2014-04-18 11:11:14 +0200 Wim Taymans <
[email protected]>
* gst/rtpmanager/gstrtpjitterbuffer.c:
* gst/rtpmanager/rtpjitterbuffer.c:
* gst/rtpmanager/rtpjitterbuffer.h:
jitterbuffer: improve EOS handling
Make a new method to disable the jitterbuffer buffering.
Rework the update_estimated_eos() method. Calculate how much time
there is left to play. If we have less than the delay of the
jitterbuffer, we disabled buffering because we might never be able to
fill the complete jitterbuffer again.
If we receive an EOS event, disable buffering. We will drain the
buffer and eventually push the EOS event out.
When we reach the estimated NPT timeout and we didn't receive an EOS
event, make one and queue it so that it can be pushed.
Fixes
https://bugzilla.gnome.org/show_bug.cgi?id=728017
Conflicts:
gst/rtpmanager/gstrtpjitterbuffer.c
=== release 1.2.4 ===
2014-04-18 12:49:26 +0200 Sebastian Dröge <
[email protected]>
* ChangeLog:
* NEWS:
* RELEASE:
* 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-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-dtmf.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-vpx.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:
* gst-plugins-good.doap:
* win32/common/config.h:
Release 1.2.4
2014-04-18 12:49:13 +0200 Sebastian Dröge <
[email protected]>
* po/af.po:
* po/az.po:
* po/bg.po:
* po/ca.po:
* po/cs.po:
* po/da.po:
* po/de.po:
* po/el.po:
* po/en_GB.po:
* po/eo.po:
* po/es.po:
* po/eu.po:
* po/fi.po:
* po/fr.po:
* po/gl.po:
* po/hr.po:
* po/hu.po:
* po/id.po:
* po/it.po:
* po/ja.po:
* po/lt.po:
* po/lv.po:
* po/mt.po:
* po/nb.po:
* po/nl.po:
* po/or.po:
* po/pl.po:
* po/pt_BR.po:
* po/ro.po:
* po/ru.po:
* po/sk.po:
* po/sl.po:
* po/sq.po:
* po/sr.po:
* po/sv.po:
* po/tr.po:
* po/uk.po:
* po/vi.po:
* po/zh_CN.po:
* po/zh_HK.po:
* po/zh_TW.po:
Update .po files
2014-04-18 04:23:26 +0200 Wim Taymans <
[email protected]>
* gst/rtpmanager/gstrtpjitterbuffer.c:
jitterbuffer: assume a full buffer when eos
Rework the logic to make buffering messages a little, make sure we
don't make the same message multiple times.
Consider the buffer full when EOS was received.
Fixes
https://bugzilla.gnome.org/show_bug.cgi?id=728017
Conflicts:
gst/rtpmanager/gstrtpjitterbuffer.c
2014-04-11 18:19:49 +0200 Josep Torra <
[email protected]>
* gst/rtp/gstrtph264depay.c:
rtph264depay: only guess AU boundaries when aren't indicated by marker
The marker bit isn't mandatory and we had in place code to guess AU
boundaries by detecting a new picture start. This guessing code
didn't work with interlaced content that has proper marker bits
to indicate the AU boundaries. It was leaking the first field buffer
and producing a corrupted output.
fixes:
https://bugzilla.gnome.org/show_bug.cgi?id=728041
2014-04-06 18:03:11 -0300 Reynaldo H. Verdejo Pinochet <
[email protected]>
* tests/check/elements/souphttpsrc.c:
tests: souphttpsrc: use SoupKnownStatusCode if needed
From libsoup docs:
Prior to 2.44 SoupStatus was called SoupKnownStatusCode,
but the individual values have always had the names they
have now.
Fixes:
https://bugzilla.gnome.org/show_bug.cgi?id=727329
2013-12-02 15:26:50 -0500 Nicolas Dufresne <
[email protected]>
* sys/v4l2/gstv4l2object.c:
v4l2object: Don't validate dimension for encoded format
We set the dimensions just in case but don't validate them
afterwards. For some codecs the dimensions are *not* in the
bitstream, IIRC VC1 in ASF mode for example.
https://bugzilla.gnome.org/show_bug.cgi?id=720568
Conflicts:
sys/v4l2/gstv4l2object.c
2014-03-29 19:13:06 -0400 Nicolas Dufresne <
[email protected]>
* sys/v4l2/gstv4l2object.c:
* sys/v4l2/gstv4l2src.c:
v4l2: Fix support for caps without width, height, framerate or format
For format like mpegts, width and height is rarely in the negotiated caps. This
patch fixes failure when setting format, and prevent introducing width, height,
framerate and format to the caps when fixating.
https://bugzilla.gnome.org/show_bug.cgi?id=725860
Conflicts:
sys/v4l2/gstv4l2object.c
2014-03-06 19:52:36 +0000 William Manley <
[email protected]>
* sys/v4l2/gstv4l2object.c:
v4l2src: Fix support for mpegts streams
It seems that GStreamer's mpegts elements (tsdemux, tsparse) require caps
`video/mpegts,systemstream=true`. As far as I can see the significance
of systemstream is to indicate that this is a container format rather than
an elementary stream. As this is the case (and I can't understand how it
could not be the case with mpegts) I add systemstream=true to v4l2src's
caps.
This allows v4l2src to be linked with tsdemux for playback from my
Hauppauge HD-PVR with the pipeline:
v4l2src ! queue ! tsdemux ! video/x-h264 ! decodebin ! xvimagesink
In combination with the next commit this fixes using Hauppauge HD-PVR with
GStreamer 1.0+.
2014-03-20 15:28:26 +0100 Ognyan Tonchev <
[email protected]>
* gst/rtp/gstrtpjpegpay.c:
jpegpay: consider header len when calculating payload len
Fixed
https://bugzilla.gnome.org/show_bug.cgi?id=726777
2014-03-06 12:06:43 -0500 Olivier Crête <
[email protected]>
* gst/rtp/gstrtpamrdepay.c:
* gst/rtp/gstrtpilbcdepay.c:
* gst/rtp/gstrtpsirendepay.c:
* gst/rtp/gstrtpspeexdepay.c:
rtp: Remove caps restrictions from RTP depayloader sink caps
Remove caps restrictions that correspond to the default and are not
required in SDP. With the new usage of having pads require a subset
of the caps, they will make the negotiation fail.
2014-03-06 11:02:09 -0500 Olivier Crête <
[email protected]>
* gst/rtp/gstrtpspeexdepay.c:
rtpspeexdepay: Remove caps restrictions for depayloader
The "encoding-params" is optional in the SDP, because we now require
a subset of the caps, it would fail caps negotiatioin if it wasn't present.
So removed it from the template caps.
2014-03-05 14:26:02 +0100 Wim Taymans <
[email protected]>
* gst/rtpmanager/rtpsession.c:
session: handle unknown SSRC in FIR
https://bugzilla.gnome.org/show_bug.cgi?id=725712
2014-03-01 01:14:35 -0300 Thiago Santos <
[email protected]>
* gst/isomp4/qtdemux.c:
qtdemux: prevent segfault after seeking with negative rate
After a seek, the position and segments info of qtdemux are
reset to null or invalid values. For reverse playback, qtdemux
will seek to a keyframe and play until the next keyframe, then
seek to another previous keyframe. This seeking function requires
the position and segments info of qtdemux to be properly initialized.
Due to a missing check for segment.rate in the movie demuxing loop
it would consider the current segment to have ended without having
it initialized, leading qtdemux to attempt to do a previous keyframe
seek without proper position and segment info data. This would cause
a segfault. This patch fixes it by correctly checking for segment
boundaries taking the segment.rate into consideration.
https://bugzilla.gnome.org/show_bug.cgi?id=725104
2014-02-27 18:55:04 -0300 Thiago Santos <
[email protected]>
* ext/jpeg/gstjpegdec.c:
jpegdec: mark all parsed frames as sync points
all jpeg frames are sync points, so mark them as such so
reverse playback can properly work with the video decoder
base class
https://bugzilla.gnome.org/show_bug.cgi?id=725104
2014-02-25 09:00:45 +0100 Santiago Carot-Nemesio <
[email protected]>
* gst/rtsp/gstrtspsrc.c:
rtspsrc: Fix deadlock when task creation is no successful
https://bugzilla.gnome.org/show_bug.cgi?id=725124
2014-02-22 21:31:21 +0100 Mark Nauwelaerts <
[email protected]>
* sys/v4l2/v4l2_calls.c:
v4l2src: handle old and odd driver behaviour when listing controls
2013-11-28 16:54:58 -0800 Darryl Gamroth <
[email protected]>
* gst/audiofx/audiofxbaseiirfilter.c:
audiofxbaseiirfilter: check if coefficients are provided inside filter lock
https://bugzilla.gnome.org/show_bug.cgi?id=719524
2014-02-19 13:35:59 -0300 Reynaldo H. Verdejo Pinochet <
[email protected]>
* gst/audioparsers/gstaacparse.c:
aacparse: make sure we have enough ADTS data
We need at least 6 bytes to pass over to _get_frame_len()
but we were just checking for a minimum of 2 bytes for the
syncword.
https://bugzilla.gnome.org/show_bug.cgi?id=724638
2014-02-19 15:19:30 +0100 Branislav Katreniak <
[email protected]>
* ext/soup/gstsouphttpsrc.c:
souphttpsrc: do not emit error when connection with unknown size ends
Commit 46fd12ae5ec53200b16dfd7f17048d6bc60fbfbc introduced connection
recovery. But when server does not specify content-size,
souphttpsrc tries to reconnect even after regular end of stream.
Http server replies with SOUP_STATUS_REQUESTED_RANGE_NOT_SATISFIABLE
but souphttpsrc still emits error instead of EOS.
https://bugzilla.gnome.org/show_bug.cgi?id=724717
Signed-off-by: Branislav Katreniak <
[email protected]>
2014-02-18 22:54:45 +0100 Stefan Sauer <
[email protected]>
* gst/audiofx/audiofirfilter.c:
* gst/audiofx/audioiirfilter.c:
* gst/level/gstlevel.c:
* gst/spectrum/gstspectrum.c:
docs: use docbook markup for xi:include
It turns out that the change in gtk-doc-1.20 which wraps the |[]| content in
CDATA break xi:inlcude examples. As in a whole jhbuild checkout these where
the only 4, we're fixing them instead.
2014-02-18 11:28:18 +0100 Stefan Sauer <
[email protected]>
* ext/pulse/pulsesink.c:
pulsesink: fix crash when getting the current-device in NULL->READY
The "goto unlock" is wrong as in this code path we haven't take the lock yet.
Fixes #724619
2014-02-14 15:27:20 -0500 William Jon McCann <
[email protected]>
* gst/audiofx/audiocheblimit.c:
* gst/udp/gstudpsrc.c:
docs: fix mismatched para tags
newer gtkdoc is more sensitive to mismatched docbook tags.
This fixes the build in master.
2014-02-12 12:39:10 +0100 Sebastian Dröge <
[email protected]>
* ext/soup/gstsouphttpsrc.c:
souphttpsrc: Retry connection if we're finished before the content size only if we actually have a content size
https://bugzilla.gnome.org/show_bug.cgi?id=722185
2014-02-11 13:25:46 +0100 Sebastian Dröge <
[email protected]>
* ext/soup/gstsouphttpsrc.c:
souphttpsrc: Add mapping for NOT_FOUND and NOT_AUTHORIZED errors
2014-02-11 13:25:22 +0100 Sebastian Dröge <
[email protected]>
* ext/soup/gstsouphttpsrc.c:
souphttpsrc: Don't duplicate status_code to GStreamer error mapping