Commit graph

25889 commits

Author SHA1 Message Date
Seungha Yang
a966cc20bc d3d11convert: Fix fallback texture copy
Fix texture copy when input texture has non-zero subresource index

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1275>
2020-05-19 17:25:05 +00:00
Seungha Yang
a24d4e654d d3d11: Add support for video rescale and rename element to d3d11convert
GstD3D11ColorConverter implementation is able to rescale video as well.
By doing colorspace conversion and rescale at once, we can save
one cycle of shader pipeline which will can save GPU resource.

Since this element can support color space conversion and rescale,
it's renamed as d3d11convert

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1275>
2020-05-19 17:25:05 +00:00
Seungha Yang
0e842c74b6 d3d11: Move scoring util method for colorspace conversion to colorconvert element
It's used only by colorconvert element.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1275>
2020-05-19 17:25:05 +00:00
Víctor Manuel Jáquez Leal
206d900573 codecs: h264decoder: chain finalize vmethod
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1238>
2020-05-19 16:57:09 +00:00
Nicolas Dufresne
540c287a73 codecparsers: h264: Only set relevant default weight values
This is minor optimization to avoid setting values we don't need. It also
makes debugging easier since only relevant values a non-zero now.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1238>
2020-05-19 16:57:09 +00:00
Nicolas Dufresne
d062677b3f codecparsers: h264: Fix default ref list size
The default in PPS was not applied properly. The default does not apply for
I-Slice and l1 default only applies for B-Slice.  This fixes the slice values
for num_ref_idx_l0_active_minus1 and num_ref_idx_l1_active_minus1.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1238>
2020-05-19 16:57:09 +00:00
Nicolas Dufresne
28e30c8493 codecs: h264decoder: Use calculated values for max_pic_num/frame_num
The parser pre-calculate these already, just use them.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1238>
2020-05-19 16:57:09 +00:00
Víctor Manuel Jáquez Leal
d2aae6bb96 codecs: h264decoder: ref pic lists as decode_slice parameters
Pass reference picture lists to decode_slice() vmethods

Change gstv4l2codech264dec and gstnvh264dec accordingly.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1238>
2020-05-19 16:57:09 +00:00
Víctor Manuel Jáquez Leal
990392d595 codecs: h264decoder: handle reference picture lists
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1238>
2020-05-19 16:57:08 +00:00
Nicolas Dufresne
7e63d1f5ad codecs: h264decoder: Port from GList to GArray
Using glist requires a lot of small allocation at runtime and also
it comes with a slow sort algorithm. As we play with that for very
frame and slices, use GArray instead. Note that we cache some arrays
in the instance as there is no support for stack allocated arrays
in GArray.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1238>
2020-05-19 16:57:08 +00:00
Nicolas Dufresne
f8736626a9 codecs: h264decoder: Make get_long_ref_by_pic_num() transfer none
We don't use the extra reference, so let's just avoid the extra
ref/unref.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1238>
2020-05-19 16:57:08 +00:00
Nicolas Dufresne
a20147f255 codecs: h264decoder: Make get_short_ref_by_pic_num() transfer none
We don't use the extra reference, so let's just avoid the extra
ref/unref.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1238>
2020-05-19 16:57:08 +00:00
Stéphane Cerveau
298e22e217 tests: fix nalutils file name
The filename was too long causing issues with ccache

Fix https://gitlab.freedesktop.org/gstreamer/gst-build/-/issues/97

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1281>
2020-05-19 15:39:50 +02:00
Jan Alexander Steffens (heftig)
055de3cdff mpegtsdemux: tests: Add simple tests for tsparse and tsdemux
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1274>
2020-05-18 14:11:40 +00:00
Jan Alexander Steffens (heftig)
9b2ed3a3fc mpegtsdemux: Close a buffer leak and simplify input_done
tsparse leaked input buffers quite badly:

    GST_TRACERS=leaks GST_DEBUG=GST_TRACER:9 gst-launch-1.0 audiotestsrc num-buffers=3 ! avenc_aac ! mpegtsmux ! tsparse ! fakesink

The input_done vfunc was passed the input buffer, which it had to
consume. For this reason, the base class takes a reference on the buffer
if and only if input_done is not NULL.

Before 34af8ed66a, input_done was used in
tsparse to pass on the input buffer on the "src" pad. That commit
changed the code to packetize for that pad as well and removed the use
of input_done.

Afterwards, 0d2e908523 set input_done
again in order to handle automatic alignment of the output buffers to
the input buffers. However, it ignored the provided buffer and did not
even unref it, causing a leak.

Since no code makes use of the buffer provided with input_done, just
remove the argument in order to simplify things a bit.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1274>
2020-05-18 14:11:40 +00:00
Mats Lindestam
bf004227ec gstcurlhttpsink: Set 'Expect: 100-continue'-header
In the upgrade of libcurl from 7.64.1 to 7.69.1 the
EXPECT_100_THRESHOLD has been increased from 1 Kb to 1 Mb
(see https://curl.haxx.se/mail/lib-2020-01/0050.html).
This caused the gstcurlhttpsink to not being able to rewind
and resend in the case, e.g. response '401 Unauthorized'.
Now the 'Expect: 100-continue'-header is explicitly set in
the gstcurlhttpsink.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1276>
2020-05-18 13:46:24 +02:00
Arun Raghavan
d0776faf47 opensles: Remove hard-coded buffer-/latency-time values
These were originally required in early Android versions, but are no
longer needed.
2020-05-16 19:23:06 +00:00
Seungha Yang
8afc283b51 mediafoundation: Refactor GstMFSourceObject implementation
* Move CoInitializeEx/CoUninitialize pair into thread function in order to
  ensure MTA COM thread
* Move common code to baseclass

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1269>
2020-05-15 09:03:13 +00:00
Seungha Yang
c29c71ae9d mediafoundation: Remove COM thread constraints from GstMFTransform object
Move CoInitializeEx/CoUninitialize pair into our dedicated thread so that
we can ensure COM thread is MTA. This will remove thread constraints
around plugin init.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1269>
2020-05-15 09:03:13 +00:00
Matthew Waters
b3cbdb1d26 amcvideodec: fix sync meta copying not taking a reference
Fixup for
9b9e39be24: amc: Fix crash when a sync_meta survives its sink
https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/603

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1272>
2020-05-15 13:54:56 +10:00
J. Kim
4ccaa1ebbb srtobject: add streamid property
The stream id starts with '#!::' according to SRT Access Control[1],
but GstURI requires URI encoded string.This commit introduces additional
property to set the id by normal string.

[1] https://github.com/Haivision/srt/blob/master/docs/AccessControl.md
2020-05-13 14:13:48 +00:00
Nirbheek Chauhan
e3d5849225 meson: Pass native: false to add_languages()
This is needed for cross-compiling without a build machine compiler
available. The option was added in 0.54, but we only need this in
Cerbero and it doesn't affect older versions so it should be ok.
Will only cause a spurious warning.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1266>
2020-05-13 13:59:36 +05:30
Alex Hoenig
0a2e026985 mpegtsmux: detect and ignore gap buffers
Fixes #1291.  Without this, when a stream has gaps and then resumes, the next buffer PTS that is written to the TS is given the PTS of the first gap.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1263>
2020-05-12 12:18:28 -04:00
Matthew Waters
6dae95d60f ccconverter: check fraction multiply for overflow
It should not happen and if it does, something went very wrong earlier

CID 1463350

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1262>
2020-05-12 16:06:32 +10:00
Matthew Waters
09cbe4cd03 ccconverter: tighten up a couple of NULL checks
CID 1463347
CID 1463346
CID 1463345

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1262>
2020-05-12 16:06:32 +10:00
Matthew Waters
ebc19d19bb ccconverter: fix unintialized read of mapped output info in error case
We only need to gst_buffer_unmap() if we have gst_buffer_map()ed.  In
most cases we can shorten the lenght of time we need to map the output
buffer.  Fix similar occurences elsewhere.

CID 1463349

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1262>
2020-05-12 16:06:32 +10:00
Matthew Waters
f077189809 ccconverter: fix uninitialized read in error case
CID 1463351

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1262>
2020-05-12 16:06:32 +10:00
Ting-Wei Lan
d10e2381c7 v4l2codecs: Fix compilation error on FreeBSD
This commit does the following things to fix compilation on FreeBSD:

1. Add required typedefs to linux/types-compat.h.
2. Remove unnecessary include linux/ioctl.h and replace linux/types.h
   with linux/types-compat.h. Both files do not exist on FreeBSD.
3. Check the header including makedev macro. FreeBSD does not have
   sys/sysmacros.h, and including it unconditionally causes error.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1259>
2020-05-11 15:17:31 +00:00
Matthew Waters
12edc0d9b8 ccconverter: implement discont handling
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1116>
2020-05-11 12:30:31 +00:00
Matthew Waters
ba1558a7ab ccconverter: use a better padding byte sequence for writing cdp
0xf8 can be interpreted as cea608 data at the beginning of a cdp packet
as the cc_valid bit is not checked when cc_valid in (0b00 or 0b01).

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1116>
2020-05-11 12:30:31 +00:00
Matthew Waters
7ed0bc539f ccconverter: split temporary storage into 3
Instead of storing the raw cc_data, store the 2 cea608 fields individually
as well as the ccp data.

Simply copying the input cc_data to the output cc_data violates a number of
requirements in the cea708 specification.  The most prominent being, that
cea608 triples must be placed at the beginning of each cdp.

We also need to comply with the framerate-dpendent limits for both the
cea608 and the ccp data which may involve splitting or merging some
cea608 data but not ccp data or vice versa.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1116>
2020-05-11 12:30:31 +00:00
Matthew Waters
3417a1709c ccconvert: compact input cc_data where possible
Skip over padding cc_data triples.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1116>
2020-05-11 12:30:31 +00:00
Matthew Waters
7d028af675 ccconverter: implement support for CDP framerate conversions
- Any format involving CDP is supported.
- Time codes (if present) are scaled as well.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1116>
2020-05-11 12:30:31 +00:00
Matthew Waters
ddc7563ac9 tests/ccconverter: test the time codes are successfully passed through
Where time codes are not stored in the caption data themselves

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1116>
2020-05-11 12:30:31 +00:00
Matthew Waters
75503017c2 ccconverter: introduce define for max cdp packet length
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1116>
2020-05-11 12:30:31 +00:00
Matthew Waters
44c874fd9e ccconverter: don't rely on external state in *_internal()
This allows using the _internal() variants for simply converting some
caption data without relying on any external state.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1116>
2020-05-11 12:30:31 +00:00
Matthew Waters
31a0bf367d ccconverter: cc_count limits are per framerate
Enforce this and add a test for cdp input being too large.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1116>
2020-05-11 12:30:31 +00:00
Matthew Waters
b25d75f201 ccconverter: refactor cdp id, fps, max_cc_count into a table
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1116>
2020-05-11 12:30:31 +00:00
Matthew Waters
afc120fa74 ccconverter: pivot to implementing generate_output
Will make a n-n buffer element much easier to implement.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1116>
2020-05-11 12:30:31 +00:00
Jan Alexander Steffens (heftig)
dc57fb7095 vulkan: Drop use of VK_RESULT_BEGIN_RANGE
This was removed in Vulkan 1.2.140.

> Shortly after 2020-04-24, we will be removing the automatically
> generated `VK_*_BEGIN_RANGE`, `VK_*_END_RANGE`, and `VK_*_RANGE_SIZE`
> tokens from the Vulkan headers. These tokens are currently defined for
> some enumerated types, but are explicitly not part of the Vulkan API.
> They existed only to support some Vulkan implementation internals,
> which no longer require them. We will be accepting comments on this
> topic in [#1230], but we strongly suggest any external projects using
> these tokens immediately migrate away from them.

[#1230]: https://github.com/KhronosGroup/Vulkan-Docs/issues/1230

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1255>
2020-05-11 12:04:08 +00:00
Sebastian Dröge
79e65951a9 audiobuffersplit: Perform discont tracking on running time
Otherwise we would have to drain on every segment event. Like this we
can handle segment events that don't cause a discontinuity in running
time to be handled without draining.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1254>
2020-05-11 07:25:39 +00:00
Sebastian Dröge
20756e3387 audiobuffersplit: Keep incoming and outgoing segments separate
We might have to drain already queued input based on the old segment
before forwarding the new segment event. The new segment is only
forwarded after a discont as otherwise we might cause unnecessary
timestamp jumps as we output buffers timestamped based on sample counts.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1254>
2020-05-11 07:25:39 +00:00
Chris Ayoup
9937101e51 webrtc: move filtering properties to webrtcice
We want webrtcbin to only expose properties that are defined in JSEP, so
these additional properties should be moved out.  In order to access
them, the webrtcice instance is exposed from webrtcbin.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1223>
2020-05-11 05:30:59 +00:00
Chris Ayoup
ca754245e9 webrtc: allow setting local IP addresses
If a local IP address is specified, ICE gathering can be much faster
in environments where there are multiple IP addreses but only some are
usable (for example, if you are running docker on the machine).

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1223>
2020-05-11 05:30:59 +00:00
Chris Ayoup
3fc8818824 webrtc: Allow toggling TCP and UDP candidates
Add some properties to allow TCP and UDP candidates to be toggled.  This
is useful in cases where someone is using this element in an environment
where it is known in advance whether a given transport will work or not
and will prevent wasting time generating and checking candidate pairs
that will not succeed.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1223>
2020-05-11 05:30:59 +00:00
Haihao Xiang
98600942e4 msdkvpp: clear the parameters after closing the session
Otherwise the stale values are used for the new process.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1159>
2020-05-11 02:16:44 +00:00
Sebastian Dröge
d6f6c51f3c spanplc: Don't segfault when retrieving the stats property without a spanplc context
For example when trying to get the property value in NULL state.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1258>
2020-05-10 08:44:09 +00:00
Sebastian Dröge
2a2e48fd9e onviftimestamp: Add missing break in set_property()
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1257>
2020-05-10 11:17:19 +03:00
Nicolas Dufresne
fac627ba1b test: h265parse: Test parsing buffer the ends with half a NAL header
This test cover the case where we are parsing, but our current buffers ends
with half the NAL header (which is 2 bytes in HEVC). Previously we would
throw an error message on the bus.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1251>
2020-05-07 14:17:28 -04:00
Nicolas Dufresne
90ca8850de h265parse: Ensure parsing ends on start-code + full header
The parser is used all over the place assuming that after calling
gst_h265_parser_identify_nalu(), the start-code found is can also be
identified. In H264 this works, because scan_for_start_code rely on
gst_byte_reader_masked_scan_uint32() that ensures that 1 byte passed the 3
bytes start code is found. But for HEVC, we need two bytes to identify the
following NAL.

This patch will return NO_NAL_END, even if a start code is found in the case
there was not enough bytes. This solution was chosen to maintain backward
compatibility, and reduce complexicity.

Fixes #1287

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1251>
2020-05-07 13:59:33 -04:00