This is needed for cross-compiling without a build machine compiler
available. The option was added in 0.54, but we only need this in
Cerbero and it doesn't affect older versions so it should be ok.
Will only cause a spurious warning.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1266>
Instead of storing the raw cc_data, store the 2 cea608 fields individually
as well as the ccp data.
Simply copying the input cc_data to the output cc_data violates a number of
requirements in the cea708 specification. The most prominent being, that
cea608 triples must be placed at the beginning of each cdp.
We also need to comply with the framerate-dpendent limits for both the
cea608 and the ccp data which may involve splitting or merging some
cea608 data but not ccp data or vice versa.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1116>
Add some properties to allow TCP and UDP candidates to be toggled. This
is useful in cases where someone is using this element in an environment
where it is known in advance whether a given transport will work or not
and will prevent wasting time generating and checking candidate pairs
that will not succeed.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1223>
When negotiating the SDP we should only connect the streams that are
actually mentioned in the SDP. All other streams are not relevant at
this time and would likely be part of a future SDP update. Fixes a
couple of the renegotiation webrtc unit tests.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1240>
Proper calculate running time for buffers that are out of current
segment and try to honor them.
A typical case is for AVTP packets coming from avtpcvfpay element, as
those may have DTS that falls out of segment (which is about PTS).
By using gst_segment_to_running_time_full(), avtpsink can properly
calculate when to transmit those buffers.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1004>
Seek events will cause new segments to be sent to avtpcvfpay, and for
flushing seeks, a pipeline running time reset. This running time
reset, which effectively changes pipeline base time, will cause
avtpcvfpay element to generate incorrect DTS for the initial set of
buffers sent after FLUSH_STOP.
This happens due the fact that base time change happens only when the
sink gets the first buffer after the FLUSH_STOP - so avtpcvfpay used
the wrong base time to do its calculations.
However, if the pipeline is paused before the seek, sink will update
base time when pipeline state goes to PLAYING again, before avtpcvfpay
gets the first buffers after the flush. Then avtpcvfpay element will be
able to normally calculate DTS for the outgoing packets.
This patch simply adds a warning message in case a flushing seek is
performed on a playing pipeline.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1004>
TSN streams are expected to send packets to the network in a well
defined "pace", which is arbitrarily defined for each stream. This pace
is defined by the "measurement interval" property of a stream.
When the AVTP CVF payloader element - avtpcvfpay - fragments a video
frame that is too big to be sent to the network, it currently defines
that all fragments should be transmitted at the same time (via DTS
property of GstBuffers generated, as sink will use those to time the
transmission of the AVTPDU). This doesn't comply with stream definition,
which also has a limit on how many packets can be sent on a given
measurement interval.
This patch solves that by spreading in time the DTS of the GstBuffers
containing the AVTPDUs. Two new properties, "measurement-interval" and
"max-interval-frames", added to avptcvfpay element so that it knows
stream measurement interval and how many AVTPDUs it can send on any of
them. More details on the method used to proper spread DTS/PTS according
to measurement interval can be found in a code commentary inside this patch.
Tests also added for the new property and behaviour.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1004>
If the remote is bundling, but we are not and remote is offering.
we cannot put the remote media sections into a bundled transport as that
is not how we are going to respond.
This specific failure case was that the remote ICE credentials were
never set on the ice stream and so ice connectivity would fail.
Technically, this whole bunde-policy=none handling should be removed
eventually when we implement bundle-policy=balanced. Until such time,
we have this workaround.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1231>
This commit introduces the AVTP Clock Reference Format (CRF) Checker
element. This element re-uses the GstAvtpCrfBase class introduced along
with the CRF Synchronizer element.
This element will typically be used along with the avtpsrc element to
ensure that the AVTP timestamp (and H264 timestamp in case of CVF-H264
packets) is "aligned" with the incoming CRF stream. Here, "aligned" means
that the timestamp value should be within 25% of the period of the media
clock recovered from the CRF stream.
The user can also set an option (drop-invalid) in order to drop any packet
whose timestamp is not within the thresholds of the incoming CRF stream.
This commit introduces the AVTP Clock Reference Format (CRF) Synchronizer
element. This element implements the AVTP CRF Listener as described in IEEE
1722-2016 Section 10.
CRF is useful in synchronizing events within different systems by
distributing a common clock. This is useful in a scenario where there are
multiple talkers who are sending data to a single listener which is
processing that data. E.g. CCTV cameras on a network sending AVTP video
streams to a base station to display on the same screen.
It is assumed that all the systems are already time-synchronized with each
other. So, the AVTP Talker essentially adjusts the AVTP Presentation Time
so it's phase-locked with the reference clock provided by the CRF stream.
There are 2 different roles of systems which participate in CRF data
exchange. A system can either be a CRF Talker, which samples it's own
clock and generates a stream of timestamps to transmit over the network, or
a CRF Listener, the system which receives the generated timestamps and
recovers the media clock from the timestamps. It then adjusts it's own
clock to align with recovered media clock. The timestamps generated by the
talker may not be continuous and the listener might have to interpolate
some timestamps to recover the media clock. The number of timestamps to
interpolate is mentioned in the CRF stream AVTPDU (Refer IEEE 1722-2016
Section 10.4 for AVTPDU structure). Only CRF Listener has been implemented
in this commit.
The CRF Sync element will create a separate thread to listen for the CRF
stream. This thread will calculate and store the average period of the
recovered media clock. The pipeline thread will use this stored period
along with the first timestamp of the latest CRF AVTPDU received to
calculate adjustment for timestamps in the audio/video streams. In case of
CRF AVTPDUs with single timestamp, two consecutive CRF AVTPDUs will be used
to figure out the average period of the recovered media clock.
In case of H264 streams, both AVTP timestamp and H264 timestamp will be
adjusted.
In the future commits, another "CRF Checker" element will be introduced
which will validate the timestamps on the AVTP Listener side. Which is why
a lot of code has been implemented as part of the gstcrfbase class.
If we are in a state where we are answering, we would start gathering
when the offer is set which is incorrect for at least two reasons.
1. Sending ICE candidates before sending an answer is a hard error in
all of the major browsers and will fail the negotiation.
2. If libnice ever adds the username fragment to the candidate for
ice-restart hardening, the ice username and fragment would be
incorrect.
JSEP also hints that the right call flow is to only start gathering when
a local description is set in 4.1.9 setLocalDescription
"This API indirectly controls the candidate gathering process."
as well as hints throughout other sections.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1226>
If we receive video buffers with non-perfect timestamps, the
caption buffers' timestamps might fall in the interval between
the end of one video buffer and the start of the next one.
Make our criteria for dropping that the caption buffer has
a timestamp older than the end of the previous video buffer,
not older than the start of the new one, unless of course
this is the first video buffer.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1207>
Use GST_OBJECT_LOCK (srtobject->element) to protect only the fields
involved in property access.
Introduce a new mutex srtobject->sock_lock to go with
srtobject->sock_cond and protect the list of callers from concurrent
access.
The previous version of the SHM export support still required a valid
EGLDisplay. The upcoming WPEBackend-FDO 1.8.x aims to remove this requirement,
hence allowing wpesrc to be used without GPU.
We should directly check the values of the `debug` and `optimization`
options instead.
`get_option('buildtype')` will return `'custom'` for most combinations
of `-Doptimization` and `-Ddebug`, but those two will always be set
correctly if only `-Dbuildtype` is set. So we should look at those
options directly.
For the two-way mapping between `buildtype` and `optimization`
+ `debug`, see this table:
https://mesonbuild.com/Builtin-options.html#build-type-options
Each srtp_stream_t is tied to an specific SSRC, so a
roc_changed flag should be kept per each SSRC in order to
properly reset RTP sequence number on ROC changes.
openssl 1.1.1e does some stricker EOF handling and will throw an error
if the EOF is unexpected (like in the middle of a record). As we are
streaming data into openssl here, it is entirely possible that we push
data from multiple buffers/packets into openssl separately.
From the openssl changelog:
Changes between 1.1.1d and 1.1.1e [17 Mar 2020]
*) Properly detect EOF while reading in libssl. Previously if we hit an EOF
while reading in libssl then we would report an error back to the
application (SSL_ERROR_SYSCALL) but errno would be 0. We now add
an error to the stack (which means we instead return SSL_ERROR_SSL) and
therefore give a hint as to what went wrong.
[Matt Caswell]
We can relax the EOF signalling to only return TRUE when we have stopped
for any reason (EOS, error).
Will also remove a spurious EOF error from previous openssl version.
Otherwise when bundling, only the changed streams would be considered as
to whether the bundled transport needs to be blocked as all streams are
inactive.
Scenario is one transceiver changes direction to inactive and as that is
the only change in transciever direction, the entire bundled transport would
be blocked even if there are other active transceivers inside the same bundled
transport that are still active.
Fix by always checking the activeness of a stream regardless of if the
transceiverr has changed direction.
The ICE gathering state can transition to complete prematurely if the
underlying ICE components complete their gathering while the initial
ICE gathering state task is queued and still pending.
In that situation, the ice gathering state task will report complete
while there are still ICE candidates queued for emission.
Prevent that by storing ICE candidates in an array and checking if
there are any pending before reporting a completed ICE gathering
state.
ICE candidates can be added to the array directly from the application
or from the webrtc main loop. Rename it to make it clear that it's
holding remote ICE candidates from the peer, and protect it with a
new mutex
As per discussion in the bug, remove the drop state from transportreceivebin.
Dropping data is necessary, but for bundled config, needs to happen
further downstream after mixed flows have been separated.
Also support switching back to BLOCK from PASS state.
https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/issues/1206
Fixes dependency issues:
FAILED: subprojects/gst-plugins-bad/ext/dash/8bd0b95@@gstdash@sha/gstdashsink.c.obj
cl @subprojects/gst-plugins-bad/ext/dash/8bd0b95@@gstdash@sha/gstdashsink.c.obj.rsp
C:\builds\ystreet\gst-plugins-base\gst-build\subprojects\gst-plugins-base\gst-libs\gst/pbutils/pbutils.h(30): fatal error C1083: Cannot open include file: 'gst/pbutils/pbutils-enumtypes.h': No such file or directory
Instead of synchronising at the ICE transport, do clock sync for the
RTP stream at the DTLS transport via the dtlssrtpenc rtp-sync
property. This avoids delaying RTCP while waiting until it is time
to output an RTP packet when rtcp-mux is enabled.
https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/issues/1212
We do not have a way to know the format modifiers to use with string
functions provided by the system. `G_GUINT64_FORMAT` and other string
modifiers only work for glib string formatting functions. We cannot
use them for string functions provided by the stdlib. See:
https://developer.gnome.org/glib/stable/glib-Basic-Types.html#glib-Basic-Types.description
F.ex.
```
../ext/dash/gstxmlhelper.c: In function 'gst_xml_helper_get_prop_unsigned_integer_64':
../ext/dash/gstxmlhelper.c:473:40: error: unknown conversion type character 'l' in format [-Werror=format=]
if (sscanf ((gchar *) prop_string, "%" G_GUINT64_FORMAT,
^~~
In file included from /builds/nirbheek/cerbero/cerbero-build/dist/windows_x86/include/glib-2.0/glib/gtypes.h:32,
from /builds/nirbheek/cerbero/cerbero-build/dist/windows_x86/include/glib-2.0/glib/galloca.h:32,
from /builds/nirbheek/cerbero/cerbero-build/dist/windows_x86/include/glib-2.0/glib.h:30,
from /builds/nirbheek/cerbero/cerbero-build/dist/windows_x86/include/gstreamer-1.0/gst/gst.h:27,
from ../ext/dash/gstxmlhelper.h:26,
from ../ext/dash/gstxmlhelper.c:22:
/builds/nirbheek/cerbero/cerbero-build/dist/windows_x86/lib/glib-2.0/include/glibconfig.h:69:28: note: format string is defined here
#define G_GUINT64_FORMAT "llu"
^
../ext/dash/gstxmlhelper.c:473:40: error: too many arguments for format [-Werror=format-extra-args]
if (sscanf ((gchar *) prop_string, "%" G_GUINT64_FORMAT,
^~~
```
In the process, we're also following the DASH MPD spec more closely
now, which specifies that ranges must follow RFC 2616 section 14.35.1:
https://tools.ietf.org/html/rfc2616#page-138
Add latency configuration logic to transportsendbin to
isolate it from the overall pipeline latency. That means that
it configures minimum latency internally based on the
latency query, and sends a latency event upstream that
matches.
Fixes https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/issues/1209
When emitting ICE candidates, also merge them to the local and
pending description so they show up in the SDP if those are
retrieved from the current-local-description and
pending-local-description properties.
https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/issues/676
- Do not send ABORTs for unexpected packets are as response to INIT
- Enable interleaving of messages of different streams
- Configure 1MB send and receive buffer for the socket
- Enable SCTP_SEND_FAILED_EVENT and SCTP_PARTIAL_DELIVERY_EVENT events
- Set SCTP_REUSE_PORT configuration
- Set SCTP_EXPLICIT_EOR and the corresponding send flag. We probably
want to split packets to a maximum size later and only set the flag
on the last packet. Firefox uses 0x4000 as maximum size here.
- Enable SCTP_ENABLE_CHANGE_ASSOC_REQ
- Disable PMTUD and set an maximum initial MTU of 1200
Calling bind() only sets up some data structures and calling connect()
only produces one packet before it returns. That packet is stored in a
queue that is asynchronously forwarded by the encoder's source pad loop,
so not much is happening there either. Especially no waiting is
happening here and no forwarding of data to other elements.
This fixes a race condition during connection setup: the connection
would immediately fail if we pass a packet from the peer to the socket
before bind() and connect() have returned.
This can't happen anymore as bind() and connect() have returned already
before both elements reach the PAUSED state, and in webrtcbin there is
an additional blocking pad probe before the decoder that does not let
any data pass through before that anyway.
The library is thread-safe by itself and potentially calls back into our
code, not only from the same thread but also from other threads. This
can easily lead to deadlocks if we try to hold our mutex on both sides.
Starting from WPEBackend-FDO 1.6.x, software rendering support is available.
This features allows wpesrc to be used on machines without GPU, and/or for
testing purpose. To enable it, set the `LIBGL_ALWAYS_SOFTWARE=true` environment
variable and make sure `video/x-raw, format=BGRA` caps are negotiated by the
wpesrc element.
Otherwise it can happen that e.g. the stream-start event is tried to be
sent as part of pushing the first buffer. Downstream might not be in
PAUSED/PLAYING yet, so the event is rejected with GST_FLOW_FLUSHING and
because it's an event would not cause the blocking pad probe to trigger
first. This would then return GST_FLOW_FLUSHING for the buffer and shut
down all of upstream.
To solve this we return GST_PAD_PROBE_DROP for all events. In case of
sticky events they would be resent again later once we unblocked after
blocking on the buffer and everything works fine.
Don't handle events specifically in sink pad blocking pad probes as here
downstream is not linked yet and we are actually waiting for the
following CAPS event before unblocking can happen.
Fixes https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/issues/1172
Without this it might happen that received data from the DTLS transport
is already passed to sctpdec before its state was set to PLAYING. This
would cause the data to be dropped, GST_FLOW_FLUSHING to be returned and
the whole DTLS transport to shut down.
Fixes https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/issues/1172
among other things.
Using a GCond can easily lead to deadlocks and only duplicates the
waiting code from gstpad.c in the best case.
In this case it actually could lead to a deadlock if both RTP and RTCP
were waiting. Only one of them would be woken up because g_cond_signal()
was used instead of g_cond_broadcast().
Change how content-length is set for HTTP POST headers, letting curl set
the header (given the content-length) instead of manually writing it.
This enables curl to know the content-length of the data.
In curl 7.66, if curl does not know the content-length (e.g. when
manually writing the header) curl will use Transfer-Encoding: chunked,
which might not be desired.
Use a double instead of a plain float for intermediary
property values, so we have enough bits to store INT_MAX
and it doesn't get rounded and wrapped to -1 when cast
back to a 32-bit integer.
Fixes criticals like
g_param_spec_int: assertion 'default_value >= minimum && default_value <= maximum' failed
when loading LADSPA plugins from the Linux Studio Plugins
Project (http://lsp-plug.in) in GStreamer.
Fixes https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/issues/1194
The avtpsink element is expected to transmit AVTPDUs at specific times,
according to GstBuffer timestamps. Currently, the transmission time is
controlled in software via the rendering synchronization mechanism
provided by GstBaseSink class. However, that mechanism may not cope with
some AVB use-cases such as Class A streams, where AVTPDUs are expected
to be transmitted at every 125 us. Thus, this patch introduces avtpsink
own mechanism which leverages the socket transmission scheduling
infrastructure introduced in Linux kernel 4.19. When supported by the
NIC, the transmission scheduling is offloaded to the hardware, improving
transmission time accuracy considerably.
To illustrate that, a before-after experiment was carried out. The
experimental setup consisted in 2 PCs with Intel i210 card connected
back-to-back running an up-to-date Archlinux with kernel 5.3.1. In one
host gst-launch-1.0 was used to generate a 2-minute Class A stream while
the other host captured the packets. The metric under evaluation is the
transmission interval and it is measured by checking the 'time_delta'
information from ethernet frames captured at the receiving side.
The table below shows the outcome for a 48 kHz, 16-bit sample, stereo
audio stream. The unit is nanoseconds.
| Mean | Stdev | Min | Max | Range |
-------+--------+---------+---------+---------+---------+
Before | 125000 │ 2401 │ 110056 │ 288432 │ 178376 |
After | 125000 │ 18 │ 124943 │ 125055 │ 112 |
Before this patch, the transmission interval mean is equal to the
optimal value (Class A stream -> 125 us interval), and it is kept the
same after the patch. The dispersion measurements, however, had
improved considerably, meaning the system is now consistently
transmitting AVTPDUs at the correct time.
Finally, the socket transmission scheduling infrastructure requires the
system clock to be synchronized with PTP clock so this patches modifies
the AVTP plugin documentation to cover how to achieve that.
This patch refactors gst_avtp_sink_start() by moving all socket
initialization code to its own function. This change prepares the code
to the next patch which will introduce avtpsink's own rendering
synchronization mechanism.
Current avtpsink code opens the AF_PACKET socket with SOCK_NONBLOCK
option. However, we actually want sendto() to block in case there isn't
available space in socket buffer.
This patch refactors both avtpsink and avtpsrc code so we use the
if_nametoindex() helper instead of building a request and issuing an
ioctl to get the if_index.
The receive bin should block buffers from reaching dtlsdec before
the dtls connection has started.
While there was code to block its sinkpads until receive_state
was different from BLOCK, nothing was ever setting it to BLOCK
in the first place. This commit corrects this by setting the
initial state to BLOCK, directly in the constructor.
In addition, now that blocking is effective, we want to only
block buffers and buffer lists, as that's what might trigger
errors, we want to still let events and queries go through,
not doing so causes immediate deadlocks when linking the
bin.
And free data with the correct free() function in the receive callback
by passing it to gst_buffer_new_wrapped_full() instead of
gst_buffer_new_wrapped().
When a pipeline is stopped (actually when the waylandsink element
state changes from PAUSED to READY) the video surface is cleared, but
the opaque black surface behind is not. Fix this by actually clearing
both surfaces.
We need the streams' pt maps updated before requesting pads
on rtpbin, because this is what will trigger the requesting
of FEC encoders, and our handler for this request looks for
the payload types in the relevant stream's pt map.
Fixes#1187
Otherwise we would start sending data to the DTLS connection before, and
the DTLS elements consider this an error.
Also RFC 8261 mentions:
o A DTLS connection MUST be established before an SCTP association can
be set up.
For us it can happen that the DTLS transports are still in the process
of connecting while the ICE transport is already completed. This
situation is not specified in the spec but conceptually that means it is
still in the process of connecting.
Fixes https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/issues/758
Previously we simply logged errors but never reported them to elements
or even to the user. Fatal errors are now properly reported.
Additionally proper connection closing is implemented based on EOS:
- dtlsenc: EOS will cause close_notify to be sent to the peer and only
if the peer also sent back close_notify we will forward the
EOS event.
- dtlsdec: EOS will be forwarded normally, this only means that the
unterlying transport was closed. On receiving a DTLS packet
containing close_notify, return EOS and send EOS downstream.
We don't have any mid before parsing the SDP, which happens after we
handled the SDP answer and that usually happens long after ICE candidate
gathering is finished.
Without this all transceivers are considered inactive and as such ICE
gathering is for active transceiver was considered complete from the
very beginning.
Fixes https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/issues/1126
We don't support stopping RTP receivers currently so let's not consider
them all stopped all the time. This fixes some of the ICE/DTLS state
change handling and specifically fixes the ICE gathering state.
Previously the ICE gathering state was immediately going from NEW to
COMPLETE because it considered all transceivers stopped and as such all
activate transceivers were finished gathering ICE candidates.
Fixes https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/issues/1126
xdg_shell fullscreen mode doesn't work for committing
xdg_surface without configure acknowledgement.
In addition, we can't set different surface setting from
acknowledged config in this mode.
AES128 support was added since nettle version 3.0
../subprojects/gst-plugins-bad/ext/hls/gsthlsdemux.h:110:10: error: field ‘ctx’ has incomplete type
struct CBC_CTX (struct aes128_ctx, AES_BLOCK_SIZE) aes_ctx;
Add static or dynamic mpd with:
- baseURL
- period
- adaptation_set
- representaton
- SegmentList
- SegmentURL
- SegmentTemplate
Support multiple audio and video streams.
Pass conformance test with DashIF.org
Instead of always going through the file system API we allow the
application to modify the behaviour. For the playlist itself and
fragments, the application can provide a GOutputStream. In addition the
sink notifies the application whenever a fragment can be deleted.
The SVT-HEVC (Scalable Video Technology[0] for HEVC) Encoder is an
open source video coding technology[1] that is highly optimized for
Intel Xeon Scalable processors and Intel Xeon D processors.
[0] https://01.org/svt
[1] https://github.com/OpenVisualCloud/SVT-HEVC
It cleans up videoFormat by combining the wl_shm_formats and
drm_formats into a single table that represents the same format.
In addition, it adds NV61 format to the waylandsink.
commit 6adfb120ab added this flag to fix
builds with `-Werror`, and afterwards it was changed to use a version
check when newer versions of openexr moved over to C++11.
However, some distros have backported patches to older openexr
versions which make it require C++11, which makes the version check
incorrect and causes an error because we passed `-Werror -std=c++98`.
Instead, directly check when usage of the header requires `-std=c++98`
with `-Werror` and override the `cpp_std` setting on the target.
Fixes https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/issues/1117
Includes a new GstVulkanHandlePool base class for pooling different
resources togther. The descriptor cache object is ported to
GstVulkanHandlePool with the exact same functionality.
A new GstVulkanFenceCache is also implemented for caching fences
which is used internally by GstVulkanDevice for creating or reusing
fences.
The existing GstVulkanTrashFenceList object now caches trash objects.
Part 1 is a base class (vkvideofilter) that handles instance, device,
queue retrieval and holding that has been moved to the library
Part 2 is a fullscreenrenderquad that is still in the plugin that
performs all of the previous vulkan-specific functionality.
Based on Stream ID, the application can accept or reject the connection,
select the desired data stream, or set an appropriate passphrase for the
connection. Example usage:
srt://127.0.0.1:1234?streamid=mystream
Current code would change any non-ok return from gst_pad_push to
GST_FLOW_ERROR, thus hiding meaningful returns such as GST_FLOW_EOS.
Tests also added.
Most of avtpcvfdepay messages are currently logged as warnings, which can
make some scenarios - such as receiving two AVTP streams on the same
pipeline - too verbose.
This patch tones those message down to INFO or DEBUG level - more in
sync with avtpaafdepay logging.
To allow curlhttpsrc to support DASH streams that use the on-demand
profile, it needs to support HTTP Range GETs. In GStreamer, the RANGE
is specified by issuing a GST_FORMAT_BYTES seek to set the start and
end of the range. curlhttpsrc needs to implement seek and set the
appropriate curl options to make it add the Range header to the
request.
The major functionality gain this provides is proper reference counting
for a descriptor set. Overall this allows us to create descriptor sets
when they are needed (or reused from a cache) without violating any of
vulkan's object synchronisation requirements.
As there are a fixed number of sets available in a pool, the number of
descriptors in elements is currently hardcoded to 32. This can be extended
in a future change to create pools on the fly if that limit is ever overrun.
Allows a cleaner control flow when there is no fence available for use
with the trash list. An always signalled fence type will always return
TRUE for gst_vulkan_fence_is_signalled.
The following build error occurs:
vkdeviceprovider.h:30:10: fatal error: gst/vulkan/vulkan.h: No such file or directory
#include <gst/vulkan/vulkan.h>
^~~~~~~~~~~~~~~~~~~~~
By passing NULL to `g_signal_new` instead of a marshaller, GLib will
actually internally optimize the signal (if the marshaller is available
in GLib itself) by also setting the valist marshaller. This makes the
signal emission a bit more performant than the regular marshalling,
which still needs to box into `GValue` and call libffi in case of a
generic marshaller.
Note that for custom marshallers, one would use
`g_signal_set_va_marshaller()` with the valist marshaller instead.
It provides to set tile-columns and tile-rows configurations. The av1
codec allows an input image frame be partitioned into separate vertical
or horizontal tile which can be encoded or decoded independently. It
helps to encode/decode parallel.
CURLOPT_DEBUGFUNCTION option replaces the standard debug function
used when CURLOPT_VERBOSE is in effect. This callback receives various debug information.
And only update the caps and stream-start event accordingly. This
ensures that we'll always forward sticky events that arrive after the
caption pad was created, and especially updates to existing sticky
events like the segment event.
Also create a proper stream id based on the upstream stream id for the
stream-start event, and make sure that all the sticky events we know are
already on the caption pad at the time it is added to the element.
clamp-to-border will return the border color which is typically black,
white or transparent. When linear filtering the edge pixels will
typeically be combined with the border color which is not typically what
we want. Especially when color converting, this removes a green box
around the edge when converting YUV->RGB.
This fixes a regression from commit "srtp: Support libsrtp2"
e9aa117200 where an internal
set of ssrc(s) was added because the libsrtp v2 keeps its
internal streams as private. But the change prevented that
ssrc(s) that not in the caps from being added to the stats.
This patch ensures that all ssrc(s) are inserted to this set
instead of only inserting those from the caps.
If the mutex is locked while running frameComplete there is a potential deadlock
bound to happen when we get a new exported images from the backend.
Fixes#1101
hlssink2 defined "max-files" property to decide the maximum number
of fragments which should be stored in disk. But we've not used
the property. Instead, the size has been maintained by "playlist-length".
Since "max-files" and "playlist-length" have different meaning,
the decision should be done by "max-files" property.
For example, an user might want expose only 3 fragments via playlist
but might want to keep more files than 3 in disk.
A classic case of not updating the next item to iterate after deleting
it from the singly linked list.
Only ever hit with a text buffer with GST_CLOCK_TIME_NONE for either the
timestamp or duration.
hlssink* elements could be finalized without EOS event, and in that case
the final playlist might not include the EXT-X-ENDLIST tag.
Since missing ENDLIST tag means it's live stream, but we did't intend it,
hlssink* elements should put the tag at the end.
In Debian, soundfonts in SF3 format (i.e. the same as SF2 format but
with Ogg/Vorbis-compressed samples) are installed into
/usr/share/sounds/sf3. Soundfonts in SF3 format are supported since
FluidSynth 1.1.7 (released in Feb 2018).
Internal sources seem to be rtp streams we are sending whereas
non-internal sources are the rtp streams we are receiving. Redo the
statistics with that in mind.
This reverts commit 68fa80e831.
Some wayland servers, especially weston, only expect empty input
region as a request to disable input.
Signed-off-by: Jeffy Chen <jeffy.chen@rock-chips.com>
The function _get_stats_from_ice_transport returns a string which must be
freed by the caller. However, _get_stats_from_dtls_transport was ignoring
the return value from this function, resulting in a leak.
Ran this with valgrind. Before this fix there was a leak of 40 bytes each
time this was called. After there was no leak.
/usr/include/mjpegtools/mpeg2enc/ontheflyratectlpass1.hh:1:9: error: '_ONTHEFLYRATECTLPASS1_HH' is used as a header guard here, followed by #define of a different macro [-Werror,-Wheader-guard]
#ifndef _ONTHEFLYRATECTLPASS1_HH
^~~~~~~~~~~~~~~~~~~~~~~~
/usr/include/mjpegtools/mpeg2enc/ontheflyratectlpass1.hh:2:9: note: '_ONTHELFYRATECTLPASS1_HH' is defined here; did you mean '_ONTHEFLYRATECTLPASS1_HH'?
#define _ONTHELFYRATECTLPASS1_HH
^~~~~~~~~~~~~~~~~~~~~~~~
_ONTHEFLYRATECTLPASS1_HH
In file included from ../subprojects/gst-plugins-bad/ext/mpeg2enc/gstmpeg2encoder.cc:31:
/usr/include/mjpegtools/mpeg2enc/ontheflyratectlpass2.hh:1:9: error: '_ONTHEFLYRATECTLPASS2_HH' is used as a header guard here, followed by #define of a different macro [-Werror,-Wheader-guard]
#ifndef _ONTHEFLYRATECTLPASS2_HH
^~~~~~~~~~~~~~~~~~~~~~~~
/usr/include/mjpegtools/mpeg2enc/ontheflyratectlpass2.hh:2:9: note: '_ONTHELFYRATECTLPASS2_HH' is defined here; did you mean '_ONTHEFLYRATECTLPASS2_HH'?
#define _ONTHELFYRATECTLPASS2_HH
^~~~~~~~~~~~~~~~~~~~~~~~
_ONTHEFLYRATECTLPASS2_HH
/usr/include/mjpegtools/mpeg2enc/encoderparams.hh:82:1: error: struct 'RateCtl' was previously declared as a class; this is valid, but may result in linker errors under the Microsoft C++ ABI [-Werror,-Wmismatched-tags]
struct RateCtl;
^
/usr/include/mjpegtools/mpeg2enc/ratectl.hh:50:7: note: previous use is here
class RateCtl
^
/usr/include/mjpegtools/mpeg2enc/encoderparams.hh:82:1: note: did you mean class here?
struct RateCtl;
^~~~~~
class
../subprojects/gst-plugins-bad/ext/aom/gstav1enc.c:415:34: warning: implicit conversion from enumeration type 'GstAV1EncEndUsageMode' to different enumeration type 'enum aom_rc_mode' [-Wenum-conversion]
av1enc->aom_cfg.rc_end_usage = DEFAULT_END_USAGE;
~ ^~~~~~~~~~~~~~~~~
../subprojects/gst-plugins-bad/ext/aom/gstav1enc.c:162:41: note: expanded from macro 'DEFAULT_END_USAGE'
#define DEFAULT_END_USAGE GST_AV1_ENC_END_USAGE_VBR
^~~~~~~~~~~~~~~~~~~~~~~~~
_decrypt_start() failure will lead to decryption failure eventually
but catching it earlier if possible. The decrpytion start failure means
that the hls plugin was built without crypto library or crypto library
does not want to accept given key and IV.
crypto libraries are not required for hlssink and hlssink2.
Also, hlsdemux with nonencrypted stream can work without crpyto.
Make an error only when users set "hls-crpyto" with non-auto option explicitly,
but no crpyto library was found.
On Windows, if libusrsctp and gstreamer are built with different
C runtimes (CRT), we cannot free memory allocated inside libusrsctp
with the `free()` function from gstreamer's CRT.
`usrsctp_freedumpbuffer()` simply calls `free()`, but because of the
way DLLs work on Windows, it will always call the free function from
the correct CRT.
Don't fixate profile caps which will choose the first profile from list.
Instead, store all profiles allowed by peer and try them until x265 can
accept one of them.
x265 does not allow user to configure a picture size smaller than
at least one CU size, and maxCUSize must be 16, 32, or 64.
Therefore, the CU size must be set according to the input resolution,
and the input resolution can not be less than 16.
When negotiating a data channel, Chrome as recent as 75 still uses SDP
based on version 05 of the SCTP SDP draft, for example:
m=application 9 DTLS/SCTP 5000
a=sctpmap:5000 webrtc-datachannel 1024
Implement support for parsing SCTP port out of SDP message with sctpmap
attribute. Fixes data channel negotiation with Chrome browser.
WebKit's websrc depends on the main-thread for download completion
rendezvous. This exposed a number of deadlocks in adaptivedemux due to
it holding the MANIFEST_LOCK during network requests, and also needing
to hold it to change_state and resolve queries, which frequently occur
during these download windows.
Make demux->running MT-safe so that it can be accessed without using the
MANIFEST_LOCK. In case a source is downloading and requires a MT-thread
notification for completion of the fragment download, a state change
during this download window will deadlock unless we cancel the downloads
and ensure they are not restarted before we finish the state-change.
Also make demux->priv->have_manifest MT-safe. A duration query happening
in the window described above can deadlock for the same reason. Other
src queries (like SEEKING) that happen in this window also could
deadlock, but I haven't hit this scenario.
Increase granularity of API_LOCK'ing in change_state as well. We need to
cancel downloads before trying to take this lock, since sink events
(EOS) will hold it before starting a fragment download.
The row based multi threading control was introduced after 1.0.0 version
of libaom released. It adds a guard to check the relevant control
definition declared. It fixes#1025
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
