This is how it was documented and how it worked before the port to GstPlay.
Without this, applications expecting signals to be emitted directly
without anything running the main context will simply not receive any
signals.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/5672>
d2d runtime seems to execute pending GPU command list
when DXGI ID2D1RenderTarget is being released, and it will invoke
d3d11 immediate context APIs. Should protect all rendering operations
and DXGI resources with lock.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/5659>
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>
The code seems to validate that the media-level fingerprint matches
the fingerprint of the previous media or of the whole session. There
is no such requirement in any RFC I found. The session-session one
is just meant to act as a fallback when there is no media-level
fingerprint.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1118>
This commit corrects the mapping relationship between RGB and BGR in GST and DRM.
The previous mapping was incorrect, causing potential color mismatches in the output.
The changes are as follows:
{WL_SHM_FORMAT_RGB888, DRM_FORMAT_RGB888, GST_VIDEO_FORMAT_BGR},
{WL_SHM_FORMAT_BGR888, DRM_FORMAT_BGR888, GST_VIDEO_FORMAT_RGB},
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/5620>
After talking with Vivia on IRC, she does not remember why the default
was FALSE and it is in my opinion preferable to stick to whatever
representation best represents time for a given framerate as a default
behavior.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/5628>
In the case of encoders and filters when importing a DMABuf, use
GstVideoInfoDmaDrm to get the drm fourcc and modifier.
In both cases, instead of keeping the original GstVideoInfoDmaDrm from caps, the
GstVideoInfo part of the structure is converted as canonical one, given the
format from the fourcc. It's kept in the way to handle V4L2 linear DMABufs and
to avoid too many changes in the current code.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/5264>
Instead of guessing the DRM format and modifier, pass a DRM video info to
gst_va_dmabuf_memories_setup().
Still, it checks for the DRM parameters in DRM info, if they are not available,
as in the case of V4L2 buffers, the part of the video info is used.
This is an API breakage, but since the plugin is still in stage, it's still
allowed.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/5264>
To import DMAbufs we used VASurfaceAttribExternalBuffers which works, but it's
not specific for DRM PRIME 2, since it lacks of many metadata. This patch
replaces VASurfaceAttribExternalBuffers with VADRMPRIMESurfaceDescriptor in
va_create_surfaces().
Still, this patch assumes linear modifier only.
The hack for RGB surfaces in I965 driver was pushed down into
va_create_surfaces() to avoid handling both structures.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/5264>
Should fix auto-generated follow-up sections like "Hierarchy" or
"Factory details" to be listed under the element name in the
table-of-contents of the document, instead of a stand-alone
"Duplex-Mode" section.
Also cleanup some spurious colon suffix after section names.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/5625>
Now that nvidia-vaapi-driver appeared and isn't yet supported by GstVA, we've to
add an allowed list of supported drivers.
This patch implements it adding a environment variable to disable this driver
check: GST_VA_ALL_DRIVERS
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/5616>
If text width ever reached 1px, for example after resizing the output window, the overlay would stop rendering
and never return again. The 1px condition itself does not seem to make much sense here anyway.
This was a chain of events: width reached 1, so the composition was set to NULL. Then, after resizing the output window,
push_frame() was called but would not attempt to renegotiate because composition is NULL. This caused the width/height
to never be updated again, as that only happens during negotiation, so the overlay was gone for good.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/5614>
When a new discoverer was created for a thread so discovery could
recurse we could end up removing the wrong discoverer info from the
cache leading to freeing it while it was still discovering URIS, which
lead to the following assertion:
``` bt
Thread 1 (Thread 0x7fcc2e1a5840 (LWP 1855496)):
#0 __pthread_kill_implementation (threadid=<optimized out>, signo=signo@entry=6, no_tid=no_tid@entry=0) at pthread_kill.c:44
#1 0x00007fcc2e9d98a3 in __pthread_kill_internal (signo=6, threadid=<optimized out>) at pthread_kill.c:78
#2 0x00007fcc2e9878ee in __GI_raise (sig=sig@entry=6) at ../sysdeps/posix/raise.c:26
#3 0x00007fcc2e96f8ff in __GI_abort () at abort.c:79
#4 0x00007fcc2ed80056 in g_assertion_message (domain=domain@entry=0x7fcc2f2c19f9 "GES", file=file@entry=0x7fcc2f2dfd68 "../subprojects/gst-editing-services/ges/ges-discoverer-manager.c", line=line@entry=20, func=func@entry=0x7fcc2f2e0030 <__func__.7> "ges_discoverer_data_free", message=message@entry=0x12dab70 "assertion failed: (data->n_uri == 0)") at ../glib/gtestutils.c:3497
#5 0x00007fcc2ede1d87 in g_assertion_message_expr (domain=domain@entry=0x7fcc2f2c19f9 "GES", file=file@entry=0x7fcc2f2dfd68 "../subprojects/gst-editing-services/ges/ges-discoverer-manager.c", line=line@entry=20, func=func@entry=0x7fcc2f2e0030 <__func__.7> "ges_discoverer_data_free", expr=expr@entry=0x7fcc2f2dfcf1 "data->n_uri == 0") at ../glib/gtestutils.c:3523
#6 0x00007fcc2f2bd5c5 in ges_discoverer_data_free (data=0x160e430) at ../subprojects/gst-editing-services/ges/ges-discoverer-manager.c:20
#7 0x00007fcc2ed8509d in g_atomic_rc_box_release_full (clear_func=0x7fcc2f2bd4f0 <ges_discoverer_data_free>, mem_block=0x160e430) at ../glib/garcbox.c:355
#8 g_atomic_rc_box_release_full (mem_block=0x160e430, clear_func=0x7fcc2f2bd4f0 <ges_discoverer_data_free>) at ../glib/garcbox.c:338
#9 0x00007fcc2eda6809 in g_hash_table_remove_internal (notify=1, key=0x10448a0, hash_table=0x12e0be0) at ../glib/ghash.c:1776
#10 g_hash_table_remove (hash_table=0x12e0be0, key=0x10448a0) at ../glib/ghash.c:1804
#11 0x00007fcc2f2bd95f in cleanup_discoverer_cb (discoverer_data=discoverer_data@entry=0x13e7000) at ../subprojects/gst-editing-services/ges/ges-discoverer-manager.c:379
#12 0x00007fcc2edbc759 in g_timeout_dispatch (source=0x15a6060, callback=0x7fcc2f2bd910 <cleanup_discoverer_cb>, user_data=0x13e7000) at ../glib/gmain.c:5121
#13 0x00007fcc2edbbe1c in g_main_dispatch (context=0x1044700) at ../glib/gmain.c:3476
#14 g_main_context_dispatch_unlocked (context=0x1044700) at ../glib/gmain.c:4284
#15 0x00007fcc2ee16d78 in g_main_context_iterate_unlocked.isra.0 (context=0x1044700, block=block@entry=1, dispatch=dispatch@entry=1, self=<optimized out>) at ../glib/gmain.c:4349
#16 0x00007fcc2edbd407 in g_main_loop_run (loop=0x12ccbd0) at ../glib/gmain.c:4551
#17 0x00007fcc2f285791 in ges_uri_clip_asset_request_sync (uri=uri@entry=0x12d7980 "file:///var/home/phil/gstreamer/build/subprojects/gst-integration-testsuites/logs/ges/scenarios/check_seek_on_very_deeply_nested_timeline/nested_timeline_depth6.xges", error=error@entry=0x7fff499015a8) at ../subprojects/gst-editing-services/ges/ges-uri-asset.c:688
#18 0x00007fcc2f28949b in ges_project_create_asset_sync (project=0x12c1c70, id=id@entry=0x12d7980 "file:///var/home/phil/gstreamer/build/subprojects/gst-integration-testsuites/logs/ges/scenarios/check_seek_on_very_deeply_nested_timeline/nested_timeline_depth6.xges", extractable_type=extractable_type@entry=Python Exception <class 'gdb.error'>: value has been optimized out , error=error@entry=0x7fff499015a8) at ../subprojects/gst-editing-services/ges/ges-project.c:959
#19 0x00007fcc2f2ba484 in _ges_get_asset_from_timeline (timeline=timeline@entry=0x12bdc80, type=type@entry=Python Exception <class 'gdb.error'>: value has been optimized out , id=id@entry=0x12d7980 "file:///var/home/phil/gstreamer/build/subprojects/gst-integration-testsuites/logs/ges/scenarios/check_seek_on_very_deeply_nested_timeline/nested_timeline_depth6.xges", error=error@entry=0x7fff49901728) at ../subprojects/gst-editing-services/ges/ges-structured-interface.c:540
#20 0x00007fcc2f2ba9b2 in _ges_add_clip_from_struct (timeline=0x12bdc80, structure=0x157f690, error=0x7fff49901728) at ../subprojects/gst-editing-services/ges/ges-structured-interface.c:697
#21 0x00007fcc2f2b6a9d in _validate_action_execute (scenario=0x15f7620, action=0x157f500) at ../subprojects/gst-editing-services/ges/ges-validate.c:922
#22 0x00007fcc2eef5c9c in gst_validate_execute_action (action=0x157f500, action_type=0x13e0500) at ../subprojects/gst-devtools/validate/gst/validate/gst-validate-scenario.c:2541
#23 gst_validate_execute_action (action_type=0x13e0500, action=0x157f500) at ../subprojects/gst-devtools/validate/gst/validate/gst-validate-scenario.c:2507
#24 0x00007fcc2eef8ce3 in execute_next_action_full (scenario=<optimized out>, message=<optimized out>) at ../subprojects/gst-devtools/validate/gst/validate/gst-validate-scenario.c:2782
#25 0x00007fcc2eef9dee in _action_set_done (action=0x157efb0) at ../subprojects/gst-devtools/validate/gst/validate/gst-validate-scenario.c:6368
#26 0x00007fcc2edbd26d in g_main_context_invoke_full (context=0x1044700, priority=200, function=0x7fcc2eef9ab0 <_action_set_done>, data=0x157efb0, notify=0x7fcc2eeea5d0 <gst_validate_action_unref>) at ../glib/gmain.c:6533
#27 0x00007fcc2eef6cf2 in gst_validate_action_set_done (action=0x157efb0) at ../subprojects/gst-devtools/validate/gst/validate/gst-validate-scenario.c:6411
#28 0x00007fcc2eef9018 in execute_next_action_full (scenario=<optimized out>, message=<optimized out>) at ../subprojects/gst-devtools/validate/gst/validate/gst-validate-scenario.c:2803
#29 0x00007fcc2eef9dee in _action_set_done (action=0x157ea60) at ../subprojects/gst-devtools/validate/gst/validate/gst-validate-scenario.c:6368
#30 0x00007fcc2edbd26d in g_main_context_invoke_full (context=0x1044700, priority=200, function=0x7fcc2eef9ab0 <_action_set_done>, data=0x157ea60, notify=0x7fcc2eeea5d0 <gst_validate_action_unref>) at ../glib/gmain.c:6533
#31 0x00007fcc2eef6cf2 in gst_validate_action_set_done (action=0x157ea60) at ../subprojects/gst-devtools/validate/gst/validate/gst-validate-scenario.c:6411
#32 0x00007fcc2eef9018 in execute_next_action_full (scenario=<optimized out>, message=<optimized out>) at ../subprojects/gst-devtools/validate/gst/validate/gst-validate-scenario.c:2803
#33 0x00007fcc2eef9dee in _action_set_done (action=0x157e510) at ../subprojects/gst-devtools/validate/gst/validate/gst-validate-scenario.c:6368
#34 0x00007fcc2edbd26d in g_main_context_invoke_full (context=0x1044700, priority=200, function=0x7fcc2eef9ab0 <_action_set_done>, data=0x157e510, notify=0x7fcc2eeea5d0 <gst_validate_action_unref>) at ../glib/gmain.c:6533
#35 0x00007fcc2eef6cf2 in gst_validate_action_set_done (action=0x157e510) at ../subprojects/gst-devtools/validate/gst/validate/gst-validate-scenario.c:6411
#36 0x00007fcc2eef9018 in execute_next_action_full (scenario=<optimized out>, message=<optimized out>) at ../subprojects/gst-devtools/validate/gst/validate/gst-validate-scenario.c:2803
#37 0x00007fcc2eef9dee in _action_set_done (action=0x157df10) at ../subprojects/gst-devtools/validate/gst/validate/gst-validate-scenario.c:6368
#38 0x00007fcc2edbd26d in g_main_context_invoke_full (context=0x1044700, priority=200, function=0x7fcc2eef9ab0 <_action_set_done>, data=0x157df10, notify=0x7fcc2eeea5d0 <gst_validate_action_unref>) at ../glib/gmain.c:6533
#39 0x00007fcc2eef6cf2 in gst_validate_action_set_done (action=0x157df10) at ../subprojects/gst-devtools/validate/gst/validate/gst-validate-scenario.c:6411
#40 0x00007fcc2eef9018 in execute_next_action_full (scenario=<optimized out>, message=<optimized out>) at ../subprojects/gst-devtools/validate/gst/validate/gst-validate-scenario.c:2803
#41 0x00007fcc2eef9dee in _action_set_done (action=0x157d9e0) at ../subprojects/gst-devtools/validate/gst/validate/gst-validate-scenario.c:6368
#42 0x00007fcc2edbd26d in g_main_context_invoke_full (context=0x1044700, priority=200, function=0x7fcc2eef9ab0 <_action_set_done>, data=0x157d9e0, notify=0x7fcc2eeea5d0 <gst_validate_action_unref>) at ../glib/gmain.c:6533
#43 0x00007fcc2eef6cf2 in gst_validate_action_set_done (action=0x157d9e0) at ../subprojects/gst-devtools/validate/gst/validate/gst-validate-scenario.c:6411
#44 0x00007fcc2eef9018 in execute_next_action_full (scenario=<optimized out>, message=<optimized out>) at ../subprojects/gst-devtools/validate/gst/validate/gst-validate-scenario.c:2803
#45 0x00007fcc2eef9dee in _action_set_done (action=0x157d3e0) at ../subprojects/gst-devtools/validate/gst/validate/gst-validate-scenario.c:6368
#46 0x00007fcc2edbd26d in g_main_context_invoke_full (context=0x1044700, priority=200, function=0x7fcc2eef9ab0 <_action_set_done>, data=0x157d3e0, notify=0x7fcc2eeea5d0 <gst_validate_action_unref>) at ../glib/gmain.c:6533
#47 0x00007fcc2eef6cf2 in gst_validate_action_set_done (action=0x157d3e0) at ../subprojects/gst-devtools/validate/gst/validate/gst-validate-scenario.c:6411
#48 0x00007fcc2eef9018 in execute_next_action_full (scenario=<optimized out>, message=<optimized out>) at ../subprojects/gst-devtools/validate/gst/validate/gst-validate-scenario.c:2803
#49 0x00007fcc2eef9dee in _action_set_done (action=0x157cf70) at ../subprojects/gst-devtools/validate/gst/validate/gst-validate-scenario.c:6368
#50 0x00007fcc2edbd26d in g_main_context_invoke_full (context=0x1044700, priority=200, function=0x7fcc2eef9ab0 <_action_set_done>, data=0x157cf70, notify=0x7fcc2eeea5d0 <gst_validate_action_unref>) at ../glib/gmain.c:6533
#51 0x00007fcc2eef6cf2 in gst_validate_action_set_done (action=0x157cf70) at ../subprojects/gst-devtools/validate/gst/validate/gst-validate-scenario.c:6411
#52 0x00007fcc2eef9018 in execute_next_action_full (scenario=<optimized out>, message=<optimized out>) at ../subprojects/gst-devtools/validate/gst/validate/gst-validate-scenario.c:2803
#53 0x00007fcc2eef9dee in _action_set_done (action=0x157cb00) at ../subprojects/gst-devtools/validate/gst/validate/gst-validate-scenario.c:6368
#54 0x00007fcc2edbd26d in g_main_context_invoke_full (context=0x1044700, priority=200, function=0x7fcc2eef9ab0 <_action_set_done>, data=0x157cb00, notify=0x7fcc2eeea5d0 <gst_validate_action_unref>) at ../glib/gmain.c:6533
#55 0x00007fcc2eef6cf2 in gst_validate_action_set_done (action=0x157cb00) at ../subprojects/gst-devtools/validate/gst/validate/gst-validate-scenario.c:6411
#56 0x00007fcc2eef9018 in execute_next_action_full (scenario=<optimized out>, message=<optimized out>) at ../subprojects/gst-devtools/validate/gst/validate/gst-validate-scenario.c:2803
#57 0x00007fcc2eef9dee in _action_set_done (action=0x157c690) at ../subprojects/gst-devtools/validate/gst/validate/gst-validate-scenario.c:6368
#58 0x00007fcc2edbd26d in g_main_context_invoke_full (context=0x1044700, priority=200, function=0x7fcc2eef9ab0 <_action_set_done>, data=0x157c690, notify=0x7fcc2eeea5d0 <gst_validate_action_unref>) at ../glib/gmain.c:6533
#59 0x00007fcc2eef6cf2 in gst_validate_action_set_done (action=0x157c690) at ../subprojects/gst-devtools/validate/gst/validate/gst-validate-scenario.c:6411
#60 0x00007fcc2eef9018 in execute_next_action_full (scenario=<optimized out>, message=<optimized out>) at ../subprojects/gst-devtools/validate/gst/validate/gst-validate-scenario.c:2803
#61 0x00007fcc2eef9dee in _action_set_done (action=0x157c220) at ../subprojects/gst-devtools/validate/gst/validate/gst-validate-scenario.c:6368
#62 0x00007fcc2edbd26d in g_main_context_invoke_full (context=0x1044700, priority=200, function=0x7fcc2eef9ab0 <_action_set_done>, data=0x157c220, notify=0x7fcc2eeea5d0 <gst_validate_action_unref>) at ../glib/gmain.c:6533
#63 0x00007fcc2eef6cf2 in gst_validate_action_set_done (action=0x157c220) at ../subprojects/gst-devtools/validate/gst/validate/gst-validate-scenario.c:6411
#64 0x00007fcc2eef9018 in execute_next_action_full (scenario=<optimized out>, message=<optimized out>) at ../subprojects/gst-devtools/validate/gst/validate/gst-validate-scenario.c:2803
#65 0x00007fcc2eef9dee in _action_set_done (action=0x15233c0) at ../subprojects/gst-devtools/validate/gst/validate/gst-validate-scenario.c:6368
#66 0x00007fcc2edbd26d in g_main_context_invoke_full (context=0x1044700, priority=200, function=0x7fcc2eef9ab0 <_action_set_done>, data=0x15233c0, notify=0x7fcc2eeea5d0 <gst_validate_action_unref>) at ../glib/gmain.c:6533
#67 0x00007fcc2eef6cf2 in gst_validate_action_set_done (action=0x15233c0) at ../subprojects/gst-devtools/validate/gst/validate/gst-validate-scenario.c:6411
#68 0x00007fcc2eef9018 in execute_next_action_full (scenario=<optimized out>, message=<optimized out>) at ../subprojects/gst-devtools/validate/gst/validate/gst-validate-scenario.c:2803
#69 0x00007fcc2eef9dee in _action_set_done (action=0x1522f80) at ../subprojects/gst-devtools/validate/gst/validate/gst-validate-scenario.c:6368
#70 0x00007fcc2edbd26d in g_main_context_invoke_full (context=0x1044700, priority=200, function=0x7fcc2eef9ab0 <_action_set_done>, data=0x1522f80, notify=0x7fcc2eeea5d0 <gst_validate_action_unref>) at ../glib/gmain.c:6533
#71 0x00007fcc2eef6cf2 in gst_validate_action_set_done (action=0x1522f80) at ../subprojects/gst-devtools/validate/gst/validate/gst-validate-scenario.c:6411
#72 0x00007fcc2eef9018 in execute_next_action_full (scenario=<optimized out>, message=<optimized out>) at ../subprojects/gst-devtools/validate/gst/validate/gst-validate-scenario.c:2803
#73 0x00007fcc2eef9dee in _action_set_done (action=0x1522ae0) at ../subprojects/gst-devtools/validate/gst/validate/gst-validate-scenario.c:6368
#74 0x00007fcc2edbd26d in g_main_context_invoke_full (context=0x1044700, priority=200, function=0x7fcc2eef9ab0 <_action_set_done>, data=0x1522ae0, notify=0x7fcc2eeea5d0 <gst_validate_action_unref>) at ../glib/gmain.c:6533
#75 0x00007fcc2eef6cf2 in gst_validate_action_set_done (action=0x1522ae0) at ../subprojects/gst-devtools/validate/gst/validate/gst-validate-scenario.c:6411
#76 0x00007fcc2eef9018 in execute_next_action_full (scenario=<optimized out>, message=<optimized out>) at ../subprojects/gst-devtools/validate/gst/validate/gst-validate-scenario.c:2803
#77 0x00007fcc2eef9dee in _action_set_done (action=0x1522190) at ../subprojects/gst-devtools/validate/gst/validate/gst-validate-scenario.c:6368
#78 0x00007fcc2edbd26d in g_main_context_invoke_full (context=0x1044700, priority=200, function=0x7fcc2eef9ab0 <_action_set_done>, data=0x1522190, notify=0x7fcc2eeea5d0 <gst_validate_action_unref>) at ../glib/gmain.c:6533
#79 0x00007fcc2eef6cf2 in gst_validate_action_set_done (action=0x1522190) at ../subprojects/gst-devtools/validate/gst/validate/gst-validate-scenario.c:6411
#80 0x00007fcc2eef9018 in execute_next_action_full (scenario=<optimized out>, message=<optimized out>) at ../subprojects/gst-devtools/validate/gst/validate/gst-validate-scenario.c:2803
#81 0x00007fcc2eef9dee in _action_set_done (action=0x1520ea0) at ../subprojects/gst-devtools/validate/gst/validate/gst-validate-scenario.c:6368
#82 0x00007fcc2edbd26d in g_main_context_invoke_full (context=0x1044700, priority=200, function=0x7fcc2eef9ab0 <_action_set_done>, data=0x1520ea0, notify=0x7fcc2eeea5d0 <gst_validate_action_unref>) at ../glib/gmain.c:6533
#83 0x00007fcc2eef6cf2 in gst_validate_action_set_done (action=0x1520ea0) at ../subprojects/gst-devtools/validate/gst/validate/gst-validate-scenario.c:6411
#84 0x00007fcc2eef9018 in execute_next_action_full (scenario=<optimized out>, message=<optimized out>) at ../subprojects/gst-devtools/validate/gst/validate/gst-validate-scenario.c:2803
#85 0x00007fcc2edbc759 in g_timeout_dispatch (source=0x14b6340, callback=0x7fcc2eef99c0 <execute_next_action>, user_data=0x15f7620) at ../glib/gmain.c:5121
#86 0x00007fcc2edbbe1c in g_main_dispatch (context=0x1044700) at ../glib/gmain.c:3476
#87 g_main_context_dispatch_unlocked (context=0x1044700) at ../glib/gmain.c:4284
#88 0x00007fcc2ee16d78 in g_main_context_iterate_unlocked.isra.0 (context=context@entry=0x1044700, block=block@entry=1, dispatch=dispatch@entry=1, self=<optimized out>) at ../glib/gmain.c:4349
#89 0x00007fcc2edb9a93 in g_main_context_iteration (context=context@entry=0x1044700, may_block=may_block@entry=1) at ../glib/gmain.c:4414
#90 0x00007fcc2ec14c3d in g_application_run (application=application@entry=0x1042ab0, argc=argc@entry=4, argv=argv@entry=0x7fff499031e8) at ../gio/gapplication.c:2577
#91 0x0000000000405dfd in real_main (argv=0x7fff499031e8, argc=4) at ../subprojects/gst-editing-services/tools/ges-launch.c:38
#92 main (argc=4, argv=0x7fff499031e8) at ../subprojects/gst-editing-services/tools/ges-launch.c:56
```
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/5619>
Suppose you have a project where GStreamer and wayland-protocols are
pulled in as dependencies via .wrap files. In that case, Meson's setup
step will fail for gst-plugins-bad with the message "Sandbox violation:
Tried to grab file viewporter.xml outside current (sub)project." To
avoid this exception, one should use Meson's `files` and `join_paths`
functions. The suggested solution is identical to how GTK 4 processes
Wayland files.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/5593>
In rare cases - notably on macOS, because of multiple GL contexts - the lack of a sync point was causing overlays
to disappear for a frame after being redrawn, or sometimes not appear at all. This change makes sure that the display
in one GL context will be correctly synchronised with the other GL context where the overlay texture was uploaded.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/5610>
The manager keeps track of one discoverer per thread and in large applications
with hundreds of threads this can significantly increase memory pressure. So we
need to periodically clean-up the unused discoverers.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/5608>
When using deeply nested timelines with the `ges:` protocol the
formatters ends up trying to do discovery from the same thread current
discovery happens, leading to infinite freeze as GstDiscoverer can't run
several discoveries at the same time.
By ensuring that when calling `gst_discoverer_discover_uri_async` no
`GstDiscoverer` is set as "thread discoverer" we know that another
discoverer will be created if discovery recurses, effectively removing
the freeze.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/5608>
A payload of 0x80 0x80 means that it's padding. It's not a good idea to
throw this away though, because of the cc_valid field.
According to CEA 10-B section 25.2.1, if cc_valid is zero, the run-in
clock and start bit should not be generated. In practice, this means
that any closed captions will be erased and the end-user TV will show
that captions are not available for this stream. This might have
undesired consequences, e.g. we were just showing a long line of
captions and we disable it before the user has had time to read it, or
you can't enable closed captions during silence/music intervals.
We cannot reliably detect whether there's a currently-silent closed
caption stream or just nothing, but we have this information coming from
upstream, so we can at least not discard it.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/5508>
When decoding stream using hardware V4L2 decoder element, in any of the
currently supported formats, the decoding will fail once frame number
1000000 is reached. The reported error clearly indicates a wrap-around
occured, instead of receiving decoded frame 1000000, frame 0 is received
from the hardware V4L2 decoder driver.
The problem is actually not in the driver itself, but rather in gstreamer,
which uses `struct v4l2_buffer` member `.timestamp` in a special way. The
timestamp of buffers with encoded data added to the SINK (input) queue of
the driver is copied by the driver into matching buffers with decoded data
added to the SOURCE (output) queue of the driver. In fact, the timestamp
is not a timestamp at all, but rather in this special case, only part of
it is used as an incrementing frame counter.
The `.timestamp` is of type `struct timeval`, which is defined in
`sys/time.h` [1]. Only the `tv_usec` member of this structure is used
for the incrementing frame counter. However, suseconds_t tv_usec [2]
may be limited to range [-1, 1000000]:
"
[XSI] The type suseconds_t shall be a signed integer type capable of
storing values at least in the range [-1, 1000000].
"
Therefore, once frame 1000000 is reached, a rollover occurs and decoding
fails.
Fix this by using both `struct timeval` members, `.tv_sec` and `.tv_usec`
with matching modular arithmetic, this way the failure would occur again
just short of 2^84 frames, which should be plenty.
[1] https://pubs.opengroup.org/onlinepubs/9699919799/basedefs/sys_time.h.html
[2] https://pubs.opengroup.org/onlinepubs/9699919799/basedefs/sys_types.h.html
A test case using stateless hardware h264 decoder, the WARN/ERROR output
in gstreamer log indicates a failure occurred. With this change, that
error no longer occurs and the WARN/ERROR are not present:
```
pc$ gst-launch-1.0 videotestsrc num-buffers=1001001 pattern=6 ! \
video/x-raw,width=16,height=16,format=I420 ! \
x264enc ! filesink location=/tmp/test.h264
dut$ GST_DEBUG="*:3" gst-launch-1.0 filesrc location=/tmp/test.h264 ! \
h264parse ! v4l2slh264dec ! fakesink
...
0:03:51.393677606 12111 0x370df400 WARN \
v4l2codecs-decoder gstv4l2decoder.c:1157:gst_v4l2_request_set_done:<v4l2decoder2> \
Requested frame 1000000, but driver returned frame 0.
0:03:51.394140597 12111 0x370df400 WARN \
v4l2codecs-decoder gstv4l2decoder.c:1157:gst_v4l2_request_set_done:<v4l2decoder2> \
Requested frame 1000001, but driver returned frame 1.
0:03:51.394425216 12111 0x370df400 WARN \
v4l2codecs-decoder gstv4l2decoder.c:1157:gst_v4l2_request_set_done:<v4l2decoder2> \
Requested frame 1000002, but driver returned frame 2.
0:03:51.394665211 12111 0x370df400 WARN \
v4l2codecs-decoder gstv4l2decoder.c:1157:gst_v4l2_request_set_done:<v4l2decoder2> \
Requested frame 1000003, but driver returned frame 3.
0:03:51.394785833 12111 0x370df400 WARN \
v4l2codecs-h264dec gstv4l2codech264dec.c:1059:gst_v4l2_codec_h264_dec_output_picture:<v4l2slh264dec0> \
error: Failed to decode frame 1000000
ERROR: from element /GstPipeline:pipeline0/v4l2slh264dec:v4l2slh264dec0: Failed to decode frame 1000000
```
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/5598>
This pair of elements, inspired from shmsink/shmsrc, send unix file
descriptors (e.g. memfd, dmabuf) from one sink to multiple source
elements in other processes.
The unixfdsink proposes a memfd/shm allocator, which causes for example
videotestsrc to write directly into memories that can be transfered to
other processes without copying.
Sponsored-by: Netflix Inc.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/5328>
There is no guarantee that g_get_user_runtime_dir() is in a tmpfs. Using
an explicit shared memory API seems safer for all POSIX platforms.
Note that Android does not have shm_open() and only added memfd_create()
since API level 30 (Android 11).
Sponsored-by: Netflix Inc.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/5328>
Detail a bit the intention behind GST_ALLOCATOR_FLAG_CUSTOM_ALLOC, even
if implementation does not currently fully follow that usage. Introduce
a new flag specifically for copying memories using the default system
allocator.
Sponsored-by: Netflix Inc.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/5328>
The src caps of the libde265 is now fixed to I420, and so if the
stream is other format, such as 4:4:4 or 10 bits format, the pipeline
will crash because the dowstream element accesses the video buffer as
I420 format.
We now restrain the input caps to "main" profile, which only contains
4:2:0 8 bits stream.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/5573>
While the minimum timeout duration is 5s, checking only every 5s means
that we would notice this 4.9s too late in the worst case.
Checking once a second reduces this considerably while keeping the
number of wakeups still low.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/5520>
Otherwise it can happen that we regularly switch back and forth between
clocks under certain circumstances for no good reason.
Also remove redundant comparison when comparing the steps removed between two
clocks.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/5520>
This can happen with the dummy "noopenh264" library that the freedesktop
flatpak runtime ships, and Fedora is planning on shipping as well. In
both cases the dummy implementation gets replaced with the actual
openh264 library that's downloaded directly from Cisco, but just to be
on safe side, this patch makes it careful to check the return values to
avoid crashing if the underlying library hasn't been swapped out yet.
The patch is taken from freedesktop-sdk and was originally written by
Valentin David <valentin.david@codethink.co.uk>.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/5581>
- Fixes a crash in tutorial 5, which happened when going back from video playback to the 'library' view, due to
ui_delegate already being destroyed at that point.
- Updates layouts to avoid navigation bar overlapping play/pause buttons. Colours are now correctly updated
based on light/dark mode being enabled, overall look and feel is improved with bigger buttons and paddings.
New button types are used, so target version is now iOS 15.0.
- Disables debug log coloring, as the default terminal in XCode does not render that anyway, so logs are now
more readable there.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/5473>
The local glib subproject doesn't exist so the glib/glib.supp file
cannot be included.
As it is needed for the do_lookup_by_name() function call, let's add the
system wide suppression file so that its version matches the installed glib
version.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/5434>
If users update geometry related properties very frequently
for a stream to be animated, redrawing on every update
can make rendering choppy or can be a performance bottleneck.
To address the issue, adding a property to control the behavior
of redrawing scene when geometry related properties are updated.
Also, do not resize swapchain on such property update, since
re-allocating backbuffer and multi-sampled render target is
unnecessary in that case.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/5575>
Other Windows applications allow window switching even when
an application window is in fullscreen mode. Also fixing
regression introduced in 15248d8b84
which makes restored window is always located at topmost
since we do not call SetWindowPos() anymore when restoring
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/5574>
ISimpleAudioVolume controls volume of corresponding audio session
and there would be only single input/output audio session
in case of share-mode, which means that it controls audio volume of the
process. Instead, use IAudioStreamVolume interface which controls
volume of the stream.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/5549>
Ignore alpha component of source (mouse cursor texture)
when blending alpha channel, otherwise the background area of source
(which has zeros) will be written to render target. Then it will result
in black rectangle if output texture is converted to premultiplied alpha
texture
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/5566>
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>
It's a bad idea trying to mix the Options from GStramer and
GTK, in addition with cli argument being a bit wonky thing for
GUI applications in general. In the rare, now, occasion
that an application wants to parse arguments, its preferable
to parse them manually and use library apis afterwards
rather than trying to combine the option groups and hope it
works.
In addition, applications should be opening files using
`g_application_open` instead of parsing random arguments.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4788>
When hotdoc documentation is enabled and opencv plugin is set as
auto-detected, but the library isn't installed, meson configuration fails
with this message:
../subprojects/gst-plugins-bad/docs/meson.build:139:21: ERROR: Unknown variable "gstopencv_dep".
This patch fixes this case defined gstopencv_dep as disabler() when
dependencies aren't found.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/5560>
Current codes try derive image in _update_image_info first, if
derive returns no error, the va_allocator->info is the one from derived
image, but in va_map_unlocked, we disable derive manner for d3d backend
because it doesn't seem to work, this will cause issue for d3d path,
i.e. possibly using derived info in va_get_image to do mapping...
This patch disables derive image for d3d backend in _update_image_info,
to ensure we only use info from va_create_image for d3d path.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/5495>
An operation is an arbitrary amount of work to be executed on the host, a
device, or an external entity such as a presentation engine.
The purpose of this object is to help on the operation's synchronization
through declaring explicit execution dependencies, and memory dependencies
between two sets of operations defined by the command’s two synchronization
scopes.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/5079>
While VkPipelineStageFlags is an enum (arguably backed as uint32 in 32bit
platforms), VkPipelineStageFlags2 is a redefinition of guint64; likewise for
VkAccessFlags and VkAccessFlags2.
This patch types both members in GstVulkanBarrierMemoryInfo as guint64 for
compatibility, so it could be used with or without synchronization2 vulkan
extension.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/5079>
Previously we were checking for opencv dep in 2 different places,
and the checks would vary in terms of how complex and exhaustive
they were.
Move the check into the libs module and reuse the result later on.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3016>
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>
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>
This field is used to store gbooleans (which are ints) but if it's
a :1 bit depth assigning ints to it changes it's value as the only
valid values are -1 and 0.
Make it a guint instead so the cast would be correct.
```
../subprojects/gst-plugins-bad/gst-libs/gst/vulkan/xcb/gstvkwindow_xcb.c:151:25: error:
implicit truncation from 'int' to a one-bit wide bit-field changes value from 1 to -1
[-Werror,-Wsingle-bit-bitfield-constant-conversion]
window_xcb->visible = TRUE;
```
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/5432>
Followup to 75872c802b , clang version
While we ignore `discarded-qualifiers` already for gcc, clang seems
to assign this error to `incompatible-pointer-types-discards-qualifiers`
so we need to ignore that as well.
```
In file included from \
../subprojects/gst-plugins-base/ext/cdparanoia/gstcdparanoiasrc.h:37: \
/usr/include/cdda/cdda_interface.h:164:3: error: initializing 'char *' with an expression \
of type 'const char [8]' discards qualifiers [-Werror, \
-Wincompatible-pointer-types-discards-qualifiers]
"Success",
^~~~~~~~~
```
See 75872c802b for more
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/5474>
This was wrongly calling the base class method, which unnecessairly took the stream lock, already taken by
handle_frame(). The drain() call in negotiate() would then wait for the output loop to pause, while that loop
is stuck waiting to take the stream lock, thus causing a deadlock.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/5521>
This element refactors functionality from gstonnxinference element,
namely separating out the ONNX inference from the subsequent analysis.
The new element runs an ONNX model on each video frame, and then
attaches a TensorMeta meta with the output tensor data. This tensor data
will then be consumed by downstream elements such as gstobjectdetector.
At the moment, a provisional TensorMeta is used just in the ONNX
plugin, but in future this will upgraded to a GStreamer API for other
plugins to consume.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4916>
- Don't try to make the parameters match `GHFunc`. Use a dedicated
callback for `g_hash_table_foreach`.
- Don't try to be clever with buffer memories. We're allocating a full
packet anyway, might as well memcpy and save on a lot of complexity.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/5496>
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>
There are a bunch of plugins that you need for webrtc support, and
it's not obvious at all to users which those are.
With this commit, srtp, sctp and dtls options will be auto-enabled if
the webrtc option is enabled.
Requires meson 1.1
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/5505>
In _gl_memory_upload_propose_allocation(), when output target is "external-oes",
then we should not provide GL allocator and pool in the allocation query.
This is because the "external-oes" kind memory can never be mapped directly
and the upstream element may misuse it.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/5468>
The propose and decide allocation vfuncs are called directly from
basetransform and need to use the locked accessor function for
retrieving a reliable reference to the GstGLContext (if available)
Fixes spurious crashes on shutdown during pad reconfiguration
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/5437>
The v4l2codecs H.265 decoder uses the
GstH265SliceHdr::entry_point_offset_minus1 array so make sure that it is not
freed before decoding the frame.
Before this patch, some H.265 input would segfault in
gst_v4l2_codec_h265_dec_fill_slice_params() when executing the line:
guint32 entry_point_offset = slice_hdr->entry_point_offset_minus1[i] + 1;
Make sure that the array is not freed before using it.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/5499>
The number of planes is a meta we carry around in the GstVideoMeta with
DMA_DRM format. In cannot be decuded correctly from knowledge of the
base format. Notably, some compression modifier may introduce an extra
plane to store the compression parameters.
So use n_planes from GstVideoMeta and pass this explicitly when
importing to EGLImage.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/5461>
The DMAbuf accept function was ensuring the in_dma_info values was valid if
the in_caps have change. But the check was bogus since the in_caps was being
modified without a pointer change. As a side effect, on the second accept
call, the drm_fourcc was reset to 0, which cause the uploader to fallback.
Fix this by ensuring we always have a valid dma_frm info directly in the
set_caps() function. Also remove the bogus caps changed check and remove any
modification to the info structure and always do that inner checks.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/5461>
DRM Modifiers are not generically transferrable from a format like NV12 to
their indirect shading format (R8 / RG88). So the helper to this do needs
to be removed from our API.
To make things worse, we support indirect formats that aren't DRM format in
the first place. Notably NV12_16L32 (aka MM21) is not (yet) a DRM format. Yet,
each plane can be indirectly imported using R8/RG88 and a detiling shader.
This patch also removes this constraint restoring zero-copy playback on
Mediatek SoC.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/5461>
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>
Update connection-speed at runtime in playbin, uridecodebin and decodebin
also do the same thing in urisourcebin.
With contributions from Philippe Normand <philn@igalia.com> (build fixes and
rebase on mono-repo).
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4713>
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>
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>
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>
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>
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>
Do not update timelevel on segment. Segment itself does not tell
anything about the amount of buffered time duration in the element
but buffer timestamp/duration is required to measure actual bufferred time.
Moreover, at the time when new segment is applied to sink/srcpad,
segment.position would point to random value.
Therefore calculating running time using the random value does not
make sense and it will result in wrong timelevel report.
This patch updates queue/queue2's timelevel measuring logic so that
it can be updated only on buffer/buffer-list/gap-event flow.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/5430>
This was causing a memory leak in cases like `gltestsrc ! gltransformation scale-x=0.5 ! glimagesink`.
Parent meta was being added in assumption that those buffers are different, which was not the case here,
creating a reference loop and never freeing the buffer.
Co-authored-by: Matthew Waters <matthew@centricular.com>
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/5445>
The caps that were sent by the caps event can be retrieved from the sinkpad
using gst_pad_get_current_caps(). This is more reliable than using cur_caps as
we know exactly which caps upstream selected when the UVC host didn't select a
format, yet.
This further allows to simplify the check, if the uvcsink has to wait for the
caps event before switching to the internal v4l2sink.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4994>
The probe passes all events except the EVENT_CAPS. Installing and removing the
probe doesn't provide any additional value.
Install an event function and always handle EVENT_CAPS. Use the caps_changed
field, to decide, if the element has to do anything special on a EVENT_CAPS.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4994>
Move the sanity checks to the beginning of the function. Make the actual effect
of the function more obvious and reset the flags in the end.
This should make it easier to understand what this function is doing.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4994>
The probe that installs the buffer probe is already on the correct pad. There is
no need for a separate function to install the probe.
While at it, change the signature of the probe functions to GstPadProbeCallback
to avoid the cast when installing the probes.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4994>
The uvcsink calculates the caps for the format that the UVC host selected. The
gst_uvc_sink_parse_cur_caps() sets these caps as cur_caps as a side effect. This
behavior is surprising as cur_caps is later updated to reflect the actually used
caps.
Just return the configured caps to avoid side effects. This makes the function
easier to understand. Update the function name to reflect the new behavior.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4994>
The only job of the event peer probe is to catch the upcoming caps event
and be able to react with the sink change. All other events that are
passing the pad shall be passed and ignored.
Since the probe is a blocking probe, there is no use in returning
with GST_PAD_PROBE_OK on other events. Otherwise the event would just
be blocked.
Since we are handling the probe removal of the probe already in the
event switch, we can remove the second explicit probe removal.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4994>
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>
Back in the mists of time[1], we switched `giostream*` elements to not close the
stream on stop() so that applications that needed a handle to the stream after
the element stopped had it.
Unfortunately, we also have cases[2] where waiting for the element to be
finalized is too late for the stream to be closed.
In order to not change the behaviour of the element, we add a property to allow
users to select the desired behaviour.
[1]: https://bugzilla.gnome.org/show_bug.cgi?id=587896
[2]: gst-plugins-rs#423
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/5372>
Formatters might call "loaded" from the `gessrc` streaming thread
meaning that the `->formatters` field need to be protected.
Several other APIs are called from gesbasedemux, in some radom
thread, so we should ensure that this is all MT. safe, and the API
makes it simple.
Co-authored-by: Philippe Normand <philn@igalia.com>
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/5431>
By using the gst_caps_set_simple() to set the format on all structures, the
compositor may create invalid combinations as the caps may contain passthrough
caps. Avoid this issue by intersecting the resul with its original.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/5386>
Adds list of formats that should be used by element in needs to passthrough
video. It contains the full list of video format plus DMA_DRM format
and will be extended in the future as needed. This patches includes 3 new
symbols:
- GST_VIDEO_FORMATS_ANY_STR
- GST_VIDEO_FORMATS_ANY
- gst_video_formats_any()
The last one can be used by bindings or for code that prefers having
GstVideoFormat values instead of strings.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/5386>
This commit ports functionality from the `rtpsrc` to make the `ristsrc`
work with dynamic payload types.
It adds two properties:
- `caps`
- `encoding-name`
These can be used to make the `ristsrc` receive other payload types than
the MPEG TS one.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/5422>
While the suspend modes NONE and PAUSED provided a low startup latency
for connecting clients they did not ensure that streams started on
fresh data.
With this property we can maintain the low startup latency of those
suspend modes while also ensuring that a stream starts on a key unit.
Furthermore, by modifying the value of a new property,
ensure-keyunit-on-start-timeout, it is possible to accept a keyunit of
a certain age but discard it if too much time has passed and instead
force a new keyunit.
Fixes#2443
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4334>
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>
Since DXVA does not support some profiles such as HEVC RExt,
vendor specific decoding API is still required.
When decoder is negotiated with d3d11 caps, decoder will convert
semi-planar frame to planar since semi-planar format (e.g.,
DXGI_FORMAT_NV12) is not supported by CUDA/D3D11 interop.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/5409>
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>
Use gst_codec_utils_caps_get_mime_codec() in pbutils for codec
strings. That function gives more elaborate RFC 6381 compatible
strings than the helper functions in gstmdphelper.c, such as
"avc1.F4000D".
Remove the helper functions, as they were only used from dashsink.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/5404>
When the property "start-time-selection" is set to "first", it
calculates the start time of the output from the buffer pts
(converting it to running time of the segment), but if the
rate is negative, the real start is not the pts, but the
pts + duration, because it plays from the end of the buffer
to it's start.
As a result of this bug, in the negative rate, when the
start-time-selection=first, the first frame is dropped
by the videoaggregator (reproduced on d3d11compositor).
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/5276>
The interaudiosrc might take buffers of different sizes from the audio adapter,
so keeping metas consistency would be an issue. So the sink now strips the audio
metas away and the src adds them back (for non-interleaved layouts only) when
taking buffers from the adapter.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/5324>
Move the GstStructure field into public struct for direct access, that's
easier than having to call a function to get it. It is not an API/ABI
breakage to extend the public structure of a GstMeta because they are
always allocated by inside GStreamer. The structure is exposed already
by gst_custom_meta_get_structure() which does not return a copy/ref, so
it is locked into holding a GstStructure forever anyway.
Also add gst_meta_register_custom_simple() because most of the time only
a name is required, tags and transform functions are more niche
use-case.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/5385>
If there are multiple Wayland event listeners in different threads we
get the formats and modifiers pushed concurrently which leads to
segfault from GArray methods. This patch protects the array.
The problem occurs e.g. when using vaapipostproc together with Qt
qmlglsink, QtWayland will get the events as well as VAAPI.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/5280>
Moves outputting frames to a task on the source pad, bringing vtdec in line with vtenc.
This brings possible performance improvements thanks to decoupling queueing new frames from outputting processed ones.
The queue length is limited to `2*DBP` to prevent decoding too far ahead compared to what we're pushing downstream.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/5163>
Right now we split the RTP header from the current buffer into a new
buffer and aggregate those buffers for later processing if the
depayloader creates an output buffer.
This is cumbersome as it happens even if none of the incoming RTP
buffers carries RTP header extensions at all just because header
aggregation has been enabled in the depayloader class.
This commit will start aggregation only in case that there really are
RTP header extensions available on an incoming RTP buffer. The check
is trivial and cheap. Once activated we keep aggregation active for
all buffers. The active state is reset on state change READY_TO_PAUSE.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/5278>
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>
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>
This was easy to trigger when testing with e.g. vtenc ! vtdec ! glimagesink and closing the sink via window button,
causing GST_FLOW_ERROR to be received by the output loop, stopping it with the queue still full. This made the
enqueue_buffer() callback to lock waiting for space in our queue, while handle_frame() was waiting for the internal
VideoToolbox queue to free up, so that VTCompressionSessionEncodeFrame could finish. As the output loop was not
running, both functions waited forever.
Fixed by 1) immediately emptying our queue when GST_FLOW_ERROR is received (like we already did with _FLUSHING)
and 2) unconditionally setting the flushing flag in finish_encoding() when it sees the output loop stopped because
of GST_FLOW_ERROR, so that enqueue_buffer() will immediately discard any new frames coming out of VideoToolbox.
Both of those make sure we never run into the both-queues-full scenario.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/5303>
As a short-term solution before full d3d12 rendering feature,
copy decoded d3d12 texture to shared d3d11 texture in order to use
existing various d3d11 implementations such as conversion, resizing,
and videosink.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/5356>
There is no need to use DRM dumb pool if buffer to
render is already a DMABuf, just import it and render it.
This fixes a DMAbuf memory leakage when waylandsink downstream
element exports DMABuf while waylandsink is configured to be
DMABuf exporter (drm-device=/drv/dri/card0):
gst-launch-1.0 v4l2src io-mode=4 ! gtkwaylandsink drm-device=/dev/dri/card0
leakage identfied with command:
watch "cat /sys/kernel/debug/dma_buf/bufinfo | grep attached "
Fixes#2729
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/5350>
There is no need to use DRM dumb pool if buffer to
render is already a DMABuf, just import it and render it.
This fixes a DMAbuf memory leakage when waylandsink downstream
element exports DMABuf while waylandsink is configured to be
DMABuf exporter (drm-device=/drv/dri/card0):
gst-launch-1.0 v4l2src io-mode=4 ! waylandsink drm-device=/dev/dri/card0
leakage identfied with command:
watch "cat /sys/kernel/debug/dma_buf/bufinfo | grep attached "
Fixes#2729
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/5350>
Don't update info's size with the VA image reported data size for single plane
images, since drivers might allocate bigger space than the strictly required to
store the image, but when we dump the buffer as is (using filesink, for example)
the produced stream is corrupted. For multi-plane images video meta is required
to read/write them.
We updated info's size because gstreamer-vaapi did it too, but the reason to
update it there was for uploading and rendering surfaces (commit c698a015).
Furthermore, this patch adds an error message if the allocated data size for the
image by the driver is lesser than the expected because it would be a buggy
driver.
Fixes: #2959
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/5308>
Even if decoder is negotiated with CUDA memory feature, if downstream
proposed no buffer pool, assume that the pool size is unknown.
And disable zero-copy if there's no more free output surface.
Or, in case of reverse playback, always copy frames.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/5338>
Even if the segmentation feature value is not updated,
the parsed "segmentation_update_map" and "segmentation_temporal_update"
values should not be cleared as it's referenced during lower
level bitstream parsing. Also, don't use assert() in parser
unless it's clearly impossible condition.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/5334>
If DPB is full already, GstH265Decoder::new_picture() might fail if
subclass uses fixed size picture pool and its size is equal to the DPB
size. Call the new_picture() after DPB is cleared in gst_h265_decoder_dpb_init()
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/5333>
Issue is that when amc was producing a codec-data buffer, a
GstVideoCodecFrame was being popped off the internal queue. This meant
that the codec-data was being associated with the first input frame and
the second (first encoded buffer) output buffer with the second input
frame. At the end (assuming one input produces one output which seems
to hold in my testing and how the encoder is currently implemented)
there would be an input frame missing and would be pushed without any
timing information. This would lead to e.g. muxers rejecting the buffer
without PTS and failing to mux.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/5330>
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>
The GST_VIDEO_FORMAT_Y410, GST_VIDEO_FORMAT_Y412_LE and GST_VIDEO_FORMAT_Y412_BE
formats in fact are packed formats, which have just 1 plane. But we have special
setting for them rather than using get_single_planar_format_gl_swizzle_order().
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/5314>
As we don't have any mapping from YUV formats + modifiers to an equivalent
emulated format (e.g. NV12 + modifier -> R8+modifier/RG88+modifier), do no
allow these formats to be used with the indirect DMABuf uploader.
Fixes#2942
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/5270>
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>
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>
This is consistent with the librtmp-based old rtmp plugin and ffmpeg.
While some servers require a valid flash-version, others are failing
with a too long or any flash-version at all.
By changing to the same default as in the old plugin and in ffmpeg,
GStreamer will at least behave the same and will work and fail with the
same servers without setting a flash-version.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/5293>
The same is done in the set_property function. This was noticed when attempting
to dump a pipeline containing glsinkbin sink=gtk4paintablesink to dot format.
Critical warnings were raised due to the missing force-aspect-ratio property on
that sink.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/5311>
It is similar to NV12 but has 10bits per channel instead of 8.
As it is supported by many modern GPUs, VA-API and an increasing
number of Wayland compositors, let's support it as well.
Also bump the required libdrm version accordingly and add a temporary
define for the WL_SHM format.
Tested with Weston, Mutter and Sway.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/5275>
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>
Adding cudaipc{src,sink} element for CUDA IPC support.
Implementation note:
* For the communication between end points, Win32 named-pipe
and unix domain socket will be used on Windows and Linux respectively.
* cudaipcsink behaves as a server, and all GPU resources will be owned by
the server process and exported for other processes, then cudaipcsrc
(client) will import each exported handle.
* User can select IPC mode via "ipc-mode" property of cudaipcsink.
There are two IPC mode, one is "legacy" which uses legacy CUDA IPC
method and the other is "mmap" which uses CUDA virtual memory API
with OS's resource handle sharing method such as DuplicateHandle()
on Windows. The "mmap" mode might be better than "legacy" in terms
of stability since it relies on OS's resource management but
it would consume more GPU memory than "legacy" mode.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4510>
If glyphrun unit is changed in a single line, there could be
overlapped background area which result in drawing background
twice. Adding geometry combine so that background geometry objects
with the same color can be merged and rendered at once
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/5179>
Fixes https://gitlab.freedesktop.org/gstreamer/gstreamer/-/issues/2900
The `reports` list was being copied as a reference, therefore, copies of
a test ended up inadvertedly sharing the same list of reports. Reports
added by one instance of the test would be reflected in all instances.
This caused a race condition where, if a test was run on repeat with
gst-validate-launcher -f, very often wrong log file was shown to the
user. For instance, gst-validate-launcher would say "test failed, see
log for iteration7", but iteration7 would contain "TEST PASSED".
Worse, the runner would add the report to that incorrect log file,
mixing problems between different executions of the tests.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/5177>
Latest MSYS2 MinGW provides these now, so we don't need to define them
if they're already present in the header.
The AudioClient3 GUID requires the Windows 10 SDK, so it's only
available in the latest MinGW, and the MinGW in Cerbero is too old.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/5155>
VA decoders implementation has been verified from 1.18 through 1.22
development cycles and also via the Fluster test framework. Similar
to other cases, we can prefer hardware over software in most cases.
At the same time, GStreamer-VAAPI decoders are demoted to NONE to
avoid collisions. The first step to their deprecation.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2312>
These 10bit formats are identical to NV12_16L32S, but 64bytes of data is being
prefixed with 16bytes data with four pixels of lower 2bits per byte. For
MT2110T, the lower two bits set so each bytes contains a column of 4 pixels,
also describe as tiled lower 2 bits. MT2110T has been chosen as a name to match
the vendor chosen name. This format is unlikely to exist for other vendors.
For MT2110R, the 2 low bits are in raster order.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3444>
When advancing the ringbuffer, store the processed CoreAudio sample
time, then interpolate the clock in the _get_delay() calls to smooth
the clock. CoreAudio's "latency" report is always a constant and
otherwise leads to the clock generating a latency-time staircase.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/5140>
Set the BufferFrame size in CoreAudio so it will deliver data
in ringbuffer segment units when recording. Otherwise, CoreAudio
will provide data in whatever granularity it wants, with no
relationship to the requested latency-time.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/5140>
The current limit is `x10`, which allows just `+20 dB` of gain.
While it may seem sufficient, this came up as a problem
in a real-world, non-specially-engineered situation,
in strawberry's EBU R 128 loudness normalization.
(https://github.com/strawberrymusicplayer/strawberry/pull/1216)
There is an audio track (that was not intentionally engineered that way),
that has integrated loudness of `-38 LUFS`,
and if we want to normalize it's loudness to e.g. `-16 LUFS`,
which is a very reasonable thing to do,
we need to apply gain of `+22 dB`,
which is larger than `+20 dB`, and we fail...
I think it should allow at least `+96 dB` of gain,
and therefore should be at `10^(96/20) ~= 63096`.
But, i don't see why we need to put any specific restriction
on that parameter in the first place, other than the fact
that the fixed-point multiplication scheme does not support volume
larger than 15x-ish.
So let's just implement a floating-point fall-back path
that does not involve fixed-point multiplication
and lift the restriction altogether?
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/5063>
Issue was that Qt was caching multiple different shadersbased on a static
variable location. This static variable location was the same no matter
the input video format and so if an item had an input video format of
RGB and another of RGBA, they would eventually end up using the same
GL shader leading to incorrect colours.
Fix by providing different static variable locations for each of the
different shaders that will be produced.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/5145>
There is currently no way for applications to know if the stream has
been properly terminated by the server or if the network connection
was disconnected as EOS is sent in both cases.
Adding a property so connection errors can be reported as errors
allowing applications to distinguish between both scenarios.
Fix#2828
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/5115>
When this flag is enabled, the transform_caps() simply set passthrough
to generate the raw caps. This is not correct, because the sink and
src have different format/drm-format fields.
We already add system memory conversion for DMABuf manner, so no more
need for this flag.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3524>
The current _nvmm_upload_transform_caps() only simply apply
"memory:NVMM" to all input caps to transform the output caps.
This is not precise and may cause problem. For example, if the
input caps include:
video/x-raw(memory:DMABuf), width=(int)1920, height=(int)1080, \
interlace-mode=(string)progressive, multiview-mode=(string)mono, \
framerate=(fraction)30/1, drm-format=(string)NV12:0x0100000000000002
it will be changed as
video/x-raw(memory:GLMemory), width=(int)1920, height=(int)1080, \
interlace-mode=(string)progressive, multiview-mode=(string)mono, \
framerate=(fraction)30/1, drm-format=(string)NV12:0x0100000000000002
For GLMemory kind caps, no drm-format should appear.
So we should let it only transforms which it can recognize.
And in _nvmm_upload_accept(), we should only accept the "memory:NVMM"
feature in input caps.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3524>
The current _directviv_upload_transform_caps() only simply apply
"memory:GLMemory" to all input caps to transform the output caps.
This is not precise and may cause problem. For example, if the
input caps include:
video/x-raw(memory:DMABuf), width=(int)1920, height=(int)1080, \
interlace-mode=(string)progressive, multiview-mode=(string)mono, \
framerate=(fraction)30/1, drm-format=(string)NV12:0x0100000000000002
it will be changed as
video/x-raw(memory:GLMemory), width=(int)1920, height=(int)1080, \
interlace-mode=(string)progressive, multiview-mode=(string)mono, \
framerate=(fraction)30/1, drm-format=(string)NV12:0x0100000000000002
For GLMemory kind caps, no drm-format should appear.
So we should let it only transforms which it can recognize.
And in _directviv_upload_accept(), we should only accept the system
memory as input caps.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3524>
The current _gl_memory_upload_transform_caps() only simply apply
"memory:GLMemory" to all input caps to transform the output caps.
This is not precise and may cause problem. For example, if the
input caps include:
video/x-raw(memory:DMABuf), width=(int)1920, height=(int)1080, \
interlace-mode=(string)progressive, multiview-mode=(string)mono, \
framerate=(fraction)30/1, drm-format=(string)NV12:0x0100000000000002
it will be changed as
video/x-raw(memory:GLMemory), width=(int)1920, height=(int)1080, \
interlace-mode=(string)progressive, multiview-mode=(string)mono, \
framerate=(fraction)30/1, drm-format=(string)NV12:0x0100000000000002
For GLMemory kind caps, no drm-format should appear.
So we should let it only transforms which it can recognize.
And in _gl_memory_upload_propose_allocation(), we should only allocate
the allocator and buffer pool for the caps with "memory:GLMemory"
feature.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3524>
The current _upload_meta_upload_transform_caps() only simply apply
"memory:GLMemory" to all input caps to transform the output caps.
This is not precise and may cause problem. For example, if the
input caps include:
video/x-raw(memory:DMABuf), width=(int)1920, height=(int)1080, \
interlace-mode=(string)progressive, multiview-mode=(string)mono, \
framerate=(fraction)30/1, drm-format=(string)NV12:0x0100000000000002
it will be changed as
video/x-raw(memory:GLMemory), width=(int)1920, height=(int)1080, \
interlace-mode=(string)progressive, multiview-mode=(string)mono, \
framerate=(fraction)30/1, drm-format=(string)NV12:0x0100000000000002
For GLMemory kind caps, no drm-format should appear.
So we should let it only transforms which it can recognize.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3524>
The current _raw_data_upload_transform_caps() only simply apply
"memory:GLMemory" to all input caps to transform the output caps.
This is not precise and may cause problem. For example, if the
input caps include:
video/x-raw(memory:DMABuf), width=(int)1920, height=(int)1080, \
interlace-mode=(string)progressive, multiview-mode=(string)mono, \
framerate=(fraction)30/1, drm-format=(string)NV12:0x0100000000000002
it will be changed as
video/x-raw(memory:GLMemory), width=(int)1920, height=(int)1080, \
interlace-mode=(string)progressive, multiview-mode=(string)mono, \
framerate=(fraction)30/1, drm-format=(string)NV12:0x0100000000000002
For GLMemory kind caps, no drm-format should appear.
So we should let it only transforms which it can recognize.
We also should recognize the system memory caps in _accept() early, if
the input is not system memory, we just return early.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3524>
Most of the time, the RGB kind formats are OpenGL native supported
format which has only one plane. They can be imported at one shot
using no matter DIRECT or INDIRECT mode.
While YUV kind formats which have multi planes have two ways to import.
They can be DIRECT imported, which requires GL_OES_EGL_image_external
extension. The output format should be RGBA and TARGET should be set
as OES after imported. The other way, they can be INDIRECT imported,
which makes each plane as a texture. In this mode, the imported textures
have different fourcc from the original format. For example, the NV12
format can be imported as a R8 texture for the first plane and RG88
texture for the second plane. The output TARGET should be sets as 2D
in this mode.
When converting sink caps to src caps, we first filter the feature of
"video/x-raw(memory:DMABuf)" and system memory. Then Based on the
external_only flag (INDIRECT mode does not care while DIRECT mode cares),
we transform the drm-format into the gst video format.
When converting src caps into sink caps, we first filter the correct
TARGET(INDIRECT mode contains 2D only while DIRECT mode contains 2D,
OES or both of them) gstructure. Then Based on the include_external flag
(INDIRECT mode always true while DIRECT mode depends on TARGET), we
transform the gst video format into drm-format.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3524>
When switching from a raw stream to an encoded stream we need to make sure the
slot is unlinked, there is code in place for this but it wasn't triggered
because the slot being reconfigured wasn't advertised as linked beforehand.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/5126>
Pass GstVideoInfoDmaDrm or GstVideoInfo whenever possible, avoiding passing
strange combination of GstVieoFormat + modifier. Even though we don't have any
at the moment, this also allow supporting GstVideoFormat that are not supported
in our DRM integration.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/5120>
When we fill a bitstream buffer the buffer might be too small to hold
the entire frame. Only resize to the filled size, preventing the
following assertion to happen.
gst_buffer_resize_range: assertion 'bufmax >= bufoffs + offset + size' failed
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/5100>
Shader compilation was failing on macOS:
gstglslstage.c:519:_compile_shader:<glslstage1> fragment shader compilation failed:
ERROR: 0:10: 'input_swizzle' : syntax error: Array size must appear after variable name
Co-authored-by: Matthew Waters <matthew@centricular.com>
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/5123>
According to libva API description, cu_qp_delta in VAConfigAttribValEncHEVCFeatures
is supposed to be used as a flag not the value of depth. And if flag enabled,
diff_cu_qp_delta_depth should be decided by log2_diff_max_min_luma_coding_block_size.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/5068>
Rework the va_map_unlocked() after we keep mapping behavior (whether to
use derive) consistent with allocator_try stage. Also remove the flag
for iHD case because pitch/stride difference between vaCreateImage and
vaDeriveImage only possibly happen on iHD by now.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/5046>
In gst_va_allocator_try, the first try is to use derive_image, if it
succeeds, we should use info from derived image to create bufferpool.
If derive fails, then try create_image and give created image info
to the pool.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/5046>
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/5021>
We were checking if the tag list is writable, but it may actually be
shared through the same event (tee upstream or multiple consumers).
Fix a bug where multiple branches have a videoflip element checking the
taglist. The first one was changing the orientation back to rotate-0
which was resetting the other instances.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/5097>
vtenc has an async output queue, which we only iterate over after another frame is enqueued.
At the very least it means we're always a frame behind the fastest possible output.
In edge cases it's also bug-prone - for example if we only have 1 frame, the downstream caps negotiation
will never happen.
This commit adds a separate task running on the source pad, which only iterates over the output queue
and pushes frames out as soon as they're put there. The queue length is limited to ensure we don't encode
too far ahead compared to what downstream can consume. Any failures that occur when pushing data downstream
will be signalled in self->downstream_ret so that other parts of code can act accordingly.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4967>
Don't assume that compositor will output only single buffer
for single input buffer. If buffer's running time is not completly
aligned to output buffer running time or duration, compositor
can generate multiple buffers. If that happens, two threads,
one is aggregator output thread and main thread were trying
to modify buffer in this test. Clear the buffer after
shutting down pipeline to avoid the race.
Fixes: https://gitlab.freedesktop.org/gstreamer/gstreamer/-/issues/2836
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/5081>
Newer macOS provides /usr/lib/pkgconfig/libpcre2-8.pc which is broken
because it says headers are in /usr/include but that directory doesn't
exist. It can only be used to find the library, which only exists on
newer macOS at /usr/lib/libpcre2-8.dylib, so it's also unusable.
So, force usage of the subproject for glib.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/5060>
Do not attempt to send a streams-selected message when reassigning
an output slot in case upstream signalled that it is handling stream selection.
In this case decodebin3 doesn't keep track of stream
collections (`dbin->collection` is NULL).
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/5059>
Sending an EOS event is actually really bad because rtpbin doesn't
handle that very well. It was only being used as a way to notify
webrtcbin to check if re-negotiation is needed.
We don't need that anymore, since changing the direction is enough to
notify webrtcbin to check for re-negotiation.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/5045>
When a video track is muted, hide the video element to differentiate
it from a track that is stuck because we stopped receiving RTP data.
Show it again when it is unmuted.
When a video track is removed, remove the video element. It will be
re-added on renegotiation.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/5045>
The current way of dma caps uses the drm-format to replace the orginal
format field. The absence of format field means it can accept all formats.
It causes problems when clipping with other old DMA or video/x-raw(ANY)
caps, the result will contain both format field and drm-format field,
which is not valid DMA caps.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4981>
This GST_VIDEO_FORMAT_DMA_DRM is introduced for DMABuf kind feature
usage. It represent the DMA DRM kind memory. And like the ENCODED
format, it should not be interpreted and mapped as normal video format.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4981>
Setting the input field on the empty slot prevents future linking of it and will
result in flow errors later on.
This was observed in WebKit's MediaStream source element, when it changes the
caps on one of its associated streams, from an encoded format to a raw video
format. The associated stream-id on the sticky stream-start event doesn´t
change, but the element creates a new GstStream with a different ID and sets it
on the stream-start event. Stream parsing is disabled in urisourcebin, so
decodebin3 handles the parsing. Without this patch we would end-up with unlinked
pads in decodebin3 after switching to the raw video format.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/5048>
The drop-frame rules are specified in “SMPTE ST 12-3:2016” and are
consistent with the traditional ones:
“
To minimize fractional time deviation from real time, the first two
super-frame numbers (00 and 01) shall be omitted from the count at the
start of each minute except minutes 00, 10, 20, 30, 40, and 50. Thus the
first eight frame numbers (0 through 7) are omitted from the count at
the start of each minute except minutes 00, 10, 20, 30, 40, and 50.
”
Where “super-frame” is a group of 4 frames for 120 FPS.
Fixes#2797
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/5028>
The current implementation copies metas without checking if the buffer
is writable.
The operation that needs to be done, replacing the input buffer and
copying the metas, is only part of that process. We create a new function
that does both.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4912>
This check fixes a critical warning that can happen when a pointer motion
happens and the video doesn't have its width/height information available.
GStreamer-Video-CRITICAL **: gst_video_center_rect: assertion 'src->h != 0' failed
#0 g_logv (log_domain=0x7ffff705e176 "GStreamer-Video", log_level=G_LOG_LEVEL_CRITICAL, format=<optimized out>, args=<optimized out>) at ../../../../Projects/jhbuild/glib/glib/gmessages.c:1422
#1 0x00007ffff7e1a81d in g_log (log_domain=<optimized out>, log_level=log_level@entry=G_LOG_LEVEL_CRITICAL, format=format@entry=0x7ffff7e77a9d "%s: assertion '%s' failed") at ../../../../Projects/jhbuild/glib/glib/gmessages.c:1460
#2 0x00007ffff7e1b749 in g_return_if_fail_warning (log_domain=<optimized out>, pretty_function=<optimized out>, expression=<optimized out>) at ../../../../Projects/jhbuild/glib/glib/gmessages.c:2930
#3 0x00007ffff701d90b in gst_video_sink_center_rect (src=..., dst=..., result=result@entry=0x7fffffffc6d0, scaling=scaling@entry=1) at ../../../../Projects/jhbuild/gstreamer/subprojects/gst-plugins-base/gst-libs/gst/video/gstvideosink.c:105
#4 0x00007fffe5652dbb in _fit_stream_to_allocated_size (result=0x7fffffffc6d0, allocation=0x7fffffffc6c0, base_widget=0x9396f0) at ../../../../Projects/jhbuild/gstreamer/subprojects/gst-plugins-good/ext/gtk/gtkgstbasewidget.c:326
#5 gtk_gst_base_widget_display_size_to_stream_size (base_widget=base_widget@entry=0x9396f0, x=1207.7109375, y=811.84765625, stream_x=stream_x@entry=0x7fffffffc720, stream_y=stream_y@entry=0x7fffffffc728) at ../../../../Projects/jhbuild/gstreamer/subprojects/gst-plugins-good/ext/gtk/gtkgstbasewidget.c:344
#6 0x00007fffe5651a4b in gst_gtk_base_sink_navigation_send_event (navigation=0x5ff990, event=0x178a730) at ../../../../Projects/jhbuild/gstreamer/subprojects/gst-plugins-good/ext/gtk/gstgtkbasesink.c:340
#7 0x00007fffe5652432 in gtk_gst_base_widget_motion_event (widget=<optimized out>, event=event@entry=0x1f14b60) at ../../../../Projects/jhbuild/gstreamer/subprojects/gst-plugins-good/ext/gtk/gtkgstbasewidget.c:404
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/5049>
The libpsl subproject wasn't building successfully and CI didn't
notice because:
1. The plugin wasn't explicitly enabled
2. Even when the plugin is explicitly enabled, the dep is not required
at build time when not building a static plugin
So fix all of these issues.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/5038>
The videoencoder base class uses getcaps() to ask a subclass for the caps in its
sink_query_default() implementation.
Replace the custom handling of the QUERY_CAPS in the v4l2videoenc with an
implementation of getcaps() that returns the caps that are supported by the
v4l2videoenc to return these caps in the query.
This getcaps() implementation also calls the provided proxy_getcaps(), which
sends a caps query to downstream. This fixes the v4l2videoenc element to respect
limits of downstream elements in a sink query.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/5034>
Given the amount of complains about artifacts when negotiating dmabuf
given incompatible drm-formats, and that there's no enough bandwidth
for a proper and quick fix in gstreamer-vaapi, this patch disables,
from decoders and postprocessor, the DMABuf caps feature.
For those who needs DMABuf can use the va elements in -bad, increasing
their ranking for autoplugging by using the environment variable
GST_PLUGIN_FEATURE_RANK=vah264dec:MAX, for example.
This can be considered a first step to the deprecation of
gstreamer-vaapi in favor of the va plugin in -bad.
Fixes: #1137
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/5010>
The videoencoder base class always uses the negotiated allocator for allocating
coded buffers and ignores the negotiated buffer pool. Therefore, the
v4l2videoenc always has to copy buffers from the pool into the allocated
output buffers.
This breaks downstream elements that want to import the CAPTURE buffers of the
v4l2videoenc, since the v4l2videoenc copies the exported CAPTURE buffers and
sends the copies downstream.
Always use the CAPTURE buffer pool for acquiring CAPTURE buffers instead of
allocating the buffers in the base class.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4230>
It's possible and normal to tear down a harness while the pipeline is
running. At the same time, it's desired for the
`gst_harness_pad_link_tear_down()` function to be synchronous.
This has created the conflict where the main thread may request a
harness to be torn down while it's in use or about to be used by a pad
in the streaming thread.
The previous implementation of `gst_harness_pad_link_tear_down()` tried
to handle this by taking the stream lock of the harnessed pad and
resetting all the pad functions while holding it. That approach was
however insufficient to handle the case where a non-serialized event
or query is being handled or about to be handled in a different thread.
This edge case was one race condition behind the flakes in the flvmux
check tests -- the rest being covered by https://gitlab.freedesktop.org/gstreamer/gstreamer/-/issues/2803.
This patch fixes the problem by adding an intermediate ref-counted
object, GstHarnessLink, which replaces the usage of the HARNESS_KEY
association. GstHarnessLink allows the pad functions such as event,
query and chain to borrow a reference to GstHarness and more
importantly, to lock the GstHarnessLink during their usage to block
(delay) its destruction until no users are left, and guarantee that any
future user will not receive an invalid GstHarness handle past its
destruction.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/5017>
This reverts commit 893e4ed0dd.
This caused regressions in existing elements which override/set things
like QoS and such in their own init functions. If the base class does
this in ::constructed() now it will override the subclass settings
again with its own, which can have unintended side-effects.
Case in point is gdkpixbufsink which disabled QoS there, and this
patch would reliably make the unit test fail in valgrind because
now frames are dropped because of QoS (when QoS should really be
disabled).
Fixes#2794
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/5006>
If the capture pool is already active, like when handling gaps at the
start of a stream, do not setup the decoder to wait for src_ch event.
Otherwise the decoder will endup waiting for that at the wrong moment
and exit the decoding thread unexpectedly.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4590>
Fix this pipeline where the tag list is not writable:
gst-launch-1.0 videotestsrc ! taginject tags="image-orientation=rotate-90" ! videoflip video-direction=auto \
! autovideosink
GStreamer-CRITICAL **: 12:34:36.310: gst_tag_list_add: assertion 'gst_tag_list_is_writable (list)' failed
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4987>
This fixes a build error if Qt was build without accessibility support:
../../../../../gstreamer/subprojects/gst-plugins-good/ext/qt/qtglrenderer.cc:
In member function 'bool GstQuickRenderer::init(GstGLContext*, GError**)':
../../../../../gstreamer/subprojects/gst-plugins-good/ext/qt/qtglrenderer.cc:341:13:
error: 'QCoreApplication' was not declared in this scope; did you mean 'QApplication'?
../../../../../gstreamer/subprojects/gst-plugins-good/ext/qt/qtglrenderer.cc:341:31:
error: 'app' was not declared in this scope
../../../../../gstreamer/subprojects/gst-plugins-good/ext/qt/qtglrenderer.cc:341:37:
error: 'QCoreApplication' is not a class, namespace, or enumeration
[...]
../../../../../gstreamer/subprojects/gst-plugins-good/ext/qt/qtglrenderer.cc:458:5:
error: 'QEventLoop' was not declared in this scope; did you mean 'QEvent'?
../../../../../gstreamer/subprojects/gst-plugins-good/ext/qt/qtglrenderer.cc:459:9:
error: 'loop' was not declared in this scope
If accessibility is enabled, the includes for QCoreApplication and QEventLoop
are indirectly pulled via QWidget.
Add the required headers as documented in [1] and [2].
[1] https://doc.qt.io/qt-5/qcoreapplication.html
[2] https://doc.qt.io/qt-5/qeventloop.html
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4815>
Previously, we would create a new GstMemory per write operation
and then append them to the GstBuffer. This would cause a reallocation
every 16 Memories which is an issue since the png encoder will usually
do write in a pattern of 4, 8 and 8k bytes repeating until the frame
is done.
Instead allocate a single GstMemory and keep writting it into it
with a manual index. Much like the jpegenc does.
Doing some basic testing With a testsrc snow pattern at 4k and 8k
the same pipeline would take ~3.30s to encode a 4k frame and ~23s
for an 8k. At 4k 0.70s/33% is taken by memory allocations, while at
8k its ~10.5s/45%.
With this patch, at 4k the pipeline takes ~2.40s and at 8k only 9.60s
making this 28% and 58% faster accordingly on my laptop, and
allocation runtime is dropped to subsecond times.
Here's the test pipeline used, increase num-buffers in image freeze
to gather more samples.
```
gst-launch-1.0 videotestsrc num-buffers=1 pattern=snow ! imagefreeze num-buffers=1 ! \
video/x-raw,width=7680,height=4320 ! pngenc ! fakesink
```
Close#2717
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4944>
lc3enc:
- encodes raw audio into lc3 format
- uses the default bitrate property and frame duration
from the caps to determine the byte count of
the encoded frames if it is not specified in
the downstream caps after negotiation
- uses the same byte count value for all the channels
- all the common session configuration parameters
are passed in the src caps
lc3dec:
- decodes an lc3 encoded audio
- sink caps should contain all the common session configuration
params
- uses frame_duration and frame_bytes (byte count) in the sink
caps as parameters along with sample rate and channel count
- byte count is same for all the channels
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4376>
`srt_rejectreason_str` doesn't give us a unique string for every
possible reason. Peers can define their own reasons and SRT just gives
us the string `"Application-defined rejection reason"` for all of them.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4948>
When gst_element_set_state is called in _setup_locked and errors, the
callback is already processed before we reach handle_current_async, and
the timer is started even though it's finished processing, which results
in a NULL pointer crash later in async_timeout_cb.
To fix this, we check that it's still processing before calling
handle_current_async.
Fixes#1683
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4936>
The aim of this example is to show how to make use of the accept-certificate
signal from a GTK GUI, and prompt user in case of invalid certificate.
There are two subtleties to be aware of:
1. the signal is emitted from the GStreamer streaming thread, therefore the
caller can't modify the GUI straight away, instead they must do it from the
main thread (eg. by using g_idle_add())
2. in case of a redirection, then a TLS failure, the caller won't know
about the redirection. Actually, it's possible to be notified of the
redirection by watching "message:element" and inspecting http-headers,
but even in that case, the signal will be received *after* the signal
"accept-certificate" (even though the redirection happened *before*).
This second point is tricky. It's not uncommon to have servers that redirect
http requests to https. So errors of the type "HTTP -> HTTPS -> TLS error"
happen, and if the caller doesn't care about redirection, they might prompt
users with a message such as "TLS error for URL http://...", which wouldn't make
much sense.
This example shows how to handle that right, by connecting to the signal
"message:element", inspecting the http-headers, and in case of redirection,
updating the TLS error dialog to indicate that the request was redirected.
Here are a few examples of streams that exhibit TLS failure (at the time of
this commit, of course):
* https://radiolive.sanjavier.es:8443/stream: unknown-ca
* https://am981.ddns.net:9005/stream.ogg: unknown-ca
* http://stream.diazol.hu:7092/zene.mp3: redir then bad-identity
* https://streaming.fabrik.fm/izwi/echocast/audio/index.m3u8: unknown-ca
(this one is a HLS stream)
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4925>
With libsoup 2.x, it was possible to know when there was a TLS failure, as
libsoup provided the "special http status code" SOUP_STATUS_SSL_FAILED.
However these special codes were dropped with libsoup 3.x: now libsoup emits
the accept-certificate signal when there's a TLS failure.
This commit adds a signal "accept-certificate" to SoupHttpSrc, which is in fact
just about forwarding the signal from SoupMessage (which is, itself, forwarded
from GTlsConnection). Note that, in case of libsoup 2.x, the signal is never
emitted.
Fixes: #2379
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4925>
Adding new subtitle overlay element. It's a bin which is wrapping
two internal elements dwritesubtitlemux and dwritetextoverlay.
* dwritesubtitlemux: A new internal element to aggregate subtitle
buffers and to attach the aggregated subtitle buffers on
video buffer as meta.
* dwritetextoverlay: Extracts/renders the subtitle meta and
discard the meta after rendering.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4934>
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/4920>
There's no reason to release GstMemory manually at all.
If we do release GstMemory, corresponding GstBuffer will be
discarded by GstBufferPool baseclass because the size is changed
to zero.
Actual cause of heavy CPU usage in case of fixed-size pool
(i.e., decoder output buffer pool) and if we remove GstMemory from
GstBuffer is that GstBufferPool baseclass is doing busy wait in acquire_buffer()
for some reason. That needs to be investigated though, discarding
and re-alloc every GstBuffer is not ideal already.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4935>
Fixes test: validate.uridecodebin.expose_raw_pad_caps
testsrcbin (currently part of debugutilsbad) is an useful element for
validate tests.
validate.uridecodebin.expose_raw_pad_caps makes use of it.
Unfortunately, because validate tests with GStreamer only run with
whitelisted plugins and `debugutilsbad` wasn't in the whitelist, the
test was failing and being auto-skipped.
This patch adds debugutilsbad to the whitelists used by validate tests
in subprojects with a validate/meson.build.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4931>
The current way of using gst_video_info_set_format() will change all
fields of the GstVideoInfo. We only need to change its format, stride
and offset fields.
In order to keep the consistency with th common drm API, we rename the
gst_va_video_info_from_dma_info() into gst_va_dma_drm_info_to_video_info().
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4883>
The current way only selects the best video format from the first
structure of the caps. The caps like:
video/x-raw(memory:VAMemory),drm-format=(string)NV12; \
video/x-raw(memory:VAMemory),format=(string){ NV12, Y210 }
Will just choose NV12 as the result, even the bitstream is 10 bits.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4928>
ges-timeline-element property getter handler was using
g_value_take_object() with internal pointers of the element as
arguments, instead of g_value_set_object().
g_value_take_object() moves the ownership of the reference; hence,
when reading "timeline" the reference ownership of timeline is moved
away from the ges-timeline-element and into the GValue.
Since GValues are temporaries that are often discarded quickly after,
this can easily lead to a double free. This was causing
gst-editing-services / pythontests to crash when running
TestTrackElements.test_ungroup_regroup() because of an innocent read of
`clip2.props.timeline` around the end of the test.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4924>
Some NLE tests were calling ges_init() from the test suite
initialization function. This was causing them to deadlock when running
with glib 2.76.3.
ges_init() causes a GThreadPool to be initialized. Even if GES at this
point doesn't request any thread to be created, since glib 2.63.4+
(see 8aeca4fa64)
the first time a GThreadPool is initialized a "pool-spawner" thread is
created, which is later used by g_thread_pool_push().
The default behavior of the GStreamer check tests is to fork for every
test case. This is not safe if any thread has been created at this
point. In this particular case, GThreadPool preserves the state that
says a "pool-spawner" thread has been created, and will have access to
its mutex and condition variable, but their queues will have different
contents as the memory has been forked. In consequence, calls to
g_thread_pool_push() will deadlock.
The deadlock will not occur if running the tests with CK_FORK=no.
This patch modifies the affected tests to only call ges_init() from
inside the test cases, fixing the deadlock.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4915>
A significant portion of the NLE test suite was often timing out due to
the tests taking way longer than necessary because the sinks were
synchronizing to the clock, which is the default behavior for
fakevideosink and fakeaudiosink.
Notable was the case of nleoperation.c:test_pyramid_operations, that ran
through a 10 second stream twice. As the default timeout is 20 seconds,
this made the test flaky.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4914>
Refusing an incoming segment in < GST_MATROSKA_READ_STATE_DATA should only be
done if the incoming segment is not in GST_FORMAT_TIME.
In GST_FORMAT_TIME, we are just storing the values and returning, so we can
invert the order of the checks.
Fixes proper segment propagation in matroska/webm DASH use-cases
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3914>
Is a seek is done on stream-collection post, there are no selected streams
yet. Therefore none would be chosen to adjust the key-unit seek.
If no streams are selected, fallback to a default stream (i.e. one which has
track(s) with GST_STREAM_FLAG_SELECT).
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3914>
When seeking is handled by the collection posting thread, there is a possibility
that some leftover data will be pushed by the stream thread.
Properly detect and reject those early segments (and buffers) by comparing it to
the main segment seqnum
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3914>
As VK_FORMAT_FEATURE_2_xxx are defined as static const variable, the
vscoce C compiler prevents the initialization of the vk_usage_map
structure with error "C2099: initializer is not a constant".
Init the structure separately.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4904>
The current way of using parent's copy_metadata() virtual function will
selectively filter out some meta such as crop meta. That virtual function
should be used when copying input buffer's meta data into output buffer,
not suitable when importing the input buffer.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4887>
When the input buffer has crop meta, and we need to do copy, we
should consider the uncropped video size and copy the full size
of video memory.
The video meta in this case should contain the full uncropped
resolution info. We can use it to create full size va buffers.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4887>
Adding Direct3D11 backend Qt6 QML videosink element, qml6d3d11sink.
Implementation details are similar to the qt6 plugin in -good
but there are a few notable differences.
* qml6d3d11sink accepts all GstD3D11 supported video formats (e.g., NV12).
* Scene graph (owned by qml6d3d11sink) will hold dedicated and sharable
RGBA texture which belongs to Qt6's Direct3D11 device, instead of sharing
GStreamer's own texture with Qt6.
* All rendering operations will be done by using GStreamer's Direct3D11 device.
Specifically, upstream texture will be copied (in case of RGBA)
or converted to the above mentioned Qt6's sharable texture.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3707>
... otherwise streams with constant size samples defined with a single
`sample_size` for all samples in the `stsz` box fall in the category
`chunks_are_samples` in `qtdemux_stbl_init`, overriding the actual
sample count.
`FOURCC_soun` would set this automatically for `compression_id == 0xfffe`,
however `compression_id` is read from the Audio Sample Entry box at an offset
marked as "pre-defined" in some version of the spec and set to 0 both by
GStreamer and FFmpeg for opus streams.
Considering the stream `sampled` flag is set explicitely by other fourcc
variants, doing so for opus seems consistent.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4903>
This will cause an integer overflow a little bit further down because we
allocate a bit more memory to allow for a NUL-terminator.
The caller should've avoided passing that much data in already as it's
not going to be a valid image and there's likely not even that much data
available.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4894>
This is a small optimization and avoids restarting the next parsing
iteration on already accepted data.
On its own it would also fix ZDI-CAN-20968 (see previous commit) but the
previous commit independently is also a valid fix for it.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4895>
The "Encapsulation of Opus in ISO Base Media File Format" [1] specifications,
§ 4.3.2 Opus Specific Box, indicates that data must be stored as big-endian.
In `build_opus_extension`, `gst_byte_writer_put*_le ()` variants were used,
causing audio streams conversion to Opus in mp4 to offset samples due to the
PreSkip field incorrect value (29ms early in our test cases).
[1] https://opus-codec.org/docs/opus_in_isobmff.html#4.3.2
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4875>
Now all codec baseclasses can inform subclasses of correct max DPB size,
and exception handling (e.g., emergency bumping in h.264) has been
improved as well. Smaller number of additional DPB frame allocation
seems to be safe.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4878>
Adding GST_CUDA_CRITICAL_ERRORS env variable so that program can be
terminated on unrecoverable error.
Example)
GST_CUDA_CRITICAL_ERRORS=2,700 gst-launch-1.0 ...
In this example, CUDA_ERROR_OUT_OF_MEMORY(2) and
CUDA_ERROR_ILLEGAL_ADDRESS(700) are registered as critical error
and program will be aborted on those errors
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4729>
If two senders use the same multicast IP and port then new_session_pad()
may try to add a srcpad to the same stream twice.
stream->srcpad is updated but gst_element_add_pad() fails the second
time. As a result stream->srcpad points to a deleted object and
access in gst_sdp_demux_stream_free() fails with a segfault.
Just ignore the second pad. Nothing useful can be done with it anyway.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4603>
- Adding bayer 10,12,14,16 bits components with 16 bits storage. These
changes only adds capabilities. Capability format string is a complete
description of the frame and pixels layout. Only mapping LE bayer
formats as v4l2 only define LE bayer formats.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4852>
The `gst_video_decoder_negotiate_pool` function expects the
`decide_allocation` function to always provide a pool and will fail to
negotiate if the pool is missing. If we return immediately (even if we
don't need to do anything special) negotiation will fail if the
downstream element does not propose a pool.
Fix by chaining up to the default `decide_allocation` function which
adds a fallback pool if one was not already proposed.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4630>
Adding DirectWrite text rendering elements
* dwriteclockoverlay: Equivalent to clockoverlay
* dwritetimeoverlay: Equivalent to timeoverlay
* dwritetextoverlay: Similar to textoverlay but subtitle is not
supported
Newly added elements support system memory and d3d11 memory
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4826>
This new property allows setting of PES stream number for AAC audio
and AVC video streams.
The stream number is subject to the following constraints:
1. it must be between 0 and 15 for video
2. it must be between 0 and 31 for audio
Currently the PES stream number is hard-coded to zero for these
stream types.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4822>
Add support for 10/12/14/16 bit depths . This consists of multiple parts.
First is the parsing of caps, which pulls out the bitness and endianness
from the video/x-bayer format.
Second, gst_bayer2rgb_split_and_upsample_horiz() is split into two similar
functions, one for 8bit bayer handling and another for 16bit bayer handling.
The content is basically identical, except one uses 8bpp and the other 16bpp
inputs and outputs, and they each use different ORC code to match. The 16bpp
variant also handles endian swapping. There is now a wrapper called
gst_bayer2rgb_split_and_upsample_horiz() which selects the correct function
based on bpp from the parser.
Third, gst_bayer2rgb_process() is extended to handle both 8bit and 16bit
bayer data. Yet again there are matching ORC functions to handle the 16bit
data. This time however the 16bit handling of data is slightly special. The
ORC is not able to emit opcodes for 'x2 mergelq', so the trick here is to
store the BG and GR longs into separate 'dtmp' temporary buffer, and then
do one more ORC post-processing step, compensate for the less-than-16bpp
bitness using left shift, and reorder them into the destination frame
using 'mergelq' .
Example usage:
```
$ gst-launch-1.0 videotestsrc ! \
video/x-bayer,width=512,height=512,format=bggr16le ! \
bayer2rgb ! \
video/x-raw,format=RGBA64_LE ! \
videoconvert ! \
autovideosink
```
Example usage:
```
$ gst-launch-1.0 videotestsrc ! \
video/x-raw,width=512,height=512,format=ARGB ! \
rgb2bayer ! \
video/x-bayer,format=bggr12le ! \
bayer2rgb ! \
video/x-raw,format=RGBA64_LE ! \
videoconvert ! \
autovideosink
```
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4686>
Add comments regarding which LINE()s point to which data in the
temporary buffer and a large comment explaining how the buffer
is processed. This will hopefully be useful to someone, as the
code is not obvious. No functional change.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4686>
Instead of passing a single element of GstBayer2RGB structure into the
gst_bayer2rgb_split_and_upsample_horiz(), pass the entire pointer and
let the funciton pick out whatever it needs out of the structure. This
is a preparatory patch. No functional change.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4686>
Pass all three parameters used by the LINE() macro to the LINE() macro
and unroll the code for readability. Add more comments regarding which
of these LINE()s point to which data in the temporary buffer to make
the code less confusing.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4686>
The j variable is used as an iterator further down in this code, but
here it can be just inlined in the macro parameters to make the code
easier to read. This is done in preparation for further changes. No
functional change.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4686>
The bayer2rgb process implemented doesn't support in-place tranform.
This element doesn't implement a "transform_ip" vmethod of
GstBaseTransform it will revert to using the "tranform" vmethod.
It's misleading to set it to TRUE, here. Change this to FALSE.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4686>
Add support for conversion to 10/12/14/16 bit bayer pattern.
The implementation is rather simplistic, just take the ARGB
input, generate 16-bit data out of it instead of 8-bit, shift
them as required by the output bitness, and apply endian swap.
Example usage:
```
$ gst-launch-1.0 videotestsrc num-buffers=1 ! \
video/x-raw,width=512,height=512,format=ARGB ! \
rgb2bayer ! \
video/x-bayer,format=bggr12le ! \
filesink location=/tmp/bayer12.raw
```
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4686>
Add support for generation of 10/12/14/16 bit bayer test pattern.
The implementation is rather simplistic, just take the ARGB
input, generate 16-bit data out of it instead of 8-bit, shift
them as required by the output bitness, and apply endian swap.
Example usage:
```
$ gst-launch-1.0 videotestsrc ! \
video/x-bayer,width=512,height=512,format=bggr12le ! \
bayer2rgb ! \
video/x-raw,format=RGBA64_LE ! \
videoconvert ! \
autovideosink
```
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4686>
Even if we don't yet know what the echo probe format is, we want to be able to
provide silence for the reverse path, so that when the probe becomes available,
there is no ambiguity around what time period the new set of samples are for.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4849>
The probe's info may not precisely match the dsp's info. For instance,
the number of channels or their layout might be different.
```
GStreamer-Audio-CRITICAL **: 16:21:32.899: the GstAudioInfo argument is not equal to the GstAudioMeta's attached info
```
This broke in d5755744c3.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4849>
Race condition without this patch:
- srcpad task is being stopped in gst_aggregator_stop_srcpad_task()
- at that moment, in pre-queue event handler, gst_pad_get_task_state()
returned GST_TASK_PAUSED
- then in srcpad task got stopped in gst_aggregator_stop_srcpad_task()
- finally srcpad task got resumed in pre-queue event handler
To address it, checks "running" flag in pre-queue event handler.
Both pre-queue stream-start event handler and "running" flag
are protected by SRC_LOCK already.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4847>
