rename gst-launch --> gst-launch-1.0
replace old elements with new elements(ffmpegcolorspace -> videoconvert, ffenc_** -> avenc_**)
fix caps in examples
https://bugzilla.gnome.org/show_bug.cgi?id=759432
The edit rate is only supposed to be the same in a source package, but there
might be multiple source packages with the same essence container. As such
just comparing the body/index SID is not sufficient.
This was completely broken before and could only work on a very constrained
set of files. After these changes it should work except for situations where
PTS != DTS, which is not handled at all in mxfdemux currently.
https://bugzilla.gnome.org/show_bug.cgi?id=759118
According to S377-1-2009c 9.2 the local tags must not be resolved from the
primer pack, which as a result means that there can't be any other tags than
statically assigned ones.
This is to support byte-stream decoder that does not remember the
PPS/SPS after a flush. This is not needed by all decoders, but is
harmless for those that do remember.
https://bugzilla.gnome.org/show_bug.cgi?id=758405
The order in which program switch must happen is:
1) drain all data on old pads (but don't push EOS)
2) add new pads (but don't push any data on them)
3) Push EOS and remove old pads
4) Start pushing data on new pads
There was one caveat in this implementation, which is that when
we activate a sparse pad (step 2) we would push a GAP event. The problem
is that, while being an event, it is actually *data*.
We therefore need to make sure pushing those GAP event is done at the step
we start pushing data.
https://bugzilla.gnome.org/show_bug.cgi?id=750402
Before we add any streams, make sure we drain all streams. This ensures
there's consistency that only "new" data will be pushed on buffers once
the new pads are added
https://bugzilla.gnome.org/show_bug.cgi?id=750402
When changing programs, the order of events needs to be the following:
* add pads from new program
* send EOS on old pads
* remove old pads
* emit 'no-more-pads'
Previously tsdemux was not doing that, and was first deactivating and
removing old pads before adding new ones.
We fix this by allowing subclasses of mpegtsbase to be able to handle
themselves the deactivation of programs. In this case tsdemux will
properly deactivate it once it has activated the new program.
https://bugzilla.gnome.org/show_bug.cgi?id=750402
The videoframe-audiolevel element acts like a synchronized audio/video "level"
element. For each video frame, it posts a level-style message containing the
RMS value of the corresponding audio frames. This element needs both video and
audio to pass through it. Furthermore, it needs a queue after its video
source.
https://bugzilla.gnome.org/show_bug.cgi?id=748259
0x04 signifies a MPEG elementary stream but according to RP2008, 0x10 should
be used for a h264 byte-stream. This also fixes compatibility of our files
with ffmpeg.
If packet->payload_unit_start_indicator is true and pointer 0, there is no
discontinuity check. Therefore there could be a previous section not complete
that need to be cleared.
https://bugzilla.gnome.org/show_bug.cgi?id=758010
The values of channel_mapping are copied by gst_codec_utils_opus_create_caps ()
but it doesn't free or take ownership of the g_new0 allocated memory. This
needs to be freed before going out of scope.
CID 1338692
buf surely isn't NULL inside the block conditional to a buffer size bigger
than (G_MAXUINT16 - 3). Plus gst_buffer_unref() checks if the buffer is
NULL and does nothing if it is.
CID 1338693
If tsdemux never receives data for a stream, the corresponding pad will never
be added and stream->active will remain FALSE. When the stream is removed, the
pad will not be unreffed and will be leaked.
https://bugzilla.gnome.org/show_bug.cgi?id=757873
The current implementation for detecting the resolution changes
on key frames is based on vp8 bitstream alignment. Avoid this
width and height parsing for vp9 bitstream, which requires proper
frame header parsing inorder to detect the resolution change (Fixme).
https://bugzilla.gnome.org/show_bug.cgi?id=757825
It is up to the element handling the seek to send flush events
downstream, otherwise we end up with a situation where upstream
would get unexpected GST_FLOW_FLUSHING
The Onvif Streaming Specification specifies that the NTP timestamps
in the Onvif extension header indicaes the absolute UTC time associated
with the access unit. But by using running time we can not achieve that,
since a frame's running time depends on the played interval, whether a
non-flushing is done, etc. Instead we have to use the stream time.
https://bugzilla.gnome.org/show_bug.cgi?id=757688
It is now possible to update the currently used ntp-offset with a
custom serialized downstream event. The element will read the ntp-offset
property when doing the state transition from READY to PAUSED and
use that offset until it receives a "GstNtpOffset" event, which also
has a "ntp-offset" attribute in that it's structure. In case the
property is not set and no event has been received, the element will
guess the npt-offset with help of the clock. If no clock can be
retrieved, the element will error out and stop the data flow.
The same event is also used for updating the D/E-bits in the RTP
extension header. The discont flag in a buffer can be set whenver a
live/network source looses a frame, but that is not the type of
discontinuity that the onvif extension header should reflect. The
header is mainly used for playback of a track concept, in which
gaps can be present, and it's those kind of gaps that should be
highlighted with the D- and E-bits.
https://bugzilla.gnome.org/show_bug.cgi?id=757688
If a buffer or a buffer list is cached, no events serialized with the
data stream should get through. The cached buffers and events should
be purged when we stop flushing.
https://bugzilla.gnome.org/show_bug.cgi?id=757688
Otherwise those symbols can conflict with external libraries when
linking everything statically for mobile targets.
Use the gst_gm_ prefix, short for gst geometric math.
https://bugzilla.gnome.org/show_bug.cgi?id=756882
Store and copy input state fields when setting the
output state of the decoder. Avoids problems like
the framerate set by an upstream element being ignored
Related to:
https://bugzilla.gnome.org/show_bug.cgi?id=756563
New subclass with a similar behaviour as the old liveadder, but
a slightly different API as the latency is in nanoseconds, not
milliseconds. Also, the new liveadder has a effective latency that
is latency + output-buffer-duration. In practice, just setting a non-zero
latency with the new audiomixer gives you the right behavior in 99% of the
cases.
Build error due to wrong argument type in debug message
aagg->priv->offset and next_offset are of type int64, but uint64
formatter is being used in logs. Changing all those to int64
https://bugzilla.gnome.org/show_bug.cgi?id=756065
When g_option_context_parse fails, context and error variables are not getting free'd
which results in memory leaks. Free'ing the same.
And replacing g_error_free with g_clear_error, which checks if the error being passed
is not NULL and sets the variable to NULL on free'ing.
https://bugzilla.gnome.org/show_bug.cgi?id=753854
The buffer timestamps in the collect function will already be
running time, don't try to convert them again to running time,
this would yield CLOCK_TIME_NONE now that the segment is shifted
to account for negative dts.
This fixes x264enc ! mpegtsmux ! hlssink, which was broken
because mpegtsmux would send a downstream key unit event with
running time NONE and then hlssink would immediately send
another one upstream and it would just be a flood of force
keyframe events in both directions after the first one. This
would then break hlssink because it uses multifilesink in
next-file=key-unit-event mode, and starting a new file after
every few kB does not work well for HLS.
The alsamidisrc element allows to get input event from ALSA MIDI
sequencer devices, and possibly convert them to sound using some
downstream element like fluiddec.
https://bugzilla.gnome.org/show_bug.cgi?id=738687
An IDX file or codec_data normally contains the original frame size of
the video. Allow upstream to provide this information by sending a
custom event, which will allow scaling the overlay correctly.
https://bugzilla.gnome.org/show_bug.cgi?id=663750
Instead of only supporting writing SPU data directly to YUV frames,
render the SPU data to an intermediate AYUV overlay buffer. The overlay
data is then attached to the video frame if downstream supports overlay
composition, otherwise the AYUV overlay is blended to the video frame.
For the PGS format, the overlay buffer size is set to the size of the
Composition Window, and its position in the overlay composition is set
to the window position. The objects to render are now cropped when the
cropping flag is set.
For the Vobsub format, the overlay buffer size is set to the size of the
Display Area.
Once rendered, the overlay composition rectangle is now moved and scaled
to fit the video output size, to avoid clipping.
https://bugzilla.gnome.org/show_bug.cgi?id=663750
We might've queued up a GAP buffer that is only partially inside the current
output buffer (i.e. we received it too late!). In that case we should only
skip the part of the GAP buffer that is inside the current output buffer, not
also the remaining part. Otherwise we forward this pad too far into the future
and break synchronization.
Derive from GstVideoSink so that preroll frames will automatically
get rendered too, unless the show-preroll-frame property is set to
FALSE. Fixes intervideosrc only picking up frames if intervideosink
is in PLAYING state.
https://bugzilla.gnome.org/show_bug.cgi?id=755049
Fix the negotiation of GstVideoOverlayComposition by checking
intersection with the peer caps, rather than just accept-caps,
which might only check the pad template.
https://bugzilla.gnome.org/show_bug.cgi?id=755113
We have to queue buffers based on their running time, not based on
the segment position.
Also return running time from GstAggregator::get_next_time() instead of
a segment position, as required by the API.
Also only update the segment position after we pushed a buffer, otherwise
we're going to push down a segment event with the next position already.
https://bugzilla.gnome.org/show_bug.cgi?id=753196
x/y/w/h are signed integers. As can be seen in GstCompositorPad.
The prototype for clamp_rectangle was wrong. This commit reverts the change
and fixes the prototype.
This reverts commit bca444ea4a.
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 by using MIN().
CID 1320707
As it's recursive, gst_pad_get_allowed_caps() may also return
empty for anything incompatible downstream. EMPTY is not valid caps
value for gst_caps_fixate(). This lead to assertion and then crash.
Ideally, the negotiate function should be re-factored to have a return
value, and we could make the negotiation fails earlier.
https://bugzilla.gnome.org/show_bug.cgi?id=754122
The obscured check in compositor was using the dimensions of the pad to clamp
the h/w of the pad instead of the output resolution, and was doing an incorrect
calculation to do so. Fix that by simplifying the whole calculation by using
corner coordinates. Also add a test for this bug which fell through the cracks,
and just skip all the obscured tests if the pad's alpha is 0.0.
https://bugzilla.gnome.org/show_bug.cgi?id=754107
The tsdemux latency should always be added to the minimum
latency (which is always a valid clock time value). The
"cleanup" in commit a1f709c2 made it so that it would not
be added if upstream reported 0 as minimum latency (as
e.g. udpsrc would). This broke playback of live mpeg-ts
streaming in some cases, leading to playback stutter due
to a too-small configured latency for the pipeline.
https://bugzilla.gnome.org/show_bug.cgi?id=751508
In gst_live_adder_chain() function, calls to gst_buffer_copy_region() can lead
to assertion as 'offset + size <= bufsize' is not respected.
Indeed 'offset' and 'size' parameters are calculated through calling gst_live_adder_length_from_duration(),
and thus gst_util_uint64_scale_int_round().
Depending on the nearest integers, rounded values 'offset' and 'size' can then trigger the assertion.
This case mainly occurs when 'skip' value is > 0 in chain function process.
https://bugzilla.gnome.org/show_bug.cgi?id=753759
It is faster than doing a query that propagates downstream and
should be enough
Elements: faac, gsmenc, opusenc, sbcenc, voamrwbenc, adpcmenc, sirenenc
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
A race condition in the state change function may cause buffers to be
unreffed while they are still used by the streaming thread in
gst_rtp_h265_pay_send_vps_sps_pps() resulting in a crash. Chain up to the
parent class first in the state change function to make sure streaming
has stopped and only then free those buffers.
https://bugzilla.gnome.org/show_bug.cgi?id=741381
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
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
(same as 744c58d71b but for the
position query)
It was only querying in time, but then trying to use dead bytes
to time conversion code.
Coverity 1139677
(Identical to commit 612cdeec80 which
was for resindvd)
When we'd see an unknown stream type, then a SDDS stream.
Then we'd get to the end of the switch with a NULL temp stream
pointer, and dereference it.
Coverity 1139708
Due to mpegts streaming nature some pads are created but are only added
later to the element. This can cause a scenario where the first stream
doesn't have an available decoder (while the next ones still pending
would have) and tsdemux will fail with not-linked as the first stream
added wouldn't be linked.
To avoid this tsdemux needs to add pads to the flowcombiner
when they are created instead of only when adding them to the
element.
gstfreeverb.c:781:29: error: using integer absolute value function 'abs' when
argument is of floating point type [-Werror,-Wabsolute-value]
if (abs (out_l2) > 0 || abs (out_r2) > 0)
They are very confusing for people, and more often than not
also just not very accurate. Seeing 'last reviewed: 2005' in
your docs is not very confidence-inspiring. Let's just remove
those comments.
This should not harm regular files, since those are the last 4 bytes of
a normal file.
This allows to handle playback of concatenated mpeg-ps files. Seeking and
duration reporting is still wrong though.
Testing mpegversion when mpegaudioversion was likely meant.
Similar tests in sys/androidmedia/gstamcaudiodec.c also test
mpegaudioversion with the same conditional code.
Coverity 206071
This component is dereferenced, and later code checking for
NULL in particular cases implies it can be NULL. This likely
does not fix the coverity warning as it was seeing another
path setting component to NULL explicitely, but this was
spotted by looking at:
Coverity 1139736
Which is actually OK from what I can see since the actual
dereference of the explicit NULL pointer will not happen
if the condition that led to the NULL pointer assignment
is met, since the assignment and defeference have mutually
exclusive tests.
Detect resolution changes on key frames, and propagate the resulting
caps to the src pad. Only the uncompressed data chunk is decoded, so
avoid using the new VP8 bitstream parsing library for now.
Signed-off-by: Gwenole Beauchesne <gwenole.beauchesne@intel.com>
Avoid possible division-by-zero while deriving the presentation timestamp
of the buffer. The base class will take care of any interpolation needs.
Signed-off-by: Gwenole Beauchesne <gwenole.beauchesne@intel.com>
* Search in current pending values first. For CBR streams we can very
easily end up having just one initial observations and then nothing
else (since the bitrate doesn't change).
* Use one group whether we are in that group *OR* if there is only
one group.
* If the group to use isn't closed (points are being accumulated in the
PCROffsetCurrent), use the latest data available for calculation
* If in the unlikelyness that all of this *still* didn't produce more
than one data point, just return the initial offset
While the calculation done in these macros will work with 64bit
integers, they will fail if working with 32bit integers.
Force the scaling up to solve that.
This amazingly didn't introduce major issues up to now, but resulted
in bogus values in debug logs.
Doing a hard flush on the packetizer will drop all observations, which
will eventually break push-based seeking (with BYTES segment) since
we won't know where to seek to anymore (new data would always be
considered as the beginning of the stream).
Turns out glib aborts on allocation failure, so this is pointless.
We'll just ignore Coverity warnings on such constructs.
This reverts commit d347809a82.
While this probably should never happen if callers are well behaved,
this avoids a crash if it does. With a warning about it. Unsure if
it'd be better to not add at all, but it should not happen...
Coverity 1139713
While it will probably not trigger, it should silence a Coverity
warning about the fail code path testing for NULLness before
freeing, where the buffer was already dereferenced. It seems
safest to keep that test, in case future goto fail statements
happen to have a NULL buffer there.
Coverity 1139851
Fixes multiple seeking issues. When doing ACCURATE or normal
non-KEYUNIT seeks, mxfdemux would just send data from the
edit unit that covered the seek position, whether that's
a keyframe or not. Decoders would only output things from
the next keyframe then, which means there's a gap between
the start of the segment and the first decoded data in
some cases. In combination with gst-editing-services this
might result in a frozen picture for the duration of that
gap at the beginning (if videorate fixes up the first
buffer's start timestamp to cover the entire gap), or
a black frame (if no videorate is used and videomixer
fills the gap). Also fixes A/V sync issue when requesting
a KEYUNIT seek.
gst_ts_demux_push_pending_data() will check if it now can activate the
stream and add the pad, we don't have to check that ourselves.
Fixes playback of very short MPEG TS files.
