2012-09-14 01:46:14 +0100 Tim-Philipp Müller <tim@centricular.net>

2012-09-14 01:46:14 +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/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-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-cutter.xml:
* docs/plugins/inspect/plugin-debug.xml:
* docs/plugins/inspect/plugin-deinterlace.xml:
* docs/plugins/inspect/plugin-dv.xml:
* docs/plugins/inspect/plugin-effectv.xml:
* docs/plugins/inspect/plugin-equalizer.xml:
* docs/plugins/inspect/plugin-flac.xml:
* docs/plugins/inspect/plugin-flv.xml:
* docs/plugins/inspect/plugin-flxdec.xml:
* docs/plugins/inspect/plugin-gdkpixbuf.xml:
* docs/plugins/inspect/plugin-goom.xml:
* docs/plugins/inspect/plugin-goom2k1.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-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-rtpmanager.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:
docs: update docs

2012-09-14 00:47:38 +0100 Tim-Philipp Müller <[email protected]>

* tests/check/elements/wavpackenc.c:
tests: push stream-start and segment events in wavpackenc test

2012-09-13 10:56:27 +0200 Wim Taymans <[email protected]>

* sys/v4l2/gstv4l2object.h:
* sys/v4l2/gstv4l2src.c:
* sys/v4l2/gstv4l2src.h:
v4l2: remove unused properties

2012-09-13 10:15:54 +0200 Wim Taymans <[email protected]>

* ext/pulse/pulsesrc.c:
pulsesrc: disable reconfigure
See https://bugzilla.gnome.org/show_bug.cgi?id=683902

2012-09-10 22:09:59 -0700 Jan Schmidt <[email protected]>

* gst/deinterlace/gstdeinterlace.c:
deinterlace: Don't treat every custom-downstream event as EOS
Don't fall through to the EOS handling after receiving a
custom-downstream event.

2012-09-12 21:05:44 +0200 Stefan Sauer <[email protected]>

* ext/cairo/gsttextoverlay.c:
* gst/avi/gstavimux.c:
* gst/flv/gstflvmux.c:
* gst/interleave/interleave.c:
* gst/isomp4/gstqtmux.c:
* gst/matroska/matroska-mux.c:
* gst/multipart/multipartmux.c:
* gst/smpte/gstsmpte.c:
* gst/videomixer/videomixer2.c:
collectpads: remove gst_collect_pads_add_pad_full
Rename gst_collect_pads_add_pad_full() to gst_collect_pads_add_pad() and fix all
invocations.

2012-09-12 17:14:46 +0200 Mark Nauwelaerts <[email protected]>

* gst/udp/gstmultiudpsink.c:
udp: add include for IPPROTO_*

2012-09-12 16:39:08 +0200 Mark Nauwelaerts <[email protected]>

* gst/udp/gstmultiudpsink.c:
udp: properly match braces and cpp directives
Fixes compilation where IPV6_TCLASS not defined.

2012-09-12 14:42:07 +0200 Edward Hervey <[email protected]>

* gst/shapewipe/gstshapewipe.c:
shapewipe: Use default query handler where needed
And clean up get_caps code while I'm at it

2012-09-12 13:28:07 +0200 Wim Taymans <[email protected]>

* gst/deinterlace/gstdeinterlace.c:
deinterlace: improve framerate transform
Handle G_MAXINT in the framerates better. If we cannot double or divide the
framerate, clamp to the smallest/largest possible value we can express instead
of failing.
Fixes https://bugzilla.gnome.org/show_bug.cgi?id=683861

2012-09-12 13:17:54 +0200 Wim Taymans <[email protected]>

* gst/deinterlace/gstdeinterlace.c:
deinterlace: small cleanup

2012-09-07 17:20:57 -0400 Youness Alaoui <[email protected]>

* gst/videomixer/blend.c:
* gst/videomixer/blend.h:
* gst/videomixer/videomixer2.c:
videomixer2: Adding nv12 and nv21 support
Fixes https://bugzilla.gnome.org/show_bug.cgi?id=683841

2012-09-12 10:18:53 +0200 Michael Smith <[email protected]>

* gst/isomp4/qtdemux.c:
* gst/isomp4/qtdemux_fourcc.h:
qtdemux: add support for prores
Fixes https://bugzilla.gnome.org/show_bug.cgi?id=683839

2012-09-12 00:16:31 +0100 Tim-Philipp Müller <[email protected]>

* tests/check/elements/rganalysis.c:
tests: fix most of the rganalysis unit tests
Before the element would post messages on the bus itself, now
the sinks do that based on the tag events they receive. But
since we don't have proper sink elements in these unit tests,
but just dangling pads, we have to post the tag messages the
test checks for ourselves.
Down from 52/55 failing to 7/52 failing.

2012-09-11 17:36:51 +0200 Mark Nauwelaerts <[email protected]>

* ext/dv/gstdvdemux.c:
* gst/avi/gstavidemux.c:
* gst/debugutils/rndbuffersize.c:
* gst/flv/gstflvdemux.c:
* gst/isomp4/qtdemux.c:
* gst/matroska/matroska-demux.c:
* gst/wavparse/gstwavparse.c:
ext, gst: only activate in pull mode if upstream is seekable

2012-09-11 15:38:23 +0200 Wim Taymans <[email protected]>

* sys/v4l2/gstv4l2src.c:
v4l2: disable renegotiation
We can't yet wait for the bufferpool to DRAIN before starting renegotiation so
disable it for now.
Fixes https://bugzilla.gnome.org/show_bug.cgi?id=682770

2012-09-11 12:48:39 +0200 Mark Nauwelaerts <[email protected]>

* tests/check/elements/rtpbin.c:
tests: rtpbin: port to the new GLib thread API

2012-09-11 12:36:56 +0200 Mark Nauwelaerts <[email protected]>

* sys/directsound/gstdirectsoundsink.c:
* sys/directsound/gstdirectsoundsink.h:
directsoundsink: port to the new GLib thread API

2012-09-11 11:59:54 +0200 Wim Taymans <[email protected]>

* gst/isomp4/qtdemux.c:
qtdemux: don't reset segment
Don't reset the segment because we need the values for accumulation. the segment
is reset at start and after a flushing seek. Fixes some problems with files with
quicktime segments.

2012-09-10 17:14:37 +0200 Wim Taymans <[email protected]>

* tests/check/elements/id3demux.c:
tests: fix id3demux test

2012-09-10 14:31:02 +0200 Mark Nauwelaerts <[email protected]>

* gst/flv/amfdefs.h:
* gst/flv/gstflvdemux.c:
* gst/rtp/gstrtpqdmdepay.c:
* gst/rtp/gstrtpsv3vdepay.c:
gst: adjust comment style

2012-09-10 14:30:42 +0200 Mark Nauwelaerts <[email protected]>

* gst/avi/gstavidemux.c:
avidemux: remove defunct commented code

2012-09-10 13:35:15 +0200 Wim Taymans <[email protected]>

* ext/pulse/pulsesrc.c:
pulsesrc: consider stream alive when not connected yet
When we start and renegotiate, there is a moment where the stream is created but
not yet connected. Make sure all functions deal with this situation correctly
instead of erroring out.
Fixes https://bugzilla.gnome.org/show_bug.cgi?id=681247

2012-09-10 12:15:25 +0200 Wim Taymans <[email protected]>

* ext/pulse/pulsesrc.c:
pulsesrc: don't fail when not negotiated yet
When get_time is called but we are not yet negotiated, return 0 instead of
posting an error. It's possible that the base class is still negotiating when
our get_time is called.

2012-09-10 11:32:25 +0200 Wim Taymans <[email protected]>

* ext/pulse/pulsesrc.c:
* sys/oss/gstosssrc.c:
* sys/oss4/oss4-source.c:
update for audio base src api change

2012-09-10 00:42:52 +0100 Tim-Philipp Müller <[email protected]>

* gst/avi/gstavimux.c:
* gst/isomp4/qtdemux.c:
video/x-3ivx and video/x-xvid -> video/mpeg,mpegversion=4
If it ever turns out that we really must use thoe specific
fourccs and not the generic one, we can still add a flavor
field to the caps later.

2012-09-07 16:15:42 +0200 Daniela <[email protected]>

* gst/rtsp/gstrtspsrc.c:
rtspsrc: avoid leak
When setup fails, make sure to cleanup afterwards.
Fixes https://bugzilla.gnome.org/show_bug.cgi?id=673509

2012-09-07 15:23:44 +0200 Mark Nauwelaerts <[email protected]>

* gst/rtp/gstrtpamrdepay.c:
rtpamrdepay: unmap rtp buffer
... thereby plugging a memleak.

2012-09-07 14:13:17 +0200 Mark Nauwelaerts <[email protected]>

* tests/check/elements/rtp-payloading.c:
tests: rtp-payloading: adjust to modified bufferlist semantics
... now implemented by buffer memory blocks.

2012-09-07 14:11:39 +0200 Mark Nauwelaerts <[email protected]>

* gst/rtp/gstrtph264pay.c:
rtph264pay: avoid crashing on NULL access in debug message

2012-09-07 14:11:02 +0200 Mark Nauwelaerts <[email protected]>

* gst/rtp/gstrtph263ppay.c:
rtph263ppay: plug caps leak

2012-09-06 17:09:20 +0200 Wim Taymans <[email protected]>

* gst/deinterlace/gstdeinterlace.c:
deinterlace: remove redundant _set_allocation call

2012-09-06 17:05:00 +0200 Mark Nauwelaerts <[email protected]>

* tests/check/elements/deinterlace.c:
tests: deinterlace: do not leak deinterlace pads

2012-09-06 17:04:39 +0200 Mark Nauwelaerts <[email protected]>

* gst/deinterlace/gstdeinterlace.c:
deinterlace: plug some leaks

2012-09-06 16:49:02 +0200 Wim Taymans <[email protected]>

* gst/deinterlace/gstdeinterlace.c:
deinterlace: reuse core function for GCD

2012-09-06 16:31:00 +0200 Mark Nauwelaerts <[email protected]>

* gst/deinterlace/gstdeinterlace.c:
deinterlace: support filter in getcaps

2012-09-06 16:30:44 +0200 Mark Nauwelaerts <[email protected]>

* gst/deinterlace/gstdeinterlace.c:
deinterlace: do not leak getcaps result

2012-09-06 16:23:28 +0200 Wim Taymans <[email protected]>

* gst/deinterlace/gstdeinterlace.c:
* gst/deinterlace/gstdeinterlace.h:
deinterlace: add support for bufferpool
Add bufferpool support to avoid a memcpy in the videosink when actively
interlacing.
Remove some commented obsolete code.

2012-09-06 13:38:52 +0200 Wim Taymans <[email protected]>

* gst/deinterlace/gstdeinterlace.c:
deinterlace: proxy allocation query in passthrough
We can let the allocation query pass when we are operating in passthrough mode.

2012-09-06 13:23:46 +0200 Wim Taymans <[email protected]>

* gst/deinterlace/gstdeinterlace.c:
deinterlace: use default event functions
instead of blindly forwarding unknown events.

2012-09-06 13:23:30 +0200 Wim Taymans <[email protected]>

* gst/deinterlace/gstdeinterlace.c:
deinterlace: small cleanups

2012-09-06 12:56:30 +0200 Wim Taymans <[email protected]>

* gst/deinterlace/gstdeinterlace.c:
deinterlace: call default query handlers
Call the default query handler instead of forwarding the query blindly. Fixes
issues of strides because of proxying the allocation query wrongly.

2012-09-06 10:42:21 +0200 Wim Taymans <[email protected]>

* sys/v4l2/gstv4l2object.c:
v4l2: remove unused code.

2012-09-06 10:42:06 +0200 Wim Taymans <[email protected]>

* ext/pulse/pulsesink.c:
pulse: improve debug

2012-09-05 11:50:05 +0200 Mark Nauwelaerts <[email protected]>

* ext/dv/gstdvdemux.c:
dvdemux: remove obsolete update newsegment handling code

2012-09-04 12:35:53 +0200 Wim Taymans <[email protected]>

* gst/videofilter/gstvideobalance.c:
videobalance: avoid deadlock
_update_properties takes the object lock and should not be called when the
object lock is already taken.

2012-09-03 12:46:03 +0100 Tim-Philipp Müller <[email protected]>

* gst/matroska/matroska-mux.c:
matroskamux: extract interlaced-ness of video track from interlace-mode field
instead of the old boolean "interlaced" field.

2012-09-03 02:51:24 +0100 Tim-Philipp Müller <[email protected]>

* gst/avi/gstavimux.c:
* gst/matroska/matroska-demux.c:
* gst/matroska/matroska-mux.c:
* gst/rtp/gstrtpmp4vpay.c:
* tests/check/elements/avimux.c:
video/x-xvid -> video/mpeg,mpegversion=4

2012-09-02 02:50:50 +0100 Tim-Philipp Müller <[email protected]>

* gst/isomp4/qtdemux.c:
* gst/matroska/matroska-demux.c:
* gst/matroska/matroska-mux.c:
text/plain + text/x-pango-markup -> text/x-raw

2012-09-02 01:31:53 +0100 Tim-Philipp Müller <[email protected]>

* ext/soup/gstsouphttpsrc.c:
* gst/matroska/matroska-demux.c:
gst_message_new_duration -> gst_message_new_duration_changed

2012-08-30 22:07:24 +0200 Wim Taymans <[email protected]>

* gst/rtpmanager/rtpsession.c:
session: also stop probatation on existing sources
Receiving an RTCP packet should also stop probation on sources we have seen
before.
Fixes https://bugzilla.gnome.org/show_bug.cgi?id=683065

2012-08-22 16:36:21 -0700 Aleix Conchillo Flaque <[email protected]>

* gst/rtpmanager/gstrtpsession.c:
* gst/rtpmanager/rtpsession.c:
* gst/rtpmanager/rtpsession.h:
* gst/rtpmanager/rtpsource.c:
* gst/rtpmanager/rtpsource.h:
* gst/rtsp/gstrtspsrc.c:
* gst/rtsp/gstrtspsrc.h:
rtp: make rtp packet probation configurable (bug #682512)

2012-08-30 12:21:01 +0200 Mark Nauwelaerts <[email protected]>

* ext/gdk_pixbuf/gstgdkpixbufoverlay.c:
gdkpixbuf: adjust to modified video overlay composition API

2012-08-30 11:30:01 +0200 Mark Nauwelaerts <[email protected]>

* gst/audioparsers/gstflacparse.c:
flacparse: fixup 0.11 port of suspect frame checking
Fixes https://bugzilla.gnome.org/show_bug.cgi?id=682959

2012-08-28 18:56:19 +0200 Mark Nauwelaerts <[email protected]>

* gst/avi/gstavidemux.c:
avidemux: avoid invalid H264 bytestream codec_data
Fixes https://bugzilla.gnome.org/show_bug.cgi?id=681369

2012-08-28 19:00:44 +0200 Mark Nauwelaerts <[email protected]>

* gst/isomp4/qtdemux.c:
qtdemux: port segment event creation to 0.11

2012-08-28 16:28:13 +0200 Mark Nauwelaerts <[email protected]>

* gst/isomp4/qtdemux.c:
qtdemux: release extra event ref when replacing pending newsegment event

2012-07-03 17:50:24 +0200 David Corvoysier <[email protected]>

* gst/isomp4/qtdemux.c:
* gst/isomp4/qtdemux_dump.c:
* gst/isomp4/qtdemux_dump.h:
* gst/isomp4/qtdemux_fourcc.h:
* gst/isomp4/qtdemux_types.c:
isomp4: add DASH tfdt box support
MPEG DASH has defined a set of new boxes to specify duration, indexes and
offsets of ISOBMFF fragments.
The Track Fragment Base Media Decode Time (tfdt) Box can in particular be
included inside a traf box to specify the absolute decode time, measured on the
media timeline, of the first sample in decode order in the track fragment.
This information can be used by the isomp4 demux to find out the current position of
an MP4 fragment in the timeline.
This patch adds code to isomp4 to:
- parse the tfdt box
- adjust the time/position member of the new segment sent when playback starts
Fixes https://bugzilla.gnome.org/show_bug.cgi?id=677535

2012-08-26 22:39:55 +0100 Tim-Philipp Müller <[email protected]>

* ext/aalib/gstaasink.c:
* ext/cairo/gstcairorender.c:
* ext/cairo/gsttextoverlay.c:
* ext/cairo/gsttimeoverlay.c:
* ext/dv/gstdvdec.c:
* ext/dv/gstdvdemux.c:
* ext/flac/gstflacenc.c:
* ext/flac/gstflactag.c:
* ext/gdk_pixbuf/gstgdkpixbufoverlay.c:
* ext/gdk_pixbuf/gstgdkpixbufsink.c:
* ext/gdk_pixbuf/pixbufscale.c:
* ext/jack/gstjackaudiosink.c:
* ext/jack/gstjackaudiosrc.c:
* ext/jpeg/gstjpegdec.c:
* ext/jpeg/gstjpegenc.c:
* ext/libcaca/gstcacasink.c:
* ext/libpng/gstpngdec.c:
* ext/pulse/pulsesink.c:
* ext/pulse/pulsesrc.c:
* ext/raw1394/gstdv1394src.c:
* ext/raw1394/gsthdv1394src.c:
* ext/soup/gstsouphttpclientsink.c:
* ext/soup/gstsouphttpsrc.c:
* ext/speex/gstspeexdec.c:
* ext/speex/gstspeexenc.c:
* ext/taglib/gstapev2mux.cc:
* ext/taglib/gstid3v2mux.cc:
* ext/wavpack/gstwavpackdec.c:
* ext/wavpack/gstwavpackenc.c:
* gst/alpha/gstalpha.c:
* gst/alpha/gstalphacolor.c:
* gst/apetag/gstapedemux.c:
* gst/audiofx/audioamplify.c:
* gst/audiofx/audiochebband.c:
* gst/audiofx/audiocheblimit.c:
* gst/audiofx/audiodynamic.c:
* gst/audiofx/audioecho.c:
* gst/audiofx/audioinvert.c:
* gst/audiofx/audiokaraoke.c:
* gst/audiofx/audiopanorama.c:
* gst/audiofx/audiowsincband.c:
* gst/audiofx/audiowsinclimit.c:
* gst/audioparsers/gstaacparse.c:
* gst/audioparsers/gstac3parse.c:
* gst/audioparsers/gstamrparse.c:
* gst/audioparsers/gstdcaparse.c:
* gst/audioparsers/gstflacparse.c:
* gst/audioparsers/gstmpegaudioparse.c:
* gst/audioparsers/gstwavpackparse.c:
* gst/autodetect/gstautoaudiosink.c:
* gst/autodetect/gstautoaudiosrc.c:
* gst/autodetect/gstautovideosink.c:
* gst/autodetect/gstautovideosrc.c:
* gst/avi/gstavidemux.c:
* gst/avi/gstavimux.c:
* gst/avi/gstavisubtitle.c:
* gst/cutter/gstcutter.c:
* gst/debugutils/gstpushfilesrc.c:
* gst/debugutils/gsttaginject.c:
* gst/debugutils/progressreport.c:
* gst/deinterlace/gstdeinterlace.c:
* gst/effectv/gstaging.c:
* gst/effectv/gstdice.c:
* gst/effectv/gstedge.c:
* gst/effectv/gstop.c:
* gst/effectv/gstquark.c:
* gst/effectv/gstradioac.c:
* gst/effectv/gstrev.c:
* gst/effectv/gstripple.c:
* gst/effectv/gstshagadelic.c:
* gst/effectv/gststreak.c:
* gst/effectv/gstvertigo.c:
* gst/effectv/gstwarp.c:
* gst/equalizer/gstiirequalizer10bands.c:
* gst/equalizer/gstiirequalizer3bands.c:
* gst/equalizer/gstiirequalizernbands.c:
* gst/flv/gstflvdemux.c:
* gst/flv/gstflvmux.c:
* gst/goom/gstgoom.c:
* gst/goom2k1/gstgoom.c:
* gst/icydemux/gsticydemux.c:
* gst/id3demux/gstid3demux.c:
* gst/imagefreeze/gstimagefreeze.c:
* gst/interleave/deinterleave.c:
* gst/interleave/interleave.c:
* gst/isomp4/atomsrecovery.c:
* gst/isomp4/gstqtmux-doc.c:
* gst/isomp4/gstqtmux.c:
* gst/isomp4/qtdemux.c:
* gst/matroska/matroska-demux.c:
* gst/matroska/matroska-mux.c:
* gst/matroska/matroska-parse.c:
* gst/matroska/webm-mux.c:
* gst/monoscope/gstmonoscope.c:
* gst/multifile/gstmultifilesink.c:
* gst/multifile/gstmultifilesrc.c:
* gst/multifile/gstsplitfilesrc.c:
* gst/multipart/multipartdemux.c:
* gst/multipart/multipartmux.c:
* gst/replaygain/gstrganalysis.c:
* gst/replaygain/gstrglimiter.c:
* gst/replaygain/gstrgvolume.c:
* gst/rtp/README:
* gst/rtpmanager/gstrtpbin.c:
* gst/rtpmanager/gstrtpjitterbuffer.c:
* gst/rtpmanager/gstrtpptdemux.c:
* gst/rtpmanager/gstrtpsession.c:
* gst/rtpmanager/gstrtpssrcdemux.c:
* gst/rtsp/gstrtspsrc.c:
* gst/shapewipe/gstshapewipe.c:
* gst/smpte/gstsmpte.c:
* gst/smpte/gstsmptealpha.c:
* gst/udp/gstudpsink.c:
* gst/udp/gstudpsrc.c:
* gst/videobox/gstvideobox.c:
* gst/videocrop/gstaspectratiocrop.c:
* gst/videocrop/gstvideocrop.c:
* gst/videofilter/gstgamma.c:
* gst/videofilter/gstvideobalance.c:
* gst/videofilter/gstvideoflip.c:
* gst/wavparse/gstwavparse.c:
* sys/directsound/gstdirectsoundsink.c:
* sys/oss/gstosssink.c:
* sys/oss/gstosssrc.c:
* sys/oss4/oss4-sink.c:
* sys/oss4/oss4-source.c:
* sys/osxaudio/gstosxaudiosink.c:
* sys/osxaudio/gstosxaudiosrc.c:
* sys/sunaudio/gstsunaudiosink.c:
* sys/sunaudio/gstsunaudiosrc.c:
* sys/v4l2/gstv4l2radio.c:
* sys/v4l2/gstv4l2sink.c:
* sys/v4l2/gstv4l2src.c:
* sys/waveform/gstwaveformsink.c:
* sys/ximage/gstximagesrc.c:
* tests/examples/cairo/cairo_overlay.c:
* tests/examples/rtp/client-H263p-AMR.sh:
* tests/examples/rtp/client-H263p-PCMA.sh:
* tests/examples/rtp/client-H263p.sh:
* tests/examples/rtp/client-H264-PCMA.sh:
* tests/examples/rtp/client-H264.sh:
* tests/examples/rtp/client-PCMA.sh:
* tests/examples/rtp/server-VTS-H263p-ATS-PCMA.sh:
* tests/examples/rtp/server-VTS-H263p.sh:
* tests/examples/rtp/server-alsasrc-PCMA.sh:
* tests/examples/rtp/server-decodebin-H263p-AMR.sh:
* tests/examples/rtp/server-v4l2-H263p-alsasrc-AMR.sh:
* tests/examples/rtp/server-v4l2-H264-alsasrc-PCMA.sh:
* tests/examples/shapewipe/shapewipe-example.c:
* tests/icles/gdkpixbufsink-test.c:
* tests/icles/videocrop-test.c:
docs: gst-launch -> gst-launch-1.0 and ffmpegcolorspace -> videoconvert

2012-08-26 22:32:54 +0100 Tim-Philipp Müller <[email protected]>

* ext/flac/gstflacdec.c:
* gst/videomixer/videomixer2.c:
docs: gst-launch-0.11 -> gst-launch-1.0

2012-08-26 22:08:54 +0100 Tim-Philipp Müller <[email protected]>

* gst/deinterlace/gstdeinterlace.c:
* tests/check/elements/deinterlace.c:
deinterlace: the field in caps is "interlace-mode" not "interlace-method"
Fix deinterlace unit test. Need to set right field on output caps.
Also remove right field (not old 0.10 "interlaced" boolean field)
from caps in unit test before comparing old and new.

2012-08-26 21:45:44 +0100 Tim-Philipp Müller <[email protected]>

* tests/check/elements/icydemux.c:
tests: fix icydemux unit test
Was waiting for a tag message on the bus, which would never
come, because elements don't post those themselves any more
but let sinks post them from tag events. Only that there are
no sinks in this unit test.

2012-08-26 21:27:00 +0100 Tim-Philipp Müller <[email protected]>

* tests/check/elements/videocrop.c:
tests: fix videocrop crop_to_1x1 unit test for GRAY8 format
Update table with pixel values with the value actually produced
by videotestsrc.

2012-08-27 09:00:45 +0200 Sjoerd Simons <[email protected]>

* ext/pulse/pulsesrc.c:
pulsesrc: Only print caps if they're provided

2012-08-24 19:43:08 +0100 Michael Rubinstein <[email protected]>

* gst/videomixer/blend.c:
videomixer: fix endianness check on systems where non-glib endianness defines are not set
On Windows LITTLE_ENDIAN without the G_ in was not defined, so the
test comes out wrong.

2012-08-22 17:23:25 +0200 Wim Taymans <[email protected]>

* gst/udp/gstmultiudpsink.c:
udpsink: don't crash on NULL error
Check if there is an error before retrieving its message.
See https://bugzilla.gnome.org/show_bug.cgi?id=682481

2012-08-22 13:30:19 +0200 Stefan Sauer <[email protected]>

* common:
Automatic update of common submodule
From 668acee to 4f962f7

2012-08-22 13:18:00 +0200 Stefan Sauer <[email protected]>

* configure.ac:
configure: bump gtk-doc req to 1.12 (mar-2009)
This allows us to e.g. unconditionally use gtkdoc-rebase.

2012-08-22 11:21:38 +0200 Martin Ertsaas <[email protected]>

* sys/osxvideo/osxvideosink.h:
* sys/osxvideo/osxvideosink.m:
osxvideosink: Make osxvideosink use the non-deprecated threading api from glib.
https://bugzilla.gnome.org/show_bug.cgi?id=682446

2012-08-14 15:40:31 +0530 Arun Raghavan <[email protected]>

* ext/pulse/pulsesrc.c:
pulsesrc: Handle negotiation events
This makes sure that we:
a) Destroy an existing stream if a negotiate() request comes in: this is
required when receiving a downstream renegotiation request after a
stream has been created.
b) Create a new stream on prepare(): this is required since we do a
setcaps() in negotiate(), which causes the stream to be dropped by a
ringbuffer release() call (this does not happen during first negotiation
since the release is only done on a running ringbuffer). The subsequent
call to ringbuffer acquire() fails because the stream was lost on
release().
https://bugzilla.gnome.org/show_bug.cgi?id=681247

2012-08-14 15:38:27 +0530 Arun Raghavan <[email protected]>

* ext/pulse/pulseutil.c:
pulse: Clear unpositioned flag when setting positions
If converting a PA channel map to gst channel positions results in a
valid set of channel positions, we clear the unpositioned flag from the
ringbuffer spec.

2012-08-14 09:37:45 +0530 Arun Raghavan <[email protected]>

* ext/pulse/pulsesrc.c:
pulsesrc: Remove redundant channel-mask setting for stereo case
The gstaudio helper libraries already take care of this case for us.

2012-08-14 09:36:30 +0530 Arun Raghavan <[email protected]>

* ext/pulse/pulsesrc.c:
pulsesrc: Don't use memset to set invalid channel positions
This itereates over the GstAudioInfo to set invalid channel positions
rather than use memset() which works right now because it assumes that
GST_AUDIO_CHANNEL_POSITION_INVALID is -1.

2012-08-22 10:30:04 +0100 Tim-Philipp Müller <[email protected]>

* ext/gdk_pixbuf/gstgdkpixbufsink.c:
gdkpixbufsink: minor docs improvement

2012-08-22 10:23:24 +0100 Tim-Philipp Müller <[email protected]>

* ext/gdk_pixbuf/Makefile.am:
* ext/gdk_pixbuf/gstgdkpixbufplugin.c:
gdkpixbuf: re-enable already-ported gdkpixbufsink

2012-08-22 10:08:08 +0100 Tim-Philipp Müller <[email protected]>

* ext/gdk_pixbuf/Makefile.am:
* ext/gdk_pixbuf/gstgdkpixbufoverlay.c:
* ext/gdk_pixbuf/gstgdkpixbufoverlay.h:
* ext/gdk_pixbuf/gstgdkpixbufplugin.c:
gdkpixbuf: port gdkpixbufoverlay element to 0.11

2012-08-22 00:00:46 +0100 Tim-Philipp Müller <[email protected]>

* configure.ac:
* ext/gdk_pixbuf/Makefile.am:
* ext/gdk_pixbuf/gstgdkpixbuf.c:
* ext/gdk_pixbuf/gstgdkpixbuf.h:
* ext/gdk_pixbuf/gstgdkpixbufdec.c:
* ext/gdk_pixbuf/gstgdkpixbufdec.h:
* ext/gdk_pixbuf/gstgdkpixbufplugin.c:
gdkpixbuf: re-enable already-ported gdkpixbuf element as gdkpixbufdec
Not sure why it as disabled exactly given that it had already
been ported (though without metas or baseclass).
Move plugin_init bits into separate source file, and rename
decoder element to gdkpixbufdec.

2012-08-21 23:25:47 +0100 Tim-Philipp Müller <[email protected]>

* ext/gdk_pixbuf/gst_loader.c:
gdkpixbuf: remove old and unused gst_loader source file
Once upon a time used to load GStreamer vids via GdkPixbuf API.

2012-08-16 16:51:16 -0700 Aleix Conchillo Flaque <[email protected]>

* gst/rtpmanager/gstrtpbin.c:
* gst/rtpmanager/gstrtpbin.h:
* gst/rtsp/gstrtspsrc.c:
* gst/rtsp/gstrtspsrc.h:
rtspsrc: make jitterbuffer drop-on-latency available (fix #682055)
Conflicts:
gst/rtsp/gstrtspsrc.h

2012-08-21 19:47:45 +0800 Huacai Chen <[email protected]>

* sys/v4l2/v4l2_calls.c:
v4l2: make gst_v4l2_fill_lists() adapt to kernel 3.3+
When do v4l2_ioctl() with VIDIOC_ENUMINPUT fails on some devices,
kernels before 3.3.0 return EINVAL, but newer kernels return ENOTTY.
This patch make those devices work well on kernel 3.3+.
Related kernel commit:
http://git.kernel.org/?p=linux/kernel/git/torvalds/linux.git;a=commit;h=07d106d0a33d6063d2061305903deb02489eba20
Signed-off-by: Huacai Chen <[email protected]>
Signed-off-by: Rui Wang <[email protected]>
Signed-off-by: Jie Chen <[email protected]>

2012-08-20 23:30:38 +0100 Tim-Philipp Müller <[email protected]>

* docs/plugins/inspect/plugin-matroska.xml:
* gst/isomp4/qtdemux.c:
* gst/matroska/matroska-demux.c:
* gst/matroska/matroska-mux.c:
video/x-dvd-subpicture -> subpicture/x-dvd

2012-08-17 20:52:42 +0100 Tim-Philipp Müller <[email protected]>

* gst/multifile/gstmultifilesrc.c:
multifilesrc: fix example pipeline in docs

2012-08-17 14:59:57 +0200 Stefan Sauer <[email protected]>

* gst/equalizer/gstiirequalizer.c:
* gst/equalizer/gstiirequalizer10bands.c:
* gst/equalizer/gstiirequalizer3bands.c:
* tests/check/elements/equalizer.c:
equalizer: enable presets for the n-band equalizer
Add a test for saving and restoring the preset.

2012-08-14 01:20:19 +0100 Tim-Philipp Müller <[email protected]>

* gst/deinterlace/gstdeinterlace.c:
deinterlace: fix not-negotiated errors on variable or missing framerate in input caps
Remove some bogus code I added during porting that would error out
on missing or variable framerates in input caps. Handle this like
we do in 0.10
Fixes test_mode_disabled_passthrough unit test check.

2012-08-12 13:16:32 +0200 Sjoerd Simons <[email protected]>

* gst/law/alaw-decode.c:
* gst/law/mulaw-decode.c:
law: Filter layout caps field
The layout caps field shouldn't be passed through to the sink pad
of {mu,a}lawdec.
https://bugzilla.gnome.org/show_bug.cgi?id=681677

2012-08-09 19:41:34 +0300 Anton Belka <[email protected]>

* ext/flac/gstflacenc.c:
flacenc: allow a TOC with single alternative top-level entry
Allow a TOC that has a single alternative top-level entry
with multiple sequence sub-entries
https://bugzilla.gnome.org/show_bug.cgi?id=540891

2012-08-09 10:31:39 +0200 Sebastian Dröge <[email protected]>

* configure.ac:
configure: And fix the GTK check to use the correct pkg-config package name

2012-08-09 10:25:38 +0200 Sebastian Dröge <[email protected]>

* configure.ac:
configure: Fix GTK required version variable name

2012-08-09 08:35:23 +0100 Matthias Clasen <[email protected]>

* sys/v4l2/gstv4l2bufferpool.c:
v4l2: fix build with recent kernels, the v4l2_buffer input field was removed
This was unused apparently and removed in the kernel in commit:
From 2b719d7baf490e24ce7d817c6337b7c87fda84c1 Mon Sep 17 00:00:00 2001
From: Sakari Ailus <[email protected]>
Date: Wed, 2 May 2012 09:40:03 -0300
Subject: [PATCH] [media] v4l: drop v4l2_buffer.input and V4L2_BUF_FLAG_INPUT
Remove input field in struct v4l2_buffer and flag V4L2_BUF_FLAG_INPUT which
tells the former is valid. The flag is used by no driver currently.
https://bugzilla.gnome.org/show_bug.cgi?id=681491
Conflicts:
sys/v4l2/gstv4l2bufferpool.c

2012-08-08 17:25:36 -0700 Olivier Crête <[email protected]>

* gst/rtp/gstrtph264pay.c:
* tests/check/elements/rtp-payloading.c:
rtph264pay: Make it actually work after cleanups

2012-08-08 17:39:07 +0200 Sebastian Dröge <[email protected]>

* ext/jpeg/gstjpegenc.c:
* gst/matroska/matroska-demux.c:
* gst/multipart/multipartmux.c:
* gst/videomixer/videomixer2.c:
gst: Set alignment at the correct place of GstAllocationParams

2012-08-08 16:25:58 +0100 Tim-Philipp Müller <[email protected]>

* configure.ac:
* win32/common/config.h:
Back to development

=== release 0.11.93 ===

2012-08-08 15:22:04 +0100 Tim-Philipp Müller <[email protected]>

* configure.ac:
* gst-plugins-good.doap:
* win32/common/config.h:
Release 0.11.93

2012-08-08 15:17:22 +0100 Tim-Philipp Müller <[email protected]>

* Makefile.am:
* win32/MANIFEST:
* win32/common/tuner-enumtypes.c:
* win32/common/tuner-enumtypes.h:
* win32/common/tuner-marshal.c:
* win32/common/tuner-marshal.h:
win32: add generated tuner-marshal/enumtypes files for v4l2src and update
And gst-indent the right rtp marshal files; add missing files to MANIFEST.

2012-08-08 15:10:37 +0100 Tim-Philipp Müller <[email protected]>

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

2012-08-08 11:31:59 +0100 Tim-Philipp Müller <[email protected]>

* ext/wavpack/gstwavpackdec.c:
* gst/rtpmanager/gstrtpssrcdemux.c:
* sys/oss4/oss4-audio.c:
* sys/v4l2/gstv4l2bufferpool.c:
* sys/v4l2/gstv4l2object.c:
Silence some 'variable may be used uninitialized' compiler warnings
When compiling with -DG_DISABLE_ASSERT

2012-08-08 10:56:51 +0100 Tim-Philipp Müller <[email protected]>

* ext/jpeg/gstjpegdec.c:
* ext/libpng/gstpngdec.c:
* gst/isomp4/gstqtmoovrecover.c:
* tests/icles/ximagesrc-test.c:
No code with side-effects inside g_assert() please

2012-08-07 11:14:21 -0700 Olivier Crête <[email protected]>

* gst/udp/gstmultiudpsink.c:
multiudpsink: Return FLUSHING instead of ERROR on unlock
If the base class asks multiudpsink to unlock, then it should return
FLUSHING, not ERROR

2012-07-26 16:19:57 +0300 Anton Belka <[email protected]>

* ext/flac/gstflacenc.c:
* ext/flac/gstflacenc.h:
flacenc: add TOC support
Add TOC as embedded cuesheets in flac files.
https://bugzilla.gnome.org/show_bug.cgi?id=54089

2012-08-07 12:12:09 +0200 Mark Nauwelaerts <[email protected]>

* gst/audioparsers/gstflacparse.c:
flacparse: generate empty vorbiscomment for complete streamheaders if needed
Fixes https://bugzilla.gnome.org/show_bug.cgi?id=681335

2012-08-06 18:02:50 -0700 Olivier Crête <[email protected]>

* gst/rtpmanager/gstrtpssrcdemux.c:
rtpssrcdemux: Block pad while it is announced.
Block the RTP pad and associated RTCP pads while they are being
announced. This it to prevent a race where one is announced and
before the callback has connected it, the other one gets a buffer.
We can't use the "padlock" of ssrcdemux because it causes deadlocks.

2012-08-06 15:00:57 +0100 Tim-Philipp Müller <[email protected]>

* common:
common: un-do accidental common update revert in commit 7b5925b5

2012-08-06 14:50:53 +0200 Mark Nauwelaerts <[email protected]>

* gst/rtp/gstrtpmparobustdepay.c:
rtpmparobustdepay: set correct data_size for generated dummy frame
... which prevents getting stuck in a loop if such one is needed.

2012-08-06 14:50:03 +0200 Mark Nauwelaerts <[email protected]>

* gst/rtp/gstrtpmparobustdepay.c:
rtpmparobustdepay: improve and fix debug statement
... so it really informs about next rather than past frame.

2012-08-06 12:34:55 +0200 Mark Nauwelaerts <[email protected]>

* gst/rtp/gstrtpmparobustdepay.c:
rtpmparobustdepay: update available bytewriter space when repositioning
... and add some more assert to catch potential surprises early on.
Fixes https://bugzilla.gnome.org/show_bug.cgi?id=680558

2012-08-04 12:47:44 +0200 Sebastian Dröge <[email protected]>

* common:
* ext/dv/gstdvdemux.c:
* gst/avi/gstavidemux.c:
* gst/flv/gstflvdemux.c:
* gst/isomp4/qtdemux.c:
* gst/matroska/matroska-demux.c:
gst: Add stream-id to stream-start events

2012-08-04 12:54:32 +0200 Sebastian Dröge <[email protected]>

* gst/matroska/matroska-demux.c:
matroskademux: Chain up to the parent class' query handler if no pad is provided

2012-08-02 01:48:29 +0200 Andoni Morales Alastruey <[email protected]>

* sys/osxvideo/osxvideosink.h:
* sys/osxvideo/osxvideosink.m:
osxvideosink: add a better detection for the main run loop

2012-07-27 16:13:49 +0200 Xavi Artigas <[email protected]>

* sys/directsound/gstdirectsoundsink.c:
directsoundsink: Do not overwrite the DS buffer when testing for AC3 support
https://bugzilla.gnome.org/show_bug.cgi?id=680706
Conflicts:
sys/directsound/gstdirectsoundsink.c

2012-08-05 16:39:23 +0100 Tim-Philipp Müller <[email protected]>

* common:
Automatic update of common submodule
From 94ccf4c to 668acee

2012-08-03 16:13:52 +0100 Olivier Crête <[email protected]>

* gst/rtpmanager/gstrtpssrcdemux.c:
rtpssrcdemux: Release lock before signalling new pad
This prevents a deadlock where something would try to push an event
through the SSRC demux from the callback, causing the pads to be iterated
and the lock taken.

2012-08-04 16:10:16 +0100 Tim-Philipp Müller <[email protected]>

* ext/flac/gstflacenc.c:
* ext/flac/gstflactag.c:
* ext/shout2/gstshout2.c:
* ext/soup/gstsouphttpsrc.c:
* ext/speex/gstspeexdec.c:
* ext/speex/gstspeexenc.c:
* gst/audioparsers/gstflacparse.c:
* gst/avi/gstavidemux.c:
* gst/avi/gstavimux.c:
* gst/debugutils/gsttaginject.c:
* gst/flv/gstflvdemux.c:
* gst/icydemux/gsticydemux.c:
* gst/isomp4/gstqtmux.c:
* gst/isomp4/qtdemux.c:
* gst/matroska/matroska-demux.c:
* gst/matroska/matroska-parse.c:
* gst/matroska/matroska-read-common.c:
* gst/wavparse/gstwavparse.c:
* tests/check/elements/apev2mux.c:
* tests/check/elements/icydemux.c:
* tests/check/elements/id3demux.c:
* tests/check/elements/id3v2mux.c:
* tests/check/elements/qtmux.c:
* tests/check/elements/rganalysis.c:
* tests/check/pipelines/tagschecking.c:
gst_tag_list_free -> gst_tag_list_unref

2012-08-03 14:10:32 +0200 Mark Nauwelaerts <[email protected]>

* gst/rtsp/gstrtspsrc.c:
rtspsrc: manage race between connection closing and flushing
... where the former can happen in task thread and the latter in mainloop
upon downward state change.

2012-08-03 14:02:23 +0200 Mark Nauwelaerts <[email protected]>

* ext/flac/gstflacdec.c:
flacdec: improve and relax audio frame parsing
... so as to properly recognize first audio frame.
Fixes https://bugzilla.gnome.org/show_bug.cgi?id=681077
Conflicts:
ext/flac/gstflacdec.c

2012-08-01 12:16:41 +0200 René Stadler <[email protected]>

* gst/isomp4/qtdemux.c:
qtdemux: fix double unref of private tag buffer

2012-07-30 17:54:51 +0300 Anton Belka <[email protected]>

* gst/wavparse/gstwavparse.c:
wavparse: create TOC as needed
Avoid creating the toc if the wav has no or empty cue chunk.
Also a small code cleanup.

2012-07-28 11:26:01 +0100 Tim-Philipp Müller <[email protected]>

* gst/wavparse/gstwavparse.c:
wavparse: update for TOC API changes

2012-07-28 11:22:43 +0100 Tim-Philipp Müller <[email protected]>

* gst/matroska/matroska-read-common.c:
matroska: update for TOC API changes

2012-07-28 11:20:08 +0100 Tim-Philipp Müller <[email protected]>

* gst/audioparsers/gstflacparse.c:
flacparse: update for TOC API changes

2012-07-28 00:19:51 +0200 Sebastian Dröge <[email protected]>

* ext/dv/gstdvdemux.c:
* ext/flac/gstflactag.c:
* ext/soup/gstsouphttpsrc.c:
* ext/wavpack/gstwavpackdec.c:
* gst/audioparsers/gstflacparse.c:
* gst/audioparsers/gstmpegaudioparse.c:
* gst/avi/gstavidemux.c:
* gst/avi/gstavisubtitle.c:
* gst/debugutils/gsttaginject.c:
* gst/flv/gstflvdemux.c:
* gst/icydemux/gsticydemux.c:
* gst/isomp4/qtdemux.c:
* gst/matroska/matroska-demux.c:
* gst/matroska/matroska-read-common.c:
* gst/multipart/multipartdemux.c:
* gst/replaygain/gstrganalysis.c:
* gst/wavparse/gstwavparse.c:
* tests/check/elements/rganalysis.c:
* tests/check/elements/rgvolume.c:
tag: Update for taglist/tag event API changes

2012-07-27 12:05:44 +0200 Mark Nauwelaerts <[email protected]>

* gst/isomp4/gstqtmux.c:
* gst/isomp4/isomp4-plugin.c:
* gst/isomp4/qtdemux.c:
qt(de)mux: pass private blob tags in a sample
... rather than a buffer, and the detailed info in the sample info
rather than caps.

2012-07-27 11:31:13 +0200 Robert Swain <[email protected]>

* gst/videocrop/gstvideocrop.c:
videocrop: Don't return NULL from _transform_caps
If _transform_caps () returns NULL, the basetransform _transform_caps
tries to call gst_caps_is_subset () with a NULL subset which hits an
assertion.

2012-07-27 11:26:18 +0200 Mark Nauwelaerts <[email protected]>

* ext/flac/gstflacenc.c:
flacenc: obtain image type from the sample info

2012-07-27 11:25:49 +0200 Mark Nauwelaerts <[email protected]>

* ext/flac/gstflacenc.c:
flacenc: remove extraneous _unref
... since we did not obtain a buffer ref from the GstSample.

2012-07-27 10:14:23 +0200 Robert Swain <[email protected]>

* ext/flac/gstflacenc.c:
flacenc: Update to use GstSample tag setting API

2012-07-26 16:34:21 +0200 Mark Nauwelaerts <[email protected]>

* gst/rtp/gstrtpmparobustdepay.c:
rtpmparobustdepay: modify buffer data rather than buffer itself

2012-07-26 16:28:33 +0200 Mark Nauwelaerts <[email protected]>

* gst/rtp/gstrtpmparobustdepay.c:
rtpmparobustdepay: avoid leaking bytewriter instance

2012-07-26 16:04:23 +0200 Robert Swain <[email protected]>

* gst/deinterlace/gstdeinterlace.c:
deinterlace: Fix timestamp adjustment and caps

2012-07-26 16:03:57 +0200 Robert Swain <[email protected]>

* gst/deinterlace/gstdeinterlace.c:
deinterlace: Fix/simplify telecine state checks

2012-07-26 12:08:58 +0200 Robert Swain <[email protected]>

* gst/deinterlace/gstdeinterlace.c:
deinterlace: Improve debug output

2012-07-26 12:08:36 +0200 Robert Swain <[email protected]>

* gst/deinterlace/gstdeinterlace.c:
deinterlace: Fix low-latency pattern locking

2012-07-24 16:19:53 +0200 Robert Swain <[email protected]>

* gst/deinterlace/gstdeinterlace.c:
deinterlace: RFF should be ignored in deinterlace
RFF only occurs on progressive frames in telecine sequences. For
deinterlace, we don't want these repeated fields as we will simply be
pushing the progressive frame and then moving on.
However, we need to consider RFF in order to correctly identify patterns
and adjust the timestamps.

2012-07-24 14:59:47 +0200 Robert Swain <[email protected]>

* gst/deinterlace/gstdeinterlace.c:
deinterlace: Improve process logic
The logic now works better if we filter orphans, then progressive, then
telecine interlaced fields which need to be woven and fall through to
interlace. Telecine interlaced fields will be regularly deinterlaced if
there is no pattern lock for us to be sure that we have a telecine
pattern.
Telecine sequences that aren't 24fps progressive with RFF flags can't
really be tested until fieldanalysis is ported.

2012-07-25 16:02:34 +0200 Mark Nauwelaerts <[email protected]>

* ext/flac/gstflacenc.c:
flacenc: only set complete output caps once
... so as to avoid downstream complaints about missing streamheaders.

2012-07-25 15:29:04 +0200 Mark Nauwelaerts <[email protected]>

* ext/flac/gstflacdec.c:
flacdec: also support S24_32 output

2012-07-25 15:28:14 +0200 Mark Nauwelaerts <[email protected]>

* ext/flac/gstflacenc.c:
flacenc: pass correct parameters to encoder lib

2012-07-25 14:57:13 +0200 Mark Nauwelaerts <[email protected]>

* ext/flac/gstflacenc.c:
flacenc: adjust to modified audioencoder getcaps helper API

2012-07-25 12:50:01 +0200 Wim Taymans <[email protected]>

* gst/rtsp/gstrtspsrc.c:
rtsp: go and stay in the loop function on PLAY
When we have a PLAY request, go into the LOOP function next. When we are
looping, keep on looping until we are told otherwise.
This fixed rtsp and TCP connections.
Fixes https://bugzilla.gnome.org/show_bug.cgi?id=680551

2012-07-25 12:49:35 +0200 Wim Taymans <[email protected]>

* gst/rtsp/gstrtspsrc.c:
rtsp: set caps after activating the pad

2012-07-25 12:49:07 +0200 Wim Taymans <[email protected]>

* gst/rtp/gstrtph264depay.c:
h264depay: small cleanups

2012-07-25 10:08:52 +0200 Wim Taymans <[email protected]>

* gst/isomp4/gstrtpxqtdepay.c:
xqtdepay: fix buffer refcount error
After pushing the buffer into the adapter, we should not let the baseclass push
it out anymore. This error was introduced while porting to 0.11.
See https://bugzilla.gnome.org/show_bug.cgi?id=680540

2012-07-24 21:41:53 +0200 Stefan Sauer <[email protected]>

* gst/level/gstlevel.c:
level: remove obsolete liboil comment

2012-07-24 21:11:18 +0200 Mark Nauwelaerts <[email protected]>

* gst/matroska/matroska-demux.c:
* gst/matroska/matroska-demux.h:
matroskademux: push mode: increase segment accuracy following seek
Conflicts:
gst/matroska/matroska-demux.c

2012-07-24 16:41:51 +0200 Mark Nauwelaerts <[email protected]>

* gst/matroska/matroska-demux.c:
matroskademux: perform proper KEY_UNIT seek also in push mode
Conflicts:
gst/matroska/matroska-demux.c

2012-07-24 19:04:39 +0100 Tim-Philipp Müller <[email protected]>

* gst/udp/gstudpsrc.c:
udpsrc: don't crash dereferencing NULL error when leaving multicast group on shutdown
Strangely enough, if we do pass an error variable to be filled, we
no longer get an error on leaving.

2012-07-24 15:55:12 +0200 Mark Nauwelaerts <[email protected]>

* gst/avi/gstavidemux.c:
avidemux: rearrange some checks to avoid NULL use

2012-07-24 15:38:24 +0200 Mark Nauwelaerts <[email protected]>

* gst/avi/gstavidemux.c:
avidemux: use same fourcc to determine caps in determining uncompressed-ness
Fixes https://bugzilla.gnome.org/show_bug.cgi?id=673898
Conflicts:
gst/avi/gstavidemux.c

2012-07-24 15:36:54 +0200 Mark Nauwelaerts <[email protected]>

* gst/avi/gstavidemux.c:
Revert "avidemux: Don't consider 0 fcc_handler as uncompressed."
This reverts commit c6b9f5b25ab435669816a07049b0e5a8f01e09ca.
fourcc GST_RIFF_rgb = 0 still leads to raw uncompressed rgb caps.
See also https://bugzilla.gnome.org/show_bug.cgi?id=673898

2012-07-24 12:10:46 +0100 Tim-Philipp Müller <[email protected]>

* ext/jpeg/gstjpegdec.c:
jpegdec: fix up example pipeline some more
No more ffmpegcolorspace

2012-07-20 16:30:00 +0300 Sreerenj Balachandran <[email protected]>

* ext/jpeg/gstjpegdec.c:
jpegdec: Fix the example gst-launch pipeline.

2012-07-24 12:33:33 +0200 Mark Nauwelaerts <[email protected]>

* gst/matroska/matroska-demux.c:
matroskademux: avoid NULL access when checking subtitle
Fixes https://bugzilla.gnome.org/show_bug.cgi?id=680388

2012-07-24 12:22:08 +0200 Edward Hervey <[email protected]>

* gst/audioparsers/gstaacparse.c:
aacparse: Reset parser when we have caps without codec_data
This ensures the detection (and proper downstream caps settings) will
actually happen when we have new incoming caps without codec_data.
This was easily triggered by streams from matroskademux which initially
provided caps with a constructed codec_data, but then pushed new caps
without the codec_data once it detected the stream was adts.

2012-07-24 09:17:09 +0200 Wim Taymans <[email protected]>

* gst/videomixer/blend.c:
* gst/videomixer/blendorc-dist.c:
* gst/videomixer/blendorc-dist.h:
* gst/videomixer/blendorc.orc:
videomixer: prefix orc functions with video_mixer_orc_

2012-07-24 09:13:48 +0200 Wim Taymans <[email protected]>

* gst/videobox/gstvideobox.c:
* gst/videobox/gstvideoboxorc-dist.c:
* gst/videobox/gstvideoboxorc-dist.h:
* gst/videobox/gstvideoboxorc.orc:
videobox: prefix orc functions with video_box_orc_

2012-07-23 18:51:00 +0200 Christian Fredrik Kalager Schaller <[email protected]>

* gst-plugins-good.spec.in:
Update spec file with latest changes

2012-07-23 17:37:58 +0200 Mark Nauwelaerts <[email protected]>

* gst/matroska/matroska-demux.c:
matroskademux: generate correct segment stream time
Fixes https://bugzilla.gnome.org/show_bug.cgi?id=680275

2012-07-23 16:42:56 +0200 Wim Taymans <[email protected]>

* gst/rtp/gstrtpj2kdepay.c:
* gst/rtp/gstrtpj2kdepay.h:
* gst/rtp/gstrtpj2kpay.c:
* gst/rtp/gstrtpj2kpay.h:
rtp: always use buffer lists

2012-07-23 15:24:17 +0200 Patricia Muscalu <[email protected]>

* gst/rtp/gstrtpmp4vpay.c:
* gst/rtp/gstrtpmp4vpay.h:
rtpmp4vpay: always enable buffer-lists

2012-07-23 15:22:24 +0200 Patricia Muscalu <[email protected]>

* gst/rtp/gstrtpjpegpay.c:
* gst/rtp/gstrtpjpegpay.h:
rtpjpegpay: always enable buffer-lists

2012-07-23 15:49:04 +0200 Wim Taymans <[email protected]>

* configure.ac:
* gst/deinterlace/gstdeinterlace.c:
deinterlace: get frame flags correctly
Also move the deinterlace plugin to ported status

2012-07-23 15:33:54 +0200 Mark Nauwelaerts <[email protected]>

* gst/matroska/matroska-demux.c:
matroskademux: proper parse recovery after seek
Fixes https://bugzilla.gnome.org/show_bug.cgi?id=680427

2012-07-23 12:39:05 +0200 Mark Nauwelaerts <[email protected]>

* gst/flv/gstflvdemux.c:
flvdemux: clear old segment event when requesting new one
Fixes https://bugzilla.gnome.org/show_bug.cgi?id=680283

2012-07-23 10:32:36 +0200 Sebastian Dröge <[email protected]>

* ext/jpeg/gstjpegdec.c:
* ext/libpng/gstpngdec.c:
ext: Update for video base classes API changes

2012-07-23 08:49:07 +0200 Alban Browaeys <[email protected]>

* gst/wavparse/gstwavparse.c:
wavparse: convert all non GST_FORMAT_BYTES to format bytes.
Convert all non GST_FORMAT_BYTES to format bytes:
fixes:
GStreamer-CRITICAL **: gst_query_set_duration: assertion `format ==
g_value_get_enum (gst_structure_id_get_value (s, GST_QUARK (FORMAT)))'
failed
when playing more than one wav stream.
gst-plugins-base/tests/icles/playback/test7 uri1.wav uri2.wav

2012-07-23 09:25:23 +0200 Sebastian Dröge <[email protected]>

* gst/wavparse/gstwavparse.c:
wavparse: Don't fail if more data then needed is available when parsing cue chunks
Fixes bug #680328.

2012-07-23 09:22:20 +0200 Sebastian Dröge <[email protected]>

* gst/wavparse/gstwavparse.c:
wavparse: Some minor cleanup to the cue/labl parsing

2012-07-23 08:45:28 +0200 Sebastian Dröge <[email protected]>

* common:
Automatic update of common submodule
From 98e386f to 94ccf4c

2012-07-19 14:55:45 +0200 Robert Swain <[email protected]>

* gst/deinterlace/gstdeinterlace.c:
* gst/deinterlace/gstdeinterlace.h:
* gst/deinterlace/gstdeinterlacemethod.c:
* gst/deinterlace/gstdeinterlacemethod.h:
* gst/deinterlace/tvtime/greedyh.c:
* gst/deinterlace/tvtime/tomsmocomp/TomsMoCompAll.inc:
deinterlace: Port to 1.0
This requires the additional INTERLACED buffer flag recently added to
-base

2012-07-20 15:18:46 +0200 Wim Taymans <[email protected]>

* gst/interleave/interleave.c:
interleave: convert the output segment to time
Convert the stored input segment to time before pushing it out.
Conflicts:
gst/interleave/interleave.c

2012-07-20 13:12:44 +0200 Wim Taymans <[email protected]>

* gst/interleave/interleave.c:
* gst/interleave/interleave.h:
interleave: try to fix segment handling
Conflicts:
gst/interleave/interleave.c

2012-07-20 15:28:21 +0200 Sebastian Dröge <[email protected]>

* gst/matroska/matroska-demux.c:
matroskademux: Non-update seeks should still make sure that reverse playback status is reset
Conflicts:
gst/matroska/matroska-demux.c

2012-07-20 15:18:21 +0200 Sebastian Dröge <[email protected]>

* gst/matroska/matroska-demux.c:
matroskademux: Properly initialize from_offset and from_time

2012-07-20 14:25:43 +0200 Sebastian Dröge <[email protected]>

* gst/matroska/matroska-demux.c:
matroskademux: We need an index and index entry for reverse playback
Reverse playback does not work with index-less files yet.

2012-07-20 14:10:41 +0200 Mark Nauwelaerts <[email protected]>

* gst/wavparse/gstwavparse.c:
wavparse: clean up push mode segment handling
Fixes https://bugzilla.gnome.org/show_bug.cgi?id=680277

2012-07-20 13:35:29 +0200 Mark Nauwelaerts <[email protected]>

* gst/isomp4/qtdemux.c:
qtdemux: properly transform incoming segment event
... which is really useful for proper push mode seeking.
Fixes https://bugzilla.gnome.org/show_bug.cgi?id=680278

2012-07-20 11:07:58 +0200 Sebastian Dröge <[email protected]>

* gst/matroska/matroska-demux.c:
* gst/matroska/matroska-demux.h:
matroskademux: Fix reverse playback for seeks without stop position
Conflicts:
gst/matroska/matroska-demux.c
gst/matroska/matroska-demux.h

2012-07-20 10:48:34 +0200 Sebastian Dröge <[email protected]>

* gst/matroska/matroska-demux.c:
matroskademux: Only take the stream_start_time into account for SET seeks
For other seeks the stream_start_time is already added to the
segment values.
Conflicts:
gst/matroska/matroska-demux.c

2012-07-08 20:36:22 +0300 Anton Belka <[email protected]>

* gst/wavparse/gstwavparse.c:
* gst/wavparse/gstwavparse.h:
wavparse: Add TOC support
Add support for:
* Cue Chunk
* Associated Data List Chunk
* Label Chunk
https://bugzilla.gnome.org/show_bug.cgi?id=677306

2012-05-09 15:58:16 +0200 Maria Giovanna Chiossa <mariagiovanna.chiossa at selexelsag.com>

* gst/rtsp/gstrtspsrc.c:
rtspsrc: also set UDP buffer size in multicast
Also set the UDP buffer size in multicast mode.
Fixes https://bugzilla.gnome.org/show_bug.cgi?id=675448

2012-07-18 23:43:59 +0100 Tim-Philipp Müller <[email protected]>

* gst/avi/gstavidemux.c:
avidemux: fix header parsing in push mode
Fix 'break' that got warped to the wrong place,
probably as part of a merge. Fixes GST_IS_BUFFER
criticals in parse_idit() when being accidentally
passed a NULL buffer because of the missing break.
gst-launch-1.0 playbin uri=http://docs.gstreamer.com/media/sintel_trailer-480i.avi

2012-07-18 22:47:22 +0200 Alban Browaeys <[email protected]>

* configure.ac:
* ext/soup/gstsouphttpsrc.c:
soup: deprecated soup_message_headers _get -> _get_one
https://bugzilla.gnome.org/show_bug.cgi?id=680206

2012-07-18 18:27:40 +0200 Edward Hervey <[email protected]>

* ext/jpeg/gstjpegdec.c:
* ext/libpng/gstpngdec.c:
jpeg/png: Call video_decoder_negotiate()

2012-07-18 17:57:59 +0200 Wim Taymans <[email protected]>

* gst/debugutils/gstpushfilesrc.c:
update for ghostpad changes

2012-07-18 11:36:27 +0200 Sebastian Dröge <[email protected]>

* gst/matroska/matroska-demux.c:
matroskademux: Pass seek rate to upstream seek events in push mode
Fixes bug #679435.
Conflicts:
gst/matroska/matroska-demux.c

2012-07-17 16:38:27 +0200 Wim Taymans <[email protected]>

* gst/isomp4/gstrtpxqtdepay.c:
* gst/rtp/gstasteriskh263.c:
* gst/rtpmanager/gstrtpjitterbuffer.c:
* gst/rtpmanager/gstrtpptdemux.c:
* gst/rtpmanager/gstrtpssrcdemux.c:
* gst/rtpmanager/rtpsession.c:
* gst/rtsp/gstrtpdec.c:
update for RTP buffer api changes

2012-07-16 11:07:44 +0200 Patricia Muscalu <[email protected]>

* gst/rtp/gstrtph264pay.c:
rtph264pay: use buffer lists
Fixes https://bugzilla.gnome.org/show_bug.cgi?id=679994

2012-07-17 10:01:54 +0200 Sebastian Dröge <[email protected]>

* gst/audioparsers/gstflacparse.c:
flacparse: Fix parsing of ISRC from the cuesheets

2012-07-05 14:15:25 +0300 Anton Belka <[email protected]>

* gst/audioparsers/gstflacparse.c:
* gst/audioparsers/gstflacparse.h:
flacparse: add TOC support
Add support embedded cuesheets in flac files.
Parsing METADATA_BLOCK_CUESHEET as TOC.
https://bugzilla.gnome.org/show_bug.cgi?id=540891

2012-07-13 14:43:31 +0200 Mark Nauwelaerts <[email protected]>

* gst/audioparsers/gstflacparse.c:
flacparse: avoid some more frame misparsing by additional header sanity check
... using a required constant blocking_strategy bit.
https://bugzilla.gnome.org/show_bug.cgi?id=679807

2012-07-13 13:51:48 +0200 Edward Hervey <[email protected]>

* ext/dv/gstdvdemux.c:
* gst/avi/gstavidemux.c:
* gst/flv/gstflvdemux.c:
* gst/isomp4/qtdemux.c:
* gst/matroska/matroska-demux.c:
demux: Push STREAM_START event when needed

2012-07-11 13:10:07 +0200 Stefan Sauer <[email protected]>

* gst/isomp4/gstqtmux.c:
qtmux: avoid warning if both ts are equal

2012-07-11 12:28:23 +0100 Tim-Philipp Müller <[email protected]>

* gst/udp/gstmultiudpsink.c:
multiudpsink: check the right size when warning about too large udp packets
What matters is the total size, not the size of any of the
individual memory chunks that make up the packet.

2012-07-10 14:38:21 +0200 Wim Taymans <[email protected]>

* gst/autodetect/gstautoaudiosink.c:
* gst/autodetect/gstautoaudiosink.h:
* gst/autodetect/gstautovideosink.c:
* gst/autodetect/gstautovideosink.h:
autodetect: proxy ts-offset properties
Proxy the ts-offset property in the audio*sink elements.
Fixes https://bugzilla.gnome.org/show_bug.cgi?id=679343

2012-07-09 16:27:10 +0200 Wim Taymans <[email protected]>

* gst/isomp4/qtdemux.c:
* sys/v4l2/gstv4l2bufferpool.c:
fix for allocator API changes

2012-07-09 12:22:02 +0200 Mark Nauwelaerts <[email protected]>

* gst/avi/gstavimux.c:
* gst/matroska/matroska-demux.c:
* gst/wavparse/gstwavparse.c:
update for riff field rename

2012-05-21 13:54:51 +0200 Mathias Hasselmann <[email protected]>

* tests/check/Makefile.am:
tests: drop redundant elements_level_LDADD line
https://bugzilla.gnome.org/show_bug.cgi?id=676302

2012-07-08 13:30:34 +0100 Tim-Philipp Müller <[email protected]>

* tests/check/elements/jpegdec.c:
tests: minor jpegdec clean-ups and fixes
Fix race condition in eos checking and a leak. And
build pipeline without parse_launch.

2012-05-21 13:53:54 +0200 Mathias Hasselmann <[email protected]>

* tests/check/Makefile.am:
* tests/check/elements/.gitignore:
* tests/check/elements/jpegdec.c:
* tests/files/image.jpg:
tests: Add some basic tests for jpegdec
https://bugzilla.gnome.org/show_bug.cgi?id=676302

2012-07-06 19:11:02 +0100 Tim-Philipp Müller <[email protected]>

* gst/rtp/gstrtph264pay.c:
rtph264pay: avoid some relocations

2012-07-06 14:49:18 +0100 Tim-Philipp Müller <[email protected]>

* gst/rtp/gstrtpmp4vpay.c:
rtpmp4vpay: remove deprecated send-config property
Use config-interval instead.

2012-07-06 14:42:19 +0100 Tim-Philipp Müller <[email protected]>

* gst/rtp/gstrtph264depay.c:
rtph264depay: remove deprecated "byte-stream" and "access-unit" properties
These will be picked automatically based on downstream caps now, so
if you want the depayloader to output a specific format, make sure
the element downstream advertises that preference or use a capsfilter
after the depayloader to force it.

2012-07-06 14:13:54 +0100 Tim-Philipp Müller <[email protected]>

* gst/rtp/gstrtph264pay.c:
rtph264pay: remove deprecated and non-functional "profile-level-id" property
This is now optionally taken from downstream caps, so can be
specified via a capsfilter after the payloader.

2012-07-06 15:07:51 +0200 Mark Nauwelaerts <[email protected]>

* gst/audioparsers/gstaacparse.c:
aacparse: perform additional sanity check before confirming ADTS format
... and tweak confusing debug message.

2012-07-06 15:29:14 +0200 Mark Nauwelaerts <[email protected]>

* gst/audioparsers/gstaacparse.c:
aacparse: remove unhelpful stray debug message

2012-07-06 13:16:00 +0100 Tim-Philipp Müller <[email protected]>

* gst/rtpmanager/gstrtpsession.c:
rtpsession: remove deprecated and unused "ntp-ns-base" property

2012-07-06 12:57:20 +0100 Tim-Philipp Müller <[email protected]>

* gst/isomp4/gstqtmux-doc.c:
docs: update isomp4 docs for gppmux -> 3gppmux change as well

2012-07-06 12:54:02 +0100 Tim-Philipp Müller <[email protected]>

* gst/isomp4/gstqtmux.c:
* gst/isomp4/gstqtmuxmap.c:
* tests/check/pipelines/tagschecking.c:
isomp4: remove gppmux, which was deprecated in favour of 3gppmux

2012-07-06 12:49:54 +0100 Tim-Philipp Müller <[email protected]>

* gst/smpte/gstsmpte.c:
smtp: remove deprecated "fps" property

2012-07-06 12:46:30 +0100 Tim-Philipp Müller <[email protected]>

* gst/multipart/multipartdemux.c:
* gst/multipart/multipartdemux.h:
multipartdemux: remove deprecated and unused "autoscan" property
Replaced by boundary=NULL.

2012-07-06 09:07:41 +0100 Tim-Philipp Müller <[email protected]>

* gst/rtp/gstrtph263ppay.c:
* tests/check/elements/rtp-payloading.c:
rtph263ppay: accept any h263 input unless downstream forces specific requirements
rtph263ppay should accept any input compatible with its sink template
caps if it just outputs to e.g. udpsink or fakesink.
rtph263ppay ! rtph263pdepay should also work with any compatible input.
This would fail before with not-negotiated errors because the get_caps
function would see the encoding-name in the depayloader's template caps
and default to baseline H.263 because there's no profile/level information
in those caps, which is the right thing to do if downstream has filtercaps
from an SDP, but not if those fields are absent because they can be
anything like with the depayloader's template caps. Makes
videotestsrc ! avenc_h263p ! rtph263ppay ! rtph263pdepay ! fakesink
work.

2012-07-05 22:57:05 +0100 Tim-Philipp Müller <[email protected]>

* tests/check/elements/rtp-payloading.c:
tests: fix h263p payload ! depayload unit test
Need to add h263version field to input caps since the
payloader sink get_caps function will contain it in the
the caps, and the stricter caps subset check requires
this to be present in the input caps as well then.

2012-07-06 11:50:50 +0200 Wim Taymans <[email protected]>

* ext/aalib/gstaasink.c:
* ext/jpeg/gstjpegenc.c:
* ext/libpng/gstpngenc.c:
* sys/v4l2/gstv4l2sink.c:
update for query api changes

2012-07-06 11:26:46 +0200 Wim Taymans <[email protected]>

* ext/dv/gstdvdec.c:
* ext/jpeg/gstjpegdec.c:
* ext/libpng/gstpngdec.c:
* gst/rtp/gstrtpvrawdepay.c:
* sys/v4l2/gstv4l2src.c:
update for query api changes

2012-07-06 11:02:24 +0200 Wim Taymans <[email protected]>

* ext/aalib/gstaasink.c:
* ext/jpeg/gstjpegenc.c:
* ext/libpng/gstpngenc.c:
* sys/v4l2/gstv4l2sink.c:
update for allocation query changes

2012-07-05 15:14:33 +0100 Tim-Philipp Müller <[email protected]>

* tests/check/elements/rgvolume.c:
tests: fix rgvolume unit test event handling
Must flush after EOS before sending more buffers or
another EOS event, or the event or buffer will be
rejected. Also send a SEGMENT event at the start
of each stream for good measure.

2012-07-05 13:13:09 +0200 Sebastian Dröge <[email protected]>

* ext/dv/gstdvdemux.c:
* gst/avi/gstavidemux.c:
* gst/flv/gstflvdemux.c:
* gst/imagefreeze/gstimagefreeze.c:
* gst/isomp4/qtdemux.c:
* gst/matroska/matroska-demux.c:
* gst/matroska/matroska-parse.c:
* gst/rtsp/gstrtspsrc.c:
* gst/wavparse/gstwavparse.c:
gst: Implement segment-done event

2012-07-05 12:35:49 +0200 Sebastian Dröge <[email protected]>

* gst/matroska/matroska-demux.c:
matroskademux: Remove the TOC query handling

2012-07-04 19:52:22 +0200 Sebastian Dröge <[email protected]>

* gst/matroska/matroska-demux.c:
* gst/matroska/matroska-mux.c:
* gst/matroska/matroska-read-common.c:
matroska: Update for new GstToc API
TOC support in matroskamux is disabled for now as it was broken anyway.

2012-07-04 23:57:18 +0100 Tim-Philipp Müller <[email protected]>

* tests/check/elements/rganalysis.c:
tests: fix rganalysis unit test event handling
Must flush after EOS before sending more buffers or
another EOS event, or the event or buffer will be
rejected. Also send a SEGMENT event at the start
of each stream for good measure.

2012-07-04 18:58:46 +0100 Tim-Philipp Müller <[email protected]>

* gst/imagefreeze/gstimagefreeze.c:
imagefreeze: clear 0 DTS on buffers output, as sinks will prefer DTS over PTS for syncing
Since the initial decoded still image buffer will have dts=pts=0, and
we only set PTS on buffers we push out, all buffers pushed out would
have a DTS of 0. Sinks, however, will prefer DTS over PTS if both are
set, and will therefore always see a timestamp of 0 no matter what
the PTS is set to.
Fixes unit test too.

2012-07-04 20:59:03 +0400 Руслан Ижбулатов <[email protected]>

* sys/directsound/gstdirectsoundsink.c:
directsoundsink: Fix query function implementation; more debugging

2012-07-04 19:41:52 +0400 Руслан Ижбулатов <[email protected]>

* sys/directsound/gstdirectsoundsink.c:
directsoundsink: Fix spec stuff in directsoundsink

2012-05-31 19:22:47 +0200 Andoni Morales Alastruey <[email protected]>

* sys/directsound/gstdirectsoundsink.c:
directsoundsink: fix access to invalid pointer in set_volume

2012-06-13 12:12:39 +0200 Sebastian Dr=C3=B6ge <[email protected]>

* sys/directsound/gstdirectsoundsink.c:
directsoundsink: Fix caps leaks

2012-05-29 11:37:59 +0000 Andoni Morales Alastruey <[email protected]>

* sys/directsound/gstdirectsoundsink.c:
directsoundsink: fix acceptcaps check

2012-05-25 10:14:57 +0000 Andoni Morales Alastruey <[email protected]>

* sys/directsound/gstdirectsoundsink.c:
directsoundsink: use helper function to check for spdif formats

2012-05-25 10:19:09 +0000 Andoni Morales Alastruey <[email protected]>

* sys/directsound/gstdirectsoundsink.c:
directsoundsink: add support for DTS

2012-05-08 16:23:42 +0200 Andoni Morales Alastruey <[email protected]>

* sys/directsound/gstdirectsoundsink.c:
directsoundsink: force 48000 kHz force AC-3 over spdif

2012-07-04 17:42:49 +0400 Andoni Morales Alastruey <[email protected]>

* sys/directsound/gstdirectsoundsink.c:
directsoundsink: add support for ac-3 over spdif

2012-07-04 12:37:40 +0100 Tim-Philipp Müller <[email protected]>

* tests/check/elements/deinterlace.c:
tests: disable deinterlace test for now, element still needs to be ported
But leave it active and print a FIXME. Porting is in progress.

2012-07-03 19:38:39 +0100 Tim-Philipp Müller <[email protected]>

* gst/interleave/deinterleave.c:
deinterleave; downgrade caps change failure debug message
Add some more info and downgrade to warning, so
it doesn't look like the unit test failed.

2012-07-03 17:52:11 +0100 Tim-Philipp Müller <[email protected]>

* gst/audiofx/audiopanorama.c:
audiopanorama: fix negotiation and unit test
Must remove a possibly-fixed channel-mask field if
we're going to set unfixed channels on the structure,
or a different channel count.

2012-07-03 17:26:26 +0200 Sebastian Dröge <[email protected]>

* gst/matroska/matroska-demux.c:
matroskademux: Only push the TOC event, the message is handled by the sinks

2012-07-03 12:47:58 +0900 Javier Jardón <[email protected]>

* tests/examples/equalizer/demo.c:
* tests/examples/spectrum/demo-audiotest.c:
* tests/icles/gdkpixbufsink-test.c:
tests: do not use deprecated gtk+ symbols
https://bugzilla.gnome.org/show_bug.cgi?id=679301

2012-07-03 09:27:17 +0100 Tim-Philipp Müller <[email protected]>

* configure.ac:
configure: require Gtk+ 3.0 for tests/examples

2012-07-03 12:57:18 +0900 Javier Jardón <[email protected]>

* gst/rtp/gstrtpL16depay.c:
* gst/rtp/gstrtpmpadepay.c:
* gst/rtp/gstrtpvorbispay.c:
* gst/rtp/gstrtpvrawdepay.c:
rtp: remove some outdated comments
https://bugzilla.gnome.org/show_bug.cgi?id=679301

2012-06-29 11:51:30 +0100 Tim-Philipp Müller <[email protected]>

* sys/osxvideo/osxvideosink.m:
osxvideosink: default to force-aspect-ratio=true

2012-06-28 20:03:05 +0100 Tim-Philipp Müller <[email protected]>

* gst/debugutils/rndbuffersize.c:
rndbuffersize: add push mode support
https://bugzilla.gnome.org/show_bug.cgi?id=656317

2012-06-28 11:29:55 +0200 David Corvoysier <[email protected]>

* gst/isomp4/qtdemux.c:
isomp4: Try to seek upstream before processing seek push event
When it receives a seek in push mode, the qtdemux should first try to push the event upstream, and only if upstream fails fall back to
its own seek logic.

2012-06-28 11:47:20 +0200 David Corvoysier <[email protected]>

* gst/isomp4/qtdemux.c:
isomp4: Allow duration queries to be forwarded upstream
When receiving a duration query for TIME format, try to query upstream, and only if upstream fails fall back to qtdemux duration handling.

2012-06-28 11:59:11 +0200 Wim Taymans <[email protected]>

* gst/rtp/gstrtph264pay.c:
* gst/rtp/gstrtph264pay.h:
rtph264pay: cleanups
Use the caps properties for alignment and format.
Remove some old properties, we always want to use bufferlists when we can now.

2012-06-28 11:32:03 +0200 Wim Taymans <[email protected]>

* gst/rtp/gstrtph264pay.c:
h264pay: prefer AVC, it's easier to parse etc

2012-06-27 09:09:32 +0200 Wim Taymans <[email protected]>

* ext/jpeg/gstjpegenc.c:
jpegenc: mark all output frames as keyframes

2012-06-26 18:48:11 +0100 Tim-Philipp Müller <[email protected]>

* gst/matroska/matroska-read-common.c:
matroska: update for GstToc API additions

2012-06-26 17:04:41 +0200 Wim Taymans <[email protected]>

* gst/matroska/matroska-demux.c:
matroska: set interlace-mode

2012-06-26 13:19:02 +0200 Wim Taymans <[email protected]>

* sys/v4l2/gstv4l2bufferpool.c:
v4l2: improve debug

2012-06-26 13:02:13 +0200 Wim Taymans <[email protected]>

* sys/v4l2/gstv4l2bufferpool.c:
Revert "v4l2: free kernel buffers before allocating new ones"
This reverts commit 1b09bc609a578e731f0dbc8f6e698e25d8f4c5f8.
Seems to make libv4l2 complain, maybe because we call REQBUFS with 0 buffers
before we allocated buffers.

2012-06-26 12:07:47 +0200 Wim Taymans <[email protected]>

* sys/v4l2/gstv4l2bufferpool.c:
v4l2: free kernel buffers before allocating new ones
See https://bugzilla.gnome.org/show_bug.cgi?id=670257

2012-06-26 12:07:29 +0200 Wim Taymans <[email protected]>

* sys/v4l2/gstv4l2src.c:
v4l2src: improve debug

2012-06-26 11:14:59 +0200 Wim Taymans <[email protected]>

* sys/v4l2/gstv4l2bufferpool.c:
v4l2: setup strides and offsets for all planes

2012-06-25 20:11:53 +0100 Tim-Philipp Müller <[email protected]>

* gst/matroska/matroska-mux.c:
matroska-mux: update for GstTocSetter changes

2012-06-25 13:31:16 +0200 Sebastian Dröge <[email protected]>

* gst/matroska/matroska-demux.c:
matroskademux: Return FALSE from queries if we can't answer POSITION/DURATION queries

2012-06-21 17:15:11 +0300 Anton Belka <[email protected]>

* gst/matroska/matroska-demux.c:
matroskademux: Return FALSE from TOC query if no TOC exists instead of an empty TOC

2012-06-24 22:51:16 +0100 Tim-Philipp Müller <[email protected]>

* gst/matroska/matroska-demux.c:
* gst/matroska/matroska-mux.c:
* gst/matroska/matroska-read-common.c:
matroska: update for GstToc API changes

2012-06-23 14:57:28 +0100 Tim-Philipp Müller <[email protected]>

* gst/rtsp/gstrtspsrc.c:
rtspsrc: update for gst_element_make_from_uri() changes

2012-06-20 12:31:01 +0200 Wim Taymans <[email protected]>

* tests/check/elements/flvdemux.c:
* tests/check/elements/flvmux.c:
* tests/check/elements/id3demux.c:
update for bus api changes

2012-06-20 10:33:42 +0200 Wim Taymans <[email protected]>

* ext/dv/gstdvdemux.c:
* gst/avi/gstavidemux.c:
* gst/debugutils/rndbuffersize.c:
* gst/flv/gstflvdemux.c:
* gst/imagefreeze/gstimagefreeze.c:
* gst/isomp4/gstqtmoovrecover.c:
* gst/isomp4/qtdemux.c:
* gst/matroska/matroska-demux.c:
* gst/rtpmanager/gstrtpjitterbuffer.c:
* gst/rtsp/gstrtspsrc.c:
* gst/wavparse/gstwavparse.c:
update for task api change

2012-06-20 09:59:49 +0200 Wim Taymans <[email protected]>

* gst/rtpmanager/gstrtpjitterbuffer.c:
* tests/examples/spectrum/demo-audiotest.c:
* tests/examples/spectrum/demo-osssrc.c:
update for clock api changes

2012-06-19 12:15:33 +0200 Josep Torra <[email protected]>

* sys/osxaudio/Makefile.am:
* sys/osxaudio/gstosxaudiosink.c:
* sys/osxaudio/gstosxaudiosink.h:
* sys/osxaudio/gstosxcoreaudio.h:
* sys/osxaudio/gstosxringbuffer.c:
* sys/osxaudio/gstosxringbuffer.h:
osxaudiosink: respect the prefered channel layout
In OSX is allowed to configure the default audio output device,
prefered channel layout and speaker positions through the tool
"Audio MIDI Setup".

2012-04-30 22:59:58 +0200 Matej Knopp <[email protected]>

* gst/matroska/matroska-demux.c:
matroska-demux: Send gap events for subtitle streams

2012-06-17 01:00:40 +0100 Tim-Philipp Müller <[email protected]>

* gst/multifile/gstsplitfilesrc.c:
splitfilesrc: fix up docs for 0.11

2012-06-16 23:29:41 +0100 Tim-Philipp Müller <[email protected]>

* gst/multifile/gstsplitfilesrc.c:
splitfilesrc: small uri handler fixup and some more docs
Get URI location using gst_uri_get_location(), so any
escaped bits get unescaped.
https://bugzilla.gnome.org/show_bug.cgi?id=609049

2012-06-17 00:59:21 +0100 Tim-Philipp Müller <[email protected]>

* gst/multifile/gstsplitfilesrc.c:
splitfilesrc: re-port to 0.11

2012-06-16 19:06:25 +0100 Bastien Nocera <[email protected]>

* gst/multifile/gstsplitfilesrc.c:
splitfilesrc: Implement splitfile:// URI scheme
https://bugzilla.gnome.org/show_bug.cgi?id=609049
Conflicts:
gst/multifile/gstsplitfilesrc.c

2012-06-14 10:43:56 +0200 Wim Taymans <[email protected]>

* gst/rtp/gstrtptheoradepay.c:
theoradepay: fix buffer memory
The memory was added to the input buffer instead of the output buffer.

2012-06-13 13:36:45 +0200 Wim Taymans <[email protected]>

* gst/rtsp/gstrtspsrc.c:
rtspsrc: Don't reset time in flush-stop
Don't reset the time in flush-stop. Live sources can do this flush in the
playing state and so the pipeline will never have a chance to update the
base_time of the elements, which only happens when going from paused to
playing.

2012-06-12 12:42:31 +0200 Josep Torra <[email protected]>

* sys/osxaudio/Makefile.am:
* sys/osxaudio/gstosxaudiosink.c:
* sys/osxaudio/gstosxaudiosink.h:
* sys/osxaudio/gstosxcoreaudio.h:
* sys/osxaudio/gstosxringbuffer.c:
* sys/osxaudio/gstosxringbuffer.h:
osxaudiosink: Add support for SPDIF output
A big refactoring to allow passthrough AC3/DTS over SPDIF.
Several random cleanups and minor fixes.

2011-09-01 15:41:26 +0100 Vincent Penquerc'h <[email protected]>

* gst/deinterlace/gstdeinterlace.c:
* gst/deinterlace/gstdeinterlace.h:
deinterlace: send QoS messages when dropping a frame
https://bugzilla.gnome.org/show_bug.cgi?id=657941

2012-06-12 16:05:40 +0200 Wim Taymans <[email protected]>

* gst/rtsp/gstrtspsrc.c:
* gst/rtsp/gstrtspsrc.h:
rtspsrc: Rework the async state handling
Always send the flushing events to the udp elements now that basesrc supports
this. This makes sure a segment event is sent correctly after a flush.
Keep track of the currently executing command and make it possible to specify
what command you want to cancel when starting a new async command.
See https://bugzilla.gnome.org/show_bug.cgi?id=677905

2012-06-11 18:24:20 +0200 Stefan Sauer <[email protected]>

* gst/equalizer/gstiirequalizer.c:
* gst/equalizer/gstiirequalizer10bands.c:
* gst/equalizer/gstiirequalizer3bands.c:
* gst/videomixer/videomixer2.c:
childproxy: update api use

2012-06-11 12:54:27 +0200 Mark Nauwelaerts <[email protected]>

* gst/matroska/matroska-demux.c:
matroskademux: always perform full seek if seek is flushing
Fixes https://bugzilla.gnome.org/show_bug.cgi?id=677838

2012-06-11 11:20:18 +0100 Tim-Philipp Müller <[email protected]>

* gst/debugutils/rndbuffersize.c:
rndbuffersize: printf format fix for long -> int change

2012-06-08 20:38:34 +0200 Hans de Goede <[email protected]>

* sys/v4l2/gstv4l2object.c:
v4l2object: Don't probe UVC devices for being interlaced
UVC devices are never interlaced, and doing VIDIOC_TRY_FMT on them
causes expensive and slow USB IO, so don't probe them for interlaced.
This shaves 2 seconds of the startup time of cheese with a Logitech
Webcam Pro 9000.
Signed-off-by: Hans de Goede <[email protected]>
Fixes https://bugzilla.gnome.org/show_bug.cgi?id=677722

2012-06-09 16:53:54 +0100 Tim-Philipp Müller <[email protected]>

* gst/debugutils/rndbuffersize.c:
debug: change rndbuffersize properties from long to int
These should all be int instead of long, to avoid bugs
when passing these as varargs with g_object_set(), and
there was no reason to use long in the first place here.
Fixes FIXME.

2012-06-08 15:54:42 +0200 Sebastian Dröge <[email protected]>

* gst/avi/gstavidemux.c:
* gst/goom/gstgoom.c:
* gst/goom2k1/gstgoom.c:
* gst/monoscope/gstmonoscope.c:
* gst/rtsp/gstrtpdec.c:
elements: Use gst_pad_set_caps() instead of manual event fiddling

2012-06-08 15:04:59 +0200 Edward Hervey <[email protected]>

* common:
Automatic update of common submodule
From 03a0e57 to 98e386f

2012-06-08 10:11:12 +0200 Wim Taymans <[email protected]>

* ext/flac/gstflacenc.c:
* ext/wavpack/gstwavpackenc.c:
* gst/audioparsers/gstwavpackparse.c:
* sys/oss4/oss4-audio.c:
* tests/check/elements/interleave.c:
update for audio api change

2012-06-07 16:12:34 +0200 Sebastian Dröge <[email protected]>

* configure.ac:
Back to development