ChangeLog
=========
2016-06-09 Sebastian Dröge <[email protected]>
* configure.ac:
releasing 1.8.2
2016-06-09 10:05:34 +0300 Sebastian Dröge <[email protected]>
* po/hr.po:
* po/pt_BR.po:
* po/sk.po:
po: Update translations
2016-06-09 09:30:48 +0900 Seungha Yang <[email protected]>
* gst/flv/gstflvdemux.c:
flvdemux: Fix unref assertion failure
Fix unref assertion failure
https://bugzilla.gnome.org/show_bug.cgi?id=767424
2016-06-07 20:53:34 -0400 Nicolas Dufresne <[email protected]>
* 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 <[email protected]>
* 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 <[email protected]>
* 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 16:04:52 -0400 Nicolas Dufresne <[email protected]>
* 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 <[email protected]>
* 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 <[email protected]>
* 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
2015-09-02 11:48:29 +0200 Philipp Zabel <[email protected]>
* 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 <[email protected]>
* sys/v4l2/gstv4l2videodec.c:
v4l2videodec: add gst_v4l2_decoder_cmd helper
https://bugzilla.gnome.org/show_bug.cgi?id=733864
2015-01-28 12:07:58 +0100 Enrico Jorns <[email protected]>
* 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 <[email protected]>
* 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 <[email protected]>
* 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 14:58:41 +0200 Philipp Zabel <[email protected]>
* 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
2016-01-27 09:57:38 +0100 Andreas Naumann <[email protected]>
* 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 <[email protected]>
* 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
2016-05-31 21:34:04 +0200 Josep Torra <[email protected]>
* 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
2016-02-10 19:56:59 +0530 Nirbheek Chauhan <[email protected]>
* sys/v4l2/gstv4l2deviceprovider.c:
v4l2: Don't leak v4l2 objects and props on probe errors
2016-05-31 17:04:32 +0300 Sebastian Dröge <[email protected]>
* 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 17:10:36 +0300 Sebastian Dröge <[email protected]>
* 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 16:38:34 +0300 Sebastian Dröge <[email protected]>
* 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
2016-03-15 03:25:26 +0530 Nirbheek Chauhan <[email protected]>
* gst/rtp/gstrtpjpegdepay.c:
rtpjpegdepay: Don't send invalid frames downstream after packet loss
or a DISCONT
After clearing the adapter due to a DISCONT, as might happen when
some packet(s)
have been lost, the depayloader was pushing data into the adapter
(which had no
header due to the clear), creating a headerless frame out of it, and
sending it
downstream. The downstream decoder would then usually ignore it;
unless there
were lots of DISCONTs from the jitterbuffer in which case the decoder
would reach
its max_errors limit and throw an element error. Now we just discard
that data.
It is probaby not worth trying to salvage this data because
non-progressive
jpeg does not degrade gracefully and makes the video unwatchable even
with
low packet loss such as 3-5%.
2016-05-25 17:11:13 +0200 Pierre Lamot <[email protected]>
* 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
<[email protected]>
* 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
2016-05-20 11:12:44 +0300 Sebastian Dröge <[email protected]>
* 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
2016-05-20 11:15:44 +0300 Sebastian Dröge <[email protected]>
* 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 10:56:52 +0300 Sebastian Dröge <[email protected]>
* 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
<[email protected]>
* 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 <[email protected]>
* 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
2016-05-12 11:52:09 +0900 Seungha Yang <[email protected]>
* gst/isomp4/qtdemux.c:
qtdemux: Parsing elst box based on version
segment_duration and media_time should be parsed based on version
of elst box. Specification defines that an elst box with version 1
has uint64 and int64 values for segment_duration and media_time,
respectively.
https://bugzilla.gnome.org/show_bug.cgi?id=766301
2016-05-15 12:30:50 +0300 Sebastian Dröge <[email protected]>
* gst/rtpmanager/rtpsession.c:
rtpsession: Take the lock already when reading the other stats, not
just for the hash table
https://bugzilla.gnome.org/show_bug.cgi?id=766025
2016-05-12 14:43:43 +0200 Patricia Muscalu <[email protected]>
* gst/auparse/gstauparse.c:
* gst/auparse/gstauparse.h:
auparse: Fix sticky event misordering warning
Make sure that src pad has caps before sending segment event.
https://bugzilla.gnome.org/show_bug.cgi?id=766359
2016-05-19 15:36:57 +0900 Seungha Yang <[email protected]>
* 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 <[email protected]>
* 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.
2016-05-15 22:07:14 +1000 Jan Schmidt <[email protected]>
* 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.
2016-05-14 23:39:22 +1000 Jan Schmidt <[email protected]>
* gst/multifile/gstsplitmuxsink.c:
* gst/multifile/gstsplitmuxsink.h:
splitmuxsink: Use GstBin async-handling instead of our own.
Set the async-handling property on GstBin to let it manage
async-handling instead of the local handling from the previous
commit. Works because of #174a5e in core
2016-05-14 18:32:52 +1000 Jan Schmidt <[email protected]>
* gst/multifile/gstsplitmuxsink.c:
* gst/multifile/gstsplitmuxsink.h:
splitmuxsink: Hide internal async state changes.
When switching fragments, hide the async-start/async-done
messages from the parent bin, as otherwise we sometimes (very rarely)
hang in PAUSED instead of returning / continuing to PLAYING
state.
2016-05-13 21:20:28 +1000 Jan Schmidt <[email protected]>
* gst/multifile/gstsplitmuxsink.c:
splitmuxsink: Remove stray carriage-return from debug
2015-04-30 14:43:04 +0200 Jesper Larsen <[email protected]>
* gst/avi/gstavimux.c:
avimux: Do not write index and header if idx is NULL
Fixes criticals with e.g.
videotestsrc num-buffers=1 ! identity drop-probability=1.0 ! avimux !
fakesink
https://bugzilla.gnome.org/show_bug.cgi?id=748700
2016-05-03 11:45:01 +0200 Havard Graff <[email protected]>
* gst/rtpmanager/gstrtpjitterbuffer.c:
* tests/check/elements/rtpjitterbuffer.c:
rtpjitterbuffer: Fix stall when receiving already lost packet
When a packet arrives that has already been considered lost as part
of a
large gap the "lost timer" for this will be cancelled. If the
remaining
packets of this large gap never arrives, there will be missing entries
in the queue and the loop function will keep waiting for these packets
to arrive and never push another packet, effectively stalling the
pipeline.
The proposed fix conciders parts of a large gap definitely lost (since
they are calculated from latency) and ignores the late arrivals.
In practice the issue is rare since large gaps are scheduled
immediately,
and for the stall to happen the late arrival needs to be processed
before this times out.
https://bugzilla.gnome.org/show_bug.cgi?id=765933
2016-05-11 09:28:13 +0300 Sebastian Dröge <[email protected]>
* gst/rtpmanager/rtpsession.c:
rtpsession: Don't notify about stats property changes while taking
the session lock
The signal handlers might want to actually get the value of the stats
property, which would take the session lock again and deadlock.
This was introduced by 2e960e70750a0cb7e1117d0c09d08597866a29ee.
https://bugzilla.gnome.org/show_bug.cgi?id=766025
2016-05-05 14:18:21 +0200 Miguel París Díaz <[email protected]>
* gst/rtpmanager/rtpsession.c:
rtpsession: Take session lock when creating stats
The access to the session hash table must happen while the session
lock is
taken, otherwise another thread might modify the hash table while
we're
creating the stats.
https://bugzilla.gnome.org/show_bug.cgi?id=766025
2016-05-04 09:30:27 +0300 Sebastian Dröge <[email protected]>
* ext/dv/gstdvdec.c:
* ext/dv/gstdvdemux.c:
dv: Use correct pixel-aspect-ratio values
The previous ones resulted in odd display aspect ratios and were
different
from the ones used by e.g. ffmpeg. The new ones now result in display
aspect
ratios of 4:3 and 16:9.
https://bugzilla.gnome.org/show_bug.cgi?id=765946
2016-05-03 21:17:01 -0300 Thiago Santos <[email protected]>
* gst/isomp4/qtdemux.c:
qtdemux: update segment when new duration is found
Otherwise the old segment will have a shorter stop time and would
cause the stream to end too early.
https://bugzilla.gnome.org/show_bug.cgi?id=765805
2016-05-04 11:37:29 -0300 Thiago Santos <[email protected]>
* gst/isomp4/qtdemux.c:
qtdemux: dismember activate_segment into 2 parts
One that updates and push a new segment, the other will move the
stream to the new segment starting position
https://bugzilla.gnome.org/show_bug.cgi?id=765805
2016-05-04 11:15:20 -0400 Xavier Claessens <[email protected]>
* gst/multifile/gstsplitmuxsink.c:
splitmuxsink: Fix deadlock case when source reaches EOS
https://bugzilla.gnome.org/show_bug.cgi?id=765072
2016-04-11 10:54:38 +0300 Sebastian Dröge <[email protected]>
* gst/udp/gstudpsrc.c:
udpsrc: Always bind to ANY when address is a multicast address and
not only on Windows
For IPv6 addresses, binding to a multicast group does not work on
Linux
either. Always bind to ANY and then later join the multicast group.
https://bugzilla.gnome.org/show_bug.cgi?id=764679
2016-04-28 16:24:52 +0300 Sebastian Dröge <[email protected]>
* gst/isomp4/gstqtmux.c:
* gst/isomp4/gstqtmuxmap.c:
qtmux: Allow MPEG-1 Layer 1 and 2 in addition to 3 in MP4
Via the MPEG-4 Part 3 spec we can support the other layers too.
Also correct the samples per frame calculation for MP3 if it's MPEG-2
or
MPEG-2.5.
https://bugzilla.gnome.org/show_bug.cgi?id=765725
2016-04-29 15:04:11 +0300 Sebastian Dröge <[email protected]>
* gst/isomp4/qtdemux.c:
* gst/isomp4/qtdemux.h:
qtdemux: Store the segment sequence number in the EOS events and
SEGMENT_DONE events/message
Also instead of storing it per stream, store it globally in the
demuxer. It's
the same for each stream anyway.
https://bugzilla.gnome.org/show_bug.cgi?id=765806
2016-04-27 20:46:34 +0300 Sebastian Dröge <[email protected]>
* gst/rtsp/gstrtspsrc.c:
* gst/rtsp/gstrtspsrc.h:
rtspsrc: Update caps for TCP whenever they change
We only changed them for UDP so far, which caused the wrong
seqnum-base and
other information to be passed to rtpjitterbuffer/etc when seeking.
This
usually wasn't that much of a problem as the code there is robust
enough, but
every now and then it causes us to drop up to 32756 packets before we
continue doing anything meaningful.
https://bugzilla.gnome.org/show_bug.cgi?id=765689
2016-04-27 20:33:38 +0300 Sebastian Dröge <[email protected]>
* gst/rtpmanager/gstrtpjitterbuffer.c:
rtpjitterbuffer: Ensure to not take caps with the wrong pt for
getting the clock-rate
Especially the caps on the pad might be out of date, and the new caps
would be
provided for the current pt via the request-pt-map signal.
https://bugzilla.gnome.org/show_bug.cgi?id=765689
2016-04-27 18:27:17 +0300 Sebastian Dröge <[email protected]>
* gst/rtsp/gstrtspsrc.c:
rtspsrc: Don't propagate spurious state change returns from internal
elements further
We handle them inside rtspsrc and override them in all other cases
anyway, so
do the same for "internal" state changes like PAUSED->PAUSED and
PLAYING->PLAYING.
This keeps unexpected NO_PREROLL to confuse state changes in GstBin.
See also https://bugzilla.gnome.org/show_bug.cgi?id=760532
https://bugzilla.gnome.org/show_bug.cgi?id=765689
2016-05-01 15:09:27 +0200 Mark Nauwelaerts <[email protected]>
* gst/avi/gstavimux.c:
avimux: set audio header rate according to calculated bps in stop_file
... now that set_fields is no longer called there by
e538608b3f90539003de21c1db238f3c9b946e30
2015-11-26 13:15:06 +0100 Dimitrios Katsaros <[email protected]>
* sys/v4l2/v4l2_calls.c:
v4l2: Change warning handling to break infinite message loop
v4l2src can cause an "infinite message loop" when a base control
exposed as a
property is not provided by the device. In these cases, if in the
warning message
handling for the bus, the GST_DEBUG_BIN_TO_DOT_FILE* category of
functions are used,
the src lookup causes a new warning to be posted on the bus, causing
a loop.
This patch changes the warning for these controls so they are not
posted on the bus.
https://bugzilla.gnome.org/show_bug.cgi?id=758703
2016-04-25 15:03:14 +0200 Mats Lindestam <[email protected]>
* gst/udp/gstmultiudpsink.c:
multiudpsink: Allow setting "socket-v6" without setting "socket" too
https://bugzilla.gnome.org/show_bug.cgi?id=764897
2016-04-27 13:53:00 +0300 Sebastian Dröge <[email protected]>
* gst/avi/gstavimux.c:
avimux: Actually store the largest audio chunk size for the VBR case
of MP2/MP3
3ea338ce271e1f6a96d2ed49d4472b091f6f8b7e changed avimux to do that,
but it
never actually kept track of the max audio chunk for MP3 and MP2.
These are
knowing the hdr.scale only after parsing the frames instead of at
setcaps
time.
2016-04-27 14:09:03 +0300 Sebastian Dröge <[email protected]>
* gst/avi/gstavimux.c:
avimux: Don't override maximum audio chunk size with the scale again
just before writing it
set_fields() should only be called in the beginning, otherwise we
will never
remember the maximum audio chunk size and write a wrong block
align... which
then causes wrong timestamps and other problems.
2016-04-22 15:02:16 +0100 Mario Sanchez Prada <[email protected]>
* ext/vpx/gstvpxenc.c:
vpxenc: Properly handle frames with too low duration
When a frame's duration is too low, calling gst_util_uint64_scale()
to scale its value can result into it being truncated to zero, which
will cause the vpx encoder to return an VPX_CODEC_INVALID_PARAM error
when trying to encode.
To prevent this from happening, we simply ignore the duration when
encoding if it becomes zero after scaling, logging a warning message.
https://bugzilla.gnome.org/show_bug.cgi?id=765391
2016-04-22 15:48:08 +0100 Tim-Philipp Müller <[email protected]>
* gst/deinterlace/gstdeinterlace.c:
deinterlace: fix description of linear interlacing method
2016-04-21 14:08:19 -0300 Thibault Saunier <[email protected]>
* gst/flv/gstflvmux.c:
flv: Handle the case where we do not get any CollectData in
handle_buffer
https://bugzilla.gnome.org/show_bug.cgi?id=765320
2016-02-09 17:17:09 +0000 Alex Ashley <[email protected]>
* gst/isomp4/qtdemux.c:
qtdemux: support seeking of CENC encrypted streams
When playing a stream that has been protected by DASH CENC, playback
will fail if a seek is performed. Qtdemux produces the error "stream
is protected using cenc, but no cenc protection system information
has been found" and playback stops.
The problem is that gst_qtdemux_reset() gets called as part of the
FLUSH during a seek. This function frees the protection_system_ids
array. When gst_qtdemux_configure_protected_caps() is called after the
seek has completed, the protection_system_ids array is empty and
qtdemux is unable to create the correct output caps for the protected
stream.
This commit changes it to only free the protection_system_ids on
hard resets.
https://bugzilla.gnome.org/show_bug.cgi?id=761787
2016-04-11 22:41:20 +0900 Seungha Yang <[email protected]>
* gst/isomp4/qtdemux.c:
qtdemux: Do not use unreliable framerate
timescale/1 is unreliable value for framerate. Due to downstream
element usually use framerate generated by qtdemux, let it be omitted
until the framerate can be reliably calculated.
https://bugzilla.gnome.org/show_bug.cgi?id=764733
2016-04-21 12:53:33 +0300 Sebastian Dröge <[email protected]>
* gst/isomp4/qtdemux.c:
* gst/isomp4/qtdemux.h:
Revert "qtdemux: expose streams with first moof for fragmented format"
This reverts commit d8bb6687ea251570c331038279a43d448167d6ad.
https://bugzilla.gnome.org/show_bug.cgi?id=764733
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