flexelint (http://www.gimpel.com/html/flex.htm) static code analyser
complained about implicit conversions from unsigned to signed, so I added
explicit conversions.
Ideally, the size parameter of gst_mpd_parse function should be unsigned,
but I don't want to change the API.
The duration_to_ms function converts a time specified by year, month, day,
hour, minute, second, millisecond to a millisecond value. Because all the
arguments are positive numbers, the result must also be positive.
This patch changes the returned value from a gint64 to a guint64 type.
Improved dash_mpd unit tests by adding new tests that parse the Period element.
Code coverage reported by lcov for dash/gstmpdparser.c is:
lines......: 43.0% (985 of 2290 lines)
functions..: 47.5% (67 of 141 functions)
According to ISO/IEC 23009-1:2014(E), chapter 5.3.2.1
"The Period extends until the PeriodStart of the next Period, or until
the end of the Media Presentation in the case of the last Period."
This means that a configured value for optional attribute period duration
should be ignored if the next period contains a start attribute or it is
the last period and the MPD contains a mediaPresentationDuration attribute.
https://bugzilla.gnome.org/show_bug.cgi?id=750797
The gst_mpdparser_get_rep_idx_with_max_bandwidth function assumes
representations are ordered by bandwidth and incorrectly returns the
first one when wanting the one with minimum bandwidth.
Corrected gst_mpdparser_get_rep_idx_with_max_bandwidth function to get the
correct representation in case max_bandwidth parameter is 0.
https://bugzilla.gnome.org/show_bug.cgi?id=751153
Added a check for a_node->ns before accessing a_node->ns->href in
gst_mpdparser_get_xml_node_namespace. This could happen if the xml
is missing the default namespace.
https://bugzilla.gnome.org/show_bug.cgi?id=750866
When the 'ignore-eos' property is set on a pad, compositor will keep resending
the last buffer on the pad till the pad is unlinked. We count the buffers
received on appsink, and if it's more than the buffers sent by videotestsrc, the
test passes.
Lets not cram everything into a single test - this would render the test name
useless for quick diagnosis. Having separate tests for the optional feature is
also verifying the behaviour when the feature is off.
Rather than one of the input pad video info's.
The test checking this was not constraining the output frame size
to ensure that the out of frame stream was not being displayed.
We verify that all the buffers on an obscured sinkpad are skipped by overriding
the map() function in the GstVideoMeta of the buffers to set a variable when
called. We also test that the buffers do get mapped when they're not obscured.
Blame^WCredit for the GstVideoMeta map() idea goes to Tim.
https://bugzilla.gnome.org/show_bug.cgi?id=746147
GstPhotography enables new paths in wrappercamerabinsrc that allows
the source to be notified about the capture caps and provide an
alternative caps if desired bypassing the negotiation (this doesn't
seem like a good idea these days). To make sure it keeps working
until we remove it from the API in favor of standard caps negotiation
features this test was added.
It adds 3 extra tests with a simple test source that will:
1) Test that capturing with ANY caps work
2) Test that capturing with a fixed caps work
3) Test that capturing with a fixed caps and having the source
pick a different resolution from GstPhotography API works
by having wrappercamerabinsrc crop the capture to the final
requested dimensions
A bitmask is 64 bits, but integer immediates are passed as int
in varargs, which happen to be 32 bit with high probability.
This triggered a valgrind jump-relies-on-uninitalized-value
report well away from the site, since it doesn't trigger on
stack accesses, and there must have been enough zeroes to stop
g_object_set at the right place.
The first output MPEG-TS packet that corresponds to a video input
buffer which had the delta flag cleared (i.e. was a keyframe)
should have the delta flag cleared as well.
This is needed e.g. by tcpserversink in order to keep track
of the last keyframe and be able to burst data to newly-
connected clients.
https://bugzilla.gnome.org/show_bug.cgi?id=706872
This reverts commit 1c77d12ce8.
"interlaced" in the caps don't mean the same thing as the SOF2 marker in the
JPEG format. This test passes because of broken behaviour.
The flush stop could have happened between the source trying
to push the segment event and the buffer, this would cause a warning.
Prevent that by taking the source's stream lock while flushing.
https://bugzilla.gnome.org/show_bug.cgi?id=742684
With the current audiomixer, the input caps need to be the same,
otherwise there is an unavoidable race in the caps negotiation. So
enforce that using capsfilters
https://bugzilla.gnome.org/show_bug.cgi?id=742684
Get rid of now-useless packetizer struct and just
call internal functions directly. Also remove
version property which is now defunct, not least
because we create the packetizer with the
version in the init function before a version
can be set.
In file included from /home/thiagoss/gst/head/gstreamer/gst/gst.h:54:0,
from /home/thiagoss/gst/head/gstreamer/libs/gst/check/gstcheck.h:34,
from elements/hlsdemux_m3u8.c:27:
../../ext/hls/gstfragmented.h:8:28: error: redundant redeclaration of ‘fragmented_debug’ [-Werror=redundant-decls]
GST_DEBUG_CATEGORY_EXTERN (fragmented_debug);
Move the definition of the category to after the declaration.
Ported from https://github.com/ylatuya/gst-plugins-bad
This still has some unit tests for alternative renditions and
seeking, which are commented out for the time being until we
support them properly.
Use the sticky events to compose the streamheader as they are the
ones that are persisted to config new pads linked. Instead of storing
them ourselves rely on the pad storage that already orders it for us
https://bugzilla.gnome.org/show_bug.cgi?id=732596
Check that conversion to byte-stream/au formats work and that we
can effectively drop broken/invalid NAL units from the resulting
access unit buffer.
https://bugzilla.gnome.org/show_bug.cgi?id=732203
Signed-off-by: Gwenole Beauchesne <gwenole.beauchesne@intel.com>
If an SEI NAL unit with a buffering_period() message is inserted
between an SPS and PPS NAL unit, check that the output buffer still
contain it. i.e. make sure that this SEI message is not dropped.
https://bugzilla.gnome.org/show_bug.cgi?id=732156
Signed-off-by: Gwenole Beauchesne <gwenole.beauchesne@intel.com>
Expose one more libcurl option: CURLOPT_SSH_HOST_PUBLIC_KEY_MD5.
This allows authenticating the server by the MD5 fingerprint of
the server's public key.
https://bugzilla.gnome.org/show_bug.cgi?id=723167
* stream-start-id is mandatory at the beginning, so add that to the
gdp headers
* caps must be sent before new segment, invert the order from legacy
0.10 code
And fix the tests as a ref is now kept for those buffers that compose
the header
Most of the tests weren't updated after the sticky events order
and stream start. Fix that and refactor those tests check that
are the same to some common functions.
Those functions still don't actually test the content but at
least now they are in a single place and can be improved
without replication
Commit 6af387cd5a made h264parse
strip a leading 0x00 byte from some output scenarios. This broke
tests as bs_to_nal test expects one more byte on the output.
Fix this by comparing the output with the expected stripped version,
too.
When outputting in AVC3 stream format, the codec_data should not
contain any SPS or PPS, because they are embedded inside the stream.
In case of avc->bytestream h264parse will push the SPS and PPS from
codec_data downstream at the start of the stream, at intervals
controlled by "config-interval" and when there is a codec_data change.
In the case of avc3->bytstream h264parse detects that there is
already SPS/PPS in the stream and sets h264parse->push_codec to FALSE.
Therefore avc3->bytstream was already supported, except for the stream
type.
In the case of bystream->avc h264parse will generate codec_data caps
from the parsed SPS/PPS in the stream. However it does not remove these
SPS/PPS from the stream. bytestream->avc3 is the same as bytestream->avc
except that the codec_data must not have any SPS/PPS in it.
|--------------+-------------+-------------------|
|stream-format | SPS in-band | SPS in codec_data |
|--------------+-------------+-------------------|
| avc | maybe | always |
|--------------+-------------+-------------------|
| avc3 | always | never |
|--------------+-------------+-------------------|
Amendment 2 of ISO/IEC 14496-15 (AVC file format) is defining a new
structure for fragmented MP4 called "avc3". The principal difference
between AVC1 and AVC3 is the location of the codec initialisation
data (e.g. SPS, PPS). In AVC1 this data is placed in the initial MOOV box
(moov.trak.mdia.minf.stbl.stsd.avc1) but in AVC3 this data goes in the
first sample of every fragment.
https://bugzilla.gnome.org/show_bug.cgi?id=702004
