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)
SRT[0] is an open source transport technology[1] that optimizes
streaming performance across unpredictable networks.
Although SRT is based on UDP, it works like connection-oriented
protocol. However, it doesn't mean that the SRT server or client
is necessarily to link to a receiver or a sender so, here, the
pairs of source and sink elements are introduced.
- srtserversink: SRT server to feed SRT stream
- srtclientsrc: SRT client to get SRT stream from srtserversink
- srtclientsink: SRT client to send SRT stream
- srtserversrc: SRT server to listen from srtclientsink
[0] https://github.com/Haivision/srt
[1] http://www.srtalliance.org/https://bugzilla.gnome.org/show_bug.cgi?id=785730
OpenJPEG 2.3 installs its headers to /usr/include/openjpeg-2.3. However,
since libopenjp2.pc seems to provide the right includedir CFLAGS at
least since version 2.1, instead of adding yet another version check,
just remove the subdir and the check for 2.2.
https://bugzilla.gnome.org/show_bug.cgi?id=788703
It is legal for a stream to reuse segments (marking discontinuities as
needed). Uplynk delivers such playlists for their placeholder loops.
Leave the URI scanning in place for playlists which have no
EXT-X-MEDIA-SEQUENCE tag. This should be harmless since the spec
requires these playlists to not be missing segments (RFC8216 6.2.2),
so we should be always matching on the first segment.
https://bugzilla.gnome.org/show_bug.cgi?id=788417
The function was basically one big if-else. Move the branch to the
one caller.
Currently, it's never called with previous_files == NULL. Assert that
this continues.
https://bugzilla.gnome.org/show_bug.cgi?id=788417
This simplifies the code a lot without any functional changes apart from
not closing the display connection. Closing the display connection is
not safe to do as it is shared between all other code in the same
process and no reference counting or anything happens at the platform
layer.
