Jan Alexander Steffens (heftig) [Mon, 5 Nov 2018 16:15:39 +0000 (17:15 +0100)]
rtph264pay: Fix delta-unit/discont handling when injecting SPS/PPS
Apply the wanted delta-unit and discont to the first packet; following
packets for this frame are always delta units and not discont.
Jan Alexander Steffens (heftig) [Mon, 5 Nov 2018 18:03:45 +0000 (19:03 +0100)]
rtph264pay: Replace fragmentation while-loop with for-loop
Jan Alexander Steffens (heftig) [Mon, 5 Nov 2018 17:57:38 +0000 (18:57 +0100)]
rtph264pay: Calculate the right max_fragments
Jan Alexander Steffens (heftig) [Mon, 5 Nov 2018 17:36:35 +0000 (18:36 +0100)]
rtph264pay: Rename payload_len to max_fragment_size
Jan Alexander Steffens (heftig) [Mon, 5 Nov 2018 17:34:40 +0000 (18:34 +0100)]
rtph264pay: Clean up _payload_nal_fragment
Jan Alexander Steffens (heftig) [Mon, 5 Nov 2018 17:06:19 +0000 (18:06 +0100)]
rtph264pay: Clean up _payload_nal
Move determining whether we need to fragment at all into the fragmenter.
Jan Alexander Steffens (heftig) [Mon, 5 Nov 2018 17:04:13 +0000 (18:04 +0100)]
rtph264pay: Clean up _payload_nal_single
Jan Alexander Steffens (heftig) [Mon, 5 Nov 2018 16:55:23 +0000 (17:55 +0100)]
rtph264pay: Extract sending fragments into _payload_nal_fragment
Jan Alexander Steffens (heftig) [Mon, 5 Nov 2018 16:49:52 +0000 (17:49 +0100)]
rtph264pay: Extract sending a single packet into _payload_nal_single
Jan Alexander Steffens (heftig) [Mon, 5 Nov 2018 16:10:03 +0000 (17:10 +0100)]
rtph264pay: Define and use FU_A_TYPE_ID
Jan Alexander Steffens (heftig) [Mon, 5 Nov 2018 16:07:06 +0000 (17:07 +0100)]
rtph264pay: Use snake_case variables
Jan Alexander Steffens (heftig) [Mon, 5 Nov 2018 16:04:14 +0000 (17:04 +0100)]
rtph264pay: Clean up whitespace and syntax
Olivier Crête [Thu, 6 Jun 2019 20:05:31 +0000 (16:05 -0400)]
rtpjitterbuffer: Unlock output if the queue is full
Thomas Bluemel [Sun, 30 Jun 2019 05:17:28 +0000 (23:17 -0600)]
rtpjitterbuffer: Ignore unsolicited rtx packets.
If an rtx packet arrives that hasn't been requested (it might
have been requested from prior to a reset), ignore it so that
it doesn't inadvertently trigger a clock skew.
Havard Graff [Sun, 30 Jun 2019 05:16:44 +0000 (23:16 -0600)]
rtpjitterbuffer: Add unit test for unsolicited rtx affecting skew
Thomas Bluemel [Thu, 13 Jun 2019 21:45:28 +0000 (15:45 -0600)]
rtpjitterbuffer: Only calculate skew or reset if no gap.
In the case of reordered packets, calculating skew would cause
pts values to be off. Only calculate skew when packets come
in as expected. Also, late RTX packets should not trigger
clock skew adjustments.
Fixes #612
Mart Raudsepp [Tue, 2 Jul 2019 18:21:05 +0000 (21:21 +0300)]
qtdemux: Provide a 30 frames lead-in for MP3
mpegaudioparse suggests MP3 needs 10 or 30 frames of lead-in (depending on
mpegaudioversion, which we don't know here), thus provide at least 30 frames
lead-in for such cases as a followup to commit
cbfa4531ee5ef.
Olivier Crête [Fri, 24 May 2019 14:31:39 +0000 (10:31 -0400)]
rtpjitterbuffer: max-dropout-time gets cast to int32
So any value over MAXINT32 gets considered as negative and is silently ignored.
Mathieu Duponchelle [Tue, 2 Jul 2019 11:00:32 +0000 (13:00 +0200)]
qtdemux: do_seek can never be called with a NULL event
Mathieu Duponchelle [Mon, 1 Jul 2019 20:38:41 +0000 (22:38 +0200)]
qtdemux: only adjust segment time when adjusting segment start
We ended up setting segment.time to segment.position when doing
reverse playback, which is obviously wrong.
Mathieu Duponchelle [Mon, 1 Jul 2019 11:54:13 +0000 (13:54 +0200)]
rtspsrc: unref the event in element seek handler
Mathieu Duponchelle [Fri, 28 Jun 2019 22:25:26 +0000 (00:25 +0200)]
rtspsrc: handle seek event on the element
Without this, the user has to wait for rtspsrc to have sent a PLAY
request and exposed its pads before seeking it.
Nicolas Dufresne [Wed, 26 Jun 2019 22:03:29 +0000 (18:03 -0400)]
multiudpsink: Add missing socket.h include
Without this include, macro like SO_BINDTODEVICE is not visible and
associated feature gets out-compiled. This also affects the support for
SO_SNDBUF.
Jan Alexander Steffens (heftig) [Mon, 24 Jun 2019 15:35:15 +0000 (17:35 +0200)]
flvmux: Clear new_tags if sending metadata in header
This avoids sending an additional metadata object right after the
headers.
Song Bing [Wed, 13 Jun 2018 21:55:29 +0000 (14:55 -0700)]
v4l2videodec: Fix drain() function return type
Return right type for drain() function.
Mart Raudsepp [Mon, 24 Jun 2019 11:28:39 +0000 (14:28 +0300)]
audioparsers: add back segment clipping to parsers that have lost it
The pre_push_frame default clipping behaviour was introduced in 2010
with commit
30be03004e82 and modified with commit
4163969a2422 in 2011,
when most parsers didn't implement a pre_push_frame yet. Not having it
meant that clipping was done by default. Those that did implement a
pre_push_frame (flacparse and mpegaudioparse) at the time, had the flag
adjusted as part of the 2011 refactor work.
All other parsers got a pre_push_frame vfunc implementation only in
2013, but seem to have forgot to keep the clipping behaviour, as
was done automatically when a pre_push_frame implementation doesn't
exist for the parser. aacparse lost it with commit
91d4abcea in
July 2013; the others in Dec 2013 as part of AUDIO_CODEC tag posting
in commits
6f89b430e,
d2ab5199b,
29f2cae12,
753d3c23a and
292780574.
Tim-Philipp Müller [Mon, 24 Jun 2019 09:42:31 +0000 (09:42 +0000)]
v4l2: fix compiler warning due to c99-ism
Jan Alexander Steffens (heftig) [Wed, 19 Jun 2019 12:28:28 +0000 (14:28 +0200)]
test: flvmux: Test changing caps with one sinkpad
These tests segfault without the preceding crash fix.
Jan Alexander Steffens (heftig) [Wed, 19 Jun 2019 12:08:06 +0000 (14:08 +0200)]
test: flvmux: Use gst_harness_sink_push_many
And check its return value.
Jan Alexander Steffens (heftig) [Wed, 19 Jun 2019 10:31:46 +0000 (12:31 +0200)]
flvmux: Simplify an if-else chain
Merge the identical branches and turn the condition around to make it
easier to read.
Jan Alexander Steffens (heftig) [Wed, 19 Jun 2019 10:28:22 +0000 (12:28 +0200)]
flvmux: Avoid crash when changing caps without both streams
mux->video_pad and mux->audio_pad can be NULL if the corresponding pad
has not been requested.
Sebastian Dröge [Wed, 12 Jun 2019 12:57:48 +0000 (15:57 +0300)]
rtpgstpay: Send caps anyway if caps are pending in the adapter but are different from the new ones
Otherwise it can happen that we receive a caps event, then another caps
event and only then buffers. We would then send out the first caps event
in the stream but mark buffers with the caps version of the second caps
event.
Sebastian Dröge [Wed, 12 Jun 2019 11:57:24 +0000 (14:57 +0300)]
rtpgstdepay: Only store the current caps and drop old caps immediately
Otherwise it can happen that we already collected 7 caps, miss the 8th
caps packet (packet loss) and then re-use the 1st caps for the following
buffers instead of the 8th caps which will likely cause errors further
downstream unless both caps are accidentally the same.
Keeping old caps around does not seem to have any value other than
potentially causing errors. We would always receive new caps whenever
they change (even if they were previous ones) and it's very unlikely
that they happen to be exactly the same as the previous ones.
Also after having received new caps or a buffer with a next caps
version, no buffers with old caps version will arrive anymore.
Jan Schmidt [Fri, 14 Jun 2019 16:00:43 +0000 (02:00 +1000)]
rtpjitterbuffer: Clear clock master before unreffing
Make sure to clear any master clock on the media_clock
before unreffing it to release the timer callback that's
updating the clock and keeping it reffed.
Jan Schmidt [Sun, 16 Jun 2019 01:07:31 +0000 (11:07 +1000)]
matroska: Initialise a video_context field to satisfy valgrind
Clear the mastering_display_info_present field explicitly
after reallocating the track context into a video context
to avoid uninitialised warnings in valgrind
Thibault Saunier [Fri, 14 Jun 2019 21:34:31 +0000 (17:34 -0400)]
docs: Fix link to strings
We can't link to #gchar* this way.
Mathieu Duponchelle [Thu, 13 Jun 2019 22:17:22 +0000 (00:17 +0200)]
jitterbuffer: unset DTS on output buffers
Mathieu Duponchelle [Wed, 22 May 2019 19:40:52 +0000 (21:40 +0200)]
splitmuxsink: set the same seqnum on flush_start / flush_stop
It's currently not made mandatory by aggregator, but it might
eventually be, and is more consistent behaviour
See https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/issues/977
Mikhail Fludkov [Thu, 13 Jun 2019 09:55:04 +0000 (11:55 +0200)]
rtpjitterbuffer: late packets shouldn't affect PTS of the following packet
If, say, a rtx-packet arrives really late, this can have a dramatic
effect on the jitterbuffer clock-skew logic, having it being reset
and losing track of the current dts-to-pts calculations, directly affecting
the packets that arrive later.
This is demonstrated in the test, where a RTX packet is pushed in really
late, and without this patch the last packet will have its PTS affected
by this, where as a late RTX packet should be redundant information, and
not affect anything.
Mikhail Fludkov [Wed, 12 Jun 2019 08:47:39 +0000 (10:47 +0200)]
rtpjitterbuffer: fix rtx delay calulation when large packet spacing
Stian Selnes [Thu, 24 Nov 2016 17:18:01 +0000 (18:18 +0100)]
rtpjitterbuffer: Fix delay for EXPECTED timers added by gaps
This patch corrects the delay set on EXPECTED timers that are added when
processing gaps. Previously the delay could be too small so that
'timout + delay' was much less than 'now', causing the following retries
to be scheduled too early. (They were sent earlier than
rtx-retry-timeout after the previous timeout.)
Havard Graff [Tue, 20 Nov 2018 15:11:12 +0000 (16:11 +0100)]
rtpjitterbuffer: don't try and calculate packet-rate if seqnum are jumping
Turns out that the "big-gap"-logic of the jitterbuffer has been horribly
broken.
For people using lost-events, an RTP-stream with a gap in sequencenumbers,
would produce exactly that many lost-events immediately.
So if your sequence-numbers jumped 20000, you would get 20000 lost-events
in your pipeline...
The test that looks after this logic "test_push_big_gap", basically
incremented the DTS of the buffer equal to the gap that was introduced,
so that in fact this would be more of a "large pause" test, than an
actual gap/discontinuity in the sequencenumbers.
Once the test was modified to not increment DTS (buffer arrival time) with
a similar gap, all sorts of crazy started happening, including adding
thousands of timers, and the logic that should have kicked in, the
"handle_big_gap_buffer"-logic, was not called at all, why?
Because the number max_dropout is calculated using the packet-rate, and
the packet-rate logic would, in this particular test, report that
the new packet rate was over 400000 packets per second!!!
I believe the right fix is to don't try and update the packet-rate if
there is any jumps in the sequence-numbers, and only do these calculations
for nice, sequential streams.
Havard Graff [Wed, 12 Jun 2019 09:16:22 +0000 (11:16 +0200)]
rtpjitterbuffer: fix unused variables
Jan Schmidt [Tue, 11 Jun 2019 16:42:42 +0000 (02:42 +1000)]
splitmuxsrc: Protect initial pad configuration with the object lock
gst_splitmux_src_activate_part() configures the pad information
before starting the pad task, but occasionally the changes it makes
to the pad are not seen in the pad task because they're not
protected by the right locking. Use the pad's object lock to
protect those variables.
Jan Schmidt [Tue, 11 Jun 2019 15:42:20 +0000 (01:42 +1000)]
splitmuxsrc: Restart pad task on a reconfigure
On a reconfigure event, restart streaming on the pad so
that switching tracks in playbin works cleanly
Jan Schmidt [Tue, 11 Jun 2019 08:40:09 +0000 (18:40 +1000)]
splitmuxsrc: Use an RW lock instead of a mutex to protect the pad list
Fix a deadlock around the pads list by using an RW lock to
allow simultaneous readers. The pad list doesn't really changes
except at startup and shutdown.
Jan Schmidt [Tue, 11 Jun 2019 13:18:24 +0000 (23:18 +1000)]
splitmuxsrc: Ignore duplicate seeks
Use the seqnum to ignore duplicated seek events.
Jan Schmidt [Tue, 28 May 2019 23:20:07 +0000 (09:20 +1000)]
splitmuxsink: Improve debug output
Make the debug output less confusing by not mentioning a src
pad when doing calculations on the sink pad side.
Improve debug around why a GOP is considered overflowing a fragment
Jan Schmidt [Tue, 28 May 2019 23:20:07 +0000 (09:20 +1000)]
splitmuxsink: Give internal queues useful names
Makes debug output more useful
Mart Raudsepp [Wed, 5 Jun 2019 20:13:33 +0000 (23:13 +0300)]
qtdemux: Provide a 2 frames lead-in for audio decoders
AAC and various other audio codecs need a couple frames of lead-in to
decode it properly. The parser elements like aacparse take care of it
via gst_base_parse_set_frame_rate, but when inside a container, the
demuxer is doing the seek segment handling and never gives lead-in
data downstream.
Handle this similar to going back to a keyframe with video, in the
same place. Without a lead-in, the start of the segment is silence,
when it shouldn't, which becomes especially evident in NLE use cases.
Mart Raudsepp [Tue, 28 May 2019 17:14:49 +0000 (20:14 +0300)]
qtdemux: remove indent exception and reindent
As the indent disabling isn't playing along for a following fix,
remove the indent disabling and reindent in a way that doesn't
look too stupid.
Philippe Normand [Fri, 8 Mar 2019 14:43:20 +0000 (14:43 +0000)]
v4l2: Fix H.264 level 3 string representation
The string_to_level function handles "3" so the level_to_string function should
do the same, to prevent caps negotiation issues.
Philippe Normand [Mon, 4 Mar 2019 11:05:29 +0000 (11:05 +0000)]
v4l2: Profile and level probing support for encoders and decoders
There used to be some profile/level support in encoders. This code was moved to
GstV4l2Codecs and is now also used for decoders. The caps templates for the
H.264, H.265, MPEG4, VP8 and VP9 encoders and decoders should now reflect the
profiles and levels advertised by the kernel.
Aaron Boxer [Mon, 3 Jun 2019 20:21:12 +0000 (16:21 -0400)]
matroskamux: fix typo in property description
Nicolas Dufresne [Tue, 4 Jun 2019 17:39:00 +0000 (13:39 -0400)]
supp: Ignore leaks caused by shout/sethostent
sethostent() seems to be using a global state and we endup with leaks from
that API when called through shout_init(). We had the option to only
ignore the shout case, but the impression is that if we have shout and
another sethostend user, as it's a global state, we may endup with a
different stack trace for the same leak. So in the end, we just ignore
memory allocated by sethostent in general.
Thibault Saunier [Tue, 30 Apr 2019 21:28:25 +0000 (17:28 -0400)]
pulse-device: Hide the alsa device provider if we provide alsa devices
Nicolas Dufresne [Tue, 21 May 2019 19:25:03 +0000 (15:25 -0400)]
rtpssrcdemux: Avoid taking streamlock out-of-band
In this change we now protect the internal srcpads list using the
stream lock and limit usage of the internal stream lock to
preventing data flowing on the other src pad type while creating
and signalling the new pad.
This fixes a deadlock with RTPBin shutdown lock. These two locks would
end up being taken in two different order, which caused a deadlock. More
generally, we should not rely on a streamlock when handling out-of-band
data, so as a side effect, we should not take a stream lock when
iterating internal links.
Damian Hobson-Garcia [Mon, 27 May 2019 09:08:54 +0000 (18:08 +0900)]
v4l2object: Orphan buffer pool on object_stop if supported
Use V4L2 buffer orphaning, on recent kernels so that
the device can be restarted immediately with
a new buffer pool during renogatiation.
Damian Hobson-Garcia [Thu, 30 May 2019 04:12:31 +0000 (13:12 +0900)]
v4l2bufferpool: Free orphaned allocator resources when buffers are released
Allocator resources cannot be freed when a buffer pool is orphaned
while its buffers are in use. They should, however, be freed once those
buffers are no longer needed. This patch disposes of any buffers
belonging to an orphaned pool as they are released, and makes sure
that the allocator is cleaned up when the last buffer is returned.
Damian Hobson-Garcia [Thu, 30 May 2019 02:13:07 +0000 (11:13 +0900)]
v4l2bufferpool: return TRUE when buffer pool orphaning succeeds
When trying to orphan a buffer pool, successfully return and unref
the pool when the pool is either successfully stopped or orphaned.
Indicate failure and leave the pool untouched otherwise.
Niels De Graef [Fri, 31 May 2019 21:04:11 +0000 (23:04 +0200)]
meson: Bump minimal GLib version to 2.44
This means we can use some newer features and get rid of some
boilerplate code using the G_DECLARE_* macros.
As discussed on IRC, 2.44 is old enough by now to start depending on it.
Sebastian Dröge [Wed, 5 Sep 2018 18:10:51 +0000 (21:10 +0300)]
qtmux: Use size of first closed caption buffer in prefill mode
It must be accurate for all samples to work in Final Cut properly, so
the best we can do is to assume that all samples are the same as the
first. Bigger samples are truncated, smaller samples are padded.
Mathieu Duponchelle [Wed, 29 May 2019 20:06:58 +0000 (22:06 +0200)]
doc: remove xml from comments
Tim-Philipp Müller [Wed, 29 May 2019 10:02:26 +0000 (11:02 +0100)]
docs: update plugins cache
And add gtk+ and qt plugins
Tim-Philipp Müller [Wed, 29 May 2019 09:58:40 +0000 (10:58 +0100)]
dv, gtk, qt, osxaudio, osxvideo, waveform: add to plugins list
Makes sure the paths for these plugins are included in the
uninstalled plugin paths list. And also for the docs.
Fixes #604
Sebastian Dröge [Thu, 18 Apr 2019 12:31:00 +0000 (15:31 +0300)]
matroskamux: Add new property to offset all streams to start at zero
This takes the timestamp of the earliest stream and offsets it so that
it starts at 0. Some software (VLC, ffmpeg-based) does not properly
handle Matroska files that start at timestamps much bigger than zero.
Closes https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/issues/449
Tim-Philipp Müller [Tue, 28 May 2019 13:13:56 +0000 (14:13 +0100)]
rtpmp4gdepay: don't spam debug log for broken ADTS-in-RTP AAC
Print warning only once.
Sebastian Dröge [Wed, 22 May 2019 15:06:04 +0000 (18:06 +0300)]
splitmuxsink: Only set running time on finalizing sink element when in async-finalize mode
There is only a single sink element in async-finalize mode, and we would
keep the running time from previous fragments set in that case. As we
don't ever set the running time for the very last fragment on EOS, this
would mean that the closing time reported for the very last fragment is
the same as the closing time of the previous fragment.
Nicolas Dufresne [Thu, 26 Mar 2015 17:08:32 +0000 (13:08 -0400)]
rtspsrc: Remove uneeded keep-alive hack
The rtsp connection code has been fixed now.
https://bugzilla.gnome.org/show_bug.cgi?id=744209
Vivia Nikolaidou [Sun, 26 May 2019 14:46:06 +0000 (17:46 +0300)]
rtpjitterbuffer: Print GstClockTimeDiff as GST_STIME_FORMAT
Mathieu Duponchelle [Sat, 25 May 2019 17:45:02 +0000 (19:45 +0200)]
doc: update plugin cache
Mathieu Duponchelle [Sat, 25 May 2019 15:25:02 +0000 (17:25 +0200)]
videomixer: the documentation for GstVideoMixer2Pad is not exposed
Mathieu Duponchelle [Sat, 25 May 2019 14:56:32 +0000 (16:56 +0200)]
doc: fix element section documentations
Element sections were not rendered anymore after the hotdoc
port, fixing this revealed a few incorrect links.
Nicolas Dufresne [Tue, 19 Feb 2019 17:15:19 +0000 (12:15 -0500)]
rtpbin: Improve RTPStorage action signal documentation
This is a tiny clarification as the storage was loosely named "storage".
This change clarify that the storage is specificaly used for received RTP
packets. This is unlike the storage found in rtprtxsend that stores a
backlog of sent RTP packets.
Seungha Yang [Sun, 5 May 2019 13:16:36 +0000 (22:16 +0900)]
matroska: Add BT2020_10, PQ and HLG transfer functions
The direct use of newly added transfer functions
Sebastian Dröge [Thu, 23 May 2019 09:38:06 +0000 (12:38 +0300)]
aasink: Generate pkg-config file for the plugin
Seungha Yang [Wed, 22 May 2019 02:01:17 +0000 (11:01 +0900)]
multifilesink: Fix documentation of max-file-duration property
The max-file-duration property works with max-duration mode
Nicolas Dufresne [Tue, 14 May 2019 21:36:14 +0000 (17:36 -0400)]
rtpsession: Always keep at least one NACK on early RTCP
We recently added code to remove outdate NACK to avoid using bandwidth
for packet that have no chance of arriving on time. Though, this had a
side effect, which is that it was to get an early RTCP packet with no
feedback into it. This was pretty useless but also had a side effect,
which is that the RTX RTT value would never be updated. So we we stared
having late RTX request due to high RTT, we'd never manage to recover.
This fixes the regression by making sure we keep at least one NACK in
this situation. This is really light on the bandwidth and allow for
quick recover after the RTT have spiked higher then the jitterbuffer
capacity.
Thibault Saunier [Thu, 16 May 2019 13:14:19 +0000 (09:14 -0400)]
docs: Stop building the doc cache by default
Fixes https://gitlab.freedesktop.org/gstreamer/gst-docs/issues/36
Thibault Saunier [Tue, 14 May 2019 02:53:59 +0000 (22:53 -0400)]
docs: Update plugins documentation cache
Thibault Saunier [Tue, 23 Apr 2019 16:28:23 +0000 (12:28 -0400)]
doc: Fix some docstrings
Thibault Saunier [Mon, 22 Oct 2018 09:39:55 +0000 (11:39 +0200)]
doc: Port documentation to hotdoc
Thibault Saunier [Mon, 12 Nov 2018 11:05:45 +0000 (08:05 -0300)]
Mark some properties as DOC_SHOW_DEFAULT
Thibault Saunier [Mon, 22 Oct 2018 09:39:24 +0000 (11:39 +0200)]
docs: Port all docstring to gtk-doc markdown
Thiago Santos [Fri, 3 May 2019 05:14:35 +0000 (22:14 -0700)]
rtspsrc: do not try to send EOS with invalid seqnum
The second udpsrc (rtcp) might not have seen the segment event if it was
not enabled or if rtcp is not available on the server. So if the
application tries to send an EOS event it will try to set an invalid
seqnum to the event.
Nicolas Dufresne [Wed, 24 Apr 2019 17:54:12 +0000 (13:54 -0400)]
rtpsource: Add more information to probation warning
Nicolas Dufresne [Wed, 24 Apr 2019 17:47:54 +0000 (13:47 -0400)]
rtpsession: Call on-new-ssrc earlier
Right now, we may call on-new-ssrc after we have processed the first
RTP packet. This prevents properly configuring the source as some
property like "probation" are copied internally for use as a
decreasing counter. For this specific property, it prevents the
application from disabling probation on auxiliary sparse stream.
Probation is harmful on sparse streams since the probation algorithm
assume frequent and contiguous RTP packets.
Seungha Yang [Tue, 19 Feb 2019 04:34:49 +0000 (13:34 +0900)]
matroskamux: Write MasteringMetadata and Max{CLL,FALL}
Enable muxing with HDR meta data if upstream provided it
Seungha Yang [Mon, 18 Feb 2019 14:28:50 +0000 (23:28 +0900)]
matroskademux: Add support parsing HDR metadata
Set SMPTE ST 2086 mastering-display-metadata and
content-light-level to caps, if any
Seungha Yang [Tue, 19 Feb 2019 09:27:23 +0000 (18:27 +0900)]
matroska: Remove white space
Sebastian Dröge [Wed, 1 May 2019 07:00:51 +0000 (10:00 +0300)]
rtprawdepay: Don't get rid of the buffer pool on FLUSH_STOP
We expect there to be a pool as long as the caps are known and
FLUSH_STOP is not resetting the caps. Getting rid of the pool would
cause assertions.
Fixes https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/issues/584
Danny Smith [Fri, 8 Feb 2019 09:09:17 +0000 (10:09 +0100)]
rtpbin: Free storage when freeing session
Sebastian Dröge [Thu, 25 Apr 2019 18:52:42 +0000 (21:52 +0300)]
matroskamux: Fix typo in error message
Sebastian Dröge [Thu, 25 Apr 2019 08:19:06 +0000 (11:19 +0300)]
imagefreeze: Only set the DISCONT flag on the first buffer after segment start
okuoku [Tue, 23 Apr 2019 17:38:32 +0000 (02:38 +0900)]
jack: Use jack_free(3) to release ports
Port objects acquired with jack_get_ports() need to be freed with
jack_free(3), not stdlib free().
On Windows, Jack may be linked against different libc than GStreamer
libraries so free()ing port objects directly might cause crash because
of libc mismatch.
Philippe Normand [Tue, 23 Apr 2019 09:10:01 +0000 (10:10 +0100)]
scaletempo: Advertise interleaved layout in caps templates
Scaletempo doesn't support non-interleaved layout. Not explicitely stating this
would trigger critical warnings and a caps negotiation failure when scaletempo
is used as playbin audio-filter.
Patch suggested by George Kiagiadakis <george.kiagiadakis@collabora.com>.
Fixes #591
Seungha Yang [Sun, 21 Apr 2019 11:12:28 +0000 (20:12 +0900)]
meson: matroska: Ensure header dependency not only library
Library existence does not guarantee header.
Robert Rosengren [Tue, 13 Nov 2018 12:48:11 +0000 (13:48 +0100)]
multidupsink: Use gst_net_utils_set_socket_tos for QoS DSCP
Util function in net library exists for setting QoS DSCP on socket, hence
use it to simplify code.
Tim-Philipp Müller [Fri, 19 Apr 2019 09:27:38 +0000 (10:27 +0100)]
Back to development
Tim-Philipp Müller [Thu, 18 Apr 2019 23:23:16 +0000 (00:23 +0100)]
Release 1.16.0