... by seeking to target offset determined by new seek segment,
rather than that of the previous segment. The latter would typically
seek back to start for a non-accurate seek, and lead to a lot
of skipping in case of an accurate seek.
If one of the inputs is live, add a latency of 2 frames to the video
stream and wait on the clock for that much time to pass to allow for the
LTC audio to be ahead.
In case of live LTC, don't do any waiting but only ensure that we don't
overflow the LTC queue.
Also in non-live LTC audio mode, flush too old items from the LTC queue
if the video is actually ahead instead of potentially waiting forever.
This could've happened if there was a bigger gap in the video stream.
According to H264 ITU standards from 06/19, GST_H264_PROFILE_HIGH_422
(profile_idc = 122) with constraint_set1_flag = 0 and
constraint_set3_flag = 0 can be mapped to high-4:2:2 or high-4:4:4.
GST_H264_PROFILE_HIGH_422 with constraint_set1_flag = 0 and
constraint_set3_flag = 1 can be mapped to high-4:2:2, high-4:4:4,
high-4:2:2-intra or high-4:4:4-intra.
The previous implementation had a very high reproducibility race where
if after a track switch, the ex-active track pad completed a buffer
chain (now returning not-linked) the flow combiner had all their pads in
non-linked state, propagating it as an error and stopping the pipeline.
By resetting the flow combiner in response to RECONFIGURE events that
race is made impossible.
Incrementing it afterwards will always have to phase_index >= 1 and we
will never be at the beginning (0) of the phase again, and thus never
reset timestamp tracking accordingly.
This was broken in bea13ef43b in 2010, and
causes interlace to run into integer overflows after 2^31 frames or
about 5 hours at 29.97fps. Due to usage of wrong types for the integers
this then causes negative numbers to be used in calculations and all
calculations spectacularly fail, leading to all following buffers to
have the timestamp of the first buffer minus one nanosecond.
Allowing the UDP elements to bind on an interface is needed in more
complex networks where there are mutiple networks interfaces without
default gateway
From the documentation of gst_base_parse_set_infer_ts, it should be
disabled for non-audio data. Currently just disabling for all video
parsers that have reordered data: h264, h265, mpeg, mpeg4, vc1. Was
already disabled in h263.
Currently tsdemux timestamps only the PTS, and only issues the DTS if
it's different. In that case, parsers tend to estimate the next DTS
based on the previous DTS and the duration, which can accumulate
rounding errors.
Clean up path objects nicely when shutting down,
first by dropping pointers to elements during dispose,
and then by making sure to drop the ref to the path object
when finalizing the switch bin.
Fixes valgrind checks in the unit test.
Coverity rightly complains that checking a pointer for NULL after
dereferencing it is pointless.
Remove the check, and to be safe, assert that gst_buffer_add_meta
returns non-NULL.
CID 1455485
The message buffers are created using `gst_rtmp_message_new` and thus
always contain a GstRtmpMeta. Add checks to appease Coverity's static
analysis.
CID 1455596
CID 1455384
According to the spec, the least significant bit is reserved and should
always we set to 1. Though, some wrong file has been found. Considering
how low important this reserved bit is, relax the validation.
Related to #660
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.
We generally always prefer the information from upstream for other
metadata (pixel-aspect-ration, etc.) and should also do so here.
Other parsers (h264parse) already do the same.
When sps parsing fails we use a fallback sps from the caps, since we
have got an sps we need to update parser state and header as in the case the
sps was successfully parsed
When sps parsing fails we use a fallback sps from the caps, since we
have got an sps we need to update parser state and header as in the case the
sps was successfully parsed
Closes#503
Set more unhandled flags to general_constraint_indicator_flags field.
The field is required for building "Codecs" parameter as defined
ISO/IEC 14496-15 Annex E. The resulting "Codecs" string might be used
in various places (e.g., HLS/DASH manifest, browser, player, etc)
This is a re-implementation of the RTP elements that are submitted in
2013 to handle RTP streams. The elements handle a correct connection
for the bi-directional use of the RTCP sockets.
https://bugzilla.gnome.org/show_bug.cgi?id=703111
The rtpsink and rtpsrc elements add an URI interface so that streams
can be decoded with decodebin using the rtp:// interface.
The code can be used as follows
```
gst-launch-1.0 videotestsrc ! x264enc ! rtph264pay config-interval=3 ! rtpsink uri=rtp://239.1.1.1:1234
gst-launch-1.0 videotestsrc ! x264enc ! rtph264pay config-interval=1 ! rtpsink uri=rtp://239.1.2.3:5000
gst-launch-1.0 rtpsrc uri=rtp://239.1.2.3:5000?encoding-name=H264 ! rtph264depay ! avdec_h264 ! videoconvert ! xvimagesink
gst-launch-1.0 videotestsrc ! avenc_mpeg4 ! rtpmp4vpay config-interval=1 ! rtpsink uri=rtp://239.1.2.3:5000
gst-launch-1.0 rtpsrc uri=rtp://239.1.2.3:5000?encoding-name=MP4V-ES ! rtpmp4vdepay ! avdec_mpeg4 ! videoconvert ! xvimagesink
```
rtpmanagerbad: add pkg-config
rtpmanagerbad: Rtp should be uppercase
rtpmanagerbad: add G_OS_WIN32 for shielding unix headers
rtpmanagerbad: remove Since from documentation
rtpmanagerbad: rename lib name from nrtp to rtpmanagerbad
rtpmanagerbad: sync meson.build with other modules
rtpmanagerbad: add Makefile.am
rtpmanagerbad: use GstElement to count pads
rtpmanagerbad: use gst_bin_set_suppressed_flags
rtpmanagerbad: check element creation
rtpmanagerbad: post message when trying to access missing rtpbin
rtpmanagerbad: return FALSE with g_return tests
rtpmanagerbad: use gsocket multicast check
rtpmanagerbad: use gst_caps_new_empty_simple iso gst_caps_from_string
rtpmanagerbad: sync with gstrtppayloads.h
rtpmanagerbad: correct media type X-GST
rtpmanagerbad: test if a compatible pad was found
rtpmanagerbad: remove evil copy of GstRTPPayloadInfo
rtpmanagerbad: add gio_dep to meson
rtpmanagerbad: revert to old glib boilerplate
GStreamer 1.16 does not yet support the newer GLib templates, so revert.
rtpmanagerbad: return GST_STATE_CHANGE_NO_PREROLL for live sources
for live sources, NO_PREROLL should be returned for PLAYING->PAUSED and
READY->PAUSED transitions.
rtpmanagerbad: use GstElement pad counting
rtpmanagerbad: just use template name to request pad
rtpmanagerbad: remove commented code
rtpmanagerbad: use funnel to send multiple streams on one socket
rtpmanagerbad: avoid beaches
beaches should only be used during the summer, so rewrite the code to
return explicitly and avoid beaches during the winter.
rtpmanagerbad: add copyright to test code
rtpmanagerbad: g_free is NULL safe
rtpmanagerbad: do not trace rtpbin
rtpmanagerbad: return NULL explitly
rtpmanagerbad: warn when data port is not even
According to RFC 3550, RTP data should be sent on even ports, while RTCP
is sent on the following odd port.
rtpmanagerbad: document port allocation in rtpsink/src
rtpmanagerbad: improve uri description
rtpmanagerbad: add comment re-use socket
rtpmanagerbad: rename gst_object_set_properties_from_uri_query
rtpmanagerbad: loan prop/val setter from rist
rtpmanagerbad: rtpsrc: fix unitialised pointer
rtpmanagerbad: fix silly typo
rtpmanagerbad: test for empty key/value
rtpmanagerbad: rtpsrc: deprecate ssrc collision to INFO
rtpmanagerbad: sync debug with rist
rtpmanagerbad: small strings allocated on stack
rtpmanagerbad: correct rename
rtpmanagerbad: add locking on prop setters/getters
Locking is added because the URI allows to access the properties too.
rtpmanagerbad: allow for RTCP through NAT
rtpmanagerbad: move gio to header file
rtpmanagerbad: free small strings too
rtpmanagerbad: ttl_mc for ttl on dynudpsink
rtpmanagerbad: add comments on the URI registered
rtpmanagerbad: correct macro after file rename
rtpmanagerbad: code style
rtpmanagerbad: handle wrong URIs in setter
rtpmanagerbad: nit URI notation correction
In an URI, the first key/value pair should not have an ampersand, the
parser did not die though.
As sections can be provided by the user through send_event
when the element state is NULL, their lifetime is expected
to match that of the muxer, and they must be preserved when
the state changes
We can have multiple TsMuxPacketInfo objects for the same PID
with user-provided sections, for example ATSC requires multiple
tables with the same PID.
This might be necessary temporarily for changing the previous settings.
Make it an actual error if the settings are like this while processing a
buffer.
It is parsing frame data and so should check the data size against the
frame header size instead of the file header size. If don't, it is
possible to drop the last frame because IVF_FILE_HEADER_SIZE is greater
than IVF_FRAME_HEADER_SIZE
This patchs add support for configuring the bonding method used. There is
two method specified
- redundant: All the RTP packets are replicated
- combined: RTP packet are evenly distributed over each links
Additionally, an application can set the "dispatcher" property in order
to implement custom dispatching method. Whenever the "dispatcher"
property is set, "bonding-method" property will be ignored.
As we can now have multiple sessions, stats need to be implemented per
session. This follow RTPSession model with sources. The stats are now:
dropped: 0
received: 0
recovered: 0
permanently-lost: 0
duplicates: 0
retransmission-requests-sent: 0
rtx-roundtrip-time: 0
session-stats:
session-id=0
rtp-from=""
rtcp-from=""
dropped=0
received=0
session-id=1
rtp-from=""
rtcp-from=""
dropped=0
received=0
. . .
session-stats is a GValueArray as there is no better alternatives.
As we can now have multiple sessions, stats need to be implemented per
session. This follow RTPSession model with sources. The stats are now:
sent-original-packets: 0
sent-retransmitted-packets: 0
session-stats:
session-id=0
sent-original-packets=0
sent-retransmitted-packets=0
round-trip-time=0
session-id=1
sent-original-packets=0
sent-retransmitted-packets=0
round-trip-time=0
. . .
session-stats is a GValueArray as there is no better alternatives.
gstmpegtsmux.c:291:3: error: implicit declaration of function ‘memmove’ [-Werror=implicit-function-declaration]
memmove (map.data + 4, map.data, map.size - 4);
^
gstmpegtsmux.c:291:3: error: incompatible implicit declaration of built-in function ‘memmove’ [-Werror]
gstmpegtsmux.c:291:3: note: include ‘<string.h>’ or provide a declaration of ‘memmove’
... and set to caps if necessary.
Note 1) the mastering display info and content light level SEI meessages
are persistent in the corresponding codec video sequence (i.e., GOP).
So any bitstream containing those SEI messages
(and also all pictures are intended to be HDR rendered) should be ensured that
each first slice of codec video sequence follows those SEI messages.
Note 2) The codec video sequence is a group an [IRAP + NoRaslOutputFlag == 1]
and following AUs which are not [IRAP + NoRaslOutputFlag == 1]
The NoRaslOutputFlag is equal to 1 for each IDR AU, BLA AU and some CRA AU.
For a CRA AU to have NoRaslOutputFlag equal to 1, following condition should required.
* When the CRA AU is the first AU in the bitstream in decoding order
* or the CRA AU is the first AU that follows an end of sequence NAL in decoding order
* or the HandleCraAsBlaFlag equal to 1.
Due to the limited context in parse element, in this commint, CRA AU will not considered as
having the NoRaslOutputFlag equal to 1. Therefore, in the worst case,
mastering-display-info and content-light-level could be cleared one GOP after
when stream was chagned from HDR to SDR.
RIST TR-06-1 is a specification for video streaming made by the VSF
group. It is using a subset of RTP specification to which some
modification has been made to improve RTX behaviour and avoid any need
for signaling. The plugin implement ristrtxsend / ristrtxreceive element
which are the RIST specific equivalent of rtprtxsend/rtprtxreceive and
ristsink / ristsrc which implement rist transmitter and receiver. The
RIST protocol is meant to be used in unidirectional way. Typically, MPEG
TS over RTP is used.
Currently we support unicast and multicast streaming according to the
specification. This patch does not include any bonding support yet. The
ristsrc element introduce rist:// URI handling in parallel to it's
property configuration interface.
Expose SEI data in the H.264 bitstream parser API and
extract closed captions and other things that are not
specified in the H.264 spec itself in the videoparser.
Based on patch by: Mathieu Duponchelle <mathieu@centricular.com>
https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/issues/940
