Havard Graff [Fri, 13 Nov 2020 20:25:42 +0000 (21:25 +0100)]
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>
Havard Graff [Fri, 13 Nov 2020 11:39:53 +0000 (12:39 +0100)]
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>
Vivia Nikolaidou [Fri, 13 Nov 2020 12:58:44 +0000 (14:58 +0200)]
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>
Tim-Philipp Müller [Thu, 12 Nov 2020 23:39:21 +0000 (23:39 +0000)]
tests: qtdemux: fix typo in caps field
timesacle -> timescale
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/815>
Tim-Philipp Müller [Thu, 12 Nov 2020 23:38:21 +0000 (23:38 +0000)]
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>
Sanchayan Maity [Mon, 14 Sep 2020 07:42:50 +0000 (13:12 +0530)]
rtp: ldacpay: Add LDAC RTP payloader
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/757>
Sebastian Dröge [Tue, 3 Nov 2020 13:58:30 +0000 (15:58 +0200)]
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>
ChrisDuncanAnyvision [Tue, 10 Nov 2020 18:18:12 +0000 (18:18 +0000)]
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>
ChrisDuncanAnyvision [Tue, 10 Nov 2020 16:17:23 +0000 (16:17 +0000)]
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>
Nirbheek Chauhan [Wed, 4 Nov 2020 13:13:04 +0000 (18:43 +0530)]
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>
Olivier Crête [Fri, 16 Oct 2020 01:42:40 +0000 (21:42 -0400)]
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>
Guillaume Desmottes [Thu, 29 Oct 2020 14:58:38 +0000 (15:58 +0100)]
docs: update plugins cache
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/530>
Guillaume Desmottes [Fri, 20 Mar 2020 12:15:33 +0000 (13:15 +0100)]
rtp: add rtpisacdepay
Depayload for the iSAC audio codec.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/530>
Guillaume Desmottes [Fri, 20 Mar 2020 12:15:33 +0000 (13:15 +0100)]
rtp: add rtpisacpay
Payload for the iSAC audio codec.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/530>
Dinesh Manajipet [Sun, 1 Nov 2020 18:36:49 +0000 (18:36 +0000)]
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>
Sebastian Dröge [Sun, 1 Nov 2020 08:30:27 +0000 (10:30 +0200)]
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>
Matthew Waters [Mon, 26 Oct 2020 01:40:49 +0000 (12:40 +1100)]
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>
Stéphane Cerveau [Thu, 29 Oct 2020 13:54:16 +0000 (14:54 +0100)]
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>
Jan Schmidt [Sat, 31 Oct 2020 01:52:04 +0000 (12:52 +1100)]
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>
Jan Schmidt [Sat, 31 Oct 2020 01:49:08 +0000 (12:49 +1100)]
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>
Jan Schmidt [Fri, 30 Oct 2020 15:19:07 +0000 (02:19 +1100)]
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>
Jan Schmidt [Thu, 29 Oct 2020 16:38:15 +0000 (03:38 +1100)]
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>
Jan Schmidt [Thu, 29 Oct 2020 16:36:51 +0000 (03:36 +1100)]
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>
Jan Schmidt [Wed, 28 Oct 2020 15:36:35 +0000 (02:36 +1100)]
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>
Jan Schmidt [Thu, 17 Sep 2020 12:56:01 +0000 (22:56 +1000)]
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>
Jan Schmidt [Wed, 28 Oct 2020 15:38:16 +0000 (02:38 +1100)]
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>
Stian Selnes [Mon, 13 Aug 2018 13:35:11 +0000 (15:35 +0200)]
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>
Stian Selnes [Fri, 30 Oct 2020 02:09:48 +0000 (03:09 +0100)]
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>
Knut Saastad [Wed, 14 Oct 2020 21:17:53 +0000 (23:17 +0200)]
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>
Knut Saastad [Wed, 14 Oct 2020 21:17:53 +0000 (23:17 +0200)]
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>
Mikhail Fludkov [Tue, 13 Oct 2020 23:28:50 +0000 (01:28 +0200)]
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>
Jan Schmidt [Tue, 20 Oct 2020 12:22:36 +0000 (23:22 +1100)]
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>
Jan Schmidt [Fri, 16 Oct 2020 13:05:15 +0000 (00:05 +1100)]
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>
Havard Graff [Tue, 27 Oct 2020 23:29:05 +0000 (00:29 +0100)]
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>
Jan Schmidt [Tue, 27 Oct 2020 12:43:49 +0000 (23:43 +1100)]
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>
Thibault Saunier [Thu, 22 Oct 2020 18:29:01 +0000 (15:29 -0300)]
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>
Arun Raghavan [Thu, 22 Oct 2020 13:17:26 +0000 (09:17 -0400)]
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>
Bastien Reboulet [Fri, 16 Oct 2020 23:05:45 +0000 (16:05 -0700)]
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>
Sebastian Dröge [Mon, 19 Oct 2020 15:23:25 +0000 (18:23 +0300)]
v4l2codec: Garbage collect old frames if they accumulate because of codec bugs
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/778>
Sebastian Dröge [Mon, 19 Oct 2020 14:56:04 +0000 (17:56 +0300)]
v4l2codec: Pass system frame number as timestamp and use it to retrieve back frames reliably
System frame numbers are supposed to be unique and correct drivers are
passing through timestamps without modification from the output/sink to the
capture/src side.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/778>
Nicolas Dufresne [Thu, 24 Sep 2020 17:13:00 +0000 (13:13 -0400)]
rtpbin: Add clear-ssrc action
This action signal will delegate to clear-ssrc onto the rtpssrcdemux element
associated with the session. This allow rtpbin users to clear pads and
elements for a specific ssrc that is known to no longer be in use. This
happens when a pad is reused in rtpsrc or ristsrc.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/736>
John-Mark Bell [Fri, 8 Sep 2017 19:02:13 +0000 (20:02 +0100)]
rtpvp8pay: payload temporally scaled bitstreams.
Co-Authored-By: Vincent Sanders <vince@pexip.com>
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/728>
Stian Selnes [Fri, 17 Nov 2017 14:11:41 +0000 (15:11 +0100)]
rtpvp8pay: Add picture-id-offset property
Add property to set the initial value for picture-id. RFC7741 says
that picture-id MAY be initialized to a random value, thus it's also
valid to simply set it to a fixed initial value. A fixed value is very
useful for testing.
Default behavior is not changed.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/728>
Mikhail Fludkov [Thu, 16 Mar 2017 14:23:28 +0000 (15:23 +0100)]
rtpvp8pay: move duplicate code to separate functions
Two new functions to modify picture id:
gst_rtp_vp8_pay_picture_id_reset - picks random picture id of
appropriate bitsize
gst_rtp_vp8_pay_picture_id_increment - increments picture id taking
care of wrapping
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/728>
John-Mark Bell [Fri, 8 Sep 2017 07:13:05 +0000 (08:13 +0100)]
vp8enc: expect bps for temporal-scalability-target-bitrate.
Consistency with target-bitrate is less surprising and with
modern libvpx additional configuration is required to make
temporal scaling work.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/728>
John-Mark Bell [Fri, 8 Sep 2017 07:19:20 +0000 (08:19 +0100)]
vp8enc: finish support for temporally scaled encoding
- introduce two new properties:
* temporal-scalability-layer-flags:
Provide fine-grained control of layer encoding to the
outside world. The flags sequence should be a multiple of
the periodicity and is indexed by a running count of encoded
frames modulo the sequence length.
* temporal-scalability-layer-sync-flags:
Specify the pattern of inter-layer synchronisation (i.e.
which of the frames generated by the layer encoding
specification represent an inter-layer synchronisation).
There must be one entry per entry in
temporal-scalability-layer-flags.
- apply temporal scalability settings and expose as buffer
metadata.
This allows the codec to allocate a given frame to the correct
internal bitrate allocator. Additionally, all the
non-bitstream metadata needed to payload a temporally scaled
stream is now attached to each output buffer as a
GstVideoVP8Meta.
- add unit test for temporally scaled encoding.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/728>
Stéphane Cerveau [Thu, 15 Oct 2020 16:21:54 +0000 (18:21 +0200)]
meson: update glib minimum version to 2.56
In order to support the symbol g_enum_to_string in various
project using GStreamer ( gst-validate etc.), the glib minimum
version should be 2.56.0.
Remove compat code as glib requirement
is now > 2.56
Version used by Ubuntu 18.04 LTS
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/774>
Mathieu Duponchelle [Wed, 14 Oct 2020 12:30:34 +0000 (14:30 +0200)]
rtpst2022-1-fecenc: fix input seqnum check
We need to cast the incremented last seqnum to guint16 for
consistent checks on wraparound
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/770>
Jan Alexander Steffens (heftig) [Sat, 12 Sep 2020 07:02:30 +0000 (09:02 +0200)]
flvmux: Correct time types
- last_dts is in milliseconds, not nanoseconds as expected for
GstClockTime. Make it a generic guint64.
- Use GstClockTime for the fields that actually contain nanoseconds.
None of them should become negative.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/766>
Sebastian Dröge [Fri, 9 Oct 2020 06:31:27 +0000 (09:31 +0300)]
rtpst2022-1-fecenc: Don't unconditionally use GLib 2.60 APIs
g_queue_clear_full() in this case.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/768>
Mathieu Duponchelle [Thu, 8 Oct 2020 16:54:55 +0000 (18:54 +0200)]
rtpulpfec: fix potential alignment issue in xor function
https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/753#note_646453
for context
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/753>
Mathieu Duponchelle [Tue, 6 Oct 2020 01:03:13 +0000 (03:03 +0200)]
rtpmanager: implement SMPTE 2022-1 FEC encoder
+ improve integration of FEC encoders in rtpbin
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/753>
Mathieu Duponchelle [Tue, 6 Oct 2020 01:13:30 +0000 (03:13 +0200)]
rtpmanager: implement SMPTE 2022-1 FEC decoder
+ improve integration of FEC decoders in rtpbin
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/753>
Olivier Crête [Wed, 8 Jul 2020 21:28:31 +0000 (17:28 -0400)]
rtpfunnel: Also forward custom sticky event
This is useful to track metadata about each group of packets
Also include a unit test
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/666>
Thibault Saunier [Tue, 29 Sep 2020 12:44:54 +0000 (09:44 -0300)]
isomp4: Rename GstQTMux to GstBaseQTMux to avoid breaking API
Since
52b63de19ada283c1180c8fc00cacb1465fdf10f the qtmux GType was
renamed GstQTMuxElement which breaks presets, revert that change.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/755>
Sebastian Dröge [Mon, 28 Sep 2020 15:25:21 +0000 (18:25 +0300)]
rtp: Fix allocations to support source-info property
Use gst_rtp_base_payload_allocate_output_buffer() instead of
gst_rtp_buffer_new_allocate() in order to allocate RTP buffer with
correct number of CSRCs according to the meta.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/612>
Stian Selnes [Fri, 23 Oct 2015 09:08:56 +0000 (11:08 +0200)]
rtpvp8pay: Fix allocation to support source-info property
Use gst_rtp_base_payload_allocate_output_buffer() in order to allocate
RTP buffer with correct number of CSRCs according to the meta.
Fixes https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/issues/314
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/612>
Matthew Waters [Mon, 28 Sep 2020 05:36:00 +0000 (15:36 +1000)]
qtmux: output the correct limits in error messages
Having the current bytes being less than the limit was confusing!
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/732>
Matthew Waters [Fri, 31 Jul 2020 06:47:37 +0000 (16:47 +1000)]
qtmux: properly support initial caps nego failure
Scenario:
- gap event causes h264parse to push made up caps that may fail checks
inside qtmux (e.g missing codec_data).
- the caps event has already been marked as received and is sticky on
the sink pad
- gst_qt_mux_pad_can_renegotiate() will retrieve the failed caps event
using gst_pad_get_current_caps() and reject the correct updated caps
with codec_data.
- Failure!
Keep track of the configured caps ourselves instead of relying on the
sticky event on the pad.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/732>
Matthew Waters [Wed, 22 Jul 2020 05:34:44 +0000 (15:34 +1000)]
qtmux: support non-seekable downstream mode
Write an mdat per buffer in that case.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/732>
Nicolas Dufresne [Wed, 23 Sep 2020 19:25:36 +0000 (15:25 -0400)]
rtpbin: Remove the rtpjitterbuffer with the stream
Since !348, the jitterbuffer was only removed with the session. This restores
the original behaviour and removes the jitterbuffer when the stream is
removed. This avoid accumulating jitterbuffer objects into the bin when a
session is reused.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/735>
Nicolas Dufresne [Wed, 23 Sep 2020 17:26:51 +0000 (13:26 -0400)]
rtpbin: Cleanup dead code
The rtpjitterbuffer is now part of the session elements, we no longer need
to do the ref_sink dance when signalling it. It is already owned by the bin
when signalled. Also, the code that handles generic session elements already
handle the ref_sink() calls since:
03dc22951bacb6fdc3868c8f801e6a52c33a745f
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/735>
Matthew Waters [Fri, 18 Sep 2020 06:09:20 +0000 (16:09 +1000)]
rtph26*depay: drop FU's without a corresponding start bit
If we have not received a FU with a start bit set, any subsequent FU
data is not useful at all and would result in an invalid stream.
This case is constructed from multiple requirements in
RFC 3984 Section 5.8 and RFC 7798 Section 4.4.3. Following are excerpts
from RFC 3984 but RFC 7798 contains similar language.
The FU in a single FU case is forbidden:
A fragmented NAL unit MUST NOT be transmitted in one FU; i.e., the
Start bit and End bit MUST NOT both be set to one in the same FU
header.
and dropping is possible:
If a fragmentation unit is lost, the receiver SHOULD discard all
following fragmentation units in transmission order corresponding to
the same fragmented NAL unit.
The jump in seqnum case is supported by this from the specification
instead of implementing the forbidden_zero_bit mangling:
If a fragmentation unit is lost, the receiver SHOULD discard all
following fragmentation units in transmission order corresponding to
the same fragmented NAL unit.
A receiver in an endpoint or in a MANE MAY aggregate the first n-1
fragments of a NAL unit to an (incomplete) NAL unit, even if fragment
n of that NAL unit is not received. In this case, the
forbidden_zero_bit of the NAL unit MUST be set to one to indicate a
syntax violation.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/730>
Seungha Yang [Sun, 20 Sep 2020 12:06:19 +0000 (21:06 +0900)]
imagefreeze: Response caps query from srcpad
... and chain up to default query handler for unhandled query types.
Unhandled query shouldn't be returned with FALSE if there's no special needs.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/731>
Matthew Waters [Wed, 16 Sep 2020 02:15:09 +0000 (12:15 +1000)]
qtmux: make documentation happy
introduce a base qtmux class that we can install documentation snippets
on instead of duplicating across alll the isomp4 elements
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/643>
Matthew Waters [Thu, 28 May 2020 09:40:24 +0000 (19:40 +1000)]
isomp4/mux: add a fragment mode for initial moov with data
Used by some proprietary software for their fragmented files.
Adds some support for multi-stream fragmented files
Flow is as follows.
1. The first 'fragment' is written as a self-contained fragmented
mdat+moov complete with an edit list and durations, tags, etc.
2. Subsequent fragments are written with a mdat+moof and each stream is
interleaved as data arrives (currently ignoring the interleave-*
properties). data-offsets in both the traf and the trun ensure
data is read from the correct place on demuxing. Data/chunk offsets
are also kept for writing out the final moov.
3. On finalisation, the initial moov is invalidated to a hoov and the
size of the first mdat is extended to cover the entire file contents.
Then a moov is written as regularly would in moov-at-end mode (the
default).
This results in a file that is playable throughout while leaving a
finalised file on completion for players that do not understand
fragmented mp4.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/643>
Matthew Waters [Thu, 25 Jun 2020 06:37:56 +0000 (16:37 +1000)]
qtdemux: increase some logging on streams and sample parsing
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/643>
Matthew Waters [Thu, 25 Jun 2020 06:35:45 +0000 (16:35 +1000)]
qtdemux: bail out when encountering an atom with a size of 0
A size 0 atom means the atom extends to the end of the file. No further
valid atoms will ever follow. Avoids a subsequent scan for an atom from
one byte earlier after encountering a size 0 atom.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/643>
Matthew Waters [Thu, 25 Jun 2020 06:33:04 +0000 (16:33 +1000)]
qtdemux: fix subsequent moof parsing after moov with valid samples
reset the moof_offset back to its original value like is done in the
error case just before.
Fixes subsequent parsing of a moof following a moov that contains valid
samples in a non-streaming fragmented mp4.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/643>
Matthew Waters [Thu, 25 Jun 2020 06:30:28 +0000 (16:30 +1000)]
qtdemux: extend edit list when fragmented
When we are fragmented, the edit list may only refer to the portion of
the media that is in the moov. Extend the edit list stop time when we
if there is only one qt segment and we are reading a fragmented file.
Fixes playback of some fragmented mp4 files generated by proprietary
programs.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/643>
Nicolas Dufresne [Tue, 15 Sep 2020 18:22:13 +0000 (14:22 -0400)]
meson: Allow overriding qt5 feature
This will allow controlling that feature from gst-build
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/729>
Olivier Crête [Wed, 18 Nov 2015 00:14:01 +0000 (19:14 -0500)]
splitmuxsrc: Implement segment query
Fixes #239
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/713>
Sebastian Dröge [Mon, 14 Sep 2020 07:15:35 +0000 (10:15 +0300)]
rtpmp4gdepay: Allow lower-case "aac-hbr" instead of correct "AAC-hbr"
Various live555 based products are using the wrong "mode" string or
seem to assume case-insensitive matching, which is wrong.
Examples for this are the Yuan SC6C0N1 mini and the Kiloview E2.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/727>
Stefan Brüns [Sat, 2 May 2020 00:21:00 +0000 (02:21 +0200)]
qtdemux: Add support for AAX encrypted audio streams
This is modelled after the DASH Common Encryption scheme, but is somewhat
simpler as more parts are fixed, i.e. just one encryption scheme.
The output caps are fixed to 'application/x-aavd'. All information
required for decryption are part of the 'adrm' atom, which is passed
on as a property. The property is attached to the buffer.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/577>
Stefan Brüns [Sat, 2 May 2020 00:20:44 +0000 (02:20 +0200)]
qtdemux: Add 'aavd' and related fourcc codes for AAX encrypted audio
The 'aavd' box is contained in the 'stsd' sample description. The 'aavd'
box follows the layout of an 'mp4a' entry, i.e. it contains a single
standard 'esds' extension box, and the two proprietary 'adrm' and 'aabd'
extension boxes.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/577>
Haakon Sporsheim [Mon, 23 Jun 2014 06:46:37 +0000 (08:46 +0200)]
vpxdec: request a sync point on decoder errors
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/725>
Camilo Celis Guzman [Sun, 13 Sep 2020 16:31:57 +0000 (18:31 +0200)]
rtp/vrawpay: use alloc_output_buffer from base class
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/726>
Ricky Tang [Mon, 7 Sep 2020 15:20:58 +0000 (23:20 +0800)]
rtspsrc: Fix push-backchannel-buffer parameter mismatch
When using python, signal parameter must match with function.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/724>
Jérôme Laheurte [Thu, 10 Sep 2020 09:24:32 +0000 (11:24 +0200)]
jpegdec: check buffer size before dereferencing. Fixes #541
Some cameras (Panacast) have buggy drivers/firmware which send
invalid JPEG frames, containing no data, which makes jpegdec
crash because it assumes the frame is at least 2 bytes long.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/723>
Jan Alexander Steffens (heftig) [Thu, 10 Sep 2020 09:11:00 +0000 (11:11 +0200)]
flvmux: Improve logging of gst_flv_mux_buffer_to_tag_internal
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/722>
Jan Alexander Steffens (heftig) [Wed, 9 Sep 2020 13:12:53 +0000 (15:12 +0200)]
flvmux: Move stream skipping to GstAggregatorPadClass.skip_buffer
Besides looking like the correct place to put this, it allows us to drop
the entire aggregator queue. The old implementation only dropped at most
one buffer for each call of aggregate.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/722>
Havard Graff [Tue, 8 Sep 2020 15:35:50 +0000 (17:35 +0200)]
v4l2object: plug memory-leak
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/718>
Mathieu Duponchelle [Fri, 28 Aug 2020 16:09:15 +0000 (18:09 +0200)]
vp9enc: expose row-mt property
With recent libvpx versions, multithreading can be enabled on
a per-tile basis, instead of on a per tile-column basis.
In combination with the new tile-rows property, this allows the
encoder to make much better use of the available CPU power.
Bump minimum libvpx version to 1.7.0
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/707>
Mathieu Duponchelle [Fri, 28 Aug 2020 15:45:48 +0000 (17:45 +0200)]
vpxenc: change default for deadline to good quality
Having the deadline set to best quality causes the encoder
to be absurdly slow, most real-life users will want the good
quality tradeoff instead.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/707>
Mathieu Duponchelle [Fri, 28 Aug 2020 15:39:47 +0000 (17:39 +0200)]
vp9enc: expose tile-columns and tile-rows properties
Based on patch by Stian Selnes <stian@pexip.com>.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/707>
Mathieu Duponchelle [Fri, 28 Aug 2020 15:35:26 +0000 (17:35 +0200)]
vpxenc: add configure_encoder virtual method
For subclasses to expose format-specific properties
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/707>
Mathieu Duponchelle [Tue, 8 Sep 2020 18:57:33 +0000 (20:57 +0200)]
splitmuxsink: fix sink pad release while PLAYING
- Release the split mux lock while removing the probes
- Flush the sinkpad to unblock other pads
- Turn check_completed_gop into a do while statement, when
waking up we want to recheck whether the current GOP is
ready for sending
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/719>
John-Mark Bell [Tue, 31 Oct 2017 09:40:33 +0000 (09:40 +0000)]
vp8enc: improve unit tests
- make test_encode_simple cope with libvpx built with
CONFIG_REALTIME_ONLY. Sadly, there's no way to detect this at
runtime beyond trying to set lag-in-frames to >0, pushing a
buffer and catching the GST_FLOW_NOT_NEGOTIATED return.
- fix bitrot in test_encode_simple_when_bitrate_set_to_zero.
- port test_encode_simple to GstHarness and introduce a separate
test for the lag-in-frames property.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/708>
Jakub Adam [Fri, 21 Aug 2020 14:03:09 +0000 (16:03 +0200)]
docs: Update plugin cache
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/705>
Jakub Adam [Tue, 24 Mar 2020 18:35:07 +0000 (19:35 +0100)]
vpx: Support GST_VIDEO_FORMAT_I422_10LE
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/705>
Jakub Adam [Tue, 24 Mar 2020 16:16:59 +0000 (17:16 +0100)]
vpx: Support GST_VIDEO_FORMAT_I420_10LE
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/705>
Jakub Adam [Mon, 23 Mar 2020 20:44:30 +0000 (21:44 +0100)]
vp9enc: support GST_VIDEO_FORMAT_Y444
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/705>
Tim-Philipp Müller [Tue, 8 Sep 2020 16:30:35 +0000 (17:30 +0100)]
ci: include template from gst-ci master branch again
Tim-Philipp Müller [Tue, 8 Sep 2020 15:58:37 +0000 (16:58 +0100)]
Back to development
Tim-Philipp Müller [Mon, 7 Sep 2020 23:05:14 +0000 (00:05 +0100)]
Release 1.18.0
Tim-Philipp Müller [Mon, 7 Sep 2020 21:39:02 +0000 (22:39 +0100)]
meson: dist pot file in tarballs
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/716>
Sebastian Dröge [Mon, 7 Sep 2020 09:13:18 +0000 (12:13 +0300)]
gst: Update for gst_video_transfer_function_*() function renaming
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/715>
Jan Alexander Steffens (heftig) [Mon, 31 Aug 2020 13:01:32 +0000 (15:01 +0200)]
flvmux: Avoid crash when best pad gets flushed
The 'best' pad might receive a flush event between us picking it and us
popping the buffer. In this case, the buffer will be missing.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/711>
Jan Alexander Steffens (heftig) [Mon, 31 Aug 2020 11:43:42 +0000 (13:43 +0200)]
flvmux: Correct breaks in gst_flv_mux_find_best_pad
The code seems to use `continue` and `break` as if both refer to the
surrounding `while` loop. But because `break` breaks out of the
`switch`, they actually have the same effect.
This may have caused the loop not to terminate when it should. E.g. when
`skip_backwards_streams` drops a buffer we should abort the aggregation
and wait for all pads to be filled again. Instead, we might have just
selected a subsequent pad as our new "best".
Replace `break` with `done = TRUE; break`, and `continue` with `break`.
Then simplify the code a bit.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/710>
Dmitriy Purgin [Wed, 13 May 2020 09:31:38 +0000 (11:31 +0200)]
gstqmlgl: build on Windows with qmake without pkgconfig; update instructions on building for Windows
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/591>