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Author SHA1 Message Date
Matthew Waters
2bed220771 webrtc: don't generate duplicate rtx payloads when bundle-policy is set 
It was possible to generate a SDP that had an RTX payload type
that matched one of the media payload types when providing caps via
codec_preferences without any sink pads.

Fixes

m=video 9 UDP/TLS/RTP/SAVPF 96
...
a=rtpmap:96 VP8/90000
a=rtcp-fb:96 nack pli
a=fmtp:96 apt=96

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/2046>
 2021-03-09 02:22:35 +00:00
Ilya Kreymer
92626535c7 webrtc ice: Add 'min/max-rtp-port' props for setting RTP port range 
default min port == 0, max port == 65535 -- if min port == 0, uses existing random port selection (range ignored)
add 'gathering_started' flag to avoid changing ports after gathering has started
validity checks: min port <= max port enforced, error thrown otherwise
include tests to ensure port range is being utilized (by @hhardy)

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/119>
 2021-03-01 14:42:17 +00:00
Olivier Crête
3a3965e5cf webrtc ice: Only ever request one component, it's always rtcpmux 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/119>
 2021-03-01 14:42:16 +00:00
Matthew Waters
b6038523c1 webrtcbin: use regular ice nomination by default 
1. We don't currently deal with an a=ice-options in the SDP which means
   we currently violate https://tools.ietf.org/html/rfc5245#section-8.1.1
   which states: "If its peer is using ICE options (present in
   an ice-options attribute from the peer) that the agent does not
   understand, the agent MUST use a regular nomination algorithm."
2. The recommendation is default to regular nomination in both RFC5245
   and RFC8445.  libnice change for this is
   https://gitlab.freedesktop.org/libnice/libnice/-/merge_requests/125
   which requires an API break in libnice.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/2031>
 2021-03-01 10:00:06 +00:00
Víctor Manuel Jáquez Leal
771645e445 vulkan: Fix elements long name. 
Fix vkcoloconvert and vkviewconvert long names.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/2034>
 2021-02-24 20:15:52 +01:00
Stéphane Cerveau
5d4e45fe36 dtls: use GST_WARNING instead of g_warning 
No need a g_warning which is failing always
with gst-inspect -a

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/2010>
 2021-02-17 23:10:55 +00:00
Thibault Saunier
927bd289e5 openh264enc: Add support for main and high profiles 
Those are supported (to a certain extent) so we should not limit
ourself to baseline

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1789>
 2021-02-11 14:58:35 +00:00
Jakub Adam
9c00d261c3 srt: preserve ABI compatibility 
Reintroduce socket descriptor parameter removed in 327ad84e to
"caller-added" and "caller-removed" signals, just set it always to zero.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/2004>
 2021-02-03 23:39:00 +01:00
Jakub Adam
327ad84e35 srt: don't pass SRT socket ID to "caller-added,removed" signals 
The caller's IP and port is enough for unique identification. Don't leak
the socket handle since using it in unadvised libsrt calls from the
application could break the SRT element.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1772>
 2021-02-03 16:23:33 +00:00
Jakub Adam
4a58af4352 srtobject: add caller address to stats structure 
In listener mode, gst_stats() returns an independent set of
statistics for every connected caller. Having the caller's IP and port
present in each structure allows to correlate the statistics with a
particular caller that has been announced by "caller-added" signal.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1772>
 2021-02-03 16:23:33 +00:00
Arun Raghavan
a417a761fd ldac: Use pkg-config instead of raw lib/header search 
The ldacBT library includes pkg-config files for the standard and ABR
libraries, so let's just use that instead of doing a header/library
search.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1990>
 2021-01-27 17:16:57 -05:00
Haihua Hu
66788366a0 dashsink: add h265 codec support 
Return hvc1 for video/x-h265 mime type in mpd helper function

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1966>
 2021-01-26 17:47:53 +00:00
Marijn Suijten
e8bb0fa062 ext/ldac: Move duplicate sampling rates into #define 
Because there was a typo in one of the duplicates already (see previous
commit) it is much safer to specify these once and only once.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1985>
 2021-01-26 11:12:28 +01:00
Marijn Suijten
3747fdb1a6 ext/ldac: Fix typo in 88200(0) stereo encoder sampling rate 
Fixes: a57681455 ("ext: Add LDAC encoder")
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1985>
 2021-01-26 11:02:21 +01:00
Matthew Waters
3ed0ee95f2 wpesrc: fix possible small deadlock on shutdown 
Problem is that unreffing the EGLImage/SHM Buffer while holding the
images_mutex lock may deadlock when a new buffer is advertised and
an attempt is made to lock the images_mutex there.

The advertisement of the new image/buffer is performed in the
WPEContextThread and the blocking dispatch when unreffing wants to run
something on the WPEContextThread however images_mutex has already been
locked by the destructor.

Delay unreffing images/buffers outside of images_mutex and instead just
clear the relevant fields within the lock.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1843>
 2021-01-25 09:15:28 +00:00
Haihua Hu
1753d2931c dashsink: fix double unref of sinkpad caps 
no need to unref caps in gst_mpd_helper_get_XXX_codec_from_mime
it will be unref in caller gst_dash_sink_get_stream_metadata()

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1981>
 2021-01-25 10:28:45 +08:00
Matthew Waters
1a53dfbd64 ldac: also look for the ldac/ldacBT.h header. 
Otherwise there will be a scenario where the library can be found but
not the header and a compilation build error will result

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1975>
 2021-01-22 09:32:51 +00:00
Mathieu Duponchelle
86c009e7aa webrtc: expose transport property on sender and receiver 
As advised by !1366#note_629558 , the nice transport should be
accessed through:

> transceiver->sender/receiver->transport/rtcp_transport->icetransport

All the objects on the path can be accessed through properties
except sender/receiver->transport. This patch addresses that.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1952>
 2021-01-13 19:22:42 +00:00
Matthew Waters
94fea694bc wpesrc: replace object lock usage with a new lock 
Using the object lock is problematic for anything that can dispatch to
another thread which is what createWPEView() does inside
gst_wpe_src_start().  Using the object lock there can cause a deadlock.

One example of such a deadlock is when createWPEView is called, but
another (or the same) wpesrc is on the WPEContextThread and e.g. posts a
bus message.  This message propagations takes and releases the object
lock of numerous elements in quick succession for determining various
information about the elements in the bin.  If the object lock is
already held, then the message propagation will block and stall bin
processing (state changes, other messages) and wpe servicing any events.

Fixes https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/issues/1490

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1934>
 2021-01-12 08:35:10 +00:00
Mathieu Duponchelle
88e007fb21 webrtcbin: try harder not to pick duplicate media ids 
On renegotiation, or when the user has specified a mid for
a transceiver, we need to avoid picking a duplicate mid for
a transceiver that doesn't yet have one.

Also assign the mid we created to the transceiver, that doesn't
fix a specific bug but seems to make sense to me.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1902>
 2021-01-08 20:22:57 +00:00
Edward Hervey
4e7f7871db srt: Define options added in later revisions 
Allows compiling the plugin against old headers.

For SRTO_BINDTODEVICE there's nothing we can do, since the value depends on
configuration options of the library. Nice.

Fixes build with libsrt < 1.4.2

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1945>
 2021-01-07 09:23:28 +01:00
Jakub Adam
6c35222973 srtobject: distinguish authentication error messages 
Use GST_RESOURCE_ERROR_NOT_AUTHORIZED code in posted error messages
related to SRT authentication (e.g. incorrect or missing password) so
that the application can recognize them more easily.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1943>
 2021-01-06 23:35:20 +00:00
Jakub Adam
ef118f3d0a srtobject: detect socket errors from srt_epoll_wait() 
On an error event, epoll wait puts the failed socket in both readfds and
writefds. We can take advantage of this and avoid explicitly checking
socket state before every read or write attempt.

In addition, srt_getrejectreason() will give us more detailed
description of the connection failure.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1943>
 2021-01-06 23:35:20 +00:00
Olivier Crête
df8d29e9c3 webrtcbin: Remove remnant of non-rtcp-mux mode 
There was some code left that wasn't used anymore.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1930>
 2021-01-06 23:02:37 +00:00
Jakub Adam
3c3e89304e srtobject: make possible to specify more sockopts in SRT URI 
Any socket option that can be passed to libsrt's srt-live-transmit
through SRT URI query string is now recognized.

Also make the code that applies options to SRT sockets more generic.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1842>
 2021-01-06 22:28:02 +00:00
Jakub Adam
5687b03438 srtsrc: fix typos 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1541>
 2021-01-06 19:21:14 +00:00
Jakub Adam
1e461b3166 srtsink: remove unused connection_mode variable 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1541>
 2021-01-06 19:21:14 +00:00
Jakub Adam
d540012091 srtobject: obey "wait-for-connection" in caller mode 
The pipeline now gets stuck in gst_srt_object_write_one() until the
receiver comes online, which may or may not be desired based on the use
case.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1836>
 2021-01-06 18:55:37 +00:00
Jakub Adam
00e44e8ed7 srtobject: post a message on the bus when broken socket is detected 
So that the application gets notified may react to it.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1935>
 2021-01-05 16:50:01 +00:00
Raghavendra
08b1485862 srt: Add authentication to srtsink and srtsrc elements 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1725>
 2021-01-04 00:03:47 +05:30
Haihua Hu
a4a532c092 dashsink: fix critical log when exit dynamic pipeline 
availability-start-time and publish-time shared the same
GstDateTime object, this object will be unref twice and
cause reference count issue. Should use g_value_dup_boxed()
to copy this object.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1928>
 2020-12-31 10:34:50 +08:00
Olivier Crête
51ef4557b5 webrtcstats: PLI/FIR/NACK direction are the opposite of the media 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1924>
 2020-12-29 15:07:03 -05:00
Sebastian Dröge
b258144c16 assrender: Don't try unlocking unlocked mutex 
When flushing right at the beginning of the video chain function or
when failing negotiation at the top of the function, the assrender mutex
would be unlocked without being previously locked.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1918>
 2020-12-29 11:19:53 +00:00
Arun Raghavan
2a5d564de3 openaptx: Drop lib prefix from option name for consistency 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1876>
 2020-12-11 22:08:01 -05:00
Igor Kovalenko
b916522382 openaptx: add aptX and aptX-HD codecs using libopenaptx 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1871>
 2020-12-11 11:55:54 +03:00
Philippe Normand
3bcb876c29 wpe: Emit load-progress messages 
The estimated-load-progress value can be used on application side to display a
progress bar for instance.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1710>
 2020-12-09 17:31:51 +00:00
Jan Alexander Steffens (heftig)
470e6989d2 srt: Don't take object lock calling gst_srt_object_get_stats 
This function takes the sock lock. This can result in a deadlock when
another thread holding the sock lock is trying to take the object lock.

Thread A (Holds object lock, wants sock lock):

    #2  gst_srt_object_get_stats at gst-plugins-bad/ext/srt/gstsrtobject.c:1753
    #3  gst_srt_object_get_property_helper at gst-plugins-bad/ext/srt/gstsrtobject.c:409
    #4  gst_srt_sink_get_property at gst-plugins-bad/ext/srt/gstsrtsink.c:95
    #5  g_object_get_property from libgobject-2.0.so.0

Thread B (Holds sock lock, wants object lock):

    #2  gst_element_post_message_default at gstreamer/gst/gstelement.c:2069
    #3  gst_element_post_message at gstreamer/gst/gstelement.c:2123
    #4  gst_element_message_full_with_details at gstreamer/gst/gstelement.c:2259
    #5  gst_element_message_full at gstreamer/gst/gstelement.c:2298
    #6  gst_srt_object_send_headers at gst-plugins-bad/ext/srt/gstsrtobject.c:1407
    #7  gst_srt_object_send_headers at gst-plugins-bad/ext/srt/gstsrtobject.c:1444
    #8  gst_srt_object_write_to_callers at gst-plugins-bad/ext/srt/gstsrtobject.c:1444
    #9  gst_srt_object_write at gst-plugins-bad/ext/srt/gstsrtobject.c:1598
    #10 gst_srt_sink_render at gst-plugins-bad/ext/srt/gstsrtsink.c:179

Fixes d2d00e07ac.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1861>
 2020-12-07 17:59:09 +00:00
Sebastian Dröge
0243afcb9d ccconverter: Add property to specify which sections to include in CDP packets 
Various software, including ffmpeg's Decklink support, fails parsing CDP
packets that contain anything but CC data in the CDP packets.

Based on this property, timecodes are not written into the CDP packets
even if they're present.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1833>
 2020-12-07 19:23:42 +02:00
Sebastian Dröge
b6debae2c0 ccconverter: Refactor code to only retrieve the timecode meta once 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1833>
 2020-12-07 09:40:52 +00:00
Edward Hervey
d137171f03 opencv: Expose retinex parameters 
Makes the plugin a tad more useful :)

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1845>
 2020-12-03 17:04:07 +01:00
Edward Hervey
339ad46b93 hlsdemux: Use actual object for logging 
i.e. the pad of the stream

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1853>
 2020-12-03 14:31:17 +00:00
Arun Raghavan
81abc4c825 curl: Remove incorrect GST_DEBUG_OBJECT() calls 
klass is not a GstObject, and these debugs print should likely not be
around anyway.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1851>
 2020-12-03 13:31:38 +00:00
Thibault Saunier
f1cf5d0683 hlssink2: Mark as Muxer 
The way it is usable by encodebin2. This is what splitmux does already.
 2020-11-30 15:16:01 -03:00
Thibault Saunier
d608636327 qroverlay: Reuse the same OverlayComposition object when possible 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1829>
 2020-11-26 14:34:34 +00:00
Thibault Saunier
ad5f812c91 qroverlay: Rework basing it on overlaycomposition 
The base class is now a bin which wraps the `overlaycomposition`
element and implements the `draw` signal.

This way we support all the video formats the GstVideoOverlayComposition
API supports and the blending code can be reused. It is also possible
to have the blending happen in the sinks now.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1829>
 2020-11-26 14:34:34 +00:00
Olivier Crête
a801018ef1 webrtc: Make ssrc map into separate data structures 
They now contain a weak reference and that could be freed later
causing strange crashes as GWeakRef are not movable.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1766>
 2020-11-24 04:27:52 +00:00
Olivier Crête
1deb034e3d webrtcstats: Get the remote-inbound stats from the right RTPSource 
This also means that we need to get the clock-rate from the codec instead
of from the RTPSource, as the remote one doesn't include a clock rate.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1766>
 2020-11-24 04:27:52 +00:00
Olivier Crête
1c1661b54f webrtcbin: Implement getting stats for a specific pad 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1766>
 2020-11-24 04:27:52 +00:00
Olivier Crête
23ea950351 webrtcstats: Also return the raw rtpsource stats for more information 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1766>
 2020-11-24 04:27:52 +00:00
Olivier Crête
b895240241 webrtcstats: Avoid copy of GstStructure 
Instead transfer the ownership to the new structure

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1766>
 2020-11-24 04:27:52 +00:00
Olivier Crête
a46c6e3a97 webrtcstats: Remove receiver side when sending 
Those are just invalid and just reflect what we sent. We'd need to parse the
RTCP XR packets from the other side to know more about those.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1766>
 2020-11-24 04:27:52 +00:00
Olivier Crête
ba0dfa52d2 webrtcstats: Extract statistics from the rtpjitterbuffer 
And expose them as standardised webrtc statistics

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1766>
 2020-11-24 04:27:52 +00:00
Olivier Crête
fc0f6db856 webrtcbin: Store the rtpjitterbuffer instances to extract stats from them 
Store them as web refs to avoid having to worry about freeing later and because
the new-jitterbuffer is on a different thread

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1766>
 2020-11-24 04:27:52 +00:00
Olivier Crête
d9d7814182 webrtcstats: Document all RTP missing fields according to the latest spec 
Just document all the missing fields and document which ones will never
be implemented because they depend on the codec or depayloader

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1766>
 2020-11-24 04:27:52 +00:00
Olivier Crête
895ea210c2 webrtcstats: RTCP computed RTT is only available at sender 
The receiver doesn't have the information to compute it.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1766>
 2020-11-24 04:27:52 +00:00
Olivier Crête
a5c3331197 webrtcstats: Remove redundant lines 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1766>
 2020-11-24 04:27:52 +00:00
Olivier Crête
5d5417f271 webrtc: Remove non rtcp-mux code 
RTCP mux is now always required by the WebRTC spec

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1765>
 2020-11-24 01:59:55 +00:00
Tim-Philipp Müller
470c79be61 qroverlay: unset executable flag on source files 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1824>
 2020-11-20 13:24:24 +00:00
Tim-Philipp Müller
53947cad29 qroverlay: fix auto detection of json-glib for plugin 
Only want to check for json-glib when libqrencode was found,
but also it shouldn't be required but depend on the option.

Fixes #1465

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1824>
 2020-11-20 13:22:48 +00:00
Olivier Crête
03d710bd40 openh264dec: Accept constrained-high and progressive-high profiles 
They're just subsets of the high profile.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1634>
 2020-11-18 15:47:36 -05:00
Jan Schmidt
92472ef088 wpe: Don't crash when running on X11. 
Don't assume the available EGL display is a wayland display -
instead, check the the GStreamer GL context is EGL, and then
use gst_gl_display_egl_from_gl_display to create a
GstGLDisplayEGL from that, which also adds refcounting
around the underlying EGLDisplay.

Fixes https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/issues/1385

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1752>
 2020-11-15 15:27:08 +00:00
Aaron Boxer
330b2c6b7c openjpegenc: store stripe offset when encoding image 
The decoder can simply read this offset after decoding
to know where to blit the stripe to the full frame

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1800>
 2020-11-12 13:53:47 +00:00
Aaron Boxer
af87da86e2 openjpegenc: take subsampling into account when calculating stripe height 
We calculate minimum of (stripe height * sub sampling) across all components
to ensure that all component dimensions are consistent with sub-sampling.
The last stripe for each component is simply the remaining height.

limit wavelet resolutions for "thin" stripes

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1800>
 2020-11-12 13:53:47 +00:00
Stéphane Cerveau
4f6b609558 openjpegenc: fix memory leak from mstream 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1800>
 2020-11-12 13:53:47 +00:00
Aaron Boxer
7463e1090c openjpegenc: fail negotation in handle_frame if alignment mismatch 
If encoder is in stripe mode, then downstream must also support stripe

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1800>
 2020-11-12 13:53:47 +00:00
Edward Hervey
dd425fe0fd adaptivedemux: Store QoS values on the element 
Storing it per-stream requires taking the manifest lock which can apparenly be
hold for aeons. And since the QoS event comes from the video rendering thread
we *really* do not want to do that.

Storing it as-is in the element is fine, the important part is knowing the
earliest time downstream.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1021>
 2020-11-11 20:18:11 +00:00
Edward Hervey
a4b20ed276 hlsdemux: Don't double-free variant streams on errors 
If an error happened switching to a new variant, we switch back to the previous
one ... except it will be unreffed when settin git.

In order to avoid such issues, keep a reference to the old variant until we're
sure we don't need it anymore

Fixes cases of double-free on variants and its contents

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1799>
 2020-11-11 19:19:28 +00:00
Sanchayan Maity
a576814553 ext: Add LDAC encoder 
LDAC is an audio coding technology developed by Sony that enables the
transmission of High-Resolution (Hi-Res) audio contents over Bluetooth.

Currently Adaptive Bit Rate (ABR) as supported by libldac encoder is not
implemented.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1621>
 2020-11-11 22:16:43 +05:30
Raul Tambre
6d300ce785 webrtc: Update libnice version requirement to 0.1.17 
Since !1366 nice_agent_get_sockets() is used, which requires 0.1.17.
Update the version requirement accordingly.

Fixes #1459.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1792>
 2020-11-11 13:41:59 +02:00
Edward Hervey
1246b448ee hlsdemux: Re-use streams if possible 
When switching variants, try to re-use existing streams/pads instead of creating
new ones. When dealing with urisourcebin and decodebin3 this is not only the
expected way but also avoids a lot of buffering/hang issues.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1757>
 2020-11-11 04:06:05 +00:00
Edward Hervey
f1fdbfc5bd m3u8: Make a debug function usable elsewhere 
The rest of the code might want to use this

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1757>
 2020-11-11 04:06:05 +00:00
Thibault Saunier
3a554f6d62 qroverlay: Generate documentation 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1730>
 2020-11-11 00:18:32 +00:00
Thibault Saunier
e7ec9986ca qroverlay: Add a qroverlay element that allows overlaying any data 
This moves `gstqroverlay.c` to `gstdebugqroverlay.c` and implements
a simple `gstqroverlay` element.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1730>
 2020-11-11 00:18:32 +00:00
Thibault Saunier
7d9f125ab1 qroverlay: Rename qroverlay to debugqroverlay 
The element is specially focus on debugging purposes and not a generique QR overlay

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1730>
 2020-11-11 00:18:32 +00:00
Thibault Saunier
964c8aa5e0 qroverlay: Factor out qroverlay logic to a base class 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1730>
 2020-11-11 00:18:32 +00:00
Thibault Saunier
4afa2d2c3d qroverlay: Factor out qroverlay logic to a base class 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1730>
 2020-11-11 00:18:32 +00:00
Thibault Saunier
e6881aefa9 qroverlay: Make subclassable 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1730>
 2020-11-11 00:18:32 +00:00
Thibault Saunier
c307f1418d qroverlay: Port to VideoFilter 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1730>
 2020-11-11 00:18:32 +00:00
Thibault Saunier
4a259e8f28 qroverlay: Make default pizel-size 3 
Otherwise zbar isn't able to read the produced qrcodes

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1730>
 2020-11-11 00:18:32 +00:00
Thibault Saunier
dc88d3ece9 qroverlay: Cleanup the way we build the json using json-glib 
And reindent the .h file removing tabs

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1730>
 2020-11-11 00:18:32 +00:00
Thibault Saunier
bf811616e0 qroverlay: Fix copyright 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1730>
 2020-11-11 00:18:32 +00:00
Thibault Saunier
3db26e3f96 qroverlay: Fix some warnings 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1730>
 2020-11-11 00:18:32 +00:00
Thibault Saunier
c743d5db23 qroverlay: Minor renaming and documentation fixes 
Matching usual namings

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1730>
 2020-11-11 00:18:32 +00:00
Thibault Saunier
af054f7015 qroverlay: Import from gst-qroverlay 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1730>
 2020-11-11 00:18:32 +00:00
Olivier Crête
da2bd55177 webrtc: Add properties to change the socket buffer sizes to ice object 
libnice doesn't touch the kernel buffer sizes. When dealing with RTP data,
it's generally advisable to increase them to avoid dropping packets locally.
This is especially important when running multiple higher bitrate streams at
the same time.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1366>
 2020-11-03 22:07:53 +00:00
Jan Schmidt
5f9897745b vkdeviceprovider: Avoid deadlock on physical device 
Don't hold the object lock on the vk physical device while
constructing a GstVulkanDevice around it, as
GstVulkanDevice can make calls on the physical device that
require the object lock.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1754>
 2020-11-03 04:28:00 +00:00
Randy Li
4e4c26af4b wlvideoformat: fix DMA format convertor 
In the most of case, this typo would work. But for
ARGB8888 and XRGB8888, which shm format is not based on fourcc,
which would never appear in format enumeration.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1751>
 2020-11-02 12:39:38 +00:00
Jan Schmidt
dbeb576531 sctp: Do downward state change logic after chaining up. 
Call the parent state_change function first when changing state
downward, to make sure that the element has stopped before cleaning
it up.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1741>
 2020-10-31 01:47:06 +00:00
Jan Schmidt
760592a29c dtls: Avoid bio_buffer assertion on shutdown. 
On shutdown, a previous iteration of dtsl_connection_process()
might be incomplete and leave a partial bio_buffer behind.

If the DTLS connection is already marked closed, drop out
of dtls_connection_process early without asserting.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1741>
 2020-10-31 01:47:06 +00:00
Jan Schmidt
af90778314 webrtc: Fix a race on shutdown. 
The main context can disappear in gst_webrtc_bin_enqueue_task()
between checking the is_closed flag and enqueueing a source on the
main context. Protect the main context with the object lock instead
of the PC lock, and hold a ref briefly to make sure it stays alive.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1741>
 2020-10-31 01:47:06 +00:00
Olivier Crête
80a56c25a6 webrtc: Set the DSCP markings based on the priority 
This matches how the WebRTC javascript API works and the Chrome implementation.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1707>
 2020-10-30 16:24:40 -04:00
Olivier Crête
0fbbdc5734 rtptransceiver: Store the SSRC of the current stream 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1707>
 2020-10-30 16:23:10 -04:00
Olivier Crête
7be09a5f22 webrtc: Save the media kind in the transceiver 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1707>
 2020-10-30 16:23:10 -04:00
Olivier Crête
e172ca5be1 webrtcbin: Remove unused function 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1707>
 2020-10-30 16:23:10 -04:00
Chris Bass
d25be0d16e ttmlparse: Handle whitespace before XML declaration 
When ttmlparse is in, e.g., an MPEG-DASH pipeline, there may be
whitespace between successive TTML documents in ttmlparse's accumulated
input. As libxml2 will fail to parse documents that have whitespace
before the opening XML declaration, ensure that any preceding whitespace
is not passed to libxml2.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1539>
 2020-10-30 07:01:52 +00:00
Chris Bass
8df2314c23 ttmlparse: Ensure only single TTML doc parsed 
The parser handles only one TTML file at a time, therefore if there are
multiple TTML documets in the input, parse only the first.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1539>
 2020-10-30 07:01:52 +00:00
Guillaume Desmottes
bfb9071081 isac: add iSAC plugin 
Wrapper on the iSAC reference encoder and decoder from webrtc,
see https://en.wikipedia.org/wiki/Internet_Speech_Audio_Codec

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1124>
 2020-10-29 16:59:18 +01:00
Randy Li
6d8133e41e waylandsink: release frame callback when destroyed 
We would use a frame callback from the surface to indicate
that last buffer is rendered, but when we destroy the surface
and that callback is not back yet, it may cause the wayland event
queue crash.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1729>
 2020-10-29 12:09:01 +00:00
Nicola Murino
77f28ee3e7 opencv: allow compilation against 4.5.x 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1709>
 2020-10-27 10:53:27 +00:00
Philippe Normand
37b7809d18 wpe: Convert launch lines to markdown and move since tag 
Seems like the examples don't appear in the generated docs because the Since tag
was badly positioned in the doc blurb.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1706>
 2020-10-18 15:17:25 +00:00
Tim-Philipp Müller
3f8d33abed hlssink2: fix and flesh out docs 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1699>
 2020-10-16 13:37:18 +00:00
Stéphane Cerveau
cbd61e28b2 meson: update glib minimum version to 2.56 
In order to support the symbol g_enum_to_string in various
project using GStreamer ( gst-validate etc.), the glib minimum
version should be 2.56.0.

Remove compat code as glib requirement
is now > 2.56

Version used by Ubuntu 18.04 LTS

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1695>
 2020-10-16 09:16:34 +00:00
Vivia Nikolaidou
94e1623434 cameracalibrate: Improve gst-inspect documentation 
Thanks to @kazz_naka on Twitter

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1691>
 2020-10-13 17:21:59 +03:00
Matthew Waters
2f29a4cde6 wpesrc: add some debug logging around WPEView creation/destruction 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1663>
 2020-10-13 08:48:05 +00:00
Matthew Waters
da18a8d93d wpesrc: fix a memory leak of the bytes 
free the previous GBytes if load-bytes is called multiple times
before view creation.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1663>
 2020-10-13 08:48:05 +00:00
Matthew Waters
356fee4dd6 wpesrc: only create webview if not already created 
e.g. _decide_allocation() can be called multiple times throughout the
element's lifetime and we only want to create the view once rather than
overwriting.

Fixes a leak of the WPEView under certain circumstances.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1663>
 2020-10-13 08:48:05 +00:00
Matthew Waters
b84c8821de wpe: free a previous pending image/shm buffer 
Don't blindly overwrite a possibly previously set buffer.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1663>
 2020-10-13 08:48:05 +00:00
Jan Alexander Steffens (heftig)
4eeff95f92 srtsrc: Prevent delay from being negative 
`delay` should be a GstClockTimeDiff since SRT time is int64_t.

All values are in local time so we should never see a srctime that's in
the future. If we do, clamp the delay to 0 and warn about it.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1674>
 2020-10-12 12:58:22 +00:00
Jan Alexander Steffens (heftig)
ec11ad9d55 srtsrc: Don't calculate a delay if the srctime is 0 
A zero srctime is a missing srctime. Apparently this can happen when
["the connection is not between SRT peers or if Timestamp-Based Packet
Delivery mode (TSBPDMODE) is not enabled"][1] so it may not apply to us,
but it's best to be defensive.

[1]: https://github.com/Haivision/srt/blob/v1.4.2/docs/API.md#sending-and-receiving

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1674>
 2020-10-12 12:58:22 +00:00
Jan Alexander Steffens (heftig)
6b2fcb52e5 srtsrc: Defend against missing clock 
If we don't have a clock, stop the source instead of asserting in
gst_clock_get_time. This can happen when the element is removed from the
pipeline while it's playing.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1674>
 2020-10-12 12:58:22 +00:00
Olivier Crête
8a0d1d85cf dtlsconnection: Ignore OpenSSL system call errors 
OpenSSL shouldn't be making real system calls, so we can safely
ignore syscall errors. System interactions should happen through
our BIO. So especially don't look at the system's errno, as it
should be meaningless.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1656>
 2020-10-10 15:34:21 +00:00
Jan Alexander Steffens (heftig)
c6eeead1e4 srt: Consume the error from gst_srt_object_write 
Instead of leaking it.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1668>
 2020-10-09 07:47:47 +00:00
Jan Alexander Steffens (heftig)
2a7fa67693 srt: Check socket state before retrieving payload size 
The connection might be broken, which we should detect instead of just
aborting the write.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1669>
 2020-10-09 07:12:04 +00:00
Jakub Adam
6f2f15b5fb x265enc: fix deadlock on reconfig 
Don't attempt to obtain encoder lock that is already held by
gst_x265_enc_encode_frame().

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1671>
 2020-10-09 06:39:36 +00:00
Edward Hervey
dd11e91c3b srtsrc: Fix timestamping 
SRT provides the original timestamp of a packet (with drift/skew corrected for
local clock), which is what should be used for timestamping the outgoing
buffers. This ensures that we output the packets with the same timestamp (and by
extension rate) as the original feed.

Also detect if packets were dropped (by checking the sequence number) and
properly set DISCONT flag on the outgoing buffer.

Finally answer the latency queries

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1658>
 2020-10-08 21:12:17 +00:00
Sebastian Dröge
cc7e98816f Revert "webrtc: Save the media kind in the transceiver" 
This reverts commit f54d8e9945.

It breaks the CI until the C# bindings are fixed.
 2020-10-08 18:53:12 +03:00
Sebastian Dröge
849839ba97 Revert "rtptransceiver: Store the SSRC of the current stream" 
This reverts commit d1da271f25.

It breaks the CI until the C# bindings are fixed.
 2020-10-08 18:53:07 +03:00
Sebastian Dröge
e65a8cbcf1 Revert "webrtcbin: Remove unused function" 
This reverts commit 39723dbe93.

It breaks the CI until the C# bindings are fixed.
 2020-10-08 18:53:04 +03:00
Sebastian Dröge
b565a7ef66 Revert "webrtc: Set the DSCP markings based on the priority" 
This reverts commit 8ba08598bb.

It breaks the CI until the C# bindings are fixed.
 2020-10-08 18:53:00 +03:00
Olivier Crête
8ba08598bb webrtc: Set the DSCP markings based on the priority 
This matches how the WebRTC javascript API works and the Chrome implementation.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1425>
 2020-10-06 16:49:08 -04:00
Olivier Crête
39723dbe93 webrtcbin: Remove unused function 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1425>
 2020-10-06 16:49:08 -04:00
Olivier Crête
d1da271f25 rtptransceiver: Store the SSRC of the current stream 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1425>
 2020-10-06 16:49:08 -04:00
Olivier Crête
f54d8e9945 webrtc: Save the media kind in the transceiver 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1425>
 2020-10-06 16:49:08 -04:00
Jan Alexander Steffens (heftig)
92dc2f4192 srt: Remove unused sa_family tracking 
Now that SRT no longer needs the family when creating the socket, this
code has become useless.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1659>
 2020-10-06 13:56:32 +02:00
Niklas Hambüchen
13c8bda531 srt: Move off deprecated srt_socket(). 
See 73ee1e1a3e/docs/API-functions.md (srt_socket)

`srt_create_socket()` was added in
4b897ba92d
and srt `v1.3.0` is the first release that has it.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1659>
 2020-10-06 13:56:32 +02:00
Jan Alexander Steffens (heftig)
4e26b447f6 srt: Register a log handler 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1659>
 2020-10-06 12:39:04 +02:00
Jan Alexander Steffens (heftig)
936f422764 srt: Avoid removing invalid sockets from the polls 
This would provoke error messages from SRT.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1659>
 2020-10-06 12:39:00 +02:00
Jan Alexander Steffens (heftig)
fda4cfd15e srt: Fix use of srt_startup 
`srt_startup` can also return 1 if it was successful. Avoid warning in
this case.

Avoid a race when checking whether we need to call it at all.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1659>
 2020-10-06 12:38:57 +02:00
Jan Alexander Steffens (heftig)
6b8c4a5f34 srt: Fix parameter types used for socket options 
The [SRT documentation][1] specifies exact types for the socket options.
Make sure we match these.

This reverts the linger workaround in commit 84f8dbd932
and extends srt_constant_params to support other types than int.

[1]: https://github.com/Haivision/srt/blob/master/docs/APISocketOptions.md

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1659>
 2020-10-06 12:36:40 +02:00
Lars Lundqvist
9ded00bcf0 curlbasesink: Add curl seek callback 
Adding functionality to handle SEEK_SET enables rewinding of sent data.
In the HTTP case, this happens after an HTTP 401 has been received from
the other end. This will result in the sent data being resent.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1616>
 2020-10-01 10:40:14 +00:00
Matthew Waters
b003387526 wpesrc: fix some caps leaks using the non-GL output 
Always chain up to the parent _stop() implementation as it unrefs some
caps (among other things).

Fixes: https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/issues/1409
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1618>
 2020-09-30 12:10:44 +00:00
Hosang Lee
f7a8ece5ef smoothstreaming: clear live adapter on seek 
In live streaming, buffers sent by souphttpsrc are pushed to the live
adapter. The buffers in the adapter are sent out of mssdemux when it
is greater than 4096 bytes.

Occasionally, when seeking in live streams, if seek occurs just
after the last data chunk was received, and if this data chunk is
smaller than 4096 bytes, it will be kept in the live adapter.
This remaining data in the live adapter will be erroneously prepended
to the new data that is downloaded after seek and pushed out.
When qtdemux receives this data, since it does not start with
a moof box, it is impossible to demux the fragment, and bogus
size error will occur.

Clear out the live adapter on seek so that no unnecessary remaining
data is pushed out together with the new fragment after seeking.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1345>
 2020-09-30 11:48:02 +00:00
Ederson de Souza
8335039ecd tests/avtp: Fix coverity issues 
Fixes sign extension issues, unchecked return values and some constant
expression results.

CID: 1465073, 1465074, 1465075, 1465076, 1465077
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1398>
 2020-09-28 18:40:43 +00:00
Ederson de Souza
38d3360edb avtp: Change "%lu" for G_GUINT64_FORMAT 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1398>
 2020-09-28 18:40:43 +00:00
raghavendra
84f8dbd932 srtobject: typecast SRTO_LINGER to linger 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1615>
 2020-09-25 22:00:26 +05:30
Philippe Normand
2e8927ce93 wpe: Plug event leak 
Handled events don't go through the default pad event handler, so they need to
be unreffed in this case.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1568>
 2020-09-21 16:39:57 +00:00
Jan Schmidt
6fc7455881 wpesrc: Don't crash if WPE doesn't generate a buffer. 
On creating a 2nd wpesrc in a new pipeline in an app that already
has a runnig wpesrc, WPE sometimes doesn't return a buffer on request,
leading to a crash. This commit fixes the crash, but not the underlying
failure - a 2nd wpesrc can still error out instead.

Partially fixes https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/issues/1386

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1568>
 2020-09-21 16:39:57 +00:00
Philippe Normand
c3659cd611 wpe: Plug SHM buffer leaks 
Fixes #1409

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1568>
 2020-09-21 16:39:57 +00:00
Philippe Normand
8ef30a9ce5 wpe: Move webview load waiting to WPEView 
As waiting for the load to be finished is specific to the WebView, it should be
done from our WPEView, not from the WPEContextThread. This fixes issues where
multiple wpesrc elements are created in sequence. Without this patch the first
view might receive erroneous buffer notifications.

Fixes #1386

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1568>
 2020-09-21 16:39:57 +00:00
Philippe Normand
b707454a5a wpe: Use proper callback for TLS errors signal handling 
The load-failed and load-failed-with-tls-errors signals expect distinct callback
signatures.

Fixes #1388

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1566>
 2020-09-21 14:11:15 +00:00
Olivier Crête
825a79f01f webrtcbin: Accept end-of-candidate pass it to libnice 
libnice now supports the concept of end-of-candidate, so use the API
for it. This also means that if you don't do that, the webrtcbin will
never declared the connection as failed.

This requires bumping the dependency to libnice 0.1.16

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1139>
 2020-09-18 18:40:58 -04:00
Olivier Crête
63f06d16db webrtcbin: Merge the RTX SSRCs from all transceivers when bundling 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1545>
 2020-09-18 14:20:03 +00:00
Marian Cichy
c145798876 avtp: avtpaafdepay: fix crash when building caps 
gst_caps_new_simple gets wrong types for rate and channel which
may lead to a crash.

As 64-bit values for rate, depth, format, channels does not
make much sense and since any other functionality in gstreamer
expects G_TYPE_INT for channels and rate, we should stick to that

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1576>
 2020-09-17 08:32:07 +00:00
Emmanuel Gil Peyrot
f97b718b4c waylandsink: Use memfd_create() when available 
This (so-far) Linux- and FreeBSD-only API lets users create file
descriptors purely in memory, without any backing file on the filesystem
and the race condition which could ensue when unlink()ing it.

It also allows seals to be placed on the file, ensuring to every other
process that we won’t be allowed to shrink the contents, potentially
causing a SIGBUS when they try reading it.

This patch is best viewed with the -w option of git log -p.

It is an almost exact copy of Wayland commit
6908c8c85a2e33e5654f64a55cd4f847bf385cae, see
https://gitlab.freedesktop.org/wayland/wayland/merge_requests/4

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1577>
 2020-09-15 19:17:12 +00:00
Matthew Waters
e2d88f0569 webrtc: propagate more errors through the promise 
Return errors on promises when things fail where available.

Things like parsing errors, invalid states, missing fields, unsupported
transitions, etc.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1565>
 2020-09-14 04:04:29 +00:00
Adam Williamson
52ef192526 opencv: set opencv_dep when option is disabled (#1406) 
The examples build file checks opencv_dep, so it still needs to
be set even if the option is disabled.

Fixes https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/issues/1406

Signed-off-by: Adam Williamson <awilliam@redhat.com>
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1570>
 2020-09-11 07:16:21 +00:00
Mathieu Duponchelle
c096d30f6b openh264dec: port to new request_sync_point() API 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1571>
 2020-09-10 23:44:50 +02:00
Mathieu Duponchelle
c58357fb66 line21enc: add remove-caption-meta property 
Similar to #GstCCExtractor:remove-caption-meta

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1554>
 2020-09-09 22:11:28 +02:00
Mathieu Duponchelle
c07e2a89ba line21enc: heavily constrain video height 
We can only determine a correct placement for the CC line
with:

* height == 525 (standard NTSC, line 21 / 22)
* height == 486 (NTSC usable lines + 6 lines for VBI, line 1 / 2)

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1554>
 2020-09-09 19:38:58 +02:00
Mathieu Duponchelle
1d416750d1 line21enc: add support for CDP closed caption meta 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1554>
 2020-09-09 19:12:11 +02:00
Jan Alexander Steffens (heftig)
3f9a7e5c73 hlssink2: Actually release splitmuxsink's pads 
It was looking at the "outer" peer of the ghost pad, not the "inner"
peer (the target).

It provided the wrong pad to gst_element_release_request_pad.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1551>
 2020-09-09 01:06:21 +00:00
Sebastian Dröge
64039cdf84 gst: Update for gst_video_transfer_function_*() function renaming 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1557>
 2020-09-07 12:14:47 +03:00
Nirbheek Chauhan
16d84a2816 webrtc: Clean up the userinfo unescaping code 
Continuation from 04fd705906. This is
easier to understand and also avoids two copies.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1547>
 2020-08-30 09:53:42 +00:00
Jonathan Matthew
2b024ec1b4 modplug: avoid division by zero 
Under some conditions, GetMaxPosition() returns zero, which should cause
position queries to fail rather than crash.
 2020-08-28 08:10:04 +10:00
trilene
04fd705906 webrtc: Unescape turnserver user and password 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1530>
 2020-08-26 23:37:17 +01:00
Tim-Philipp Müller
b48702e4bc sctp: usrsctp: increase DIAG_MSG_LEN to accomodate longer file path 
Fixes "‘%s’ directive output truncated writing XX bytes into
a region of size NN [-Wformat-truncation=]" compiler warnings.

https://github.com/sctplab/usrsctp/pull/521

Fixes #1389

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1540>
 2020-08-26 00:00:24 +01:00
Philippe Normand
2caa3e0230 wpe: skip glbasesrc decide_allocation when non-GL caps are negotiated 
Checking for GL caps features in gl_start() was done too late in case the parent
class fails to setup a working GL context. The element now determines if GL
support should be enabled during the decide-allocation query handling.

Additionally, when no GL context was found, we need to handle the element
cleanup because in that situation glbasesrc won't call gl_stop.

Fixes https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/issues/1376

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1532>
 2020-08-24 20:59:50 +00:00
Matthew Waters
e15a8fcbdd webrtc/datachannel: clear the error after use 
Fixes a memory leak

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1535>
 2020-08-24 17:02:35 +10:00
Matthew Waters
7489addc0a webrtc/datachannel: free previous protocol/label fields 
Fixes a memory leak

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1535>
 2020-08-24 17:02:35 +10:00
Matthew Waters
e5a2e3ac4c sctpdec: unref after retrieving the static pad template 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1535>
 2020-08-24 17:02:35 +10:00
Matthew Waters
9011539940 webrtc/ice: resolve .local candidates internally 
Requires the system's DNS resolver to support mdns resolution.

Fixes interoperablity with recent versions of chrome/firefox that
produce .local address in for local candidates.

Fixes https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/issues/1139
 2020-08-20 13:01:17 +10:00
J. Kim
8e9f8c7f2c srtobject: set error when canceled waiting for a caller 
To propagate error, this commit sets a reason. Otherwise, the function
caller should check if `error` is NULL when the return value is not normal.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1522>
 2020-08-19 12:01:37 +00:00
J. Kim
ebdb3447ce srtobject: fix typo, s/errorj/error 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1522>
 2020-08-19 11:31:41 +00:00
Vivia Nikolaidou
dc58065dfb fdkaacenc: Implement flush function 
The internal fdk encoder always produces 1024 bytes even with no input,
so special care should be taken to not drain it twice.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1515>
 2020-08-17 23:39:39 +03:00
Jan Alexander Steffens (heftig)
76c171509e fdkaacenc: Refactor layout selection code 
No functional change.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1359>
 2020-08-17 10:19:56 +02:00
Jan Alexander Steffens (heftig)
5e68124981 fdkaacenc: Move channel layouts to gstfdkaac.c 
In preparation of sharing them with the decoder. Iteration of the
channel layouts needs to be changed to use a sentinel element.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1359>
 2020-08-17 08:07:00 +00:00
Matthew Waters
2d31aba78d vulkan: docs annotation updates 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1506>
 2020-08-15 02:55:30 +00:00
Philippe Normand
314a8c023f wpe: WebView and WebContext handling fixes 
The WPEThreaded view is now split in 2 classes:
- WPEContextThread handles the persistent WebKit thread, where all WebKit API
calls should be handled.
- WPEView: is created from the WPEContextThread. It handles the WebView and
maintains the public interface on which wpesrc relies. This is the facade for
the WebView, basically. It takes care of dispatching API calls into the context
thread.

With these fixes it is now possible to create (and reuse) mutlple wpesrc
elements during the application lifetime.

Fixes #1372

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1484>
 2020-08-14 09:41:56 +00:00
Sebastian Dröge
7ef393d5ff sctp: fix build with GST_DISABLE_GST_DEBUG 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1465>
 2020-08-14 01:48:33 +01:00
Tim-Philipp Müller
80a0da9698 sctp: hook up internal copy of libusrsctp to build 
Add option 'sctp-internal-usrsctp' so people can choose
to build againts the distro version instead.

Fixes https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/issues/870

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1465>
 2020-08-14 01:33:28 +01:00
Tim-Philipp Müller
f4538e24b6 sctp: import internal copy of usrsctp library 
There are problems with global shared state and no API stability
guarantees, and we can't rely on distros shipping the fixes we
need. Both firefox and Chrome bundle their own copies too.

Imported from https://github.com/sctplab/usrsctp,
commit 547d3b46c64876c0336b9eef297fda58dbe1adaf
Date: Thu Jul 23 21:49:32 2020 +0200

Fixes https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/issues/870

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1465>
 2020-08-14 01:32:45 +01:00
Seungha Yang
91f9490529 cccombiner: Correct sink_query chain up and fix caps leaks 
Don't chain up to src_query() from sink_query() method, and
returned caps by gst_static_pad_template_get_caps() needs to be
cleared.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1513>
 2020-08-13 20:42:51 +09:00
Hosang Lee
d9dda36e02 smoothstreaming: start closer to the edge in live streams 
It is more appropriate to start closer to the live edge in
live streams. Some live streams maintain a large dvr window
(over few hours in some cases), so starting from the first
fragment will be too far away from the live edge.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1346>
 2020-08-10 16:13:30 +09:00
Sebastian Dröge
1d1b3eb8b4 cccombiner: Update for additional info parameter to the "samples-selected" signal 
See https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/590

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1498>
 2020-08-07 17:53:14 +00:00
Mathieu Duponchelle
93a54093ec mpeg2enc: add disable-encode-retries property 
MJPEG Tools may reencode pictures in a second pass to stick
closer to the target bitrate. This can result in slower than
real-time encoding for full HD content in certain situations,
as entire GOPs need reencoding when the reference picture is
reencoded.

See https://sourceforge.net/p/mjpeg/bugs/141/ for background

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1491>
 2020-08-06 17:13:03 +00:00
Mathieu Duponchelle
674ad01016 mpeg2enc: report a latency 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1491>
 2020-08-06 17:13:03 +00:00
Mathieu Duponchelle
2dfceac9fc mpeg2enc: finalize GstVideoEncoder port 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1491>
 2020-08-06 17:13:03 +00:00
Tim-Philipp Müller
c9d10e2277 mpeg2enc: store video encoder instance directly in stream writer class 
Instead of storing the pad and then only using it to get the
element.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1491>
 2020-08-06 17:13:03 +00:00
Tim-Philipp Müller
8a745529c7 mpeg2enc: remove unused streamwriter member 'buf' 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1491>
 2020-08-06 17:13:03 +00:00
Tim-Philipp Müller
0c28c406cc mpeg2enc: remove some unused code 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1491>
 2020-08-06 17:13:03 +00:00
Tim-Philipp Müller
7f6eb54d42 mpeg2enc: remove code paths for older mjpegtools versions 
Gets rid of lots of code paths that no one has built,
used or tested for ages, and makes code more maintainable.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1491>
 2020-08-06 17:13:03 +00:00
Alban Browaeys
79d90b4fd2 mpeg2enc: initial port to GstVideoEncoder base class 
https://bugzilla.gnome.org/show_bug.cgi?id=685414

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1491>
 2020-08-06 17:13:03 +00:00
Sebastian Dröge
e70ec38000 srt: Add support for using hostnames instead of IP addresses 
If an address can't be parsed as IP address, try resolving it via
GResolver instead. SRT URIs more often than not contain hostnames and
without trying to resolve them we won't be able to handle such URIs.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1493>
 2020-08-06 07:29:14 +00:00
Mathieu Duponchelle
1522e40397 cccombiner: update to new samples selection API 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1497>
 2020-08-05 18:16:32 +02:00
Jordan Petridis
cee211123a opencv: compile with -Wno-format-nonliteral 
opencv plugin is pulling a header which makses clang++ 10
complain a lot and blocks -werror.

```
/usr/include/opencv4/opencv2/flann/logger.h:83:36: error: format string is not a string literal [-Werror,-Wformat-nonliteral]
        int ret = vfprintf(stream, fmt, arglist);
                                   ^~~
```

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1494>
 2020-08-05 12:17:06 +00:00
Jordan Petridis
92f9567737 gstlv2utils.c: avoid implicit float to int conversion 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1487>
 2020-08-04 11:37:52 +00:00
Jordan Petridis
5705301ed5 gstladspautils.c: avoid implicit float to int conversion 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1487>
 2020-08-04 11:37:52 +00:00
Francisco Javier Velázquez-García
97b5951d25 srtobject: Add support for IPv6 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1477>
 2020-08-03 21:46:04 +00:00
Francisco Javier Velázquez-García
1ba379ded0 srtobject: Reset parameters before setting URI 
This makes `gst_srt_object_validate_parameters` work properly since
`localaddress` and `localport` will be missing if the URL did not
provide them.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1477>
 2020-08-03 21:46:04 +00:00
Francisco Javier Velázquez-García
096c60f9c5 srtobject: Simplify gst_srt_object_set_*_value 
This fixes `gst_srt_object_set_string_value` in particular because the
value might not be a static string.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1477>
 2020-08-03 21:46:04 +00:00
Francisco Javier Velázquez-García
1a8e2cf981 srtobject: Store passphrase like other parameters 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1477>
 2020-08-03 21:46:04 +00:00
Nirbheek Chauhan
d4ca8820e7 webrtc, rtmp2: Warn if the user or password aren't escaped 
If the user/pass aren't escaped, the userinfo will be ambiguous and we
won't know where to split. We will accidentally get it right if the :
belongs in the password.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1481>
 2020-08-03 18:12:50 +00:00
Nirbheek Chauhan
827afa206d webrtc, rtmp2: Fix parsing of userinfo in URI strings 
While parsing the string, `gst_uri_from_string()` also unescapes the
userinfo. This is bad if your username contains a `:` character, since
we will then split the userinfo at the wrong location when parsing it.

To fix this, we can use the new `gst_uri_from_string_escaped()` API
that was added in
https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/583

Fixes https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/issues/831

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1481>
 2020-08-03 18:12:50 +00:00
Sebastian Dröge
6e412d42c7 hlssink2: Don't assert if we don't have a current location when receiving the fragment-closed message 
This can happen if the application did not provide an output stream for
the fragment and didn't handle the error message before splitmuxsink
decided to consider the fragment closed.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1469>
 2020-08-03 11:23:36 +00:00
Nicola Murino
8544f3928e opencv: allow compilation against 4.4.x 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1482>
 2020-08-01 17:38:42 +00:00
Mathieu Duponchelle
265128e7f7 cccombiner: implement samples selection API 
Call gst_aggregator_selected_samples() after identifying the
caption buffers that will be added as a meta on the next video
buffer.

Implement GstAggregator.peek_next_sample.

Add an example that demonstrates usage of the new API in
combination with the existing buffer-consumed signal.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1390>
 2020-07-30 23:10:33 +00:00
Mathieu Duponchelle
480ede1aa7 wpesrc: timestamp buffers when working with SHM buffers 
GLBaseSrc::fill() will take care of that when dealing with
images, but as we don't chain up when dealing with SHM buffers
this needs to be done in order for GLBaseSrc::get_times() to
work appropriately.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1476>
 2020-07-30 08:09:47 +00:00
Mathieu Duponchelle
ad57b8040f wpe: fix ready signalling 
Receiving the WEBKIT_LOAD_COMMITTED event doesn't actually
mean we have committed an SHM buffer / image yet.

As this is the condition we are interested in, check it
instead.

Also wrap g_cond_wait in a loop for extra correctness points.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1476>
 2020-07-30 08:09:47 +00:00
Damian Hobson-Garcia
deefedd002 waylandsink: Update stale GstBuffer references in wayland buffer cache 
"waylandsink: use GstMemory instead of GstBuffer for cache lookup"
changes the cache key to GstMemory, but the cached data still needs
a pointer to the GstBuffer to control the buffer lifecycle.
If the GstMemory used as the cache key moves from one GstBuffer to
another, the pointer in the cached data will be out-of-date.

Update the current GstBuffer pointer for each frame so that it always
represents the currently in use (from attach to release) GstBuffer
for each wl_buffer.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1473>
 2020-07-28 11:35:53 +00:00
Tim-Philipp Müller
798249dc9f directfb: suppress compiler warning from directfb headers 
On debian sid, directfb 1.7.7

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1467>
 2020-07-25 19:47:43 +01:00