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
We used to have the same enum to represent H265 profiles and idc values.
Those are no longer the same with extension profiles defined from
version 2 of the spec.
Split those enums so the semantic of each is clearer and we'll be able
to add extension profiles to GstH265Profile.
Also add gst_h265_profile_tier_level_get_profile() to retrieve the
GstH265Profile from the GstH265ProfileTierLevel. It will be used to
implement the detection of extension profiles.
https://bugzilla.gnome.org/show_bug.cgi?id=793876
Measures the audio latency between the source pad and the sink pad by
outputting period ticks on the source pad and measuring how long they
take to arrive on the sink pad.
Very useful for quantifying latency improvements in audio pipelines.
This plugin was particularly useful during development of the
low-latency features of the wasapi plugin.
https://bugzilla.gnome.org/show_bug.cgi?id=793839
This is a wrapper around fakesink that will advertise GstVideoMeta
and other meta API in order to achieve zero-copy whenever possible.
his new element is useful when doing performance testing with
video stream and don't want the sink capability to change the
upstream behaviour.
https://bugzilla.gnome.org/show_bug.cgi?id=793624
The pnmenc was not mapping the input buffers as video buffers. Because
of this, the video frame stride was not being set based on frame but
based on the caps, which make the assumption that the strides are a
power of 4. For input that is not a power of 4, this would lead to a
SIGSEGV.
https://bugzilla.gnome.org/show_bug.cgi?id=793419
The inter plugin originated in 0.10, which had only one timestamp. As a
result, during the port to 1.0, the DTS were left undefined. This can cause
subtle bugs with basesrc, which can end up incorrectly picking DTS over PTS
and producing output buffers with incorrect timestamps.
https://bugzilla.gnome.org/show_bug.cgi?id=791347
This keep-it-simple plugin is useful when you want to pipe arbitrary
data to a different pipeline within the same process. Some advantages
over appsink/appsrc, the inter elements, etc:
* Ease of use. Buffers, events, and caps are transmitted as-is without
copying or serialization.
* Enables zerocopy (especially DMABUF) transparently without any
special-casing.
* Enables usage with sinks or elements that are unreliable and may
throw errors and need re-initialization, such as a network sink, a
USB device sink (v4l2), etc.
* Transmits arbitrary data, not just audio/video/subs
* Can easily implement 1 producer pipeline -> N dynamic consumer
pipelines within a single process when combined with the `tee`
element.
All queries, events, buffers, and buffer lists are proxied. State
changes, clocks, and base times for the two pipelines are independent
since the upstream and downstreams continue to be different pipelines.
https://bugzilla.gnome.org/show_bug.cgi?id=788200
gdpdepay element uses the decide_allocation to fetch the downstream
allocator. Nonetheless it is possible that allocate uses a custom
alloc function, which is not usable by gdpdepay, crashing later the
application when the allocater buffer is NULL.
This patch checks for the allocator flags and reset it if the
allocator has a custom alloc function.
https://bugzilla.gnome.org/show_bug.cgi?id=789476
When querying downstream for allocation, and the source caps hasn't
set its caps, using ANY by default, it raises a critical message in
console:
CRITICAL **: gst_video_info_from_caps: assertion 'gst_caps_is_fixed (caps)' failed
This patch bails out decide_allocation() if the caps aren't fixed.
https://bugzilla.gnome.org/show_bug.cgi?id=789476
This information could be used for example to pick a decoder supporting
a specific chroma and/or bit depth, like 4:2:2 10 bits.
It can also be used to inform earlier decoder about the format it is
about to decode.
https://bugzilla.gnome.org/show_bug.cgi?id=792039
This fixes issues where wavparse would query the file size upstream
and assert because the file size is way smaller then what the WAVE
header says. This patch disable or cane a handful of queries that
make no sense to forward.
https://bugzilla.gnome.org/show_bug.cgi?id=791811
This plugin is useful when you want to pipe arbitrary data to
a different pipeline within the same process. Buffers, events, and caps
are transmitted as-is without copying or manipulation.
"avwait-status" is posted when avwait starts or stops passing through
data (e.g. because target-timecode and end-timecode respectively have
been reached). The attached structure includes a "dropping" boolean (set
to TRUE if we are currently dropping data, FALSE otherwise), and a
"running-time" GST_CLOCK_TIME which contains the running time of the
change.
https://bugzilla.gnome.org/show_bug.cgi?id=790170
Reordering of packets is not very common in networks, and the delay
functions will always introduce reordering if delay > packet-spacing,
so by setting allow-reordering to FALSE you guarantee that the packets
are in order, while at the same time introducing delay/jitter to them.
