Packets of a given PID are meant to have sequential continuity counters
(modulo 16). If there are not sequential, this is the sign of a broken
stream, which we then consider as a discontinuity.
But if that new packet is a frame start (PUSI is true), then we can resume
from that packet without any damage.
Sometimes, one wants to force a clock on some pipelines - for instance,
when testing TSN related pipelines, one usually uses GstPtpClock or
CLOCK_REALTIME (assuming system realtime clock is in sync with network
one). Until now, one needs to write an application for that - not
difficult, but quite boring if one just wants to test something. This
patch presents a new element to help that: clockselect.
clockselect is a pipeline with two properties to select a clock. One
property, "clock-id", enables one to choose between "monotonic",
"realtime", "ptp" or "default" clock - where default keeps pipeline
behaviour of choosing a clock based on its elements. The other property,
"ptp-domain" gives one the choice of which PTP domain should be used.
Some very simple tests also added for this new element.
By passing NULL to `g_signal_new` instead of a marshaller, GLib will
actually internally optimize the signal (if the marshaller is available
in GLib itself) by also setting the valist marshaller. This makes the
signal emission a bit more performant than the regular marshalling,
which still needs to box into `GValue` and call libffi in case of a
generic marshaller.
Note that for custom marshallers, one would use
`g_signal_set_va_marshaller()` with the valist marshaller instead.
The peer is already gone when pad_removed_cb() called, so the ghost cannot
be removed. Use g_object_set_data() instead to remember the ghost pad.
Copied from similar code in GstRTPBin.
Since we are not requiring that profile equals GST_JPEG2000_PARSE_PROFILE_BC_SINGLE,
(as the standard requires) we can allow profile to be null. We relax this condition because
OpenJPEG can't create broadcast profiles.
1. only recalculate src codec format if sink caps change
2. use correct value for "jp2c" magic in J2C box ID
3. only parse J2K magic once, and store result
4. more sanity checks comparing caps to parsed codec
This adds two properties:
* scte-35-pid: If not 0, enables the SCTE-35 support for the current
program. This will write the proper PMT and send SCTE-35 NULL
commands (i.e. heartbeats) at a regular interval
* scte-35-null-interval: This specifies the interval at which the
NULL commands should be sent
Sending SCTE-35 commands is done by creating the appropriate SCTE-35
GstMpegtsSection and then sending them on the muxer. See the
associated example
We were unconditionally adding top-level descriptors in the PMT which
were only related to bluray support for PS3 (from 10 years ago).
These should be re-added conditionally
This went un-noticed for 6 years :( The issue is that for short
sections (without subtables and CRC), we would always fail when
checking whether we had enough data or not and then default to the
long section checking.
Use the long section checking would then cause interesting side-effects
for short sections (such as believing they were already seen and therefore
would be dropped/ignored).
This allows selecting whether we continue updating our last known
upstream timecode whenever a new one arrives or instead only keep the
last known one and from there on count up.
This uses the last known upstream timecode (counted up per frame), or
otherwise zero if none was known.
The normal last-known timestamp uses the internal timecode as fallback
if no upstream timecode was ever known.
The convert-error signal is emitted whenever we get a GstFlowReturn
other than GST_FLOW_OK. The handler can then decide what to convert that
into - for instance, return the same GstFlowReturn to not convert it.
The default handler will act according to the ignore-error,
ignore-notlinked, ignore-notnegotiated and convert-to properties. If a
handler is connected, these properties are ignored.
In file included from ../../../../dist/linux_x86_64/include/gstreamer-1.0/gst/gstbin.h:27,
from ../../../../dist/linux_x86_64/include/gstreamer-1.0/gst/gst.h:35,
from ../gst/rtp/gstrtpsink.h:23,
from ../gst/rtp/gstrtpsink.c:49:
In function ‘gst_rtp_sink_start’,
inlined from ‘gst_rtp_sink_change_state’ at ../gst/rtp/gstrtpsink.c:509:11:
../../../../dist/linux_x86_64/include/gstreamer-1.0/gst/gstelement.h:422:18: error: ‘%s’ directive argument is null [-Werror=format-overflow=]
422 | gchar *__txt = _gst_element_error_printf text; \
../gst/rtp/gstrtpsink.c:476:3: note: in expansion of macro ‘GST_ELEMENT_ERROR’
476 | GST_ELEMENT_ERROR (self, RESOURCE, NOT_FOUND,
| ^~~~~~~~~~~~~~~~~
../gst/rtp/gstrtpsink.c: In function ‘gst_rtp_sink_change_state’:
../gst/rtp/gstrtpsink.c:477:37: note: format string is defined here
477 | ("Could not resolve hostname '%s'", remote_addr),
| ^~
In file included from ../../../../dist/linux_x86_64/include/gstreamer-1.0/gst/gstbin.h:27,
from ../../../../dist/linux_x86_64/include/gstreamer-1.0/gst/gst.h:35,
from ../../../../dist/linux_x86_64/include/gstreamer-1.0/gst/rtp/gstrtpdefs.h:27,
from ../../../../dist/linux_x86_64/include/gstreamer-1.0/gst/rtp/rtp.h:25,
from ../gst/rist/gstristsink.c:72:
In function ‘gst_rist_sink_setup_rtcp_socket’,
inlined from ‘gst_rist_sink_start’ at ../gst/rist/gstristsink.c:658:10,
inlined from ‘gst_rist_sink_change_state’ at ../gst/rist/gstristsink.c:801:13:
../../../../dist/linux_x86_64/include/gstreamer-1.0/gst/gstelement.h:422:18: error: ‘%s’ directive argument is null [-Werror=format-overflow=]
422 | gchar *__txt = _gst_element_error_printf text; \
../gst/rist/gstristsink.c:595:3: note: in expansion of macro ‘GST_ELEMENT_ERROR’
595 | GST_ELEMENT_ERROR (sink, RESOURCE, NOT_FOUND,
| ^~~~~~~~~~~~~~~~~
../gst/rist/gstristsink.c: In function ‘gst_rist_sink_change_state’:
../gst/rist/gstristsink.c:596:37: note: format string is defined here
596 | ("Could not resolve hostname '%s'", remote_addr),
| ^~
Allows for "low latency" mpeg-ts mode which is not standard, but somewhat common.
For this to work the sender has to put timestamps at a higher frequency than the spec requires.
PES packets with size 0 are unbounded, and
could therefore overflow the 32-bit size
accumulator.
Add a 32MB limit, which is larger than
any PES packet should ever get. If one does,
then output a 32MB chunk and continue.
A guint32 greater than 2^31 would be interpreted as negative by
gst_util_uint64_scale_int() and critical. Use the 64-bit integer version
of the function instead.
Don't signal a pipeline error when processing incomplete
j2pk PES packets that are too small. That can happen normally
during a DISCONT and shouldn't shut down the whole pipeline
Remove some custom and incomplete seek calculation
logic in favour of gst_segment_do_seek(), and
short-circuit any actual seeking or recalculation
if the position didn't change and just send an updated
segment directly.
This removes the custom seeking logic in favour of
using standard core seek handling.
The default behaviour of rtponviftimestamp is to drop buffers
outside the segment. This creates obvious problems for reverse
playback.
The ONVIF specification unfortunately doesn't describe how to handle
that specific use case, but we can expose a property to let the
user disable the dropping behaviour, and forward these buffers with
a G_MAXUINT64 ONVIF timestamp.
Also modify rtponvifparse to handle such timestamps appropriately.
We reject caps with other framerates as it's impossible to generate
timecodes unless we actually know a constant framerate. Reflect this
also in the pad template caps.
Instead of using a static hardcoded PCR interval, allow the user
to configure it.
Also revert back the default to a 40 ms interval, that was changed
in recent patches for no good reason.
There's no point in working with invalid LTC timestamps as all future
calculations will be wrong based on this, and invalid LTC timestamps can
sometimes be read via the audio input.
This patch just enforces boudaries for the access to the
standard_deviation array (64 floats). Such case can be
seen with a corrupted stream, where there's no hope to
obtain a valid decoded frame anyway.
https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/issues/1002
The D bit is meant to be set whenever there is a discontinuity
in transmission, and directly maps to the DISCONT flag.
The E bit is not meant to be set on every buffer preceding a
discontinuity, but only on the last buffer of a contiguous section
of recording. This has to be signaled through the unfortunately-named
"discont" field of the custom NtpOffset event.
Based on a patch by
Georg Lippitsch <glippitsch@toolsonair.com>
Vivia Nikolaidou <vivia@toolsonair.com>
Using libltc from https://github.com/x42/libltc
We now have a single property to select the timecode source that should
be applied, and for each timecode source the timecode is updated at
every frame. Then based on a set mode, the timecode is added to the
frame if none exists already or all existing timecodes are removed and
the timecode is added.
In addition the real-time clock is considered a proper timecode source
now instead of only allowing to initialize once in the beginning with
it, and also instead of just taking the current time we now take the
current time at the clock time of the video frame.
The SPS parsing functions take a parse_vui_param flag
to skip VUI parsing, but there's no indication in the output
SPS struct that the VUI was skipped.
The only caller that ever passed FALSE seems to be the
important gst_h264_parser_parse_nal() function, meaning - so the
cached SPS were always silently invalid. That needs changing
anyway, meaning noone ever passes FALSE.
I don't see any use for saving a few microseconds in
order to silently produce garbage, and since this is still
unstable API, let's remove the parse_vui_param.
Fix a recently introduced segfault. Don't de-reference a NULL
SPS pointer when attempting to update source caps before SPS
has been seen in the stream.
Interleaved frames can be fragmented between
incoming frames. Thus, we can have multiple
frames within the single input frame, as well as
incomplete frame. Now it preserves parsing
state and handle both situations.
Fixes#991
The length of the TCP payload is the IP plus TCP header length
subtracted from the IP datagram length specified in the IP header.
Prior to this, the size was calculated incorrectly, considering
all data after TCP header as a payload till the end of a packet.
Fixes#995
If recording is set to FALSE after the last audio or video buffer and
before the EOS event then recording stop is never signalled.
Similarly, we should signal recording stop once both audio and video are
EOS, regardless of the recording property, as there's nothing to be
recorded anymore.
