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)
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.
This test takes a long time. It tests ca. 8900 conversion
combinations, and then it also runs each conversion for
at least 100ms in order to come up with some kind of benchmark.
Remove the benchmarking from the unit test, we have a separate
benchmarking tool for that now.
Also split the conversions into groups and run those as
separate checks, which allows better parallelisation at
the runner level (normal runs and when using valgrind).
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
We're creating buffers with one sample here for some reason. The
actual value of the segment stop is irrelevant for what we're testing
here, so lower it to 10ms so that we create fewer buffers which speeds
things up on slow machines and in valgrind.
../subprojects/gst-plugins-base/tests/check/elements/audiorate.c(192): warning C4047
Meaningful validation at that point seems to checking output GstAudioFormat
of gst_audio_format_from_string()
This will only duplicate buffers if the gap between two consecutive
buffers is up to fill-until nsec. If it's larger, it will only output
the new buffer and mark it as discont.
New casts to avoid the the warnings mentioned below. While at it, move
some existing casts (introduced at 61bc909189) to use
GPOINTER_TO_INT too.
[458/673] Compiling C object 'tests/check/7d01337@@libs_video@exe/libs_video.c.obj'.
../tests/check/libs/video.c: In function 'fourcc_get_size':
../tests/check/libs/video.c:160:10: warning: cast from pointer to integer of different size [-Wpointer-to-int-cast]
return (unsigned long) p->endptr;
^
In file included from ../tests/check/libs/video.c:32:
../tests/check/libs/video.c: In function 'test_video_formats':
../tests/check/libs/video.c:563:39: warning: cast from pointer to integer of different size [-Wpointer-to-int-cast]
fail_unless_equals_int (size, (unsigned long) paintinfo.endptr);
^
And more.
https://gitlab.freedesktop.org/gstreamer/gst-plugins-base/merge_requests/94
With commit 3f184c3abc, the gst_dir variable becomes unusable in
windows build. Moving it to linux scope to avoid warning:
[433/673] Compiling C object 'tests/check/7d01337@@libs_profile@exe/libs_profile.c.obj'.
../tests/check/libs/profile.c: In function 'profile_suite':
../tests/check/libs/profile.c:688:10: warning: unused variable 'gst_dir' [-Wunused-variable]
gchar *gst_dir;
^~~~~~~
Also fix a typo in the comment.
It is really easy to break the API and insert a new video format in the
middle of the enum instead of at the end. This minimal test should catch
the most obvious errors. Ideally, this test should be updated after new
format have been added, so that it won't allow further modification to
the enumeration API.
rtpbasedepayload.c:126:5: error: unknown conversion type character 'z' in format [-Werror=format]
profile.c:688:10: error: unused variable 'gst_dir' [-Werror=unused-variable]
Allow fallback to orc subproject if any.
Additionally 'dependencies' keyword is removed from find_library,
because it's invalid keyword for find_library.
Binding the vertex array to 0 will unbind everything else already.
In the previous order older versions of the Intel GL driver caused
errors to be printed for every single call when disabling the vertex
attrib arrays after binding the vertex array to 0.
ISO 14496-3 defines that audioObjectType 5 is a special case that
indicates SBR is present and that an additional field has to be
parsed to find the true audioObjectType.
There are two ways of signaling SBR within an AAC stream - implicit
and explicit (see [1] section 4.2). When explicit signaling is used,
the presence of SBR data is signaled by means of the SBR
audioObjectType in the AudioSpecificConfig data.
Normally the sample rate is specified by an index into a
table of common sample rates. However index 0x0f is a special case
that indicates that the next 24 bits contain the real sample rate.
[1] https://www.telosalliance.com/support/A-closer-look-into-MPEG-4-High-Efficiency-AACFixes#39
The old API would only assert or return an invalid timecode, the new API
returns a boolean or NULL. We can't change the existing API
unfortunately but can at least deprecate it.
... instead of hardcoded ':', since G_SEARCHPATH_SEPARATOR_S
varies depending on OS (e.g., ':' for *nix and ';' for Windows).
Note that, when the seperator is not specified explicitly, Meson
will use ';' for Windows and ':' for *nix respectively.
According to RFC3611, the extended report blocks in XR packet can
have variable length. To visit each block, the iterator should look
into block header. Once XR type is extracted, users can parse the
detailed information by given functions.
Loss/Duplicate RLE
The Loss RLE and the Duplicate RLE have same format so
they can share parsers. For unit test, randomly generated
pseudo packet is used.
Packet Receipt Times
The packet receipt times report block has a list of receipt
times which are in [begin_seq, end_seq).
Receiver Reference Time paser for XR packet
The receiver reference time has ntptime which is 64 bit type.
DLRR
The DLRR report block consists of sub-blocks which has ssrc, last RR,
and delay since last RR. The number of sub-blocks should be calculated
from block length.
Statistics Summary
The Statistics Summary report block provides fixed length
information.
VoIP Metrics
VoIP Metrics consists of several metrics even though they are in
a report block. Data retrieving functions are added per metrics.
https://bugzilla.gnome.org/show_bug.cgi?id=789822
The unit test makes mixed usage of ret value. Sometimes its does
stores an enum and at other moment a boolean. Also fix test
using boolean instead of the correct enum value.
https://bugzilla.gnome.org/show_bug.cgi?id=783521
This removes the crossfade-ratio property and replaces it with an
operator property. Currently this implements the following operators:
- SOURCE: Copy over the source and don't look at the destination
- OVER: Default blending of the source over the destination
- ADD: Like OVER but simply adding the alpha instead
See the example for how to implement crossfading with this.
https://bugzilla.gnome.org/show_bug.cgi?id=797169
The previous failure was a timeout which was due to the sending pipeline
pushing test buffer *before* the remote client was accepted. We would
therefore never get the buffer on the other side.
While the client socket would indeed appear as "connected", this doesn't
mean that the remote server side did "accept" it (which is where we then
add it to the list of remote parties to which data will be sent).
The problem isn't with the element implementation, but to the nature of
TCP 3-way handshake.
In order to make the test reliable, wait for the sink to have accepted
the remote client (by checking the number of handles) before sending out
test buffers.
Add a source-info property that will read/write meta to the buffers
about RTP source information. The GstRTPSourceMeta can be used to
transport information about the origin of a buffer, e.g. the sources
that is included in a mixed audio buffer.
A new function gst_rtp_base_payload_allocate_output_buffer() is added
for payloaders to use to allocate the output RTP buffer with the correct
number of CSRCs according to the meta and fill it.
RTPSourceMeta does not make sense on RTP buffers since the information
is in the RTP header. So the payloader will strip the meta from the
output buffer.
https://bugzilla.gnome.org/show_bug.cgi?id=761947
Add a helper to set the interlacing mode while creating the GstVideoInfo
in addition to format and resolution. Using this helper will ensure that
size is correctly calculated for split-field interlacing mode.
https://bugzilla.gnome.org/show_bug.cgi?id=796106
Add a new interlace mode enum to represent buffers containing a single
field of an interlaced video in a buffer. The name is based on the
equivalent video format in the V4L2 API, V4L2_FIELD_ALTERNATE:
https://01.org/linuxgraphics/gfx-docs/drm/media/uapi/v4l/field-order.html
Since caps fields are optional, we also introduce a new caps feature,
"format:Interlaced" that always goes with "alternate" interlace mode to ensure
that caps for this incompatible format are incompatible with other interlaced
and progressive video caps.
https://bugzilla.gnome.org/show_bug.cgi?id=796106
This allows consumers of the gstgl dependency where gstgl is optional
to do things like:
config_data.set('HAVE_GST_GL', gstgl_dep.found())
deps = [gstvideo_dep, gstgl_dep]
meaning they can still use the dep unconditionally. With the
disabler we would just disable the whole target even if the
gstgl part was an optional extra. We can add an option to
dependency() later to let users/consumers of the dep decide
if they want a not-found dependency or a disabler instead.
The fomula, 'offset = time / rate', is correct only if
the rate is never changed. When the rate is changed,
the offset should be re-calculated based on the previous
offset.
https://bugzilla.gnome.org/show_bug.cgi?id=791269
Because audioconvert can now convert between interleaved and non-interleaved,
this pipeline fails on the upstream capsfilter not being able to fixate its
output caps. This is unavoidable.
e4bf9ed8f0 was not quite right and changed
the wrong thing. Intead we needed to change the multiplication order
and should have kept the previous to/from matrices as is done in this
patch.
