ChangeLog
=========
2014-11-06 Sebastian Dröge <sl...@coaxion.net>
* configure.ac:
releasing 1.4.4
2014-11-01 12:18:02 +0100 Aurélien Zanelli <aurelien.zane...@darkosphere.fr>
* ext/vpx/gstvp8utils.h:
vpx: remove compatibility defines
We are guaranteed to have VPX_IMG_FMT_I420, VPX_PLANE_Y,
VPX_PLANE_U and VPX_PLANE_V as we require libvpx > 1.1.0.
https://bugzilla.gnome.org/show_bug.cgi?id=739476
2014-11-01 11:59:26 +0000 Tim-Philipp Müller <t...@centricular.com>
* gst/rtp/gstrtpmp2tpay.c:
rtpmp2tpay: fix up template caps so we can output the default pt 33
Add fixed payload type for mp2t to template caps as well, so
our output caps match the advertised default pt. Fixes a
regression from 1.2.
There's still something wrong with caps negotiation though,
rtpmp2tpay payload=96 ! fakesink will not output caps with
payload=96.
2014-10-27 11:08:20 +0100 Sebastian Dröge <sebast...@centricular.com>
* tests/check/elements/aacparse.c:
aacparse: Fix unit test now that we always have profile/level in the
caps
2014-10-26 11:47:25 +0100 Sebastian Dröge <sebast...@centricular.com>
* gst/audioparsers/gstaacparse.c:
aacparse: Always set profile/level on the caps
We have the information already, so why not use it?
2014-10-30 15:37:36 -0700 Aleix Conchillo Flaqué <al...@oblong.com>
* gst/rtsp/gstrtspsrc.c:
rtspsrc: mikey related memory leaks
https://bugzilla.gnome.org/show_bug.cgi?id=739430
2014-10-28 21:32:06 +0000 Tim-Philipp Müller <t...@centricular.com>
* ext/pulse/pulsedeviceprovider.h:
* sys/v4l2/gstv4l2deviceprovider.h:
* sys/v4l2/gstv4l2tuner.h:
pulse, v4l2: add missing G_END_DECLS in some places
2014-10-22 22:50:54 +0530 Arun Raghavan <a...@accosted.net>
* ext/pulse/pulsesink.c:
pulsesink: Temporarily disable stream status posting
We need a mechanism in PulseAudio to allow running code outside the
mainloop lock. Then we'd be able to post to the bus (taking the
GST_OBJECT_LOCK), without worrying about locking order with the
mainloop
lock, which is the current cause of deadlocks while trying to post the
stream status messages.
https://bugzilla.gnome.org/show_bug.cgi?id=736071
2014-10-07 15:29:33 +0200 Aurélien Zanelli <aurelien.zane...@parrot.com>
* sys/v4l2/gstv4l2bufferpool.c:
v4l2bufferpool: cleanly handle streamon failure for output device
On streamon failure, the queued buffer is not released from the
bufferpool class point of view because it is queued to the driver and
the flush logic is not performed since we are not in streaming state.
It causes the v4l2 bufferpool to always return that stop method failed
and to leak v4l2 objects and buffers.
This commit solve this by performing the flush logic in error case, ie
flushing the allocator and restoring queued buffer state to
non-queued.
https://bugzilla.gnome.org/show_bug.cgi?id=738102
2014-10-08 10:31:21 +0200 Aurélien Zanelli <aurelien.zane...@parrot.com>
* sys/v4l2/gstv4l2bufferpool.c:
v4l2bufferpool: implement dispose method
Unref objects in dispose method rather than in finalize in order to
prevent circular reference.
https://bugzilla.gnome.org/show_bug.cgi?id=738102
2014-10-08 10:35:14 +0200 Aurélien Zanelli <aurelien.zane...@parrot.com>
* sys/v4l2/gstv4l2bufferpool.c:
v4l2bufferpool: check that allocator is non null when stopping pool
Otherwise, we could dereference NULL allocator when the stop method is
called by the GstBufferPool's finalize method.
https://bugzilla.gnome.org/show_bug.cgi?id=738102
2014-10-09 12:15:05 -0400 Nicolas Dufresne <nicolas.dufre...@collabora.com>
* sys/v4l2/gstv4l2sink.c:
v4l2sink: Implement unlock/unlock_stop
This will prevent deadlocks, but will also properly flush the pool
and allocator
when going to READY state. It should also fix issues reported on
mailing list
when seeking is performed.
https://bugzilla.gnome.org/show_bug.cgi?id=738152
2014-10-25 12:36:02 +0100 Tim-Philipp Müller <t...@centricular.com>
* gst/rtpmanager/gstrtpjitterbuffer.c:
rtpjitterbuffer: fix crash on some 32-bit systems
Make sure to pass right number of bits to gst_structure_new()
which is a vararg function.
Fixes elements/rtpaux unit test on ppc32.
2014-10-24 23:48:30 +0100 Tim-Philipp Müller <t...@centricular.com>
* gst/interleave/interleave.c:
interleave: intersect result with filter caps in caps query
Fixes crash in audiotestsrc because of an unsupported format
getting negotiated on big-endian systems with
audiotestsrc ! interleave ! audioconvert ! wavenc
2014-10-22 15:28:44 +0200 Ananda <ana...@latelier23.com>
* ext/speex/gstspeexdec.c:
* ext/speex/gstspeexenc.c:
speex: Fix segfault when resetting the codecs multiple times
https://bugzilla.gnome.org/show_bug.cgi?id=738793
2014-10-21 13:10:24 +0200 Wim Taymans <wtaym...@redhat.com>
* gst/rtpmanager/gstrtpjitterbuffer.c:
rtpjitterbuffer: make debug line less confusing
2014-10-03 17:28:06 -0700 Aleix Conchillo Flaqué <al...@oblong.com>
* gst/rtsp/gstrtspsrc.c:
rtspsrc: set full stream caps on internal src TCP pads
Set the complete stream caps on the TCP internal src pads. Otherwise,
ptdemux will not properly detect the caps change.
https://bugzilla.gnome.org/show_bug.cgi?id=737868
2014-10-17 22:23:27 +0200 Sjoerd Simons <sjo...@luon.net>
* gst/rtpmanager/gstrtpmux.c:
* tests/check/elements/rtpmux.c:
rtpmux: Don't set PROXY_CAPS flag on the src pad
rtpmux behaves like a funnel in that it forwards whatever upstream is
sending buffers. So setting proxy caps doesn't make sense as the
upstream don't have to have compatible caps, thus resulting in an
empty
caps set as a result of a caps query. Instead set fixed caps just
as funnel does.
https://bugzilla.gnome.org/show_bug.cgi?id=738722
2014-10-20 11:57:38 +0530 Vineeth T M <vineeth...@samsung.com>
* gst/videobox/gstvideobox.c:
videobox: critical error when element properties set as max/min
left, right, top, bottom can be set from range of -2147483648 to
2147483647
when i launch the videobox element with that values, it gives a
critical error
(gst-check-1.0:29869): GStreamer-CRITICAL **:
gst_value_set_int_range_step: assertion 'start < end' failed
This happens because min cannot be equal to max.
https://bugzilla.gnome.org/show_bug.cgi?id=738838
2014-10-11 11:18:42 +1100 David Sansome <m...@davidsansome.com>
* gst/equalizer/gstiirequalizer.c:
equalizer: Don't call iirequalizer's transform_ip in passthrough mode
It tries to map the read-only buffer with GST_MAP_READWRITE and
crashes.
https://bugzilla.gnome.org/show_bug.cgi?id=737886
2014-10-02 14:26:08 +0530 Nirbheek Chauhan <nirbh...@centricular.com>
* ext/soup/gstsouphttpclientsink.c:
souphttpclientsink: Fix lifetime of stream headers and queued buffers
Stream headers are updated whenever ::set_caps is called, so we can't
assume
they'll be valid before the message body is written out. We *can*
assume that
for queued buffers, but SOUP_MEMORY_STATIC is still wrong for those.
Also, add some debug logging for stream header interactions.
https://bugzilla.gnome.org/show_bug.cgi?id=737771
2014-10-02 03:26:22 +0200 Matej Knopp <matej.kn...@gmail.com>
* gst/audioparsers/gstaacparse.c:
aacparse: fix memory leak when prepending ADTS headers
https://bugzilla.gnome.org/show_bug.cgi?id=737761
2014-10-02 10:10:11 +0300 Sebastian Dröge <sebast...@centricular.com>
* gst/wavenc/gstwavenc.c:
wavenc: Send CAPS event after the pad was activated
Otherwise the CAPS event will be dropped and we never configure any
caps at
all, leading to weird behaviour in many situations. Especially header
rewriting is not going to work if a capsfilter is after wavenc.
https://bugzilla.gnome.org/show_bug.cgi?id=737735
2014-10-01 23:12:30 +0530 Nirbheek Chauhan <nirbh...@centricular.com>
* ext/soup/gstsouphttpclientsink.c:
souphttpclientsink: Add some more useful debug logging
2014-10-01 23:05:03 +0530 Nirbheek Chauhan <nirbh...@centricular.com>
* ext/soup/gstsouphttpclientsink.c:
souphttpclientsink: Free queued buffers in ::reset
::render sets a new callback for writing out new buffers only if
there aren't
already buffers queued for writing with a previously-scheduled
callback.
However, if the previously-scheduled callback is interrupted by a
state change
(either manually or due to an error) and there are still buffers in
the queue,
restarting the pipeline will result in buffers being queued forever,
and no
callbacks will ever be scheduled, and no buffers will be written out.
https://bugzilla.gnome.org/show_bug.cgi?id=737739
2014-09-30 11:28:39 +0300 Sebastian Dröge <sebast...@centricular.com>
* ext/vpx/gstvp8enc.c:
vp8enc: finish() and drain() should return a GstFlowReturn
2014-09-30 11:35:12 +0300 Sebastian Dröge <sebast...@centricular.com>
* ext/vpx/gstvp8enc.c:
* ext/vpx/gstvp9enc.c:
vp8enc/vp9enc: Protect the encoder with a mutex in all situations
2014-09-30 11:31:43 +0300 Sebastian Dröge <sebast...@centricular.com>
* ext/vpx/gstvp9enc.c:
vp9enc: Allow caps renegotiation
https://bugzilla.gnome.org/show_bug.cgi?id=726329
2014-03-14 12:59:02 +0100 Jose Antonio Santos Cadenas <santoscade...@gmail.com>
* ext/vpx/gstvp8enc.c:
vp8enc: Allow caps renegotiation
https://bugzilla.gnome.org/show_bug.cgi?id=726329
2014-09-29 22:48:16 +0530 Arun Raghavan <a...@accosted.net>
* ext/pulse/pulsesink.c:
* ext/pulse/pulsesrc.c:
pulse: Add some documentation about threading and synchronisation
This gives a quick introduction to how the pulsesink/pulsesrc code
interacts with the pa_threaded_mainloop that we start up to
communicate
with the server.
2014-09-29 20:18:08 +0530 Arun Raghavan <a...@accosted.net>
* ext/pulse/pulsesink.c:
pulsesink: Make emitting stream status messages synchronous
The stream status messages are emitted in the PA mainloop thread,
which
means the mainloop lock is taken, followed by the Gst object lock (by
gst_element_post_message()). In all other locations, the order of
locking is reversed (this is unavoidable in a bunch of cases where the
object lock is taken by GstBaseSink or GstAudioBaseSink, and then we
get
control to take the mainloop lock).
The only way to guarantee that the defer callback for stream status
messages doesn't deadlock is to either stop posting those messages, or
make sure that the message emission is completed before we proceed to
any point that might take the object lock before the mainloop lock
(which is what we do after this patch).
https://bugzilla.gnome.org/show_bug.cgi?id=736071
2014-10-10 18:30:07 -0400 Olivier Crête <olivier.cr...@ocrete.ca>
* gst/rtpmanager/rtpsource.c:
* gst/rtpmanager/rtpsource.h:
rtpsource: Rename seqnum-base to seqnum-offset in caps
This was modified back in 1.0 in GstRtpBasePayload
2014-10-10 17:30:24 -0400 Olivier Crête <olivier.cr...@ocrete.ca>
* gst/rtpmanager/gstrtpmux.c:
* gst/rtpmanager/gstrtpmux.h:
* tests/check/elements/rtpmux.c:
rtpmux: clock-base and seqnum-base -> timestamp-offset and
seqnum-offset
These were renamed in GstRTPBasePayload in 1.0
2014-10-10 18:11:19 -0400 Olivier Crête <olivier.cr...@ocrete.ca>
* gst/dtmf/gstrtpdtmfsrc.c:
* tests/check/elements/dtmf.c:
rtpdtmfsrc: clock-base and seqnum-base -> timestamp-offset and
seqnum-offset
These were renamed in GstRTPBasePayload in 1.0
Download
========
https://download.gnome.org/sources/gst-plugins-good/1.4/gst-plugins-good-1.4.4.tar.xz
(2.89M)
sha256sum: 2df90e99da45211c7b2525ae4ac34830a9e7784bd48c072c406c0cf014bdb277
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