alignment works like in mpegtsmux, joining several MpegTS packets into
one buffer. Default value of 0 joins as many as possible for each
incoming buffer, to optimise CPU usage.
If we have no DTS but a PTS then this means both are the same, and we
should update the last_ts with the PTS. Only if both are unknown then we
don't know the current position and should not update it at all.
Previously we would always update the last_ts to GST_CLOCK_TIME_NONE if
the DTS is unknown, which caused the position to jump around and to
cause spurious gap events to be sent.
Instead of doing it on each packet and doing it based on the distance to
the previous SCR instead of based on the DTS.
Previously we would send gap events for audio all the time if the SCR
distance was 400ms because the threshold for audio is 300ms and by only
ever updating the position when the SCR updates we would always be 100ms
above the threshold and send needless gap events.
This fixes audio glitches on various files caused by gap events.
Some raw h264 encoded files trigger the assignment of wrong PTS to buffers
when some SEI data is provided. This change prevents it to happen.
Also ensure this behavior is being tested.
We might have some old timecodes that are in the future now and have to
drop those to make sure that our queue is correctly ordered and we don't
have multiple timecodes for the same running time.
Directly read them out of the decoder as soon as we passed audio and
then store them in a queue that we handle internally together with their
timestamps. This cleans up memory management and gives us proper control
over the queue instead of guessing how the queue inside the LTC decoder
actually works and when it overflows.
And also introduce 6 instead of 2 frames of latency compared to the LTC
audio input as that seems to be an upper bound for how much the LTC
library is lagging behind.
As the H265/H264 bitstream can support multiple slices,
mastering_display_info_state and content_light_level_state
should be changed only on first slice segment.
Fix#1152
... 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
when computing timecode metas. Depending on the value of that flag,
n_frames is to be interpreted as a number of fields or a number of
frames. As GstVideoTimeCodeMeta always deals with frames, we want
to scale that number when needed.
This debug code will help determine why certain instances of closed
captions that are present in the Picture User Data are not actually
processed by the pipeline
In 7c767f3fcd , stream creation was
refactored to occur before potential program creation. This created
issues with pipelines such as:
gst-launch-1.0 videotestsrc ! video/x-raw, format=I420, width=640, height=640, framerate=25/1 ! \
x264enc ! hlssink2 target-duration=1
eg.: gst_buffer_copy_into: assertion 'bufsize >= offset + size' failed
As this reordering was actually not needed for the purpose of allowing
to specify a PCR stream, this reverts the reordering part of the
initial commit.
The MPEG-TS packetiser should use the upstream DTS for
skew correction when running in that mode, as the DTS
carries the upstream arrival time. The PTS (if it's
set at all) is less useful, and can be invalid.
gstladspa.c:360:5: error: zero-length ms_printf format string [-Werror=format-zero-length]
vad_private.c:108:3: error: this decimal constant is unsigned only in ISO C90 [-Werror]
gstdecklinkvideosink.cpp:478:32: error: comparison between 'BMDTimecodeFormat {aka enum _BMDTimecodeFormat}' and 'enum GstDecklinkTimecodeFormat' [-Werror=enum-compare]
win/DeckLinkAPI_i.c:72:8: error: extra tokens at end of #endif directive [-Werror]
win/DeckLinkAPIDispatch.cpp:35:10: error: unused variable 'res' [-Werror=unused-variable]
gstwasapiutil.c:733:3: error: format '%x' expects argument of type 'unsigned int', but argument 8 has type 'DWORD' [-Werror=format]
gstwasapiutil.c:733:3: error: format '%x' expects argument of type 'unsigned int', but argument 9 has type 'guint64' [-Werror=format]
kshelpers.c:446:3: error: missing braces around initializer [-Werror=missing-braces]
kshelpers.c:446:3: error: (near initialization for 'known_property_sets[0].guid.Data4') [-Werror=missing-braces]
The way FlowCombiner combines the FLUSH doesn't work in the case
we have several "sinkpads" since any flush return FLUSH. But in the
case we have a seek where on one branch flush is done, we should
just say OK otherwise we might return FLUSHING to a src that has already
been seeked and is ready to process new buffers
/usr/bin/ld: .libs/libgstremovesilence_la-vad_private.o: in function `vad_set_threshold':
./gst/removesilence/vad_private.c:108: undefined reference to `pow'
/usr/bin/ld: .libs/libgstremovesilence_la-vad_private.o: in function `vad_get_threshold_as_db':
./gst/removesilence/vad_private.c:114: undefined reference to `log10'
vps/sps/pps in codec_data shouldn't be considered as inband data.
Otherwise, h26{4,5}parse never insert them to nal when transform
(packetized to byte-stream) use case
Similar change as the on I did in h264parse. We want to process SEI
recovery point as keyframe so muxers will mark them as seek points and
decoders will be able to start decoding from them rather than waiting
for an IDR.
https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/issues/790
The spec states that "recovery point SEI message assists a decoder in
determining when the decoding process will produce acceptable
pictures for display after the decoder initiates random access or after the
encoder indicates a broken link in the coded video sequence."
Mark those as keyframes so muxers will mark them as seek points and
decoders will be able to start decoding from them rather than waiting
for an IDR.
https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/issues/790
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
If the first audio buffer to be dropped started right between two video
buffers (after the end of the first but before the start of the second,
as is often the case with N/1001 video frame rates), we would miss
sending the dropping=true message.
https://bugzilla.gnome.org/show_bug.cgi?id=797248
tsdemux expects a custom descriptor (GST_MTS_DESC_AC3_AUDIO_STREAM)
to detect a stream as AC3 and not EAC3.
Note that tsdemux expects this descriptor because mpegtsmux writes
a stream with a HDMV registration descriptor.
Fixes:
gst-launch-1.0 -v audiotestsrc ! avenc_ac3 ! ac3parse ! mpegtsmux ! \
tsdemux ! ac3parse ! avdec_ac3 ! audioconvert ! autoaudiosink
https://bugzilla.gnome.org/show_bug.cgi?id=797220
Previously it was dispatched before the last video buffer, and audio
buffers would follow afterwards. It's misleading to send the
dropping=true message before both streams have really stopped, it can
lead to races when someone is e.g. waiting for that message to send EOS.
Also added some debug output.
https://bugzilla.gnome.org/show_bug.cgi?id=797145
Direct applying the commit 7bb6443. This could fix also unexpected
nal dropping when nonzero "config-interval" is set.
(e.g., gst-launch-1.0 videotestsrc ! x265enc key-int-max=30 !
h265parse config-interval=30 ! avdec_h265 ! videoconvert ! autovideosink)
Similar to the h264parse, have_{vps,sps,pps} variables will be used
for deciding on when to submit updated caps or not, and rather mean
"have new SPS/PPS to be submitted?"
See also https://bugzilla.gnome.org/show_bug.cgi?id=732203https://bugzilla.gnome.org/show_bug.cgi?id=754124
If we drain after a discont, the discont time given by the stream
synchronizer is already the time after the discontinuity. But we need to
drain all pending data based on the previous discont time instead.
The case is properly handled a few lines below by dropping the buffer.
We shouldn't perpetually block the audio chain function until the
target-timecode is reached.
https://bugzilla.gnome.org/show_bug.cgi?id=796906
This change allow setting timestamp on streams that would otherwise have
no timestamp. This is useful to make a video from bunch of JPEG files. An
example of such pipeline would be:
gst-launch-1.0 multifilesrc location=%05d.jpeg caps=image/jpeg,framerate=30/1 \
! jpegparse ! fakesink silent=0 -v
It works like a valve in front of the actual avwait. When recording ==
TRUE, other rules are then examined. When recording == FALSE, nothing is
passing through.
https://bugzilla.gnome.org/show_bug.cgi?id=796836
255 will easily become 0 in the blending function as they expect
the maximum value to be 255.
Can be reproduce with
gst-launch-1.0 videotestsrc pattern=ball ! c.sink_0 \
videotestsrc pattern=snow ! c.sink_1 \
compositor name=c \
sink_0::zorder=0 sink_1::zorder=1 sink_0::crossfade-ratio=0.5 \
background=black ! \
videoconvert ! xvimagesink
crossfade-ratio +/- 0.001 makes it work correctly and the same happens
at e.g. 0.25, 0.75, N*0.0625
https://bugzilla.gnome.org/show_bug.cgi?id=796846
Adds AV01 FOURCC to the list of allowed media files, in order to allow
parsing the IVF Container holding AV1 content.
At a later point dynamic resolution change can be supported - therefore
the sequence header OBU and frame header OBU of AV1 file must be parsed,
which can be done in future with the help of gst-lib gstav1parse.
https://bugzilla.gnome.org/show_bug.cgi?id=796677
This moves all the conversion related code to a single place, allows
less code-duplication inside compositor and makes the glmixer code less
awkward. It's also the same pattern as used by GstAudioAggregator.
The aggregated_frame is now called prepared_frame and passed to the
prepare_frame and cleanup_frame virtual methods directly. For the
currently queued buffer there is a method on the video aggregator pad
now.
Unless we only have sparse streams. In this case we will consider them.
It fixes a bug happening when first observed timestamp comes from a
sparse stream and other streams don't have a valid timestamp, yet. Thus
leading the timestamp from sparse stream to be the start of the
following segment. In this case, if the timestamp is really bigger than
non-sparse stream (audio/video), it will lead the pipeline to clip
samples from the non-parse stream.
https://bugzilla.gnome.org/show_bug.cgi?id=744469
Scene detection determines, how many scenes have changed in a video.
It compared the previous frame with present frame to find out the score and a
threshold is calculated for the same.
I have added an intermediate condition which helps in improving the positive
detections.
https://bugzilla.gnome.org/show_bug.cgi?id=735094
We were assuming that NULL pool meant that downstream didn't reply.
Update the pool index 0 instead of adding at the end. Otherwise we ended
up letting basesrc decide, which would pick the blocksize as a size
(4096) instead of the image size.
https://bugzilla.gnome.org/show_bug.cgi?id=795327
This server uses an unknown 003.889 protocol version. This patch fixes
the version validation in order to simply fallback to 3.3 as suggested
by the spec.
We would mark all streams with FLAG_UNSELECT as we would check
the pointer for non-NULLness not the dereferenced stream number
(and the pointer is always non-NULL). The intention here was
presumably to mark the first stream of each type as SELECT and
the others as UNSELECT by default.
CID 1434970.
This is a simple Bin that will expose audiotestsrc or videotestsrc
based on what is asked by the user either through the GstURIHandler
API or through the "stream-types" property.
This element also provides GstStream and GstStreamCollection
so it is nicely usable from playbin3.
https://bugzilla.gnome.org/show_bug.cgi?id=795366
pcapparse cannot parse fragmented IP packets correctly, in particular it
will get confused when trying to parsing fragments as standalone frames
in two ways:
1. the first fragment will have the packet length greater than the
frame size and will always be discarded;
2. fragments with non-zero offsets will be interpreted as full packets
and the first part of their raw payload data will be parsed as the
transport protocol header, resulting in bogus values for addresses
and ports, thus evading the properties filtering on those values.
This can make it difficult for users to see why the data does not get
downstream.
So be more explicit and just bail out when fragmented packets are
encountered.
https://bugzilla.gnome.org/show_bug.cgi?id=795284
If the 'enable-last-sample' property is enabled, fakevideosink will keep
a reference on last rendered buffer which may lead to buffer starvation
in the pipeline.
Request one extra buffer in this case so we always have a buffer flying
in the pipeline.
https://bugzilla.gnome.org/show_bug.cgi?id=795109
When starting up we need to initialise things *before*
streaming starts, so before we chain up to the parent
class in the state change function. And when we shut
down the element, we need to reset things after streaming
has stopped, so after we chain up to the parent class
in the state change function.
Possibly related to memory leak in:
https://bugzilla.gnome.org/show_bug.cgi?id=794353
