Commit graph

785 commits

Author SHA1 Message Date
Tim-Philipp Müller
1a0df09c8f test: souphttpsrc: update test_icy_stream url
The old URL would result in multiple redirects, which
causes memory leaks with old versions of libsoup as
used on the CI.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6049>
2024-02-04 14:00:01 +01:00
Tim-Philipp Müller
600e105e64 Back to development
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/5990>
2024-01-25 00:18:12 +00:00
Tim-Philipp Müller
3e41b8f18b Release 1.22.9 2024-01-24 18:21:13 +00:00
Dan Searles
19d3b14f51 rtspsrc: fix ttl setting for udpsink[1]
Fix ttl setting being incorrectly applied to udpsink[0] rather
than to udpsink[1].

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/5967>
2024-01-23 23:52:25 +00:00
Dan Searles
ba5692005d rtspsrc: set multicast-iface on udpsinks
Copy rtspsrc property multicast-iface to its udpsinks to
allow messages over those sinks back to the server to work (and
prevent 'Network unreachable' warnings).

Closes: #3239
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/5967>
2024-01-23 23:52:25 +00:00
Xavier Claessens
0ce3a2782c v4l2src: Consider framerate during caps selection
This simplifies the way it picks the closest caps to preference and take into
consideration the framerate to avoid picking high resolution at 5fps or so.
Simply calculate a "distance" of caps A and B from the preference and put
closest first, sorting by framerate first.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/5956>
2024-01-22 18:52:57 +00:00
Philippe Normand
36518d81e7 vpxdec: Use appropriate domain and code for decoding errors
STREAM domain and DECODE error is commonly used in other decoders. ENCODE is for
encoders.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/5918>
2024-01-13 18:21:41 +00:00
Sanchayan Maity
daa60e39f9 rtphdrext-clientaudiolevel: Fix level value being written by the extension
When level value is greater than 127, it was being clamped but this clamped
value was not the one being actually used. For level values greater than 127
this resulted in an incorrect value being used. As an example, a level value
of 187, after and'ed with 0x7F, it would result in 0x3B being reported as the
level value.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/5894>
2024-01-07 12:58:27 +00:00
Sebastian Dröge
a4dc820899 rtpsession: Only warn once if configured latency needs to be known but isn't yet
Otherwise we would warn about this once for every single packet until
the LATENCY event is received.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/5874>
2023-12-29 12:46:23 +01:00
Chao Guo
0587c2754e v4l2object: clear old fds in poll when closing v4l2object
When reopening a v4l2 device, the v4l2object->poll will include some old fds,
which was assigned to this device before. If the pipeline opens multiple v4l2
devices, the old fd may been assigned to other v4l2 devices when reopening
devices.

This will cause the timing of the pipeline become confusing when polling devices,
leading functional abnormalities.

Therefore, when closing v4l2object, remove the old fds in poll to ensure that the
pipeline timing is normal.

Signed-off-by: Chao Guo <chao.guo@nxp.com>
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/5840>
2023-12-19 17:18:58 +00:00
Tim-Philipp Müller
79cdbc37d5 Back to development
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/5826>
2023-12-18 13:52:12 +00:00
Tim-Philipp Müller
4af14db10e Release 1.22.8 2023-12-18 12:09:37 +00:00
Arun Raghavan
a0558cf8d4 rtp: Fix incorrect RTP channel order lookup by name
The g_ascii_strcasecmp() logic is inverted, since it returns 0 on equality.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/5817>
2023-12-16 11:04:57 +00:00
Tim-Philipp Müller
44503fc88c matroska-demux: fix accumulated base offset in segment seeks
When doing a segment seek, the base offset in the new segment
would be increased by segment.position which is basically the
timestamp of the last packet. This does not include the duration
of the last packet though, so might be slightly shorter than the
actual duration of the clip or the requested segment.

Increase the base offset by the segment duration instead when
accumulating segments, which is more correct as it doesn't cut
off the last frame and makes the effective loop segment duration
consistent with the actual duration returned from a duration
query.

In case a segment stop was specified it's also possible that
some data was sent beyond the stop that's necessary for decoding
so the base offset increment should be based on that then and
not on the timestamp of the last buffer pushed out.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/5787>
2023-12-09 14:06:53 +00:00
Guillaume Desmottes
ef734f1134 qtdemux: fix bug report URL
Using PACKAGE_BUGREPORT as in other modules.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/5786>
2023-12-09 12:45:25 +00:00
Hosang Lee
4356d4262e qtdemux: Don't overflow sample index
Don't reduce sample index if it is already at 0.
Assigning -1 to a guint32 variable causes unexpected behavior.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/5751>
2023-12-01 15:54:46 +00:00
Hosang Lee
f4ed87283b qtdemux: Fix reverse playback for pcm audio stream
Some raw lpcm or adpcm may have larger sample sizes than the max
buffer size value set.
Trimming the buffer causes bogus size error on reverse playback.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/5745>
2023-12-01 09:54:41 +00:00
jeri.li
390d43dcde v4l2bufferpool: add lock as atomic operation for seek
When seek flush, gst v4l2 buffer pool flush is not atomic which will
lead double enqueue buffer (qbuf) issue, and v4l2 buffer pool qbuf is
also not atomic which will lead no free buffer found in the pool.
1. add lock for calculate enqueue number in streamon function
2. add lock for v4l2 capture end streamoff in pool flush function
3. lock the whole funciton of v4l2 buffer pool qbuf, then the buffer
   pool index and qbuf operation are atomic

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/5695>
2023-11-20 21:30:39 +00:00
Robin Gustavsson
97f3ed0f3b rtpklvdepay: Recover after invalid fragmented KLV unit
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/5686>
2023-11-17 10:16:50 +00:00
Piotr Brzeziński
15e9b513da 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/5674>
2023-11-15 19:44:38 +00:00
Tim-Philipp Müller
faf6edc026 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/5662>
2023-11-15 00:10:35 +00:00
Dongyun Seo
23bc229e7f 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/5664>
2023-11-14 22:21:16 +00:00
Tim-Philipp Müller
b1067c023d Back to development
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/5650>
2023-11-13 14:57:09 +00:00
Tim-Philipp Müller
4d13eddc8b Release 1.22.7 2023-11-13 11:04:22 +00:00
Thibault Saunier
f4a5b6623b 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/5611>
2023-11-06 16:21:08 +00:00
Johan Adam Nilsson
6a3e0ccd95 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/5597>
2023-11-04 18:48:26 +00:00
Tim-Philipp Müller
96326904f9 pngenc: mark output frames as I-frames
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/5564>
2023-10-27 18:58:29 +00:00
Tim-Philipp Müller
567e3b3d22 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/5564>
2023-10-27 18:58:29 +00:00
Matthias Fuchs
7c49ce1c21 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/5515>
2023-10-20 00:36:26 +00:00
Stéphane Cerveau
87292c6026 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/5487>
2023-10-15 19:00:36 +01:00
Guillaume Desmottes
81178eab42 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/5465>
2023-10-12 00:28:50 +01:00
Nicolas Dufresne
f223ea076d 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/5463>
2023-10-11 16:48:51 +01:00
Sebastian Dröge
10d42caef2 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/5436>
2023-10-04 19:06:33 +01:00
Florian Zwoch
80d62798c3 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/5414>
2023-09-28 12:04:11 +02:00
Albert Sjölund
c53376cbfc 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/5402>
2023-09-27 11:27:02 +00:00
Hou Qi
ca2f7a4254 v4l2videodec: Correctly free caps to avoid memory leak
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/5388>
2023-09-25 14:57:09 +02:00
Tim-Philipp Müller
0cccb1c9a6 Back to development
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/5371>
2023-09-20 19:41:00 +01:00
Tim-Philipp Müller
ddb4fbe431 Release 1.22.6 2023-09-20 18:10:57 +01:00
Olivier Blin
ea2e829452 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/5354>
2023-09-19 14:09:46 +01:00
Nicolas Dufresne
92820a8cf2 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/5319>
2023-09-12 17:44:44 +01:00
Nicolas Dufresne
bce3053598 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/5319>
2023-09-12 17:42:16 +01:00
Nicolas Dufresne
914e752015 v4l2: bufferpool: Do not resize compressed buffer
Avoid resizing compressed buffer to their maximum size. This fixes a
regression that caused valid but very large streams to be generated.

Fixes #2953

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/5319>
2023-09-12 17:42:06 +01:00
Sebastian Dröge
2bd21a6f40 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/5318>
2023-09-12 10:40:33 +01:00
Ming Qian
551fe9c9b1 v4l2object: clear format lists if source change event is received
If decoder notify a source change event when the capture format is
changed, not the resolution changed.

then gst_v4l2_object_acquire_format will retuen false due to
unsupported format.

we need to clear the format lists in the source change flow,
and reenumerate format list

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/5223>
2023-08-22 23:59:10 +01:00
Nicolas Dufresne
426bf399d4 v4l2: allocator: Don't close foreign dmabuf
Imported dmabuf are not being duped, so they should never be closed. Instead,
we ensure their live time by having strong reference on their original
buffer. This should fix potential flickering due to dmabuf being closed
too early.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/5217>
2023-08-21 23:52:57 +01:00
Jan Alexander Steffens (heftig)
d956a7e2d1 qt: Unbreak build with qt-egl enabled but viv_fb missing
Avoids an error message when the feature is explicitly enabled:

    ERROR: Feature qt-egl cannot be enabled: gstreamer-gl-viv_fb-1.0 is required

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/5200>
2023-08-18 12:45:14 +00:00
Wang Chuan
19a0463903 gstadaptivedemux: fix memory leak
GstQuery leaks when using invalid url

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/5156>
2023-08-12 15:17:46 +00:00
Nirbheek Chauhan
15a180189a meson: Fix searching of qt5/qt6 tools with qmake
If the pkg-config files are broken, we want to ensure that qmake is
used. This can easily happen on macOS with the official Qt binaries.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/5149>
2023-08-04 16:40:21 +01:00
Alicia Boya García
654ca283a7 qtdemux: Fix premature EOS when some files are played in push mode
Fixes https://gitlab.freedesktop.org/gstreamer/gstreamer/-/issues/2771

This EOS branch exists so that if a seek with a stop is made, qtdemux
stops accepting bytes from the sink after the entire requested playback
range is demuxed, as otherwise we could keep download content that is
not being used.

This patch fixes two flaws that were present in that EOS check:

1) A comparison was made between track time and movie time without conversion.
This made the check trigger early in files with edit lists. This patch fixes
this by converting the track PTS to movie PTS (stream time) for the check.

2) To avoid sending a EOS prematurely when the segment stop is within a GOP and
B-frames are present, the check for EOS should only be done for keyframes. I
gather this was already the intention with the existing code, but because it
used `stream->on_keyframe` instead of the local variable `keyframe` the old
code was checking if the *previous* frame was a keyframe.

It's interesting to note that these two flaws in the old code mask each other
in most cases: the track PTS will have reached the movie end PTS, but EOS would
only be sent if the previous frame was a keyframe. A simple case where they
wouldn't mask each other, reproducing the bug, is a sequence of 3 frame GOPs
with structure I-B-P.

The following validateflow tests have been added to future-proof the
fix:

 * validate.test.mp4.qtdemux_ibpibp_non_frag_pull.default
 * validate.test.mp4.qtdemux_ibpibp_non_frag_push.default

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/5114>
2023-08-03 20:22:51 +00:00
Nicolas Dufresne
40311bf665 v4l2: Fix support for left and top padding
In the current implementation, we support for most pixel format left
and top padding by changing the offset in the video meta. Though, to
align driver bytesused to the offset, we recalculate the offset, which
removed the modification we did before.

Instead, save the plane size, and truncate the driver reported bytesused
to the expected size, which ensures that the offsets still match. This
should also fix issues were the buffer size ended up bigger then the
pool size due to driver introduced padding.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/5132>
2023-08-01 19:54:57 +01:00