Commit graph

119302 commits

Author SHA1 Message Date
Sebastian Dröge
121e52886b videoparsers: Don't verbosely warn about CEA_708_PROCESS_EM_DATA_FLAG not being set
And the same for CEA_708_PROCESS_CC_DATA_FLAG. This is not really a
problem and was polluting logs with warnings for every single frame.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6336>
2024-03-12 21:26:18 +00:00
L. E. Segovia
71510860af meson: Require tinyalsa >= 1.1.0 when building its plugin
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6311>
2024-03-12 20:30:11 +00:00
L. E. Segovia
9c8549c31c tinyalsasink: Fix missing const and deprecations with tinyalsa v2
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6311>
2024-03-12 20:30:11 +00:00
Piotr Brzeziński
e9802f5f41 macos: Add Apple AAC encoder (atenc)
Adds the `atenc` element capable of encoding AAC-LC audio, using the AudioToolbox framework.
It's able to encode up to 7.1 channel configurations.
Comes with basic knobs for rate control (bitrate for CBR, quality for VBR).

Support for more profiles (LD, HE-AAC) should be simple, but is not included here because of bugs
with parsing of the AudioSpecificConfig.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6254>
2024-03-12 19:50:06 +00:00
Nicolas Dufresne
bcad005d05 glupload: Do not propose allocators with sysmem
None of the GL allocators actually offer a generic alloc() implementation. As a
side effect, they cannot be offered as they don't work with generic video
buffer pool.

Our specialized buffer pool can be dropped by tee or alphacombine as sharing the
same buffer pool over two branch is not supported by the pool API.

Fixes #3372

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6327>
2024-03-12 19:02:54 +00:00
Seungha Yang
c9aaf39279 cuda,d3d11,d3d12bufferpool: Disable preallocation
Do not chain up to parent's GstBufferPool::start() which will do
preallocation. We don't want it to be preallocated
since there are various cases where negotiated downstream buffer pool is
not used at all (e.g., zero-copy decoding, IPC elements).

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6326>
2024-03-12 18:07:29 +00:00
Antonio Larrosa
edb7b787d8 gitlint: Allow curly brackets in commit prefix
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6319>
2024-03-12 16:58:07 +00:00
Antonio Larrosa
7b8fa42f8a va{h264,h265,av1}enc: fix potential crash on devices without rate control
This fixes a crash in `gst_va_h264_enc_class_init` and `gst_va_h265_enc_class_init`
(and probably also in gst_va_av1_enc_class_init) when calling
`g_object_class_install_properties (object_class, n_props, properties);`

When rate_control_type is 0, the following code is executed in :

```
  } else {
    n_props--;
    properties[PROP_RATE_CONTROL] = NULL;
  }
```

n_props has initially a value of N_PROPERTIES but PROP_RATE_CONTROL
is not the last element in the array, so it's making
g_object_class_install_properties fail to iterate over the
properties array.

This applies the same fix to gstvah264enc.c, gstvah265enc.c and
gstvaav1enc.c.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6319>
2024-03-12 16:58:07 +00:00
Antonio Larrosa
bd97973ce0 registry, ptp: Canonicalize the library path returned by dladdr
On systems using UsrMerge (like openSUSE or Fedora), /lib64 is
a symlink to /usr/lib64. So dladdr is returning the path to
the gstreamer library in /lib64 in priv_gst_get_relocated_libgstreamer.
Later gst_plugin_loader_spawn tries to build the path to the
gst-plugin-scanner helper from /lib64 and ends up trying to use
/lib64/../libexec/gstreamer-1.0/gst-plugin-scanner which doesn't exist.

By canonicalizing the path with a call to realpath, gst-plugin-scanner
is found correctly under
/usr/lib64/../libexec/gstreamer-1.0/gst-plugin-scanner

Similar change applied to gstreamer/libs/gst/net/gstptpclock.c

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6322>
2024-03-12 15:51:36 +00:00
Nirbheek Chauhan
77831d6142 gsturi: Sort by feature name to break a feature rank tie
This matches autoplug in other places such as decodebin, otherwise we
will pick "randomly" based on the order in which plugins are
registered, which is mostly dependent on the order in which readdir()
returns items.

So let's make it predictable.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6227>
2024-03-12 14:25:10 +00:00
Jurijs Satcs
6a9bf8592a mpegtsmux: allow to disable SCTE NULL by setting interval to 0
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6284>
2024-03-12 11:15:58 +00:00
Piotr Brzeziński
d3fba31da0 macos: Move atdec from applemedia (-bad) to osxaudio (-good)
osxaudio has a few helper methods potentially useful in atdec (or future atenc), like GStreamer -> CoreAudio
channel mapping. Doesn't make sense to duplicate them in applemedia, and atdec is the only audio-oriented
element there anyway.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6223>
2024-03-12 09:55:10 +00:00
Matthew Waters
a26b363d3e closedcaption: produce valid cea608 padding by default
Cea608 (valid) padding removal is available on the input side of ccconverter
or configurable on cccombiner.  cccombiner can now configure whether
valid or invalid cea608 padding is used and for valid padding, how long
after valid non-padding to keep sending valid padding.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6300>
2024-03-12 02:49:01 +00:00
Piotr Brzeziński
3243c5fe94 audiovisualizer: Don't wrap temporary memory in buffers
Avoids potentially ending up with the buffermemory pointing to already-freed or reused addresses.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6320>
2024-03-11 18:18:01 +00:00
Piotr Brzeziński
9c084faa75 qtdemux: Fix wrapping temporary memory in buffers
That memory can disappear at any moment, doesn't cost much to just copy those few bytes.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6320>
2024-03-11 18:18:01 +00:00
Piotr Brzeziński
15e0affc98 audioencoder: Avoid wrapping temporarily mapped memory with a GstBuffer and passing that to subclass
Memory from gst_adapter_map() could live shorter than the GstMemory that the GstBuffer wraps around it, which in lucky
cases 'just' caused a re-use of the same memory for multiple (potentially still in use!) input buffers, but could easily
end up pointing to an already-freed memory.

Manifested when an AudioToolbox encoder kept getting silence inserted in seemingly random circumstances, turned out
to be the memory being re-used by GStreamer at the same time that the AT API was processing it...

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6320>
2024-03-11 18:18:01 +00:00
Edward Hervey
0f1dfc2db0 playbin3: Remove un-needed URI NULL check
This will mimic the playbin2 behaviour, which sets the "next" entry to be
NULL.

The biggest impact this has is that when going back to READY the current play
entry will be discarded (instead of being kept around for when you go back to
PAUSED/PLAYING).

Fixes #3371

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6324>
2024-03-11 17:33:04 +00:00
Mikhail Rudenko
05ef1bbc06 rtsp-stream: clear sockets when leaving bin
Since commit 4d86f994, when setting an RTSP media both shared and
reusable, streaming cannot be restarted after the first time all the
clients disconnect. That happens because the sockets (unlike
addresses) of GstRTSPStream are not cleared in
gst_rtsp_stream_leave_bin, and on restart sockets and addresses are
not allocated in gst_rtsp_stream_allocate_udp_sockets, and then the
check in create_sender_part fails. Fix this by clearing sockets in
gst_rtsp_stream_leave_bin.

Fixes gstreamer/gst-rtsp-server#113

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6325>
2024-03-11 18:22:38 +03:00
He Junyan
861c1a44be va: av1enc: Init the output_frame_num when resetting gf group
Fixes: #3359
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6308>
2024-03-11 12:38:57 +00:00
Edward Hervey
5f7062136d decodebin3: Handle race switching on pending streams
find_slot_for_stream_id() will return a slot which has the request stream-id as
active_stream *or* pending_stream (i.e. the slot on which that stream is
currently being outputted or will be outputted).

When figuring out which slot to use (if any) we want to consider stream-id
which *will* appear on a given slot which isn't outputting anything yet the same
way as if we didn't find a slot yet.

Fixes races when doing intensive state changes

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6270>
2024-03-11 11:13:59 +00:00
Edward Hervey
e03e2308d7 decodebin3: Clear select streams seqnum when resetting
At this point there's definitely no pending select streams

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6270>
2024-03-11 11:13:59 +00:00
Edward Hervey
344acfe4e8 decodebin3: Only post collection message on actual updates
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6270>
2024-03-11 11:13:59 +00:00
Edward Hervey
33fe063f50 decodebin3: Clear the global collection when resetting
This avoids having stray collections when re-using decodebin3

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6270>
2024-03-11 11:13:59 +00:00
Edward Hervey
086ecb008f avviddec: Fix how we get back the codec frame
With the new copy_opaque system, the corresponding frame is stored in the
picture opaque ref.

This also handles the case where the "regular" opaque might be empty in the
case of "DECODE_ONLY" frames, since it that field is set in `get_buffer2()`
which might not be called for those frames

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6301>
2024-03-11 10:17:41 +00:00
Edward Hervey
eacd5c1cb1 avviddec: Improve debug statements
Add SFN to better track what is going on

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6301>
2024-03-11 10:17:41 +00:00
Nirbheek Chauhan
3bed35c342 rtspsrc: Don't invoke close when stopping if we've started cleanup
When we're doing a state change from PLAYING to NULL, first we invoke
gst_rtspsrc_loop_send_cmd_and_wait (..., CMD_CLOSE, ...) during
PAUSED_TO_READY which will schedule a TEARDOWN to happen async on the
task thread.

The task thread will call gst_rtspsrc_close(), which will send the
TEARDOWN and once it's complete, it will call gst_rtspsrc_cleanup()
without taking any locks, which frees src->streams.

At the same time however, the state change in the app thread will
progress further and in READY_TO_NULL it will call gst_rtspsrc_stop()
which calls gst_rtspsrc_close() a second time, which accesses
src->streams (without a lock again), which leads to simultaneous
access of src->streams, and a segfault.

So the state change and the cleanup are racing, but they almost always
complete sequentially. Either the cleanup sets src->streams to NULL or
_stop() completes first. Very rarely, _stop() can start while
src->streams is being freed in a for loop. That causes the segfault.

This is unlocked access is unfixable with more locking, it just leads
to deadlocks. This pattern has been observed in rtspsrc a lot: state
changes and cleanup in the element are unfixably racy, and that
foundational issue is being addressed separately via a rewrite.

The bandage fix here is to prevent gst_rtspsrc_stop() from accessing
src->streams after it has already been freed by setting src->state to
INVALID.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6302>
2024-03-11 09:15:50 +00:00
Edward Hervey
73152b53ff decodebin3: Provide clear error message if no decoders present
If we don't do this we will end up with a more cryptic error message (not-linked
error from some upstream component).

Fixes #3198

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6317>
2024-03-11 09:17:09 +01:00
Chris Spencer
1032d58187 vkmemory: invalidate non-coherent memory when mapping for read
Mapping non-coherent memory does not implicitly invalidate the host caches.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6310>
2024-03-11 01:55:44 +00:00
Chris Spencer
9412565221 vulkan/operation: use timeline semaphore fallback if sync2 not supported
gst_vulkan_operation_add_dependency_frame does not fall back to the
timeline semaphore implementation if VK_KHR_synchronization2 is compiled
in, but not supported by the driver.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6309>
2024-03-11 00:58:40 +00:00
Chris Spencer
7701e9ffeb vulkan/operation: add missing unlock
gst_vulkan_operation_add_dependency_frame does not release its lock if
support for VK_KHR_timeline_semaphore/VK_KHR_synchronization2 is compiled
in, but not supported by the driver.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6309>
2024-03-11 00:58:40 +00:00
Jordan Petridis
95bafc4934 rsvg: Add direct dependency on cairo
We include cairo.h in the element so we should also
declare it in meson.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6306>
2024-03-08 19:01:30 +02:00
Seungha Yang
2e1eaaec5e ges: Fix critical warning
GStreamer-CRITICAL **: 20:44:38.256: gst_debug_log_full_valist:
assertion 'category != NULL' failed

Make sure debug category initialized.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6304>
2024-03-08 21:00:51 +09:00
François Laignel
7d5bb1ea7a webrtc: add all SSRC attributes getting CAPS for a PT
The transport stream only returned the CAPS for the first matching PT entry
from the `ptmap`. Other SSRC with the same PT where not included. For a stream
which bundled multiple audio streams for instance, only the first SSRC was
knowed to the SSRC demux and downstream elements.

This commit adds all the `ssrc-` attributes from the matching PT entries.

The RTP jitter buffer can now find the CNAME corresponding its SSRC even if it
was not the first to be registered for a particular PT.

The RTP PT demux removes `ssrc-*` attributes cooresponding to other SSRCs
before pushing SSRC specific CAPS to downstream elements.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6119>
2024-03-08 10:28:15 +00:00
François Laignel
d83184cf9a sdp: accept empty attribute value represented as a NULL pointer
Some empty media attribute values are set to an empty string, others as a NULL
pointer. It seems that code is able to deal with both, except for the UTF8
validation.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6119>
2024-03-08 10:28:15 +00:00
François Laignel
aeaef7a7f0 webrtcbin: RFC5576 - early CNAME support
See RFC5576: have CNAME available to the rtpjitterbuffer before the the first
RTCP SR is received, for rapid synchronization. Similar to what was done for
RTSP (last 2 commits) of [MR 2132].

[RFC5576]: https://www.rfc-editor.org/rfc/rfc5576
[MR 2132]: https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2132

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6119>
2024-03-08 10:28:15 +00:00
Jan Schmidt
d3e79077dc identity: Don't refuse seeks unless single-segment=true
identity only needs to configure the internal seek segment if it's
aggregating upstream segments into 1. If it's not, don't break
other seek behaviour by refusing (for example) instant-rate change
seeks.

Fixes: #3363
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6299>
2024-03-08 17:07:25 +11:00
Michael Tretter
5b3082257e meson: Fix description in qt options
The qt-x11 description contains a copy/paste error from the qt-wayland option.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6292>
2024-03-08 02:14:11 +00:00
Seungha Yang
4db7eb0290 d3d12screencapturesrc: Add support for WGC API
Adding support for window and monitor capturing by using
Windows Graphics Capture API.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6256>
2024-03-08 01:05:24 +09:00
Seungha Yang
63ef405131 d3d12memory: Implement NT handle caching and custom user data support
Same as the d3d11 memory implementation.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6256>
2024-03-08 01:03:28 +09:00
Mathieu Duponchelle
04077ce906 onvif: tests: check for T flag on all packets
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/5173>
2024-03-07 14:02:33 +00:00
Mathieu Duponchelle
519546aea3 rtpgstpay: flush on EOS
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/5173>
2024-03-07 14:02:33 +00:00
Mathieu Duponchelle
0631a59803 rtponviftimestamp: make sure to set E and T bits on last buffer of lists
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/5173>
2024-03-07 14:02:33 +00:00
Sebastian Dröge
b88d69b722 rtpgstpay: Delay pushing of event packets until the next buffer
And also re-timestamp them with the current buffer's PTS.

Not doing so keeps the timestamps of event packets as
GST_CLOCK_TIME_NONE or the timestamp of the previous buffer, both of
which are bogus.

Making sure that (especially) the first packet has a valid timestamp
allows putting e.g. the NTP timestamp RTP header extension on it.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/5173>
2024-03-07 14:02:33 +00:00
Seungha Yang
37578454b9 avviddec: Fix interlaced mode detection
Fixing regression introduced by the commit b46559102b

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6285>
2024-03-07 11:53:07 +00:00
Elizabeth Figura
e2167867d5 qtdemux: Do not set channel-mask to zero
Leave it uninitialized, so that the downstream decoder will initialize it appropriately. Setting it to zero is wrong.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6225>
2024-03-07 12:52:30 +02:00
Jan Schmidt
f53dbb28b2 rtspsrc: Parse Speed/Scale before Range in responses
Parse the speed and scale in the server's response
*before* the range, so that the range start/stop
are swapped (or not swapped) correctly based
on the server's actual chosen values. Otherwise,
the old rate from the segment is used - what the
last seek asked for, but not necessarily what
the server chooses.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6248>
2024-03-06 17:50:53 +00:00
Jan Schmidt
57013e1a7c rtspsrc: Handle queries and events with no manager
When doing direct output with no session manager, we still
want to respond to queries and events from downstream, so
install the handlers

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6248>
2024-03-06 17:50:53 +00:00
Jan Schmidt
4d2f000125 rtspsrc: return NO_PREROLL on PLAYING->PAUSED too
When transitioning back to PAUSED and rtspsrc is live, return
NO_PREROLL so the pipeline knows to skip preroll here too.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6248>
2024-03-06 17:50:53 +00:00
Jan Schmidt
8356bd04a8 rtponviftimestamp: Use gst_segment_to_stream_time_full()
In the situation where playback starts from a keyframe before
the target playback segment, then the first buffers will be
outside the configured segment and gst_segment_to_stream_time()
will return GST_CLOCK_TIME_NONE unconditionally.

If drop-out-of-segment is false, the RTP buffers will not be
dropped, but will be sent witout ONVIF extension timestamps
and given GST_CLOCK_TIME_NONE timestamps on the receiver.

Instead, use gst_segment_to_stream_time_full() to extrapolate
stream time outside the segment so that such buffers still
get assigned their correct timestamps on the receiver.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6248>
2024-03-06 17:50:53 +00:00
Jan Schmidt
4b107b60e7 dvbsubenc: Fix bottom field size calculation
Don't accidentally include the stuffing byte (if present)
into the bottom field size. It should only be included in the
total segment length.

Fixes problems with FFmpeg not rendering the subtitles
with a stuffing byte, giving a "Invalid object location!" error.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6250>
2024-03-06 16:47:38 +00:00