While the standard is a bit vague about whether the padding,
extension and marker bits should be protected:
> The usage, by senders and receivers, of the following bits shall
> be defined by the associated video/audio transport standards:
It is obviously necessary and useful for some formats (eg VP8)
that those indeed be protected.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/839>
The current encoder will hang when EOS comes. When we call the
gst_vaapi_encoder_encode_and_queue(), we should release the stream
lock, just like what we do in gst_vaapiencode_handle_frame().
The deadlock happens when: The input thread holding the stream lock
is using gst_vaapi_encoder_create_coded_buffer() to acquire a coded
buffer, while the output thread which holding the coded buffer resource
is acquiring the stream lock in _push_frame() to push the data to
down stream element.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer-vaapi/-/merge_requests/404>
https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/798
introduced a check in the need-new-fragment logic to avoid starting a
new fragment unless there has been some data on the reference stream,
but the check is done against the number of bytes that have been
received on the input, not the number that were released for output
into the current fragment.
Fix the check to remember and test against bytes that have been sent
for output.
This also fixes a problem where starting a new fragment fails to
request a new filename from the format-location signal.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/833>
These options allow to select a set of features from a given
plugin with the following syntax:
-Dgst-full-plugins=plugin1;plugin10
-Dgst-full-elements=plugin2:element1,element2
-Dgst-full-typefind-functions=plugins3:func
-Dgst-full-device-providers=plugin4,dp1
-Dgst-full-dynamic-types=plugin5:dt1
By default all the enabled plugin are registered and
gst-full-plugins will allow to include only a set of plugin
If a feature(element, typefind etc.) is selected from a plugin,
the plugin is removed from the plugins list.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-build/-/merge_requests/199>
Until now we were enforcing that only 1 signal GSource was attached
the bus but we could attach as many GSource with `gst_bus_create_watch`
as we wanted... but in the end only 1 GSource will ever be dispatched for
a given `GstMessage` leading to totally broken behavior.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/718>
The GIR parser does not want any empty line after the function or macro
name line.
Fixes the following warning:
[309/4246] Generating Gst-1.0.gir with a custom command
../subprojects/gstreamer/gst/gstelement.h:57: Warning: Gst: "@element" parameter unexpected at this location:
* @element: The element name in lower case, with words separated by '_'.
^
../subprojects/gstreamer/gst/gstelement.h:84: Warning: Gst: "@e" parameter unexpected at this location:
* @e: The element name in lower case, with words separated by '_'.
^
../subprojects/gstreamer/gst/gstelement.h:106: Warning: Gst: "@e" parameter unexpected at this location:
* @e: The element name in lower case, with words separated by '_'.
^
../subprojects/gstreamer/gst/gstdeviceprovider.h:32: Warning: Gst: "@d_p" parameter unexpected at this location:
* @d_p: The device provider name in lower case, with words separated by '_'.
^
../subprojects/gstreamer/gst/gstdynamictypefactory.h:28: Warning: Gst: "@t_n" parameter unexpected at this location:
* @t_n: The dynamic type name in lower case, with words separated by '_'.
^
../subprojects/gstreamer/gst/gsttypefind.h:34: Warning: Gst: "@type_find" parameter unexpected at this location:
* @type_find: The type find name in lower case, with words separated by '_'.
^
../subprojects/gstreamer/gst/gsttypefind.h:61: Warning: Gst: "@t_f" parameter unexpected at this location:
* @t_f: The type find name in lower case, with words separated by '_'.
^
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/719>
First: the buffer is pushed on the encoding filesink only *after* the
checkpoints, leading to wrong ordering in expectations.
The second one is about EOS being posted on the bus *before* the
stop action is executed, leading to the action never being executed.
That action was plain useless as the default behavior of validate
scenario is to internally create a "stop" action on EOS (unless
disabled by user).
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-base/-/merge_requests/968>
Otherwise, when rtpm2src cancels an inflight operation that has a queued
message stored, then the rtmp connection operation is not stopped.
If the cancellation occurs during rtmp connection start up, then
rtpm2src does not have any way of accessing the connection object as it
has not been returned yet. As a result, rtpm2src will cancel, the
connection will still be processing things and the
GMainContext/GMainLoop associated with the outstanding operation will be
destroyed. All outstanding operations and the rtmpconnection object will
therefore be leaked in this case.
Fixes: https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/issues/1425
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1862>
gst_vaapi_encoder_h264_get_pending_reordered() does not consider the
case for HIERARCHICAL_B mode. The pipeline:
gst-launch-1.0 videotestsrc num-buffers=48 ! vaapih264enc prediction-type=2 \
keyframe-period=32 ! fakesink
get a assert:
ERROR:../gst-libs/gst/vaapi/gstvaapiencoder_h264.c:1996:reflist1_init_hierarchical_b:
assertion failed: (count != 0)
The last few B frames are not fetched in correct order when HIERARCHICAL_B
is enabled.
We also fix a latent bug for normal mode. The g_queue_pop_tail() of B frames
make the last several frames encoded in reverse order. The NAL of last few
frames come in reverse order in the bit stream, though it can still output
the correct image.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer-vaapi/-/merge_requests/405>
Add a way for applications to specify that the underlying file is
growing which implies that the source won't EOS when reaching the end
of the file but instead start monitoring it and start reading it again
whenever a change is detected.
Also add a validate test to check the behavior
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-base/-/merge_requests/937>
We've been allowing only a few known chroma-site values such as
jpeg (not co-sited), mpeg2 (horizontally co-sited) and
dv (co-sited on alternate lines). That's insufficient for
representing all possible chroma-site values. By this commit,
we can represent any combination of chroma-site flags.
But, an exception here is that any combination with
GST_VIDEO_CHROMA_SITE_NONE will be considered as invalid value.
For any combination of chroma-site flags,
gst_video_chroma_to_string() method is deprecated in order to
return newly allocated string via a new gst_video_chroma_site_to_string()
method. And for consistent API naming, gst_video_chroma_from_string()
is also deprecated. Newly written code should use
gst_video_chroma_site_from_string() instead.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-base/-/merge_requests/927>
This function takes the sock lock. This can result in a deadlock when
another thread holding the sock lock is trying to take the object lock.
Thread A (Holds object lock, wants sock lock):
#2 gst_srt_object_get_stats at gst-plugins-bad/ext/srt/gstsrtobject.c:1753
#3 gst_srt_object_get_property_helper at gst-plugins-bad/ext/srt/gstsrtobject.c:409
#4 gst_srt_sink_get_property at gst-plugins-bad/ext/srt/gstsrtsink.c:95
#5 g_object_get_property from libgobject-2.0.so.0
Thread B (Holds sock lock, wants object lock):
#2 gst_element_post_message_default at gstreamer/gst/gstelement.c:2069
#3 gst_element_post_message at gstreamer/gst/gstelement.c:2123
#4 gst_element_message_full_with_details at gstreamer/gst/gstelement.c:2259
#5 gst_element_message_full at gstreamer/gst/gstelement.c:2298
#6 gst_srt_object_send_headers at gst-plugins-bad/ext/srt/gstsrtobject.c:1407
#7 gst_srt_object_send_headers at gst-plugins-bad/ext/srt/gstsrtobject.c:1444
#8 gst_srt_object_write_to_callers at gst-plugins-bad/ext/srt/gstsrtobject.c:1444
#9 gst_srt_object_write at gst-plugins-bad/ext/srt/gstsrtobject.c:1598
#10 gst_srt_sink_render at gst-plugins-bad/ext/srt/gstsrtsink.c:179
Fixes d2d00e07ac.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1861>
Various software, including ffmpeg's Decklink support, fails parsing CDP
packets that contain anything but CC data in the CDP packets.
Based on this property, timecodes are not written into the CDP packets
even if they're present.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1833>
