The wpe element is used to produce a video texture representing a web page
rendered off-screen by WPE. This element can be used to overlay HTML on top of
another video stream for instance.
The latter is going away in libfdk-aac 2.0.0. Instead, MPEG-style output
is always non-interleaved and WAV-style output is always interleaved.
Earlier libfdk-aac also defaults interleaving accordingly.
Since our reordering looks at the associated PCE indices instead of the
actual channel order, we're agnostic to the mapping.
For https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/issues/825
Currently master code of gst1-plugins-bad use plain-string host name while passing it to
libnice agent: nice_agent_set_relay_info() in gstwebrtcice.c while adding turn_server(_add_turn_server).
It is observered that if we don't convert the host parameter by using gst_uri_get_host, it fails in libnice agent(0.1.14-1).
Code does, actually, set the host correctly but while passing params to nice_agent_set_relay_info, it uses incorrect one.
https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/issues/823
It fails to build only on Mac OSX with the following error.
In file included from ../subprojects/gst-plugins-bad/ext/opencv/gstopencv.cpp:45:
../subprojects/gst-plugins-bad/ext/opencv/gstcameracalibrate.h:96:38: error: a space is required between consecutive right angle brackets (use '> >')
std::vector<std::vector<cv::Point2f>> imagePoints;
^~
> >
1 error generated.
Fix: #817
As suggested in [the SSL_get_error manpage][1]. Upgrade the message to a
warning if the errno isn't 0 (success). The latter apparently means the
transport encountered an EOF (shutdown) without the shut down handshake
on the (D)TLS level. This happens quite often for otherwise normal DTLS
connections.
[1]: https://www.openssl.org/docs/man1.1.1/man3/SSL_get_error.html
Print out all errors from the OpenSSL error queue instead of just
looking at the topmost error. Using the callback interface also removes
the need for formatting using a buffer on the stack.
This reverts commit 73ebdb888e.
This isn't needed and it breaks srtpenc ! srtpdec, specifying the
roll-over counter manually is an advanced feature.
Also revert "srtp: Add "roc" caps field to the gst-launch example"
This reverts commit 67ae35813b.
https://bugzilla.gnome.org/show_bug.cgi?id=765079
ext/sctp/ext@sctp@@gstsctp@sha/sctpassociation.c.obj: In function `receive_cb':
/var/lib/jenkins/workspace/cerbero-cross-mingw32/workdir/sources/windows_x86/gst-plugins-bad-1.0-1.15.0.1/_builddir/../ext/sctp/sctpassociation.c:692: undefined reference to `_imp__ntohl@4'
Expanded to support image format to YV12/I422/I444. It's related to the
color bit-depth and profile of the codec. It can make configuring
appropriate profile according to bit-depth and format.
https://bugzilla.gnome.org/show_bug.cgi?id=791674
With prenegotiated channels, the data-channel protocol is not used and
instead the channel's negotiation is intended to be performed out of band in
some application-specific manner.
Comes with test!
This means that we will reject all operations before we've transitioned
into READY.
This also fixes the tests using the default GMainContext in the NULL
state instead of the webrtcbin internal GMainContext and thread. Also
removes a potential ordering race where on the element transitioning to
READY, an operations could have been queued on two different threads and
removing a guarentee on operation ordering.
It might be possible that if we set webrtcbin to the NULL state some
tasks (idle sources) are still executed and they might even freeze. The freeze
is caused because the webrtcbin tasks don't hold a reference to webrtcbin and
if it's last unref inside the idle source itself this will not allow the main
loop to finish because the main loop is waiting on the idle source to finish.
We now start and stop webrtcbin thread when changing states. This will allow
the idle sources to finish properly.
https://bugzilla.gnome.org/show_bug.cgi?id=797251
Fixes a race where the task could attempt to set
stream-start/caps/segment before the pad was active and would be
dropped resulting in a 'data-flow before stream-start' warning.
It is possible and often desirable to pass multiple ICE relays
to libnice agents, the "turn-server" property, while convenient
to use from the command line, does not allow that.
This adds a new action signal, "add-turn-server" to address that.
https://bugzilla.gnome.org/show_bug.cgi?id=797012
We now have options for all plugins, so we will just disable these in
the cerbero recipe instead. These require external deps, so they won't
affect gst-build either.
Although RTMP_ConnectStream() was failed, librtmp's internal memory
is not freed by RTMP_ConnectStream(), so RTMP_Close() should be called
before RTMP_Free()
https://bugzilla.gnome.org/show_bug.cgi?id=797058
Worst case it will be empty. This fixes a crash when the base class
calls data_received() when the stream is neither is_isobmff or
has_isoff_ondemand_profile.
https://bugzilla.gnome.org/show_bug.cgi?id=796745
gst_curl_http_src_remove_queue_item() can free qelement and then
we get an invalid memory reference when we do qelement->next a
couple of lines below. Take the next pointer earlier so that we can
safely free.
This fixes an issue with SSA/ASS subtitles, where subtitles
would fail to appear if there was already a subtitle on screen.
This was because `struct _GstAssRender` had a single
`GstBuffer *subtitle_pending` member. This meant that
the assrender context could only be aware of one subtitle
at a time.
This patch changes the subtitle_pending member to a
linked list of pending subtitles.
The `gst_ass_render_chain_text` function no longer needs
to care about whether there are already subtitles pending,
it simply appends new subtitles to the list.
The `gst_ass_render_chain_video` function has been modified
to handle the list of pending subtitles.
Finally, the `gst_ass_render_pop_text` function has been
modified to pop the entire list of pending subtitles.
https://bugzilla.gnome.org/show_bug.cgi?id=735944
When compiling with clang-6 this error raises:
raw_decoder.c:411:1: error: unused function 'cpr1204_crc'
[-Werror,-Wunused-function]
This patch only comments it out.
https://bugzilla.gnome.org/show_bug.cgi?id=796957
When compiling with clang-6 this error pops out:
raw_decoder.c:1011:62: error: implicit conversion from enumeration
type 'const vbi_modulation' to different enumeration type
'vbi3_modulation' [-Werror,-Wenum-conversion]
This is because function vbi3_bit_slicer_set_params() sets
vbi3_modulation as enum type parameter, nonetheless vbi_modulation
enum is passed. Both enums looks semantically equal, thus the fix is a
simple cast.
https://bugzilla.gnome.org/show_bug.cgi?id=796957
This is the native format that is in use by the webrtc audio processing
library internally, so this avoids internal {de,}interleaving and
format conversion (S16->F32 and back)
https://bugzilla.gnome.org/show_bug.cgi?id=793605
This uses the new path for OpenCV headers. OpenCV now have
master headers files per modules, which reduce the amount of
required includes. Note that HIGHGUI was included to get the
imgcodecs includes, which I fixed, though the master header is
missing the C headers, so I included that directly. All the
image stuff should be ported to C++ eventually. Finally, this
patch also update the header checks to reflect the modules that
are really being used.
... instead of doing it ourselves. Otherwise, we should add more
logic here (such as checking GstClock and etc) which was already provided by
GstBaseSrc.
https://bugzilla.gnome.org/show_bug.cgi?id=796842
Relaxed the wl_shell interface constrains, so application that
pass via GstContext the wl_surface can use waylandsink in a
compositor without wl_surface and zwp_fullscreen_shell.
Added support for zwp_fullscreen_shell.
https://bugzilla.gnome.org/show_bug.cgi?id=796772
When scanning paths for LADSPA plugins, don't try and load
every random file as a module, as g_module_open ends up throwing
errors on Windows.
Use a G_MODULE_SUFFIX and GST_EXTRA_MODULE_SUFFIX suffix check as
we do for GStreamer plugins.
https://bugzilla.gnome.org/show_bug.cgi?id=796450
Refactor transportsendbin, and change the way
pads are blocked on dtlssrtpenc so that they
don't interfere with state changes.
As well as being easier to read, this fixes
spurious failures shutting down webrtcbin
if DTLS negotiation hasn't completed yet.
Move the errant piece of dtlssrtpenc state change
management from dtlstransport in the Webrtc libs,
into the transportsendbin that does the rest of
the element management so it's all in one place.
The `CV_RGB` macro is now in `imgproc.hpp`.
Fixes:
../subprojects/gst-plugins-bad/ext/opencv/gsthanddetect.cpp:497:40: error: ‘CV_RGB’ was not declared in this scope
cvCircle (img, center, radius, CV_RGB (0, 0, 200), 1, 8, 0);
^~~~~~
When negotiation is triggered by receiving caps on our sink pad
probes, we could encounter a race condition where need-negotiation
is emitted and the application requires the creation of an offer
before the current caps were actually updated.
This led to retrieving incomplete caps when creating the offer,
using find_codec_preferences -> pad_get_current_caps.
Instead, as we save the caps in the probe callback anyway, it is better
and thread safe to use these if they were set.
https://bugzilla.gnome.org/show_bug.cgi?id=796801
Matches the output from a similar glimagesink pipeline when
rotating from an upstream gltransformation passed through
the affine transformation meta with xpos/ypos being set.
https://bugzilla.gnome.org/show_bug.cgi?id=794401
Fixes random crashes when an allocated webrtcbin isn't
given fresh 0-filled memory in its allocation. It works
mostly because GMutex and GCond are automatically initialised
in that case.
Move freeing of the pad blocks back to before we call the
GstBin state change function, as there's something racy
going on on the build server otherwise, where the pads don't
unblock during downward state changes.
This is a bit of a stab in the dark, since I can't recreate
the build server failure locally.
Release references in pad blocks and release the memory in the
dispose function too, in case the state change doesn't get
run (because calling the parent state change fails).
When changing state downward, we can't set pads
to inactive if they are blocked, it will deadlock
trying to acquire the streaming lock.
Just calling the parent state change function
will do the correct things to unblock probes and
set the pad inactive, so let it do that and
remove the probes after the parent state change
function has run
https://bugzilla.gnome.org/show_bug.cgi?id=796682
When max is GST_CLOCK_TIME_NONE in the query, it should not
be set in the query handler, this otherwise could lead to
impossible situations, where the minimum latency ended up
greater than the maximum.
https://bugzilla.gnome.org/show_bug.cgi?id=796603
The flush function immediately returned when pitch->next_buffer_offset
was 0.
This is clearly wrong, as next_buffer_offset can be 0 when a single
input buffer has been received, and no output buffer has been produced
before receiving EOS.
Simply remove that condition.
https://bugzilla.gnome.org/show_bug.cgi?id=796603
This lets users call gst_pad_get_current_caps on newly-added
pads to easily determine what to plug them into.
We cannot copy sticky events unconditionally in core,
see #719437https://bugzilla.gnome.org/show_bug.cgi?id=796387
This new element allows decoding and overlaying CEA-708 Closed Caption
streams over video.
* Supports CDP and cc_data closedcaption/x-cea-708 streams
* Uses pango to render CC stream
* Support GstVideoOverlayComposition meta if downstream supports is
Tested on various test files.
Remains to be fixed/improved:
* Switch to GstByteReader (for code safety)
* Switch to GString (instead of manual pango string construction)
* Move pango/rendering code outside of main 708 decoder file (so
that actual CC parser/decoder can be (re)used in other scenarios).
Initial patches and improvements by:
* CableLabs RUIH-RI Team <ruihri@cablelabs.com>
* Steve Maynard <steve@secondstryke.com>
* cjun.wang" <cjun.wang@samsung.com>
https://bugzilla.gnome.org/show_bug.cgi?id=704881
zvbi switched to a lot more flexible CC detection in VBI.
The problem is that it returns a *lot* of non-VBI lines as containing
CC which isn't the case.
Current code from zapping/zvbi as of 2018-03-14. Files copied
are all LGPL v2+.
Changes from original zvbi code:
* Switch to gst-debug logging system
* Use glib for endianness detection
* Fix compilation warnings
Allows extracting GstVideoCaptionMeta from a stream and outputs
it to a standalone stream.
Part of a new 'ext' closedcaption plugin, since more features are
going to be added, which will depend on external dependencies such
as pango.
On debian system headers trigger compiler warnings like these,
don't error out on them:
/usr/include/directfb/direct/os/linux/glibc/waitqueue.h:95:1: note: previous definition of ‘direct_waitqueue_signal’ was here
Explicitly cast to void* because GCC 8 is (rightfully) upset that this is
"writing to an object of type ‘...’ with no trivial copy-assignment".
Caused by the new "class-memaccess" warning
This moves all the conversion related code to a single place, allows
less code-duplication inside compositor and makes the glmixer code less
awkward. It's also the same pattern as used by GstAudioAggregator.
The aggregated_frame is now called prepared_frame and passed to the
prepare_frame and cleanup_frame virtual methods directly. For the
currently queued buffer there is a method on the video aggregator pad
now.
Previously we assumed that the texture ID is going to be valid even
after unmapping the frame, as it was immediately unmapped before even
being used. Now we only unmap once we're done with the texture.
During element shutdown, the srtp encryption session
object can be cleaned up. In that case, return GST_FLOW_FLUSHING
from the chain function. Also properly return GST_FLOW_ERROR
upstream during actual errors.
https://bugzilla.gnome.org/show_bug.cgi?id=790508
Store a PTS of a highlight event directly into the event structure,
rather than the GST_EVENT_TIMESTAMP that will probably be removed
in GStreamer 2.0, and is hardly used.
https://bugzilla.gnome.org/show_bug.cgi?id=761477
If that threshold is reached, `iqa` will emit an ERROR message on the
bus, stopping any processing.
This way we can do a simpler comparison with gst-validate and the
process will error out if the specified threshold is reached.
https://bugzilla.gnome.org/show_bug.cgi?id=795428
We don't want to reset the muxer, otherwise the continuity counter will
reset after each segment and some software gets confused. We want to
create a continuous stream.
https://bugzilla.gnome.org/show_bug.cgi?id=794816
There are two issues, both related to dependency checking with the meson
support for the ladspa plugin.
With autotools, lrdf is handled like an optional dependency. But with
meson it is required. This makes the meson support less flexible and
inconsistent with autotools.
When autotools is used it properly checks if ladspa.h is available.
But with meson it does not, instead it treats lrdf as the main
dependency. This could cause a build failure if lrdf is installed, but
the ladspa sdk is not.
https://bugzilla.gnome.org/show_bug.cgi?id=794350
Strictly speaking, the TTML spec requires that text backgrounds extend
only to the font height of the related text, rather than to the vertical
distance between lines. The result of this is that there will typically
be vertical gaps between line backgrounds through which moving video can
be seen. Since this was unnacceptable to some content providers, v1.0.1
of the IMSC spec (which profiles TTML) adds a new attribute,
itts:fillLineGap[1], that allows content authors to specify that clients
should extend text backgrounds such that there are no gaps between
lines. This attribute is also going to be included in the next release
of EBU-TT-D.
This patch adds support for fillLineGap to ttmlparse and ttmlrender.
[1] https://www.w3.org/TR/ttml-imsc1.0.1/#itts-fillLineGaphttps://bugzilla.gnome.org/show_bug.cgi?id=787071
Fixes ffeb09e4ab
if (sscanf(...)) { // != 0
error;
}
Is not correct where != 0 indicates some kind of success.
Check instead that the correct number of elements were slurped.
SDP's are generated and consumed according to the W3C PeerConnection API
available from https://www.w3.org/TR/webrtc/
The SDP is either created initially from the connected
sink pads/attached transceivers as in the case of generating an offer or
intersected with the connected sink pads/attached transceivers as in
the case for creating an answer. In both cases, the rtp payloaded streams
sent by the peer are exposed as separate src pads.
The implementation supports trickle ICE, RTCP muxing, reduced size RTCP.
With contributions from:
Nirbheek Chauhan <nirbheek@centricular.com>
Mathieu Duponchelle <mathieu@centricular.com>
Edward Hervey <edward@centricular.com>
https://bugzilla.gnome.org/show_bug.cgi?id=792523
By removing the indirection to the main loop completely when receiving
the peer certificate. For reference, the on-decoder-key signal does not
have a redirection.
We call the base class first as this will remove the pad from
the aggregator, thus stopping misc callbacks from being called,
one of which (process_textures) will recreate the vertex_buffer
if it is destroyed
https://bugzilla.gnome.org/show_bug.cgi?id=760873
For libsrtp 1, add defines that translate the new namespaced identifiers
to the old unnamespaced ones. Also move the code for setting and getting
a stream's ROC into two compat functions that match libsrtp2's API.
It seems that libsrtp2 properly supports changing the ROC without having
to touch the sequence numbers afterwards, given that srtp_set_stream_roc
sets a pending_roc field, so the entire roc_changed dance should not be
needed anymore. The compat functions for libsrtp 1 just contain our
preexisting hacks, however, so it's still needed there.
libsrtp2 has no means of discovering the streams in the session, so to
create the stats structure we need to iterate over our own set of SSRCs.
For this we also need to re-add the previously removed ssrcs_set to the
encoder.
https://bugzilla.gnome.org/show_bug.cgi?id=776901
Fix regression when used in combination with new flvmux which was
ported to GstAggregator, and which sends plain video/x-flv caps
before sending full caps that include streamheaders.
Instead of a massive if/else/if/else/if/else/...:
* Use a common cleanup path for allocated items just before leaving
the function (which will be free-d only if we're not dealing with
a delayed SPU).
* "goto" that cleanup path wherever needed
CID #1427096
CID #1427114
In file included from ../../../gst-plugins-bad/ext/gl/gstopengl.c:47:0:
../../../gst-plugins-bad/ext/gl/gstglmixerbin.h:25:29: fatal error: gst/video/video.h: No such file or directory
This is to mimic LV2 and what is commonly documented over the
web. We also completely track these directories when updating
the cache now. Unlike LV2, the plugins are flat in the plugin
directories, so no need for the recursive lookup. This also fixes
support for Fedora and other architecture using lib64 as a libdir.
While keeping it simple, this patch tries and mimic lilv default path.
It does not matter if some path are duplicated due to symlink because in
the end it's lilv that will walk these paths. The worst case is that we
update our cache more often then strictly needed.
https://bugzilla.gnome.org/show_bug.cgi?id=791717
The AVERAGE-BANDWIDTH attribute in the EXT-X-STREAM-INF tag represents
the average segment bit rate of the Variant Stream, while the BANDWIDTH
attribute represents the peak segment bit rate of the Variant Stream.
(https://tools.ietf.org/html/draft-pantos-http-live-streaming-23#section-4.3.4.2)
Using the average bit rate instead of the peak bit rate for variant switching
is more efficient and appropriate. Sometimes due to VBR encoding,
the BANDWIDTH may represent a value way above the average bit rate,
which could result to players not switching to that variant stream
although network bandwidth is sufficiently available.
https://bugzilla.gnome.org/show_bug.cgi?id=790821
gstsrt.c: In function ‘gst_srt_client_connect_full’:
gstsrt.c:151:6: error: ‘sock’ may be used uninitialized in this function [-Werror=maybe-uninitialized]
if (sock != SRT_INVALID_SOCK) {
https://bugzilla.gnome.org/show_bug.cgi?id=791302
When compiling with clang, an enum conversion error is triggered
since GstVideoFrameFlags are not GstVideoFlags.
This patch sets GST_VIDEO_FRAME_FLAG_NONE to the added video meta.
https://bugzilla.gnome.org/show_bug.cgi?id=791251
This patch adds code to gldownload to export the image as a
dmabuf if requested. The element now exposes memory:DMABuf as
a cap feature, and if it is selected, the element exports the
texture to an EGL image and then a dmabuf. It also implements a
fallback to system memory download in case the exportation failed.
https://bugzilla.gnome.org/show_bug.cgi?id=776927
We change the video info base on the received buffer. We need to
rollback these changes whenever we want to copy into our internal
pool of buffers.
https://bugzilla.gnome.org/show_bug.cgi?id=790057
The SHM interface does not allow passing arbitrary strides and offsets,
for this reason, we simply disable this feature from the proposed pool.
This fixes video artifact seen when using the FFMPEG based video
decoder.
https://bugzilla.gnome.org/show_bug.cgi?id=790057
This reverts commit 47fd4d391e.
This patch is incorrect. It doesn't actually compile, and causes a crash
because the viv-fb window implementation needs a native EGL handle
to pass to fbCreateWindow, but the GstGLDisplayEGL handleis actually
an EGLDisplay now (and gets cast to the wrong type)
