Commit graph

19155 commits

Author SHA1 Message Date
Mathieu Duponchelle
081509e030 rtpst2022-1-fec-*: protect additional RTP header fields
While the standard is a bit vague about whether the padding,
extension and marker bits should be protected:

> The usage, by senders and receivers, of the following bits shall
> be defined by the associated video/audio transport standards:

It is obviously necessary and useful for some formats (eg VP8)
that those indeed be protected.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/839>
2020-12-12 09:29:15 +00:00
Jan Schmidt
2d24a45c89 splitmuxsink: Unit test - check format/opened/closed sequence
Check the sequence of format-location/fragment-opened/fragment-closed
events is respected. There should be 1 format-location call for each
fragment-opened message, and 1 fragment-closed for each.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/833>
2020-12-12 03:28:56 +11:00
Jan Schmidt
d7a9a844f6 splitmuxsink: Fix for 'reference bytes muxed' check.
https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/798
introduced a check in the need-new-fragment logic to avoid starting a
new fragment unless there has been some data on the reference stream,
but the check is done against the number of bytes that have been
received on the input, not the number that were released for output
into the current fragment.

Fix the check to remember and test against bytes that have been sent
for output.

This also fixes a problem where starting a new fragment fails to
request a new filename from the format-location signal.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/833>
2020-12-12 03:28:19 +11:00
Jan Schmidt
67f70af1bb splitmuxsink: Add debug for fragment opened/closed msgs
When posting fragment-opened and fragment-closed messages,
put a debug statement in the logs

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/833>
2020-12-09 01:03:01 +11:00
Jan Schmidt
df8b147e75 splitmuxsink: Convert asserts into element errors.
Change some g_assert into element errors so that they can be
caught and the pipeline shut down.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/833>
2020-12-09 01:03:01 +11:00
Matthew Waters
656af79130 rtpmanager: update for rtp header extensions
Provide an implementation of the transport-wide-cc header extension and
use it in rtpfunnel.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/808>
2020-12-04 13:24:19 +11:00
Jose Quaresma
2a5d7b18a0 rpicamsrc: add vchostif library as it is required to build successful
fix: undefined reference to `vc_gencmd'
/usr/src/debug/gstreamer1.0-plugins-good/1.18.1-r0/build/../gst-plugins-good-1.18.1/sys/rpicamsrc/RaspiCamControl.c:1440: undefined reference to `vc_gencmd'

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/818>
2020-12-01 18:24:58 +00:00
Marijn Suijten
030b1b3fa5 tests/rtp-payloading: Use new AudioFormatInfo::fill_silence function
The function is renamed to be properly associated with AudioFormatInfo
(its instance) instead of AudioFormat (an unrelated enum), see [1] for
the rename itself.

[1]: https://gitlab.freedesktop.org/gstreamer/gst-plugins-base/-/merge_requests/940
2020-11-26 10:06:25 +02:00
Nirbheek Chauhan
552da8569b deinterlace: Enable x86 assembly with nasm on MSVC
We need to remove x86inc.asm from the list of compiled assembly files
because it is not supposed to be compiled separately. It is directly
included by yadif.asm, and it exports no symbols.

The object file was getting ignored on all platforms except on msvc
where it was causing a linker hang when building with debugging
enabled because the object file had no debug symbols (or similar).
We've seen this before in FFmpeg too, which uses nasm:
https://gitlab.freedesktop.org/gstreamer/meson-ports/ffmpeg/-/merge_requests/46

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/825>
2020-11-24 22:11:50 +05:30
Matthew Waters
9d14ae4740 qml: add some docs on display and contexts
Especially considering some dynamic pipeline scenarios.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/822>
2020-11-20 07:49:40 +00:00
Tim Schneider
2c76f03fe2 rpicamsrc: Added "src->started = FALSE;" to gst_rpi_cam_src_stop
Makes the element reusable multiple times after a state change back to READY.
Fixes #105

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/823>
2020-11-20 02:15:15 +11:00
Bing Song
8a0a7d932a v4l2: caps negotiate wrong as interlace feature
gst_caps_simplify() will move interlace format before normal video
format. It will cause caps negotiate prefer interlaced caps which
isn't expected. Seperate normal caps and interlaced caps and then
merge it will keep prefer progress video format.
Add ARGB/BGRA for interlaced caps.

Fixes https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/issues/802

Part-of <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/813>

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/813>
2020-11-16 15:12:28 +00:00
Havard Graff
79748dab2b rtpsession: never send on a non-internal source
This will end up as a "received" packet, due to the code in
source_push_rtp, which will think this is a packet being received.

Instead drop the packet and hope that either:
1. Something upstream responds to the GstRTPCollision event and changes
   SSRC used for sending.
2. That the application responds to the "on-ssrc-collision" signal, and
   forces the sender (payloader) to change its SSRC.
3. That the BYE sent to the existing user of this SSRC will respond to
   the BYE, and that we timeout this source, so we can continue sending
   using the chosen SSRC.

The test reproduces a scenario where we previously would have sent
on a non-internal source.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/817>
2020-11-13 21:35:58 +01:00
Havard Graff
97ced29277 rtpsource: rewrite timeout-check to avoid underflow
If current_time is < collision_timeout, we get an uint64 underflow, and
the check will trigger prematurely.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/817>
2020-11-13 21:30:06 +01:00
Vivia Nikolaidou
5a2f9d510f aacparse: Fix caps change handling
In baseparse we set the fixed caps flag on all src pads, therefore the
source pad caps query in get_allowed_caps will return the current caps.
Current caps won't necessarily intersect with the new caps (e.g. sample
rate change). Replace get_allowed_caps with peer_query_caps.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/816>
2020-11-13 13:10:05 +00:00
Tim-Philipp Müller
f5310ce346 tests: qtdemux: fix typo in caps field
timesacle -> timescale

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/815>
2020-11-12 23:40:13 +00:00
Tim-Philipp Müller
2ce5909f3a tests: qtdemux: fix crash on 32-bit architectures
Fixes https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/issues/803

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/815>
2020-11-12 23:40:13 +00:00
Sanchayan Maity
8c3ec64473 rtp: ldacpay: Add LDAC RTP payloader
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/757>
2020-11-11 22:59:19 +05:30
Sebastian Dröge
e3c16d0194 qmlglsink: Keep old buffers around a bit longer if they were bound by QML
We don't know exactly when QML will stop using them but it should be
safe to unref them after at least 2 more buffers were bound.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/810>
2020-11-11 11:37:27 +00:00
ChrisDuncanAnyvision
d9ea3346f3 rtspsrc: Ensure same group-id used for both TCP/UDP stream-start events
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/811>
2020-11-10 18:18:12 +00:00
ChrisDuncanAnyvision
e5f5e712c6 rtspsrc: Use consistent URI hashed stream-id for UDP and TCP/Interleaved streams
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/811>
2020-11-10 16:23:17 +00:00
Nirbheek Chauhan
35eabc3f95 meson: Enable some MSVC warnings for parity with GCC/Clang
This makes it easier to do development with MSVC by making it warn
on common issues that GCC/Clang error out for in our CI configuration.

Continuation from https://gitlab.freedesktop.org/gstreamer/gst-build/-/merge_requests/223

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/809>
2020-11-04 18:43:04 +05:30
Olivier Crête
99723bc1c1 rtpsource: Report for which local SSRC is a remote RB reporting on
This is useful in the Bundle case because there may be multiple local
and remote SSRCs in the same session.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/776>
2020-11-03 12:35:54 -05:00
Guillaume Desmottes
473a70bb21 docs: update plugins cache
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/530>
2020-11-03 09:51:27 +01:00
Guillaume Desmottes
ba3919ecb2 rtp: add rtpisacdepay
Depayload for the iSAC audio codec.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/530>
2020-11-03 09:51:27 +01:00
Guillaume Desmottes
a1e7b1fd61 rtp: add rtpisacpay
Payload for the iSAC audio codec.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/530>
2020-11-03 09:51:27 +01:00
Dinesh Manajipet
aa264e6b07 qmlglsink: Set qtitem's implicit width/height
This can be useful to let the layouts automatically resize qtitem
and also easily query a video's width/height from QML

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/802>
2020-11-02 13:53:46 +00:00
Sebastian Dröge
917bf649cc flvmux: Release pads via GstAggregator
See https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/797

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/801>
2020-11-02 08:46:21 +00:00
Matthew Waters
9d74a60810 qtmux: support muxing multiple codec_data for h264/h265
Each codec_data is put into its own SampleTableEntry inside the stsd.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/787>
2020-11-02 03:32:50 +00:00
Stéphane Cerveau
d664f400aa navseek: add hold_eos property
This property will tell the element to hold
the EOS event and keep it until the next
keystroke.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/792>
2020-11-01 15:19:46 +01:00
Jan Schmidt
81ecf076e8 splitmuxsrc: Fix comment in a test
Fix a comment in the splitmuxsrc robust muxing test so it
describes the test properly.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/798>
2020-10-31 02:50:51 +00:00
Jan Schmidt
35cc0df53d splitmuxsink: Change EOS catching logic.
Add a new state for ending the overall stream, and use it to decide
whether to pass the final EOS message up the bus instead of dropping
it. Fixes a small race that makes the testsuite sometimes not generate
the last fragment(s) sometimes because the wrong EOS gets
allowed through too early.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/798>
2020-10-31 02:50:51 +00:00
Jan Schmidt
d12fa00195 splitmuxsink: Don't use the element state lock
Using the element state lock to avoid splitmuxsink shutting
down while doing element manipulations can lead to a deadlock on
shutdown if a fragment switch happens at exactly the wrong moment.

Use a private mutex and a shutdown boolean instead.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/798>
2020-10-31 02:50:51 +00:00
Jan Schmidt
41ca3b4e43 splitmuxsink: Don't busy loop on a non-ready pad.
If a pad gets into the check_completed_gop method and then
the underlying conditions change on the reference context,
things could get stuck in a busy loop when the context should
instead jump back out and wait for more data.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/798>
2020-10-31 02:50:51 +00:00
Jan Schmidt
5ac4fdeb7a splitmuxsrc: Mark running=false on shutdown.
Make sure that any late gst_element_call_async() callbacks
know that the elements is shutting down and bail out instead
of operating on the element we're trying to stop.

Fixes a spurious test failure in elements_splitmuxsrc

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/798>
2020-10-31 02:50:51 +00:00
Jan Schmidt
f0c24319de splitmuxsink: Forward EOS messages from async fragments.
Re-enable forwarding EOS messages from fragments that are completing
asynchronously, so that splitmuxsink itself won't go EOS until they
are complete. This was disabled to work around a bug in core that
is fixed in
https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/683

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/798>
2020-10-31 02:50:50 +00:00
Jan Schmidt
1316dd9c65 splitmuxsink: Never start a new fragment with no reference buffers
If there has been no bytes from the reference stream muxed into
the current fragment, then time can't have advanced, there's no
GOP... this fragment would be broken or empty, so wait for some
data on the reference buffer.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/798>
2020-10-31 02:50:50 +00:00
Jan Schmidt
537e7d873a qtmux: Chain up when releasing pad, and fix some locking.
Release pads by calling up into aggregator so it can do the right
things. Don't clean up the pad until after that.

 Add some missing locks around some accesses to shared pad state.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/797>
2020-10-31 02:01:10 +00:00
Stian Selnes
95579a00c0 rtpvp9depay: Improve SVC parsing, aggregate all layers
- Fix start and end of picture to support multiple layers. Start of
  picture is the first packet of the base layer, while end of picture
  is when the marker bit is set (last packet of the enhancement
  layers).
- All "layers" (aka "frames") of a picture are pushed downstream in a
  single buffer when picture is complete.
- Forgive SID=0 for enhancement layers (invalid, but Chrome and
  Firefox sends it)

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/773>
2020-10-30 17:46:30 +01:00
Stian Selnes
d77fcf251b rtpvp8depay: Send lost events when marker bit is missing
This means the previous frame was incomplete.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/796>
2020-10-30 03:43:19 +01:00
Knut Saastad
fa505867a9 rtpvp9depay: detect incomplete frames and bail out
If a packet with the B bit set arrives but we haven't received
a packet with the marker or E bits set to end the previous frame,
we know the current frame was incomplete.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/795>
2020-10-30 01:31:19 +00:00
Knut Saastad
b22514d469 rtpvp9depay: detect incomplete frames and bail out
If a packet with the B bit set arrives but we haven't received
a packet with the marker or E bits set to end the previous frame,
we know the current frame was incomplete.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/769>
2020-10-29 19:56:07 +01:00
Mikhail Fludkov
346b077ae0 rtpvp*depay: possibly forward might-have-been-fec PacketLost events
This is ad adaptation of a Pexip patch for dealing with spurious
GstRTPPacketLost events caused by lost ulpfec packets: as FEC packets
under that scheme are spliced in the same sequence domain as the media
packets, it is not generally possible to determine whether a lost packet
was a FEC packet or a media packet.

When upstreaming pexip's ulpfec patches, we decided to drop all lost
events at the base depayloader level, and where the original patch
from pexip was making use of picture ids and marker bits to determine
whether a packet should be forwarded, this patch makes use of those
to determine whether they should be dropped instead (by removing their
might-have-been-fec field).

Spurious lost events coming out of the depayloader can cause the
decoder to stop decoding until the next keyframe and / or request a new
keyframe, and while this is not desirable it makes sense to forward
that information when we have other means to determine whether a lost
packet was indeed a FEC packet, as is the case with VP8 / VP9 payloads
when they carry a picture id.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/769>
2020-10-29 19:56:07 +01:00
Jan Schmidt
b066441e21 rtph264depay: Preserve SPS/PPS arrival order.
Even if SPS/PPS haven't changed, make sure to move them to the
end of the tracking array if needed, so we always know what the
most recent entries are, in case we need to discard the oldest
when generating codec_data.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/775>
2020-10-29 14:09:21 +00:00
Jan Schmidt
2623404744 rtph264depay: Warn when max SPS/PPS are collected in AVC mode.
The AVC codec_data has a flaw that it can only accomodate
31 SPS headers, even though H.264 can have 32, and 255 PPS,
when there can be 256 in H.264. When streaming RTP some
clients like to cycle through SPS/PPS ids when changing
configuration and can eventually accumulate a full set.

In that case, we have no choice but to discard one (oldest)
entry, or else the count written into the codec_data is wrong
and downstream decoding failures ensue.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/775>
2020-10-29 14:09:21 +00:00
Havard Graff
63c7a9ae43 rtpjitterbuffer: don't send multiple instant RTX for the same packet
Due to us not properly acknowleding the time when the last RTX was sent
when scheduling a new one, it can easily happen that due to the packet
you are requesting have a PTS that is slightly old (but not too old when
adding the latency of the jitterbuffer), both its calculated second and
third (etc.) timeout could already have passed. This would lead to a burst
of RTX requests, which acts completely against its purpose, potentially
spending a lot more bandwidth than needed.

This has been properly reproduced in the test:
test_rtx_not_bursting_requests

The good news is that slightly re-thinking the logic concerning
re-requesting RTX, made it a lot simpler to understand, and allows us
to remove two members of the RtpTimer which no longer serves any purpose
due to the refactoring. If desirable the whole "delay" concept can actually
be removed completely from the timers, and simply just added to the timeout
by the caller of the API. But that can be a change for a another time.

The only external change (other than the improved behavior around bursting
RTX) is that the "delay" field now stricly represents the delay between
the PTS of the RTX-requested packet and the time it is requested on,
whereas before this calculation was more about the theoretical calculated
delay. This is visible in three other RTX-tests where the delay had
to be adjusted slightly. I am confident however that this change is
correct.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/789>
2020-10-28 01:22:24 +01:00
Jan Schmidt
0e84f42055 matroska-mux: Fix sparse stream crash
https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/656
introduced an invalid memory access when debug is enabled, by casting
the wrong pointer to a GstCollectPad. Fixing that showed the original
change was incorrect and leads to an infinite loop in the
testsuite. This patch fixes both problems.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/788>
2020-10-28 01:55:06 +11:00
Thibault Saunier
e0b708ea4a vpx: Fix the check to unfixed/unknown framerate to set bitrate
0/1 means unknown framerate not X/0 (which is illegal).

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/782>
2020-10-22 15:53:24 -03:00
Arun Raghavan
b4a713ff2d rtputils: Count metas with an empty tag list for copying/keeping
The GstMetaInfos registered in core do not set their tags to NULL, but
instead use an empty list (non-NULL list with a single NULL value).
Let's check explicitly for that so as to not miss some metas.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/779>
2020-10-22 09:19:53 -04:00
Bastien Reboulet
27ecd2c30d qmlglsink: fix crash when created/destroyed in quick succession
The crash is caused by a race condition where the render thread
calls a method on the QtGLVideoItem instance that was
previously destroyed by the main thread.
Also, less frequently, QtGLVideoItem::onSceneGraphInitialized
is called when QQuickItem::window is null, also causing a crash.

Fixes #798

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/777>
2020-10-20 18:12:20 +00:00