There's no reason for it to inherit from GstObject apart from
locking, which is easily replaced, and inheriting from
GInitiallyUnowned made introspection awkward and needlessly
complicated.
We pass-through the video as is, only putting a GstMeta on it from the
caption sinkpad.
This fixes negotation problems caused by not passing through caps
queries in both directions.
Also handle CAPS/ACCEPT_CAPS queries directly for the caption pad
instead of proxying.
Make declare/define a function consistent.
Note that GstBaseTransform::set_caps should return gboolean
Compiling C object subprojects/gst-plugins-bad/ext/vulkan/f3f9d6b@@gstvulkan@sha/vkviewconvert.c.obj.
../subprojects/gst-plugins-bad/ext/vulkan/vkviewconvert.c(644):
warning C4133: '=': incompatible types - from 'GstFlowReturn (__cdecl *)(GstBaseTransform *,GstCaps *,GstCaps *)'
to 'gboolean (__cdecl *)(GstBaseTransform *,GstCaps *,GstCaps *)'
The SPS parsing functions take a parse_vui_param flag
to skip VUI parsing, but there's no indication in the output
SPS struct that the VUI was skipped.
The only caller that ever passed FALSE seems to be the
important gst_h264_parser_parse_nal() function, meaning - so the
cached SPS were always silently invalid. That needs changing
anyway, meaning noone ever passes FALSE.
I don't see any use for saving a few microseconds in
order to silently produce garbage, and since this is still
unstable API, let's remove the parse_vui_param.
This patch adds to the CVF depayloader the capability to regroup H.264
fragmented FU-A packets.
After all packets are regrouped, they are added to the "stash" of H.264
NAL units that will be sent as soon as an AVTP packet with M bit set is
found (usually, the last fragment).
Unrecognized fragments (such as first fragment seen, but with no Start
bit set) are discarded - and any NAL units on the "stash" are sent
downstream, as if a SEQNUM discontinuty happened.
This patch introduces the AVTP Compressed Video Format (CVF) depayloader
specified in IEEE 1722-2016 section 8. Currently, this depayloader only
supports H.264 encapsulation described in section 8.5.
Is also worth noting that only single NAL units are handled: aggregated
and fragmented payloads are not handled.
As stated in AVTP CVF payloader patch, AVTP timestamp is used to define
outgoing buffer DTS, while the H264_TIMESTAMP defines outgoing buffer
PTS.
When an AVTP packet is received, the extracted H.264 NAL unit is added to
a "stash" (the out_buffer) of H.264 NAL units. This "stash" is pushed
downstream as single buffer (with NAL units aggregated according to format
used on GStreamer, based on ISO/IEC 14496-15) as soon as we get the AVTP
packet with M bit set.
This patch groups NAL units using a fixed NAL size lenght, sent downstream
on the `codec_data` capability.
The "stash" of NAL units can be prematurely sent downstream if a
discontinuity (a missing SEQNUM) happens.
This patch reuses the infra provided by gstavtpbasedepayload.c.
Based on `mtu` property, the CVF payloader is now capable of properly
fragmenting H.264 NAL units that are bigger than MTU in several AVTP
packets.
AVTP spec defines two methods for fragmenting H.264 packets, but this
patch only generates non-interleaved FU-A fragments.
Usually, only the last NAL unit from a group of NAL units in a single
buffer will be big enough to be fragmented. Nevertheless, only the last
AVTP packet sent for a group of NAL units will have the M bit set (this
means that the AVTP packet for the last fragment will only have the M
bit set if there's no more NAL units in the group).
This patch introduces the AVTP Compressed Video Format (CVF) payloader
specified in IEEE 1722-2016 section 8. Currently, this payload only
supports H.264 encapsulation described in section 8.5.
Is also worth noting that only single NAL units are encapsulated: no
aggregation or fragmentation is performed by the payloader.
An interesting characteristic of CVF H.264 spec is that it defines an
H264_TIMESTAMP, in addition to the AVTP timestamp. The later is
translated to the GST_BUFFER_DTS while the former is translated to the
GST_BUFFER_PTS. From AVTP CVF H.264 spec, it is clear that the AVTP
timestamp is related to the decoding order, while the H264_TIMESTAMP is
an ancillary information to the H.264 decoder.
Upon receiving a buffer containing a group of NAL units, the avtpcvfpay
element will extract each NAL unit and payload them into individual AVTP
packets. The last AVTP packet generated for a group of NAL units will
have the M bit set, so the depayloader is able to properly regroup them.
The exact format of the buffer of NAL units is described on the
'codec_data' capability, which is parsed by the avtpcvfpay, in the same
way done in rtph264pay.
This patch reuses the infra provided by gstavtpbasepayload.c.
This patch introduces the avtpsrc element which implements a typical
network source. The avtpsrc element receives AVTPDUs encapsulated into
Ethernet frames and push them downstream in the GStreamer pipeline.
Implementation if pretty straightforward since the burden is implemented
by GstPushSrc class.
Likewise the avtpsink element, applications that utilize this element
must have CAP_NET_RAW capability since it is required by Linux to open
sockets from AF_PACKET domain.
This patch introduces the avtpsink elements which implements a typical
network sink. Implementation is pretty straightforward since the burden
is implemented by GstBaseSink class.
The avtpsink element defines three new properties: 1) network interface
from where AVTPDU should be transmitted, 2) destination MAC address
(usually a multicast address), and 3) socket priority (SO_PRIORITY).
Socket setup and teardown are done in start/stop virtual methods while
AVTPDU transmission is carried out by render(). AVTPDUs are encapsulated
into Ethernet frames and transmitted to the network via AF_PACKET socket
domain. Linux requires CAP_NET_RAW capability in order to open an
AF_PACKET socket so the application that utilize this element must have
it. For further info about AF_PACKET socket domain see packet(7).
Finally, AVTPDUs are expected to be transmitted at specific times -
according to the GstBuffer presentation timestamp - so the 'sync'
property from GstBaseSink is set to TRUE by default.
This patch introduces the AAF depayloader element, the counterpart from
the AAF payloader. As expected, this element inputs AVTPDUs and outputs
audio raw data and supports AAF PCM encapsulation only.
The AAF depayloader srcpad produces a fixed format that is encoded
within the AVTPDU. Once the first AVTPDU is received by the element, the
audio features e.g. sample format, rate, number of channels, are decoded
and the srcpad caps are set accordingly. Also, at this point, the
element pushes a SEGMENT event downstream defining the segment according
to the AVTP presentation time.
All AVTP depayloaders will share some common code. For that reason, this
patch introduces the GstAvtpBaseDepayload abstract class that implements
common depayloader functionalities. AAF-specific functionalities are
implemented in the derived class GstAvtpAafDepay.
This patch introduces the AVTP Audio Format (AAF) payloader element from
the AVTP plugin. The element inputs audio raw data and outputs AVTP
packets (aka AVTPDUs), implementing a typical protocol payloader element
from GStreamer.
AAF is one of the available formats to transport audio data in an AVTP
system. AAF is specified in IEEE 1722-2016 section 7 and provides two
encapsulation mode: PCM and AES3. This patch implements PCM
encapsulation mode only.
The AAF payloader working mechanism consists of building the AAF header,
prepending it to the GstBuffer received on the sink pad, and pushing the
buffer downstream. Payloader parameters such as stream ID, maximum
transit time, time uncertainty, and timestamping mode are passed via
element properties. AAF doesn't support all possible sample format and
sampling rate values so the sink pad caps template from the payloader is
a subset of audio/x-raw. Additionally, this patch implements only
"normal" timestamping mode from AAF. "Sparse" mode should be implemented
in future.
Upcoming patches will introduce other AVTP payloader elements that will
have some common code. For that reason, this patch introduces the
GstAvtpBasePayload abstract class that implements common payloader
functionalities, and the GstAvtpAafPay class that extends the
GstAvtpBasePayload class, implementing AAF-specific functionalities.
The AAF payloader element is most likely to be used with the AVTP sink
element (to be introduced by a later patch) but it could also be used
with UDP sink element to implement AVTP over UDP as described in IEEE
1722-2016 Annex J.
This element was inspired by RTP payloader elements.
This patch introduces the bootstrap code from the AVTP plugin (plugin
definition and init) as well as the build system files. Upcoming patches
will introduce payloaders, source and sink elements provided by the AVTP
plugin. These elements can be utilized by a GStreamer pipeline to
implement TSN audio/video applications.
Regarding the plugin build system files, both autotools and meson files
are introduced. The AVTP plugin is landed in ext/ since it has an
external dependency on libavtp, an opensource AVTP packetization
library. For further information about libavtp check [1].
[1] https://github.com/AVnu/libavtp
The agent itself will take a ref on the property setter, so we'll be
left with two references to the certificate object, when actually there
should be only one
When WPEBackend-fdo >= 1.3.0 is detected, the threaded view now relies on the
wpe_fdo_egl_exported_image API instead of the EGLImageKHR-based API which is
going to be deprecated in 2.26. The GLib sources created by the view now use the
default priority as well, the custom priority is no longer required.
Regression introduced by b4bdcf15b7
This commit prevents the handshake from reaching dtlsdec when
the receive state of the receive bin is set to DROP (for example
when transceivers are sendonly).
This preserves the intent of the commit, by blocking the bin
at its sinks until the receive state is no longer BLOCK, but
makes sure the handshake still goes through, by only dropping
data at the src pads, as was the case before.
The timestamp/PTS alone is meaningless without the segment and usually
applications care about the running-time or stream-time.
This also keeps the messages in sync with the spectrum and level
elements.
Header data must be forwarded to downstream, but if demux does not finish
to finding type (e.g., ts, mp4 and etc), this header data can be cleared
by _stream_clear_pending_data(). Moreover, although demux finish downloading
header data, still it has fragment date to be downloaded, fragment sequence
shouldn't be advanced yet at that moment.
https://bugzilla.gnome.org/show_bug.cgi?id=776928
Including gl.h from WPEThreadedView.h leads to GST_LEVEL_DEFAULT detected as
redefined. The proposed fix is to include config.h from the CPP implementation
file and disable gl.h inclusion in the header, by using forward declarations.
1. The spec indicates that the notification should occur near the end of
'setting the description' processing
2. The current location with the drop of the lock could cause the 'check
if negotiation is needed' logic to execute and become confused about
the state of the webrtcbin's current local descriptions.
In the bad case, the following assertions could be hit:
g_assert (trans->mline < gst_sdp_message_medias_len (webrtc->current_local_description->sdp));
g_assert (trans->mline < gst_sdp_message_medias_len (webrtc->current_remote_description->sdp));
Moving the signalling state change later in the set description task
means that checking for a renegotiation will early abort as the
signalling state is not STABLE before the session description and
transceivers have been updated.
The start of each segment is relative to the Period start, minus
the presentation time offset.
As specified in section 5.3.9.6 of the MPEG DASH specification:
The value of the @t attribute minus the value of the
@presentationTimeOffset specifies the MPD start time of
the first Segment in the series.
dashdemux was not taking account of presentationTimeOffset and in
some methods was not taking into account the Period start time.
This commit modifies the segment->start value to always be
relative to the MPD start time (zero for VOD,
availabilityStartTime for live streams). This makes all uses of
the segment list consistent.
Fixes#841
If both data channels become ready simultaneously, then the two integer
read-add-update cycles can execute concurrently and only ever increment
once instead of the required twice. Use an atomic add instead.
It is very possible for badly behaving signalling or peers to send
us ICE candidates before we receive an SDP. While we had consideration
for that on the first set SDP, subsequent SDP's could result in
misconfigured ICE transports. Expand the previous code to also take
into account reconfigurations.
Limitations:
- No transport changes at all (ICE, DTLS)
- Codec changes are untested and probably don't work
- Stream removal doesn't remove transports (i.e. non-bundled transports
will stay around until webrtcbin is shutdown)
- Unified Plan SDP only. No Plan-B support.
Currently, if one was to set -Dhls-crypto to either libgcrypt or openssl
instead of auto, the following lines would fail because hls_crypto_dep is not
yet set:
if not hls_crypto_dep.found() and ['auto', 'libgcrypt'].contains(hls_crypto)
if not hls_crypto_dep.found() and ['auto', 'openssl'].contains(hls_crypto)
Instead, change "if not hls_crypto_dep.found()" to "if not have_hls_crypto"
which fixes the error.
Use a timeout to limit that amount of time we wait after the compositor
for the initial configure event. Compositor are support to emit a
configure event before any wl_buffer can be attached. The problem is
that Weston strongly enforce this, while gnome-shell simply does not
emit such an event.
When buffer is used by compositor, we don't need attach it and hold one
more reference. Just check used_by_compositor, just return if it is true.
Assert error log is not need, this is normal behavior.
... when it has not yet been connected to.
Also, a condition variable is not a semaphore, so a lock/wait/unlock
sequence is inherently racy without any state checking. So switch to
a different lock and check the intended state.
Some GIR annotations were incorrect or even missing. The former isn't
good for bindings, while the latter is especially annoying for signal
handlers, as that means your arguments will get the wrong names in the
rendered documentation.
GST_VIDEO_BUFFER_FLAG_INTERLACED and GST_VIDEO_BUFFER_FLAG_TFF
flags are needed when processing SCTE 20 closed captions for an interlaced
stream, when we need to convert back to analog, in which case we need to match
the caption to the top or bottom field
... if (x265 version >= 1.9) requirement is satisfied.
The SEI messages were supported since x265 version 1.8
but there was API change from version 1.9
(contentLightLevelInfo was renamed to maxCLL and maxFALL)
This change makes it possible to create more than just 5 webrtc
data channels. The maximum number of data channels is exactly
DEFAULT_NUMBER_OF_SCTP_STREAMS / 2, therefore the limit is now
512.
Use proper API to flush libass events when we do
a flushing seek, and also do it in FLUSH_STOP
rather than FLUSH_START, so we can be sure
streaming has stopped.
Fixes seeking back in time.
Something seems to have changed in libass that
renders the old manual way of flushing events
ineffective and libass then seems to ignore
timestamps that are older than the ones last
seen then if we do it the old way.
Fixes#916
With latest XDG shell, we need to fait for the surface to have been
configured before we can attach a buffer to it. This is being enforce by
Weston with an error.
Fixes#933
Fixes the following error.
gstccconverter.c:677:7: error: variable 'len' is used uninitialized whenever 'if' condition is false
[-Werror,-Wsometimes-uninitialized]
if (flags & 0x40) {
^~~~~~~~~~~~
gstccconverter.c:698:10: note: uninitialized use occurs here
return len;
^~~
gstccconverter.c:677:3: note: remove the 'if' if its condition is always true
if (flags & 0x40) {
^~~~~~~~~~~~~~~~~~
gstccconverter.c:572:12: note: initialize the variable 'len' to silence this warning
guint len;
^
= 0
Prior to this, cccombiner stopped consuming video buffers when
data wasn't arriving on its caption pad. In a live situation,
when aggregator is timing out we should still output whatever
video buffers are present, even if no caption buffers can be
aggregated with them.
Since the addition of BUNDLE support, the pads and the transceivers
share a single transport stream. When getting stats from the stream,
filter by the ssrc of the current pad to avoid merging the stats for
different pads.
Fixes https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/issues/889
It is necessary to implement this vmethod, as when the src pad
is marked as reconfigure, the base class will reset to src caps,
and the default update_src_caps simply queries the caps allowed
downstream without taking into account the caps set by
gst_aggregator_set_src_caps.
To make curlhttpsrc behave more like souphttpsrc, set the
BUFFER_OFFSET in its output buffers to match the segment
start. This means that in a HTTP RANGE request, the BUFFER_OFFSET
will match the value in the RANGE request.
To make it closer to a drop-in replacement for souphttpsrc,
expose the same gst_error_message_with_details as souphttpsrc,
so that applications can received the HTTP status code and reason
when an error occurs.
curlhttpsrc uses a single thread running the
gst_curl_http_src_curl_multi_loop() function to handle receiving
data and messages from libcurl. Each instance of curlhttpsrc adds
an entry into a queue in GstCurlHttpSrcMultiTaskContext and waits
for the multi_loop to perform the HTTP request.
Valgrind has shown up race conditions and memory leaks:
1. gst_curl_http_src_change_state() does not wait for the multi_loop
to complete before going to the NULL state, which means that
an instance of GstCurlHttpSrc can be released while
gst_curl_http_src_curl_multi_loop() still has a reference to it.
2. if multiple elements try to be removed from the queue at once,
only the last one is deleted.
3. source->caps is leaked
4. curl multi_handle is leaked
5. leak of curl_handle if URI not set
6. leak of http_headers when reusing element
7. null pointer dereference in negotiate caps
8. double-free of the default user-agent string
9. leak of multi_task_context.task
This commit changes the logic so that each element has a connection
status, which is used by the multi_loop to decide when to remove an
element from its queue. An instance of curlhttpsrc will not enter
the NULL state until its reference has been removed from the queue.
When shutting down the curl multi loop, the memory allocated from the
call to curl_multi_init() is now released.
When gstadaptivedemux uses a URI source element, it will re-use
it for multiple requests, moving it between READY and PLAYING
between each request. curlhttpsrc was leaking the http_headers
structure in this use case.
The gst_curl_http_src_negotiate_caps() function extracts the
"response-headers" field from the http_headers, but did not check
that this field might be NULL.
If the user-agent property is set, the global user-agent string
was freed. This caused a double-free error if the user-agent is
ever set a second time during the execution of the process.
There are situations within curlhttpsrc where the code needs
both the global multi_task_context mutex and the per-element
buffer_mutex. To avoid deadlocks, it is vital that the order in
which these are requested is always the same. This commit modifies
the locking order to always be in the order:
1. multi_task_context.task_rec_mutex
2. buffer_mutex
Fixes#876
Instead of creating a region, adding nothing to it, setting that as the
input region and destroying the region, you can instead just pass NULL
to wl_surface_set_input_region for the same effect.
Fixes#702
If bundle was used in combination with rtx, only the bundled transport
stream would have correctly configured rtx parameters.
Iterate over the payloads upfront in the bundled case to ensure the
correct payload mapping is set for the RTX elements.
With refactoring, supporting passphrase was removed accidently.
This commit re-enables srt encryption and validates 'passphrase'
by checking the return value of 'srt_setsockopt'.
fix: #694
Fix following build error
../subprojects/gst-plugins-bad/ext/openh264/gstopenh264dec.cpp(76): error C2121:
Note that msvc usually complains #if inside macro
gstladspa.c:360:5: error: zero-length ms_printf format string [-Werror=format-zero-length]
vad_private.c:108:3: error: this decimal constant is unsigned only in ISO C90 [-Werror]
gstdecklinkvideosink.cpp:478:32: error: comparison between 'BMDTimecodeFormat {aka enum _BMDTimecodeFormat}' and 'enum GstDecklinkTimecodeFormat' [-Werror=enum-compare]
win/DeckLinkAPI_i.c:72:8: error: extra tokens at end of #endif directive [-Werror]
win/DeckLinkAPIDispatch.cpp:35:10: error: unused variable 'res' [-Werror=unused-variable]
gstwasapiutil.c:733:3: error: format '%x' expects argument of type 'unsigned int', but argument 8 has type 'DWORD' [-Werror=format]
gstwasapiutil.c:733:3: error: format '%x' expects argument of type 'unsigned int', but argument 9 has type 'guint64' [-Werror=format]
kshelpers.c:446:3: error: missing braces around initializer [-Werror=missing-braces]
kshelpers.c:446:3: error: (near initialization for 'known_property_sets[0].guid.Data4') [-Werror=missing-braces]
Replace legacy usage of DecodeFrame2 API in favour of the
recommended DecodeFrameNoDelay()
This fixes problems with DecodeFrame2() not (currently) returning
all frames in main/high streams with B-frames, and reduces latency -
previously openh264 would not return a decoded frame until the next
call to DecodeFrame2(). DecodeFrameNoDelay() returns them immediately.
If the last frame(s) produce errors, then we need to drop them
or else we spin forever failing to decode a frame and thinking
it'll get better if we wait for more data that's never coming.
The fdkaac decoder supports 6.1 / 7.1 channels with downmixer
since v0.1.4. Old versions can use AAC_PCM_OUTPUT_CHANNELS
instead of AAC_PCM_MAX_OUTPUT_CHANNELS.
Fixes#873
This is the equivalent of iceTransportPolicy in the RTCConfiguration
dictionary.
Only two values are implemented:
* all: default behaviour
* relay: only gather relay candidates
The third member of the iceTransportPolicy enum, "public", is
obsolete.
0a350c610d broke the build by only
building enum types with meson. It also removed gstsrt.c from the list
of sources, causing the plugin to fail to load.
squash! srt: Fix autotools build
gstsrtobject.c: In function ‘gst_srt_object_close’:
gstsrtobject.c:1036:7: error: function called through a non-compatible type [-Werror]
(GDestroyNotify) g_closure_unref);
/usr/include/glib-2.0/glib/gmem.h:121:8: note: in definition of macro ‘g_clear_pointer’
(destroy) (_ptr); \
^~~~~~~
gstsrtobject.c:1038:7: error: function called through a non-compatible type [-Werror]
(GDestroyNotify) g_closure_unref);
/usr/include/glib-2.0/glib/gmem.h:121:8: note: in definition of macro ‘g_clear_pointer’
(destroy) (_ptr); \
^~~~~~~
Arch Linux
gcc 8.2.1 20181127
glib 2.58.2
We have srt{client,server}{src,sink} elements in accordance to the
norm of the connection oriented protocols. However, SRT connection
mode can be changed by uri parameters so it requires an integrated
element to handle the parameters.
fix: #740
As a side-effect we can now actually store the line offset in the
line21dec element, and have to perform fewer transformations in the
decklink elements (which were also buggy as they assumed a single byte
triplet per meta).
When waylandsink is used on some other thread than the main wayland
client thread, the waylandsink implementation is vulnerable to a
condition related to registry and surface events which handled in
seperated event queue.
The race that may happen is that after a proxy is created, but
before the queue is set, events meant to be emitted via the yet to
set queue may already have been queued on the wrong queue.
Wayland 1.11 introduced new API that allows creating a proxy
wrappper which can help to avoid this race condition.
It depends on the framerate how many cc_data byte pairs are allowed per
frame, and the framerate is also needed for converting into the CDP or
MCC format as the framerate is part of the header metadata.
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.
