Just like for the seconds field, there are no limitations on the hours and
minutes fields. The specification for xml schema duration fields doesn't forbid
specifying durations with only (huge) minutes or hours values.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2951>
When picking an available payload type, we need to pick one that is
available across all media.
The previous code, when multiple media were present, looked at the first one,
noticed it had pt 96 as the media pt, then simply looked at the next media,
noticed it didn't, and decided 96 was available.
Instead, check if the pt is used by any of the media, if it is, decide
it is not available and go to the next pt. I'm fairly sure that was the
original intent.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2984>
When updating a manifest during live playback, preserve the current
representation for each stream.
During update_fragment_info, if the current representation changed
because it couldn't be matched, trigger a caps change and new
header download.
This reverts commit e0e1db212f
and reapplies "dashdemux: Fix issue when manifest update sets slow start
without passing necessary header & caps changes downstream" with
changes.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2920>
Timers for RTX packets are dealt with later in update_rtx_timers(), and
timers for non-RTX packets would potentially also be unscheduled a
second time from there so avoid that.
Also don't shadow the timer variable from the outer scope but instead
make use of it directly.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2973>
Check back pressure of a stream transport before popping buffer from its backlog.
If the stream transport is not experiencing back pressure, the buffer can be popped from backlog and pushed to client.
Fixes:#1298
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2936>
The address/port is pre-defined by the caller of the function, so
retrying is only going to loop forever.
Ideally the multicast address should be checked after allocating but
this doesn't happen currently, so it's better to error out cleanly then
to loop forever trying the same address.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2975>
This avoids getting in a bunch of corner cases. We'd have to insert
a "rejected" line from the start as a place-holder to get around this,
but the rest of the code just becomes more complicated, so just
disallow it for now.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2439>
If the buffer is not msdk_buffer, we can try to directly import the
attached memory (i.e. va mem and dmabuf mem) by applying the common
uitl function: import_to_msdk_function ().
Here add a flag "from_qdata" in GstMsdkSurface to handle the cropping case,
we should avoid updating the crop values when msdk_surface is from the
memory's qdata, because the crop info from this surface is the already
updated one.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2498>
When input buffer is of dmabuf memory but not a msdk buffer (i.e., the
allocator is not msdk_allocator), then we can try to get fd of this mem,
create the corresponding va surface and wrap it as mfx surface.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2498>
If read_one or write_one was called but the stream closed before it could
read/write a whole packet, read_one/write_one would hang indefinitely,
consuming 100% CPU. This commit fixes that by treating a short read/write
as an error.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2964>
We were checking possible bind flags for the DXGI format
of the source texture but that's never applied to
the destination texture desc.
Just use the already configured bind (and misc) flags of source texture
for the destination texture allocation without additional check.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2950>
Setting the content-type property shall override internally detected MIME
types, to make it possible to do as following example (where audio/basic to be
used prior to audio/x-mulaw):
gst-launch-1.0 ... ! mulawenc ! audio/x-mulaw,rate=8000,channels=1 !
curlhttpsink location=<url> content-type=audio/basic
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2732>
* Private header name is changed to gstd3d11-private.h to follow
naming convention
* Add Since mark everywhere
* Update member variable names to be consistent with the other
object implementations in this library
* Correct outdated documentation
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2945>
Theoretically having elements in locked state should not have any effect
at all when the surrounding bin is doing state changes. However
previously a state change error of a locked element would cause the
bin's state change to also fail, which is clearly not intended.
State change failures of locked elements are to be handled by whoever
set the element to locked state. By always returning them here it is
impossible for the owner of the element to handle state change failures
gracefully without potentially affecting the whole pipeline's state
changes.
Non-failure returns are still returned as-is as the distinction between
ASYNC/NO_PREROLL/SUCCESS has big consequences on the state changes of
the bin and overall pipeline. Theoretically SUCCESS should also be
returned in all cases but I can't estimate the effects this would have
on the overall pipeline.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2932>
Current default G_MAXINT is not a correct value under any circumstances.
This creates an issue with screen capture, during which we currently do
not get any framerate info causing G_MAXINT to show up, where elements
downstream can possibly misbehave - for example, `vtenc` causes
a kernel panic.
Replace with 30/1 to avoid such scenarios.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2944>
The current handle_frame() does not return the real error that happens
in decode_scan and decode_frame, which makes the pipeline continue with
the error and may trigger asserting later.
We also return the error when decode_quant_table or decode_huffman_table
fails.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2938>
Add an example to show the usage of present singal.
In this example, a text overlay with alpha blended background
will be rendered on swapchain's backbuffer by using
Direct3D11, Direct2D, and DirectWrite APIs.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2923>
The "present" signal will be emitted just before the
IDXGISwapChain::Present() call. The client can perform additional
GPU operation with given GstD3D11Device object and
ID3D11RenderTargetView handle. Or, the client can read back
the scene to be displayed on window using the signal.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2923>
libsoup 3.0.x dispatches using a single source attached when the session
is created, so we need to create the session with the same context that
our download thread is later using.
2.74 or 3.1 will dispatch a response using the context which sent the
request. However, for any context other than the one that created the
session, this will also create and destroy sources, so there's still
some slight performance benefit.
Fixes: https://gitlab.freedesktop.org/gstreamer/gstreamer/-/issues/1384
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2913>
This allows an application to provide their own opened DRM device
fd handle to kmssink. For example, an application can lease
multiple fd's from a DRM master to display on different CRTC
outputs at the same time with multiple kmssink instances.
Specifying the fd property is not allowed when driver-name
and/or bus-id properties are specified.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2807>
Handle select-streams and seek events in an element
level send_event() vfunc, so they can be received
before any source pads are created.
This allows preferred streams to be selected before
segment downloading starts.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2912>
Nouveau driver currently only exposes postproc entry. But
vaapidecodebin is registered independent if there are decoders or not,
exposing a segmentation fault.
This patch removes the encoder/decoder/codec arrays if no entries are
found, and if no decoders are found vaapidecodebin is not
registered. Also for vaapipostproc if no postproc entry is found.
Also, if general decoder, used by vaapidecodebin, doesn't have a sink
pad string, don't register the glib type.
Fixes: #1349
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2865>
Without this change cleanup function for g_autoptr is not defined for
GstPlayMediaInfo, GstPlaySignalAdapter, GstPlayVideoRenderer,
GstPlayVideoOverlayVideoRenderer and GstPlayVisualization. Cleanup
function was defined in gstplay.h, but missing in other header files.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2888>
When stopping the element, make sure the pad task
is stopped before destroying the part readers.
Closes a race where the pad task might access
a freed pointer.
Also add a guard against this sort of thing
by holding a ref to the reader in the pad loop.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2901>
When playing live, it's possible that one stream reaches
the end of the available playback window and goes to sleep
waiting for a manifest update, and the manifest update
introduces a new period. In that case, the sleeping
stream needs to wake up and go 'properly' EOS before we
can advance the input to the new period.
Accordingly, make sure that a stream's last_ret value
is not marked as EOS if it's just sleeping waiting for a live
manifest update.
Also fix the output loop to go back and re-check if it's
time to switch to the next period after dequeuing and
discarding an EOS event.
https://livesim.dashif.org/livesim/periods_20/testpic_2s/Manifest.mpd
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2895>
The parent refcount is of the *transformed* buffer, not the input
buffer.
Also update the docs to clarify that @transbuf is the transformed
buffer, and not the buffer on which a transformation is being
performed.
Due to this bug, modifying the structure of a meta that has been
copied to another buffer fails with:
gst_structure_set: assertion 'IS_MUTABLE (structure) || field == NULL' failed
Add a test for the same.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2890>
Newer compilers ( clang 15 ) have turned stricter and errors out instead
of warning on implicit function declations
Fixes
gstssaparse.c:297:12: error: call to undeclared library function 'isspace' with type 'int (int)'; ISO C99 and later do not support implicit function declarations [-Wimplicit-function-declaration]
while (isspace(*t))
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2879>
Ideally new() functions should simply call g_object_new() and not much
else, so let's do that here and handle all the construction properly in
a GObject way.
Now a play object created via g_object_new() is actually usable.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2880>
Ideally new() functions should simply call g_object_new() and not much
else, so let's do that here and handle all the construction properly in
a GObject way.
Now a player object created via g_object_new() is actually usable.
In addition, also fix the video-renderer property so that reading it
returns an object of the correct type.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2880>
That is, get rid of unnecessary and wrong special-casing.
This could always use gst_rtsp_url_get_request_uri_with_control() but as
we only have the control base URI as string it is easier to just call
gst_uri_join_strings().
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2868>
This was showing up as a memory leak in GTK's
gstreamer media backend:
40 bytes in 1 blocks are definitely lost in loss record 18,487 of 40,868
at 0x484586F: malloc (vg_replace_malloc.c:381)
by 0x50D5278: g_malloc (gmem.c:125)
by 0x50EDBA5: g_slice_alloc (gslice.c:1072)
by 0x50EFBCC: g_slice_alloc0 (gslice.c:1098)
by 0x51F2F45: g_type_create_instance (gtype.c:1911)
by 0x51DAE37: g_object_new_internal (gobject.c:2011)
by 0x51DC080: g_object_new_with_properties (gobject.c:2181)
by 0x51DCB20: g_object_new (gobject.c:1821)
by 0x9855F86: UnknownInlinedFun (gstplayer-wrapped-video-renderer.c:109)
by 0x9855F86: gst_player_new (gstplayer.c:579)
Fixes https://gitlab.freedesktop.org/gstreamer/gstreamer/-/issues/1374
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2875>
Otherwise we won't send the protection packets for the last few
packets when a stream ends.
Also send EOS on the FEC src row pad immediately, and on the FEC src
column pad after draining is complete. This makes it so that the FEC
src pads on rtpbin behave the same way as the RTCP src pads on rtpbin
when EOS is received on the send_rtp_sink pad.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2863>
When a new segment event arrives, it immediately updates
the current stored segment, which was used for calculating
the running time of the current text buffer for every
passing video frame. This means a segment that arrives
after the text buffer might get used to (mis)calculate
the running times subsequently.
Instead, calculate and store the right running time
using the current segment when storing the buffer. Later
the stored segment can get freely updated.
This fixes the case where pieces of video and text streams
are seamlessly concatenated and fed through the text overlay.
Previously, it could lead to the current text buffer suddenly
have a massive running time and blocking all further input.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2802>
