Commit graph

119845 commits

Author SHA1 Message Date
Sebastian Dröge
d263a8d2fe rtspsrc: Handle the case of * as session-wide control URL from the SDP
Just like the comment above says this is supposed to indicate that the
same URL should be used as for the connection so far. If encountering
this case simply do nothing.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6982>
2024-06-01 16:15:53 +03:00
Sebastian Dröge
6f984939c4 rtspsrc: Also handle rtsps:// and similar URLs as absolute in other places
Previously a direct comparison with `rtsp://` was performed, which
didn't catch cases like `rtsps://`.

Fixes https://gitlab.freedesktop.org/gstreamer/gstreamer/-/issues/3563

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6982>
2024-06-01 16:15:53 +03:00
Sebastian Dröge
dfc03b9a2e rtspsrc: Don't try the SETUP workaround for broken servers with absolute control URIs
Previously only control URIs that started with "rtsp://" were ignored
but it makes more sense to ignore all absolute URIs.

gst_uri_is_valid() conveniently checks for exactly that.

See https://gitlab.freedesktop.org/gstreamer/gstreamer/-/issues/3563

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6982>
2024-06-01 16:15:53 +03:00
Martin Nordholts
03b6efcaf5 gst_debug: Add missing gst_debug_log_id_literal() dummy with gst_debug=false
E.g. gst_debug_log_literal() already has a dummy variant.
gst_debug_log_id_literal() is simply missing, which can
cause link errors for project using gstreamer with
gst_debug=false.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6979>
2024-06-01 11:52:32 +03:00
Samuel Thibault
8447c1d386 ptp-helper: Add GNU/Hurd support
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6974>
2024-05-31 11:16:12 +03:00
Seungha Yang
5118e657b6 d3d12memory: Fix staging buffer alignment
Not all GPUs can support arbitrary offset of
D3D12_PLACED_SUBRESOURCE_FOOTPRINT when copying GPU memory between
texture and buffer. Instead of calculating size/offset per plane,
calculate the entire size and offsets at once.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6973>
2024-05-30 16:47:35 +03:00
Jakub Adam
c305fe7a35 glcolorconvert: update existing sync meta if outbuf has one
Instead of always adding a new one, which means the buffer could end up
with multiple sync meta instances.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6962>
2024-05-30 08:35:17 +00:00
Edward Hervey
48f63a9c64 hlsdemux2: Minor refactoring of starting segment check
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6961>
2024-05-30 07:59:25 +00:00
Edward Hervey
421832e506 hlsdemux2: Be more tolerant when matching segments with PDT
Some servers might not provide 100% matching PDT when doing updates, or accross
variants. This would cause the code matching segments using PDT to fail if the
segment PDT was 1 microsecond (or whatever small value) before the candidate
segment. And would pick the (wrong) following segment as the matching one.

In order to be more tolerant when matching, we instead check whether the
candidate segment is within the first segment of the segment we are trying to
match.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6961>
2024-05-30 07:59:25 +00:00
Edward Hervey
e7ab454cf5 hlsdemux2: Fix failure to find a replacement segment on resync
If we end up with a segment with an internal time that varies from the supposed
one, this could be for two reasons:
* We guess-timated the wrong segment to go to when advancing or switching
  variants. In that case we try to find the actual segment to go to (just before
  this change).
* There was a complete playlist change (for whatever reason) and we can't find a
  replacement. In that case we want to carry on playback from this position but
  need to remember that we moved (by setting the stream to DISCONT, and
  resetting the new mapping).

Fixes playback on several broken stream

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6961>
2024-05-30 07:59:25 +00:00
Edward Hervey
12e8874f88 hlsdemux2: Refactor update of GstHLSTimeMap values
This was also missing transferring the PDT if present

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6961>
2024-05-30 07:59:25 +00:00
Edward Hervey
e9214e9afc hlsdemux2: Fix parsing of EXT-X-DISCONTINUITY-SEQUENCE:0
Since the default value of `m3u8->discont_sequence` (before parsing of the
playlist data) was 0 .. we would never properly detect the presence of that
field if it was present with a value of 0.

This would later on cause havoc in playlist synchronization where we would
assume it didn't have a discontinuity sequence specified (whereas it did, and it
was 0).

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6961>
2024-05-30 07:59:25 +00:00
Edward Hervey
2560ac6998 hlsdemux2: Increase tolerance for discontinuity detection
A lot of streams will do a poor job of estimating proper duration of fragments
in the playlist, but over several fragments have it correct.

Instead of constantly trying to realign the estimated stream time, allow for a
more realistic tolerance of 3-4 video frames

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6961>
2024-05-30 07:59:25 +00:00
Edward Hervey
5ec5323c1f hlsdemux2: Ensure a discont will be set when resetting for lost sync
This is to ensures we inform the demuxer/parsers that what follows is not contiguous

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6961>
2024-05-30 07:59:25 +00:00
Edward Hervey
dadf2ec56c hlsdemux2: Fix handling of variant switching and playlist updates
When updating playlists, we want to know whether the updated playlist is
continuous with the previous one. That is : if we advance, will the next
fragment need to have the DISCONT buffer set on it or not.

If that happens (because we switched variants, or the playlist all of a sudden
changed) we remember that there is a pending discont for the next fragment. That
will be used and resetted the next time we get the fragment information.

Previously this was only partially done. And it was racy because it was set
directly on `GstAdaptiveDemux2Stream->discont` when a playlist was updated,
instead of when the next fragment was prepared.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6961>
2024-05-30 07:59:25 +00:00
Edward Hervey
726f2d8dc0 adaptivedemux2: Only set DISCONT on beginning of fragments
This avoids accidentally setting it in the middle of a fragment, which could
cause havoc in demuxer/parsers

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6961>
2024-05-30 07:59:25 +00:00
Edward Hervey
59582e2ffe hlsdemux2: Fix getting starting segment on live playlists
When dealing with live streams, the function was assuming that all segments of
the playlist had valid stream_time. But that isn't TRUE, for example in the case
of failing to synchronize playlists.

Fixes losing sync due to not being able to match playlist on updates

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6961>
2024-05-30 07:59:25 +00:00
Seungha Yang
0ca5517d80 d3d12encoder: Do not print error log for not-supported feature
gst_d3d12_result() will print message with ERROR level if failed.
Use FAILED/SUCCEEDED macros instead, since not-supported feature
is not a critical error

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6963>
2024-05-30 00:03:28 +00:00
Sergey Krivohatskiy
63367659f2 flacparse: fix buffer overflow in gst_flac_parse_frame_is_valid
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6960>
2024-05-29 20:24:45 +00:00
Tim-Philipp Müller
03cfca1033 Back to development after 1.24.4 2024-05-29 13:51:27 +03:00
Tim-Philipp Müller
9137f539a0 Release 1.24.4 2024-05-29 13:44:50 +03:00
Sebastian Dröge
def150ed2c gstreamer: parse: Don't assume that child proxy child objects are GstObjects
The name is already passed via the signal parameters so it doesn't have
to be retrieved again via GstObject API, which would crash on other
GObjects. Child proxy child objects can be any kind of GObject and the
code here otherwise handles this correctly already.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6951>
2024-05-29 11:14:11 +03:00
Sebastian Dröge
93a2026584 gstreamer: ptp-helper: Use u64 instead of c_ulong for ifa_flags on Solaris/Illumos
See https://gitlab.freedesktop.org/gstreamer/gstreamer/-/issues/3553#note_2429400

Patch by Marcel Telka <marcel@telka.sk>.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6950>
2024-05-29 11:02:26 +03:00
Sebastian Dröge
367d693f22 gstreamer: ptp-helper: Use if_nametoindex and setsockopt on Solaris / Illumos too
Patch by Marcel Telka <marcel@telka.sk>.

Fixes https://gitlab.freedesktop.org/gstreamer/gstreamer/-/issues/3552

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6947>
2024-05-29 01:54:29 +03:00
Sebastian Dröge
c36296895f gstreamer: ptp-helper: Don't import Context trait multiple times unnecessarily
This only affected the Solaris / Illumos code path.

Patch by Marcel Telka <marcel@telka.sk>.

Fixes https://gitlab.freedesktop.org/gstreamer/gstreamer/-/issues/3551

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6947>
2024-05-29 01:54:29 +03:00
Sebastian Dröge
c97ec122d9 gstreamer: ptp-helper: Use c_ulong for ifa_flags on Solaris/Illumos
Based on a patch by Marcel Telka <marcel@telka.sk>.

Fixes https://gitlab.freedesktop.org/gstreamer/gstreamer/-/issues/3553

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6947>
2024-05-29 01:54:29 +03:00
Sebastian Dröge
895ee6f72e gstreamer: Solaris/Illumos require linking to libnsl / libsocket for various socket APIs
Patch by Tim Mooney <Tim.Mooney@ndsu.edu> from OpenIndiana/oi-userland

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6947>
2024-05-29 01:54:29 +03:00
Philippe Normand
1caa041c91 webrtcbin: Allow session level setup attribute in SDP
An SDP answer can declare its setup attribute at the session level or at the
media level. Until this patch we were validating only the latter case and an
assert was raised in the former case.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6945>
2024-05-28 15:44:21 +00:00
Sebastian Dröge
3d9fd9926c typefind: Fix handling of ID_ODD_SIZE in WavPack typefinder
Chunks are always starting on an even position and this flag only
specifies that the last byte of the chunk is not valid.

Fixes https://gitlab.freedesktop.org/gstreamer/gstreamer/-/issues/3569

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6944>
2024-05-28 17:47:22 +03:00
Sebastian Dröge
b77de8f6f2 dtlsconnection: Fix overflow in timeout calculation on systems with 32 bit time_t
If a timeout of more than 4295s was scheduled, the calculation would
overflow and a too short timeout would be used instead.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6920>
2024-05-25 08:03:22 +00:00
Sebastian Dröge
4116127217 clock: Fix 32 bit assertions in GST_TIME_TO_TIMEVAL and GST_TIME_TO_TIMESPEC
On various 32 bit systems, time_t is actually 64 bits while long is
still only 32 bits. The macro would wrongly trigger its assertion in
this case if a value with more than 68 years worth of seconds is
converted.

Examples are various newer 32 bit platforms and old ones that are
compiled with -D_TIME_BITS=64.

Also statically assert that time_t is either 32 or 64 bits. Other values
might need adjustments in the macro.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6919>
2024-05-25 10:07:32 +03:00
He Junyan
e7e6472a31 kmssink: Do not close the DRM prime handle twice
The prime_fds for multi planes may be the same. For example, on Intel's
platform, the NV12 surface may have the same FD for the plane0 and the
plane1. Then, the DRM_IOCTL_GEM_CLOSE will close the same handle twice
and get an "Invalid argument 22" error the second time.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6916>
2024-05-23 23:08:36 +00:00
Daniel Stone
75ad05b518 wayland: Use wl_display_create_queue_with_name
Wayland 1.23 and above allow us to attach names to an event queue, which
are printed out when debugging. Do this to make the logs easier to read.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6915>
2024-05-23 23:28:52 +01:00
Yacine Bandou
1b191d1d8d streamsynchronizer: Fix deadlock when streams have been flushed before others start
To simplify the description, I'm assuming we only have two streams: video and audio.

For the video stream, we have the following events :
- STREAM_START => stream->wait set to true
- NEW_SEGMENT(1) => blocked waiting in gst_stream_synchronizer_wait
- FLUSH_START => unblocked
- FLUSH_STOP => stream->wait reset to false
- NEW_SEGMENT(2) => not waiting, since stream->wait is false

Then for the audio stream, we have the following events :
- STREAM_START => stream->wait set to true
- NEW_SEGMENT(2) => blocked waiting in gst_stream_synchronizer_wait for ever.

Note: The first NEW_SEGMENT event and the FLUSH_START, FLUSH_STOP events of the audio stream
are dropped before being received by the streamsynchronizer element, because the decodebin audio pad src
is not yet linked to the playsink audio pad sink.

To fix this deadlock, we don't reset stream->wait to false in the FLUSH_STOP event when it is not
waiting for the EOS of the other streams.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6887>
2024-05-23 17:51:02 +01:00
He Junyan
a084bedd58 vabaseenc: delete the useless frame counter fields
They are used to calculate the PTS and DTS before, no usage now.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6786>
2024-05-23 16:47:55 +01:00
He Junyan
3c26c0bc33 vabaseenc: Do not set the min_pts
Because all the va encoders improved their PTS/DTS algorithm, now
it is impossible to generate minus DTS. So no underflow will happen
and we do not need to set a 1000 hour offset now.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6786>
2024-05-23 16:47:48 +01:00
Backport Bot
607dadbc53 Revert "tests/d3d11: add concurrency test for gstd3d11device"
This reverts commit 203f6b00d4.

Revert test that was added with reverted commit as well.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6907>
2024-05-23 16:37:01 +01:00
Seungha Yang
a648f0da81 Revert "d3d11device: protect device_lock vs device_new"
This reverts commit 0cb12db96c
(i.e. commit 926d5366b9 on main).

AcquireSRWLockExclusive seems to be acquiring lock in exclusive mode
when the same lock is combined with write lock access.
Reverting the commit because of this is unexpected behavior
and unavoidable OS bug.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6907>
2024-05-23 16:36:45 +01:00
He Junyan
7526919fb3 vah265enc: Let FORCE_KEYFRAME be IDR frame rather than just I frame
The FORCE_KEYFRAME frame which has GST_VIDEO_CODEC_FRAME_FLAG_FORCE_KEYFRAME
bit set should be the sync point. So we should let it be an IDR frame to begin
a new GOP, rather than just promote it to an I frame.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6857>
2024-05-23 16:29:47 +01:00
He Junyan
5e24324f4f vah264enc: Let FORCE_KEYFRAME be IDR frame rather than just I frame
The FORCE_KEYFRAME frame which has GST_VIDEO_CODEC_FRAME_FLAG_FORCE_KEYFRAME
bit set should be the sync point. So we should let it be an IDR frame to begin
a new GOP, rather than just promote it to an I frame.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6857>
2024-05-23 16:29:47 +01:00
He Junyan
af88e87eec examples: vaenc-dynamic: support force key frame setting
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6857>
2024-05-23 16:29:40 +01:00
He Junyan
77455b50d3 vah265enc: Fix a memory leak when destroying the object
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6913>
2024-05-23 16:24:13 +01:00
He Junyan
2dd3ce721a vah265enc: Use a FIFO queue to generate DTS
The base parse will infer the DTS by itself, so we need to make DTS
offset before PTS in order to avoid DTS bigger than PTS. We now use
a FIFO queue to store all PTS and assign it to DTS by an offset.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6913>
2024-05-23 16:24:13 +01:00
He Junyan
09d07f13f9 vah264enc: Use a FIFO queue to generate DTS
The base parse will infer the DTS by itself, so we need to make DTS
offset before PTS in order to avoid DTS bigger than PTS. We now use
a FIFO queue to store all PTS and assign it to DTS by an offset.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6913>
2024-05-23 16:24:13 +01:00
Seungha Yang
9a9650aeb2 cudamemory: Fix offset of subsampled planar formats
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6910>
2024-05-23 13:52:28 +01:00
Sebastian Dröge
620d5cb5d6 av1enc: Use 1/90000 as timebase and don't use the framerate at all
This mirrors the behaviour in vp8enc / vp9enc and is generally more
useful than using any framerate from the caps as it provides some degree
of accuracy if the stream doesn't have timestamps perfectly according to
the framerate.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6909>
2024-05-23 11:10:14 +00:00
Sebastian Dröge
afe74a0181 av1enc: Fix last timestamp tracking so it actually works
This behaves exactly the same as in vp8enc / vp9enc now.

Fixes https://gitlab.freedesktop.org/gstreamer/gstreamer/-/issues/3546

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6909>
2024-05-23 11:10:14 +00:00
Sebastian Dröge
36a2eb0f03 gtk: Fail initialization of the sink if GTK4 is already initialized in the same process
Initializing GTK3 and GTK4 in the same process does not work and is not
supported.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6905>
2024-05-23 10:30:47 +01:00
Piotr Brzeziński
cdaf50de8f osxaudio: Avoid using private APIs on iOS
Turns out AudioConvertHostTimeToNanos and AudioGetCurrentHostTime are macOS-only APIs, which prevents apps using
GStreamer on iOS from being accepted into App Store.

This commit replaces those functions with a manual version of what they do - mach_absolute_time() for the current time,
and data from mach_timebase_info() at the beginning to convert host timestamps to nanoseconds.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6899>
2024-05-22 11:31:32 +01:00
Jordan Petridis
47afb4564b tests/check: Avoid using "bool" for the variable name
Glib 2.82 will be aliasing [1] TRUE and FALSE to the C99
definitions, which means it will be including stdbool.h

As such, having variables named "bool" causes issues
since it conflicts with the symbol defined in stdbool.h

[1] https://gitlab.gnome.org/GNOME/glib/-/merge_requests/4001

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6896>
2024-05-22 00:18:51 +01:00