unlock_stop() is expected to be run while the streaming thread is idle. To
guaranty this is the case, we should take the streamlock, but its not
possible to take this lock during state transitions from PAUSED to
PLAYING as the wait function that we want to terminate is holding it.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4961>
After a create() call, which may have returned FLUSHING or a filled buffer,
if it possible that we detect that we are now in pause. As live sourced
don't produce data in pause, drop the buffer is any and later retry creating
a buffer. This will ensure that we resume from pause while avoiding displaying
ancient frame.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4961>
When the pipeline goes from Playing to Paused, this change will invoke
unlock in the derived class. When the pipeline goes from Paused to
Playing, this change will invoke unlock_stop in the derived class.
This feature was removed in commit 523de1a9 and is now being restored.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4961>
Fixes regression introduced in ba61160d6c,
where running check tests with gst-validate-launcher -f would trigger
this exception:
AttributeError: 'GstCheckTest' object has no attribute 'reports'.
Did you mean: 'reporter'?
The member `reports` is meant to be just part of GstValidateTest, but
not other subclasses, even though a usage is still found in the base
class GstTest in the method test_end().
This patch introduces an override of the methods copy() and test_end()
in GstValidateTest so that `reports` is copied and cleared respectively,
but only for validate tests.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/5281>
This is to fix an infinitely blocked upstream streaming thread if
* upstream has fixed-size buffer pool, some H/W decoders for example
* downstream returned flow error without releasing buffer
When the fixed-size buffer pool hits its configured max-buffers and
also downstream of queue returned flow error without releasing corresponding
buffer, upstream has no chance to run the next processing loop
because it will be blocked by acquire_buffer(), and therefore
downstream flow will not be propagated to upstream.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/5023>
Old versions of mesa doesn't support VASurfaceAttribDRMFormatModifiers. To
solve it, by just ignoring the modifiers assuming that linear is accepted and
produced, the creation of frames will be tried again without that attribute.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/5256>
This patch removes the code duplication of input buffer importation, in all the
va elements that import video frames. It defines a synthetic object whose
members are required to create a new input buffer and do the importation of the
upstream buffer.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/5257>
Fixes a potential GPU stall if an immediately freed texture/buffer is
attempted to be reused immediately by the CPU, e.g. when uploading.
Problematic scenario is this:
1. element does GPU processing reading from texture
2. frees the buffer back to the pool
3. pool acquire returns the just released buffer
4. GPU processing then has to wait for the previous GPU operation to
complete causing a stall
If there was a reliable way to know whether a buffer had been finished
with across all GPU drivers, we would use it. However as that does not
exist, this workaround is to keep the released buffer unusable until the
next released buffer.
This is the same approach as is used in the qml (Qt5) elements.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/5144>
Setting the surface source rectangle has been omitted so far. As a side effect
surface created with padded width/height are being scaled down. Fix this using
the viewporter source rectangle configuration. This can later be enhanced
to support crop meta.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/5259>
When we consider the DMA kind caps as input, the input_state->info
only contains the video format of GST_VIDEO_FORMAT_DMA_DRM, which
is not enough for va plugins. The new info in base encoder contains
the correct video info after the DMA caps parsing.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/5189>
Since d3d11convert and its variant elements does not enable basetransform's
passthrough, passthrough allocation query needs to be handled
manually in order to respect downstream element's min/max buffer
requirement.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/5255>
When using `gst_sdp_media_set_media_from_caps` on `application/x-rtp` caps
without `clock-rate` it wrongly reports missing payload type even if `payload`
is present in the caps.
This seems to be a copy&paste error from the error message for missing payload
type.
When using payload=10, both `clock-rate` and some other media properties are
defined by the RTP standard so I was wondering whether I could omit `clock-rate`
and was confused about the error message.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/5250>
* Library versioning should not be used for plugins since it will add
-{version}.dll suffix (and versioned libraries on Linux with symlink).
Then the library file name and plugin init function name mismatch
will result in blacklisted plugin.
* Don't define BUILDING_GST_CODECS, makes no sense
* Don't define G_LOG_DOMAIN, which should be used only for libraries,
not plugins
* Depends on gstcodecparsers libary, not gstcodecs.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/5249>
This page has been only sporadically updated for a decade, and it is
unlikely to be updated properly anytime soon. Update the top half, and
add a note about the tutorial section being out of date.
The trigger for this was a question on the mailing list about Windows
11 support, since it's not listed in the supported platforms list.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/5239>
when playing some codec such as matroska with vp9 codec,
demuxer will save information like video_mastering_display_info
and video_content_light_level in caps that decoder need,
v4l2videodecoder can use it by calling V4L2_CTRL_CLASS_COLORIMETRY
ioctl.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4403>
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/5218>
This elements pass RTP packets along unchanged and appear as a RTP
payloader element.
This is useful, for example when using the gstreamer-rtsp-server
library, in the case where you are receiving RTP packets from a
different source and want to serve them over RTSP. Since the
gst-rtsp-server library expect the element marked as payX to be a RTP
payloader element and assumes certain properties are available.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/5204>
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/5101>
Now that we can split GStreamer buffers over multiple v4l2 buffer, we may
endup waiting for these buffers to be processed. Avoid waiting for any of
the parts being processed. As a side effect, the pool will now try to
grow if the number of buffers is not sufficient, and will fail
otherwise.
This fixes a hang if the very first frame did not fit. In this case, the
driver will retrain that buffer until the capture is setup, but
GStreamer won't setup the capture until process() function have
returned.
Fixes https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/5100
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/5143>
Fix warnings from bindings changes in various plugin
examples
Fix the python mixer plugin by ensuring that PIL
is not holding a reference to mapped GstBuffer memory.
Port the filesrc example from old_examples
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/5187>
The hack enforcing strictly increasing timestamps was, according to the
code comments, because librtmp was confused with backwards timestamps.
rtmp2sink is not using librtmp as rtmpsink did, so this is no longer
required.
Also changing the timestamps is causing audio glitches when streaming to
Youtube.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/5212>
Add gst_audio_ring_buffer_set_errored() that will mark the
ringbuffer as errored only if it is currently started or paused,
so gst_audio_ringbuffer_stop() can be sure that the error
state means that the ringbuffer was started and needs stop called.
Fixes a crash with osxaudiosrc if the source element posts
an error, because the ringbuffer would not get stopped and CoreAudio
would continue trying to do callbacks.
Also, anywhere that modifies the ringbuffer state, make sure to
use atomic operations, to guarantee their visibility
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/5205>
Section 3.4 in RFC8835 states that if a WebRTC endpoint uses an HTTP
proxy to access the Internet it MUST include the "ALPN" header. This
commit adds this header.
By default the ALPN used when connecting to the TURN/TCP server via a
proxy is set to "webrtc". It can be changed by adding an alpn url
option for the http-proxy. For example:
http://user:pass@my.http.proxy.com:8080?alpn=c-webrtc
This will add the header "ALPN: c-webrtc" to the HTTP proxy CONNECT
request.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4212>
By default, macOS attempts to run lldb against a misbehaving process to handle the crash. This does not play well
with the SISEGV/SIGQUIT handler we add in gst-launch/gst-validate. The 'spinning' mechanism causes the lldb
and debugserver processes ran by macOS to misbehave, taking 100% CPU and rendering both themselves and the GStreamer
instance frozen and very hard to effectively kill. macOS's Activity Monitor is also unusable while this is happening.
This patch takes the quickest possible solution of just disabling those signal handlers entirely on macOS.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/5190>
Adds gst_queue_array_sort for sorting and gst_queue_array_push_sorted{,struct} for pushing in a sorted order.
All three functions accept a comparison GCompareDataFunc along with optional user_data to pass to it.
In gst_queue_array_sort a small workaround was needed to correctly sort non-struct arrays. Like what _find() already
does, we need to dereference our pointers first, to make sure we can use the same comparison functions everywhere.
This is done via a small wrapper around the provided comparison function.
The array can also wrap around (tail ends up 'before' the head), in which case we have to reorder the array (similar to
what do_expand() does) to then be able to use an existing sorting function, like g_qsort_with_data().
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/5112>
If a depayloader aggregates multiple RTP buffers into one buffer only
the last RTP buffer was checked for header extensions. Now the
depayloader remembers all RTP packets pushed before a output buffer is
pushed and checks all RTP buffers for header extensions.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4979>
Don't call wait_event() at all for gap events, as basesink will
end up waiting for the time that the gap event would be rendered
out at the audio device. There's no need to render it at all,
just treat it as a handy point to resync the audio if needed,
let the ringbuffer render silence, and place the next buffer
into the ringbuffer where it belongs.
The only thing we really need to do is make sure the ringbuffer
and clock are running, and wait for preroll.
Fixes https://gitlab.freedesktop.org/gstreamer/gstreamer/-/issues/2749
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/5178>
