basedepayload generates its own segment in a pretty unconventional
manner, relying on information in the caps such as npt-start or
npt-stop, usually set by rtspsrc.
In ONVIF mode, rtspsrc will generate the correct segment and this
logic in rtpbasedepayload will not be needed, this commit allows
rtspsrc to signal that through the caps.
While we can convert between all formats apart from the rate, we
actually need to make sure that we comply with a) the rate of the first
configured pad and b) also all the allowed rates from downstream.
We were previously only fixating the rate in the getcaps
implementation when downstream was requiring a discrete value,
causing negotiation to fail when upstream was capable of rate
conversion, but not made aware that it had to occur.
Instead of fixating the rate, we can simply update our sink
template caps with whatever GValue the downstream caps are holding
as their rate field.
Allows negotiation to successfully complete with pipelines such as:
audiotestsrc ! audio/x-raw, rate=48000 ! audioresample ! audiomixer name=m ! \
audio/x-raw, rate={800, 1000} ! autoaudiosink \
audiotestsrc ! audio/x-raw, rate=44100 ! audioresample ! m.
... and also as known as ITU-T H.273.
The conversion has been handled per plugin for now. That causes
code duplication a lot also some plugins might not be updated with newly introduced
color{matrix,transfer,primaries} enum value(s).
Instead of handling it per plugin, centralized handling can remove such
code duplication and make plugins be up-to-dated.
The extmap attribute allows mapping RTP extension header IDs to
well-known RTP extension header specifications. See RFC8285 for details.
We store the extmap attribute either as string in the caps
extmap-X=extensionname
where X is the integer extension header ID, or as 3-tuple of strings
extmap-X=<direction,extensionname,extensionattributes>
where direction or extensionattributes are allowed to be the empty
string.
Both formats are allowed because usually only the extension name is
given and it's much simpler to handle in caps.
We use this property in gst_gl_display_egl_from_gl_display, to set
foreign_display for the new GstGLDisplayEGL instance. This fixes a
problem where gst_gl_display_egl_finalize calls EglTerminate on a
pre-existing EGL connection.
It seems that eglCreatePlatformWindowSurfaceEXT is failing (with
EGL_BAD_ALLOC) because it thinks an EGL surface has already been created
for the wl_egl_window. The reason is that the "driver_private" field of
the wl_egl_window is getting clobbered by the function
wl_proxy_set_queue().
Since a wl_egl_window is not a wl_proxy, it shouldn't be passed to
wl_proxy_set_queue(). It just wraps a wl_surface (which is a wl_proxy).
And it looks like the queue for that surface is getting set earlier on
in the function anyway.
See https://gitlab.freedesktop.org/gstreamer/gst-plugins-base/issues/621#note_184582
Body_offset mean that so much data have been written.
Without this patch n_vectors somtimes becomes one more than it should
and then there will be an vector that have a random size causing
writev_bytes to cause a "Bad address" error.
This patch fixes the following critical warning:
CRITICAL **: 11:33:32.843: Unknown GL format 0x0 provided
It would happen during the setup of a second pipeline involving the DMABuf
uploader, typically with a v4l2src element. The warning was raised because the
uploader had a cached EGLImage already filled but the formats were not
synchronized accordingly.
The "field-order" is related for all interlace_mode modes except the
"progressive" mode. So instead of or'ing each mode we can use the
already supported GST_VIDEO_INFO_IS_INTERLACED macro.
This makes a pipeline below works:
little endian:
gst-launch-1.0 videotestsrc ! video/x-raw,format=P010_10LE ! glimagesink
big endian:
gst-launch-1.0 videotestsrc ! video/x-raw,format=P010_10BE ! glimagesink
gst_meta_api_type_register() assumes that the last tags element is null, but it wasn't
==17422==ERROR: AddressSanitizer: global-buffer-overflow on address 0x7f4e2a67c998 at pc 0x7f4e2a0c92ac bp 0x7ffcc41f80b0 sp 0x7ffcc41f80a0
READ of size 8 at 0x7f4e2a67c998 thread T0
#0 0x7f4e2a0c92ab in gst_meta_api_type_register ../subprojects/gstreamer/gst/gstmeta.c:94
#1 0x7f4e2a5582c3 in gst_video_afd_meta_api_get_type ../subprojects/gst-plugins-base/gst-libs/gst/video/video-anc.c:1146
#2 0x404c7c in invoke_get_type (/home/ubuntu/gst-build/build/tmp-introspect5gv1rovo/GstVideo-1.0+0x404c7c)
#3 0x406b5c in dump_irepository (/home/ubuntu/gst-build/build/tmp-introspect5gv1rovo/GstVideo-1.0+0x406b5c)
#4 0x407089 in main (/home/ubuntu/gst-build/build/tmp-introspect5gv1rovo/GstVideo-1.0+0x407089)
#5 0x7f4e295b4b6a in __libc_start_main (/lib/x86_64-linux-gnu/libc.so.6+0x26b6a)
#6 0x404479 in _start (/home/ubuntu/gst-build/build/tmp-introspect5gv1rovo/GstVideo-1.0+0x404479)
0x7f4e2a67c998 is located 40 bytes to the left of global variable 'tags' defined in '../subprojects/gst-plugins-base/gst-libs/gst/video/video-anc.c:1232:25' (0x7f4e2a67c9c0) of size 24
0x7f4e2a67c998 is located 0 bytes to the right of global variable 'tags' defined in '../subprojects/gst-plugins-base/gst-libs/gst/video/video-anc.c:1141:25' (0x7f4e2a67c980) of size 24
SUMMARY: AddressSanitizer: global-buffer-overflow ../subprojects/gstreamer/gst/gstmeta.c:94 in gst_meta_api_type_register
Add max-reorder property to make the old hard coded reordering limit of
100 configurable. It's particularly useful in some scenarios to set
max-reorder=0 to disable the behavior that the depayloader will drop
packets.
Note that although the default value is 100, the default limit has
increased with one because of the changed if-test. This was done to
allow the max-reorder value to be more intuitive. See tests.
Instead of checking if the requested GL API is GLES2 (because ANY can
be set) the string is matched with the GLES2 prefix, and if so, then
the string is offset.
RFC 7826 recommends (but does not require) starting at 0,
but at least one known server implementation fails to copy
request sequence numbers <1 into responses due to an
incorrect null check.
The server known to exhibit this behavior is the Parrot
Streaming Server, serving video from their UAV devices.
A fix has been submitted upstream as well:
https://github.com/Parrot-Developers/librtsp/pull/2
The Parrot developers are known to have tested with LibVLC.
In WireShark debugging, LibVLC appears to start with a CSeq
of 2, which is likely why this bug went unnoticed.
This reverts 487595a7d6, which set this to 0 citing the
RFC. The switch to 0 was thus a recent one; it's therefore
possible server implementors relied on the previous
GStreamer client behavior in their tests as well.
Fixes#624.
Since we started depending on GLib 2.44, we can be sure this macro is
defined (it will be a no-op on compilers that don't support it). For
plugins we should just start using `G_DECLARE_FINAL_TYPE` which means we
no longer need the macro there, but for most types in base/gst-libs we
don't want to break ABI, which means it's better to just keep it like it
is (and use the `#ifdef` instead).
The problem is that Gobject Introspections does not understand the const
gfloat matrix[16] as an matrix but as an array of gfloasts but as just
one gfloat.
To fix this i added the annotation to the parameter
descriptions.
This came up in the case where v4l2 sets caps with colorimetry=NULL, and
then tries to parse back the colorimetry, causing a crash in
gst_video_get_colorimetry() because of g_str_equal(). We fix this by
making sure the only caller of the function never calls it with a null
colorimetry string.
SMPTE ST 2084 transfer characteristics (a.k.a ITU-R BT.2100-1 perceptual quantization, PQ)
is used for various HDR standard.
With ST 2084, we can represent BT 2100 (Rec. 2100). BT 2100 defines
various aspect of HDR such as resolution, transfer functions, matrix, primaries
and etc. It uses BT2020 color space (primaries and matrix) with PQ or HLG
transfer functions.
The code for this is mostly lifted from audiobuffersplit, it
allows use cases such as keeping the buffers output by compositor
on one branch and audiomixer on another perfectly aligned, by
requiring the compositor to output a n/d frame rate, and setting
output-buffer-duration to d/n on the audiomixer.
The old output-buffer-duration property now simply maps to its
fractional counterpart, the last set property wins.
Packed 10 bits per each R, G and B channel with MSB 2bits alpha channel.
This format is mapped to Windows' DXGI_FORMAT_R10G10B10A2_UNORM format which is
required for 10bits HDR rendering.
Note that this RGB10A2_LE format is R - B channel swapped version of BGR10A2_LE
... if subclass didn't update values. Note that the mastering-display-info
and content-light-level might be updated by user defined value (e.g., encoding option).
Introduce HDR signalling methods
* GstVideoMasteringDisplayInfo: Representing display color volume info.
Defined by SMPTE ST 2086
* GstVideoContentLightLevel: Representing content light level specified in
CEA-861.3, Appendix A.
Closes https://gitlab.freedesktop.org/gstreamer/gst-plugins-base/issues/400
By using strtoul(), invalid values will get mapped to MAXULONG and we
would have to check errno. They won't get mapped to 0.
To solve this, use the signed g_ascii_strtoll(). This will map errors to
0 or G_MAXINT64 or G_MININT64, and the valid range for GstDateTime is >
0 and <= 9999 so we can directly check for this here.
Fixes https://gitlab.freedesktop.org/gstreamer/gstreamer/issues/384
As part of commit 808e7127, we prefixed the `GstWlWindow`'s `shell`
field with wl_, to differentiate it from the other types of shells a
Wayland compositor might support. However, this is apparently a struct
that we expose to our users, so changing it means we have an API break.
Fixes https://gitlab.freedesktop.org/gstreamer/gst-plugins-base/issues/592
Add the possible to limit the Content-Length
Define an appropriate request size limit and reject requests exceeding
the limit (413 Request Entity Too Large)
When the glupload element renegotiates the caps, set_caps will reset the
method_impl to NULL, but the method will be kept. transform_caps tries
to use the method_impl to transform the caps, because a method is set,
but will segfault.
Make rtspconnection a little more strict to RFC2326.
Make sure that CSeq is in every RTSP message and that CSeq is valid.
Also break the build_next loop if any parsing fails, By acting on
the builder->status code.
video-anc.h💯 Error: GstVideo: identifier not found on the first line:
* Active Format Description (AFD) support
^
video-anc.h:207: Error: GstVideo: identifier not found on the first line:
* Bar data support
^
video-anc.h:228: Warning: GstVideo: "@top_bar_flag" parameter unexpected at this location:
* @top_bar_flag : flag indicating presence of top bar field
^
This is inconsistent with other add_meta methods such as
gst_buffer_add_video_meta , which will return NULL without
logging when gst_video_info_set_format fails.
It is up to the caller to check the return value of the
function, and log if appropriate.
It's invalid to have a 'interlace-mode=alternate' without the Interlaced caps
feature as well.
Modify gst_video_info_from_caps() to reject such case so we can easily
spot them in bugged elements.
gst_gl_memory_setup_buffer() was marked as introspectable=0
anyway, so might just as well mark it as '(skip)' and suppress
the warning. Reason is the (element-type gpointer) on wrapped_data.
gstglmemory.c:1426: Warning: GstGL: gst_gl_memory_setup_buffer: argument wrapped_data: Missing (element-type) annotation
gstglmemory.c:1426: Warning: GstGL: gst_gl_memory_setup_buffer: argument wrapped_data: Missing (element-type) annotation
egl/gstegl.h:40: Warning: GstGL: symbol='EGL_EGLEXT_PROTOTYPES': Unknown namespace for symbol 'EGL_EGLEXT_PROTOTYPES'
gstaudiometa.c:382: Warning: GstAudio: gst_buffer_add_audio_meta: return value: Invalid non-constant return of bare structure or union; register as boxed type or (skip)
The function rtcp_packet_min_length() returns a length for each known type
and -1 for unknown types. This change fixes the test accordingly and silences
the following warning.
gstrtcpbuffer.c:567:12: error: comparison of constant -1 with expression of type 'GstRTCPType' is always false
[-Werror,-Wtautological-constant-out-of-range-compare]
if (type == -1)
Fix the following warnings by adding casts.
gstdiscoverer.c:1801:17: error: format specifies type 'unsigned long' but the argument has type 'off_t' (aka 'long long') [-Werror,-Wformat]
location, file_status.st_size, file_status.st_mtime);
^~~~~~~~~~~~~~~~~~~
gstdiscoverer.c:1801:38: error: format specifies type 'long long' but the argument has type '__darwin_time_t' (aka 'long') [-Werror,-Wformat]
location, file_status.st_size, file_status.st_mtime);
https://gitlab.freedesktop.org/gstreamer/gst-plugins-base/issues/570
Before a gap event is pushed downstream a segment event must be pushed
since the gap event can cause packet concealment downstream and hence
data flow. Since concealment before receiving any data packets usually
doesn't make any sense, the gap event is not sent downstream.
Alternatively one could generate a default caps and segment event, but
no need to complicate things until it's proven necessary.
https://bugzilla.gnome.org/show_bug.cgi?id=773104https://gitlab.freedesktop.org/gstreamer/gst-plugins-base/issues/301
The former code allowed an attacker to create a heap overflow by
sending a longer than allowed session id in a response and including a
semicolon to change the maximum length. With this change, the parser
will never go beyond 512 bytes.
Using a single condition variable for synchronization across all GL
messages is very slow on Windows and uses up to 20% CPU usage in some
workloads due to lock contention and false broadcasts.
Using per-message event handles reduces the CPU usage to negligible
amounts despite having to allocate a new event handle for each
message.
Implement the prepare and check functions according to the
documentation by returning TRUE when events should be dispatched
via the dispatch function.
As wl_display_read_events never blocks we can call it unconditionally
without looking at the poll status.
This simplifies the implementation and gets rid of a race where the
mainloop could get blocked due to nobody actually reading the events
from the wayland connection.
The ->skip_buffer implementation in videoaggregator replicates
the behaviour of the aggregate method to determine whether a
buffer can be skipped
(https://bugzilla.gnome.org/show_bug.cgi?id=781928).
This fixes a typo that made it so the start time of the buffer
was calculated against the output segment, not the segment of
the relevant sinkpad, which caused buffers to be skipped when
for example a sinkpad had received a segment which base had
been modified by a pad offset somewhere along the way.
This simply makes the calculation of the buffer start time
identical to the calculation in aggregate()
Doing so involves retrieving the current viewport from OpenGL which as
with any glGet operation, is expensive.
This means that the various sinks need to reset the viewport on draw.
In the process, fix resizing on cocoa.
If we only ever make it to READY, transform_caps can create an
internal convert object that will never be freed by basetransform's
stop vmethod (PAUSED->READY).
This allows us to output audio samples without discarding
any input frames, which is useful for some formats/codecs
(e.g. the MonkeysAudio decoder implementation in ffmpeg
which will might return e.g. 16 output buffers for an
input buffer for certain files).
In the past decoder implementations just concatenated
the returned audio buffers until a full frame had been
decoded, but that's no longer possible to do efficiently
when the decoder returns audio samples in non-interleaved
layout.
Allowing subframes to be output before the entire input
frame is decoded can also be useful to decrease startup
latency/delay.
https://gitlab.freedesktop.org/gstreamer/gst-libav/issues/49
The use of mediump as a specifier in GLSL shaders will have limited
resolution and when used as texture coordinates may become inaccurate
over texture sizes of 1024.
The function fill_bytes could sometimes return a value greater than zero
and in the same time set the GError.
Function read_bytes calls fill_bytes in a while loop. In the special
case above it would call fill_bytes with error already set.
Thus resulting in "GError set over the top of a previous GError".
Solved this by clearing GError when return value is greater than zero.
Actions are taken depending on error type by caller of read_bytes. Eg.
with EWOULDBLOCK gst_rtsp_source_dispatch_read will try to read the
missing bytes again (GST_RTSP_EINTR )
https://gitlab.freedesktop.org/gstreamer/gst-plugins-base/issues/445
