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2016-07-06 Sebastian Dröge <slomo@coaxion.net>
2016-07-06 11:22:53 +0300 Steven Hoving <sh@bigbrother.nl>
* gst/rtsp/gstrtspsrc.c:
rtspsrc: Fix error messages to first convert to doubles before division
2016-07-06 10:18:30 +0300 Sebastian Dröge <sebastian@centricular.com>
* po/da.po:
* po/hr.po:
* po/pt_BR.po:
* po/sk.po:
po: Update translations
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2016-07-05 21:11:35 +0300 Sebastian Dröge <sebastian@centricular.com>
* gst/rtsp/gstrtspsrc.c:
rtspsrc: Set to PLAYING after a seek again after setting up the segment and everything else
There's a small window for a race condition otherwise.
2016-07-04 17:45:40 +0200 Sebastian Dröge <sebastian@centricular.com>
* tests/check/elements/qtmux.c:
qtmux: Use complete AAC caps with codec_data in the tests
2016-07-04 16:58:38 +0200 Sebastian Dröge <sebastian@centricular.com>
* gst/audioparsers/gstaacparse.c:
aacparse: Reject raw AAC if no codec_data is found in the caps
If necessary, a demuxer will have to invent something here but this is only a
problem with non-conformant files anyway.
2016-07-04 16:55:32 +0200 Sebastian Dröge <sebastian@centricular.com>
* gst/isomp4/qtdemux.c:
qtdemux: Invent AAC codec_data if none is present
Without, raw AAC can't be handled and we have some information available in
the decoder that most likely allows us to decode the stream in one way or
another. This is the same code already used by matroskademux for the same
reasons, and ffmpeg/vlc play such files just fine too by guesswork.
2016-07-04 14:54:13 +0200 Sebastian Dröge <sebastian@centricular.com>
* gst/isomp4/gstqtmux.c:
qtmux: Reject raw AAC caps without codec_data
The resulting file is not going to be playable without guesswork and raw caps
should always have codec_data.
2016-05-10 15:48:49 +0200 Edward Hervey <edward@centricular.com>
qtdemux: Handle upstream GAP in push-mode/time segment
This is to handle cases where upstream handles the fragmented streaming in TIME
segments and sends us data with gaps within fragments. This would happen when dealing
with trick-modes.
When upstream (push-based, TIME SEGMENT) wishes to send discontinuous samples,
it must obey the following rules:
* The buffer containing the [moof] must have a valid GST_BUFFER_OFFSET
* The buffers containing the first sample after a gap:
* MUST start at the beginning of a sample,
* MUST have the DISCONT flag set,
* MUST have a valid GST_BUFFER_OFFSET relative to the beginning of the fragment.
https://bugzilla.gnome.org/show_bug.cgi?id=767354
2016-07-01 11:54:57 +0100 Tim-Philipp Müller <tim@centricular.com>
* sys/v4l2/v4l2-utils.c:
v4l2: fix potential double-free of error debug string
gst_v4l2_clear_error() doesn't work like g_clear_error(), it
doesn't NULLify the pointer, so set freed debug string to NULL
so it doesn't get freed again if gst_v4l2_clear_error() is
called twice on the error.
CID 1362901
2016-07-01 10:05:00 +0000 Brad Lackey <blackey@gmail.com>
* gst/rtsp/gstrtspsrc.c:
rtspsrc: Don't disable UDP protocols on redirecting
https://bugzilla.gnome.org/show_bug.cgi?id=768232
2016-07-01 17:28:17 +0900 Seungha Yang <sh.yang@lge.com>
* gst/isomp4/qtdemux.c:
qtdemux: Push caps only when it was updated
Commit 7873bede3134b15e5066e8d14e54d1f5054d2063 caused new caps
event per moof without consideration of duplication.
https://bugzilla.gnome.org/show_bug.cgi?id=768268
2016-06-30 15:01:46 +0200 Jonas Holmberg <jonashg@axis.com>
* gst/rtp/gstrtph265depay.c:
rtph265depay: fix invalid memory access
10 bytes was allocated for stream_format but size of "byte-stream" is
more. Use g_strdup() instead.
https://bugzilla.gnome.org/show_bug.cgi?id=753760
2016-06-29 23:31:20 +0200 Sebastian Dröge <sebastian@centricular.com>
* ext/shout2/gstshout2.c:
shout2: Use a non-timer GstPoll
Otherwise set_flushing() will have undefined semantics and nowadays causes a
g_critical() to warn about that.
2016-06-19 02:08:25 -0300 Thiago Santos <thiagossantos@gmail.com>
* ext/soup/gstsouphttpsrc.c:
* ext/soup/gstsouphttpsrc.h:
souphttpsrc: dynamically adjust blocksize
Update the blocksize depending on how much is obtained from a read
of the input stream. This avoids doing too many reads in small chunks
when larger amounts of data are available and also prevents using
a very large memory area to read a small chunk of data.
https://bugzilla.gnome.org/show_bug.cgi?id=767833
2016-06-28 16:44:50 +0300 Sebastian Dröge <sebastian@centricular.com>
* gst/udp/gstudpsrc.c:
udpsrc: Windows has no ipi_spec_dst in struct in_pktinfo
2016-06-28 15:15:14 +0300 Sebastian Dröge <sebastian@centricular.com>
* gst/udp/gstudpsrc.c:
udpsrc: #define __APPLE_USE_RFC_3542 to be able to use IPV6_PKTINFO on OSX/iOS
2016-06-28 15:08:04 +0300 Sebastian Dröge <sebastian@centricular.com>
* gst/udp/gstudpsrc.c:
udpsrc: Move #includes around to a) work around broken glibc header and b) Windows
2016-06-28 14:25:03 +0300 Sebastian Dröge <sebastian@centricular.com>
* gst/udp/gstudpsrc.c:
udpsrc: Fix compilation on Windows and *BSD/OSX
2016-06-23 20:21:59 +0300 Sebastian Dröge <sebastian@centricular.com>
* gst/udp/gstudpsrc.c:
udpsrc: Filter out multicast packets that are not for our multicast address
https://bugzilla.gnome.org/show_bug.cgi?id=767980
2016-06-28 10:57:27 +0300 Sebastian Dröge <sebastian@centricular.com>
* gst/rtsp/gstrtspsrc.c:
rtspsrc: When seeking, consider the current element state or pending state instead of the RTSP state
If we consider the RTSP state, what can happen is that it is PLAYING but the
element already asynchronously tried to PAUSE and it just did not happen yet.
We would then override this setting to PAUSED (while the element actually is
in PAUSED) and set the RTSP state to PLAYING again. This would then cause us
to produce packets while the sinks are all PAUSED, piling up thousands of
packets in the rtpjitterbuffer and other elements and finally failing.
2016-06-27 09:20:35 +0300 Sebastian Dröge <sebastian@centricular.com>
* gst/flv/gstflvdemux.c:
flvdemux: Add comment about H263/MPEG4P2 being non-standard for FLV
They are however supported by ffmpeg and apparently used out there.
https://bugzilla.gnome.org/show_bug.cgi?id=768006
2016-06-24 14:48:53 +0300 Vivia Nikolaidou <vivia@ahiru.eu>
* gst/flv/gstflvdemux.c:
flvdemux: Add support for H263 and MPEG4 part2
https://bugzilla.gnome.org/show_bug.cgi?id=768006
2016-06-21 17:10:56 -0400 Nicolas Dufresne <nicolas.dufresne@collabora.com>
* docs/plugins/Makefile.am:
* docs/plugins/gst-plugins-good-plugins-docs.sgml:
* docs/plugins/gst-plugins-good-plugins-sections.txt:
* docs/plugins/gst-plugins-good-plugins.args:
* docs/plugins/gst-plugins-good-plugins.hierarchy:
Update plugins doc
This is partly automated using "make update" in docs/plugins, but also
required manual merge. Additionally, missing plugins and elements have
been added.
2016-06-21 17:51:38 +0100 Tim-Philipp Müller <tim@centricular.com>
* tests/check/elements/splitmux.c:
tests: splitmux: skip tests if theora or ogg plugins are not available
https://bugzilla.gnome.org/show_bug.cgi?id=767861
2016-06-21 11:46:13 -0400 Nicolas Dufresne <nicolas.dufresne@collabora.com>
* common:
Automatic update of common submodule
From ac2f647 to f363b32
2016-06-21 07:40:42 -0400 Aaron Boxer <boxerab@gmail.com>
* gst/rtp/gstrtpj2kpay.c:
gstrtpj2kpay: use tile bit and tile number to determine if there are multiple tiles in packet
Now we don't have to rely on a special value for the tile number.
https://bugzilla.gnome.org/show_bug.cgi?id=767817
2016-06-21 09:34:56 +0100 Tim-Philipp Müller <tim@centricular.com>
* gst/rtp/gstrtpj2kpay.c:
rtpj2kpay: fix compiler warning on OS/X
gstrtpj2kpay.c:364:21: error: implicit truncation from 'int' to bitfield changes value from -1 to 65535
https://bugzilla.gnome.org/show_bug.cgi?id=767817
2016-06-21 09:34:37 +0100 Tim-Philipp Müller <tim@centricular.com>
* docs/plugins/gst-plugins-good-plugins.hierarchy:
* docs/plugins/gst-plugins-good-plugins.interfaces:
* docs/plugins/gst-plugins-good-plugins.prerequisites:
* docs/plugins/inspect/plugin-avi.xml:
* docs/plugins/inspect/plugin-deinterlace.xml:
* docs/plugins/inspect/plugin-rtp.xml:
docs: update
2016-05-16 17:31:58 +0200 Guillaume Desmottes <guillaume.desmottes@collabora.co.uk>
* tests/check/elements/capssetter.c:
* tests/check/elements/icydemux.c:
* tests/check/elements/jpegenc.c:
* tests/check/elements/level.c:
* tests/check/elements/multifile.c:
* tests/check/elements/qtmux.c:
* tests/check/elements/rtprtx.c:
* tests/check/elements/udpsrc.c:
fix buffer leaks in tests
Need to call gst_check_drop_buffers() to release the buffers exchanged
during the test.
https://bugzilla.gnome.org/show_bug.cgi?id=766561
2016-05-17 12:52:43 +0300 Guillaume Desmottes <guillaume.desmottes@collabora.co.uk>
* tests/check/elements/interleave.c:
interleave: fix message leaks in test
Flush the bus when cleaning up so pending messages are destroyed.
https://bugzilla.gnome.org/show_bug.cgi?id=766561
2016-05-17 12:58:06 +0300 Guillaume Desmottes <guillaume.desmottes@collabora.co.uk>
* tests/check/elements/videomixer.c:
videomixer: fix event leaks in test
https://bugzilla.gnome.org/show_bug.cgi?id=766561
2016-05-13 15:12:22 +0200 Guillaume Desmottes <guillaume.desmottes@collabora.co.uk>
* tests/check/elements/deinterleave.c:
deinterleave: fix leaks
- Flush the bus so messages aren't leaked
- Fix pad leak
https://bugzilla.gnome.org/show_bug.cgi?id=766561
2016-06-17 15:29:16 +0300 Sebastian Dröge <sebastian@centricular.com>
* gst/rtp/gstrtph264pay.c:
rtph264pay: Deprecated sprop-parameter-set property
This is supposed to be either in the codec_data (avc stream format) or inside
the stream, and we extract it from there. It should not be set from a
property as it's stream specific.
https://bugzilla.gnome.org/show_bug.cgi?id=767789
2016-06-17 12:16:32 -0700 Aleix Conchillo Flaqué <aleix@oblong.com>
* gst/rtsp/gstrtspsrc.c:
rtspsrc: make all srtp encoder properties explicit
The Session Data Protocol doesn't allow specifying a cipher for the
SRTCP, so it will use the SRTP one. In the "srtpenc" element the cipher
"aes-128-icm" is the default for SRTP and SRTCP, but if we want to have
an SRTCP with the "aes-256-icm" cipher then we also need to set the SRTP
cipher to "aes-256-icm", otherwise "aes-128-icm" will be used instead.
https://bugzilla.gnome.org/show_bug.cgi?id=767799
2016-06-17 19:59:13 +0100 Tim-Philipp Müller <tim@centricular.com>
* ext/soup/gstsoup.c:
soup: work around frequent deadlocks in GLib type initialisation
.. by registering the types from the plugin init function. This
seems to help, but we'll see if it's enough (might need similar
things elsewhere).
https://bugzilla.gnome.org/show_bug.cgi?id=693911
https://bugzilla.gnome.org/show_bug.cgi?id=674885
2016-06-17 16:08:08 +0300 Sebastian Dröge <sebastian@centricular.com>
* gst/isomp4/gstqtmux.c:
qtmux: The prores variant is stored in the variant field, not format
And the caps in the sink pad template already used variant (only).
2016-06-17 13:00:48 +0200 Jonas Holmberg <jonashg@axis.com>
* gst/rtp/gstrtph265pay.c:
* gst/rtp/gstrtph265pay.h:
rtph265pay: Remove sprop-parameter-sets property
There is no valid use case when this property is needed since the values
must be in either codec_data or buffer data.
https://bugzilla.gnome.org/show_bug.cgi?id=753760
2016-06-10 16:17:26 +0200 Jonas Holmberg <jonashg@axis.com>
* docs/plugins/scanobj-build.stamp:
* gst/rtp/gstrtph265pay.c:
rtph265pay: Read NALU type the same way everywhere
Cosmetic change to read NALU type in gst_rtp_h265_pay_decode_nal() the
same way as in other places.
https://bugzilla.gnome.org/show_bug.cgi?id=753760
2016-06-17 13:58:33 +0200 Aurélien Zanelli <aurelien.zanelli@parrot.com>
* gst/rtpmanager/rtpjitterbuffer.h:
rtpjitterbuffer: fix RTPJitterBufferMode documentation
Documentation lacks '@' before each enum values and there was an extra
line after symbol section which confuses GTK-Doc parser.
https://bugzilla.gnome.org/show_bug.cgi?id=767788
2016-05-23 10:18:48 +0200 Miguel París Díaz <mparisdiaz@gmail.com>
* gst/rtpmanager/rtpsession.c:
rtpsession: take the lock when changing stats
https://bugzilla.gnome.org/show_bug.cgi?id=766025
2016-06-15 11:19:43 +0200 Jürgen Slowack <jurgen.slowack@barco.com>
* gst/rtp/gstrtph265pay.c:
rtph265: fix NAL unit type parsing and SPS/PPS/VPS detection
Fixes sps/pps/vps insertion via the config-interval property.
https://bugzilla.gnome.org//show_bug.cgi?id=767680
2016-06-11 12:16:03 +0300 Sebastian Dröge <sebastian@centricular.com>
* tests/check/pipelines/simple-launch-lines.c:
simple-launch-lines: Use correct JPEG2000 caps
2016-06-10 13:43:09 +0100 Tim-Philipp Müller <tim@centricular.com>
* gst/flv/gstflvdemux.c:
flvdemux: fix indentation
2016-06-10 13:42:01 +0100 Tim-Philipp Müller <tim@centricular.com>
* gst/flv/gstflvdemux.c:
flvdemux: fix date parsing when there are trailing spaces
Fixes parsing of "Thu May 11 15:57:46 2006 ".
https://bugzilla.gnome.org/show_bug.cgi?id=767496
2016-05-13 15:08:24 -0400 Aaron Boxer <boxerab@gmail.com>
* gst/rtp/gstrtpj2kcommon.h:
* gst/rtp/gstrtpj2kdepay.c:
* gst/rtp/gstrtpj2kpay.c:
gstrtpj2k: set sampling field required by RFC
This field is now required in the sink caps.
https://bugzilla.gnome.org/show_bug.cgi?id=766236
2016-06-09 09:30:48 +0900 Seungha Yang <sh.yang@lge.com>
* gst/flv/gstflvdemux.c:
flvdemux: Fix unref assertion failure
Fix unref assertion failure
https://bugzilla.gnome.org/show_bug.cgi?id=767424
2016-05-14 14:46:17 +0200 Olivier Crête <olivier.crete@collabora.com>
* gst/rtpmanager/gstrtpjitterbuffer.c:
rtpjitterbuffer: Work with non-TIME segments
With non-time segments, it now assumes that the arrival time of packets
is not relevant and that only the RTP timestamp matter and it produces
an output segment start at running time 0.
https://bugzilla.gnome.org/show_bug.cgi?id=766438
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2016-06-07 20:53:34 -0400 Nicolas Dufresne <nicolas.dufresne@collabora.com>
* ext/libpng/gstpngdec.c:
pngdec: Wait for segment event before checking it
The heuristic to choose between packetise or not was changed to use the
segment format. The problem is that this change is reading the segment
during the caps event handling. The segment event will only be sent
after. That prevented the decoder to go in packetize mode, and avoid
useless parsing.
https://bugzilla.gnome.org/show_bug.cgi?id=736252
2016-06-06 17:00:22 -0400 Nicolas Dufresne <nicolas.dufresne@collabora.com>
* ext/jpeg/gstjpegdec.c:
jpegdec: Wait for segment event before checking it
The heuristic to choose between packetise or not was change to use the
segment format. The problem is that this change is reading the segment
during the caps event handling. The segment event will only be sent
after. That prevented the decoder to go in packetize mode, and avoid
useless parsing.
https://bugzilla.gnome.org/show_bug.cgi?id=736252
2016-06-07 16:42:09 -0400 Nicolas Dufresne <nicolas.dufresne@collabora.com>
* sys/v4l2/gstv4l2videodec.c:
v4l2videodec: Keep part of the input buffer
Instead of completely getting rid of the input buffer, copy
the metadata, the flags and the timestamp into an empty buffer.
This way the decoder base class can copy that information again
to the output buffer.
https://bugzilla.gnome.org/show_bug.cgi?id=758424
2016-06-07 16:41:58 -0400 Nicolas Dufresne <nicolas.dufresne@collabora.com>
* sys/v4l2/gstv4l2videodec.c:
v4l2videodec: Coding style fixes
2016-06-07 16:09:23 -0400 Nicolas Dufresne <nicolas.dufresne@collabora.com>
* sys/v4l2/gstv4l2object.c:
v4l2object: Coding style fixes
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2016-06-07 16:04:52 -0400 Nicolas Dufresne <nicolas.dufresne@collabora.com>
* sys/v4l2/gstv4l2object.c:
* sys/v4l2/gstv4l2object.h:
* sys/v4l2/gstv4l2sink.c:
* sys/v4l2/gstv4l2src.c:
* sys/v4l2/gstv4l2transform.c:
* sys/v4l2/gstv4l2videodec.c:
v4l2: Add an error return to _try/_set_format
This way one can easily ignore errors. Previously, error were always
posted ont he bus.
https://bugzilla.gnome.org/show_bug.cgi?id=766172
2016-06-07 16:01:55 -0400 Nicolas Dufresne <nicolas.dufresne@collabora.com>
* sys/v4l2/v4l2-utils.c:
* sys/v4l2/v4l2-utils.h:
v4l2-util: Introduce GstV4l2Error
This is to allow returning an error that can easily be sent as
message to the application if the element needs it. Using this
also allow ignoring errors.
https://bugzilla.gnome.org/show_bug.cgi?id=766172
2016-06-07 12:41:19 -0400 Nicolas Dufresne <nicolas.dufresne@collabora.com>
* sys/v4l2/gstv4l2src.c:
v4l2src: Avoid decide allocation on active pool
v4l2src will renegotiate only if the format have changed. As of now,
it's not possible to change the allocationw without resetting the
camera. To avoid unwanted side effect, simply keep the old allocation
if no renegotiation is taking place. This fixes assertion and possible
failures in USERPTR or DMABUF import mode (when using downstream pools).
https://bugzilla.gnome.org/show_bug.cgi?id=754042
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2016-04-28 13:44:49 +0200 Edward Hervey <bilboed@bilboed.com>
* gst/isomp4/qtdemux.c:
* gst/isomp4/qtdemux.h:
qtdemux: Show state name in debugging
Makes it easier to trace what's going on
2016-05-10 15:45:42 +0200 Edward Hervey <bilboed@bilboed.com>
* gst/isomp4/qtdemux.c:
qtdemux: Remove useless variable
That variable is only needed for a debug statement, move it there
2016-05-10 15:10:36 +0200 Edward Hervey <bilboed@bilboed.com>
* gst/isomp4/qtdemux.c:
* gst/isomp4/qtdemux.h:
qtdemux: Add/Fix comments on the various structure variables
No variables were added/removed. This was just a good excuse to:
* Comment what most variables are used for (and when)
* Order them in such a way as to show first the common variables used
in all cases, followed by those only used in push-mode
2016-05-10 15:07:40 +0200 Edward Hervey <bilboed@bilboed.com>
* gst/isomp4/qtdemux.c:
qtdemux: Remove unused structure
Let's just remove it, been commented for 7+ years :)
2015-09-02 11:48:29 +0200 Philipp Zabel <p.zabel@pengutronix.de>
* sys/v4l2/gstv4l2videodec.c:
v4l2videodec: use decoder stop command instead of queueing empty buffers
Only if the decoder stop command fails, keep queueing empty buffers to
signal end of stream as before.
https://bugzilla.gnome.org/show_bug.cgi?id=733864
2014-12-12 14:31:36 +0100 Peter Seiderer <ps.report@gmx.net>
* sys/v4l2/gstv4l2videodec.c:
v4l2videodec: add gst_v4l2_decoder_cmd helper
https://bugzilla.gnome.org/show_bug.cgi?id=733864
2016-06-01 20:28:39 +0300 Sebastian Dröge <sebastian@centricular.com>
* gst/isomp4/qtdemux.c:
qtdemux: Forward segments directly if we are operating in PUSH mode on fragmented streams
We shouldn't go through segment activation as we will only have a limited
understanding of how the whole stream timeline looks like from the moof. We
only know about the current fragment, while upstream knows about the whole
stream.
This fixes seeking in DASH streams, both for seeks after the current moof and
for seeks into the current moof. The former would fail because the moof ends
and we can't activate any segment, the latter would cause a segment that stops
at the moof end, and no further fragments would be played because we end up
being EOS.
https://bugzilla.gnome.org/show_bug.cgi?id=767071
2016-06-06 17:54:10 -0400 Nicolas Dufresne <nicolas.dufresne@collabora.com>
* sys/v4l2/gstv4l2transform.c:
v4l2transform: Use looser caps for upstream
When we fixate for upstream, try to not introduce new fields when not
needed. This was imported from videoconvert element.
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2015-01-28 12:07:58 +0100 Enrico Jorns <ejo@pengutronix.de>
* sys/v4l2/gstv4l2transform.c:
gstv4l2transform: format fixation for preferring passthrough
* If outgoing format is unfixated, try to set it to input format.
* Call gst_caps_fixate () at end of fixation routine
https://bugzilla.gnome.org/show_bug.cgi?id=766719
2016-05-20 12:49:53 +0200 Philipp Zabel <p.zabel@pengutronix.de>
* sys/v4l2/gstv4l2transform.c:
v4l2transform: allow to change pixel aspect ratio
Scalers may change width and height independently,
allow to change pixel aspect ratio.
https://bugzilla.gnome.org/show_bug.cgi?id=766712
2016-05-20 12:32:25 +0200 Philipp Zabel <p.zabel@pengutronix.de>
* sys/v4l2/gstv4l2transform.c:
v4l2transform: fix scaling in case of fixed pixel aspect ratio
To change pixel aspect ratio from DAR to PAR, the necessary scaling factor
is DAR/PAR, not DAR*PAR.
For good measure, add debug output similar to the fixed-width and
fixed-height cases.
https://bugzilla.gnome.org/show_bug.cgi?id=766711
2016-05-13 16:39:25 +0200 Philipp Zabel <p.zabel@pengutronix.de>
* sys/v4l2/gstv4l2object.c:
v4l2object: fill colorimetry in gst_v4l2_object_acquire_format
Instead of relying on the default colorimetry chosen by
gst_video_info_set_format(), set info.colorimetry from the
values returned by G_FMT. This allows decoders to propagate
their input colorimetry downstream.
https://bugzilla.gnome.org/show_bug.cgi?id=766383
2016-05-18 10:17:12 +0200 Philipp Zabel <p.zabel@pengutronix.de>
* sys/v4l2/gstv4l2object.c:
v4l2object: refactor gst_v4l2_object_get_colorspace to take a v4l2_format parameter
Move the extraction of colorimetry parameters from struct v4l2_format and the
setting of the identity matrix for RGB formats into the function to avoid code
duplication.
https://bugzilla.gnome.org/show_bug.cgi?id=766383
2016-05-13 14:58:41 +0200 Philipp Zabel <p.zabel@pengutronix.de>
* sys/v4l2/gstv4l2videodec.c:
v4l2videodec: use visible size, not coded size, for downstream negotiation filter
gst_v4l2_probe_caps() returns the coded size, not the visible size. Subtract
the known padding from probed caps with the coded size before using them as
filter for caps negotiation with downstream elements.
https://bugzilla.gnome.org/show_bug.cgi?id=766382
2016-05-13 14:45:02 +0200 Philipp Zabel <p.zabel@pengutronix.de>
* sys/v4l2/gstv4l2object.c:
v4l2object: use G_SELECTION instead of G_CROP in gst_v4l2_object_acquire_format
The gst_v4l2_object_acquire_format() function is used by v4l2videodec to obtain
the currently set capture format. Since G_FMT returns the coded size, the
visible size needs to be obtained from the compose rectangle in order to
negotiate it with downstream elements. The G_CROP call hasn't worked on mem2mem
capture queues for a long time. Instead use the G_SELECTION call to obtain the
compose rectangle and only fall back to G_CROP for ancient kernels.
https://bugzilla.gnome.org/show_bug.cgi?id=766381
2016-01-27 09:57:38 +0100 Andreas Naumann <anaumann@ultratronik.de>
* sys/v4l2/gstv4l2sink.c:
v4l2sink: Use V4L2_BUF_TYPE_VIDEO_OUTPUT_OVERLAY if driver advertises it.
On modern kernels, the G/S_FMT ioctls will always fail using
V4L2_BUF_TYPE_VIDEO_OVERLAY with VFL_DIR_TX (e.g. real overlay out drivers)
since this is not the intented use (rather rx, according to v4l2 API doc).
Probably this is why the Video Output Overlay interface was created, so if
the driver advertises it we might as well use.
For old kernels (pre 2012) the old way might still work so keeping this for
compatibility.
https://bugzilla.gnome.org/show_bug.cgi?id=761165
2016-06-06 18:52:01 +0100 Kieran Bingham <kieran@bingham.xyz>
* sys/v4l2/gstv4l2object.c:
v4l2object: Use non-deprecated V4L2 type for RGB15
Support for the updated V4L2_PIX_FMT_XRGB555 was added in commit
2538fee2fd8fdb74b05f0a511281bc4707e7cc44 however, when setting the format
for use in v4l2 ioctls, the old deprecated format is still used. Convert
this to the new accepted format type, as the preferred format.
https://bugzilla.gnome.org/show_bug.cgi?id=767300
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2016-05-04 14:50:32 +0200 Michael Olbrich <m.olbrich@pengutronix.de>
* gst/matroska/matroska-demux.c:
matroskademux: preserve seek flags
Without this some flags get lost in streaming mode.
https://bugzilla.gnome.org/show_bug.cgi?id=767194
2016-06-06 10:47:52 +0300 Sebastian Dröge <sebastian@centricular.com>
* ext/soup/Makefile.am:
* ext/soup/gstsouphttpclientsink.c:
* ext/soup/gstsouphttpsrc.c:
* ext/soup/gstsouphttpsrc.h:
Revert "WIP revert soup"
This reverts commit fdac3a7a231f3848665636cf8122f96103b46e3b.
Was not supposed to be pushed but a local workaround for
https://bugzilla.gnome.org/show_bug.cgi?id=693911#c13
2016-06-03 13:09:35 +0200 Miguel París Díaz <mparisdiaz@gmail.com>
* gst/rtpmanager/rtpsource.c:
rtpsource: complete warn log with SSRC
https://bugzilla.gnome.org/show_bug.cgi?id=767195
2016-05-31 15:29:13 +0300 Sebastian Dröge <sebastian@centricular.com>
* ext/soup/Makefile.am:
* ext/soup/gstsouphttpclientsink.c:
* ext/soup/gstsouphttpsrc.c:
* ext/soup/gstsouphttpsrc.h:
WIP revert soup
2016-06-03 13:18:31 +0300 Sebastian Dröge <sebastian@centricular.com>
* ext/dv/gstdvdemux.c:
dvdemux: Unref seek event in any case
It would be leaked if no seek handler was currently set.
2016-06-03 10:49:17 +0300 Sebastian Dröge <sebastian@centricular.com>
* ext/dv/gstdvdemux.c:
* ext/dv/gstdvdemux.h:
dvdemux: Properly set event/message sequence numbers based on the previous seek
See https://bugzilla.gnome.org/show_bug.cgi?id=765935
https://bugzilla.gnome.org/show_bug.cgi?id=767157
2016-06-03 10:36:32 +0300 Sebastian Dröge <sebastian@centricular.com>
* ext/dv/gstdvdemux.c:
* ext/dv/gstdvdemux.h:
dvdemux: Remember if upstream had a time segment and if not properly create time segments
Previously the segment.time was wrong, and the position was not updated
correctly, resulting in seeks in PUSH mode with upstream providing a BYTES
segment to not work at all.
https://bugzilla.gnome.org/show_bug.cgi?id=767157
2016-06-03 09:54:53 +0300 Sebastian Dröge <sebastian@centricular.com>
* ext/dv/gstdvdemux.c:
dvdemux: Implement SEEKING query so we can actually seek if upstream can't seek in TIME
https://bugzilla.gnome.org/show_bug.cgi?id=767157
2016-06-02 14:19:15 +0300 Sebastian Dröge <sebastian@centricular.com>
* ext/dv/gstdvdemux.c:
dvdemux: Recalculate the frame offsets at the beginning of each BYTE segment and whenever upstream gives us a timestamp
This fixes seeking in DV streams where upstream operates in PUSH mode with a
TIME segment (e.g. avidemux). Without this, we would generate wrong durations
and timestamps after a seek.
https://bugzilla.gnome.org/show_bug.cgi?id=767157
2016-06-02 13:53:44 +0300 Sebastian Dröge <sebastian@centricular.com>
* ext/dv/gstdvdemux.c:
* ext/dv/gstdvdemux.h:
dvdemux: Pass-through buffer DISCONT flags
https://bugzilla.gnome.org/show_bug.cgi?id=767157
2016-06-02 16:16:45 -0400 Olivier Crête <olivier.crete@collabora.com>
* gst/rtp/gstrtpvp9depay.c:
rtpvp9depay: Don't assert on flexible mode packets
Instead just post a warning on the bus for now.
2016-06-02 15:03:17 +0200 Guillaume Desmottes <guillaume.desmottes@collabora.co.uk>
* tests/check/elements/rtpbin.c:
tests: rtpbin: fix caps leak
https://bugzilla.gnome.org/show_bug.cgi?id=767156
2016-06-02 15:00:01 +0200 Guillaume Desmottes <guillaume.desmottes@collabora.co.uk>
* tests/check/elements/amrparse.c:
tests: amrparse: clean up test
- use GST_CHECK_MAIN() to reduce boilerplate
- unref the input caps using a teardown function to prevent leaks
https://bugzilla.gnome.org/show_bug.cgi?id=767156
2016-05-20 15:22:35 +0200 Edward Hervey <edward@centricular.com>
* gst/deinterlace/gstdeinterlace.c:
* gst/deinterlace/gstdeinterlace.h:
deinterlace: Ensure DISCONT flag is properly propagated
The output of deinterlace at startup, or when receiving a new DISCONT
buffer, should have the DISCONT flag set on the first buffer.
2016-05-31 21:34:04 +0200 Josep Torra <adn770@gmail.com>
* sys/v4l2/gstv4l2bufferpool.c:
v4l2src: check for valid size on raw video buffers
Discard buffers that doesn't contain enough data when dealing
with raw video inputs.
https://bugzilla.gnome.org/show_bug.cgi?id=767086
2016-05-31 17:10:36 +0300 Sebastian Dröge <sebastian@centricular.com>
* gst/isomp4/qtdemux.c:
qtdemux: Use the demuxer segment instead of a new one for MSS streams
Upstream might have told us something about the to be expected segment, so
let's use that information instead of coming up with a [0,-1] segment.
https://bugzilla.gnome.org/show_bug.cgi?id=767071
2016-05-31 17:04:32 +0300 Sebastian Dröge <sebastian@centricular.com>
* gst/isomp4/qtdemux.c:
qtdemux: Only activate segments and send SEGMENT events if we have streams
But in that case also remove the pending newsegment event, otherwise we would
later send a possibly outdated event.
https://bugzilla.gnome.org/show_bug.cgi?id=767071
2016-05-31 16:53:50 +0300 Sebastian Dröge <sebastian@centricular.com>
qtdemux: In PULL mode, nothing is ever going to send us a SEGMENT event
https://bugzilla.gnome.org/show_bug.cgi?id=767071
2016-05-31 16:38:34 +0300 Sebastian Dröge <sebastian@centricular.com>
* gst/isomp4/qtdemux.c:
qtdemux: Don't override TIME segments from upstream that we just saw
The point of d8fb7a9c96b108814beeaa0e63f818d4648c7fe9 was to not have any
spurious segments stored for later if we do BYTES->TIME conversion, but
overriding any TIME segments from upstream does not make any sense.
See https://bugzilla.gnome.org/show_bug.cgi?id=763165
https://bugzilla.gnome.org/show_bug.cgi?id=767071
2015-07-16 09:48:46 +0530 Prashant Gotarne <ps.gotarne@samsung.com>
* gst/multifile/gstmultifilesrc.c:
multifilesrc: set position as offset from start-index
query position in GST_FORMAT_BUFFER returns
offset from start-index rather than index.
https://bugzilla.gnome.org/show_bug.cgi?id=752462
2016-05-27 12:49:32 +0100 Tim-Philipp Müller <tim@centricular.com>
* tests/check/pipelines/simple-launch-lines.c:
* tests/files/Makefile.am:
* tests/files/gradient.j2k:
tests: add unit test for JPEG-2000 rtp payloader leak
https://bugzilla.gnome.org/show_bug.cgi?id=766870
2016-05-25 17:11:13 +0200 Pierre Lamot <pierre.lamot@openwide.fr>
* gst/rtp/gstrtpj2kpay.c:
rtpj2kpay: Fix buffer memory leak
Input buffer memory was not unmapped
https://bugzilla.gnome.org/show_bug.cgi?id=766870
2016-05-18 12:12:15 +0300 Guillaume Desmottes <guillaume.desmottes@collabora.co.uk>
* sys/v4l2/gstv4l2object.c:
v4l2object: fix caps leak
gst_v4l2_object_probe_caps() was taking an extra ref on the returned
caps for no reason.
https://bugzilla.gnome.org/show_bug.cgi?id=766610
2016-05-22 20:14:18 +0100 Tim-Philipp Müller <tim@centricular.com>
* gst/videocrop/gstvideocrop.c:
videocrop mark crop properties as mutable in playing state
2016-05-20 16:47:35 +0300 Guillaume Desmottes <guillaume.desmottes@collabora.co.uk>
* ext/soup/gstsouphttpsrc.c:
souphttpsrc: fix buffer leak when flushing
When early returning in gst_soup_http_src_read_buffer() because the
element is FLUSHING, we need to unmap and unref the buffer which was just created.
https://bugzilla.gnome.org/show_bug.cgi?id=766718
2016-05-20 11:15:44 +0300 Sebastian Dröge <sebastian@centricular.com>
* gst/isomp4/qtdemux.c:
qtdemux: Set seek event seqnum on all SEGMENT events
Some were forgotten.
See https://bugzilla.gnome.org/show_bug.cgi?id=765935
2016-05-20 11:12:44 +0300 Sebastian Dröge <sebastian@centricular.com>
* gst/avi/gstavidemux.c:
* gst/avi/gstavidemux.h:
avidemux: Pass through seek event seqnums in all SEGMENT/EOS events and SEGMENT_DONE messages/events
See https://bugzilla.gnome.org/show_bug.cgi?id=765935
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2016-05-20 10:56:52 +0300 Sebastian Dröge <sebastian@centricular.com>
* gst/matroska/matroska-demux.c:
matroskademux: Set seek event seqnum in EOS and SEGMENT_DONE messages/events
Also actually store the seqnum in pull mode seeks.
See https://bugzilla.gnome.org/show_bug.cgi?id=765935
2016-05-17 13:40:38 +0300 Guillaume Desmottes <guillaume.desmottes@collabora.co.uk>
* gst/deinterlace/gstdeinterlace.c:
deinterlace: fix caps leak
The caps returned by gst_pad_get_current_caps() was never unreffed when
not early returning.
Fix a leak with the elements/deinterlace test.
https://bugzilla.gnome.org/show_bug.cgi?id=766558
2016-01-25 16:25:51 +0100 Mikhail Fludkov <misha@pexip.com>
* gst/rtpmanager/rtpsession.c:
* tests/check/Makefile.am:
* tests/check/elements/rtpsession.c:
rtpsession: don't act on suspicious BYE RTCP
Some endpoints (like Tandberg E20) can send BYE packet containing our
internal SSRC. I this case we would detect SSRC collision and get rid
of the source at some point. But because we are still sending packets
with that SSRC the source will be recreated immediately.
This brand new internal source will not have some variables incorrectly
set in its state. For example 'seqnum-base` and `clock-rate` values will be
-1.
The fix is not to act on BYE RTCP if it contains internal or unknown
SSRC.
https://bugzilla.gnome.org/show_bug.cgi?id=762219
2015-11-15 14:54:28 +0100 Mikhail Fludkov <misha@pexip.com>
* tests/check/elements/rtpsession.c:
rtpsession: Add test for locking of the stats signal
Keeping the lock while emitting the stats signal introduces potential
deadlock in those situations when the signal callback wants the access
to rtpsession's properties which also requre the lock.
https://bugzilla.gnome.org/show_bug.cgi?id=762216
2016-05-19 15:36:57 +0900 Seungha Yang <sh.yang@lge.com>
* gst/matroska/matroska-demux.c:
matroskademux: don't hold object lock whilst pushing out headers
matroskademux would take the GST_OBJECT_LOCK in
- gst_matroska_demux_push_codec_data_all()
- gst_matroska_demux_query()
Some parse element such as FLAC checks upstream seekability, and
there is some use cases that matroska-demux is linked to a parse element
(e.g.,FLAC format) without intermediate elements (e.g., queue).
In this case, matroska-demux never returns from _push_codec_data_all()
because the parser can return only after it receives the response to
the upstream query, but that's not going to happen because it's
deadlocked.
Elements must not hold the object lock whilst pushing out events
or data.
https://bugzilla.gnome.org/show_bug.cgi?id=766645
2016-05-19 12:43:01 +0300 Sebastian Dröge <sebastian@centricular.com>
* ext/soup/gstsouphttpclientsink.c:
souphttpclientsink: Set sent_buffers and streamheader_buffers to NULL after freeing
Otherwise we might use an already freed list later and crash or worse.
2016-05-18 18:32:57 +0100 Tim-Philipp Müller <tim@centricular.com>
* gst/udp/gstudpsrc.c:
udpsrc: fix Since version for new "loop" property
2016-05-16 16:18:37 +0200 Guillaume Desmottes <guillaume.desmottes@collabora.co.uk>
* gst/rtsp/gstrtpdec.c:
rtpdec: fix clock leak
gst_system_clock_obtain() returns a new ref.
https://bugzilla.gnome.org/show_bug.cgi?id=766521
2016-05-17 05:33:35 +0100 Tim-Philipp Müller <tim@centricular.com>
* gst/udp/gstudpsrc.c:
udpsrc: add doc blurb with since marker for new "loop" property
2015-11-13 15:52:35 +0100 Dimitrios Katsaros <patcherwork@gmail.com>
* gst/avi/gstavimux.c:
avimux: add support for png
https://bugzilla.gnome.org/show_bug.cgi?id=758059
2016-05-15 22:07:14 +1000 Jan Schmidt <jan@centricular.com>
* gst/multifile/gstsplitmuxpartreader.c:
splitmuxsrc: Connect to demux signals before activating
Fix a race in splitmuxsrc by properly connecting to the
demuxer signals we're interested in *before* setting it running.
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2016-05-15 13:31:37 +0300 Sebastian Dröge <sebastian@centricular.com>
* docs/plugins/gst-plugins-good-plugins.args:
* docs/plugins/gst-plugins-good-plugins.signals:
* 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:
docs: Update for git master
2016-05-15 12:16:23 +0200 Olivier Crête <olivier.crete@collabora.com>
* gst/rtp/gstrtpmp4gpay.c:
* gst/rtp/gstrtpmp4gpay.h:
rtpmp4gpay: Don't produce timestamps based on byte count
The GST_BUFFER_OFFSET of output buffers returned to GstRtpBasePayload
should reflect the number of "samples" in the unit of the RTP clock in this
buffer. If this is not true, then it shouldn't be set.
https://bugzilla.gnome.org/show_bug.cgi?id=761943
2016-05-15 12:24:03 +0200 Edward Hervey <bilboed@bilboed.com>
* gst/matroska/matroska-mux.c:
matroska-mux: Fix strcmp usage
Just use g_strcmp0 which can handle NULL entries
2016-03-04 10:14:00 +0100 Carlos Rafael Giani <dv@pseudoterminal.org>
* ext/soup/gstsouphttpsrc.c:
souphttpsrc: Use audio/x-unaligned-raw instead of audio/x-raw for L16 data
Directly setting audio/x-raw caps leads to problems when the delivered
data blocks do not align properly at sample boundaries (for example, a
data block with 391 bytes). So, instead, set audio/x-unaligned-raw to
let a parser be autoplugged.
https://bugzilla.gnome.org/show_bug.cgi?id=689460
2016-05-12 11:52:09 +0900 Seungha Yang <sh.yang@lge.com>