Make more flexible. There is an extra
gethostbyname2_r@@GLIBC_2.2.5 (getXXbyYY_r.c:217)
in the trace on the build bots (F30).
Fixes the -base and -good valgrind jobs on the 1.16 branch CI.
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.
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.
Continuation of 4fd7a2c783
We check the availability of the high precision floats in GLSL shaders
which involves an OpenGL call and thus is required to be executed on the
OpenGL thread.
The tests were not respecting that and could fail on more strict
drivers.
Tests update for 675415bf2e
Fixes https://gitlab.freedesktop.org/gstreamer/gst-plugins-base/issues/590
We check the availability of the high precision floats in GLSL shaders
which involves an OpenGL call and thus is required to be executed on the
OpenGL thread.
The tests were not respecting that and could fail on more strict
drivers.
Tests update for 675415bf2e
Fixes https://gitlab.freedesktop.org/gstreamer/gst-plugins-base/issues/590
valgrind gets confused with the following piece of code:
var37.i = ORC_CLAMP_SL((orc_int64)var33.i + (orc_int64)var34.i);
Where all variables are orc_int32
If the last WebVTT cue does not have an empty line after it, or if it
does not end with a newline at all, it does not get pushed out and it
won't be displayed.
gst_sub_parse_sink_event() already handles the issue for other subtitle
formats, enable handling it for GST_SUB_PARSE_FORMAT_VTT too.
While at it also add a test for this case.
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
