Commit graph

859 commits

Author SHA1 Message Date
Guillaume Desmottes
3ae2904f3d flvmux: rename 'new_tags' to 'new_metadata'
The metadata contains more than just tags: resolution, framerate, etc.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4730>
2023-05-30 08:27:18 +02:00
Guillaume Desmottes
853fad001e flvmux: add some logs when input is changing
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4730>
2023-05-30 08:27:18 +02:00
Michael Olbrich
2197cdc289 flvmux: use the correct timestamp to calculate wait times
Since c0bf793c05 ("flvmux: Set PTS based on
running time") the timestamp of the output buffer is already in running
time. So using that for 'srcpad->segment.position' does not work correctly
because gst_aggregator_simple_get_next_time() will convert it again with
gst_segment_to_running_time().
This means that the timestamp returned by
gst_aggregator_simple_get_next_time() may be incorrect. For example, if
flvmux is added to a already runinng pipeline then the timestamp is too
small and gst_aggregator_wait_and_check() returns immediately. As a result,
buffers may be muxed in the wrong order.

To fix this, use the PTS of the incoming buffer instead of the outgoing
buffer. Also add the duration as get_next_time() is supposed to return the
timestamp of the next buffer, not the current one.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4701>
2023-05-29 14:56:13 +00:00
Michael Olbrich
285811e7a7 jpegdec: be stricter when detecting interlaced video
There are broken(?) mjpeg videos that are incorrectly detected as
interlaced. This happens because 'info.height > height' (e.g. 1088 > 1080).

In the interlaced case info.height is approximately 'height * 2' but not
exactly because height is a multiple of DCTSIZE. Make the check more
restrictive but take the rounding effect into account.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4696>
2023-05-25 18:34:34 +00:00
Michael Olbrich
59290feca4 jpegdec: decode the correct number of lines for interlaced frames
For interlaced jpeg, gst_jpeg_dec_decode_direct() is called twice, once for each
field. In this case, stride[n] is plane_stride[n] * 2 to ensure that only every
other line is written. So the loop must stop at height / num_fields.

If the frame is really interlaced then continuing beyound this, is not harmful,
because jpeg_read_raw_data() will do nothing and return 0, so am info message is
printed.

However, if the frame is not actually interlaced, just misdetected as interlaced
then there is still data available from the second half of the frame. Now
line[0][j] is set to the scratch buffer. If the scratch buffer is not allocated
(because the height is a multiple of v_samp[0] * DCTSIZE) then the result is a
segfault due to a null-pointer dereference.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4696>
2023-05-25 18:34:34 +00:00
YURI FEDOSEEV
8dd51501d0 v4l2videoenc: support force keyframe event in v4l2 encoder
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4663>
2023-05-24 12:42:24 +00:00
Ruben Gonzalez
059965fe53 doc: Fix newline char between authors
Found running `gst-inspect-1.0 -a |& grep -v ":" | grep @`

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4682>
2023-05-20 05:48:23 +00:00
Nicolas Dufresne
0c9ab49579 v4l2: videodec: Fix stalls on empty buffer
Drivers may signal end of sequence using an empty buffer and LAST buffer
set, or just an empty buffer on certain legacy implementation. When this
occured, we'd send GST_V4L2_FLOW_LAST_BUFFER were the code expected
GST_FLOW_EOS. Stop abusing GST_FLOW_EOS and port all the code to the new
GST_V4L2_FLOW_LAST_BUFFER.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4669>
2023-05-19 23:06:06 +00:00
Sebastian Dröge
d5a0cfc563 qtdemux: Add support for SpeedHQ video codec
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3982>
2023-05-19 07:16:03 +00:00
Matthew Waters
3f4bfa097a qml6: add a mixer element
Can take multiple input streams and a qml scene and layout the input
videos inside the qml scene.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4609>
2023-05-19 01:48:57 +00:00
Shengqi Yu
5da9a8e2f4 v4l2object: fix some errors in probe_caps_for_fromat
1, there is a mistake when print stepwise.max_height, fix it
2, modify the calculation of width and height under the step wise
condition

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4562>
2023-05-18 13:45:11 +00:00
Ruben Gonzalez
5c0f6b88d8 README.md: fix current version
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4662>
2023-05-18 06:25:50 +00:00
Hou Qi
783ebbeecb v4l2videoenc: fix set format failure when needs reset encoder
In cases that encoder needs to reset format, there is race while draining.
v4l2videoenc finish() sends CMD_STOP command to driver, and desire to return
GST_FLOW_OK. But at this time, encoder CAPTURE may have dequeued the last
buffer and got eos. finish() return value changes to be GST_FLOW_EOS which
causes set format fail. So there is no need to check return value for finish()
when set format.

Also need to flush encoder after draining to make sure flush is finished.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4495>
2023-05-17 17:59:29 +00:00
Sebastian Dröge
99285bb566 qtmux: Fix extraction of CEA608 data from S334-1A packets
The index is already incremented by 3 every iteration so multiplying it
by 3 additionally on each array access is doing it twice and does not
work.

This caused invalid files to be created if there's more than one CEA608
triplet in a buffer, and out of bounds memory reads.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4634>
2023-05-16 11:29:45 +00:00
Jan Schmidt
131d59518e splitmuxsrc: Make PTS contiguous by preference
Make splitmuxsrc deal better with stream reordering by
making the largest observed PTS contiguous in the
next fragment. Previously, it selected DTS, but then
aligned that with the segment start of the next fragment,
which holds PTS values - leading to glitches in
streams that don't have PTS = DTS at the start.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4637>
2023-05-16 04:34:16 +00:00
Sebastian Dröge
bb2c5981fe pulse: Change bitfield booleans to normal gbooleans
Assigning TRUE (1) to a signed 1 bit integer will cause truncation
from 1 to -1 because the only non-zero value that can be stored is -1
due to how two's-complement works.

As this is a proper GObject let's not bother with all this and simply
use a normal gboolean instead.

../subprojects/gst-plugins-good/ext/pulse/pulsesink.c:1490:19: warning: implicit truncation from 'int' to a one-bit
        wide bit-field changes value from 1 to -1 [-Wsingle-bit-bitfield-constant-conversion]
  pbuf->in_commit = TRUE;
                  ^ ~~~~

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4617>
2023-05-14 15:58:35 +00:00
Sebastian Dröge
f9a3b3eacf rtpjitterbuffer: Fix uninitialized variable compiler warning
It could indeed be used uninitialized, but only if one of the
g_return_val_if_fail() caused an early return.

../subprojects/gst-plugins-good/gst/rtpmanager/rtpjitterbuffer.c: In function ‘rtp_jitter_buffer_append_query’:
../subprojects/gst-plugins-good/gst/rtpmanager/rtpjitterbuffer.c🔢10: warning: ‘head’ may be used uninitialized
      [-Wmaybe-uninitialized]
 1234 |   return head;
      |          ^~~~
../subprojects/gst-plugins-good/gst/rtpmanager/rtpjitterbuffer.c:1232:12: note: ‘head’ was declared here
 1232 |   gboolean head;
      |            ^~~~

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4616>
2023-05-14 14:26:05 +00:00
Piotr Brzeziński
5e45a1b1bd macos: Set activation policy in osxvideosink and glimagesink
Upon creating a window, glimagesink and osxvideosink now set the policy to
NSApplicationActivationPolicyRegular, which lets us show an icon in the Dock
for convenience and appear in the top menu bar like other apps.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4573>
2023-05-12 01:14:44 +02:00
Piotr Brzeziński
f60c87769f macos: Remove old NSApp workaround related code
This is no longer needed since the introduction of `gst_macos_main()` in 1.22.
Before that existed, we had a patch for GLib in Cerbero, which did work but made it
impossible to update GLib at all. The code being removed was a fail-safe in case of
running without said patch being applied. It's no longer needed, since for macOS
we just wrap our GStreamer with an NSApplication using `gst_macos_main()`.

Warnings will be displayed if no NSApp/NSRunLoop is found wherever needed,
pointing the user towards using the new API.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4366>
2023-05-11 20:30:19 +02:00
Tim-Philipp Müller
0c4a702e82 qtdemux: add unit test for edit list regression
File is the mp4 file from #2549 with the mdat atom
zeroed out and compressed. We compress twice because
apparently compressing 5MB of zeroes effectively in
one run is too difficult for gzip.

https://gitlab.freedesktop.org/gstreamer/gstreamer/-/issues/2549

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4560>
2023-05-11 16:45:37 +00:00
Mathieu Duponchelle
3d3d2ed447 Revert "qtdemux: fix conditions for end of segment in reverse playback"
This reverts commit 9deb3c27ac.

The test case that was described in the associated MR
(https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/262)
remains adequately fixed by a related MR that was merged later
(https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/275).

It introduced incorrect logic that broke edit lists as described in
https://gitlab.freedesktop.org/gstreamer/gstreamer/-/issues/2549

Fixes: https://gitlab.freedesktop.org/gstreamer/gstreamer/-/issues/2549
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4560>
2023-05-11 16:45:37 +00:00
Piotr Brzeziński
560d20a2c0 osxvideosink: fix deadlock upon closing output window
Invoking gst_osx_video_sink_osxwindow_destroy() can currently cause a deadlock
because showFrame() keeps trying to get the same lock as well. Moving the lock
closer to where it's actually needed seems to be enough to fix the issue for now.

Reported-by: Alexande B <abobrikovich@gmail.com>
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4559>
2023-05-11 06:35:02 +00:00
François Laignel
6675ed9aae rtpmanager/rtsession: data race leading to critical warnings
This is a fix for a data race leading to:

> GLib-CRITICAL: g_hash_table_foreach:
>   assertion 'version == hash_table->version' failed

Identified sequence:

* `rtp_session_on_timeout` acquires the lock on `session` and proceeds with its
  processing.
* `rtp_session_process_rtcp` is called (debug log : received RTCP packet) and
  attempts to acquire the lock on `session`, which is still held by
  `rtp_session_on_timeout`.
* as part of an hash table iterator, `rtp_session_on_timeout` transitively
  invokes `source_caps` which releases the lock on `session` so as to call
  `session->callbacks.caps`.
* Since `rtp_session_process_rtcp` was waiting for the lock to be released, it
  succeeds in acquiring it and proceeds with `rtp_session_process_rr` which
  transitively calls `g_hash_table_insert` via `add_source`.
* After `source_caps` re-acquires the lock and gives the control flow back to
  `rtp_session_on_timeout`, the hash table iterator is changed, resulting in the
  assertion failure.

This commits copies `sess->ssrcs[sess->mask_idx]` and iterates on the copy so
the iterator is not affected by a concurrent change due to the lock being
released in the `source_caps` callback.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4555>
2023-05-09 16:05:29 +00:00
Philippe Normand
fd194a0a2b rtpdtmfdepay: Classify as RTP element
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4582>
2023-05-09 15:18:47 +00:00
Philippe Normand
a51fd006e6 rtpdtmfsrc: Classify as RTP source
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4582>
2023-05-09 15:18:47 +00:00
Nirbheek Chauhan
93be699ab2 meson: Add more qt options and eliminate all automagic
The qt5 and qt6 plugins will now correctly error out if you enable the
option, and you can also now explicitly ensure that wayland, x11,
eglfs support is actually functional by enabling the options. It was
too easy to build non-functional support for these.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4537>
2023-05-09 13:18:38 +00:00
Tim-Philipp Müller
8b9f1278b2 jack: tone down log ERRORs in case no JACK server is running
jackaudiosink and jackaudiosrc have a rank and might be plugged
as part of auto-plugging inside playbin and playsink or the
autoaudiosink/autoaudiosrc elements, so we don't really want to
spew ERROR log messages in that case, which is consistent with
what alsasink and pulseaudiosink do.

This is less noticable on Linux because pulseaudiosink has a
higher and alsasink which has the same rank comes before jack
in the alphabet.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4545>
2023-05-08 21:20:20 +00:00
Mathieu Duponchelle
020fd3d14d videoflip: fix setting of method property at construction time
Since c2f890ab, element properties are gathered from the parse-launch
line and passed at object construction.

This caused the following issue to happen in videoflip:

* videoflip installed a CONSTRUCT property named method, now deprecated
* videoflip now also overrides that property with a video-direction
  property

GObject construction causes method to be set first at construct time,
with the user-provided value, then video-direction with the default
value.

The user-provided value was thus overridden, causing a regression.

Fix by not installing the properties as CONSTRUCT, and explicitly
implementing constructed() instead in order to ensure that we do still
call gst_video_flip_set_method() at least once during construction.

Fixes https://gitlab.freedesktop.org/gstreamer/gstreamer/-/issues/2529

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4536>
2023-05-05 08:57:04 +00:00
Camilo Celis Guzman
0cee3cd833 rtpvp8pay: rtpvp9pay: access picture_id property atomically
Atomically set and get the picture_id. This changeset only atomically gets
the picture-id when such property is queried on the element, on every other
place where it is accessed internally it is accessed directly.

This is because there is no MT scenario where we would be modifying this value
and reading it internally in parallel.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4530>
2023-05-05 07:45:19 +00:00
Camilo Celis Guzman
e4d8cda9a1 rtpvp8pay, rtpvp9pay: increment PictureID on FLUSH_START
In recent versions of Chrome (M106) a change on their jitter buffer means that
they are very susceptible to PictureID discontinuities.

Then avoid at all cost resetting the PictureID. Moreover, according to
the RFCs for VP8 and VP9 payloads; the PictureID can start off at any
random value. So there is no logical problem of incrementing it here
rather than resetting it, as long as it is a different PictureID.

WebRTC's recent corruption issue:
https://bugs.chromium.org/p/webrtc/issues/detail?id=15101

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4530>
2023-05-05 07:45:19 +00:00
Camilo Celis Guzman
f159fd8568 rtpvp8pay, rtpvp9pay: expose picture-id as a property
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4530>
2023-05-05 07:45:19 +00:00
Camilo Celis Guzman
38d5899eba rtpvp9pay: tests: remove unused struct and argument on test
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4530>
2023-05-05 07:45:19 +00:00
Camilo Celis Guzman
11187a81c3 rtpvp9pay: add picture-id-offset property
Bring the VP9 payloader in sync in this regard to the VP8 payloader

Allowing setting the picture id to a known value is useful when testing.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4530>
2023-05-05 07:45:19 +00:00
Camilo Celis Guzman
7cffb40c2e rtpvp9pay: minor refactor of PictureID logic
This brings the logic inline with the vp8pay

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4530>
2023-05-05 07:45:19 +00:00
Camilo Celis Guzman
a79616ea7a rtpvp8pay: avoid reseting PictureID if NO_PICTURE_ID mode is set
There is no logical change here, as `& (1 << nbits) - 1` would produce also 0
when NO_PICTURE_ID mode is choosen. However, this avoid computing a random
integer that is actually unused.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4530>
2023-05-05 07:45:19 +00:00
Camilo Celis Guzman
7dd6375c5e rtpvp8pay, rtpvp9pay: use GType like name for PictureIDMode
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4530>
2023-05-05 07:45:19 +00:00
Xabier Rodriguez Calvar
021572de93 qtdemux: emit no-more-pads after pruning old pads
If we don't do that, clients can rely on this signal to see the final pad
topology but it won't be the real one as some of them will disappear after
emitting that signal. This can happen after injecting a different init segment.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4535>
2023-05-03 12:06:00 +00:00
Nicolas Dufresne
3bd43672ec v4l2: device provider: Fix GMainLoop leak
On very quick start/stop, the mainloop may never be run. As a side
effect, our idle stop function is not really being ran, so we can't rely
on that to free the main loop. Simply unref the mainloop when the
thread have completely stop.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4521>
2023-05-03 10:04:58 +00:00
Carlos Rafael Giani
3fbcf5fcf3 qtdemux: Only set appsink sync property and check for async state changes
By keeping async to TRUE, a deadlock is avoided where the appsink is
filled with data after a flushing seek but before its PAUSED->PLAYING
state change finishes. If that happens, the appsink is stuck, because
its internal condition variable waits for the appsink to have more room
for data. The basesink's preroll lock is held during this, and it also
tries to acquire that lock during the state change -> deadlock.
By keeping async to TRUE, this flood of data does not happen.

Also, setting the max-buffers property to 1 is unnecessary - the test
runner will anyway detect excess memory usage if it happens.

Other property adjustments turned out to just be redundant.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4200>
2023-05-03 08:47:56 +00:00
Carlos Rafael Giani
0071c97128 qtdemux: Add audio clipping meta when playing gapless m4a content
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4200>
2023-05-03 08:47:55 +00:00
Carlos Rafael Giani
51ebda4df5 qtdemux: use qtdemux debug category instead of default in qtdemux_tags.c
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4200>
2023-05-03 08:47:55 +00:00
Tim-Philipp Müller
83026f6289 amrnb, amrwbdec: move AMR-NB and AMR-WB plugins to -good
Fedora ships these libraries as part of the main distribution now,
and they are decades old anyway so don't implement any of the newer
features.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4512>
2023-05-02 23:33:12 +00:00
François Laignel
5ef2ce69ff rtpmanager/rtsession: race conditions leading to critical warnings
While testing the [implementation for insertable streams] in `webrtcsink` &
`webrtcsrc`, I encountered critical warnings, which turned out to result from
two race conditions in `rtpsession`. Both race conditions produce:

> GLib-CRITICAL: g_hash_table_foreach:
>   assertion 'version == hash_table->version' failed

This commit fixes one of the race conditions observed.

In its simplest form, the test consists in 2 pipelines and a Signalling server:

* pipelines_sink: audiotestsrc ! webrtcsink
* pipelines_src: webrtcsrc ! appsrc

1. Set `pipelines_sink` to `Playing`.
2. The Signalling server delivers the `producer_id`.
3. Initialize `pipelines_src` to establish a session with `producer_id`.
4. Set `pipelines_src` to `Playing`.
5. Wait for a buffer to be received by the `appsrc`.
6. Set `pipelines_src` to `Null`.
7. Set `pipelines_sink` to `Null`.

The race condition happens in the following sequence:

* `webrtcsink` runs a task to periodically retrieve statistics from `webrtcbin`.
  This transitively ends up executing `rtp_session_create_stats`.
* `pipelines_sink` is set to `Null`.
* In `Paused` to `Ready`, `gst_rtp_session_change_state()` calls
  `rtp_session_reset()`.
* The assertion failure occurs when `rtp_session_reset` is called while
  `rtp_session_create_stats` is executing.

This is because `rtp_session_create_stats` acquires the lock on `session` prior
to calling `g_hash_table_foreach`, but `rtp_session_reset` doesn't acquire the
lock before calling `g_hash_table_remove_all`.

Acquiring the lock in `rtp_session_reset` fixes the issue.

[implementing insertable streams support]: https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1176

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4528>
2023-05-02 21:56:39 +00:00
Xabier Rodriguez Calvar
66c15bc753 qtdemux: Fix segfault in cenc sample grouping
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4523>
2023-05-02 11:32:01 +02:00
Nicolas Dufresne
51fa6a2656 v4l2: pool: Flush events on capture queue
Unfortunately streamoff does not flush the events, and this can cause all
sort of issues. Flush events on capture queue. We also return
GST_V4L2_FLOW_RESOLUTION_CHANGE in case a resolution change was seen.
This allow skipping streamon(capture) on flush, which could lead to a
configuration miss-match, or failure if the buffers aren't of the right
size.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4437>
2023-05-01 15:08:10 -04:00
Nicolas Dufresne
00492234bd v4l2: videodec: Detect flushes while setting up the capture
As we missed the fact we were flushing, we could create and activate
that buffer pool, and wait on it, causing a hang. We detect that we
are flushing by checking the related pad state.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4437>
2023-05-01 13:45:39 -04:00
Nicolas Dufresne
c9841a5383 v4l2: bufferpool: Don't copy buffer when flushing
Threshold handling can race with flushing operation. This can lead to
avoidable buffer copies. Simply check and return the flushing status.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4437>
2023-05-01 13:45:16 -04:00
Nicolas Dufresne
c6be3d7505 v4l2: videodec: Don't forcibly drain on resolution changes
Let the driver detects the change and reconfigure the capture side
transparently from there. This avoid reallocation of the output buffers,
and eliminates the need to stop and restart the capture task. This is
only happening if the driver have support for this, otherwise the old
behaviour is maintained.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4437>
2023-05-01 13:44:53 -04:00
Nicolas Dufresne
f58d5dfd30 v4l2: videodec: Remove the spurious srccaps probe
We don't need to probe the srccaps in set_format() anymore, this
handled already in the capture thread while setting up the capture
queue.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4437>
2023-05-01 13:44:41 -04:00
Nicolas Dufresne
4a53beeb1f v4l2: videodec: Improve few logs
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4437>
2023-05-01 13:44:37 -04:00