Use (1 << 31) as the multiplier for int<->float conversions. This makes
sure that int->float conversions always end up with floats between
[-1.0, 1.0].
For the conversion from float to int, this multiplier will give the complete
int range after we perform clipping.
Change the unit test to take this into consideration.
Fixes https://bugzilla.gnome.org/show_bug.cgi?id=755301
Encrypted RTP buffers may contain encrypted padding, hence it's
necessary to have an option to relax the validation in order to
successfully map the buffer.
When the flag GST_RTP_BUFFER_MAP_FLAG_SKIP_PADDING is set
gst_rtp_buffer_map() will map the buffer like if padding is not
present.
https://bugzilla.gnome.org/show_bug.cgi?id=752705
While creating caps in audiointerleave tests, bitmask is being set as 0x9
This is resulting in segmentation fault. Fix the same by typecasting to guint64
https://bugzilla.gnome.org/show_bug.cgi?id=755840
Push all pending events before pushing the gap. This ensures the
segment is pushed before the gap so it can be properly translated
to the running time
Includes unit test.
https://bugzilla.gnome.org/show_bug.cgi?id=753360
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
The padding (if any) is included in the length of the last packet, see
RFC 3550.
Section 6.4.1:
padding (P): 1 bit
If the padding bit is set, this individual RTCP packet contains
some additional padding octets at the end which are not part of
the control information but are included in the length field. The
last octet of the padding is a count of how many padding octets
should be ignored, including itself (it will be a multiple of
four).
Section A.2:
* The padding bit (P) should be zero for the first packet of a
compound RTCP packet because padding should only be applied, if it
is needed, to the last packet.
* The length fields of the individual RTCP packets must add up to
the overall length of the compound RTCP packet as received.
https://bugzilla.gnome.org/show_bug.cgi?id=751883
Add flags and enums to support multiview signalling in
GstVideoInfo and GstVideoFrame, and the caps serialisation and
deserialisation.
videoencoder: Copy multiview settings from reference input state
Add gst_video_multiview_* support API and GstVideoMultiviewMeta meta
https://bugzilla.gnome.org/show_bug.cgi?id=611157
According to this section of the rfc.
https://tools.ietf.org/html/rfc5506#section-3.4.2
The validation should be updated to accept more types of RTCP
packages, with this mask change feedback packages will be also
accepted.
Change-Id: If5ead59e03c7c60bbe45a9b09f3ff680e7fa4868
The original 0/1 framerate must still be allowed to be configured
on the upstream side of videorate, otherwise future caps renegotiation
is going to fail.
https://bugzilla.gnome.org/show_bug.cgi?id=750032
[API] gst_discoverer_info_to_variant
[API] gst_discoverer_info_from_variant
[API] GstDiscovererSerializeFlags
+ Serializes as a GVariant
+ Adds a test
+ Does not serialize potential GstToc (s)
https://bugzilla.gnome.org/show_bug.cgi?id=748814
A bitmask is 64 bits, but integer immediates are passed as int
in varargs, which happen to be 32 bit with high probability.
This triggered a valgrind jump-relies-on-uninitalized-value
report well away from the site, since it doesn't trigger on
stack accesses, and there must have been enough zeroes to stop
g_object_set at the right place.
Remove all the bus watch and main loop code from the block_deadlock
test, it's not needed: neither pipeline will ever post an EOS or ERROR
message on the bus, and we're the only ones posting an error, from a
timeout. Might just as well just sleep for a bit and then do whatever
we want to do.
Don't gratuitiously set tcase timeout, just use whatever is the
default (or set via the environment).
Make individual pipeline runs shorter.
Check for valgrind and only do a handful iterations when running
in valgrind, not 100 (each iteration takes about 4s on a core i7).
Make videotestsrc output smaller buffers than the default resolution,
we don't care about the buffer contents here anyway.
Fixes test timeouts when run in valgrind.
On slower systems, or under high system load (e.g. check-valgrind),
the sending_buffers_with_9_gstmemories test would sometimes fail,
because the read call only returns 32 bytes instead of the full
36 bytes expected. This is because multisocketsink might end up
doing a partial write of 32 bytes first, and then write the
missing 4 bytes later, but since we don't wait for all of data
to be written, there's a short window where our read call in the
unit test might then only receive the 32 bytes written so far,
which makes it deeply unhappy.
Instead, make sure we loop to read all bytes.
This test sets a rather short timeout, increase this when
we run under valgrind. Also add a short sleep to the
fakesrc ! fakesink pipeline to avoid thrashing the CPU,
which would often not stop the main loop when it should.
Also fix wrong (0.10) return value from pad probe callback.
In case upstream does not provide videorate with framerate information,
it will detect the current framerate from the buffer it received,
but if downstream forces the use of variable framerate (most probably
through the use of a caps filter with framerate = 0 / 1), videorate will
respect that.
And add some unit tests
https://bugzilla.gnome.org/show_bug.cgi?id=734424
When generating segment, we can't assume the first buffer is actually
the first expected one. If it's not, we need to adjust the segment to
start a bit before.
Additionally, we if don't know when the stream is suppose to have
started (no clock-base in caps), it means we need to keep everything in
running time and only rely on jitterbuffer to synchronize.
https://bugzilla.gnome.org/show_bug.cgi?id=635701
This provides notification that the socket in use was closed by the peer
and gives an opportunity to replace it with a new one which is not
closed, allowing reading from many sockets in order.
I use this in pulsevideo to implement reconnection logic to handle the
pulsevideo service dieing, such that is can be restarted without
disrupting downstream.
Fixes https://bugzilla.gnome.org/show_bug.cgi?id=739546
`socketsrc` can be considered a source counterpart to `multisocketsink`.
It can be considered a generalization of `tcpclientsrc` and
`tcpserversrc`: it contains all the logic required to communicate over
the socket but none of the logic for creating the sockets/establishing
the connection in the first place, allowing the user to accomplish this
externally in whatever manner they wish making it applicable to other
types of sockets besides TCP.
This commit essentially copies the implementation directly from
tcpserversrc. Later patches will tidy the implementation up and
re-implement `tcpclientsrc` and `tcpserversrc` in terms of `socketsrc`.
See https://bugzilla.gnome.org/show_bug.cgi?id=739546
If a buffer is made up of non-contiguous `GstMemory`s `gst_buffer_map`
has to copy all the data into a new `GstMemory` which is contiguous. By
mapping all the `GstMemory`s individually and then using scatter-gather
IO we avoid this situation.
This is a preparatory step for adding support to multisocketsink for
sending file descriptors, where a GstBuffer may be made up of several
`GstMemory`s, some of which are backed by a memfd or file, but I think this
patch is valid and useful on its own.
Fixes https://bugzilla.gnome.org/show_bug.cgi?id=746150
Should wait state change complete before start another state change.
Can't ensure can received async-done message when state change from PLAYING to PAUSED.
https://bugzilla.gnome.org/show_bug.cgi?id=736655
Don't feed 64-bit integer variable into vararg function that expects
an unsigned integer to go with GST_TAG_TRACK_NUMBER. This would
cause crashes on 32-bit platforms, and if not that then test
failures if the comparisons fail later (at least on big endian
platforms).
Test that a pipeline can change from PLAYING to PAUSED and back in
the following scenarios:
1. One track reach EOS after pushed some buffers while another track
still pushes buffers
2. One track reach EOS without buffers while another track still pushes
buffers
https://bugzilla.gnome.org/show_bug.cgi?id=736655
The flush stop could have happened between the source trying
to push the segment event and the buffer, this would cause a warning.
Prevent that by taking the source's stream lock while flushing.
https://bugzilla.gnome.org/show_bug.cgi?id=742684
With the current audiomixer, the input caps need to be the same,
otherwise there is an unavoidable race in the caps negotiation. So
enforce that using capsfilters
https://bugzilla.gnome.org/show_bug.cgi?id=742684
Allows subclasses to do custom caps query replies.
Also exposes the standard caps query handler so subclasses can just
extend on top of it instead of reimplementing the caps query proxying.
https://bugzilla.gnome.org/show_bug.cgi?id=741263
Refactor the encoder's caps query proxying function to a common place
and use it in the videodecoder to proxy downstream restrictions.
The new function is private to the gstvideo lib.
https://bugzilla.gnome.org/show_bug.cgi?id=741263
The set_format vfunc does not pass ownership of the caps
to the decoder, so we mustn't unref the caps there.
gst_event_new_caps() does not take ownership of the caps
passed, so we must unref the caps afterwards.
Fixes leaks when running test in valgrind in 1.4 branch.
Add test to check rendering of overlays of different sizes
that are completely or partially outside the video surface.
Once the overlay is blended to the video, verify if the
position of the blended overlay is as expected, by comparing
the pixels of the blended video with the expected values.
https://bugzilla.gnome.org/show_bug.cgi?id=739281
Make an ORC version of the 2x vertical upsampling code.
Improve unit tests, test chroma up and down sampling.
memset buffer in conversion to make valgrind happy.
There don't seem to be any unit tests for the socket handling elements. As
I am about to attempt some refactorings I've added some basic tests which
exercise some of the happy-paths in tcpclientsrc, tcpserversrc,
tcpserversink and tcpclientsink. They should let me know if I've caused
serious breakage.
They are far from exhaustive but are sufficient for me to have caught a few
memory-leaks in the existing code.
https://bugzilla.gnome.org/show_bug.cgi?id=739544
Combine multiplies in 4x filters.
Rename conversion functions to make them nicer in orc.
Add ORC versions for various downsampling algorithms
Add unit test chroma resampler
Make a more complete pack/unpack test, check if the image after
pack/unpack has the same color and precision, and has correctly
duplicated subsampled pixels.
Add a video scaler object build on top of the resampler. It has
implementation to deal with interlaced video as well as horizontal and
vertical scaling functions.
Move the conversion code used in videoconvert to the video library
and expose a simple but generic API to do arbitrary conversion. It can
currently do colorspace conversion but the plan is to add videoscale to
it as well.
See https://bugzilla.gnome.org/show_bug.cgi?id=732415
Adds a new test to textoverlay to make sure it can properly handle
elements that have ANY caps but fail to add the overlay meta in
the allocation query.
This test verifies that textoverlay won't use the caps features even
knowing that the overlay meta is accepted when querying the downstream
caps because it also needs downstream to confirm by putting the meta
in the allocation query.
https://bugzilla.gnome.org/show_bug.cgi?id=735800
Make textoverlay negotiate caps more correctly.
1) Check what caps we received in the video-sink
2) If it already has the overlay meta -> use it directly
3) If it doesn't, textoverlay try adding the overlay meta and using it,
if downstream doesn't support it, just use what is received in the
video-sink
4) Check if the allocation query also supports the meta to enable
really using it
Before it wasn't really doing renegotiation of any kind, just
re-checking if it should use the overlay meta or not
Also had to update the caps in the test as memory:SystemMemory seems
to be required when you use a caps feature otherwise intersection/subset
checks will fail.
https://bugzilla.gnome.org/show_bug.cgi?id=733916
Set up a fakesink with a pad probe to replace the missing encoder to detect
if encoding was really required and only error out in this case. Otherwise
just let passthrough branch work.
This delays the error posting from the set_state function to when buffers
are really flowing. Unit test updated accordingly
https://bugzilla.gnome.org/show_bug.cgi?id=650652
Make the MIKEY message and payload objects miniobjects so that they have
a GType and are refcounted.
We can reuse the dispose method to clear our payload objects.
Add some annotations.
Implement a copy function for the MIKEY message.
Fix the unit test.
Fixes https://bugzilla.gnome.org/show_bug.cgi?id=732589
With most decoder libraries, and especially when accessing codecs via
OpenMAX or similar APIs, we don't have the ability to properly related
the output buffers to a number of input samples. And could e.g. get
a fractional number of input buffers decoded at a time.
Previously this would in the end lead to an error message and stopped
playback. Change it to a warning message instead and try to handle it
gracefully. In theory the subclass can now get timestamp tracking
wrong if it completely misuses the API, but if on average it behaves
correct (and gst-omx and others do) it will continue to work properly.
Also add a test for the new behaviour.
We don't change it in the encoder yet as that requires more internal logic
changes AFAIU and I'm not aware of a case where this was a problem so far.
Aggregate buffering messages to only post the lower value
to avoid setting pipeline to playing while any multiqueue
is still buffering.
There are 3 scenarios where the entries should be removed from
the list:
1) When decodebin is set to READY
2) When an element posts a 100% buffering (already implemented)
3) When a multiqueue is removed from decodebin.
For item 3 we don't need to handle it because this should only
happen when either 1 is hapenning or when it is playing a
chained file, for which number 2 should have happened for the
previous stream to finish
https://bugzilla.gnome.org/show_bug.cgi?id=726423
The KEMAC payload actually needs to have subpayloads and the key should
go into the KEY_DATA subpayload. Add support for subpayloads and
implement the KEY_DATA payload.
Add some pointers to the conversion functions that allow us to add
encryption and decryption later.
MIKEY is defined in RFC 3830 and is used to exchange SRTP encryption
parameters between a sender and a receiver in a secure way.
This library implements a subset of the features, enough to implement
RFC 4567, using MIKEY in SDP and RTSP.
Check that even if the subclass doesn't call set_output_format, the base
class should use upstream provided caps to fill the output caps that is
pushed before the gap event is forwarded, otherwise it ends again fixating
the rate and channels to 1.
https://bugzilla.gnome.org/show_bug.cgi?id=722144
Adds a test that simulates a scenario where the first buffers after
a segment can't be decoded and the decoder asks for those frames
to be released. The videodecoder base class should make sure that
the events attached to those first buffers are pushed even if the
buffers aren't going to be.
https://bugzilla.gnome.org/show_bug.cgi?id=721835
Add a simple playback test that makes sure that video decoder pushes
buffers in the same order it receives and that it respects the
set timestamps and durations
With the latest GLib, g_source_remove() complains about not finding
the timeout source with the given ID here, since it was already
destroyed by returning FALSE from the timeout callback. Also return
FALSE from the bus watches when we don't want to be called any more.
Wait for thread to exit before starting to free the
to_push list, otherwise thread might check the final
to_push->next node only after we've freed it already.
We already have internally the information on what type of stream (audio,
video, container, subtitle, ...) a certain caps is.
Instead of forcing callers to specify which CODEC_TAG category a certain
caps is, use that information to make a smart choice.
Does not break previous behaviour of gst_pb_utils_add_codec_description_to_tag_list
(if tag is specified it will be used, if caps is invalid it will be rejected,
...).
https://bugzilla.gnome.org/show_bug.cgi?id=702215
The function gst_rtp_buffer_get_payload can not be used in Python
because it lacks necessary length parameter. This patch adds a new
function, gst_rtp_buffer_get_payload_bytes, to use from Python
bindings. The new function has the advisory "Rename to:" annotation
so it can replace the gst_rtp_buffer_get_payload whan creating
bindings.
The function gst_rtp_buffer_get_extension_bytes is also added. It wraps
gst_rtp_buffer_get_extension_data which doesn't work in Python due to
incomplete annotation and because it returns the length as number of
32-bit words.
https://bugzilla.gnome.org/show_bug.cgi?id=698562
The array of factories should not contain a NULL element at the end
since the number of arguments is determined via G_N_ELEMENTS and the
NULL will be used as an argument to gst_element_factory_make() if
the other sources in the list weren't usable.
This allows getting a pad for a specific encoding profile, which can
be useful when there are several stream profiles of the same type.
Also update the encodebin unit tests so that we check that the returned
pad has the right caps.
https://bugzilla.gnome.org/show_bug.cgi?id=689845
This reverts commit adc9694ed7.
No need to restrict the conversion, we can handle interlace correctly. We
basically unpack each field, then convert each field to the target colorspace
and pack and interleave each field to the target format. We also disable any
fast path that can't deal with interlaced formats.
We were setting the query-func on the sink-pad, which got overwritten when
adding the new pad to collect pads. Instead register our query-func with the
collect pads object. This fixes filter caps. Add a test for it.
These override the variants without version suffix. Makes
'make check' work properly in environments that set the
suffixed variant for 1.0, such as jhbuild.
jhbuild already sets $GST_PLUGIN_PATH_1_0 which overrides $GST_PLUGIN_PATH. Set
both for the tests to see the locally built elements. Fixes 'make check' in
jhbuild.
A return value of FALSE here indicates that we don't have control-values. In
0.10 we were returning the default value of the property. Now we don't fill an
array with defaults in the ControlBinding, but leave it up to the element to
handle this case.
The behaviour is sensibly changed here. Instead of purely falling when a
preset is set on the #GstEncodingProfile, we now make sure that the
element that is plugged corresponds to the one specified as preset. Then,
if we have a preset_name, we use it, if it fails, we fail (we might rather
just keep working even without setting the element properties?)
+ Add tests that it behave correctly
Enhance current code to prefer an exact match on sample depth if
possible. Also ignore GST_AUDIO_FORMAT_FLAG_UNPACK when checking
equality on the flags.
Allocate header, payload and padding in separate memory blocks in
gst_rtp_buffer_allocate().
don't use part of the payload data as storage for the extension data but store
it in a separate memory block that can be enlarged when needed.
Rework the one and two-byte header extension to make it reserve space for the
extra extension first.
Fix RTP unit test. Don't map the complete buffer or make assumptions on the
memory layout of the underlaying implementation. We can now always add extension
data because we have a separate memory block for it.
There won't be a tag messages on the bus, because tags
are now sent downstream for sinks to post on the bus,
and there's no sink involved here that would do that.
Secondly, the audio decoder base class only sends the
tags out once it has received some non-header data as
input, which is not something we're providing here.
Fix invalid memory access caused by broken pointer arithmetic.
If we have a uint16_t *tmpbuf and add n * dest->stride to it, we
skip twice as much as we intended to because dest->stride is in
bytes and not in pixels. This made us write beyond the end of
our allocated temp buffer, and made the unit test crash.
Use multifdsink for pipes instead of multisocketsink,
to avoid "creating GSocket from fd 9: Socket operation
on non-socket "criticals from Gio. Test still fails,
but it fails in a different way now.
Add support RTP buffers with multiple memory blocks. We allow one block for the
header, one for the extension data, N for data and one memory block for the
padding.
Remove the validate function, we validate now when we map because we need to
parse things in order to map multiple memory blocks.
Since we now use videoconvert, which supports these.
Unfortunately videoscale still crashes with 64-bit formats
right now because of a too small temp buffer, but I'm sure
someone is going to fix this real soon now, just like the
other unit tests.
Make the uri property getter return the next uri, like it was configured in the
setter.
Make a new current-uri and current-suburi property that reflects the currently
playing uri and suburi.
Fixes https://bugzilla.gnome.org/show_bug.cgi?id=676665
Video base classes and theora plugin still needs to be ported again
Conflicts:
docs/libs/gst-plugins-base-libs-docs.sgml
docs/libs/gst-plugins-base-libs-sections.txt
docs/libs/gst-plugins-base-libs.types
ext/theora/gsttheoradec.c
ext/theora/gsttheoradec.h
ext/theora/gsttheoraenc.c
ext/theora/gsttheoraenc.h
gst-libs/gst/video/Makefile.am
gst-libs/gst/video/video.c
gst-libs/gst/video/video.h
gst/playback/gsturidecodebin.c
tests/check/libs/video.c
tests/check/pipelines/theoraenc.c
win32/common/libgstvideo.def
... such as setting input caps, ensuring to unmap and bearing in
mind that gst_buffer_make_writable usually does not provide
a separate memory copy/area.
