Commit graph

1023 commits

Author SHA1 Message Date
Robin Gustavsson
38a8411bdf rtpklvdepay: Recover after invalid fragmented KLV unit
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4816>
2023-11-17 09:01:10 +00:00
Sebastian Dröge
db77deef00 rtpjitterbuffer: Add new "rfc7273-reference-timestamp-meta-only" property
If this property is enabled then the jitterbuffer will do the normal PTS
calculations according to the configured mode instead of making use of
the RFC7273 media clock.

The timestamp calculated from the RFC7273 media clock will only be
stored in the reference timestamp meta, if addition of that meta is enabled.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/5512>
2023-11-16 15:23:29 +00:00
Sebastian Dröge
eae3ef7461 rtpjitterbuffer: Add new rfc7273-use-system-clock property
When this property is used, it is assumed that the system clock is
synced close enough to the media clock used by an RFC7273 stream.

As long as both clocks are at most a few seconds from each other this
will give the correct results and avoids having to create an actual
network clock that has to sync first.

If the system clock is actually synchronized to the media clock then
everything will behave exactly the same, otherwise the reference
timestamp meta will be correct but the buffer timestamps will be off by
the difference between the two clocks.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/5512>
2023-11-16 15:23:29 +00:00
Sebastian Dröge
2956ba48fc rtpjitterbuffer: Improve handling of media clocks
Do more checks for clock equality than just checking pointers. The same
NTP/PTP clock might be used as pipeline clock but a new instance, so
instead also check what clock they are synced to.

Also handling setting / resetting of the media clock and pipeline clock
correctly by resetting the media clock's state accordingly.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/5512>
2023-11-16 15:23:29 +00:00
Piotr Brzeziński
4037334143 qtdemux: Ignore raw audio streams when adjusting seek
Because we treat raw audio chunks/samples as keyframes, they were interfering
with seek time adjustment.
Became apparent when the accompanying video stream was I-frame only,
for example ProRes.
Since raw audio streams can be seeked freely, it's fine to just ignore them here,
giving priority to the real keyframes in the video stream.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4946>
2023-11-15 07:55:27 +00:00
Dongyun Seo
8db184085a dcaparse: keep upstream buffer meta
Some audio decoders cannot decode DTS stream if there is no
valid timestamp. So, keep upstream buffer meta.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/5655>
2023-11-14 16:51:44 +09:00
Olivier Crête
c2a357c867 rtpopusdepay: set resync flag
- Set re-sync flag on output buffer when rtp had the marker flag set.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/5529>
2023-11-10 21:45:13 +00:00
Philippe Normand
1fc2bd8032 adaptivedemux2-stream: Use gst_clear_object when releasing collection
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/5606>
2023-11-08 09:16:55 +00:00
Johan Adam Nilsson
808c27b4cc wavparse: fix buffer leak with adtl tag
Fixes: https://gitlab.freedesktop.org/gstreamer/gstreamer/-/issues/3020
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/5595>
2023-11-03 19:38:38 +00:00
robert
e3e8147a74 ximagesrc: fix xnavigation linking issue
Fixes #3083

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/5589>
2023-11-03 17:36:58 +00:00
Seungha Yang
5e147ed3b8 meson: Fix MSVC build with GST_DISABLE_GST_DEBUG
MSVC does not understand Wno-unused

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/5585>
2023-11-03 13:31:03 +00:00
Sebastian Dröge
2dd65d8715 mpg123audiodec: Update rank from MARGINAL to PRIMARY
This is our primary MP3 decoder after mad got removed.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/5590>
2023-11-02 14:17:06 +00:00
robert
737c32b9b6 ximagesrc: fix compile-time warning and XInitThreads()
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/5493>
2023-11-01 09:17:24 +00:00
Tim-Philipp Müller
f6c40bb15c pngenc: mark output frames as I-frames
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/5546>
2023-10-27 05:47:37 +00:00
Tim-Philipp Müller
d69885e0f7 pngenc: output one frame only in snapshot mode
In snapshot mode pngenc should output exactly one frame
and then return FLOW_EOS to upstream. If upstream sends
more input frames before shutting down, it should keep
returning FLOW_EOS but not output any more encoded frames.

After a flushing seek it should output frames again though.

Fixes #3069.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/5546>
2023-10-27 05:47:37 +00:00
Shengqi Yu
25c00b5ba2 v4l2object: scale the encoded sizeimage based on maximum resolution
The default 2MB ENCODED_BUFFER_SIZE can't support some 4K video playback. We now
detect the driver reported maximum resolution and choose an appropriate
default bitstream size accordingly. For 4K video these results in around 4MB
buffer instead of 2MB.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4549>
2023-10-23 14:10:56 +00:00
Matthias Fuchs
2bbc2a4c52 qml6glsrc: sync on the streaming thread
After rendering a QML scene the qml6glsrc element copies the contents of
the scene to a GStreamer buffer. This happens on the Qt render thread.
Then it attaches a sync point to the destination buffer. This sync point
must be awaited by other threads which use the buffer later on. The
current implementation relies on the downstream elements to wait for the
sync point. However, there are situation where this does not work. The
GstBaseTransform e.g. copies the buffer metadata (which overwrites the
sync point without waiting for it) *before* waiting for the sync point.

This commit waits for the sync point inside the qml6glsrc element before
sending it downstream. The wait command is issued on the streaming
thread with the pipeline OpenGL context, i.e. it will synchronize with
the GStreamer OpenGL thread.

This is a port of the original fix for the qmlglsrc element.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/5519>
2023-10-23 08:43:16 +00:00
Tim-Philipp Müller
654f3370a0 meson: Bump GLib requirement to >= 2.64
This includes fixes to make GstBus watches non-racy.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2126>
2023-10-22 10:48:12 +01:00
Tim-Philipp Müller
136c82d735 flacenc: signal in output caps that the output is framed
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/5524>
2023-10-22 00:25:50 +00:00
Tim-Philipp Müller
bce1d121ba rtpac3depay: should output audio/x-ac3 not audio/ac3
audio/x-ac3 is the canonical media format in GStreamer.
audio/ac3 is sometimes accepted as input (e.g. in rtpac3pay
or ac3parse), but shouldn't be output.

Fixes #3038.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/5472>
2023-10-19 13:27:58 +00:00
Matthias Fuchs
24ae3de107 qmlglsrc: sync on the streaming thread
After rendering a QML scene the qmlglsrc element copies the contents of
the scene to a GStreamer buffer. This happens on the Qt render thread.
Then it attaches a sync point to the destination buffer. This sync point
must be awaited by other threads which use the buffer later on. The
current implementation relies on the downstream elements to wait for the
sync point. However, there are situation where this does not work. The
GstBaseTransform e.g. copies the buffer metadata (which overwrites the
sync point without waiting for it) *before* waiting for the sync point.

This commit waits for the sync point inside the qmlglsrc element before
sending it downstream. The wait command is issued on the streaming
thread with the pipeline OpenGL context, i.e. it will synchronize with
the GStreamer OpenGL thread.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/5506>
2023-10-19 08:19:05 +00:00
Robert Ayrapetyan
3d807d4f6d ximagesrc: add navigation support
Add a basic navigation support:
- mouse events (buttons/move)
- keyboard events (keys)

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/5273>
2023-10-13 23:34:54 +00:00
Jordan Petridis
5f7a37f21e qt6: if def newer symbosl in QRhiTexture
version 6.4 added QRhiTexture::RGB10A2 but we depend on an older
version of qt in meson, and we can keep compiling with older Qt6
versions still.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/5475>
2023-10-12 22:57:35 +00:00
Stéphane Cerveau
7c7a90b99d imagesequencesrc: fix regular image deadlock
With one regular image file path provided (without %05d),
the element was stuck in a dead loop counting the frames:

gst_image_sequence_src_count_frames

This allows to display any image file out of the element
for a given number of buffers.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/5471>
2023-10-12 22:06:02 +00:00
Matthew Waters
7b491f382c build/qt6: properly error/skip build if the qsb tool is not found
Fixes https://gitlab.freedesktop.org/gstreamer/gstreamer/-/issues/3032

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/5459>
2023-10-12 12:58:26 +00:00
Michael Tretter
0563a25494 v4l2videoenc: unconditionally activate the OUTPUT pool
If the v4l2videoenc receives an QUERY_ALLOCATION, it must not propose a
currently used pool, because it cannot be sure that the allocation query came
from exactly the same upstream element. The QUERY_ALLOCATION will not contain
the internal OUTPUT pool.

The upstream element (the basesrc) detects that the newly proposed pool differs
from the old pool. It deactivates the old pool and switches to the new pool.

If there was a format change, a new OUTPUT buffer pool will be allocated in
gst_v4l2_object_set_format_full() and the CAPTURE task will be stopped to switch
the format. If there hasn't been a format change,
gst_v4l2_object_set_format_full() will not be called. The old pool will be kept
and reused.

Without a format change, the processing task continues running.

This leads to the situation that the processing task is running, but the OUTPUT
buffer pool (the old pool) is deactivated. Therefore, the encoder is not able to
get buffers from the OUTPUT pool and encoding cannot continue.

This situation can be triggered by sending a RECONFIGURE event without a format
change.

Resolve this situation by ensuring that the OUTPUT buffer pool is always
activated when frames arrive at the encoder.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4235>
2023-10-11 19:35:54 +00:00
Michael Tretter
41ce99ebab v4l2videoenc: fix activation of internal pool
Fix the buffer pool activation if the driver does not support VIDIOC_CREATE_BUFS
the same way as it was fixed for the v4l2videodec.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4235>
2023-10-11 19:35:54 +00:00
Michael Tretter
5e72e1985a v4l2videoenc: rename OUTPUT pool to opool
There is a CAPTURE pool in the same function. While the CAPTURE pool is called
cpool, using pool for the OUTPUT pool is confusing.

Using opool for the OUTPUT pool makes it more obvious, which pool is used.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4235>
2023-10-11 19:35:54 +00:00
Guillaume Desmottes
a56aabc773 flvmux: set the src segment position as running time
We were already converting the pad last timestamp to running time but
not the segment position.
This segment position is used by gst_aggregator_simple_get_next_time()
to compute the waiting time when aggregating.

Those waiting times were wrong in my live pipeline using the system
clock, resulting in the aggregator to never wait at all.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/5460>
2023-10-11 15:20:18 +00:00
Nicolas Dufresne
bcfbdfbbca v4l2: Fix tiled formats stride conversion
While adding arbitrary tile support, a round up operation was badly
converter. This caused the Y component of the stride to be 0. This
eventually lead to a crash in glupoad preceded by the following
assertion.

  gst_gl_buffer_allocation_params_new: assertion 'alloc_size > 0' failed

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/5458>
2023-10-11 14:13:53 +00:00
Thibault Saunier
049859c2cb adaptivedemux2: Do not submit_transfer when cancelled
There is a race condition where transfer has not been submitted yet while the
request is cancelled which leads to the transfer state going back to
`DOWNLOAD_REQUEST_STATE_OPEN` and the user of the request to get signalled about
its completion (and the task actually happening after it was cancelled) leading
to assertions and misbehaviours.

To ensure that this race can't happen, we start differentiating between the
UNSENT and CANCELLED states as in the normal case, when entering `submit_request`
the state is UNSENT and at that point we need to know that it is not because
the request has been cancelled.

In practice this case lead to an assertion in
`gst_adaptive_demux2_stream_begin_download_uri` because in a previous call to
`gst_adaptive_demux2_stream_stop_default` we cancelled the previous request and
setup a new one while it had not been submitted yet and then got a `on_download_complete`
callback called from that previous cancelled request and then we tried to do
`download_request_set_uri` on a request that was still `in_use`, leading to
something like:

```
 #0: 0x0000000186655ec8 g_assert (request->in_use == FALSE)assert.c:0
 #1: 0x00000001127236b8 libgstadaptivedemux2.dylib`download_request_set_uri(request=0x000060000017cc00, uri="https://XXX/chunk-stream1-00002.webm", range_start=0, range_end=-1) at downloadrequest.c:361
 #2: 0x000000011271cee8 libgstadaptivedemux2.dylib`gst_adaptive_demux2_stream_begin_download_uri(stream=0x00000001330f1800, uri="https://XXX/chunk-stream1-00002.webm", start=0, end=-1) at gstadaptivedemux-stream.c:1447
 #3: 0x0000000112719898 libgstadaptivedemux2.dylib`gst_adaptive_demux2_stream_load_a_fragment [inlined] gst_adaptive_demux2_stream_download_fragment(stream=0x00000001330f1800) at gstadaptivedemux-stream.c:0
 #4: 0x00000001127197f8 libgstadaptivedemux2.dylib`gst_adaptive_demux2_stream_load_a_fragment(stream=0x00000001330f1800) at gstadaptivedemux-stream.c:1969
 #5: 0x000000011271c2a4 libgstadaptivedemux2.dylib`gst_adaptive_demux2_stream_next_download(stream=0x00000001330f1800) at gstadaptivedemux-stream.c:2112
```

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/5435>
2023-10-05 20:55:00 +00:00
Nicolas Dufresne
fc4bb5585f doc: Update plugin cache for added DMA_DRM format
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/5386>
2023-10-03 21:13:00 +00:00
Nicolas Dufresne
aaed9272c1 video-filters: Fix passthrough with ANY caps feature
With the support for DRM modifiers, passthrough caps must now include DMA_DRM
format, otherwise pipeline using thhese filters unconditionally may fail
to negotiate.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/5386>
2023-10-03 21:13:00 +00:00
Sebastian Dröge
8af9cd9b1a docs: Update plugins caches
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/5412>
2023-10-02 09:39:21 +03:00
Sebastian Dröge
abdd1967ad flacenc: Correctly handle up to 255 cue entries
The counter was using a signed 8 bit integer, which was overflowing
after 127 entries. That was then passed as an unsigned 32 bit integer to
libflac, which caused it to be converted to a huge unsigned number.
That then caused an invalid memory access inside libflac.

As a bonus, signed integer overflow is undefined behaviour.

Instead, use an unsigned 8 bit integer. Once this overflows the existing
code already catches it and stops adding the cue. While FLAC__metadata_object_cuesheet_insert_track()
takes an unsigned 32 bit integer for the track number, FLAC__StreamMetadata_CueSheet_Track is
limiting it to an unsigned 8 bit integer.

Fixes https://gitlab.freedesktop.org/gstreamer/gstreamer/-/issues/2921

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/5420>
2023-09-30 15:46:52 +00:00
Dominique Leroux
7affa01e05 osxaudio: add individual elements registration for gst-full compatibility
Found that osxaudiosink could not be added standalone in gst-full build
using
-Dgst-full-elements=osxaudio:osxaudiosink because element registration
was
done at the plugin level. Now src/sink elements and deviceprovider have
their
individual registration.

Copied/adapted from the alsa plugin.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/5419>
2023-09-28 21:44:48 +00:00
Stéphane Cerveau
80cc1fcc03 mpdhelper: remove useless code
The audio/video codec name from mime type should be retrieved from
gst_codec_utils_caps_get_mime_codec instead

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/5404>
2023-09-28 18:31:07 +00:00
Xavier Claessens
0ab48250a9 GstCustomMeta: Use simplified API where possible
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/5385>
2023-09-27 18:46:34 +00:00
Florian Zwoch
4a9a9ed9fc adaptivedemux2: Call GTasks's return functions for blocking tasks
Gio/Task states the following:

If a GTask has been constructed and its callback set, it is an error to
not call g_task_return_*() on it. GLib will warn at runtime if this
happens (since 2.76).

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/5395>
2023-09-27 15:56:08 +00:00
Albert Sjölund
47dbdea469 souphttpsrc: Chain finalize call to parent
GstSoupSession finalize does not chain parent finalize,
causing it to leak memory, shown under g freeze notify.
In finalize method, ensure all branches call to parent
finalize.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/5398>
2023-09-27 09:01:43 +02:00
Daniel Moberg
0e6cd64232 rtspsrc: Property for adding custom http request headers
This commit adds a property which enables adding custom http request headers to
the rtspsrc element. Added headers will be appended to http requests
made during http tunneling.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/5268>
2023-09-26 06:35:43 +00:00
Stijn Last
4bda59f88d deinterlace: greedy, improve quality
scanlines->m1 = same line of the previous field
scanlines->t0 = line above of the current field
scanlines->b0 = line below of the current field
scanlines->mp = same line of the next field

Deinterlacing a field weaved frame:
When deinterlacing the top field, the next bottom field is available
(part of the same frame). but when deinterlacing the bottom field,
the next top field (part of the next frame) is not available and
scanlines->mp equals NULL.

In this case it's better to use greedy algorithm using the prevous field
(twice) rather then linear interpolation of the current field.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/5331>
2023-09-25 06:40:47 +00:00
Hou Qi
be9d9371b7 v4l2videodec: Correctly free caps to avoid memory leak
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/5379>
2023-09-24 12:50:01 +00:00
Seungha Yang
69d1679914 video: Add GBR 16bits formats
Adding 16bits planar RGB formats

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/5375>
2023-09-23 13:12:55 +00:00
Sebastian Dröge
2a2ef23829 rtpsource: Don't store invalid running times and calculate with it
If we end up with GST_CLOCK_TIME_NONE as running time for an RTP packet
then this can't be used for bitrate estimation, and also not for
constructing the next RTCP SR. Both would end up with completely wrong
values, and an RTCP SR with wrong values can easily break
synchronization in receivers.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/5329>
2023-09-23 07:39:00 +00:00
Piotr Brzeziński
f3d98341e3 qml: Fix leftover reference to gstqsgtexture
Made it impossible to build with qmake as per the readme. The file was renamed to gstqsgmaterial a while ago.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/5357>
2023-09-19 23:55:45 +00:00
Olivier Blin
4b891639da pulsedeviceprovider: fix incorrect usage of GST_ELEMENT_ERROR
The provider is not a GStreamer element.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/5349>
2023-09-19 14:13:49 +02:00
Sebastian Dröge
fcd591c1af rtpjitterbuffer: Avoid integer overflow in max saveable packets calculation with negative offset
The timestamp offset can be negative, and it can be a bigger negative
number than the latency introduced by the rtpjitterbuffer so the overall
timeout offset can be negative.

Using the negative offset for calculating how many packets can still
arrive in time when encountering a lost packet in an equidistant stream
would then overflow and instead of considering fewer packets lost a lot
more packets are considered lost.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/5296>
2023-09-12 08:38:53 +00:00
Nicolas Dufresne
c1e03081c0 v4l2: object: Handle video helper return value
gst_video_info_set_interlaced_format() can return an error if the
width/height causes integer overflow. Handle this case, so that we can
fail cleanly. This has been experienced while testing an in-progress
driver.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/5286>
2023-09-11 15:05:34 -04:00
Nicolas Dufresne
353cb2da92 v4l2: bufferpool: Avoid warnings on empty last buffer
Some drivers will push an buffer flagged LAST but empty. In decoder
case, this results in an "producing too many buffer" warning, even
though the result is entirely correct. Detect this case in order to
signal EOS earlier and avoid this warning.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/5286>
2023-09-11 18:08:21 +00:00