Free the existing descriptor array, if any, before replacing it.
Fix leaks with the
validate.file.playback.scrub_forward_seeking.test-mpeg2-mp3_mxf scenario.
https://bugzilla.gnome.org/show_bug.cgi?id=748580
If the stream which is about to be removed still has a ref on a tag list we
should drop it.
Fix a leak which was occasionally happening with the
validate.file.playback.change_state_intensive.tron_en_ge_aac_h264_ts scenario.
https://bugzilla.gnome.org/show_bug.cgi?id=748576
Replace videocrop ! videoscale ! capsfilter with the digitalzoom
bin that has the same pipeline internally and already updates
the capsfilter automatically when caps change, removing this code
from wrappercamerabinsrc and making it cleaner.
Avoids one extra uneeded renegotiation if the elements are already
configured to their final property values when the caps event
goes through.
Also avoids hitting bug https://bugzilla.gnome.org/show_bug.cgi?id=748344
It contains videocrop ! videoscale ! capsfilter and implements digital
zooming.
At this moment, it is a private element of the camerabin plugin.
This will remove some code used in wrappercamerabinsrc to make
code clearer and digitalzoom can potentially be used by other
applications in the future, it has nothing camerabin specific.
wrappercamerabinsrc has a videocrop element to be used for
zooming and for cropping when input caps is different when used
with the GstPhotography interface. The zooming part needs
the following elements:
capsfilter ! videocrop ! videoscale ! capsfilter
The capsfilters should always have the same caps to ensure the
zooming is done and preserves dimensions, unless when it is needed
to do more cropping due to input dimensions those caps
need to be modified accordingly to preserve the output dimensions.
This, however, makes it hard to get caps negotiation to work properly
as we need to have different caps in the capsfilters to account for
the extra cropping needed. It could be simple for fixed caps but it
gets tricky with unfixed ones.
To solve this, this patch splits the zooming and dimension reduction
cropping into 2 separate videocrop elements. The first one does
the dimension cropping, which is only needed when the GstPhotography
API is used and the source provides a caps that is different than
what is requested, while the second is dedicated to zoom crop only.
The first part of the pipeline goes from:
src ! videoconvert ! capsfilter ! videocrop ! videoscale ! capsfilter
to
src ! videocrop ! videoconvert ! capsfilter ! videocrop ! videoscale ! capsfilter
It might add an extra overhead in the image capture as the image might need
to be cropped twice but this can be solved by enabling videocrop to use
crop metas so only the later one does the real cropping.
It also makes the code a bit simpler.
Remove tee and output-selector and just link the source
pad to the outputs we want as needed.
The way we need to prioritize caps negotiation and allocation
queries depending on the mode enabled is too custom to be
handled using tee and output-selector.
This provides more flexibility and doesn't get in the way of proper
handling of negotiation and allocation queries.
The detection for missing format/alignment is done way before this
codepath is reached (at which point we have already decided of a
format and alignment).
CID #1232800
When block width property is set to 0, exception occurs.
This happens due to divide by zero errors in calculations.
block width property can never be 0. Hence adjusting the minimum value to 1.
https://bugzilla.gnome.org/show_bug.cgi?id=744188
Such seeks are used to change playback rate and we do not want
to alter the position in that case, so we bypass the flush/seek
logic, and set things up so a new segment is scheduled to be
regenerated.
https://bugzilla.gnome.org/show_bug.cgi?id=735100
This will happen when the PMT changes, replacing streams with
new ones. In that case, we need to accumulate the running time
from the previous chain in the segment base.
https://bugzilla.gnome.org/show_bug.cgi?id=745102
Reset the internal segment before freeing it.
mxf_index_table_segment_parse() allocates data inside the segment
(like segment->delta_entries) which have to be freed using
mxf_index_table_segment_reset().
https://bugzilla.gnome.org/show_bug.cgi?id=746803
Also:
- Don't modify size on early buffer
The size is the size of the buffer, not of remaining part.
- Use the input caps when manipulating the input buffer
Also store in in the sink pad
- Reply to the position query in bytes too
- Put GAP flag on output if all inputs are GAP data
- Only try to clip buffer if the incoming segment is in time or samples
- Use incoming segment with incoming timestamp
Handle non-time segments and NONE timestamps
- Don't reset the position when pushing out new caps
- Make a number of member variables private
- Correctly handle case where no pad has a buffer
If none of the pads have buffers that can be handled, don't claim to be EOS.
- Ensure proper locking
- Only support time segments
https://bugzilla.gnome.org/show_bug.cgi?id=740236
When the timeout is reached, only ignore pads with no buffers, iterate
over the other pads until all buffers have been read. This is important
in the cases where the input buffers are smaller than the output buffer.
https://bugzilla.gnome.org/show_bug.cgi?id=745768
Correctly calculate alpha in a few places by dividing by 255,
not 256.
Fix the argb and bgra blending functions to avoid an off-by-one
error in the calculations, so painting with alpha = 0xff doesn't
ever bleed through from behind
Currently the alignment property just makes sure that we
output things in multiples of align*packet_size bytes, but
with no clear maximum size. When streaming MPEG-TS over
UDP one wants buffers with a maximum packet size of 1316.
The alignment property so far would just output buffers
that are a multiple of 1316 then.
Instead we now make the alignment property output
individual buffers with the alignment size, which
is entirely backwards compatible with the expected
behaviour up until now. For efficiency reason
collect all those buffers in a buffer list and
send that downstream.
Also collect data to push downstream in a buffer
list from the adapter if we don't align things,
which is still more efficient because of the
silly way the muxer currently creates output
packets.
https://bugzilla.gnome.org/show_bug.cgi?id=722129
Actually accumulate the sample counter to check the accumulated error
between actual timestamps and expected ones instead of just resetting
the error back to 0 with every new buffer.
Also don't reset discont_time whenever we don't resync. The whole point of
discont_time is to remember when we first detected a discont until we actually
act on it a bit later if the discont stayed around for discont_wait time.
https://bugzilla.gnome.org/show_bug.cgi?id=746032
This allows us to handle new segment events correctly; either by dropping
buffers or inserting silence; for example if the offset is changed on an srcpad
connected to audiomixer.
If the video source happens to allow max-zoom to be greater than our maximum hard coded
value of 10 then the user cannot set anything greater than our maximum specified in the
param spec. We have to update our param spec to prevent glib from capping the value
https://bugzilla.gnome.org/show_bug.cgi?id=745740
This reverts commit d387cf67df.
The analysis was wrong: The first 20ms of latency are introduced by the source
already and put into the latency query, making it only necessary to cover the
additional 20ms of audiomixer inside audiomixer.
This prevents it from going into passthrough after receiving 2
byte-stream caps (different ones) as it would keep the have_pps and
have_sps set to true and would just go into passthrough without
updating its caps.
This patch makes it reset its stream information to restart properly
when new caps are received.
https://bugzilla.gnome.org/show_bug.cgi?id=745409
To avoid useless renegotiation of the pipe we can check for
negotiated caps on src_filter and compare it with requested
filter. If the caps intersect, avoid restart.
Signed-off-by: Oleksij Rempel <bug-track@fisher-privat.net>
https://bugzilla.gnome.org/show_bug.cgi?id=672610
Let's assume a source that outputs outputs 20ms buffers, and audiomixer having
a 20ms output buffer duration. However timestamps don't align perfectly, the
source buffers are offsetted by 5ms.
For our ASCII art picture, each letter is 5ms, each pipe is the start of a
20ms buffer. So what happens is the following:
0 20 40 60
OOOOOOOOOOOOOOOO
| | | |
5 25 45 65
IIIIIIIIIIIIIIII
| | | |
This means that the second output buffer (20 to 40ms) only gets its last 5ms
at time 45ms (the timestamp of the next buffer is the time when the buffer
arrives). But if we only have a latency of 20ms, we would wait until 40ms
to generate the output buffer and miss the last 5ms of the input buffer.
The two branches of the if conditional are identical, which means in all cases
the same gst_asf_put_guid() will be executed. Do it directly.
CID #1226448
the calculations for detecting the videomark is being repeated
in for loop unnecessarily. Moving this outside of for loop
such that the code need not be executed evertime the loop is executed.
https://bugzilla.gnome.org/show_bug.cgi?id=744778
Value stored in ret will be ovewritten in the next iteration of the loop. Which
means it is never used.
Plus a style issue to make gst-indent happy and allow the commit.
Don't use private GMutex implementation details to check
whether it has been freed already or not. Just turn dispose
function into finalize function which will only be called
once, that way we can just clear the mutex unconditionally.
the calculations for drawing the videomark is being repeated
in for loop unnecessarily. Moving this outside of for loop
such that the code need not be executed evertime the loop is executed.
https://bugzilla.gnome.org/show_bug.cgi?id=744371
Always update the segment and not only for accurate seeking and always
send a new segment event after seeks.
For non-accurate force a reset of our segment info to start from
where our seek led us as we don't need to be accurate
https://bugzilla.gnome.org/show_bug.cgi?id=743363
Detect invisible pixels, and skip gstspu_vobsub_blend_comp_buffers()
when there are only invisible pixels. This significantly reduces the
CPU load in cases of DVDs which don't use the clip_rect to exclude
processing for parts of the screen where the video is visible.
https://bugzilla.gnome.org/show_bug.cgi?id=667221
There's no reason why audiomixer should override the segment
base of upstream with whatever value it got from a SEEK event,
or even worse... with 0 if there was no SEEK event yet. This
broke synchronization if upstream provided a segment base other
than 0, e.g. when using pad offsets.
Also that this code did things conditional on the element's state
should've been a big warning already that something is just wrong.
If this breaks anything else now, let's fix it properly :)
Also don't do fancy segment position trickery when receiving a
segment event. It's just not correct.
The flush is called on discont and we shouldn't output a new segment
each time a discont happens. So this commit remove the mark for a new
segment when flushing streams by propagating the 'hard' flag passed
on the flusing from the base class.
https://bugzilla.gnome.org/show_bug.cgi?id=743363
Instead of using the GST_OBJECT_LOCK we should have
a dedicated mutex for the pad as it is also associated
with the mutex on the EVENT_MUTEX on which we wait
in the _chain function of the pad.
The GstAggregatorPad.segment is still protected with the
GST_OBJECT_LOCK.
Remove the gst_aggregator_pad_peak_unlocked method as it does not make
sense anymore with a private lock.
https://bugzilla.gnome.org/show_bug.cgi?id=742684
Reduce the number of locks simplify code, what is protects
is exposed, but the lock was not.
Also means adding an _unlocked version of gst_aggregator_pad_steal_buffer().
https://bugzilla.gnome.org/show_bug.cgi?id=742684
Assignment is done to variable segment.stop when the intention was to assign to
local variable stop. Instead of overwriting it, the value is now clamped and
segment.stop is set to it soon after.
CID #1265772
AIFF chunks are supposed to be even aligned.
Aligning the SSND chunk will allow the aiff muxer to properly write
chunks (like the ID3 one) at the end of the file.
https://bugzilla.gnome.org/show_bug.cgi?id=727402
This did not actually work since the video_buffer was set to NULL after
the first black frame.
Reported by: Patrik Oldsberg <patrik.oldsberg@ericsson.com>
This patch calls gst_h264_parser_parse_subset_sps() when a
SPS subset NAL type is found.
All the bits required for parsing the SPS subset in NALs were
already there, just we need to call them when the this NAL type
is found.
With this parsing, the number of views (minus 1) attribute is
filled, which was a requirement for negotiating the stereo-high
profile.
https://bugzilla.gnome.org/show_bug.cgi?id=743174
Initial support for MVC NAL units. It is only needed to propagate the
complete set of NAL units downstream at this time.
https://bugzilla.gnome.org/show_bug.cgi?id=696135
Signed-off-by: Sreerenj Balachandran <sreerenj.balachandran@intel.com>
Signed-off-by: Gwenole Beauchesne <gwenole.beauchesne@intel.com>
klass->setup (scope) will always return TRUE since all children of this class
do so, no need to store the return. Besides, the value is overwritten a few
lines down before it is used.
Change helps keep files in sync after:
-base commit a91d521a36
ret is assigned but not used and in the next cycle of the loop it is overwritten
with default_prepare_output_buffer (). If there is a flow error the function
should return instead.
CID #1226475
Value from left_luminance is assigned to out_luminance here, but that stored
value is not used before it is overwritten in the next cycle of the loop.
Removing assignation.
CID #1226473
found is initialized to FALSE to then only be used in two conditional statements
that will always be false, making the blocks inside them dead code. Looking back
in the file's history the setting of the variable's value before it is checked
was dropped as part of the port to 0.11, bringing that value setting back.
https://bugzilla.gnome.org/show_bug.cgi?id=742638
CLAMP checks both if value is '< 0' and '> max'. Value will never be a negative
number since it is an unsigned integer. Removing that check and only checking if
it is bigger than max and setting it appropriately.
Also converting the previous instance of this into MIN() for consistency.
CID 1139793
Some video bitstreams report a too restrictive set of profiles. If a video
decoder was to strictly follow the indicated profile, it wouldn't support that
stream, whereas it could in theory and in practice. So we should relax the
profile restriction for allowing the decoder to get connected with parser.
https://bugzilla.gnome.org/show_bug.cgi?id=739992
CLAMP checks both if y is '< 0' and '> h1'. y will never be a negative number
since it is an unsigned integer. Removing that check and only checking if it
bigger than h1 and setting it to that max approprietaly.
CID 1139792
type is truncated to 0-31 with "& 0x1f", but right after that it is checks if
the value is equivalent to GST_H265_NAL_VPS, GST_H265_NAL_SPS, and
GST_H265_NAL_PPS (which are 32, 33, and 34 respectively). Obviously, this will
never be True if the value is maximum 31 after the truncation.
The intention of the code was to truncate to 0-63.
After further investigation the previous commit is wrong. The code intended to
check if the type is 39 or the ranges 41-44 and 48-55. Just like gsth265parse.c
does. Type 40 would not be complete.
nal_type is the index for a GstH265NalUnitType enum. There are two types of dead
code here:
First, after checking if nal_type is >= 39 there are two OR conditionals that
check if the value is in ranges higher than that number, so if nal_type >= 39
falls in the True branch those other conditions aren't checked and if it falls
in the False branch and they are checked, they will always also be False. They
are redundant.
Second, the enum has a range of 0 to 40. So the checks for ranges higher than 41
should never be True.
Removing this redundant checks.
CID 1249684
Everytime a buffer is being provided from baseparse, we are parsing all the data from the beginning.
But since we would have already parsed some of the data in the previous iterations,
it doesnt make much sense to keep parsing the same everytime.
Hence skipping the data which is already read in previous iterations to improve the parsing performance.
https://bugzilla.gnome.org/show_bug.cgi?id=740058
The image capture mutex and the pad object lock would cause a race
if the pad query was made right when the image probe was running.
The image probe needs the capture mutex and the querying would need
the pad object lock.
It might be racy with the image probe thread as it uses the capture
mutex just like the start-capture handler from camerabin. The
start-capture would be waiting for the source's streaming thread
to stop to be able to set the source state to ready while the
probe would be blocked waiting to acquire the capture mutex.
It causes a deadlock.
Don't rely on core implementation details, which are private and
may change. It's also not needed here, the performance impact is
close to none. Also copy buffer before changing its metadata.
Get rid of some indirections and inefficiencies,
just payload things directly which gives us more
control over what memory is allocated where and
how and makes things much simpler. In particular,
we can now allocate the payload header plus the
GstMemory to represent it in one go.
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.
Add function to calculate a payload CRC across multiple memories
so we don't have to merge buffers with multiple memories just to
calculate the CRC. Also make CRC calculation function static,
since it's not used outside dataprotocol.h and move special-casing
of length = 0 -> CRC = 0 into CRC function (from caller).
Perhaps more importantly, since payload CRC is off by default:
don't map buffer (and possibly merge memories in the process)
if we are not going to use it to calculate a CRC anyway.
This can happen if this is a live pipeline and no source produced any buffer
and sent no caps until the an output buffer should've been produced according
to the latency.
When this is TRUE, we really have to produce output. This happens
in live mixing mode when we have to output something for the current
time, no matter if we have enough input or not.
Some video bitstreams report a too restrictive set of profiles. If a video
decoder was to strictly follow the indicated profile, it wouldn't support that
stream, whereas it could in theory and in practice. So we should relax the
profile restriction for allowing the decoder to get connected with parser.
https://bugzilla.gnome.org/show_bug.cgi?id=739992
When dealing with random-access content (such as files), we initially
search for the last PCR in order to figure out duration and to handle
other position estimation such as those used in seeking.
Previously, the code looking for that last PCR would search in the last
640kB of the file going forward, and stop at the first PCR encountered.
The problem with that was two-fold:
* It wouldn't really be the last PCR (it would be the first one within
those last 640kB. In case of VBR files, this would put off duration
and seek code slightly.
* It would fail on files with bitrates higher than 52Mbit/s (not common)
Instead this patch modifies that code by:
* Scanning over the last 2048kB (allows to cope with streams up to 160Mbit/s)
* Starts by the end of the file, going over chunks of 300 MPEG-TS packets
* Doesn't stop at the first PCR detected in a chunk, but instead records all
of them, and only stop searching if there was "at least" one PCR within
that chunk
This should improve duration reporting and seeking operations on VBR files
https://bugzilla.gnome.org/show_bug.cgi?id=708532
Sometimes rawparse does not handle the seeking query
properly, the rawparse should send the query upstream
first. For example, upstream could support seeking in
TIME format (but not in BYTE format), so the BYTE format
seeking query that rawparse sends in push mode would
fail.
https://bugzilla.gnome.org/show_bug.cgi?id=722764
Read PNG data chunk in one go by letting the parser
base class know the size we need, so that it doesn't
drip-feed us small chunks of data (causing a lot of
reallocs and memcpy in the process) until we have
everything.
Improves parsing performance of very large PNG files
(65MB) from ~13 seconds to a couple of millisecs.
https://bugzilla.gnome.org/show_bug.cgi?id=736176
This commit add an helper to convert a frame to frame-layer format and
use it to implement these two stream-format conversion:
- asf --> sequence-layer-frame-layer
- asf --> frame-layer
In simple/main profile, we basically have a raw frame, so building a
frame layer isn't too complicated. But in advanced profile, the first
frame-layer should contain sequence-header, entrypoint, and frame and
each keyframe should contain entrypoint, so we have to handle these
carefully.
https://bugzilla.gnome.org/show_bug.cgi?id=738526
Add an helper to check that output stream-format is coherent with
profile and header-format. It also check if we know how to do the
conversion if the input stream-format differs from selected
output-format.
So, in case output stream-format is not allowed, it will now fail at
negotiation rather than in pre_push_frame.
https://bugzilla.gnome.org/show_bug.cgi?id=738526
This commit introduces an helper to convert an ASF frame to BDUs format with
startcodes and use this helper to implements following stream-format
conversions:
- asf --> bdu
- asf --> sequence-layer-bdu
- asf --> sequence-layer-raw-frame
https://bugzilla.gnome.org/show_bug.cgi?id=738526
It add the support of following stream-format conversion:
- bdu --> sequence-layer-bdu
- bdu-frame --> sequence-layer-bdu-frame
- frame-layer --> sequence-layer-frame-layer
For these conversion, the only requirements is to push a sequence-layer
buffer prior to data.
https://bugzilla.gnome.org/show_bug.cgi?id=738526
It prepares the template for stream-format conversion and it implements
the following conversion:
- sequence-layer-bdu --> bdu
- sequence-layer-bdu-frame --> bdu-frame
- sequence-layer-frame-layer --> frame-layer
Work is done in the pre_push_frame() method.
https://bugzilla.gnome.org/show_bug.cgi?id=738526
gstinteraudiosrc.c: In function 'gst_inter_audio_src_create':
gstinteraudiosrc.c:339:27: error: variable 'buffer_samples' set but not used [-Werror=unused-but-set-variable]
guint64 period_samples, buffer_samples;
^
The whole not_linked optimisation is really a bit dodgy here, but
let's leave it in place for now and at least start pushing data
again when a pad got linked later, in which case we should get a
RECONFIGURE event.
Current CLAMP checks both if the value is below 0 or above 255. Considering it
is an unsigned value it can never be less than zero, so that comparison is
unnecessary. Switching to using if just for the upper bound.
CID #1139796
Value from left_luminance is assigned to out_luminance here, but that stored
value is not used before it is overwritten in the next cycle of the loop.
Removing assignation.
CID #1226473
As a consequence, tsdemux won't remove its pads anymore on EOS.
Fixes the case when mpegtsbase is not able to process new packets
after EOS as the corresponding pids aren't known anymore because
the programs were removed and the pes/psi were kept, preventing the
PAT to be parsed again.
https://bugzilla.gnome.org/show_bug.cgi?id=738695
It was using a 24000/24000/48000, but I think it meant to use
24000/32000/48000. Not 100% sure...
https://en.wikipedia.org/wiki/G.722.1 has the list of supported
bitrates. It's not clear whether the "flag" code maps to this,
however.
Coverity 206072
This parses the frame_packing_arragement() payload in SEI message.
This information can be used by decoders to appropriately rearrange the
samples which belong to Stereoscopic and Multiview High profiles.
https://bugzilla.gnome.org/show_bug.cgi?id=685215
Signed-off-by: Sreerenj Balachandran <sreerenj.balachandran@intel.com>
Signed-off-by: Gwenole Beauchesne <gwenole.beauchesne@intel.com>
Assume that small backward PCR jumps are just from upstream packet
mis-ordering and don't reset timestamp tracking state - assuming that
things will be OK again shortly.
Make the threshold for detecting discont between sequential buffers
configurable and match the smoothing-latency setting on tsparse
to better cope with data bursts.
When the set-timestamps property is set, use PCRs on the provided
(or autodetected) pcr-pid to apply (or replace) timestamps on the
output buffers, using piece-wise linear interpolation.
This allows tsparse to be used to stream an arbitrary mpeg-ts file,
or to smooth jittery reception timestamps from a network stream.
The reported latency is increased to match the smoothing latency if
necessary.
Otherwise a magic capsfilter after the source is required with
exactly the same caps as the input.
This would've failed before with invalid buffer sizes:
gst-launch-1.0 videotestsrc ! intervideosink intervideosrc ! "video/x-raw,width=640,height=480" ! xvimagesink
Audiomixer blocksize, cant be 0, hence adjusting the minimum value to 1
timeout value of aggregator is defined with MAX of MAXINT64,
but it cannot cross G_MAXLONG * GST_SECOND - 1
Hence changed the max value of the same
https://bugzilla.gnome.org/show_bug.cgi?id=738845
Signal sparse streams properly in stream-start event and force sending
of pending sticky events which have been stored on the pad already and
which otherwise would only be sent on the first buffer or serialized
event (which means very late in case of subtitle streams). Playsink in
playbin waits for stream-start or another serialized event, and if we
don't do this it will wait for the multiqueue to run full before
starting playback, which might take a couple of seconds.
https://bugzilla.gnome.org/show_bug.cgi?id=734040
All pads of a stream are now added at the beginning. In order to cope with
streams that don't get any data (forever or for a long time) we detect gaps
and push out GAP events when needed.
Cleanups and commenting by Jan Schmidt <jan@centricular.com>
https://bugzilla.gnome.org/show_bug.cgi?id=734040
Some VC1 decoder can have different caps according to wmv format, ie
WMV3 or WVC1.
So instead of keeping the first available caps, we interserct with
current WMV format.
https://bugzilla.gnome.org/show_bug.cgi?id=738532
When stream-format is ASF or sequence-layer-raw-frame, we basically have
a raw frame so we can parse it to extract some information such the
keyframe flag. The only requirement is to have a valid sequence-header.
This commit parse the frame header and set the DELTA_UNIT buffer flag in
case the frame is not a keyframe.
https://bugzilla.gnome.org/show_bug.cgi?id=738519
frame-layer header is represented as a sequence of 32 bit unsigned
integer serialized in little-endian byte order, so framesize is on the
first 3 bytes.
SMPTE 421M Annex L.
https://bugzilla.gnome.org/show_bug.cgi?id=738243
Also, strictly speaking, these numbers aren't DLT_*; they are LINKTYPE_* because
libpcap translates from internal OS-specific DLT_ numbering to the portable
LINKTYPE_ number space when writing files.
https://bugzilla.gnome.org/show_bug.cgi?id=738206
GST_BASE_PARSE_FRAME_FLAG_PARSING value is wrong, and the same flag is
not being used presently. Hence changing the value and commenting it out.
This needs to be included in baseparse.h later on
https://bugzilla.gnome.org/show_bug.cgi?id=737411
There are unnecessary definitions for disabling deprecation warnings.
Since GLIB_DISABLE_DEPRECATION_WARNINGS is not needed anymore in these files,
removing the same.
https://bugzilla.gnome.org/show_bug.cgi?id=737559
If a discontinuity in the stream is detected, data is discarded until
a new PES starts. If the first packet after the discontinuity is also
the start of a PES, there is no reason to discard the packets.
https://bugzilla.gnome.org/show_bug.cgi?id=737569
Or not negotiated in the case we would be actually not negotiated
Currently we are getting assertions from
gst_pb_utils_add_codec_description_to_tag_list because of NULL
caps.
https://bugzilla.gnome.org/show_bug.cgi?id=737186
If we don't have a seq_layer_buffer, we also don't have a valid
seq_layer because there are set together in
gst_vc1_parse_handle_seq_layer().
So when output header format is sequence-layer and when we don't have a
seq_layer_buffer, we forge one from seq_hdr.
https://bugzilla.gnome.org/show_bug.cgi?id=736781
Sequence-layer and frame-layer are serialized in little-endian byte
order except for STRUCT_C and framedata fields as described in SMPTE 421M Annex
L.
https://bugzilla.gnome.org/show_bug.cgi?id=736750
packetized mode is being set when framerate is being set
which is not correct. Changing the same by checking the
input segement format. If input segment is in TIME it is
Packetized, and if it is in BYTES it is not.
https://bugzilla.gnome.org/show_bug.cgi?id=736252
This commit fix several issues with sequence layer header forging on
update_caps():
- 0x00000004 unsigned integer is before STRUCT_C.
- Set reserved bits of STRUCT_C to their values for simple/main
profiles in sequence layer header format and ASF header format.
- Sequence layer shall be represented as a sequence of 32 bits unsigned
integers and shall be serialized in little-endian byte order except
for STRUCT_C which shall be serialized in big-endian byte-order.
See SMPTE 421M Annex L for more details about sequence layer format.
https://bugzilla.gnome.org/show_bug.cgi?id=736474
packet_length is defined as a guint16 in the PESHeader structure. This
definition match the specification. But since we add 6 bytes to the
packet_length value (length of start_code + stream_id + packet_length),
we can overflow the guint16 when the value in the PES header is greater
than 65529.
So use a guint32 instead of a guint16 to avoid overflow.
https://bugzilla.gnome.org/show_bug.cgi?id=736490
In gst_data_uri_src_create(), buf cannot be NULL, hence
else if (*buf != NULL) will be invalid so removing the
else if condition and adding a check to unreference buf
in else condition, just in case
https://bugzilla.gnome.org/show_bug.cgi?id=735861
gst_zebra_stripe_transform_frame_ip_planarY
gst_zebra_stripe_transform_frame_ip_YUY2
gst_zebra_stripe_transform_frame_ip_AYUV
all above 3 functions do the same functionality except for offset and pixel stride.
Hence moving the functionality to a single funtion.
https://bugzilla.gnome.org/show_bug.cgi?id=735032
Do more elaborate validation of the input caps: what fields
are required and/or not allowed. Don't assume AVC3 format
input without codec_data field is byte-stream format. Fix
up some now-unreachable code (CID 1232800).
It should try to use bytestream in these cases that the format
is set to _FORMAT_NONE as it seems that is what the 'else' clause
for bytestream can handle (by defaulting to _FORMAT_BYTESTREAM).
gst_pad_get_pad_template_caps() returns a reference which is unreferenced,
so creating a copy using gst_caps_copy() results in a reference leak.
Fixes https://bugzilla.gnome.org/show_bug.cgi?id=734528
32 bit integers are going to overflow, especially the PCR offset to
the first PCR will overflow after about 159 seconds. This makes playback
of streams stop at 159 seconds as suddenly the timestamps are starting
again from 0. Now we have a few more years time until it happens again
and 64 bits are too small.
They are kept until the probes are removed but they will never be
removed as the refcount of the element won't get to 0 because the
probes own references (cyclic refs). As the probes should only be
running as long as the element is running there is no need to
secure a ref for them.
Removes 3 leaked refs of wrappercamerabinsrc
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
The notify signal is triggered when caps is changed. But instead of
proxying the fixed caps, we query for the caps. Hence, when we go to
READY state, we endup setting template caps on the proxied caps
filter instead of NULL, which leads to negoitation failure. Correctly
proxy NULL caps if this is the new caps. Fixes not negotiated error
when running in cheese. Also fix a leak of caps string in one of the
trace.
https://bugzilla.gnome.org/show_bug.cgi?id=732741
We can still get OOB events while stopping the watchdog element, and while
stopping it we destroy the main context.
Also let the GSource own a reference to the element for additional safety.
https://bugzilla.gnome.org/show_bug.cgi?id=732554
Always use a GstAdapter when collecting access units (alignment="au")
in either byte-stream or avcC format. This is required to properly
preserve config headers like SPS and PPS when invalid or broken NAL
units are subsequently parsed.
More precisely, this fixes scenario like:
<SPS> <PPS> <invalid-NAL> <slice>
where we used to reset the output frame buffer when an invalid or
broken NAL is parsed, i.e. SPS and PPS NAL units were lost, thus
preventing the next slice unit to be decoded, should this also
represent any valid data.
https://bugzilla.gnome.org/show_bug.cgi?id=732203
Signed-off-by: Gwenole Beauchesne <gwenole.beauchesne@intel.com>
Carefully track cases when skipping broken or invalid NAL units is
necessary. In particular, always allow NAL units to be processed
and let that gst_h264_parse_process_nal() function decide on whether
the current NAL needs to be dropped or not.
This fixes parsing of streams with SEI NAL buffering_period() message
inserted between SPS and PPS, or SPS-Ext NAL following a traditional
SPS NAL unit, among other cases too.
Practical examples from the H.264 AVC conformance suite include
alphaconformanceG, CVSE2_Sony_B, CVSE3_Sony_H, CVSEFDFT3_Sony_E
when parsing in stream-format=byte-stream,alignment=au mode.
https://bugzilla.gnome.org/show_bug.cgi?id=732203
Signed-off-by: Gwenole Beauchesne <gwenole.beauchesne@intel.com>
Improve parser state tracking by introducing new flags reflecting
it: "got-sps", "got-pps" and "got-slice". This is an addition for
robustness purposes.
Older have_sps and have_pps variables are kept because they have
a different meaning. i.e. they are used for deciding on when to
submit updated caps or not, and rather mean "have new SPS/PPS to
be submitted?"
Signed-off-by: Gwenole Beauchesne <gwenole.beauchesne@intel.com>
Use gst_h264_parser_identify_nalu_unchecked() to identify the next
NAL unit. We don't want to parse the full NAL unit, but only the
header bytes and possibly the first RBSP byte for identifying the
first_mb_in_slice syntax element.
Also fix check for failure when returning from that function. The
only success condition for that is GST_H264_PARSER_OK, so use it.
https://bugzilla.gnome.org/show_bug.cgi?id=732154
Signed-off-by: Gwenole Beauchesne <gwenole.beauchesne@intel.com>
The gst_h264_parse_pps() function dynamically allocates the slice
group ids map array, so that needs to be cleared before parsing a
new PPS NAL unit again, or when it is no longer needed.
Likewise, a clean copy to the internal NAL parser state needs to be
performed so that to avoid a double-free corruption.
https://bugzilla.gnome.org/show_bug.cgi?id=707282
Signed-off-by: Gwenole Beauchesne <gwenole.beauchesne@intel.com>
The recovery point SEI message helps a decoder in determining if the
decoding process would produce acceptable pictures for display after
the decoder initiates random access or after the encoder indicates
a broken link in the coded video sequence.
This is not used in the h264parse element, but it could help debugging.
https://bugzilla.gnome.org/show_bug.cgi?id=723380
It was previously a mix and match of both variants, introducing just too much
confusion.
The prefix are from now on:
* GstMpegts for structures and type names (and not GstMpegTs)
* gst_mpegts_ for functions (and not gst_mpeg_ts_)
* GST_MPEGTS_ for enums/flags (and not GST_MPEG_TS_)
* GST_TYPE_MPEGTS_ for types (and not GST_TYPE_MPEG_TS_)
The rationale for chosing that is:
* the namespace is shorter/direct (it's mpegts, not mpeg_ts nor mpeg-ts)
* the namespace is one word under Gst
* it's shorter (yah)
In ISO/IEC 14496-15, the minimum size of a HEVCDecoderConfigurationRecord
(i.e., the contents of a hvcC box) is 23 bytes. However, the code in h265parse
checks that the size of this data is not less than 28 bytes, and it refuses to
accept caps if the check fails. The result is that standards-conformant streams
that don't carry any parameter sets in their hvcC boxes won't play.
https://bugzilla.gnome.org//show_bug.cgi?id=731783
When wrapover/reset occur, we end up with a small window of time where
the PTS/DTS will still be using the previous/next time-range.
In order not to return bogus values, return GST_CLOCK_TIME_NONE if the
PTS/DTS value to convert differs by more than 15s against the last seen
PCR
https://bugzilla.gnome.org/show_bug.cgi?id=674536
Using 32bit unsigned values for corrected pcr/offset meant that we
potentially ended up in bogus values
Furthermore, refpcr - refpcroffset could end up being negative, which
PCRTIME_TO_GSTTIME() can't handle (and returned a massive positive value)
Co-Authored by: Thibault Saunier <tsaunier@gnome.org>
From a high level perspective, the new process for seeking h264
streams is as follows:
1) Rewind the stream until we find the first I-slice of a frame,
and mark its offset in the stream.
2) Rewind the stream until we find SPS and PPS informations,
to make sure the subsequent parser is up to date.
3) Accumulate optionnal SEI NAL units on the way.
4) Push the SPS, PPS and SEI units before the new keyframe.
https://bugzilla.gnome.org/show_bug.cgi?id=675132
This should always be set for valid files when we get there,
and checking this avoids having ad hoc checks further down
in several places.
Coverity 1139698
If _set_current_pcr_offset gets called after a flushing seek, we ended
up using the current group for delta calculation ... whereas we should
be using the first group to calculate shifts.
Also add an early exit if there are no changes to apply
When working in push mode, we need to be able to evaluate the duration
based on a single group of observations.
To do that we use the current group values
When handling the PTS/DTS conversion in new groups, there's a possibility
that the PTS might be smaller than the first PCR value observed, due to
re-ordering.
When using the current group, only apply the wraparound correction when we
are certain it is one (i.e. differs by more than a second) and not when it's
just a small difference (like out-of-order PTS).
https://bugzilla.gnome.org/show_bug.cgi?id=731088
When we receive sticky events from upstream, always return TRUE.
Fixes the issue where we receive custom sticky events (such as "uri")
and no pads are created yet.
Since all the other timestamp tracking now gets reset on a discont,
it makes sense to wait for a PCR and timestamp buffers like when
playback first starts
