst/audiofx/audiowsincband.c:
audiowsincband: Fix range of kernel elements (lim -> lim-1)

2011-04-13 18:00:44 +0200 Sebastian Dröge <[email protected]>

* gst/audiofx/audiowsinclimit.c:
audiowsinclimit: Add some more braces to make the code more readable

2011-04-11 18:40:30 -0500 Jordi Burguet-Castell <[email protected]>

* gst/audiofx/audiowsinclimit.c:
audiowsinclimit: Fix range of kernel elements (lim -> lim-1) in high/low-pass filters

2011-04-13 17:49:22 +0200 Sebastian Dröge <[email protected]>

* gst/audiofx/audiowsincband.c:
audiowsincband: Add new windowing functions: gaussian, cos and hann

2011-04-11 18:41:43 -0500 Jordi Burguet-Castell <[email protected]>

* gst/audiofx/audiowsinclimit.c:
audiowsinclimimt: Add new windows to high/low-pass filters: gaussian, cosine, hann

2011-04-13 16:47:05 +0100 Tim-Philipp Müller <[email protected]>

* gst/matroska/matroska-demux.c:
matroskademux: set stream-format=byte-stream on h264 caps if there's no codec data
https://bugzilla.gnome.org/show_bug.cgi?id=606662

2011-04-13 16:37:07 +0100 Thiago Santos <[email protected]>

* gst/quicktime/gstqtmux.c:
* gst/quicktime/gstqtmuxmap.c:
qtmux: restrict h264 some more to only accept AU-aligned AVC
https://bugzilla.gnome.org/show_bug.cgi?id=606662

2011-04-13 17:11:26 +0200 Sebastian Dröge <[email protected]>

* gst/audioparsers/gstmpegaudioparse.c:
mpegaudioparse: The VBRI header is always at offset 0x20, independent of MPEG version
Also clean up advancing of the data pointer a bit.
Fixes bug #647659.

2011-04-13 15:18:11 +0100 Tim-Philipp Müller <[email protected]>

* gst/quicktime/gstqtmux.c:
* gst/quicktime/gstqtmuxmap.c:
* tests/check/Makefile.am:
* tests/check/elements/qtmux.c:
qtmux: add variant-less video/quicktime to source pad template caps
This is needed for automatic transcoding using encodebin. Our typefinder
does not always add a variant to the found caps, and encodebin needs
an *exact* match to the caps on the source pad template, so we need
to add the variant-less video/quicktime caps to the template as well
for encodebin to be able to find it. Add unit test for this as well.
https://bugzilla.gnome.org/show_bug.cgi?id=642879

2011-04-13 16:17:41 +0200 Sebastian Dröge <[email protected]>

* ext/flac/gstflacenc.c:
flacenc: Properly interprete the result of strcmp()

2011-04-13 16:09:04 +0200 Sebastian Dröge <[email protected]>

* ext/flac/gstflacenc.c:
flacenc: Don't store image tags inside the vorbiscomments and the flac metadata
Instead only store them inside the flac metadata. There's
no point in storing them twice and the flac metadata is
still the official way to store image tags inside flac.

2011-04-13 12:38:15 +0100 Tim-Philipp Müller <[email protected]>

* tests/check/elements/.gitignore:
* tests/check/pipelines/.gitignore:
tests: ignore new qtmux-related test binaries

2011-04-13 11:25:11 +0100 Tim-Philipp Müller <[email protected]>

* docs/plugins/Makefile.am:
* docs/plugins/gst-plugins-good-plugins-docs.sgml:
* docs/plugins/gst-plugins-good-plugins-sections.txt:
* docs/plugins/inspect/plugin-quicktime.xml:
* gst/quicktime/Makefile.am:
* gst/quicktime/gstqtmuxplugin.c:
* gst/quicktime/quicktime.c:
* tests/check/Makefile.am:
quicktime: move qtmux plugin from -bad to -good
https://bugzilla.gnome.org/show_bug.cgi?id=636699

2011-04-04 12:21:23 +0200 Mark Nauwelaerts <[email protected]>

* gst/quicktime/gstqtmux.c:
qtmux: more helpful debug error message when no needed duration on input buffers
Fixes #646256.

2011-03-21 10:56:51 -0300 Thiago Santos <[email protected]>

* gst/quicktime/atoms.c:
* gst/quicktime/atoms.h:
* gst/quicktime/gstqtmux.c:
qtmux: Adding GstTagXmpWriter interface
Adds GstTagXmpWriter interface support to qtmux

2011-03-22 20:53:08 +0100 Mark Nauwelaerts <[email protected]>

* gst/quicktime/gstqtmux.c:
qtmux: use running time for synchronization
See also #432612.

2011-03-10 16:03:58 +0100 Mark Nauwelaerts <[email protected]>

* gst/quicktime/gstqtmux.c:
qtmux: provide for PTS metadata when so configured
... and not only when sort-of feeling like it.
In any case, if it turns out all really is in order,
and presumably DTS == PTS, then no ctts will be produced anyway.

2011-03-10 16:02:42 +0100 Mark Nauwelaerts <[email protected]>

* gst/quicktime/gstqtmux.c:
qtmux: also track original PTS buffer timestamp in reorder dts-method

2011-02-21 12:14:59 +0100 Edward Hervey <[email protected]>

* gst/quicktime/gstqtmux.c:
Revert "Check that collectpads exists before removing pad"
This reverts commit 6d8740476ccd3a3498dc4f18c19733643825c7b8.
Depends on a core commit that was reverted

2011-02-20 23:57:19 -0800 David Schleef <[email protected]>

* gst/quicktime/gstqtmux.c:
Check that collectpads exists before removing pad
The core now calls release pad from finalize, at which point
the collectpads might have already been freed.

2011-01-13 11:28:32 -0300 Thiago Santos <[email protected]>

* tests/check/elements/qtmux.c:
test: qtmux: Tests qtmux reuse
Forces the use of qtmux after it has been put to PLAYING and back
to NULL once
https://bugzilla.gnome.org/show_bug.cgi?id=639338

2011-01-13 15:27:36 +0100 Mark Nauwelaerts <[email protected]>

* gst/quicktime/gstqtmux.c:
qtmux: set src pads when starting file
... rather than at _init time, so they are also available following a
pad (de)activation cycle.
https://bugzilla.gnome.org/show_bug.cgi?id=639338

2011-01-03 17:24:23 +0100 Mark Nauwelaerts <[email protected]>

* gst/quicktime/gstqtmux.c:
* gst/quicktime/gstqtmux.h:
qtmux: adjust nasty case timestamp tracking
That is, all sorts of problems arise with re-ordered input timestamps that
tend to defy automagic handling for every case, so allow for a few variations
that can be tried depending on circumstances.
Also try to document accordingly.
Also fixes #638288.

2010-12-30 21:48:41 +0200 Felipe Contreras <[email protected]>

* gst/quicktime/gstqtmux.c:
qtmux: get rid of timestamp overprotectiveness
Signed-off-by: Felipe Contreras <[email protected]>

2011-01-03 16:56:57 +0100 Mark Nauwelaerts <[email protected]>

* gst/quicktime/atoms.c:
* gst/quicktime/atoms.h:
* gst/quicktime/atomsrecovery.c:
* gst/quicktime/gstqtmux.c:
qtmux: simplify and fix pts_offset storing
In particular, only write a ctts atom if and only if ever a non-zero offset.

2011-01-03 10:43:15 +0100 Mark Nauwelaerts <[email protected]>

* gst/quicktime/gstqtmux.c:
qtmux: add some more documentation

2010-12-03 15:23:00 +0100 Mark Nauwelaerts <[email protected]>

* gst/quicktime/atoms.c:
* gst/quicktime/atoms.h:
* gst/quicktime/gstqtmux.c:
* gst/quicktime/gstqtmux.h:
qtmux: remove large-file property
Rather, auto-determine if 64-bits fields are needed for a valid result, and
stick to plain 32-bits if not needed.
API: GstQTMux:large-file (removed)

2010-12-19 12:53:34 +0100 Sebastian Dröge <[email protected]>

* gst/quicktime/gstqtmux.c:
qtmux: Free AtomInfo structs

2010-12-19 12:50:30 +0100 Sebastian Dröge <[email protected]>

* gst/quicktime/gstqtmux.c:
qtmux: Free tag string after use

2010-12-19 12:12:25 +0100 Sebastian Dröge <[email protected]>

* tests/check/pipelines/tagschecking.c:
tagschecking: Fix some more memory leaks

2010-12-17 19:41:25 +0200 Lasse Laukkanen <[email protected]>

* gst/quicktime/gstqtmux.c:
qtmux: allow zero duration tracks

2010-12-03 18:09:41 +0100 Mark Nauwelaerts <[email protected]>

* gst/quicktime/gstqtmux.c:
qtmux: add documentation

2010-12-01 10:45:49 +0100 David Hoyt <[email protected]>

* gst/quicktime/gstqtmux.c:
qtmux: handle msvc ftruncate incompatibility
Fixes #636185.

2010-11-27 16:07:19 -0600 Alejandro Gonzalez <[email protected]>

* gst/quicktime/gstqtmux.c:
qtmux: gst_qtmux_check_difference verify before subtract
Avoid negative overflow by checking the order of operands
on subtraction of unsigned integers.
https://bugzilla.gnome.org/show_bug.cgi?id=635878

2010-11-19 17:55:36 +0100 Mark Nauwelaerts <[email protected]>

* gst/quicktime/gstqtmux.c:
qtmux: remove remnant of obsolete property

2010-11-19 15:18:58 +0100 Mark Nauwelaerts <[email protected]>

* tests/check/elements/qtmux.c:
tests: qtmux: also unit test fragmented file cases

2010-07-30 12:48:29 +0200 Marc-André Lureau <[email protected]>

* gst/quicktime/gstqtmux.c:
* gst/quicktime/gstqtmux.h:
qtmux: allow specifying trak timescale
This is mainly because Smoothstreaming client are broken and don't
take the TimeScale property into account.

2010-11-19 17:41:41 +0100 Mark Nauwelaerts <[email protected]>

* gst/quicktime/atoms.c:
* gst/quicktime/atoms.h:
* gst/quicktime/gstqtmux.c:
qtmux: include sdtp atoms for ismv fragmented files
Based on patch by Marc-André Lureau <[email protected]>

2010-11-19 19:17:45 +0100 Mark Nauwelaerts <[email protected]>

* gst/quicktime/gstqtmux.c:
qtmux: enable default fragmented file for ismlmux

2010-09-02 13:58:05 +0200 Marc-André Lureau <[email protected]>

* gst/quicktime/atoms.h:
* gst/quicktime/ftypcc.h:
* gst/quicktime/gstqtmuxmap.c:
* gst/quicktime/gstqtmuxmap.h:
qtmux: add ismlmux, for fragmented isml major brand

2010-11-19 14:44:45 +0100 Mark Nauwelaerts <[email protected]>

* gst/quicktime/gstqtmux.c:
qtmux: finalize sinkpads list

2010-07-22 19:40:07 +0200 Marc-André Lureau <[email protected]>

* gst/quicktime/gstqtmux.c:
qtmux: add moov in streamheader

2010-08-06 13:26:27 +0200 Marc-André Lureau <[email protected]>

* gst/quicktime/gstqtmux.c:
* gst/quicktime/gstqtmux.h:
qtmux: add streamable property to avoid building fragmented mfra index

2010-11-18 16:48:06 +0100 Mark Nauwelaerts <[email protected]>

* gst/quicktime/atoms.c:
* gst/quicktime/atoms.h:
* gst/quicktime/gstqtmux.c:
* gst/quicktime/gstqtmux.h:
qtmux: add mfra to fragmented file
Based on patch by Marc-André Lureau <[email protected]>

2010-11-15 15:17:59 +0100 Mark Nauwelaerts <[email protected]>

* gst/quicktime/atoms.c:
* gst/quicktime/atoms.h:
* gst/quicktime/gstqtmux.c:
* gst/quicktime/gstqtmux.h:
qtmux: optionally create fragmented file
In this mode, an initial empty moov (containing only stream metadata) is written,
followed by fragments containing actual data (along with required metadata).
New fragments are started either at keyframe (if such are sparse) or when
property configured duration exceeded.
Based on patch by Marc-André Lureau <[email protected]>
Fixes #632911.

2010-11-15 15:12:45 +0100 Mark Nauwelaerts <[email protected]>

* gst/quicktime/atoms.c:
qtmux: use helper to set atom flags from given uint

2010-11-09 16:49:07 +0100 Mark Nauwelaerts <[email protected]>

* gst/quicktime/gstqtmux.c:
qtmux: refactor configuring and sending of moov
Based on patch by Marc-André Lureau <[email protected]>

2010-11-09 15:54:44 +0100 Mark Nauwelaerts <[email protected]>

* gst/quicktime/gstqtmux.c:
qtmux: refactor extra top-level atom handling
Also check a bit more for possible errors, and free proper items in such case.

2010-11-09 15:01:15 +0100 Mark Nauwelaerts <[email protected]>

* gst/quicktime/gstqtmux.c:
qtmux: refactor slightly using buffer helper

2010-11-05 13:48:57 +0100 Mark Nauwelaerts <[email protected]>

* gst/quicktime/gstqtmux.c:
qtmux: fix misinforming comment

2010-11-05 12:08:15 +0100 Mark Nauwelaerts <[email protected]>

* gst/quicktime/atoms.c:
* gst/quicktime/atoms.h:
* gst/quicktime/gstqtmux.c:
qtmux: delegate mvex handling to atoms
... which keeps qtmux simpler.

2009-09-28 16:11:35 +0200 Marc-André Lureau <[email protected]>

* gst/quicktime/atoms.c:
* gst/quicktime/atoms.h:
* gst/quicktime/gstqtmux.c:
qtmux: add mvex/trex in header if fragmented
One "trex" is added per "trak". We don't support default values,
but the "trex" box is mandatory.

2009-09-28 13:01:30 +0200 Marc-André Lureau <[email protected]>

* gst/quicktime/fourcc.h:
qtmux: add a couple of fourcc for fragmented mp4

2010-11-05 11:08:01 +0100 Mark Nauwelaerts <[email protected]>

* gst/quicktime/gstqtmux.c:
qtmux: avoid removing temp file when error occurred

2009-09-30 17:16:30 +0200 Marc-André Lureau <[email protected]>

* gst/quicktime/gstqtmux.c:
qtmux: truncate buffer file after each send

2009-09-28 16:53:51 +0200 Marc-André Lureau <[email protected]>

* gst/quicktime/gstqtmux.c:
qtmux: remove temp file when reset/finalize

2010-10-19 13:43:14 +0300 Stefan Kost <[email protected]>

* gst/quicktime/gstqtmoovrecover.c:
various (gst): add missing G_PARAM_STATIC_STRINGS flags
Canonicalize property names as needed.

2010-10-13 17:47:29 +0200 Mark Nauwelaerts <[email protected]>

* gst/quicktime/gstqtmux.c:
qtmux: prevent infinite loop when adjusting framerate
Fixes #632070.

2010-10-03 23:45:46 -0300 Thiago Santos <[email protected]>

* gst/quicktime/gstqtmux.c:
qtmux: Add G_PARAM_STATIC_STRINGS
Add G_PARAM_STATIC_STRINGS to qtmux properties

2010-09-15 17:54:49 -0300 Thiago Santos <[email protected]>

* gst/quicktime/atoms.c:
* gst/quicktime/atoms.h:
* gst/quicktime/fourcc.h:
* gst/quicktime/gstqtmux.c:
* gst/quicktime/gstqtmux.h:
qtmux: Follow xmp serialization guidelines closer
qt and isom variants have different ways of serializing
xmp, follow these guidelines.
Those can be found in Adobe's xmp docs.

2010-08-16 12:36:24 +0200 Mark Nauwelaerts <[email protected]>

* gst/quicktime/gstqtmux.c:
qtmux: autodetect out-of-order input timestamps and determine DTS accordingly
Favour using input buffer timestamps for DTS, but fallback to using buffer
duration (accumulation) if input ts detected out-of-order.
Fixes #624212.

2010-07-28 16:15:53 +0200 Marc-André Lureau <[email protected]>

* gst/quicktime/gstqtmux.c:
qtmux: use caps bitrate at last chance
If we didn't get the stream's bitrate from one of the atoms,
try getting it from the caps as a last resort.
https://bugzilla.gnome.org/show_bug.cgi?id=625496

2010-07-28 16:12:11 +0200 Marc-André Lureau <[email protected]>

* gst/quicktime/atoms.c:
qtmux: btrt - max bitrate before average
According to iso base media file format, the max bitrate
is before the avg
https://bugzilla.gnome.org/show_bug.cgi?id=625496

2010-07-06 14:48:08 +0530 Arun Raghavan <[email protected]>

* gst/quicktime/atoms.c:
* gst/quicktime/atoms.h:
* gst/quicktime/gstqtmux.c:
qtmux: Write 'btrt' atom for H.264 media if possible
This writes out the optional 'btrt' atom (MPEG4BitrateBox) for H.264
media if either or both of average and maximum bitrate are available for
the stream.
https://bugzilla.gnome.org/show_bug.cgi?id=623678

2010-07-05 14:09:50 +0530 Arun Raghavan <[email protected]>

* gst/quicktime/atoms.c:
* gst/quicktime/atoms.h:
* gst/quicktime/gstqtmux.c:
* gst/quicktime/gstqtmux.h:
qtmux: Write avg/max bitrate to ESDS if available
This collects the 'bitrate' and 'maximum-bitrate' tags on the
corresponding pad and uses these to populate these fields in the ESDS
where applicable.
https://bugzilla.gnome.org/show_bug.cgi?id=623678

2010-07-02 12:45:20 +0200 Edward Hervey <[email protected]>

* gst/quicktime/gstqtmux.c:
qtmux: Don't use bogus codec/format tags
https://bugzilla.gnome.org/show_bug.cgi?id=623365

2010-06-25 20:19:20 -0300 Thiago Santos <[email protected]>

* gst/quicktime/gstqtmux.c:
qtmux: Write uint tags that don't have a complement
Write uint tags that have complements (e.g. track-number/
track-count) even when we only have one of them available
and set the other one to 0.
Fixes #622484

2010-06-21 19:39:54 +0200 Edward Hervey <[email protected]>

* gst/quicktime/gstqtmux.c:
qtmux: Remove the pad from our internal list before calling collectpads
Previously we would end up with the collectpaddata structure already freed.
This would result in a bogus iteration of mux->sinkpads (all the
GstQTPad being freed) and it wouldn't be removed from that list.
Finally, due to it not being removed from that list, we would end up
calling a bogus gst_qt_mux_pad_reset on those structures => SEGFAULT

2010-05-12 18:50:34 -0700 David Schleef <[email protected]>

* gst/quicktime/fourcc.h:
* gst/quicktime/gstqtmux.c:
* gst/quicktime/gstqtmuxmap.c:
qtmux: Add VP8

2010-05-11 13:15:37 +0100 Tim-Philipp Müller <[email protected]>

* tests/check/pipelines/tagschecking.c:
tests: don't fail tagschecking test if qtdemux is not available or too old

2010-03-27 09:46:30 +0000 Tim-Philipp Müller <[email protected]>

* gst/quicktime/gstqtmuxplugin.c:
qtmux: use GStreamer package name and origin in the plugin info

2010-03-23 17:34:30 -0300 Thiago Santos <[email protected]>

* tests/check/pipelines/tagschecking.c:
tests: tagschecking: New tags tests
Adds new tags checking tests.

2010-03-25 00:20:54 +0000 Tim-Philipp Müller <[email protected]>

* gst/quicktime/gstqtmux.c:
qtmux: init debug category before using it

2010-03-22 16:56:03 +0100 Benjamin Otte <[email protected]>

* gst/quicktime/atoms.c:
Add -Wold-style-definition
and fix the warnings

2010-03-22 13:16:33 +0100 Benjamin Otte <[email protected]>

* gst/quicktime/atoms.c:
* gst/quicktime/gstqtmuxmap.h:
* tests/check/elements/qtmux.c:
Add -Wwrite-strings
and fix its warnings

2010-03-21 21:39:18 +0100 Benjamin Otte <[email protected]>

* gst/quicktime/atoms.c:
* gst/quicktime/atoms.h:
* gst/quicktime/atomsrecovery.c:
* gst/quicktime/descriptors.c:
* tests/check/elements/qtmux.c:
* tests/check/pipelines/tagschecking.c:
Add -Wmissing-declarations -Wmissing-prototypes to configure flags
And fix all warnings

2010-03-18 17:30:26 +0100 Benjamin Otte <[email protected]>

* gst/quicktime/gstqtmoovrecover.c:
* gst/quicktime/gstqtmux.c:
gst_element_class_set_details => gst_element_class_set_details_simple

2010-03-12 11:28:51 -0300 Thiago Santos <[email protected]>

* tests/check/pipelines/tagschecking.c:
tests: tagschecking: Improvements and new geo-location tests
Makes some improvements to tagschecking.c, making it use
fakesrc instead of videotestsrc and allowing to set input
caps so that more muxers can be used. Previously we could
only use those that accepted raw video caps.
Also adds some tests for geo-location tags

2010-03-12 10:53:36 -0300 Thiago Santos <[email protected]>

* gst/quicktime/gstqtmux.c:
qtmux: Use xmp on mp4mux and gppmux too
Do not restrict xmp to qtmux, but use it too
on mp4mux and gppmux

2010-03-05 13:33:37 -0300 Thiago Santos <[email protected]>

* tests/check/pipelines/tagschecking.c:
check: tagschecking: tests for tags serialization in muxers
Adds a check unit test that aims to test tags serialization
and deserialization consistency (in muxers). It provides a
basic function that allows one to easily specify tags, a
muxer and a demuxer and a test will be done to check if
the tags have been consistently muxed and demuxed

2010-02-22 16:45:34 -0300 Thiago Santos <[email protected]>

* gst/quicktime/atoms.c:
* gst/quicktime/atoms.h:
* gst/quicktime/fourcc.h:
* gst/quicktime/gstqtmux.c:
qtmux: add xmp support
Adds xmp metatags adding to qtmux.
Fixes #609539

2010-03-11 17:17:15 +0000 Tim-Philipp Müller <[email protected]>

* gst/quicktime/gstqtmoovrecover.c:
qtmux: fix GST_ELEMENT_ERROR usage
We need to pass (NULL) rather than NULL for empty arguments.

2010-03-10 10:23:23 -0600 Rob Clark <[email protected]>

* gst/quicktime/gstqtmoovrecover.c:
qtmux: fix compile error
gst/quicktime/gstqtmoovrecover.c:268: warning: format not a string literal and no format arguments
https://bugzilla.gnome.org/show_bug.cgi?id=612454

2010-02-22 19:38:15 -0300 Thiago Santos <[email protected]>

* gst/quicktime/gstqtmuxmap.c:
qtmux: Rename 'avc-sample' to 'avc' in caps
Fixes #606662

2010-02-26 11:50:25 -0800 Michael Smith <[email protected]>

* gst/quicktime/gstqtmux.c:
qtmux: Take lock around use of (non-threadsafe) tagsetter interface.

2010-02-22 16:51:00 -0300 Thiago Santos <[email protected]>

* gst/quicktime/atoms.c:
qtmux: write all udta children atoms
UDTA might have META and other children atoms
together, write them all.

2010-02-22 10:48:11 -0300 Thiago Santos <[email protected]>

* gst/quicktime/gstqtmux.c:
* gst/quicktime/gstqtmux.h:
qtmux: Use internal sink pads list
Due to GstCollectPads sink pads list being not reliably
iteratable (when not inside the collected function) this
patch adds a sink pads list to qtmux to be used when iterating
sink pads on reset function.
Fixes #609055

2010-02-16 17:13:09 +0100 Mark Nauwelaerts <[email protected]>

* gst/quicktime/atoms.c:
qtmux: prevent leaking hdlr name

2010-02-16 16:24:12 +0100 Mark Nauwelaerts <[email protected]>

* gst/quicktime/atoms.c:
* gst/quicktime/atoms.h:
* gst/quicktime/gstqtmux.c:
* gst/quicktime/gstqtmuxmap.c:
qtmux: support for ALAC
Fixes #580731.

2010-02-16 14:19:04 +0100 Mark Nauwelaerts <[email protected]>

* gst/quicktime/atoms.c:
qtmux: refactor building stsd entry 'wave' extension

2010-02-08 11:51:52 -0300 Thiago Santos <[email protected]>

* gst/quicktime/atomsrecovery.c:
qtmux: atomsrecovery: Fix compilation problem
Fixes a compilation error due to unused function result.

2009-12-12 16:07:15 -0300 Thiago Santos <[email protected]>

* gst/quicktime/atoms.c:
* gst/quicktime/atoms.h:
* gst/quicktime/atomsrecovery.c:
* gst/quicktime/atomsrecovery.h:
* gst/quicktime/fourcc.h:
* gst/quicktime/gstqtmoovrecover.c:
* gst/quicktime/gstqtmoovrecover.h:
* gst/quicktime/gstqtmux.c:
* gst/quicktime/gstqtmux.h:
* gst/quicktime/gstqtmuxplugin.c:
qtmux: Adds moov recovery feature
Adds a new property to qtmux that sets a path to a file to write
and update data about the moov atom (that is not writen till the
end of the file). If the pipeline/app crashes during execution it
might be possible to recover the movie using the qtmoovrecover element.
qtmoovrecover is an element that is also a pipeline. It is not
meant to be used with other elements (it has no pads). It is merely
a tool/utilitary to recover unfinished qtmux files.
Fixes #601576

2010-01-27 19:06:53 -0800 Michael Smith <[email protected]>

* gst/quicktime/atoms.c:
qtmux: for fixed-sample size streams (PCM audio, etc) don't allocate an enormous buffer that we then won't use at all.

2010-01-27 15:37:37 -0800 Michael Smith <[email protected]>

* gst/quicktime/gstqtmux.c:
qtmux: handle muxing adpcm correctly.

2010-01-22 13:36:04 -0800 Michael Smith <[email protected]>

* gst/quicktime/atoms.c:
qtmux: Set the mdia hdlr name field to what quicktime uses. Fix writing it since it's not null-terminated. Improves compatibility with some hardware players.

2010-01-22 13:30:07 -0800 Michael Smith <[email protected]>

* gst/quicktime/gstqtmux.c:
qtmux: endianness in gstreamer is an int, not boolean.

2010-01-26 17:54:28 +0100 Mark Nauwelaerts <[email protected]>

* gst/quicktime/atoms.c:
* gst/quicktime/atoms.h:
qtmux: streamline moov data memory storage
In particular, use arrays rather than (double) linked lists.

2010-01-26 13:44:04 -0300 Thiago Santos <[email protected]>

* gst/quicktime/gstqtmux.c:
qtmux: g_free is NULL safe

2010-01-20 13:30:48 +0100 Benjamin Otte <[email protected]>

* gst/quicktime/descriptors.c:
* gst/quicktime/descriptors.h:
* gst/quicktime/properties.c:
[cleanup] Various style and cleanups
Various fixes for gtk-doc warnings and making functions without
arguments take void as parameter.

2010-01-14 08:09:03 -0300 Thiago Santos <[email protected]>

* gst/quicktime/atoms.c:
* gst/quicktime/gstqtmux.c:
qtmux: Actually use new caps info on renegotiation
Following the previous qtmux commit, this patch tries
to use the new info added to the caps to fill the 'trak'
atom's fields and children atoms. This way qtmux will
use the late added 'codec_data' when h264parse adds
it in the following pipeline:
videotestsrc num-buffers=200 ! x264enc byte-stream=true ! \
h264parse output-format=0 ! qtmux ! \
filesink location=test.mov

2010-01-13 23:33:51 -0300 Thiago Santos <[email protected]>

* gst/quicktime/atoms.c:
* gst/quicktime/gstqtmux.c:
qtmux: Do caps renegotiation when it only adds fields
Qtmux can accept caps renegotiation if the new caps is a
superset of the old one, meaning upstream added new info to
the caps. This patch still doesn't make qtmux update any
atoms info from the new info, but at least it doesn't
reject the new caps anymore.
A pipeline that reproduces this use case is:
videotestsrc num-buffers=200 ! x264enc byte-stream=true ! \
h264parse output-format=0 ! qtmux ! \
filesink location=test.mov

2010-01-13 19:30:45 +0100 Mark Nauwelaerts <[email protected]>

* gst/quicktime/gstqtmux.c:
qtmux: provide request pads under wider conditions
Fixes #606859.

2010-01-13 10:35:00 -0300 Thiago Santos <[email protected]>

* gst/quicktime/gstqtmuxmap.c:
qtmux: Only accept avc-sample h264
qtmux and mp4mux should only accept h264 in avc-sample
format

2010-01-11 13:13:41 -0300 Thiago Santos <[email protected]>

* gst/quicktime/gstqtmux.c:
* gst/quicktime/gstqtmuxmap.c:
Rename aac's stream-format 'none' to 'raw'
Renames aac's stream-format from previous commits from none to
raw

2010-01-11 10:34:32 -0300 Thiago Santos <[email protected]>

* gst/quicktime/gstqtmux.c:
* gst/quicktime/gstqtmuxmap.c:
qtmux: Only accept stream-format='none' aac
Only accept raw aac streams (stream-format=none) to avoid
generating invalid files.
Fixes #604925

2009-12-28 11:34:35 +0200 Stefan Kost <[email protected]>

* gst/quicktime/gstqtmux.h:
qtmux: also add .h file changes to unbreak the build

2009-12-27 23:51:50 +0200 Stefan Kost <[email protected]>

* gst/quicktime/gstqtmux.c:
qtmux: use correct names from template for request pads
The pads where names pad0, pad1, ...

2009-12-27 23:32:58 +0200 Stefan Kost <[email protected]>

* gst/quicktime/gstqtmux.c:
qtmux: move errors _new_pad to the end

2009-12-21 13:58:30 -0300 Thiago Santos <[email protected]>

* gst/quicktime/gstqtmux.c:
qtmux: Accept non-paired uint tags
Adds support for unpaired unsigned interger tags

2009-12-21 12:05:37 -0300 Thiago Santos <[email protected]>

* gst/quicktime/fourcc.h:
* gst/quicktime/gstqtmux.c:
qtmux: Adds new tags
Maps more tags that are already posted by qtdemux
Fixes #599759

2009-12-10 22:20:45 -0300 Thiago Santos <[email protected]>

* gst/quicktime/atoms.c:
* gst/quicktime/atoms.h:
* gst/quicktime/fourcc.h:
* gst/quicktime/gstqtmux.c:
* gst/quicktime/gstqtmux.h:
* gst/quicktime/gstqtmuxmap.c:
qtmux: support more of j2k
Reads the new caps added to qtdemux by commit
c917d65e6df0b5d585f905c7ad78a8a0a44b2cb0
and adds its corresponding atoms.
Also adds support for image/x-jpc as it is the same
as image/x-jp2, except that the buffers need to be
boxed inside a jp2c isom box before muxing. To solve
this the QTPads now have a function that (if
not NULL) is called when a buffer is collected. This
function returns a replacement to the current collected
buffer.
Fixes #598916

2009-12-10 16:53:19 -0300 Thiago Santos <[email protected]>

* gst/quicktime/fourcc.h:
* gst/quicktime/gstqtmux.c:
* gst/quicktime/gstqtmux.h:
qtmux: Maps 'classification' tag for 3gpp files
Adds the mapping of 'classification' tags to writing of
'clsf' atoms for gppmux.
Based on a patch by: Lasse Laukkanen <[email protected]>

2009-12-08 17:59:04 -0800 Michael Smith <[email protected]>

* gst/quicktime/atoms.c:
* gst/quicktime/gstqtmux.c:
qtmux: remove c++ comments and add some more comments.

2009-12-08 17:55:56 -0800 Michael Smith <[email protected]>

* gst/quicktime/atoms.c:
* gst/quicktime/atoms.h:
* gst/quicktime/fourcc.h:
* gst/quicktime/gstqtmux.c:
* gst/quicktime/gstqtmuxmap.c:
qtmux: add ima adpcm support

2009-11-25 21:41:27 -0300 Thiago Santos <[email protected]>

* gst/quicktime/gstqtmux.c:
qtmux: replace _scale with _scale_round
Use the rounding version for improved sync between streams.
Small variations in the duration when muxing might lead to
cumullative wrong timestamping when demuxing.
Fixes #602936

2009-11-24 16:16:56 -0300 Thiago Santos <[email protected]>

* gst/quicktime/gstqtmux.c:
qtmux: use timestamps for muxing
Try to use timestamps even when the stream has out of order
timestamps, only fall back to durations when we detect an
out of order buffer. Improves sync between streams.

2009-11-19 18:28:52 -0300 Thiago Santos <[email protected]>

* gst/quicktime/gstqtmux.c:
qtmux: fix missing debug argument
Adds a missing debug argument

2009-11-19 11:36:14 +0100 Mark Nauwelaerts <[email protected]>

* gst/quicktime/gstqtmux.c:
qtmux: fix misinforming debug statement

2009-11-19 11:14:57 +0100 Mark Nauwelaerts <[email protected]>

* gst/quicktime/gstqtmux.c:
qtmux: ensure writable buffer metadata before setting caps

2009-10-29 08:36:02 -0300 Thiago Santos <[email protected]>

* gst/quicktime/atoms.c:
* gst/quicktime/atoms.h:
* gst/quicktime/fourcc.h:
* gst/quicktime/gstqtmux.c:
* gst/quicktime/gstqtmuxmap.c:
qtmux: support for SVQ3
Adds support for muxing SVQ3 content. Usually this format
has decoder info that must be passed in the 'seqh' field
in the caps. It is also good to add the gama atom to make
quicktime not crash.
Fixes #587922

2009-11-17 09:26:05 -0300 Thiago Sousa Santos <thiagoss@redmoon.(none)>

* gst/quicktime/gstqtmux.c:
qtmux: do not leak a string
Frees a string after use. Also does some code organization

2009-11-16 14:57:53 -0300 Thiago Sousa Santos <thiagoss@redmoon.(none)>

* gst/quicktime/atoms.c:
qtmux: do not add size to the pointer variable
Do not wrongly add the result of the function to the
pointer to the buffer size. Instead, check the result
to see if the serialization was ok.
Based on a patch by: "Carsten Kroll <[email protected]>"
Fixes #602106

2009-11-06 10:34:39 -0300 Thiago Santos <[email protected]>

* gst/quicktime/atoms.c:
* gst/quicktime/atoms.h:
* gst/quicktime/gstqtmux.c:
* gst/quicktime/gstqtmux.h:
qtmux: handle 'late' streams
When muxing streams, some can start later than others. qtmux
now handle this by adding an empty edts entry with the
duration of the 'lateness' to the stream's trak.
It tolerates a stream to be up to 0.1s late.
Fixes #586848

2009-11-05 21:35:56 -0300 Thiago Santos <[email protected]>

* gst/quicktime/atoms.c:
* gst/quicktime/atoms.h:
qtmux: adds the EDTS and ELTS atoms to atoms.c
These atoms will be useful for signaling streams
that start later in the file. As well for adding
edit lists if needed sometime later.

2009-11-06 00:46:12 -0300 Thiago Santos <[email protected]>

* gst/quicktime/atoms.c:
* gst/quicktime/gstqtmux.c:
qtmux: Adding some ifs for protection
Adding somes ifs to protect against warning conditions
that might happen when upstream element is not sane
Fixes #600895

2009-10-16 10:47:32 -0300 Thiago Santos <[email protected]>

* gst/quicktime/ftypcc.h:
* gst/quicktime/gstqtmux.c:
* gst/quicktime/gstqtmux.h:
* gst/quicktime/gstqtmuxmap.c:
* gst/quicktime/gstqtmuxmap.h:
gppmux: Add support for 3gr6
Keep track of the chunk durations to be able to add 3gr6
brand if it is a faststart file and the longest chunk is
smaller than a sec. Implemented according to 3gpp
TS 26.244 v6.4.0 (2005-09)
Fixes #584361

2009-10-15 21:11:16 -0300 Thiago Santos <[email protected]>

* gst/quicktime/gstqtmux.c:
qtmux: Only push ftyp later (in faststart mode)
In faststart mode, there is no need to send the ftyp
right at the beginning of the stream. Waiting and sending it
only later (when the moov atom is ready to be sent) provides
us with more information about the stream and we can better
select the compatible brands.

2009-10-15 17:51:39 -0300 Thiago Santos <[email protected]>

* gst/quicktime/gstqtmux.c:
qtmux: Improve error message
Improve error message when we can't get or estimate the
timestamp/duration of a buffer

2009-09-29 15:47:13 +0200 Marc-André Lureau <[email protected]>

* gst/quicktime/atoms.c:
qtmux: fix flags_as_uint to flags[]

2009-08-04 12:58:35 +0200 Jan Urbanski <[email protected]>

* gst/quicktime/gstqtmux.c:
qtmux: Don't require endianness field for 8 bit raw audio
Fixes bug #590360.

2009-06-25 08:38:21 +0200 Edward Hervey <[email protected]>

* gst/quicktime/atoms.c:
qtmux: Remove unused variable.

2009-06-25 08:38:10 +0200 Edward Hervey <[email protected]>

* gst/quicktime/gstqtmux.c:
qtmux: Fix debug statement.

2009-06-11 15:54:42 +0200 Mark Nauwelaerts <[email protected]>

* gst/quicktime/atoms.c:
* gst/quicktime/gstqtmux.c:
* gst/quicktime/gstqtmux.h:
qtmux: only use (64-bit) extended (mdat) atom size if needed. Fixes #585319.

2009-06-10 14:46:14 +0200 Mark Nauwelaerts <[email protected]>

* gst/quicktime/gstqtmux.c:
qtmux: set default movie timescale to microsecond units

2009-06-10 13:24:20 +0200 Mark Nauwelaerts <[email protected]>

* gst/quicktime/atoms.c:
qtmux: compress/optimize stsc writing

2009-06-10 12:42:44 +0200 Mark Nauwelaerts <[email protected]>

* gst/quicktime/atoms.c:
* gst/quicktime/atoms.h:
* gst/quicktime/fourcc.h:
* gst/quicktime/gstqtmux.c:
* gst/quicktime/gstqtmuxmap.c:
qtmux: add 3GP style tagging (and refactor appropriately)

2009-06-01 23:00:44 +0200 Mark Nauwelaerts <[email protected]>

* gst/quicktime/atoms.c:
* gst/quicktime/atoms.h:
* gst/quicktime/fourcc.h:
* gst/quicktime/gstqtmux.c:
qtmux (and variants): handle pixel-aspect-ratio. Fixes #584358.

2009-06-01 22:42:08 +0200 Mark Nauwelaerts <[email protected]>

* gst/quicktime/atoms.c:
* gst/quicktime/atoms.h:
* gst/quicktime/ftypcc.h:
* gst/quicktime/gstqtmuxmap.c:
gppmux: enhance ftyp brand heuristic. Fixes #584360.

2009-05-28 13:56:10 +0200 Mark Nauwelaerts <[email protected]>

* gst/quicktime/fourcc.h:
* gst/quicktime/gstqtmux.c:
qtmux: use different stsd atom type for H263 for ISO and QT variants
Fixes #584114.

2009-05-15 01:54:44 -0300 Thiago Santos <[email protected]>

* gst/quicktime/atoms.c:
[qtmux] Fixes segfault when adding a blob as first tag.
Moves tags data initialization to the function that actually appends
the tags to the list. Fixes #582702
Also fixes some style caught by the pre-commit hook.

2009-05-10 21:21:36 +0200 Mark Nauwelaerts <[email protected]>

* gst/quicktime/gstqtmuxmap.c:
gppmux: Add MPEG-4 part 2 to supported formats. Fixes #581593.

2009-05-07 17:53:42 +0100 Christian Schaller <[email protected]>

* gst/quicktime/gstqtmux.c:
Add ranks to various muxers and encoders in -bad

2009-04-30 14:43:36 -0300 Thiago Santos <[email protected]>

* gst/quicktime/gstqtmuxmap.c:
qtmux: changes caps of src pads to video/quicktime, variant=something
Take a look at bug #580005 for further info.

2009-04-24 18:53:36 -0300 Thiago Santos <[email protected]>

* gst/quicktime/gstqtmuxmap.c:
mp4mux: Changes src caps to application/x-iso-mp4
Fixes #580005

2009-03-25 21:24:44 +0100 Mark Nauwelaerts <[email protected]>

* gst/quicktime/gstqtmux.c:
qtmux: fix reusing element
State change to READY and then back to PAUSED should still provide
the proper structures as are otherwise freshly available following
a request_new_pad.
Pointed out by Thiago Santos.

2009-03-23 11:17:39 +0100 Wim Taymans <[email protected]>

* gst/quicktime/gstqtmux.c:
qtmux: fix includes for lseek
--

2009-03-20 14:20:16 +0100 LRN <lrn1986 at gmail dot com>

* gst/quicktime/gstqtmux.c:
win32: fix seeking in large files
Use _lseeki64() on Windows to seek in large files.
Fixes #576021.

2009-03-02 10:57:35 +0100 Edward Hervey <[email protected]>

* gst/quicktime/gstqtmux.c:
qtmux: Be a bit more verbose in our debug message when failing to renegotiate

2009-01-28 13:25:14 +0100 Mark Nauwelaerts <[email protected]>

* gst/quicktime/atoms.c:
* gst/quicktime/atoms.h:
* gst/quicktime/gstqtmux.c:
* gst/quicktime/gstqtmuxmap.c:
Additional media type support in qtmux (and friends).
Support AMR and H263 for both qtmux and gppmux,
and add extensions in sample table description.

2009-01-09 21:59:48 +0000 David Schleef <[email protected]>

gst/quicktime/gstqtmuxmap.c: Add video/x-qt-part and video/x-m4-part to caps so schroenc/schroparse can use it. Fixes #5...
Original commit message from CVS:
* gst/quicktime/gstqtmuxmap.c: Add video/x-qt-part and video/x-m4-part
to caps so schroenc/schroparse can use it. Fixes #566958

2008-12-19 18:53:47 +0000 Mark Nauwelaerts <[email protected]>

gst/quicktime/gstqtmux.c: Do not tempt or suggest to violate gst_collect_pads API specification.
Original commit message from CVS:
* gst/quicktime/gstqtmux.c: (gst_qt_mux_change_state):
Do not tempt or suggest to violate gst_collect_pads API specification.

2008-12-19 18:33:47 +0000 Mark Nauwelaerts <[email protected]>

gst/quicktime/: Dual license qtmux LGPL/MIT. Fixes #564232.
Original commit message from CVS:
* gst/quicktime/atoms.c:
* gst/quicktime/atoms.h:
* gst/quicktime/descriptors.c:
* gst/quicktime/descriptors.h:
* gst/quicktime/fourcc.h:
* gst/quicktime/ftypcc.h:
* gst/quicktime/gstqtmux.c:
* gst/quicktime/gstqtmux.h:
* gst/quicktime/gstqtmuxmap.c:
* gst/quicktime/gstqtmuxmap.h:
* gst/quicktime/properties.c:
* gst/quicktime/properties.h:
Dual license qtmux LGPL/MIT. Fixes #564232.

2008-12-16 16:26:52 +0000 Stefan Kost <[email protected]>

Totally remove the internal taglists and fully use tagsetter. Fixes various tag muxing issues.
Original commit message from CVS:
* ext/celt/gstceltenc.c:
* ext/celt/gstceltenc.h:
* ext/metadata/gstmetadatamux.c:
* gst/quicktime/gstqtmux.c:
* gst/quicktime/gstqtmux.h:
Totally remove the internal taglists and fully use tagsetter. Fixes
various tag muxing issues.

2008-12-01 16:37:45 +0000 Mark Nauwelaerts <[email protected]>

gst/quicktime/atoms.c: Fix mj2 sample description metadata construction.
Original commit message from CVS:
* gst/quicktime/atoms.c: (build_jp2h_extension):
Fix mj2 sample description metadata construction.

2008-11-18 01:09:09 +0000 David Schleef <[email protected]>

gst/quicktime/gstqtmux.c: Quiet a debugging message that I recently added.
Original commit message from CVS:
* gst/quicktime/gstqtmux.c: Quiet a debugging message that I recently
added.

2008-11-15 02:56:31 +0000 David Schleef <[email protected]>

gst/quicktime/gstqtmux.*: Use dts from GST_BUFFER_OFFSET_END() for video/x-qt-part.
Original commit message from CVS:
* gst/quicktime/gstqtmux.c:
* gst/quicktime/gstqtmux.h:
Use dts from GST_BUFFER_OFFSET_END() for video/x-qt-part.

2008-11-14 21:24:51 +0000 Mark Nauwelaerts <[email protected]>

gst/quicktime/: Revert previous commit.
Original commit message from CVS:
* gst/quicktime/atoms.c:
* gst/quicktime/atoms.h:
* gst/quicktime/descriptors.c:
* gst/quicktime/descriptors.h:
* gst/quicktime/fourcc.h:
* gst/quicktime/ftypcc.h:
* gst/quicktime/gstqtmux.c:
* gst/quicktime/gstqtmux.h:
* gst/quicktime/gstqtmuxmap.c:
* gst/quicktime/gstqtmuxmap.h:
* gst/quicktime/properties.c:
* gst/quicktime/properties.h:
Revert previous commit.

2008-11-14 20:38:18 +0000 Mark Nauwelaerts <[email protected]>

gst/quicktime/: Dual license LGPL/MIT, as apparently supposed to.
Original commit message from CVS:
* gst/quicktime/atoms.c:
* gst/quicktime/atoms.h:
* gst/quicktime/descriptors.c:
* gst/quicktime/descriptors.h:
* gst/quicktime/fourcc.h:
* gst/quicktime/ftypcc.h:
* gst/quicktime/gstqtmux.c:
* gst/quicktime/gstqtmux.h:
* gst/quicktime/gstqtmuxmap.c:
* gst/quicktime/gstqtmuxmap.h:
* gst/quicktime/properties.c:
* gst/quicktime/properties.h:
Dual license LGPL/MIT, as apparently supposed to.

2008-11-14 20:17:10 +0000 Mark Nauwelaerts <[email protected]>

gst/quicktime/: Cut detour in sample description extension construction.
Original commit message from CVS:
* gst/quicktime/atoms.c: (build_esds_extension),
(build_mov_aac_extension), (build_jp2h_extension),
(build_codec_data_extension):
* gst/quicktime/atoms.h:
* gst/quicktime/fourcc.h:
* gst/quicktime/gstqtmux.c: (gst_qt_mux_audio_sink_set_caps),
(gst_qt_mux_video_sink_set_caps):
* gst/quicktime/gstqtmuxmap.c: (gst_qt_mux_map_format_to_header):
Cut detour in sample description extension construction.
Also actually implement ISO JPEG2000 mj2 format.

2008-11-11 19:31:35 +0000 Mark Nauwelaerts <[email protected]>

tests/check/: Add unit test for qtmux.
Original commit message from CVS:
* tests/check/Makefile.am:
* tests/check/elements/qtmux.c: (setup_src_pad),
(teardown_src_pad), (setup_qtmux), (cleanup_qtmux),
(check_qtmux_pad), (GST_START_TEST), (qtmux_suite), (main):
Add unit test for qtmux.

2008-11-11 19:24:12 +0000 Mark Nauwelaerts <[email protected]>

gst/quicktime/gstqtmux.c: Add some more safety/sanity checks in tag manipulation.
Original commit message from CVS:
* gst/quicktime/gstqtmux.c: (gst_qt_mux_add_metadata_tags):
Add some more safety/sanity checks in tag manipulation.

2008-11-08 02:00:58 +0000 Thiago Sousa Santos <[email protected]>

Copy qtmux from revision 148 of the gst-qtmux repository.
Original commit message from CVS:
patch by: Thiago Sousa Santos <[email protected]>
* configure.ac:
* gst/quicktime/Makefile.am:
* gst/quicktime/atoms.c:
* gst/quicktime/atoms.h:
* gst/quicktime/descriptors.c:
* gst/quicktime/descriptors.h:
* gst/quicktime/fourcc.h:
* gst/quicktime/ftypcc.h:
* gst/quicktime/gstqtmux.c:
* gst/quicktime/gstqtmux.h:
* gst/quicktime/gstqtmuxmap.c:
* gst/quicktime/gstqtmuxmap.h:
* gst/quicktime/properties.c:
* gst/quicktime/properties.h:
Copy qtmux from revision 148 of the gst-qtmux repository.
Fixes #550280.

2011-04-12 18:25:34 +0100 Tim-Philipp Müller <[email protected]>

* Android.mk:
* configure.ac:
* docs/plugins/Makefile.am:
* docs/plugins/inspect/plugin-quicktime.xml:
* gst/qtdemux/LEGAL:
* gst/qtdemux/Makefile.am:
* gst/qtdemux/gstrtpxqtdepay.c:
* gst/qtdemux/gstrtpxqtdepay.h:
* gst/qtdemux/qtatomparser.h:
* gst/qtdemux/qtdemux.c:
* gst/qtdemux/qtdemux.h:
* gst/qtdemux/qtdemux.vcproj:
* gst/qtdemux/qtdemux_dump.c:
* gst/qtdemux/qtdemux_dump.h:
* gst/qtdemux/qtdemux_fourcc.h:
* gst/qtdemux/qtdemux_lang.c:
* gst/qtdemux/qtdemux_lang.h:
* gst/qtdemux/qtdemux_types.c:
* gst/qtdemux/qtdemux_types.h:
* gst/qtdemux/qtpalette.h:
* gst/qtdemux/quicktime.c:
* gst/quicktime/LEGAL:
* gst/quicktime/Makefile.am:
* gst/quicktime/gstrtpxqtdepay.c:
* gst/quicktime/gstrtpxqtdepay.h:
* gst/quicktime/qtatomparser.h:
* gst/quicktime/qtdemux.c:
* gst/quicktime/qtdemux.h:
* gst/quicktime/qtdemux.vcproj:
* gst/quicktime/qtdemux_dump.c:
* gst/quicktime/qtdemux_dump.h:
* gst/quicktime/qtdemux_fourcc.h:
* gst/quicktime/qtdemux_lang.c:
* gst/quicktime/qtdemux_lang.h:
* gst/quicktime/qtdemux_types.c:
* gst/quicktime/qtdemux_types.h:
* gst/quicktime/qtpalette.h:
* gst/quicktime/quicktime.c:
* po/POTFILES.in:
qtdemux: rename directory to quicktime to match plugin name
In preparation for qtmux moving to -good.

2011-04-12 11:49:54 +0200 Mark Nauwelaerts <[email protected]>

* gst/flv/gstflvdemux.c:
flvdemux: simplify framerate fraction calculation

2011-01-24 15:45:28 -0600 Leonardo Sandoval <[email protected]>

* gst/flv/gstflvdemux.c:
* gst/flv/gstflvdemux.h:
flvdemux: add width, height and framerate to caps when present on onMetaData
Fixes #640483.

2010-08-24 13:57:55 +0200 Pascal Buhler <[email protected]>

* gst/rtpmanager/gstrtpssrcdemux.c:
rtpssrcdemux: Unknown SSRC is not fatal
https://bugzilla.gnome.org/show_bug.cgi?id=646966

2010-08-24 13:54:58 +0200 Pascal Buhler <[email protected]>

* gst/rtpmanager/rtpsession.c:
rtpsession: Number of active sources should be updated whenever the status of the source changes to active
Forward-ported by Olivier Crête
https://bugzilla.gnome.org/show_bug.cgi?id=646965

2010-06-23 11:29:58 +0200 Havard Graff <[email protected]>

* gst/rtpmanager/rtpsession.c:
rtpmanager: ignore a BYE if it is sent with our internal SSRC
https://bugzilla.gnome.org/show_bug.cgi?id=646964

2010-01-29 09:49:48 -0300 Thiago Santos <[email protected]>

* gst/qtdemux/qtdemux.c:
qtdemux: Adds more h264 fields to its caps
Adds alignment=au and stream-format=avc to h264 caps
Fixes #606662

2011-04-11 12:44:19 +0300 Stefan Kost <[email protected]>

* configure.ac:
* ext/jack/gstjackaudiosink.c:
* ext/jack/gstjackaudiosrc.c:
jack: also handle deprecations for jack 1.9.7
Jack 1.9.7 was released 20.Mar.2011, need to handle the deprecated api for this
version too.

2011-04-10 18:56:52 -0400 Thibault Saunier <[email protected]>

* Android.mk:
* android/NOTICE:
* android/apetag.mk:
* android/avi.mk:
* android/flv.mk:
* android/gst/rtpmanager/gstrtpbin-marshal.c:
* android/gst/rtpmanager/gstrtpbin-marshal.h:
* android/gst/udp/gstudp-enumtypes.c:
* android/gst/udp/gstudp-enumtypes.h:
* android/gst/udp/gstudp-marshal.c:
* android/gst/udp/gstudp-marshal.h:
* android/icydemux.mk:
* android/id3demux.mk:
* android/qtdemux.mk:
* android/rtp.mk:
* android/rtpmanager.mk:
* android/rtsp.mk:
* android/soup.mk:
* android/udp.mk:
* android/wavenc.mk:
* android/wavparse.mk:
* gst/alpha/Makefile.am:
* gst/apetag/Makefile.am:
* gst/audiofx/Makefile.am:
* gst/auparse/Makefile.am:
* gst/autodetect/Makefile.am:
* gst/avi/Makefile.am:
* gst/cutter/Makefile.am:
* gst/debugutils/Makefile.am:
* gst/deinterlace/Makefile.am:
* gst/effectv/Makefile.am:
* gst/equalizer/Makefile.am:
* gst/flv/Makefile.am:
* gst/flx/Makefile.am:
* gst/goom/Makefile.am:
* gst/goom2k1/Makefile.am:
* gst/icydemux/Makefile.am:
* gst/id3demux/Makefile.am:
* gst/imagefreeze/Makefile.am:
* gst/interleave/Makefile.am:
* gst/law/Makefile.am:
* gst/level/Makefile.am:
* gst/matroska/Makefile.am:
* gst/monoscope/Makefile.am:
* gst/multifile/Makefile.am:
* gst/multipart/Makefile.am:
* gst/qtdemux/Makefile.am:
* gst/replaygain/Makefile.am:
* gst/rtp/Makefile.am:
* gst/rtpmanager/Makefile.am:
* gst/rtsp/Makefile.am:
* gst/shapewipe/Makefile.am:
* gst/smpte/Makefile.am:
* gst/spectrum/Makefile.am:
* gst/udp/Makefile.am:
* gst/videobox/Makefile.am:
* gst/videocrop/Makefile.am:
* gst/videofilter/Makefile.am:
* gst/videomixer/Makefile.am:
* gst/wavenc/Makefile.am:
* gst/wavparse/Makefile.am:
* gst/y4m/Makefile.am:
android: Make it ready for androgenizer
Remove the android/ top dir
Fixe the Makefile.am to be androgenized
To build gstreamer for android we are now using androgenizer which generates the needed Android.mk files.
Androgenizer can be found here: http://git.collabora.co.uk/?p=user/derek/androgenizer.git

2011-04-05 21:14:43 +0200 Haakon Sporsheim <[email protected]>

* gst/rtp/gstrtpgstpay.c:
rtpgstpay: declare frag_offset to hold 32bits.
As specified in documenation above and below.
https://bugzilla.gnome.org/show_bug.cgi?id=646954

2011-04-09 12:41:48 +0200 Havard Graff <[email protected]>

* gst/rtpmanager/gstrtpsession.c:
rtpsession: fix wrongly applied patch
Obviously recv_rtp_sink does not have much to do with send_rtcp_src...
See commit 046ff170.
https://bugzilla.gnome.org/show_bug.cgi?id=647263

2011-04-08 15:59:58 +0100 Tim-Philipp Müller <[email protected]>

* gst/audioparsers/gstaacparse.c:
* gst/audioparsers/gstac3parse.c:
* gst/audioparsers/gstamrparse.c:
* gst/audioparsers/gstdcaparse.c:
* gst/audioparsers/gstmpegaudioparse.c:
audioparsers: update for set_frame_props -> set_frame_rate API change

2011-04-08 00:03:21 +0100 Tim-Philipp Müller <[email protected]>

* tests/check/Makefile.am:
* tests/check/elements/.gitignore:
tests: hook up audioparser unit tests

2011-04-07 18:30:49 +0200 Mark Nauwelaerts <[email protected]>

* gst/audioparsers/gstmpegaudioparse.c:
mpegaudioparse: relax sync match a bit when draining
... to at least allow initial caps change (but no further caps jitter).
Fixes unit test again after previous change.

2011-04-07 15:21:10 +0100 Tim-Philipp Müller <[email protected]>

* docs/plugins/gst-plugins-good-plugins.args:
* docs/plugins/gst-plugins-good-plugins.hierarchy:
* docs/plugins/gst-plugins-good-plugins.interfaces:
* docs/plugins/gst-plugins-good-plugins.prerequisites:
* docs/plugins/inspect/plugin-avi.xml:
* docs/plugins/inspect/plugin-cairo.xml:
* docs/plugins/inspect/plugin-flv.xml:
* docs/plugins/inspect/plugin-matroska.xml:
* docs/plugins/inspect/plugin-monoscope.xml:
* docs/plugins/inspect/plugin-png.xml:
* docs/plugins/inspect/plugin-video4linux2.xml:
* docs/plugins/inspect/plugin-videofilter.xml:
docs: update for changes in git

2011-04-07 15:20:19 +0100 Tim-Philipp Müller <[email protected]>

* docs/plugins/Makefile.am:
* docs/plugins/gst-plugins-good-plugins-docs.sgml:
* docs/plugins/gst-plugins-good-plugins-sections.txt:
* docs/plugins/inspect/plugin-audioparsers.xml:
docs: add audioparsers to docs

2011-04-07 15:07:15 +0100 Tim-Philipp Müller <[email protected]>

* gst/audioparsers/gstaacparse.c:
* gst/audioparsers/gstaacparse.h:
* gst/audioparsers/gstamrparse.c:
* gst/audioparsers/gstamrparse.h:
* gst/audioparsers/plugin.c:
aacparse, amrparse: gst_fooparse_xyz -> gst_foo_parse_xyz to match GstFooParse
See moving-plugins checklist.

2011-04-07 14:43:42 +0100 Tim-Philipp Müller <[email protected]>

* configure.ac:
* gst/audioparsers/Makefile.am:
* gst/audioparsers/plugin.c:
audioparsers: hook up to build

2011-04-07 13:26:41 +0100 Tim-Philipp Müller <[email protected]>

* gst/audioparsers/Makefile.am:
* gst/audioparsers/gstaacparse.c:
* gst/audioparsers/gstaacparse.h:
* gst/audioparsers/gstac3parse.c:
* gst/audioparsers/gstac3parse.h:
* gst/audioparsers/gstamrparse.c:
* gst/audioparsers/gstamrparse.h:
* gst/audioparsers/gstdcaparse.c:
* gst/audioparsers/gstdcaparse.h:
* gst/audioparsers/gstflacparse.c:
* gst/audioparsers/gstflacparse.h:
* gst/audioparsers/gstmpegaudioparse.c:
* gst/audioparsers/gstmpegaudioparse.h:
audioparsers: port to new GstBaseParse in core

2011-04-04 20:55:39 +0200 Mark Nauwelaerts <[email protected]>

* gst/audioparsers/gstmpegaudioparse.c:
mpegaudioparse: require tighter sync match when draining

2011-04-01 14:47:43 +0200 Sebastian Dröge <[email protected]>

* gst/audioparsers/gstmpegaudioparse.c:
* gst/audioparsers/gstmpegaudioparse.h:
mpegaudioparse: Parse encoder delay and encoder padding from the LAME header if present

2011-03-09 23:06:14 +0530 Arun Raghavan <[email protected]>

* gst/audioparsers/plugin.c:
dcaparse: Bump rank to primary+1
Seems to work fine with a reasonably wide range of media, so bumping
rank.

2011-03-23 22:02:37 +0530 Arun Raghavan <[email protected]>

* gst/audioparsers/gstdcaparse.c:
* gst/audioparsers/gstdcaparse.h:
dcaparse: Expose frame size in caps
This exports the size of the frame (number of bytes from one sync point
to the next) as the "frame_size" field in caps.

2011-03-09 23:03:10 +0530 Arun Raghavan <[email protected]>

* gst/audioparsers/gstdcaparse.c:
* gst/audioparsers/gstdcaparse.h:
dcaparse: Expose block size in caps
This sets the "block_size" field on caps as the number of samples
encoded in one frame.

2011-03-16 15:53:13 +0000 Tim-Philipp Müller <[email protected]>

* gst/audioparsers/gstmpegaudioparse.c:
mpegaudioparse: add FIXME for making the base class use xing seek tables better

2011-03-14 18:25:25 +0100 Sebastian Dröge <[email protected]>

* gst/audioparsers/gstdcaparse.c:
* gst/audioparsers/gstdcaparse.h:
dcaparse: Add depth and endianness to the caps
Some decoders can only handle specific endianness or a fixed
depth and this allows better negotiation.
Fixes bug #644208.

2011-02-26 13:53:44 -0800 David Schleef <[email protected]>

* gst/audioparsers/gstaacparse.c:
Revert "aacparse: allow parsed frames on sink pad"
This reverts commit e49b89d5c5a1244fa0dcb8bb4996e38fb9bff9e5.

2011-02-23 17:25:03 -0800 David Schleef <[email protected]>

* gst/audioparsers/gstaacparse.c:
aacparse: allow parsed frames on sink pad

2010-10-13 16:12:02 -0700 David Schleef <[email protected]>

* tests/check/elements/parser.c:
tests: fix baseparse test

2010-10-13 15:39:55 -0700 David Schleef <[email protected]>

* gst/audioparsers/Makefile.am:
* gst/audioparsers/gstaacparse.h:
* gst/audioparsers/gstac3parse.h:
* gst/audioparsers/gstamrparse.h:
* gst/audioparsers/gstbaseparse.c:
* gst/audioparsers/gstbaseparse.h:
* gst/audioparsers/gstdcaparse.h:
* gst/audioparsers/gstflacparse.h:
* gst/audioparsers/gstmpegaudioparse.h:
baseparse: Create baseparse library

2011-02-07 14:46:57 +0100 Mark Nauwelaerts <[email protected]>

* gst/audioparsers/gstbaseparse.c:
baseparse: tune QUERY_SEEKING response
Even if we currently do not have a duration yet, assume seekable if
it looks like we'll likely be able to determine it later on
(which coincides with needed information to perform seeking).
Fixes #641047.

2011-02-08 23:39:24 +0530 Arun Raghavan <[email protected]>

* gst/audioparsers/gstbaseparse.c:
baseparse: Update min/max bitrate before first posting them
This avoids posting an initial min-bitrate of G_UINTMAX and max-bitrate
of 0.
https://bugzilla.gnome.org/show_bug.cgi?id=641857

2011-02-08 23:50:13 +0530 Arun Raghavan <[email protected]>

* gst/audioparsers/gstmpegaudioparse.c:
* gst/audioparsers/gstmpegaudioparse.h:
mpegaudioparse: Post CBR bitrate as nominal bitrate
Even if VBR headers are missing, we can't guarantee that a stream is in
fact a CBR stream, so it's safer to let baseparse calculate the average
bitrate rather than assume a CBR stream. However, in order to make
/some/ metadata available before the requisite number of frames have
been parsed, this posts the bitrate from the non-VBR headers as the
nominal bitrate.
https://bugzilla.gnome.org/show_bug.cgi?id=641858

2010-09-06 14:10:11 +0200 Mark Nauwelaerts <[email protected]>

* gst/audioparsers/gstamrparse.c:
amrparse: a valid amr-wb frame should not have reserved frame type index
See #639715.

2011-01-27 16:52:34 +0100 Mark Nauwelaerts <[email protected]>

* gst/audioparsers/gstac3parse.c:
ac3parse: improve handling of dependent substream frames
In particular, timestamps of these should track main-stream timestamps.

2011-01-21 14:53:39 +0100 Mark Nauwelaerts <[email protected]>

* gst/audioparsers/gstbaseparse.c:
baseparse: tune default duration estimate update interval
Rather than a fixed default frame count, estimate frame count to aim for
an interval duration depending on fps if available, otherwise use old
fixed default.

2011-01-14 15:16:04 +0100 Mark Nauwelaerts <[email protected]>

* gst/audioparsers/gstbaseparse.c:
baseparse: reverse playback; mind keyframes for fragment boundary

2011-01-13 15:26:21 +0100 Mark Nauwelaerts <[email protected]>

* gst/audioparsers/gstamrparse.c:
amrparse: properly check for sufficient available data prior to access

2011-01-12 14:40:37 +0100 Mark Nauwelaerts <[email protected]>

* gst/audioparsers/gstbaseparse.c:
baseparse: ensure non-empty candidate frames

2011-01-11 15:24:23 +0100 Mark Nauwelaerts <[email protected]>

* gst/audioparsers/gstbaseparse.c:
baseparse: clarify some debug statements

2011-01-11 15:24:02 +0100 Mark Nauwelaerts <[email protected]>

* gst/audioparsers/gstbaseparse.c:
baseparse: properly track upstream timestamps
... rather than with a delay.

2011-01-11 15:23:29 +0100 Mark Nauwelaerts <[email protected]>

* gst/audioparsers/gstbaseparse.c:
baseparse: need proper frame duration to obtain sensible frame bitrate

2011-01-11 15:22:51 +0100 Mark Nauwelaerts <[email protected]>

* gst/audioparsers/gstbaseparse.c:
baseparse: proper initial values for index tracking variables

2011-01-11 12:05:13 +0100 Mark Nauwelaerts <[email protected]>

* gst/audioparsers/gstbaseparse.c:
baseparse: arrange for consistent event handling

2011-01-10 16:59:59 +0100 Mark Nauwelaerts <[email protected]>

* gst/audioparsers/gstbaseparse.h:
baseparse: header style cleaning

2011-01-10 17:07:38 +0100 Mark Nauwelaerts <[email protected]>

* gst/audioparsers/gstbaseparse.c:
baseparse: provide some more initial frame metadata in parse_frame
... and document accordingly.

2011-01-10 16:56:36 +0100 Mark Nauwelaerts <[email protected]>

* gst/audioparsers/gstaacparse.c:
* gst/audioparsers/gstbaseparse.c:
* gst/audioparsers/gstbaseparse.h:
* gst/audioparsers/gstflacparse.c:
baseparse: refactor passthrough into format flags
Also add a format flag to signal baseparse that subclass/format can provide
(parsed) timestamp rather than an estimated one. In particular, such "strong"
timestamp then allows to e.g. determine duration.

2011-01-10 15:34:48 +0100 Mark Nauwelaerts <[email protected]>

* gst/audioparsers/gstaacparse.c:
* gst/audioparsers/gstac3parse.c:
* gst/audioparsers/gstamrparse.c:
* gst/audioparsers/gstbaseparse.c:
* gst/audioparsers/gstbaseparse.h:
* gst/audioparsers/gstdcaparse.c:
* gst/audioparsers/gstflacparse.c:
* gst/audioparsers/gstmpegaudioparse.c:
baseparse: introduce a baseparse frame to serve as context
... and adjust subclass parsers accordingly

2011-01-07 16:39:51 +0100 Mark Nauwelaerts <[email protected]>

* gst/audioparsers/gstbaseparse.c:
* gst/audioparsers/gstbaseparse.h:
baseparse: restrict duration scanning to pull mode and avoid extra set_caps call

2011-01-07 15:58:49 +0100 Mark Nauwelaerts <[email protected]>

* gst/audioparsers/gstbaseparse.c:
* gst/audioparsers/gstbaseparse.h:
baseparse: update some documentation
Also add some more debug.

2011-01-06 11:41:44 +0100 Mark Nauwelaerts <[email protected]>

* gst/audioparsers/gstbaseparse.c:
baseparse: allow increasing min_size for current frame parsing only
Also check that subclass actually either directs to skip bytes or
increases expected frame size to avoid going nowhere in bogus
indefinite looping.

2011-01-14 15:26:37 +0100 Mark Nauwelaerts <[email protected]>

* gst/audioparsers/gstbaseparse.c:
baesparse: fix refactor regression in loop based parsing

2011-01-06 11:16:56 +0100 Mark Nauwelaerts <[email protected]>

* gst/audioparsers/gstbaseparse.c:
baseparse: pass all available data to subclass rather than minimum
Also reduce some adapter calls and add a few debug statements.

2010-12-10 15:59:49 +0100 Mark Nauwelaerts <[email protected]>

* gst/audioparsers/gstbaseparse.c:
baseparse: fix reverse playback handling

2010-12-10 14:56:13 +0100 Mark Nauwelaerts <[email protected]>

* gst/audioparsers/gstbaseparse.c:
baseparse: minor typo and debug statement cleanup

2010-12-10 14:40:05 +0100 Mark Nauwelaerts <[email protected]>

* gst/audioparsers/gstbaseparse.c:
* gst/audioparsers/gstbaseparse.h:
baseparse: reduce locking
... which is either already mute and/or implicitly handled by STREAM_LOCK.

2011-01-14 14:08:38 +0100 Mark Nauwelaerts <[email protected]>

* gst/audioparsers/gstbaseparse.c:
baseparse: avoid loop in frame locating interpolation

2011-01-19 18:26:30 +0100 Mark Nauwelaerts <[email protected]>

* gst/audioparsers/gstflacparse.c:
flacparse: mind gst_buffer_unref not liking NULL
Fixes #639950.

2011-01-14 16:30:11 -0300 Thiago Santos <[email protected]>

* gst/audioparsers/gstbaseparse.c:
audioparsers: baseparse: Be careful to not lose the event ref
Don't unref the event if it hasn't been handled, because the caller
assumes it is still valid and might reuse it.
I ran into this problem when transcoding an AVI (with mp3 inside)
to gpp.
https://bugzilla.gnome.org/show_bug.cgi?id=639555

2011-01-13 17:10:13 +0000 Tim-Philipp Müller <[email protected]>

* gst/audioparsers/gstdcaparse.c:
dcaparse: fix sync word for 14-bit little endian coding
Fix copy'n'paste bug that made us look for the raw little endian
sync word twice instead of looking for the 14-bit LE sync word
as well. Fixes parsing of such streams (see #636234 for sample file).

2011-01-13 16:27:04 +0000 Tim-Philipp Müller <[email protected]>

* gst/audioparsers/gstbaseparse.c:
docs: minor baseparse docs/comment fixes
Remove copy'n'paste leftovers.

2011-01-06 12:49:43 +0100 Edward Hervey <[email protected]>

* gst/audioparsers/gstflacparse.c:
flacparse: Fix unitialized variable on macosx

2010-12-13 15:17:29 +0100 Mark Nauwelaerts <[email protected]>

* gst/audioparsers/gstac3parse.c:
ac3parse: relax bsid checking
... to the widest possible spec interpretation.
Fixes #637062.

2010-12-03 18:11:56 +0100 Mark Nauwelaerts <[email protected]>

* gst/audioparsers/gstaacparse.c:
* gst/audioparsers/gstac3parse.c:
* gst/audioparsers/gstamrparse.c:
audioparsers: update some documentation

2010-12-03 18:11:38 +0100 Mark Nauwelaerts <[email protected]>

* gst/audioparsers/gstmpegaudioparse.c:
mpegaudioparse: add to documentation

2010-12-03 18:11:09 +0100 Mark Nauwelaerts <[email protected]>

* gst/audioparsers/gstdcaparse.c:
dcaparse: add to documentation

2010-11-08 19:58:31 +0100 Mark Nauwelaerts <[email protected]>

* gst/audioparsers/gstbaseparse.c:
baseparse: increase keyframe awareness
... which is not particular relevant for audio parsing, but more so
in video cases. In particular, auto-determine if dealing with video (caps).

2010-12-01 15:28:53 +0100 Mark Nauwelaerts <[email protected]>

* gst/audioparsers/gstac3parse.c:
* gst/audioparsers/gstac3parse.h:
ac3parse: use proper EAC-3 caps

2010-11-30 15:41:02 +0100 Mark Nauwelaerts <[email protected]>

* gst/audioparsers/gstbaseparse.c:
baseparse: avoid unexpected stray metadata

2010-11-30 15:40:28 +0100 Mark Nauwelaerts <[email protected]>

* gst/audioparsers/gstbaseparse.c:
baseparse: use proper _NONE output value when applicable

2010-11-25 18:56:42 +0100 Edward Hervey <[email protected]>

* gst/audioparsers/gstaacparse.c:
* gst/audioparsers/gstamrparse.c:
* gst/audioparsers/gstbaseparse.c:
audioparsers: Remove dead assignments

2010-11-25 17:14:23 +0100 Andoni Morales Alastruey <[email protected]>

* gst/audioparsers/gstbaseparse.c:
audioparse: fix possible division-by-zero
https://bugzilla.gnome.org/show_bug.cgi?id=635786

2010-11-17 16:23:42 +0100 Mark Nauwelaerts <[email protected]>

* gst/audioparsers/gstbaseparse.c:
baseparse: use correct offset when adding index entry
... bearing in mind that BUFFER_OFFSET is media specific and may not
reflect the basic offset after having been parsed.

2010-11-17 14:30:09 +0100 Mark Nauwelaerts <[email protected]>

* gst/audioparsers/gstbaseparse.c:
baseparse: enhancements for timestamp marked framed formats
That is, as such formats allow subclass to extract position from frame,
it is possible to extract duration (if not otherwise provided)
from (near) last frame, and a seek can fairly accurately target the required
position.
Fixes #631389.

2010-11-16 17:06:14 +0100 Mark Nauwelaerts <[email protected]>

* gst/audioparsers/gstbaseparse.c:
baseparse: refactor frame scanning peformed by _loop

2010-11-16 18:04:00 +0100 Mark Nauwelaerts <[email protected]>

* gst/audioparsers/gstbaseparse.c:
baseparse: slightly optimize sending of pending newsegment events

2010-11-16 17:04:35 +0100 Mark Nauwelaerts <[email protected]>

* gst/audioparsers/gstbaseparse.c:
baseparse: minor fixes and enhancements
Arrange for upstream as well as downstream flushing when seeking.
Also determine upstream size as well as seekability. Adjust some comments
to reality and employ debug statement in proper order.

2010-11-17 15:33:36 +0100 Mark Nauwelaerts <[email protected]>

* gst/audioparsers/gstaacparse.c:
aacparse: minor cleanups

2010-11-17 15:24:37 +0100 Mark Nauwelaerts <[email protected]>

* gst/audioparsers/gstaacparse.c:
aacparse: fix regression in ADIF src caps setting

2010-11-16 12:11:53 +0100 Mark Nauwelaerts <[email protected]>

* gst/audioparsers/gstflacparse.c:
* gst/audioparsers/gstflacparse.h:
flacparse: parse seektable
Fixes #631389 (partially).

2010-11-16 12:08:54 +0100 Mark Nauwelaerts <[email protected]>

* gst/audioparsers/gstflacparse.c:
flacparse: minor refactor and enable default baseparse segment clipping

2010-11-09 19:38:25 +0100 Mark Nauwelaerts <[email protected]>

* gst/audioparsers/gstmpegaudioparse.c:
mpegaudioparse: fix silly leak in _reset

2010-10-29 14:08:58 +0200 Mark Nauwelaerts <[email protected]>

* gst/audioparsers/gstbaseparse.c:
baseparse: use only upstream duration if it provides one

2010-10-25 14:15:50 +0200 Mark Nauwelaerts <[email protected]>

* gst/audioparsers/gstbaseparse.c:
baseparse: reflow update_bitrate code
... which makes local variables represent real state better, and avoids
triggering unneeded updates/actions.

2010-10-25 14:13:51 +0200 Mark Nauwelaerts <[email protected]>

* gst/audioparsers/gstbaseparse.c:
baseparse: add some debug statements

2010-10-19 23:25:54 +0100 Tim-Philipp Müller <[email protected]>

* gst/audioparsers/gstdcaparse.c:
dcaparse: init variable to make osx build bot happy
gstdcaparse.c: In function 'gst_dca_parse_check_valid_frame':
gstdcaparse.c:246: warning: 'best_sync' may be used uninitialized in this function

2010-10-19 00:15:20 +0100 Tim-Philipp Müller <[email protected]>

* gst/audioparsers/Makefile.am:
* gst/audioparsers/gstdcaparse.c:
* gst/audioparsers/gstdcaparse.h:
* gst/audioparsers/plugin.c:
audioparsers: add very basic dts/dca parser
Still some issues, e.g. with seekable queries in totem, but also
processing already-chunked input (created with matroskademux ! gdppay).

2010-10-14 16:48:21 +0200 Mark Nauwelaerts <[email protected]>

* gst/audioparsers/gstac3parse.c:
ac3parse: properly parse e-ac3 frame header
Also add a few debug statements.

2010-10-13 11:00:01 +0200 Mark Nauwelaerts <[email protected]>

* gst/audioparsers/gstflacparse.c:
flacparse: tweak setting buffer metadata; avoid timestamp jitter
Fixes #631993.

2010-10-12 18:07:49 +0200 Mark Nauwelaerts <[email protected]>

* gst/audioparsers/gstaacparse.c:
* gst/audioparsers/gstaacparse.h:
aacparse: streamline src caps setting
In particular, also set src caps whenever changes in stream warrant doing so.

2010-10-12 10:28:33 +0200 Sebastian Dröge <[email protected]>

* tests/check/elements/flacparse.c:
flacparse: Adjust unit tests to new flacparse behaviour
Garbage after frames is now included in the frames because flacparse
has no easy way to detect the real end of a frame. Decoders are
expected to everything after the frame because only decoding the
bitstream will reveal the real end of the frame.
Fixes bug #631814.

2010-10-12 10:27:53 +0200 Sebastian Dröge <[email protected]>

* gst/audioparsers/gstflacparse.c:
flacparse: Don't drop the last frame if it is followed by garbage
See bug #631814.

2010-10-11 17:49:46 +0200 Mark Nauwelaerts <[email protected]>

* gst/audioparsers/gstbaseparse.c:
baseparse: perform bitrate handling and posting after newsegment sending

2010-10-11 17:36:19 +0200 Mark Nauwelaerts <[email protected]>

* gst/audioparsers/gstbaseparse.c:
baseparse: immediately post subclass provided bitrate

2010-10-11 17:06:48 +0200 Mark Nauwelaerts <[email protected]>

* gst/audioparsers/gstflacparse.c:
flacparse: fix parsing with unknown framesizes
Fixes #631814 (mostly).

2010-10-07 23:37:36 +0200 Sebastian Dröge <[email protected]>

* gst/audioparsers/gstflacparse.c:
flacparse: Simplify frame header parsing by using lookup tables
Based on a patch by Felipe Contreras.
See bug #631200.

2010-10-07 23:28:08 +0200 Sebastian Dröge <[email protected]>

* gst/audioparsers/gstflacparse.c:
* gst/audioparsers/gstflacparse.h:
flacparse: Don't parse the complete FLAC frames but only look for valid frame headers
Thanks to Felipe Contreras for the suggestion. This is partially
based on his patches and makes flacparse more than 3.5 times faster.
Looking for valid frame headers is unlikely to give false positives
because every frame header is at least 9 bytes long, contains a
14 bit sync code and a 8 bit checksum over the first 8 bytes.
Fixes bug #631200.

2010-10-06 18:32:51 +0200 Sebastian Dröge <[email protected]>

* gst/audioparsers/gstflacparse.c:
flacparse: Really post tags only after the initial newsegment event
The first newsegment event will be send by the first call to
gst_base_parse_push_buffer() if necessary, posting the tags
before that is not a good idea. Instead do it from the
GstBaseParse::pre_push_buffer vfunc.

2010-10-05 11:17:52 +0100 Tim-Philipp Müller <[email protected]>

* gst/audioparsers/gstbaseparse.c:
Revert "baseparse: add skip property"
This reverts commit b5a3d60363d837a10f0533c141ec93d10b742312.
Reverting this for now, since no one really seems to remember why this
property exists or what it could possibly be good for. It seems to have
been in the original mp3parse since the beginning of time and was back-
ported from there.

2010-10-04 10:41:52 +0200 Sebastian Dröge <[email protected]>

* gst/audioparsers/gstflacparse.c:
flacparse: Fix uninitialized variable compiler warnings
These warnings are wrong, the variables are only used if they were
initialized by the bit reader.

2010-09-14 02:48:58 +0300 Felipe Contreras <[email protected]>

* gst/audioparsers/gstflacparse.c:
flacparse: fix picture parsing
Signed-off-by: Felipe Contreras <[email protected]>

2010-10-03 23:54:49 +0200 Sebastian Dröge <[email protected]>

* gst/audioparsers/gstflacparse.c:
flacparse: Push tags before the header buffers are pushed

2010-08-02 20:50:21 +0300 Felipe Contreras <[email protected]>

* gst/audioparsers/gstflacparse.c:
flacparse: trivial caps fix
Signed-off-by: Felipe Contreras <[email protected]>

2010-10-03 23:50:29 +0200 Sebastian Dröge <[email protected]>

* gst/audioparsers/gstbaseparse.c:
audioparser: Let the format string agree with the parameters to fix compiler warning

2010-10-03 15:41:20 +0200 Sebastian Dröge <[email protected]>

* gst/audioparsers/gstac3parse.c:
ac3parse: Use unchecked versions of the bitreader get functions
We didn't check the return values anyway...

2010-09-22 15:44:43 +0530 Arun Raghavan <[email protected]>

* gst/audioparsers/gstbaseparse.c:
baseparse: Fix debug output
We lose the reference to the buffer after gst_pad_push(), so the debug
print should happen before.
https://bugzilla.gnome.org/show_bug.cgi?id=622276

2010-10-01 12:34:55 +0200 Mark Nauwelaerts <[email protected]>

* tests/check/elements/flacparse.c:
* tests/check/elements/parser.c:
* tests/check/elements/parser.h:
audioparsers: add flacparse unit test
... and tweak parser test helper in the process.

2010-09-29 16:12:42 +0200 Mark Nauwelaerts <[email protected]>

* gst/audioparsers/gstbaseparse.c:
baseparse: support reverse playback
... in pull mode or upstream driven.

2010-09-27 12:16:43 +0200 Mark Nauwelaerts <[email protected]>

* gst/audioparsers/gstbaseparse.c:
baseparse: remove done TODOs and update documentation

2010-09-25 14:40:54 +0200 Mark Nauwelaerts <[email protected]>

* gst/audioparsers/gstbaseparse.c:
baseparse: use determined seekability in answering SEEKING query

2010-09-25 14:32:06 +0200 Mark Nauwelaerts <[email protected]>

* gst/audioparsers/gstbaseparse.c:
baseparse: add skip property

2010-09-25 13:59:39 +0200 Mark Nauwelaerts <[email protected]>

* tests/check/elements/ac3parse.c:
* tests/check/elements/mpegaudioparse.c:
audioparsers: add ac3parse and mpegaudioparse unit test

2010-09-25 13:59:18 +0200 Mark Nauwelaerts <[email protected]>

* gst/audioparsers/Makefile.am:
* gst/audioparsers/gstmpegaudioparse.c:
* gst/audioparsers/gstmpegaudioparse.h:
* gst/audioparsers/plugin.c:
mpegaudioparse: initial version
... adequately equivalent to mp3parse, so lets boldly set it
to higher rank.

2010-09-25 14:01:07 +0200 Mark Nauwelaerts <[email protected]>

* gst/audioparsers/gstaacparse.c:
aacparse: set minimum frame size at _start
... rather than one time at _init.

2010-09-25 13:50:51 +0200 Mark Nauwelaerts <[email protected]>

* tests/check/elements/aacparse.c:
* tests/check/elements/amrparse.c:
* tests/check/elements/parser.c:
* tests/check/elements/parser.h:
audioparsers: refactor existing unit tests using common helper

2010-09-22 15:07:09 +0200 Mark Nauwelaerts <[email protected]>

* gst/audioparsers/gstaacparse.c:
* gst/audioparsers/gstac3parse.c:
* gst/audioparsers/gstamrparse.c:
* gst/audioparsers/gstbaseparse.c:
* gst/audioparsers/gstbaseparse.h:
baseparse: use _set_frame_props to configure frame lead_in and lead_out
... provided a corresponding decoder with sufficient leading and following
frames to carry out full decoding for a particular segment.

2010-09-22 14:13:17 +0200 Mark Nauwelaerts <[email protected]>

* gst/audioparsers/gstaacparse.c:
* gst/audioparsers/gstac3parse.c:
* gst/audioparsers/gstamrparse.c:
* gst/audioparsers/gstbaseparse.c:
* gst/audioparsers/gstbaseparse.h:
* gst/audioparsers/gstflacparse.c:
baseparse: use _set_duration to configure duration update interval
... as it logically belongs there as one or the other; either subclass
can provide a duration, or an estimate must be made (reguarly updated).

2010-09-22 13:55:20 +0200 Mark Nauwelaerts <[email protected]>

* gst/audioparsers/gstbaseparse.c:
baseparse: localize use of provided fps information

2010-09-22 12:13:12 +0200 Mark Nauwelaerts <[email protected]>

* gst/audioparsers/gstbaseparse.c:
baseparse: seek table and accurate seek support

2010-09-21 13:57:10 +0200 Mark Nauwelaerts <[email protected]>

* gst/audioparsers/gstbaseparse.c:
baseparse: proper and more extended segment and seek handling
That is, loop pause handling, segment seek support, newsegment for gaps, etc

2010-09-21 10:57:04 +0200 Mark Nauwelaerts <[email protected]>

* gst/audioparsers/gstbaseparse.c:
* gst/audioparsers/gstbaseparse.h:
baseparse: add index support

2010-09-21 09:59:56 +0200 Mark Nauwelaerts <[email protected]>

* gst/audioparsers/gstbaseparse.c:
baseparse: refactor state reset

2010-09-20 16:39:37 +0200 Mark Nauwelaerts <[email protected]>

* gst/audioparsers/gstbaseparse.c:
baseparse: prevent indefinite resyncing

2010-09-20 13:57:55 +0200 Mark Nauwelaerts <[email protected]>

* gst/audioparsers/gstbaseparse.c:
baseparse: specific EOS handling if no output so far

2010-09-20 13:31:57 +0200 Mark Nauwelaerts <[email protected]>

* gst/audioparsers/gstbaseparse.c:
baseparse: adjust _set_frame_prop documentation and set default as claimed

2010-09-20 13:30:54 +0200 Mark Nauwelaerts <[email protected]>

* gst/audioparsers/gstbaseparse.c:
baseparse: fix bitrate copy-and-paste and update heuristic

2010-09-17 18:33:29 +0200 Mark Nauwelaerts <[email protected]>

* gst/audioparsers/gstbaseparse.c:
baseparse: post duration message if average bitrates is updated

2010-09-17 18:24:22 +0200 Mark Nauwelaerts <[email protected]>

* gst/audioparsers/gstaacparse.c:
* gst/audioparsers/gstbaseparse.c:
* gst/audioparsers/gstbaseparse.h:
baseparse: remove is_seekable vmethod and use a set_seek instead
Seekability, like duration, etc is unlikely to change (frequently), and
the default assumption covers most cases, so let subclass set when needed.
At the same time, allow subclass to indicate if it has seek-metadata (table)
available, and possibly have it provide an average bitrate.

2010-09-17 17:35:40 +0200 Mark Nauwelaerts <[email protected]>

* gst/audioparsers/gstac3parse.c:
ac3parse: remove redundant default is_seekable

2010-09-17 17:21:46 +0200 Mark Nauwelaerts <[email protected]>

* gst/audioparsers/gstbaseparse.c:
* gst/audioparsers/gstbaseparse.h:
baseparse: add another hook for subclass prior to pushing buffer
... and allow subclass to perform custom segment clipping, or to
emit tags or messages at this time.

2010-09-17 17:19:37 +0200 Mark Nauwelaerts <[email protected]>

* gst/audioparsers/gstbaseparse.c:
baseparse: 0 converts to 0 by default

2010-09-16 18:56:46 +0200 Mark Nauwelaerts <[email protected]>

* gst/audioparsers/gstbaseparse.c:
* gst/audioparsers/gstbaseparse.h:
basepase: refactor conversion using helper function and export default convert

2010-09-16 18:35:47 +0200 Mark Nauwelaerts <[email protected]>

* gst/audioparsers/gstbaseparse.c:
baseparse: streamline query handling

2010-09-16 11:51:20 +0200 Mark Nauwelaerts <[email protected]>

* gst/audioparsers/gstbaseparse.c:
* gst/audioparsers/gstbaseparse.h:
baseparse: cleanup struct and remove unused member

2010-08-16 11:04:37 +0200 Mark Nauwelaerts <[email protected]>

* gst/audioparsers/plugin.c:
audioparsers: increase ranks to enable auto-plugging
Because we can, and should, have some shakedown testing before having
these make it into -good later on ...

2010-09-22 16:07:24 +0530 Arun Raghavan <[email protected]>

* gst/audioparsers/gstbaseparse.c:
baseparse: Allow chaining of subclass event handlers
This allows the child class to chain its event handler with
GstBaseParse, so that subclasses don't have to duplicate all the default
event handling logic.
https://bugzilla.gnome.org/show_bug.cgi?id=622276

2010-08-27 18:35:10 +0200 Sebastian Dröge <[email protected]>

* gst/audioparsers/gstbaseparse.c:
baseparse: Don't use GST_FLOW_IS_FATAL()
Also don't post an error message for UNEXPECTED and do it
for NOT_LINKED.

2010-09-06 14:12:00 +0200 Mark Nauwelaerts <[email protected]>

* gst/audioparsers/gstbaseparse.c:
baseparse: non-TIME seek event is simply not handled

2010-06-15 15:34:05 +0200 Mark Nauwelaerts <[email protected]>

* gst/audioparsers/gstbaseparse.c:
baseparse: fix seek event ref handling

2010-06-15 15:33:37 +0200 Mark Nauwelaerts <[email protected]>

* gst/audioparsers/gstbaseparse.c:
baseparse: prevent arithmetic overflows in pull mode buffer cache handling

2010-06-15 15:32:34 +0200 Mark Nauwelaerts <[email protected]>

* gst/audioparsers/gstbaseparse.c:
baseparse: fix seek handling
Allow a few more seek event type combinations, and really use the result
of gst_segment_set_seek to perform the seek. Also add some debug.

2010-04-12 18:07:29 +0200 Edward Hervey <[email protected]>

* tests/check/elements/aacparse.c:
* tests/check/elements/amrparse.c:
check: Don't re-declare 'GList *buffers' in the tests
It's an external which lives in gstcheck.c. Redeclaring it makes some
compilers/architectures think the 'buffers' in the individual tests are
a different symbol... and therefore we end up comparing holodecks with
oranges.

2010-03-26 18:56:49 +0000 Arun Raghavan <[email protected]>

* gst/audioparsers/gstbaseparse.c:
baseparse: Don't emit bitrate tags too early
We wait to parse a minimum number of frames (10, arbitrarily) before
emiting bitrate tags so that our early estimates are not wildly
inaccurate for streams that start with a silence. If the stream ends
before that, we just emit the tags anyway.
While it _would_ be nicer to be specify the threshold to start pushing
the tags in terms of duration, this would introduce more complexity than
this merits.
https://bugzilla.gnome.org/show_bug.cgi?id=614991

2010-03-26 18:58:35 +0100 Sebastian Dröge <[email protected]>

* gst/audioparsers/gstflacparse.c:
* gst/audioparsers/gstflacparse.h:
flacparse: Optionally check the overall frame checksums too before accepting a frame as valid
This is optional because it's a quite expensive operation and it's very
unlikely that a non-frame is detected as frame after the header CRC check
and checking all bits for valid values. The overall frame checksums are
mainly useful to detect inconsistencies in the encoded payload.

2010-03-26 18:42:28 +0100 Sebastian Dröge <[email protected]>

* gst/audioparsers/gstflacparse.c:
flacparse: Check the CRC-8 of the headers before accepting a frame as valid
This makes false-positives during seeking much less likely and detection of
them much faster.

2010-03-26 18:20:24 +0100 Sebastian Dröge <[email protected]>

* gst/audioparsers/gstbaseparse.c:
baseparse: Set the last stop to the buffer starttime if the duration is invalid
...instead of not setting it at all.

2010-03-26 18:19:00 +0100 Joshua M. Doe <[email protected]>

* gst/audioparsers/gstbaseparse.c:
baseparse: Send NEWSEGMENT event with correct start and position
Instead of taking the last stop (which could be buffer endtime instead
of starttime) always take the buffer starttime.
Fixes bug #614016.

2010-03-26 16:49:01 +0000 Arun Raghavan <[email protected]>

* gst/audioparsers/gstflacparse.c:
flacparse: Fix buffer refcount issue
When called from the GST_FLAC_PARSE_STATE_HEADERS case,
gst_flac_parse_hand_headers() does a gst_buffer_set_caps() on a buffer
with refcount > 1. This change handles this case by making the buffer
metadata_Writable.
https://bugzilla.gnome.org/show_bug.cgi?id=614037

2010-03-25 17:09:17 +0000 Tim-Philipp Müller <[email protected]>

* gst/audioparsers/gstbaseparse.c:
* gst/audioparsers/gstbaseparse.h:
audioparsers: remove unused GstBaseParseClassPrivate structure

2010-03-25 12:55:02 +0000 Arun Raghavan <[email protected]>

* gst/audioparsers/gstflacparse.c:
flacparse: Make bitrate estimation more accurate
This implements the get_frame_overhead() vfunc so that baseparse can
make more accurate bitrate estimates.

2010-03-25 11:48:46 +0000 Arun Raghavan <[email protected]>

* gst/audioparsers/gstaacparse.c:
aacparse: Fix bitrate calculation
This patch adds the get_frame_overhead() vfunc so that baseparse can
accurately calculate the min/avg/max bitrates for aacparse.
Note: The bitrate was being incorrectly calculated for ADTS streams
(it's not in the header as the code suggests).

2010-03-25 11:22:58 +0000 Arun Raghavan <[email protected]>

* gst/audioparsers/gstbaseparse.c:
* gst/audioparsers/gstbaseparse.h:
audioparsers: Add bitrate calculation to baseparse
This makes baseparse keep a running average of the stream bitrate, as
well as the minimum and maximum bitrates. Subclasses can override a
vfunc to make sure that per-frame overhead from the container is not
accounted for in the bitrate calculation.
We take care not to override the bitrate, minimum-bitrate, and
maximum-bitrate tags if they have been posted upstream. We also
rate-limit the emission of bitrate so that it is only triggered by a
change of >10 kbps.

2010-03-22 16:56:03 +0100 Benjamin Otte <[email protected]>

* tests/check/elements/amrparse.c:
Add -Wold-style-definition
and fix the warnings

2010-03-21 21:39:18 +0100 Benjamin Otte <[email protected]>

* tests/check/elements/aacparse.c:
* tests/check/elements/amrparse.c:
Add -Wmissing-declarations -Wmissing-prototypes to configure flags
And fix all warnings

2010-03-18 17:30:26 +0100 Benjamin Otte <[email protected]>

* gst/audioparsers/gstaacparse.c:
* gst/audioparsers/gstamrparse.c:
gst_element_class_set_details => gst_element_class_set_details_simple

2010-01-14 11:50:33 +0100 Mark Nauwelaerts <[email protected]>

* gst/audioparsers/gstbaseparse.c:
audioparsers: rename baseparse GType name to avoid possible conflicts

2010-01-12 18:55:53 +0100 Edward Hervey <[email protected]>

* gst/audioparsers/gstflacparse.c:
flacparse: Initialize variables.
Fixes build on $#@*( macosx

2010-01-11 22:41:57 +0300 �� <[email protected]>

* gst/audioparsers/gstaacparse.c:
* gst/audioparsers/gstamrparse.c:
win32: Include config.h before anything else. Fix mpegdemux LIBADD
Because config.h defines __MSVCRT_VERSION__, which should be defined
before inclusion of any system header.
Also fixes mpegdemux Makefile.am LIBADD typo.
Fixes #606665

2010-01-11 13:20:26 -0300 Thiago Santos <[email protected]>

* gst/audioparsers/gstaacparse.c:
aacparse: Also add stream-format to template caps
Do not forget to add stream-format to template caps
off aacparse

2010-01-11 13:13:41 -0300 Thiago Santos <[email protected]>

* gst/audioparsers/gstaacparse.c:
* tests/check/elements/aacparse.c:
Rename aac's stream-format 'none' to 'raw'
Renames aac's stream-format from previous commits from none to
raw

2010-01-11 12:10:02 -0300 Thiago Santos <[email protected]>

* tests/check/elements/aacparse.c:
aacparse: update tests to stream-format changes
Updates aacparse unit tests to check for stream-format
correctness as well.

2010-01-11 10:51:18 -0300 Thiago Santos <[email protected]>

* gst/audioparsers/gstaacparse.c:
aacparse: Add stream-format to output caps
Adds stream-format field to output caps

2010-01-05 15:05:05 +0100 Mark Nauwelaerts <[email protected]>

* gst/audioparsers/gstaacparse.c:
* gst/audioparsers/gstamrparse.c:
* gst/audioparsers/gstbaseparse.c:
audioparsers: documentation fixes

2010-01-05 15:04:38 +0100 Mark Nauwelaerts <[email protected]>

* gst/audioparsers/gstac3parse.c:
ac3parse: add documentation

2010-01-05 14:48:49 +0100 Mark Nauwelaerts <[email protected]>

* gst/audioparsers/gstflacparse.c:
* gst/audioparsers/gstflacparse.h:
flacparse: add documentation

2009-12-21 18:29:43 +0100 Mark Nauwelaerts <[email protected]>

* gst/audioparsers/gstflacparse.c:
flacparse: perform additional frame checks when resyncing

2010-01-05 16:35:52 +0100 Mark Nauwelaerts <[email protected]>

* gst/audioparsers/gstflacparse.c:
flacparse: fix (multiple channel) frame parsing

2010-01-05 16:35:44 +0100 Mark Nauwelaerts <[email protected]>

* gst/audioparsers/gstflacparse.c:
flacparse: declare unparsed input and parsed output

2009-12-21 18:19:23 +0100 Mark Nauwelaerts <[email protected]>

* gst/audioparsers/gstac3parse.c:
ac3parse: fix scanning for next syncword

2009-12-21 18:18:39 +0100 Mark Nauwelaerts <[email protected]>

* gst/audioparsers/gstbaseparse.c:
baseparse: adjust seek handling and newsegment sending
Perform sanity check on type of seek, and only perform one that is
appropriately supported. Adjust downstream newsegment event
to first buffer timestamp that is sent downstream.

2009-12-21 11:59:45 +0100 Mark Nauwelaerts <[email protected]>

* gst/audioparsers/gstbaseparse.c:
baseparse: minor refactor cleanup
Also add some debug logging.

2009-12-18 21:05:11 +0100 Mark Nauwelaerts <[email protected]>

* gst/audioparsers/gstflacparse.c:
flacparse: locate next sync code more efficiently

2009-12-18 21:04:12 +0100 Mark Nauwelaerts <[email protected]>

* gst/audioparsers/gstflacparse.c:
flacparse: baseparse takes care of handling leftover pieces

2009-12-18 21:02:40 +0100 Mark Nauwelaerts <[email protected]>

* gst/audioparsers/gstbaseparse.c:
baseparse: implement leftover draining in pull mode

2009-12-17 12:45:36 +0100 Mark Nauwelaerts <[email protected]>

* gst/audioparsers/gstflacparse.c:
flacparse: set _OFFSET and _OFFSET_END on outgoing buffers

2009-12-17 12:44:20 +0100 Mark Nauwelaerts <[email protected]>

* gst/audioparsers/Makefile.am:
* gst/audioparsers/gstflacparse.c:
* gst/audioparsers/gstflacparse.h:
* gst/audioparsers/plugin.c:
audioparsers: move 'flacparse' into it

2009-12-16 18:38:33 +0100 Mark Nauwelaerts <[email protected]>

* gst/audioparsers/gstbaseparse.c:
baseparse: provide default conversion using bps if no fps available
Also store estimated duration as such, rather than pretending otherwise
(e.g. set by subclass).

2009-12-18 13:30:29 +0100 Mark Nauwelaerts <[email protected]>

* gst/audioparsers/gstbaseparse.c:
baseparse: check for remaining data when draining in push mode

2009-12-18 13:30:07 +0100 Mark Nauwelaerts <[email protected]>

* gst/audioparsers/gstbaseparse.c:
baseparse: fix pull mode cache size comparison

2009-12-18 13:01:17 +0100 Edward Hervey <[email protected]>

* gst/audioparsers/gstac3parse.c:
ac3parse: Fix unitialized variable.

2009-12-17 14:46:01 +0000 Christian Schaller <[email protected]>

* gst/audioparsers/Makefile.am:
Update spec file and fix ac3parser header listing in Makefile.am

2009-12-11 10:25:16 -0800 Michael Smith <[email protected]>

* gst/audioparsers/gstbaseparse.c:
audioparse: fix a format string as reported on irc.

2009-11-23 16:34:50 +0100 Mark Nauwelaerts <[email protected]>

* gst/audioparsers/gstac3parse.c:
ac3parse: ensure sufficient data available for parsing

2009-10-29 15:19:04 +0100 Mark Nauwelaerts <[email protected]>

* gst/audioparsers/gstac3parse.c:
ac3parse: extract and use some more details for Enhanced Ac-3 streams

2009-10-29 15:18:37 +0100 Mark Nauwelaerts <[email protected]>

* gst/audioparsers/gstbaseparse.c:
* gst/audioparsers/gstbaseparse.h:
baseparse: custom bufferflag indicates not to count frame in stats

2009-10-28 14:08:43 +0100 Mark Nauwelaerts <[email protected]>

* gst/audioparsers/gstac3parse.c:
ac3parse: perform additional frame checks when resyncing

2009-10-28 14:07:17 +0100 Mark Nauwelaerts <[email protected]>

* gst/audioparsers/gstac3parse.c:
ac3parse: inform base parser of frame duration

2009-10-27 16:16:50 +0100 Mark Nauwelaerts <[email protected]>

* gst/audioparsers/gstac3parse.c:
ac3parse: improve src caps settings

2009-11-27 17:59:03 +0100 Mark Nauwelaerts <[email protected]>

* gst/audioparsers/Makefile.am:
* gst/audioparsers/gstac3parse.c:
* gst/audioparsers/gstac3parse.h:
* gst/audioparsers/plugin.c:
ac3parse: initial version
MARGINAL rank for now; might take some time for some (useful)
framed=true/false to appear here and there.

2009-11-26 18:34:45 +0100 Mark Nauwelaerts <[email protected]>

* gst/audioparsers/gstamrparse.c:
* gst/audioparsers/gstamrparse.h:
amrparse: use (default) time handling of baseparser class

2009-11-26 18:15:21 +0100 Mark Nauwelaerts <[email protected]>

* gst/audioparsers/Makefile.am:
* gst/audioparsers/gstamrparse.c:
* gst/audioparsers/gstamrparse.h:
* gst/audioparsers/plugin.c:
audioparsers: move 'amrparse' into it

2009-11-27 17:27:32 +0100 Mark Nauwelaerts <[email protected]>

* gst/audioparsers/gstbaseparse.c:
audioparsers: reference GstBaseParse now lives here

2009-11-28 18:13:31 +0100 Mark Nauwelaerts <[email protected]>

* gst/aacparse/Makefile.am:
* gst/aacparse/gstaacparse.c:
* gst/aacparse/gstaacparse.h:
* gst/aacparse/gstbaseparse.c:
* gst/aacparse/gstbaseparse.h:
* gst/aacparse/plugin.c:
* gst/audioparsers/Makefile.am:
* gst/audioparsers/gstaacparse.c:
* gst/audioparsers/gstaacparse.h:
* gst/audioparsers/gstbaseparse.c:
* gst/audioparsers/gstbaseparse.h:
* gst/audioparsers/plugin.c:
audioparsers: rename 'aacparse' plugin to generic 'audioparsers' plugin

2009-11-26 17:04:43 +0100 Mark Nauwelaerts <[email protected]>

* gst/aacparse/Makefile.am:
* gst/aacparse/gstaacparse.c:
* gst/aacparse/plugin.c:
aacparse: separate plugin registration and rename plugin

2009-11-26 17:04:36 +0100 Mark Nauwelaerts <[email protected]>

* gst/aacparse/gstaacparse.c:
aacparse: ensure sufficient data available before accessing

2009-11-05 14:31:40 +0100 Mark Nauwelaerts <[email protected]>

* gst/aacparse/gstaacparse.c:
* gst/aacparse/gstaacparse.h:
aacparse: use (default) time handling of baseparser class

2009-10-29 15:19:35 +0100 Mark Nauwelaerts <[email protected]>

* gst/aacparse/gstaacparse.c:
aacparse: fixup comments to C-style

2009-10-29 16:05:00 +0100 Mark Nauwelaerts <[email protected]>

* gst/aacparse/gstbaseparse.c:
baseparse: reset passthrough mode to default (disabled) on activation

2009-10-29 15:16:59 +0100 Mark Nauwelaerts <[email protected]>

* gst/aacparse/gstbaseparse.c:
baseparse: ensure buffer metadata is writable

2009-10-28 14:06:13 +0100 Mark Nauwelaerts <[email protected]>

* gst/aacparse/gstbaseparse.c:
* gst/aacparse/gstbaseparse.h:
baseparse: fix/enhance DISCONT marking
In particular, consider DISCONT == !sync, and allow subclass to query
sync state, as it may want to perform additional checks depending
on whether sync was achieved earlier on.
Also arrange for subclass to query whether leftover data is being drained.

2009-11-23 15:48:25 +0100 Mark Nauwelaerts <[email protected]>

* gst/aacparse/gstbaseparse.c:
* gst/aacparse/gstbaseparse.h:
baseparse: add timestamp handling, and default conversion
In particular, (optionally) provide baseparse with a notion of frames per second
(and therefore also frame duration) and have it track frame and byte counts.
This way, subclass can provide baseparse with fps and have it provide default
buffer time metadata and conversions, though subclass can still install
callbacks to handle such itself.

2009-10-28 12:02:03 +0100 Mark Nauwelaerts <[email protected]>

* gst/aacparse/gstbaseparse.c:
baseparse: documentation fixes

2009-10-28 12:00:08 +0100 Mark Nauwelaerts <[email protected]>

* gst/aacparse/gstbaseparse.c:
baseparse: use_fixed_caps for src pad
After all, stream is as-is, and there is little molding to downstream's
taste that can be done. If subclass can and wants to do so, it can
still override as such.

2009-11-20 17:32:13 +0100 Julien Moutte <[email protected]>

* gst/aacparse/gstbaseparse.c:
aacparse: Fix compilation warnings

2009-10-11 11:22:11 +0200 Josep Torra <[email protected]>

* gst/aacparse/gstaacparse.c:
* gst/aacparse/gstbaseparse.c:
aacparse: fix warnings in macosx snow leopard

2009-09-25 17:02:53 +0200 Mark Nauwelaerts <[email protected]>

* gst/aacparse/gstaacparse.c:
* gst/aacparse/gstbaseparse.c:
* gst/aacparse/gstbaseparse.h:
aacparse: forego (bogus) parsing of already parsed (raw) input

2009-08-07 13:07:17 +0200 Mark Nauwelaerts <[email protected]>

* gst/aacparse/gstbaseparse.c:
baseparse: prevent infinite loop when draining

2009-08-07 13:06:28 +0200 Mark Nauwelaerts <[email protected]>

* gst/aacparse/gstbaseparse.c:
baseparse: fix minor memory leak

2009-07-14 14:08:04 +0200 Sebastian Dröge <[email protected]>

* gst/aacparse/gstbaseparse.c:
* gst/aacparse/gstbaseparse.h:
aacparse: Add function for the baseparse subclass to push buffers downstream
Also handle the case gracefully where the subclass decides to drop
the first buffers and has no caps set yet. It's still required to
have valid caps set when the first buffer should be passed downstream.

2009-07-14 14:07:44 +0200 Sebastian Dröge <[email protected]>

* gst/aacparse/gstbaseparse.c:
baseparse: Fix seek event leaking

2009-06-18 12:13:28 +0200 Mark Nauwelaerts <[email protected]>

* gst/aacparse/gstaacparse.c:
aacparse: ADIF: do not send bogus timestamps, leave to downstream (decoder)

2009-06-01 15:53:27 +0100 Tim-Philipp Müller <[email protected]>

* gst/aacparse/gstaacparse.c:
aacparse: fix sample rate extraction from codec data
In one case we extracted the sample rate index from the codec data
and saved it as sample rate rather than getting the real sample
rate from the table. Fix that, and also make sure we don't access
non-existant table entries by adding a small helper function that
guards against out-of-bounds access in case of invalid input data.

2009-06-01 14:02:33 +0100 Tim-Philipp Müller <[email protected]>

* gst/aacparse/gstaacparse.c:
aacparse, amrparse: remove bogus gst_pad_fixate_caps() calls

2009-06-01 13:56:18 +0100 Tim-Philipp Müller <[email protected]>

* gst/aacparse/gstbaseparse.c:
baseparse: propagate return value of GstBaseParse::set_sink_caps()
gst_base_parse_sink_setcaps() presumably should fail if the subclass
returns FALSE from its ::set_sink_caps() function.

2009-06-01 13:47:01 +0100 Tim-Philipp Müller <[email protected]>

* gst/aacparse/gstbaseparse.c:
baseparse: don't try to GST_LOG an already-freed caps string
The proper way to log caps is via GST_PTR_FORMAT anyway.

2009-06-01 13:05:35 +0100 Tim-Philipp Müller <[email protected]>

* gst/aacparse/gstaacparse.c:
* tests/check/elements/aacparse.c:
aacparse: set channels and rate on output caps, and keep codec_data
Create output caps from input caps, so we maintain any fields we
might get on the input caps, such as codec_data or rate and channels.
Set channels and rate on the output caps if we don't have input caps
or they don't contain such fields. We do this partly because we can,
but also because some muxers need this information. Tagreadbin will
also be happy about this.

2009-05-26 19:43:53 +0200 Mark Nauwelaerts <[email protected]>

* gst/aacparse/gstbaseparse.c:
baseparse: fix debug category

2009-04-27 22:39:15 +0200 Mark Nauwelaerts <[email protected]>

* gst/aacparse/gstbaseparse.c:
baseparse: fix (regression in) newsegment handling
(aacparse, amrparse, flacparse). Fixes #580133.

2009-04-07 04:53:02 +0300 René Stadler <[email protected]>

* gst/aacparse/gstbaseparse.c:
baseparse: Fix slightly broken buffer-in-segment check (aacparse, amrparse, flacparse)

2009-04-05 03:50:19 +0300 René Stadler <[email protected]>

* gst/aacparse/gstbaseparse.c:
baseparse: Fix push mode seeking (aacparse, amrparse)
Sending the flush-start event forward before taking the stream lock actually
works, in contrast to deadlocking in downstream preroll_wait (hunk 1).
After that we get the chain function being stuck in a busy loop. This is fixed
by updating the minimum frame size inside the synchronization loop because the
subclass asks for more data in this way (hunk 2).
Finally, this leads to a very probable crash because the subclass can find a
valid frame with a size greater than the currently available data in the
adapter. This makes the subsequent gst_adapter_take_buffer call return NULL,
which is not expected (hunk 3).

2009-03-31 16:07:46 +0200 Mark Nauwelaerts <[email protected]>

* gst/aacparse/gstbaseparse.c:
baseparse: Delay newsegment as long as possible.
If newsegment is sent (too) early, caps may not yet be fixed/set,
and downstream may not have been linked.

2009-03-19 01:17:25 +0200 René Stadler <[email protected]>

* gst/aacparse/gstaacparse.c:
aacparse: Fix busyloop when seeking. Fixes #575388
The problem is that after a discont, set_min_frame_size(1024) is called when
detect_stream returns FALSE. However, detect_stream calls check_adts_frame
which sets the frame size on its own to something larger than 1024. This is the
same situation as in the beginning, so the base class ends up calling
check_valid_frame in an endless loop.

2009-03-19 00:32:40 +0200 René Stadler <[email protected]>

* gst/aacparse/gstaacparse.c:
aacparse: Refactor check_valid_frame to expose broken code
Just moving code around and removing an unhelpful/misleading comment.

2009-02-27 11:24:37 +0200 Stefan Kost <[email protected]>

* gst/aacparse/gstbaseparse.c:
baseparse: revert last change and properly fix
Baseparse internaly breaks the semantics of a _chain function by calling it with
buffer==NULL. The reson I belived it was okay to remove it was that there is
also an unchecked access to buffer later in _chain. Actually that code is wrong,
as it most probably wants to set discont on the outgoing buffer.

2009-02-26 11:02:06 +0200 Stefan Kost <[email protected]>

* gst/aacparse/gstbaseparse.c:
baseparse: remove checks for buffer==NULL
Accordifn to docs for GstPadChainFunction buffer cannot be NULL. If we would
leave the check, we would also need more such check below.

2009-02-11 00:15:43 +0200 René Stadler <[email protected]>

* gst/aacparse/gstaacparse.c:
aacparse: Fix license specified in plugin details.

2009-01-30 18:18:10 +0000 Jan Schmidt <[email protected]>

* gst/aacparse/gstbaseparse.c:
Fix the return value of the default parse_frame function.
Fix the return value of the default parse_frame function in both
copies of GstBaseParse

2009-01-23 16:00:10 +0200 Stefan Kost <[email protected]>

* gst/aacparse/gstaacparse.c:
Log aac details found in codec_data.

2008-11-13 17:24:58 +0000 Wim Taymans <[email protected]>

gst/aacparse/gstaacparse.c: Don't autoplug aacparse until it works.
Original commit message from CVS:
* gst/aacparse/gstaacparse.c: (plugin_init):
Don't autoplug aacparse until it works.

2008-11-13 15:20:15 +0000 Stefan Kost <[email protected]>

tests/check/: Add unit tests for new parsers.
Original commit message from CVS:
* tests/check/Makefile.am:
* tests/check/elements/aacparse.c:
* tests/check/elements/amrparse.c:
Add unit tests for new parsers.

2008-11-13 14:21:39 +0000 Stefan Kost <[email protected]>

gst/: Fix baseparse type name.
Original commit message from CVS:
* gst/aacparse/gstbaseparse.c:
* gst/amrparse/gstbaseparse.c:
Fix baseparse type name.

2008-11-13 12:59:34 +0000 Stefan Kost <[email protected]>

Add two new baseparse based parsers (aac and amr) from Bug #518857.
Original commit message from CVS:
* configure.ac:
* gst/aacparse/Makefile.am:
* gst/aacparse/gstaacparse.c:
* gst/aacparse/gstaacparse.h:
* gst/aacparse/gstbaseparse.c:
* gst/aacparse/gstbaseparse.h:
* gst/amrparse/Makefile.am:
* gst/amrparse/gstamrparse.c:
* gst/amrparse/gstamrparse.h:
* gst/amrparse/gstbaseparse.c:
* gst/amrparse/gstbaseparse.h:
Add two new baseparse based parsers (aac and amr) from Bug #518857.

2011-03-20 01:08:38 +0100 Havard Graff <[email protected]>

* gst/rtpmanager/gstrtpjitterbuffer.c:
jitterbuffer: Make src_query MT-safe
It is possible that the element might be going down while the event arrives

2011-04-08 15:22:47 +0200 Sebastian Dröge <[email protected]>

* ext/jpeg/gstjpegdec.c:
jpegdec: Unref event if the parent element disappeared

2011-04-08 15:22:19 +0200 Sebastian Dröge <[email protected]>

* gst/rtpmanager/gstrtpjitterbuffer.c:
jitterbuffer: Unref event if the parent element disappeared

2011-03-21 16:04:34 +0100 Havard Graff <[email protected]>

* ext/jpeg/gstjpegdec.c:
jpegdec: Make upstream events MT-safe

2011-03-21 16:04:34 +0100 Havard Graff <[email protected]>

* gst/rtpmanager/gstrtpjitterbuffer.c:
jitterbuffer: Make upstream events MT-safe

2011-04-08 15:20:51 +0200 Sebastian Dröge <[email protected]>

* gst/rtpmanager/gstrtpjitterbuffer.c:
* gst/rtpmanager/gstrtpptdemux.c:
* gst/rtpmanager/gstrtpsession.c:
* gst/rtpmanager/gstrtpssrcdemux.c:
rtp: Unref events if the parent element disappeared

2011-01-06 18:24:36 +0100 Ole André Vadla Ravnås <[email protected]>

* gst/rtpmanager/gstrtpjitterbuffer.c:
* gst/rtpmanager/gstrtpptdemux.c:
* gst/rtpmanager/gstrtpsession.c:
* gst/rtpmanager/gstrtpssrcdemux.c:
rtpmanager: fix pad callbacks so they handle when parent goes away
1) We need to lock and get a strong ref to the parent, if still there.
2) If it has gone away, we need to handle that gracefully.
This is necessary in order to safely modify a running pipeline. Has been
observed when a streaming thread is doing a buffer_alloc() while an
application thread sends an event on a pad further downstream, and from
within a pad probe (holding STREAM_LOCK) carries out the pipeline plumbing
while the streaming thread has its buffer_alloc() in progress.

2010-11-26 15:20:04 +0100 Havard Graff <[email protected]>

* gst/rtpmanager/gstrtpsession.c:
rtpsession: make iterate_internal_links MT-safe

2011-04-08 14:35:04 +0200 Sebastian Dröge <[email protected]>

* ext/pulse/pulsesink.c:
Revert "Pulsesink: Allow chunks up to bufsize instead of segsize"
This reverts commit 1e2c1467ae042a3c6bb1a6bc0c07aeff13ec5edb.
The commit causes pulsesink to ignore the latency-time baseaudiosink property.

2011-04-08 11:13:07 +0200 Alexey Fisher <[email protected]>

* gst/rtp/gstrtpspeexpay.c:
rtpspeexpay: Do not transmitt samples with GAP flag
If we get GAP samples, there is no need to transmitt it.
In some situations, microphone is muted, we can drop net traffick
usage to ~1 kbit/s. Without patch it will stay ~20 kbit/s

2011-04-08 11:11:58 +0200 Alexey Fisher <[email protected]>

* ext/speex/gstspeexenc.c:
speexenc: Use speex intern silence detection
Speex has build in silence detection. If speex_encode_int returns 0,
than there is silence and sample do not need to be transmitted.
This work only if vbr=1 and dtx=1 optionas are enabled.
So if we get 0, we add GAP flag to the sample.

2011-04-05 17:12:28 +0200 Wim Taymans <[email protected]>

* gst/rtsp/gstrtspsrc.c:
rtspsrc: handle * control correctly
Parse session control attributes when no media control attribute is
present. Threat * control attributes as an empty string, just like the
spec says.
Fixes #646800

2011-04-05 14:28:54 +0200 Sebastian Dröge <[email protected]>

* gst/matroska/matroska-mux.c:
matroskamux: Add support for A-Law and µ-Law
Fixes bug #646567.

2011-04-05 09:44:01 +0200 Jon Nordby <[email protected]>

* configure.ac:
* ext/jack/gstjackaudiosink.c:
* ext/jack/gstjackaudiosrc.c:
jack: Fix build with jack 0.120.1
9544622674c0d0a3147a9b51145159b02eec68e9 checked
for 0.120.2 and later, but the deprecation was introduced in
0.120.1

2011-04-05 12:05:19 +0300 Stefan Kost <[email protected]>

* sys/v4l2/gstv4l2radio.h:
* sys/v4l2/gstv4l2src.h:
* sys/v4l2/gstv4l2xoverlay.c:
docs: fix docuemntation warnings (and reindent)

2011-04-04 17:34:17 +0200 Alessandro Decina <[email protected]>

* gst/videomixer/blendorc-dist.c:
* gst/videomixer/blendorc-dist.h:
videomixer: update orc dist files

2011-04-04 15:57:10 +0300 Stefan Kost <[email protected]>

* common:
Automatic update of common submodule
From 1ccbe09 to c3cafe1

2011-03-01 14:08:12 +0530 Arun Raghavan <[email protected]>

* ext/pulse/pulsesink.c:
pulsesink: Always call pa_stream_new_with_proplist()
pa_stream_new_with_proplist() can take a NULL proplist, so we don't need
to concern ourselves with whether it's NULL or not.

2011-04-04 11:33:10 +0200 Mark Nauwelaerts <[email protected]>

* gst/rtsp/gstrtspsrc.c:
rtspsrc: perform post-flush state tricks downstream to upstream
... so downstream is set when upstream resumes data flow.

2011-04-04 11:27:29 +0200 Mark Nauwelaerts <[email protected]>

* gst/rtsp/gstrtspsrc.c:
rtspsrc: distribute new base_time to manager children following flush seek
... by forcing a state changed to PLAYING, which should otherwise be a
no-op as elements should already be in that state.
In particular, jitterbuffer needs new base_time as soon as possible to perform
proper timing (e.g. eos timeout handling) and can't wait for the new base_time
that will be distributed when the whole pipeline returns to PLAYING.
See bug #646397.

2011-04-04 11:35:59 +0200 Mark Nauwelaerts <[email protected]>

* gst/rtpmanager/gstrtpjitterbuffer.c:
Revert "jitterbuffer: reset element base_time upon flush"
This reverts commit f84b8a69cba9c538f5546869cb4ef454ad5efb9d.
Fixes bug #646397.

2011-04-04 10:31:44 +0100 Zaheer Abbas Merali <[email protected]>

* gst/flv/gstflvdemux.c:
* gst/flv/gstflvmux.c:
flv: Specify the only possible stream-format for h264 in the pad templates.

2011-04-04 10:07:42 +0200 Sebastian Dröge <[email protected]>

* gst/qtdemux/qtdemux.c:
qtdemux: Check for invalid (empty) classification info entity strings
Otherwise the classification string can be empty and gst_tag_list_add() will
complain or have a \0 in the first four bytes, which is wrong too.

2011-04-04 10:01:26 +0200 Sebastian Dröge <[email protected]>

* gst/qtdemux/qtdemux.c:
qtdemux: Year 0 is not a valid year for GDate and the proleptic gregorian calendar

2011-04-01 13:18:55 +0200 Sebastian Dröge <[email protected]>

* ext/flac/gstflacenc.c:
flacenc: Add support for writing METADATA_BLOCK_PICTURE blocks for GST_TAG_IMAGE and GST_TAG_PREVIEW_IMAGE

2011-04-01 11:33:54 +0200 Sebastian Dröge <[email protected]>

* gst/videomixer/videomixer.c:
* gst/videomixer/videomixer2.c:
videomixer[2]: Use orc_memset() instead of memset()

2011-01-19 18:06:45 -0700 Lane Brooks <[email protected]>

* gst/videomixer/videomixer.c:
* gst/videomixer/videomixer.h:
videomixer: Add transparent background option for alpha channel formats

2011-01-19 12:07:17 -0700 Lane Brooks <[email protected]>

* gst/videomixer/blend.c:
* gst/videomixer/blend.h:
* gst/videomixer/blendorc.orc:
* gst/videomixer/videomixer2.c:
* gst/videomixer/videomixer2.h:
videomixer2: Add transparent background option for alpha channel formats
This option allows the videomixer2 element to output a valid alpha
channel when the inputs contain a valid alpha channel. This allows
mixing to occur in multiple stages serially.
The following pipeline shows an example of such a pipeline:
gst-launch videotestsrc background-color=0x000000 pattern=ball ! video/x-raw-yuv,format=$fourcc$AYUV ! videomixer2 background=transparent name=mix1 ! videomixer2 name=mix2 ! ffmpegcolorspace ! autovideosink videotestsrc ! video/x-raw-yuv,format=$fourcc$AYUV ! mix2.
The first videotestsrc in this pipeline creates a moving ball on a
transparent background. It is then passed to the first videomixer2.
Previously, this videomixer2 would have forced the alpha channel to
1.0 and given a background of checker, black, or white to the
stream. With this patch, however, you can now specify the background
as transparent, and the alpha channel of the input will be
preserved. This allows for further mixing downstream, as is shown in
the above pipeline where the a second videomixer2 is used to mix in a
background of an smpte videotestsrc. So the result is a ball hovering
over the smpte test source. This could, of course, have been
accomplished with a single mixer element, but staged mixing is useful
when it is not convenient to mix all video at once (e.g. a pipeline
where a foreground and background bin exist and are mixed at the final
output, but the foreground bin needs an internal mixer to create
transitions between clips).
Fixes bug #639994.

2011-03-31 13:25:00 +0200 Mark Nauwelaerts <[email protected]>

* ext/pulse/pulsesink.c:
pulsesink: also uncork during EOS waiting (and after EOS is rendered)
Pulsesink was recently changed to defer uncorking until there is data
to write. This condition will however never occur when EOS in being
rendered (since that marks the end of data). Changing to PAUSED state
while EOS is being waited on results in a hang: pausing corks the
stream, which will never be undone since there is no more data when
going back to PLAYING. If pulsesink is the clock provider, deadlock
ensues since time doesn't continue in corked state and the clock id
for EOS wait never fires.
Fixes #645961.

2011-03-29 16:33:43 +0200 Sebastian Dröge <[email protected]>

* tests/check/elements/rtpbin.c:
rtpbin: Don't try to request the same request pad twice

2011-03-28 23:46:47 +0100 Tim-Philipp Müller <[email protected]>

* ext/flac/gstflacdec.c:
* ext/flac/gstflacdec.h:
flacdec: fix issues with large metadata blocks when streaming unframed flac
Parse metadata blocks when handling unparsed flac in push mode. This
works around a bunch of issues with the flac decoder when handling
metadata blocks that are larger than the max. flac framesize, which
coverart blocks often are. We need to have all the data for these
blocks available when we pass data to libflac.
http://gstreamer-devel.966125.n4.nabble.com/Flac-files-that-will-playback-but-not-stream-td3338198.html#a3395276
https://bugzilla.gnome.org/show_bug.cgi?id=566769

2011-03-27 21:39:50 +0200 Jan Urbański <[email protected]>

* gst/flv/gstflvdemux.c:
* gst/flv/gstflvdemux.h:
flvdemux: Do not build an index if upstream is not seekable
An index is not useful if upstream cannot handle seeks and building it
for infinite files, for instance FLV streams, results in a memory leak.

2011-03-27 01:19:58 +0300 Alexey Chernov <[email protected]>

* docs/plugins/Makefile.am:
* docs/plugins/gst-plugins-good-plugins-docs.sgml:
* docs/plugins/gst-plugins-good-plugins-sections.txt:
* docs/plugins/inspect/plugin-video4linux2.xml:
* sys/v4l2/Makefile.am:
* sys/v4l2/gstv4l2.c:
* sys/v4l2/gstv4l2radio.c:
* sys/v4l2/gstv4l2radio.h:
v4l2: new v4l2radio element to control analog radio devices
https://bugzilla.gnome.org/show_bug.cgi?id=640118

2011-03-25 22:22:43 +0100 Sebastian Dröge <[email protected]>

* common:
Automatic update of common submodule
From 193b717 to 1ccbe09

2011-03-25 14:56:06 +0200 Stefan Kost <[email protected]>

* common:
Automatic update of common submodule
From b77e2bf to 193b717

2011-03-25 12:53:43 +0200 Stefan Kost <[email protected]>

* ext/cairo/Makefile.am:
cairo: fix the name of the *-marshall.list file to unbreak make distcheck

2011-03-25 09:31:03 +0100 Sebastian Dröge <[email protected]>

* common:
Automatic update of common submodule
From d8814b6 to b77e2bf

2011-03-25 09:06:16 +0100 Sebastian Dröge <[email protected]>

* common:
Automatic update of common submodule
From 6aaa286 to d8814b6

2011-03-25 00:10:56 +0200 Stefan Kost <[email protected]>

* gst/spectrum/gstspectrum.c:
* gst/spectrum/gstspectrum.h:
spectrum: refactor processing loop for block based operation
Previously the chain function was working sample frame based. In each cycle it
was checking if it is time to run a fft or if it is time to send a message.
Now we changed the data transform functions to work on a block of data and
calculate the max length until either {end-of-data, do-fft, do-msg}. This allows
us also to avoid the duplicated code for the single and multi-channel case (as
the transformers have the same signature now).

2011-03-24 23:47:33 +0200 Stefan Kost <[email protected]>

* configure.ac:
jack: unbreak the build for jack2 users
Jack2 (versions 1.X.X) does only have that API in svn. Limmit the use of the new
API for jack1 versions.

2011-03-24 18:49:19 +0200 Stefan Kost <[email protected]>

* common:
Automatic update of common submodule
From 6aec6b9 to 6aaa286

2011-03-24 14:14:09 +0200 Stefan Kost <[email protected]>

* gst/spectrum/gstspectrum.c:
spectrum: fix the error accumulation and frames_todo handling
Even though we wrap around the accumulated second, we still need to add the
error in the same cycle. Increase the todo in the same conditional as afterwards
the accumulated error will be below one second.

2011-03-24 13:53:12 +0200 Stefan Kost <[email protected]>

* gst/spectrum/gstspectrum.c:
spectrum: fix broken code resulting for a wrong splitup of changes

2011-03-22 16:29:53 +0200 Stefan Kost <[email protected]>

* gst/spectrum/gstspectrum.c:
* gst/spectrum/gstspectrum.h:
spectrum: simplify the have_interval calculation
Move some of the conditions to the places where the dependent variables change.

2011-03-22 16:26:45 +0200 Stefan Kost <[email protected]>

* gst/spectrum/gstspectrum.c:
spectrum: use local var for input_data function
Avoid dereferencing the input_data from the instance from within an inner loop.

2011-03-23 16:34:16 +0100 Sebastian Dröge <[email protected]>

* ext/speex/gstspeexdec.c:
* ext/speex/gstspeexdec.h:
speexdec: Get and use streamheader from the caps if possible
This allows playback of streams where the streamheader buffers
were dropped from the stream for some reason.

2011-03-22 19:36:31 +0100 Mark Nauwelaerts <[email protected]>

* gst/flv/gstflvmux.c:
flvmux: use running time for synchronization
Fixes #432612.

2011-03-22 19:36:21 +0100 Mark Nauwelaerts <[email protected]>

* gst/matroska/matroska-mux.c:
matroskamux: use running time for synchronization
Fixes #432612.

2011-03-22 19:35:58 +0100 Mark Nauwelaerts <[email protected]>

* gst/avi/gstavimux.c:
avimux: use running time for synchronization
See bug #432612.

2011-03-22 12:53:22 +0100 Luis de Bethencourt <[email protected]>

* configure.ac:
configure.ac: redundant uses of AC_MSG_RESULT()
cleaned the redundant uses of AC_MSG_RESULT() in configure.ac

2011-03-18 19:34:57 +0100 Luis de Bethencourt <[email protected]>

* autogen.sh:
autogen: wingo signed comment

2011-03-16 10:43:47 +0100 Robert Swain <[email protected]>

* ext/jack/gstjackaudiosink.c:
jackaudiosink: Fix typo from 9544622674c0d0a3147a9b51145159b02eec68e9

2011-03-16 09:38:43 +0100 Sebastian Dröge <[email protected]>

* gst/matroska/matroska-demux.c:
* gst/matroska/matroska-mux.c:
matroska: Mark tag mapping tables as static const

2011-03-16 09:37:58 +0100 Sebastian Dröge <[email protected]>

* gst/matroska/matroska-mux.c:
matroskamux: Use ARTIST instead of AUTHOR for GST_TAG_ARTIST

2011-03-16 09:35:50 +0100 Sebastian Dröge <[email protected]>

* gst/matroska/matroska-demux.c:
* gst/matroska/matroska-ids.h:
matroskademux: Use ARTIST Matroska tag instead of AUTHOR for GST_TAG_ARTIST
AUTHOR only existed in an old version of the spec and ARTIST is
the new replacement for this. We are still reading both to still
be compatible with old files.
Fixes bug #644875.

2011-03-15 20:19:48 +0000 Tim-Philipp Müller <[email protected]>

* tests/check/elements/videofilter.c:
tests: enable more formats in videofilter unit test, check more resolutions

2011-03-14 19:14:07 -0400 Youness Alaoui <[email protected]>

* gst/videofilter/gstvideoflip.c:
videoflip: Fix buffer overflow bug for odd resolutions and Y422 colorspaces
https://bugzilla.gnome.org/show_bug.cgi?id=644773

2011-03-15 19:36:01 +0200 Vincent Penquerc'h <[email protected]>

* ext/speex/gstspeexdec.c:
speexdec: silence warning message when appropriate
If we did not know how many frames to expect, then we get an unexpected
end of stream when trying to decode more frames that are there, if there
are leftover bits to pad to the next byte

2011-03-14 19:14:07 -0400 Youness Alaoui <[email protected]>

* gst/videofilter/gstvideoflip.c:
videoflip: Add support for YUY2, UVYV and YVYU colorspaces
https://bugzilla.gnome.org/show_bug.cgi?id=644773

2011-03-15 09:43:35 +0000 Tim-Philipp Müller <[email protected]>

* tests/check/elements/videofilter.c:
tests: in videofilter unit test also check with 'odd' widths and heights
And only use one test suite.

2011-03-14 19:28:07 +0100 Sebastian Dröge <[email protected]>

* ext/speex/gstspeexdec.c:
speexdec: Always process the number of frames per packet as specified in the header
Looking at the remaining bits in the bitstream after decoding a
single frame can't be used as loop condition. The remaining
bits might not give a complete frame and the speex decoder will
then output nothing but access uninitialized memory, which leads
to valgrind warnings.
Fixes bug #644669.

2011-03-14 15:46:50 +0100 Andoni Morales Alastruey <[email protected]>

* gst/matroska/matroska-mux.c:
matroskamux: return TRUE from sink pad event function for tag events, which are handled
https://bugzilla.gnome.org/show_bug.cgi?id=644730

2011-03-12 00:44:31 +0530 Philip Jägenstedt <[email protected]>

* ext/pulse/pulsesink.c:
pulsesink: Better fix for deadlock on failed connect
This reverts the previous fix that would cause a double-unlock when the
stream connect failed.
https://bugzilla.gnome.org/show_bug.cgi?id=644510

2011-03-11 23:06:31 +0530 Arun Raghavan <[email protected]>

* ext/pulse/pulsesink.c:
pulsesink: Fix deadlock if connecting to PA fails
Commit dd4ec22e introduced a deadlock in the failure path while trying
to connect to PulseAudio. This makes sure we drop the lock on the
resource mutex to avoid this.
https://bugzilla.gnome.org/show_bug.cgi?id=644510

2011-03-11 16:59:10 +0200 Stefan Kost <[email protected]>

* tests/check/Makefile.am:
tests: order state-test blacklist and add jack elements
Jack audio src/sink elements recently got moved from bad and should be excluded
from the test (like the other device specific source and sinks).
Fixes #644288

2011-03-11 13:47:26 +0100 Sebastian Dröge <[email protected]>

* ext/dv/gstdvdemux.c:
dvdemux: Chain up to the parent class' ::send_event for non-seek events

2011-03-11 13:46:05 +0100 Sebastian Dröge <[email protected]>

* ext/dv/gstdvdemux.c:
dvdemux: Fix refcount issues with the seek event
Fixes bug #642963.

2011-03-11 09:54:02 +0000 Tim-Philipp Müller <[email protected]>

* ext/pulse/pulsesink.c:
docs: fix pulsesink gtk-doc markup

2011-03-11 10:29:08 +0100 Philippe Normand <[email protected]>

* configure.ac:
* ext/jack/gstjackaudiosink.c:
* ext/jack/gstjackaudiosrc.c:
jack: fix build against jack 0.120.2
jack_port_get_total_latency() has been deprecated in favor of
jack_port_get_latency_range().
https://bugzilla.gnome.org/show_bug.cgi?id=644477

2011-03-10 14:29:25 +0200 Stefan Kost <[email protected]>

* gst/spectrum/gstspectrum.c:
spectrum: more comments and tune and logging

2011-03-10 14:15:42 +0200 Stefan Kost <[email protected]>

* gst/spectrum/gstspectrum.c:
spectrum: avoid unneccesary extra fft runs
Before it was possible that we run an extra fft when the time for sending a new
message is due. Only do this if we have not run the fft for the interval at all.

2011-03-10 14:12:01 +0200 Stefan Kost <[email protected]>

* gst/spectrum/gstspectrum.c:
spectrum: only scale the vectors that we are processing
Phase is not produced by default, so lets not scale it unconditionally to save a
few cycles.

2011-03-10 14:10:25 +0200 Stefan Kost <[email protected]>

* gst/spectrum/gstspectrum.c:
* gst/spectrum/gstspectrum.h:
spectrum: put number of channels to instance variable
When freeing data the format might have changed. Thus we need to remember for
which format we allocated memory.

2011-03-10 10:27:14 +0200 Stefan Kost <[email protected]>

* gst/spectrum/gstspectrum.c:
spectrum: update doc review stamp

2011-03-10 10:22:29 +0200 Stefan Kost <[email protected]>

* gst/spectrum/gstspectrum.c:
* gst/spectrum/gstspectrum.h:
spectrum: use function pointers for data readers
Don't check the format for each sample frame to read. We can make that decission
in _setup already. This is still not ideal as we call the function per frame.
Ideally we determine how many samples we can copy and have a loop in the input
reader. As an alternative we might also consider to use the fft variants for the
various formats and not convert to float for all cases - we would still need to
mix or deinterleave though.

2011-03-09 17:07:47 +0100 Mark Nauwelaerts <[email protected]>

* gst/rtsp/gstrtspsrc.c:
* gst/rtsp/gstrtspsrc.h:
rtspsrc: improve recovery from failed seek
In case server-side fails to perform seek, i.e. PLAY at non-zero requested
position, recovery so far would arrange for streaming to continue, albeit
having lost position tracking in the process. So, query position prior
to seek and use upon failed seek.

2011-03-09 16:51:00 +0100 Mark Nauwelaerts <[email protected]>

* gst/rtpmanager/gstrtpjitterbuffer.c:
jitterbuffer: handle position query

2011-03-09 16:57:28 +0200 Stefan Kost <[email protected]>

* gst/spectrum/gstspectrum.c:
* gst/spectrum/gstspectrum.h:
spectrum: multi-channel support
Add a boolean multi-channel property with a default of FALSE. When set to TRUE
the element won't mix all input channels to mono, but instead run a FFT on each
channel. In that case the result message would contain a 2 dimensional array
of channel x data for magnitude and phase.
API: GstSpectrum:multi-channel
https://bugzilla.gnome.org/show_bug.cgi?id=593482

2011-03-09 16:55:56 +0200 Stefan Kost <[email protected]>

* gst/spectrum/gstspectrum.c:
spectrum: more xrefs in the docs

2011-03-09 12:41:15 +0200 Stefan Kost <[email protected]>

* gst/spectrum/gstspectrum.c:
spectrum: factor out the code that accumulated samples into the ring-buffer
Use a separate function to read a sample frame into a ringbuffer slot. In the
future we can use format-specific function pointer to avoid the reoccuring
format checks.

2011-03-09 12:38:52 +0200 Stefan Kost <[email protected]>

* gst/spectrum/gstspectrum.c:
spectrum: pull format to temp var to improve readability of lines using it

2011-03-09 12:20:11 +0200 Stefan Kost <[email protected]>

* gst/spectrum/gstspectrum.c:
spectrum: code cleanup for copying data to ring-buffer
Rename fp to is_float and restructure if-else part for handling the different formats.

2011-03-09 11:40:48 +0200 Stefan Kost <[email protected]>

* gst/spectrum/gstspectrum.c:
* gst/spectrum/gstspectrum.h:
spectrum: add a GstSpecrtumChannel context structure
We now keep the fft data that is related to one channel in a separate structure
to prepare for multichannel support. We also refactor the code to operate more
often on the channel context.

2011-03-09 11:18:19 +0200 Stefan Kost <[email protected]>

* gst/spectrum/gstspectrum.c:
spectrum: call the instance var spectrum instead of filter

2011-03-09 11:14:37 +0200 Stefan Kost <[email protected]>

* gst/spectrum/gstspectrum.c:
spectrum: don't value we already took from the gvalue

2011-03-08 16:28:27 +0000 Tim-Philipp Müller <[email protected]>

Merge ad-hoc release branch '0.10.28'