The SPS struct might be filled out by a call to
gst_h264_parser_parse_subset_sps, which fills out
dynamically allocated data and requires a call
to gst_h264_sps_clear() to free it. Also make sure
to clear out any allocated SPS data when returning
an error.
https://bugzilla.gnome.org/show_bug.cgi?id=753306
The payloader didn't copy anything so far, the depayloader copied every
possible meta. Let's make it consistent and just copy all metas without
tags or with only the video tag.
https://bugzilla.gnome.org/show_bug.cgi?id=751774
When playing mts files with embedded subtitles, the buffer is mapped,
but not unmapped at the end resulting in a memory leak.
Also unref event in handle_dvd_event as it takes ownership of the event.
https://bugzilla.gnome.org/show_bug.cgi?id=753539
When playing mts files with embedded subtitles, there are few event leaks.
Events are supposed to be transfer full. So if not forwarding the event,
they need to be freed.
https://bugzilla.gnome.org/show_bug.cgi?id=753539
h264parse and gstrtph264depay do the same, let's keep the behaviour
consistent. As we now include the codec_data inside the stream, this causes
less caps renegotiation.
https://bugzilla.gnome.org/show_bug.cgi?id=753228
rtph264depay does the same and this fixes decoding of some streams with 32
SPS (or 256 PPS). It is allowed to have SPS ID 0 to 31 (or PPS ID 0 to 255),
but the field in the codec_data for the number of SPS or PPS is only 5
(or 8) bit. As such, 32 SPS (or 256 PPS) are interpreted as 0 everywhere.
This looks like a mistake in the part of the spect about the codec_data.
This causes an assertion and would lead to getting a NULL instead
of a buffer. Without proper checking this would easily lead to a
segfault.
Related to rpth264depay: https://bugzilla.gnome.org/show_bug.cgi?id=737199
String parameters are expected to be passed as (f0r_param_string *),
which actually map to char**. In the filters this is evaluated as
(*(char**)param) which currently lead to crash when passing char*.
Remove the special case for string, all types, including char* as
passed as a reference.
https://phabricator.freedesktop.org/T83
Check if downstream is seekable via a SEEKING query and output a
BYTE segment if we want to seek back to fix up the headers later,
but if we're streaming send a TIME segment instead (which goes
down better with e.g. asfmux ! rtpasfpay).
https://bugzilla.gnome.org/show_bug.cgi?id=719553
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=747613
In the case where you have a source giving the GstAggregator smaller
buffers than it uses, when it reaches a timeout, it will consume the
first buffer, then try to read another buffer for the pad. If the
previous element is not fast enough, it may get the next buffer even
though it may be queued just before. To prevent that race, the easiest
solution is to move the queue inside the GstAggregatorPad itself. It
also means that there is no need for strange code cause by increasing
the min latency without increasing the max latency proportionally.
This also means queuing the synchronized events and possibly acting
on them on the src task.
https://bugzilla.gnome.org/show_bug.cgi?id=745768
VPS is not mandatory, and need not check for its presence before setting
the caps. Because of the check, in streams which don't have VPS,
sticky event mishandling happens.
https://bugzilla.gnome.org/show_bug.cgi?id=752807
In media to caps function, reserved_keys array is being used for variable i,
leading to GLib-CRITICAL **: g_ascii_strcasecmp: assertion 's1 != NULL' failed
changed it to variable j
https://bugzilla.gnome.org/show_bug.cgi?id=753009
Skip keys from the fmtp, which we already use ourselves for the
caps. Some software is adding random things like clock-rate into
the fmtp, and we would otherwise here set a string-typed clock-rate
in the caps... and thus fail to create valid RTP caps
https://bugzilla.gnome.org/show_bug.cgi?id=753009
The PID on a pad shouldn't change on a state change, only
if the pad is freed and a new one created. Clearing the PID
prevented mpegtsmux from being reused, because all packets
would end up muxed in PID 0
https://bugzilla.gnome.org/show_bug.cgi?id=752999
Accumulate streamheader packets in reverse into the
GList for efficiency, and reverse the list once when
processing.
Improves muxing speed when there are a lot of
streamheaders.
Don't throw away AU delimiter(s) that precede the SPS/PPS. Should
fix MPEG-TS playback on iOS/Quicktime when muxing streams that
already have AU delimiters.
See https://bugzilla.gnome.org/show_bug.cgi?id=736213 for getting
h264parse to insert AU delimiters when they don't already
exist.
We need to sync the pad values before taking the aggregator and pad locks
otherwise the element will just deadlock if there's any property changes
scheduled using GstController since that involves taking the aggregator and pad
locks.
Also add a test for this.
https://bugzilla.gnome.org/show_bug.cgi?id=749574
ret is declared just to initialize to TRUE and overwrite with the value of
vret. We can return the value of vret directly. vret is TRUE unless the
forward_event_func sets it to FALSE.
We only want to do a hard reset of the observations if we're working
with TIME segments in push mode. For BYTE segment we want to keep
the observations (in order to do seeks in push-mode).
When in push mode, we want to discard all previous observations from the
mpegtspacketizer when we get a DISCONT buffer.
This avoids trying to calculate bogus timestamps (estimating them using old
PCR observations).
We only do a hard reset in push-mode. In pull-mode we still need the observations
(in order to seek properly)
This is not public API, use g_assert() instead of
g_return_if_fail(), so that it's compiled out in
releases. It's only called from our code, with &foo.
Introduced by c4c9fe60b pcapparse: Take buffer directly from the adapter
Using gst_adapter_take_buffer_fast() can lead to buffers that are
made up of multiple memories with the first memory smaller than the
RTP header size, which violates assumptions GstRtpBaseDepayloader
makes, namely that the complete RTP header will be in the first
memory. This leads to such packets being dropped when feeding
them from pcapparse to RTP depayloaders. Use take_buffer() so
we get buffers with a single memory.
Introduced by c4c9fe60b pcapparse: Take buffer directly from the adapter
Flush any trailing bytes after the payload from the adapter as well,
otherwise we'll read a bogus packet size from the adapter next and
then everything goes downhill from there.
https://bugzilla.gnome.org/show_bug.cgi?id=751879
Can be used to fix misbehaving sinks. It will pass through all buffers
until it encounters GST_FLOW_ERROR or GST_FLOW_NOT_NEGOTIATED (configurable).
At that point it will unref the buffers and return GST_FLOW_NOT_LINKED
(configurable) - until the next READY_TO_PAUSED or FLUSH_STOP.
https://bugzilla.gnome.org/show_bug.cgi?id=750098
Move the pixel-aspect-ratio calculations higher up in caps
determination, so the results are available for a call to
gst_video_multiview_guess_half_aspect() when stereoscopic video
is detected.
Use QOS messages to update rendered and dropped frame stats. This is
the only accurate method. The old method didn't take max-lateness and
latency into account.
After few iteration, this variable became the same as dts. It's not
the min as the name says, but the dts of the current buffer. Simply
remove and place with dts. Also move the debug trace to actually
print the signed version of the running-time dts.
after e000a6f0a4, there is build error in bad plugins
this happens because, GST_CLOCK_STIME_IS_VALID () is being checked for pad_data
but it expects a GstClockTime parameter. Changing the check to 'dts'
https://bugzilla.gnome.org/show_bug.cgi?id=750961
The segment should start at first PTS, and the vairable name lower_pts
state so correctly. Though we where using the first DTS instead. This
could lead to small desynchronization of video stream.
https://bugzilla.gnome.org/show_bug.cgi?id=740575
Use the saved DTS, make it signed and pass that to the stream muxer. This
preserves the running time sign. All usage of -1 as invalid TS are now
replaced with G_MININT64. Negative values will be seen as wrap-around
point, but the delta between PTS and DTS will remain correct. Demuxers
don't care about absolute values, they only cares about deltas.
https://bugzilla.gnome.org/show_bug.cgi?id=740575
There was code to detect backward dts, but the marker min_dts
was never set. Setting it enable this feature that prevents
potential integer overflow when generating TS.
https://bugzilla.gnome.org/show_bug.cgi?id=740575
Wait until at least one keyframe has been parsed before
deciding to switch to passthrough mode, in case the
stream contains SEI messages that supplement the output
caps - for example by providing stereoscopic information
We were off by one byte in the matching
It should be (using 24 bit matching):
* startcode : 0000 0000 0000 0000 1000 00xx
* mask (bin) : 1111 1111 1111 1111 1111 1100
* mask (hex) : f f f f f c
* match : 0 0 0 0 8 0
https://bugzilla.gnome.org/show_bug.cgi?id=750685
In case of the videomark being partially or fully outside,
an error was bein thrown saying, mark width is more than video width.
And when the width, offset properties are set to maximum it resulted in crash.
Instead of throwing error, added logic to detect the mark
in case of partial visibility or dont show the mark when it is outside.
https://bugzilla.gnome.org/show_bug.cgi?id=743908
In case of the videomark being partially or fully outside, an error was being
thrown saying the mark width is more than video width. And when the width,
offset properties are set to maximum it resulted in crash. Instead of throwing
an error, add logic to detect the mark in case of partial visibility or don't
show the mark when it is outside.
https://bugzilla.gnome.org/show_bug.cgi?id=743908
Chinese broadcaster encapsulate AVS video codec into MPEG2-TS. They
use the stream_id 0x42 to identify AVS video streams. It should be noted
that this id is currently within the ISO reserved range, hence it's
utilisation is unofficial.
https://bugzilla.gnome.org/show_bug.cgi?id=727731
Value of res is reset to FALSE in each iteration of the while loop. We want to
conserve TRUE if any pad event succeeded until we arrive to done.
Also, buf is set to the value of *outbuf twice. Removing the first assignment
since the second one is outside of a conditional.
Like SPS/PPS they do contain information which will be needed to
decode the following data (as per definition of the flag)
Also ensures that the series of SPS/PPS/SEI NALU before a keyframe
can be considered as one contiguous header
In the same way we do it for the DELTA_UNIT flag
This allows downstream elements to know whether a given mpeg-ts
packet contains a corresponding HEADER elementary unit
Timestamps should start at the segment start, rather than 0, so
we need to not subtract the first timestamp. This makes the sink
correctly account for running time when switching PMTs where a
stream starts not quite at zero, causing timing offsets that can
become noticeable and causing dropped frames after a few times.
A new program object is created to replace an existing one
in the programs hash table, so its refcount needs to match.
With the default of 0 refcount on creation, the next PAT
change will cause that refcount to be both incremented and
decremented (assuming the new PAT references that stream too),
which will cause the program to be destroyed.
https://bugzilla.gnome.org/show_bug.cgi?id=748412
In the H263 spec, CPFMT is present only if the use of a custom
picture format is signalled in PLUSEPTYPE and UFEP is "001",
so we need to check params->format and only if the value is
6 (custom source format) the CPFMT should be read, otherwise
it's not present and wrong data will be parsed.
When reading the CPFMT, the width and height were not
calculated correctly (wrong bitmask).
https://bugzilla.gnome.org//show_bug.cgi?id=749253
Rather than one of the input pad video info's.
The test checking this was not constraining the output frame size
to ensure that the out of frame stream was not being displayed.
No need to call gst_remove_silence_reset() in gst_remove_silence_init() because
vad_new() already calls this function. Since there are no more uses of
_silence_reset(), we can remove it altogether.
Don't use the apis in codec-utils to extract the profile and level
syntax elements since it is wrong if there are emulation prevention
bytes existing in the byte-stream data.
https://bugzilla.gnome.org/show_bug.cgi?id=747613
It's a waste of resources to map it if it won't be converted
or used at all. Since we moved the frame mapping down, we need
to use the GST_VIDEO_INFO accessor macros now in the code above
that instead of the GST_VIDEO_FRAME accessor macros.
https://bugzilla.gnome.org/show_bug.cgi?id=746147
For each frame, compare the frame boundaries, check if the format contains an
alpha channel, check opacity, and skip the frame if it's going to be completely
overwritten by a higher zorder frame. The check is O(n^2), but that doesn't
matter here because the number of sinkpads is small.
More can be done to avoid needless drawing, but this covers the majority of
cases. See TODOs. Ideally, a reverse painter's algorithm should be used for
optimal drawing, but memcpy during compositing is small compared to the CPU used
for frame conversion on each pad.
https://bugzilla.gnome.org/show_bug.cgi?id=746147
Don't use the apis in codec-utils to extract the profile,tier and level
syntax elements since it is wrong if there are emulation prevention
bytes existing in the byte-stream data.
https://bugzilla.gnome.org/show_bug.cgi?id=747613
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
