Olivier Crête [Wed, 23 Jan 2019 23:27:06 +0000 (18:27 -0500)]
rtp session: Add test for collision loopback detection
Ignore further collisions if the remote SSRC change with ours, it's
probably because someone is sending us back the packets we send out.
Olivier Crête [Wed, 23 Jan 2019 23:14:23 +0000 (18:14 -0500)]
rtpsession tests: Add test for third-party collision detection
Add tests to validate the code that ignores the same packets coming
from 2 different sources (an third-party collision).
Olivier Crête [Wed, 23 Jan 2019 22:19:15 +0000 (17:19 -0500)]
rtpsession: Add test for collision on incoming packets
Make sure that the collision is properly detected on incoming packets.
Olivier Crête [Wed, 23 Jan 2019 22:09:27 +0000 (17:09 -0500)]
rtpsession test: Verify that on-ssrc-collision message is emitted
Olivier Crête [Wed, 23 Jan 2019 21:58:22 +0000 (16:58 -0500)]
rtpsession: Also send conflict event when sending packet
If the conflict is detected when sending a packet, then also send an
upstream event to tell the source to reconfigure itself.
Also ignore the collision if we see more than one collision from the same
remote source to avoid problems on loops.
Song Bing [Mon, 15 Apr 2019 23:32:03 +0000 (16:32 -0700)]
v4l2transform: set right buffer count.
Set right buffer count to avoid one buffer.
Olivier Crête [Thu, 27 Jun 2019 23:47:41 +0000 (19:47 -0400)]
rtph265pay: Also immediately send packet if it is a suffix NAL
Immediately send packet if it contains any suffix NAL, this is required
in case they come after the VCL nal to not have to wait until the next frame.
Olivier Crête [Thu, 27 Jun 2019 23:46:01 +0000 (19:46 -0400)]
rtph265pay: Don't drop second byte of NAL header
At least keep 2 bytes per NAL even if the second one is 0, the
second byte of the NAL header could very well be 0.
Olivier Crête [Wed, 26 Jun 2019 20:42:44 +0000 (16:42 -0400)]
rtph26xpay: Avoid print when there is no bundle at end of packet
Olivier Crête [Wed, 26 Jun 2019 20:25:01 +0000 (16:25 -0400)]
rtph26xpay: Wait until there is a VCL or suffix NAL to send
With unit tests.
Olivier Crête [Wed, 19 Jun 2019 21:16:03 +0000 (17:16 -0400)]
rtph265pay test: Add unit tests for aggregation
Olivier Crête [Tue, 18 Jun 2019 23:07:38 +0000 (19:07 -0400)]
rtph265pay: Implement Aggregation packets
Align with rtph264pay
Olivier Crête [Tue, 18 Jun 2019 19:03:09 +0000 (15:03 -0400)]
rtph264pay test: Add unit tests for aggregation
Olivier Crête [Tue, 18 Jun 2019 17:45:15 +0000 (13:45 -0400)]
rtph264pay: Report latency when in maximal aggregation mode
Olivier Crête [Mon, 17 Jun 2019 15:31:53 +0000 (11:31 -0400)]
rtph264pay: Default to not adding latency when aggregating
Send the bundle as soon as there is one VCL unit in the packet at
the end of an incoming buffer.
The DELTA_UNIT flag is not reliable, so ignore it.
Olivier Crête [Fri, 14 Jun 2019 20:54:23 +0000 (16:54 -0400)]
rtp-payloading test: Fix working to 1.0 buffers instead of groups
Olivier Crête [Thu, 13 Jun 2019 22:07:35 +0000 (18:07 -0400)]
rtph265pay: Replace fragmentation while-loop with for-loop
Align with rtph264pay
Olivier Crête [Thu, 13 Jun 2019 21:42:05 +0000 (17:42 -0400)]
rtph265pay: Rename payload_len to max_fragment_size
Align to rtph264pay
Olivier Crête [Thu, 13 Jun 2019 21:30:08 +0000 (17:30 -0400)]
rtph265pay: Clean up _payload_nal
Move determining whether we need to fragment at all into the
fragmenter.
Align with rtph264pay
Olivier Crête [Thu, 13 Jun 2019 21:23:26 +0000 (17:23 -0400)]
rtph265pay: Extract sending fragments into _payload_nal_fragment
Align with rtph264pay
Olivier Crête [Thu, 13 Jun 2019 20:22:57 +0000 (16:22 -0400)]
rtph265pay: Extract sending a single packet into _payload_nal_single
Align with rtph264pay
Olivier Crête [Thu, 13 Jun 2019 20:14:31 +0000 (16:14 -0400)]
rtph265pay: Define and use FU_A_TYPE_ID
Align with rtph264pay
Olivier Crête [Thu, 13 Jun 2019 20:08:37 +0000 (16:08 -0400)]
rtph265pay: Use snake_case variables
Align with rtph264pay
Olivier Crête [Thu, 13 Jun 2019 20:04:39 +0000 (16:04 -0400)]
rtph265pay: Clean up whitespace and syntax
Align with rtph264pay
Jan Alexander Steffens (heftig) [Tue, 3 Jul 2018 17:39:25 +0000 (19:39 +0200)]
rtph264pay: Support STAP-A bundling
Add a new property "do-aggregate"* to the H.264 RTP payloader which
enables STAP-A aggregation as per [RFC-6184][1]. With aggregation enabled,
packets are bundled instead of sent immediately, up until the MTU size.
Bundles also end at access unit boundaries or when packets have to be
fragmented.
*: The property-name is kept generic since it might apply more widely,
e.g. STAP-B or MTAP.
[1]: https://tools.ietf.org/html/rfc6184#section-5.7
Closes https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/issues/434
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