Compositor does not support it currently and it needs special support
for handling this correctly, and is rather non-trivial to implement for
all formats.
For frame->buffer, baseparse is doing that automatically for us. For
frame->output_buffer it doesn't and assumes that the subclass is already
doing that. Consistency!
This is useful e.g. if audio buffers should be exactly the duration of a
video frame, or if a audio buffers should never be too large because of
latency constraints.
The element is taking a fractional buffer duration, to allow working
with e.g. 1001/30000 as output duration and it accumulates rounding
errors in the buffer durations and compensates for them by making some
buffers one sample larger than the others.
https://bugzilla.gnome.org/show_bug.cgi?id=774689
We will allocate a screen area of width*height*bpp bytes, however this
calculation can easily overflow if too high width or height are given
inside the stream. Nonetheless we would just assume that enough memory
was allocated, try to fill it and overwrite as much memory as wanted.
Also allocate the screen area filled with zeroes to ensure that we start
with full-black and not any random (or not so random) data.
https://scarybeastsecurity.blogspot.gr/2016/11/0day-poc-risky-design-decisions-in.html
Ideally we should just remove this plugin in favour of the one in
gst-libav, which generally seems to be of better code quality.
https://bugzilla.gnome.org/show_bug.cgi?id=774533
Type cast has higher precedence than bitwise shift, so the third
argument will truncate to 8 bits and then shift right by 8 bits
resulting in constant zero.
https://bugzilla.gnome.org/show_bug.cgi?id=774293
Consistently use GST_ROUND_UP_4(width) as stride for
bayer buffers. Bayer data will usually come in widths
that are multiples of 4 anyway, so hopefully this
should not have any adverse impact on anyone in
practice.
Before, bayer2rgb required input buffers to are sized
accordingly, but then didn't actually round up when
calculating row offsets. rgb2bayer didn't use a rounded
stride nor buffer size.
https://bugzilla.gnome.org/show_bug.cgi?id=752014
rawvideoparse wouldn't error out on not-negotiated,
but would just keep on going, because it didn't pass
the flow return value back to the parent class and
thus upstream, so the source wouldnt' stop streaming.
MSVC warns about this because it's a C++ compiler, and this actually
results in useful things such as the incorrect 'gboolean' return value
for functions that return GstFlowReturn, so let's do explicit
conversions to reduce the noise and increase its efficacy.
With MSVC, this gives the following warning:
warning C4305: 'function': truncation from 'double' to 'gfloat'
Apparently, MSVC does not figure out what type to use for constants
based on the assignment. This warning is very spammy, so let's try to
fix it.
In file included from ../subprojects/gst-plugins-base/gst-libs/gst/video/video.h:27:0,
from ../subprojects/gst-plugins-bad/gst/segmentclip/gstvideosegmentclip.c:25:
../subprojects/gst-plugins-base/gst-libs/gst/video/video-format.h:27:39: fatal error: gst/video/video-enumtypes.h: No such file or directory
#include <gst/video/video-enumtypes.h>
^
compilation terminated.
https://ci.gstreamer.net/job/GStreamer-master-meson/269/console
In M2TS mode, we need an extra 4 bytes in the buffer, so need
to ensure the buffer can contain these. The allocation site
does not know the mode, so this is done in all cases.
This was a regression.
We only have a upstream-id via STREAM_START if we were in push-mode.
In pull-mode we need to create one.
Note: It would be good to eventually have that method (copied from
gst_pad_get_stream_id_internal()) public in the future
For each MpegTSBaseStream, we have a GstStream object which
subclasses can extend with information.
For each program a GstStreamCollection is created with all
GstStream from each stream.
When dealing with TIME-based input, the incoming stream could have
potentially changed completely.
In order to check whether it did or not, we need to re-check all sections
(PAT, PMT...). If it didn't, we will keep using the existing streams/pad,
and if it did we will act as if there was a program switch.
Fixes HLS streaming with decodebin3/playbin3
The default value of D-bit is changed to TRUE so discontinuity
is set for initial request and seek request as well.
Only set the e_bit flag for the CUSTOM_DOWNSTREAM event if
a cached buffer exists.
https://bugzilla.gnome.org/show_bug.cgi?id=770221
EAC3 bit streams shall be identified with a stream_type value of 0x87 when
transmitted as PES streams conforming to ATSC-published standards. It is specified
in ATSC Standard A/52.
https://bugzilla.gnome.org/show_bug.cgi?id=770528
The headers passed as parametter are relative to the build dir
basically "../subproject/gst-plugins-bad/gst-libs/gst/mpegts/XXX.h"
but that does not match what is needed at build time when building as
subproject, also we always add current dir as include_dir so we are
safe including directly.
And link mpegtsdemux against the 'math' library as it is needed.
https://github.com/mesonbuild/meson
With contributions from:
Tim-Philipp Müller <tim@centricular.com>
Matej Knopp <matej.knopp@gmail.com>
Jussi Pakkanen <jpakkane@gmail.com> (original port)
Highlights of the features provided are:
* Faster builds on Linux (~40-50% faster)
* The ability to build with MSVC on Windows
* Generate Visual Studio project files
* Generate XCode project files
* Much faster builds on Windows (on-par with Linux)
* Seriously fast configure and building on embedded
... and many more. For more details see:
http://blog.nirbheek.in/2016/05/gstreamer-and-meson-new-hope.htmlhttp://blog.nirbheek.in/2016/07/building-and-developing-gstreamer-using.html
Building with Meson should work on both Linux and Windows, but may
need a few more tweaks on other operating systems.
_stdint.h is generated by Autotools and we don't really need it. All
supported platforms now ship with stdint.h. The only stickler was MSVC,
and since Visual Studio 2015 it also ships stdint.h now.
After seeking in aiff files the information about the data end offset is
discarded, leading to audio artifacts with metadata chunks at the end of
a file.
This patch retains the end offset information after a seek event.
https://bugzilla.gnome.org//show_bug.cgi?id=769376
timecodewait receives a timecode as an argument (either as string or as
GstVideoTimeCode - one is gst-launch-friendly and the other is code-friendly),
and it will drop all audio and video buffers until that timecode has been
reached.
https://bugzilla.gnome.org/show_bug.cgi?id=766419
When draining a program, we might send a newsegment event on the pads
that are going to be removed (and then the pending data).
In order to do that, calculate_and_push_newsegment() needs to know
what list of streams it should take into account (instead of blindly
using the current one).
All callers to calculate_and_push_newsegment() and push_pending_data()
can now specify the program on which to act (or NULL for the default
one).
Fixing the following warning when generating documentation:
xml/element-gaussianblur.xml:72: element refsect2: validity error :
ID GstGaussianBlur already defined
<refsect2 id="GstGaussianBlur" role="typedef">
^
Warning: multiple "IDs" for constraint linkend: GstGaussianBlur.
DOC Fixing cross-references
Fixing the following warning when generating documentation:
xml/element-chromium.xml:74: element refsect2: validity error :
ID GstChromium already defined
<refsect2 id="GstChromium" role="typedef">
^
Warning: multiple "IDs" for constraint linkend: GstChromium.
DOC Fixing cross-references
When skipping data, check if they are filler bytes. If so, drop the
data instead of skipping. We don't want to output filler bytes, but they
shouldn't cause a discontinuity.
https://bugzilla.gnome.org/show_bug.cgi?id=768125
If the input alignment claims AU alignment, each received
buffer should contain a complete video frame, so never hold over parts
of buffers for later processing. Also reduces latency, as packets
are parsed/converted and output immediately instead of 1 buffer
later.
Fixes a problem where an (arguably disallowed) padding byte on the
end of a buffer is detected as an extra byte in the following
start code, and messes up the timestamping that should apply to
that start code.
This is an automatic update with manual merges of running
"make update" in the doc/plugins directory. This should help
later maintenance of the plugins doc. A lot of plugin are
not referenced yet in the doc. Will come later.
And always set the sampling field on the src caps, if necessary guessing a
correct value for it from the colorspace field.
Also, did some cleanup: removed sampling enum - redundant.
https://bugzilla.gnome.org/show_bug.cgi?id=766236
The heuristic to choose between packetise or not was changed to use the
segment format. The problem is that this change is reading the segment
during the caps event handling. The segment event will only be sent
after. That prevented the decoder to go in packetize mode, and avoid
useless parsing.
https://bugzilla.gnome.org/show_bug.cgi?id=736252
A simple fix for the problem of creating new pads with duplicate
names when switching program, easier than the alternative of
trying to work out which pads might persist and manage that.
See https://bugzilla.gnome.org/show_bug.cgi?id=758454
Remove code that dealt with odd strides separately - there's
not really any overhead to just using 1 codepath for both matched
and unmatched stride output.
Add separate codepaths for BE vs LE GRAY16 input so they're
handled properly
As is done everywhere else, and avoids setting bogus values
And remove useless *<val> checks (we always provide valid values and
it's an internal function).
CID #1320700
This helps in cases where raw audio data is being delivered, but the
buffers do not come in sample aligned sizes. The new unalignedaudioparse
bin can be autoplugged and configures an internal audioparse element to
align the data. audioparse itself gets support for audio/x-unaligned-raw
input caps; the output caps then contain the same information, except that
the name is changed to audio/x-raw (since audioparse aligns the data).
This ensures that souphttpsrc ! audioparse still works.
https://bugzilla.gnome.org/show_bug.cgi?id=689460
When scanning for SCR / PTS / DTS, handle the case where
the pack header is followed by the optional system header,
so we can correctly collect timestamps in such cases.
https://bugzilla.gnome.org/show_bug.cgi?id=623860
When the file size is smaller than the configured 4MB scan
limit for timestamps, don't underflow the guard variable
when checking if it's time to stop.
Limit the backward SCR scan to the same 4MB as the PTS scan.
Add some comments.
Adds a new function to mpegts lib to create a iso639 language
descriptor from a language and use it in mpegtsmux to add
a language descriptor to audio streams that have a language set.
https://bugzilla.gnome.org/show_bug.cgi?id=763647
When the sub-class is delaying deactivation of the old program,
but it has the same program number as the new program, don't
overwrite the old program in the hash table and then steal
the new program back out of it. Instead, add the new program to
the hash table after handling removal of the old one.
This way we can use the base class for buffer allocation, hence use
fill() instead of create() virtual. This also adds a strict check on the
select pool buffer size as we don't support strides and padding.
This is based on initial patch proposed by Sebastien Dröge, from which I
also fixed a buffer pool leak.
https://bugzilla.gnome.org/show_bug.cgi?id=763441
As we currently only use the server reported "natural" format, caps
negotiation should simply be limited to telling the base class which
format to use. Fix the negotiation by moving the associated code
into negotiate() virtual function. Also, use gst_base_src_set_caps()
rather then setting it on the pad directly. Also protect against this
method being called multiple time (we can't renegotiate for now).
This change also moves some network code that was being run during the
application state change call, to be run on the streaming thread.
https://bugzilla.gnome.org/show_bug.cgi?id=739598
Although it's not very well documented, g_input_stream_read_all() will
set the number of bytes read to 0 if the connection is closed rather
then returning an error.
This prevents recursion on error. This used to happen as we
don't change the state when something fails. We end up running
and failing in the same state forever.
Using GSocketClient we can simplify a lot the read/write operation.
This also provide an GSocketConnection (a GIOStream) which can then
be used with the GTlsClientConnection for secure connections. Note
that we use _write_all() to ensure all bytes have been read. This is
to follow the fact the none of the _send() calls check the return
value.
When the security cannot be negotiated, the server returns
security type of 0 (failure). In that case, the next step is
to read the error reason string.
We get into this code path if the profile is already constrained-baseline and
downstream does not support constrained-baseline. So we should try baseline
and the other compatible profiles.
https://bugzilla.gnome.org/show_bug.cgi?id=764448
Request pads are requested by applications and as such should only be released
by them again. Instead of releasing them when stopping the muxer, just clear
their state so that they can be used again when starting the muxer again.
https://bugzilla.gnome.org/show_bug.cgi?id=763862
The parser handles the downstream force-key-unit event incorrectly,
it tries to parse it as an upstream force-key-unit event, does not
check the return value, and then uses uninitialized memory in
"all_headers" boolean variable.
https://bugzilla.gnome.org/show_bug.cgi?id=763793
When the sub-class claims a program for later freeing, make
sure it's not left in the hash table, or it can cause crashes on shutdown.
Make sure tsdemux frees any program it has kept around at shutdown
if it wasn't freed already.
https://bugzilla.gnome.org/show_bug.cgi?id=763503
This is a regression from since mpegvideoparser was switched to
use the codecparsing library.
The problem is that the high bit of the profile_and_level is used
to specify non-hierarchical profiles and levels. Unfortunately we
were discarding that information.
Expose that escape bit, and use it in the element
https://bugzilla.gnome.org/show_bug.cgi?id=763220
In some cases, the PTS might be smaller than the first observed PCR
value which causes element to apply wraparound leading to bogus
timestamp. To solve this, we only apply it if the PTS-PCR difference is
greater that 1 second to be sure that it's a real wraparound.
Moreover, using unsigned 32 bits values to handle wrapover could end up
with bogus value, so it use pts value to handle it.
Also, convert pcr time to gst time before comparing it to pts.
Since refpcr is expressed in PCR time base while pts is expressed in GStreamer
time.
https://bugzilla.gnome.org/show_bug.cgi?id=743259
Enabling passthorugh mode is causing multiple issue:
For nal aligned multiresoluton streams, passthrough mode
make h264parse unable to advertise the new resoultions.
Also causing issues while parsing MVC streams which have two
separate layers (base-view and non-base-view).
This fix is only a temporary workaround.
For MVC, proper fixes needed in many places:
(handle prefix nal unit, handle non-base-view slice nal extension,
fix the picture_start detection for multi-layer-mvc streams etc)
https://bugzilla.gnome.org/show_bug.cgi?id=758656
Remove calls to gst_pad_has_current_caps() which then go on to call
gst_pad_get_current_caps() as the caps can go to NULL in between. Instead just
use gst_pad_get_current_caps() and check for NULL.
https://bugzilla.gnome.org/show_bug.cgi?id=759539
Set fallback channel layout on files with more than two
channels. Not clear where to retrieve the real layout from
or what the default layout is for AIFF files, the spec
only seems to specify some layout for up to 6 channels
and the file in question doesn't have a CHAN chunk.
https://bugzilla.gnome.org/show_bug.cgi?id=676425
This fixes a couple of issues regarding the output of (request)
per-program pads output:
We would never push out PAT sections (ok, that was one reallly stupid
mistake. I guess nobody ever uses this feature ...).
In the case where the PMT section of a program was bigger than one
packet, we would only end up pushing the last packet of that PMT. Which
obviously results in the resulting stream never containing the proper
(complete) PMT.
The problem was that the program is only started (in the base class)
after the PMT section is completely parsed. When dealing with single-program
pads, tsparse only wants to push the PMT corresponding to the requested
program (and not the other ones). tsparse did that check by looking
at the streams of the program...
... but that program doesn't exist for the first packets of the initial
PMT.
The fix is to use the base class program information (if it parsed the
PAT it already has some information, like the PMT PID for a given program)
if the program hasn't started yet.
In addition to the fact that it's a sane thing to do for multi-source
pad elements, it also avoids the situation where just using a request
pad (and not the main static pad) would result in the processing
stopping.
tsdemux is not able to handle negative playback rates.
But in mpegtsbase, the same check is not being done.
added a check to not handle negative rate while seeking unless
the same is handled upstream.
https://bugzilla.gnome.org/show_bug.cgi?id=758516
Since commit b77f8e172a the new value
assigned to mview_mode hasn't been used. That commit changed the following
"if" check to an "else if", which means the original value of mview_mode
is used.
When converting from avc to byte-stream, there will not be any codec_data
in the src caps. Remove it before the equality check to avoid sending caps
events downstream on every SPS/PPS change.
https://bugzilla.gnome.org/show_bug.cgi?id=761014
If we have a stream that contains an unchanging SPS/PPS for every video frame,
we don't need to to constantly query downstream for it's supported caps if the
current caps are compatible with the negotiated caps.
https://bugzilla.gnome.org/show_bug.cgi?id=761014
When the framesize is not specified, we try and calculate a size from
the strides and offset information. This was done with the sum of
offsets + the size of the last frame. That is just wrong method. We also
need to account for video meta that may be flipping two planes. An
example is if you convert I420 to YV12 by flipping the two last offsets.
https://bugzilla.gnome.org/show_bug.cgi?id=760270
To make parser work with image having non-standard strides, plane
offsets or with padding between images.
For now, since element doesn't check for videometa, we can't directly
push buffers when these properties are set so it convert the frame
in the pre_push_buffer method to remove any custom padding.
https://bugzilla.gnome.org/show_bug.cgi?id=760270
Allows the subclass to completely override the chosen src caps.
This is needed as videoaggregator generally has no idea exactly
what operation is being performed.
- Adds a fixate_caps vfunc for fixation
- Merges gst_video_aggregator_update_converters() into
gst_videoaggregator_update_src_caps() as we need some of its info
for proper caps handling.
- Pass the downstream caps to the update_caps vfunc
https://bugzilla.gnome.org/show_bug.cgi?id=756207
When sps data is NULL, the buffer allocated and mapped is not being freed.
In this scenario there is no need to allocate the buffer as we are supposed to return NULL.
https://bugzilla.gnome.org/show_bug.cgi?id=761070
It's useful enough already to be used in other elements for audio aggregation,
let's give people the opportunity to use it and give it some API testing.
https://bugzilla.gnome.org/show_bug.cgi?id=760733
It's not enough to have timeout or event based VPS/SPS/PPS information
sent in RTP packets. There are some scenarios when key frames may appear
more frequently than once a second, in which case the minimum timeout
for "config-interval" of 1 second for sending VPS/SPS/PPS isn't enough.
It might also be desirable in general to make sure the VPS/SPS/PPS is
available with every keyframe (packet loss aside), so receivers can
actually pick up decoding immediately from the first keyframe if
VPS/SPS/PPS is not signaled out of band.
This commit adds the possibility to send VPS/SPS/PPS with every key frame.
This mode can be enabled by setting "config-interval" property to -1. In
this case the payloader will add VPS, SPS and PPS before every key (IDR)
frame.
https://bugzilla.gnome.org/show_bug.cgi?id=757892
The MPEG standard (ISO-13880-1) says the reserve bits need to be set
to one (2.1.64). This is causing transport streams to fail validation
on some systems.
https://bugzilla.gnome.org/show_bug.cgi?id=760127
This works usually in this place, unless the compiler optimizes things in
interesting ways in which case it causes stack corruption and crashes later.
The compiler in question here is clang with -O1, which seems to pack the stack
a bit more and causes writing to the guint as pointer to overwrite map.memory,
which then later crashes during unmapping of the memory.
While encoding the frame in ASCII mode, per component four bytes are needed
and after every 20 bytes, a \n will be added. So the calculation should be
size = size * (4 + 1 / 20). This should exclude the header being written.
Since header is also being included in the calculations, memory mishandlings
are happening.
https://bugzilla.gnome.org/show_bug.cgi?id=759520
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
