Commit graph

6908 commits

Author SHA1 Message Date
cketti
2408fbe92d curlsmtpsink: Use correct email date format
See https://www.rfc-editor.org/rfc/rfc5322.html#section-3.3

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1317>
2020-06-05 08:22:34 +00:00
Matthew Waters
8396e16f85 ccconverter: signal cea608 padding as invalid
Outputting a valid but null cea608 byte pair may cause some issues with
some checksum packets.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1318>
2020-06-05 07:36:22 +00:00
Matthew Waters
bfd03537f0 ccconverter: also copy buffer metadata when draining
Fixes buffers without PTS/DTS/meta/etc when receiving an EOS with data
still stored in the internal scratch buffer.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1318>
2020-06-05 07:36:22 +00:00
Matthew Waters
00bbaff371 ccconverter: Output the limit hit in debug lines
Fix two case of the input triplet limit not applying in cea608 -> cdp
conversion.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1318>
2020-06-05 07:36:22 +00:00
Thibault Saunier
d9ffa3b3b2 doc: Fix spelling of GstWebRTCICE 2020-06-04 13:33:16 -04:00
Sebastian Dröge
74f2f733be plugins: Use gst_type_mark_as_plugin_api() for all non-element plugin types 2020-06-04 13:33:16 -04:00
Peter Workman
b98712c44a srtobject: continue polling or report error on failed receive
fixes #1277

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1260>
2020-06-03 09:31:42 +00:00
Jan Alexander Steffens (heftig)
a1bc9d4319 srt: Make logging regarding callers more useful
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1273>
2020-06-03 04:23:14 +00:00
Sebastian Dröge
b25d153c34 webrtc: Add GstWebRTCDataChannel to the library API
This makes it more discoverable for bindings and allows bindings to
generate static API for the signals and functions.

Fixes https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/issues/1168

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1313>
2020-06-02 21:04:37 +00:00
Ederson de Souza
bafdade207 avtp: Ensure that the avtp plugin is only built on Linux
It uses some Linux only features. This also prevents gst-build trying to
get libavtp on non-Linux environments.
2020-06-01 11:37:16 -07:00
Tim-Philipp Müller
00caf46e3f vulkan: fix use of assert() with older meson versions
Follow-up to !1307

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1308>
2020-05-28 22:50:00 +01:00
Tim-Philipp Müller
b75ad03313 vulkan: don't run tests or build lib if plugin isn't actually built
The unit tests only checked for vulkan_dep.found(), which can
be true if the libs are there but glslc was not found, in which
case the plugin wouldn't be built and the unit tests would fail
because of missing vulkan plugins.

Doesn't really make much sense to build the vulkan integration lib
either if we're not going to build the vulkan plugin, so just disable
both for now if glslc is not available.

Fixes #1301

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1307>
2020-05-28 19:07:32 +01:00
Matthew Waters
67ae885d4c webrtc: handle an ice-lite remote offer
When the remote peer offers an ice-lite SDP, we need to configure our
ICE negotiation to be in controlling mode as the peer will not be.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1304>
2020-05-28 19:57:45 +10:00
Jan Schmidt
918ed75944 srt: Don't leak the connection_poll_id on close()
Attempting to reach an inactive SRT peer in caller mode
was leaking an fd every few seconds in the gst_srt_object_close()/open()
loop.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1293>
2020-05-24 10:47:59 +00:00
Thibault Saunier
5a2c358a2b pitch: Remove useless restriction on number of channel
It handles any number of channels just fine

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1292>
2020-05-22 18:18:28 -04:00
Stéphane Cerveau
3f16353d64 meson: add libopenjp2 fallback for openjpeg
As a wrap is now available in gst-build, the fallback
can be used.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1270>
2020-05-21 08:36:59 +00:00
Ederson de Souza
a754c67c05 avtp: Add libavtp fallback dependence
So that libavtp can be found if added as subproject on gst-build.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1271>
2020-05-21 00:14:08 +00:00
Mats Lindestam
bf004227ec gstcurlhttpsink: Set 'Expect: 100-continue'-header
In the upgrade of libcurl from 7.64.1 to 7.69.1 the
EXPECT_100_THRESHOLD has been increased from 1 Kb to 1 Mb
(see https://curl.haxx.se/mail/lib-2020-01/0050.html).
This caused the gstcurlhttpsink to not being able to rewind
and resend in the case, e.g. response '401 Unauthorized'.
Now the 'Expect: 100-continue'-header is explicitly set in
the gstcurlhttpsink.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1276>
2020-05-18 13:46:24 +02:00
J. Kim
4ccaa1ebbb srtobject: add streamid property
The stream id starts with '#!::' according to SRT Access Control[1],
but GstURI requires URI encoded string.This commit introduces additional
property to set the id by normal string.

[1] https://github.com/Haivision/srt/blob/master/docs/AccessControl.md
2020-05-13 14:13:48 +00:00
Nirbheek Chauhan
e3d5849225 meson: Pass native: false to add_languages()
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>
2020-05-13 13:59:36 +05:30
Matthew Waters
6dae95d60f ccconverter: check fraction multiply for overflow
It should not happen and if it does, something went very wrong earlier

CID 1463350

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1262>
2020-05-12 16:06:32 +10:00
Matthew Waters
09cbe4cd03 ccconverter: tighten up a couple of NULL checks
CID 1463347
CID 1463346
CID 1463345

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1262>
2020-05-12 16:06:32 +10:00
Matthew Waters
ebc19d19bb ccconverter: fix unintialized read of mapped output info in error case
We only need to gst_buffer_unmap() if we have gst_buffer_map()ed.  In
most cases we can shorten the lenght of time we need to map the output
buffer.  Fix similar occurences elsewhere.

CID 1463349

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1262>
2020-05-12 16:06:32 +10:00
Matthew Waters
f077189809 ccconverter: fix uninitialized read in error case
CID 1463351

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1262>
2020-05-12 16:06:32 +10:00
Matthew Waters
12edc0d9b8 ccconverter: implement discont handling
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1116>
2020-05-11 12:30:31 +00:00
Matthew Waters
ba1558a7ab ccconverter: use a better padding byte sequence for writing cdp
0xf8 can be interpreted as cea608 data at the beginning of a cdp packet
as the cc_valid bit is not checked when cc_valid in (0b00 or 0b01).

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1116>
2020-05-11 12:30:31 +00:00
Matthew Waters
7ed0bc539f ccconverter: split temporary storage into 3
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>
2020-05-11 12:30:31 +00:00
Matthew Waters
3417a1709c ccconvert: compact input cc_data where possible
Skip over padding cc_data triples.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1116>
2020-05-11 12:30:31 +00:00
Matthew Waters
7d028af675 ccconverter: implement support for CDP framerate conversions
- Any format involving CDP is supported.
- Time codes (if present) are scaled as well.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1116>
2020-05-11 12:30:31 +00:00
Matthew Waters
75503017c2 ccconverter: introduce define for max cdp packet length
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1116>
2020-05-11 12:30:31 +00:00
Matthew Waters
44c874fd9e ccconverter: don't rely on external state in *_internal()
This allows using the _internal() variants for simply converting some
caption data without relying on any external state.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1116>
2020-05-11 12:30:31 +00:00
Matthew Waters
31a0bf367d ccconverter: cc_count limits are per framerate
Enforce this and add a test for cdp input being too large.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1116>
2020-05-11 12:30:31 +00:00
Matthew Waters
b25d75f201 ccconverter: refactor cdp id, fps, max_cc_count into a table
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1116>
2020-05-11 12:30:31 +00:00
Matthew Waters
afc120fa74 ccconverter: pivot to implementing generate_output
Will make a n-n buffer element much easier to implement.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1116>
2020-05-11 12:30:31 +00:00
Chris Ayoup
9937101e51 webrtc: move filtering properties to webrtcice
We want webrtcbin to only expose properties that are defined in JSEP, so
these additional properties should be moved out.  In order to access
them, the webrtcice instance is exposed from webrtcbin.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1223>
2020-05-11 05:30:59 +00:00
Chris Ayoup
ca754245e9 webrtc: allow setting local IP addresses
If a local IP address is specified, ICE gathering can be much faster
in environments where there are multiple IP addreses but only some are
usable (for example, if you are running docker on the machine).

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1223>
2020-05-11 05:30:59 +00:00
Chris Ayoup
3fc8818824 webrtc: Allow toggling TCP and UDP candidates
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>
2020-05-11 05:30:59 +00:00
Sebastian Dröge
d6f6c51f3c spanplc: Don't segfault when retrieving the stats property without a spanplc context
For example when trying to get the property value in NULL state.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1258>
2020-05-10 08:44:09 +00:00
Sebastian Dröge
77784c7aba musepackdec: Don't fail all queries if no sample rate is known yet
The sample rate is only needed for the POSITION/DURATION queries and we
would otherwise fail important queries like the CAPS query.

Fixes https://gitlab.freedesktop.org/gstreamer/gstreamer/-/issues/498

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1248>
2020-05-06 08:51:38 +00:00
Luka Blaskovic
4cf362e2df opencv: allow compilation against 4.3.x
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1235>
2020-05-06 06:49:08 +00:00
Matthew Waters
02c8e66ff1 webrtc: fix an off-by-one calculating low-threshold
We were not signalling low-threshold when the previous amount was at
exactly the low-threshold mark.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1247>
2020-05-06 15:49:58 +10:00
Matthew Waters
18de5f8f04 webrtc: remove debugging leftover
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1247>
2020-05-06 15:49:58 +10:00
Matthew Waters
50644f5718 webrtc: always reply to a promise
Otherwise, we defeat the purpose of a promise.

We were not replying when the state was closed.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1240>
2020-05-06 02:53:27 +00:00
Matthew Waters
1f395e3ddb webrtc: name threads based on the element name
Makes debugging a busy loop possibly easier

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1240>
2020-05-06 02:53:27 +00:00
Matthew Waters
d552c6556c webrtc: correctly use the pad template
GstHarness uses this for releasing request pads correctly. Fixes
numerous leaks in the webrtc unit tests.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1240>
2020-05-06 02:53:27 +00:00
Matthew Waters
46176fbcc7 webrtc: Fix a couple of renegotiation races
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>
2020-05-06 02:53:27 +00:00
Edward Hervey
75289d83a1 iqa: Fix all leaks in error path
CID #1456049
CID #1456080
CID #1456083

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1244>
2020-05-05 17:33:20 +00:00
Matthew Waters
3baf0d5dc4 sctp: enable usrsctp debug when supported
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1234>
2020-05-05 03:38:06 +00:00
Ederson de Souza
b68e47968b avtpsink: Log that AVTPDU transmission failure is due lateness
As ENOBUFS is not really clear about what is going on, let's check
socket error queue to see if packets are being dropped due being late.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1004>
2020-05-02 17:42:15 +00:00
Ederson de Souza
32281ddd33 avtpsink: Accept buffers that fall out of segment
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>
2020-05-02 17:42:15 +00:00
Ederson de Souza
7edaeb3fae avtpcvfpay: Warn about timestamp issues on non-flushing seek
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>
2020-05-02 17:42:15 +00:00
Ederson de Souza
12838af353 avtpcvfpay: Ensure NAL fragments are transmitted following stream specs
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>
2020-05-02 17:42:15 +00:00
Matthew Waters
b266652043 webrtcbin: also mark data channel transports as active
Fixes negotiation of a bundled sdp with only a data channel.

Without marking the transport as active, we would never unblock the
transportreceivebin and thus no data would ever reach us.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1231>
2020-05-01 03:13:46 +00:00
Matthew Waters
ce9b41f5d4 webrtcbin: fix bundle none case with remote offer bundling
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>
2020-05-01 03:13:46 +00:00
Vedang Patel
e47fa2006f avtp: Introduce the CRF Check element
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.
2020-04-30 23:31:25 +00:00
Vedang Patel
12ad2a4bcd avtp: Introduce the CRF Sync Element
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.
2020-04-30 23:31:25 +00:00
krivoguzovVlad
b769af0c4f Update gstsrtobject.c
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/999>
2020-04-30 18:57:13 +00:00
Matthew Waters
80ede09193 webrtcbin: only start gathering on local descriptions
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>
2020-04-30 14:47:55 +00:00
Seungha Yang
0b102d22ec webrtc: Correct symbol visibility to fix build warning on Windows
GstWebRTCDataChannel is fully internal of plugin

webrtcdatachannel.c(50): warning C4273: 'gst_webrtc_data_channel_get_type': inconsistent dll linkage

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1225>
2020-04-30 10:27:47 +00:00
Thibault Saunier
a3595f7e0f lv2: Namespace global variables and explicitly make them private
And fix a LV2_PORT_GROUPS__rearLeft/LV2_PORT_GROUPS__rearRight typo

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1125>
2020-04-29 19:49:45 +00:00
Debarshi Ray
a0cd455dd0 lv2: Make it build with -fno-common
GCC 10 defaults to -fno-common. This means that global variables shared
across multiple translation units should be declared as 'extern' in
header files and defined in exactly one C file. See:
https://gcc.gnu.org/gcc-10/porting_to.html

https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1125

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1125>
2020-04-29 19:49:45 +00:00
Seppo Yli-Olli
90f374dd0c openh264: memcmp return value 0 means match
Commit e2aa76db79 introduced version
check guard for OpenH264 binary. There was a boolean error in
memcmp so matching OpenH264 was erroneously rejected.
Fixes #1278

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1219>
2020-04-27 15:40:10 +00:00
Mathieu Duponchelle
62d1a3a143 cccombiner: don't drop buffers on video timestamp discontinuities
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>
2020-04-24 08:47:50 +00:00
Mathieu Duponchelle
f02300eef5 cccombiner: handle gap buffers adequately
- Don't try to map them as actual CC data, that was raising
  a critical

- Consume video buffers up to the end of the gap

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1207>
2020-04-24 08:47:50 +00:00
Guillaume Desmottes
4e9030a0b6 spanplc: add 'stats' property
Allow users to retrieve the number of samples, and their duration,
generated using PLC.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1210>
2020-04-23 23:15:29 +00:00
Seppo Yli-Olli
e2aa76db79 Have strict version check for OpenH264 to avoid ABI issues
This fixes #1274 and no longer trusts soname alone

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1206>
2020-04-23 22:52:23 +00:00
Jan Alexander Steffens (heftig)
97c05d3f4b
srt: Accumulate total bytes sent/received over all connections/callers
So we don't lose them. Split gst_srt_object_open_internal for internal
reconnections that don't reset the accumulated bytes.
2020-04-15 10:42:48 +02:00
Jan Alexander Steffens (heftig)
d19b3fccb5
srt: Fix type of bytes-received-lost
The field is a uint64_t.
2020-04-15 10:42:47 +02:00
Jan Alexander Steffens (heftig)
132e3a1af9
srt: Remove use of closures for signal emission
It seems overly complicated.
2020-04-15 10:42:47 +02:00
Jan Alexander Steffens (heftig)
d2d00e07ac
srt: Clean up locking
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.
2020-04-15 10:42:47 +02:00
Jan Alexander Steffens (heftig)
37ee389913
srt: Remove trailing whitespace 2020-04-15 10:42:47 +02:00
Philippe Normand
991bcb22d5 wpe: Add support for SHM without requiring EGLDisplay
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.
2020-04-13 11:53:16 +00:00
J. Kim
04f3f4be4f srtobject: fix mutex lock target
GstSRTObject is a structure that has an actual GstElement
which is extended to srt{src,sink}.
2020-04-13 15:23:46 +09:00
Zeid Bekli
663cd44ef0 srtp: Added support for BYE packet
SRTCP can't get SSRC from BYE packet, this will make srtpdec element
to drop the package. Adding support to get the SSRC from BYE packets.
2020-04-09 15:11:19 +00:00
Stéphane Cerveau
d59bd5f674 dash: fix VARARGS coverity error
va_end was not called in every code path due to
g_return_val_if_fail.

API usage errors  (VARARGS)
va_end was not called for "myargs".

CID: 1461294
2020-04-08 20:02:57 +00:00
worldofpeace
f10b424418 meson: build with neon 0.31
No API/ABI changes https://github.com/notroj/neon/blob/0.31.0/NEWS#L3
2020-04-03 18:50:16 -04:00
Nirbheek Chauhan
387b6df948 meson: Don't use get_option('buildtype')
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
2020-04-03 17:07:47 +05:30
Miguel Paris
45a1070203 srtpdec: reduce log level for replay cases
These are normal cases, so DEBUG level is enough.
2020-04-01 17:45:15 +00:00
Miguel París Díaz
ed71e262b0 srtpdec: do not warning old replay errors
Reordered packets producing decrypting errors are very normal,
so we should filter which errors are warning and which not.
2020-04-01 17:45:15 +00:00
Miguel Paris
075ff1e8b0 srtpdec: fix reseting RTP sequence number on ROC changes
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.
2020-04-01 16:49:44 +02:00
Seungha Yang
770a851e03 x265enc: Update for video-hdr struct change
See the change of -base https://gitlab.freedesktop.org/gstreamer/gst-plugins-base/-/merge_requests/594
2020-04-01 05:18:11 +00:00
Matthew Waters
8da177c0bf dtls/connection: fix EOF handling with openssl 1.1.1e
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.
2020-03-27 11:43:53 +11:00
Matthew Waters
319a5e5779 webrtc: mark streams as active on renegotiation as well.
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.
2020-03-25 14:46:15 +11:00
Philippe Normand
26f76dd927 wpe: Enable SHM support for new stable WPEBackend-FDO release
1.5.0 was the development version.
2020-03-23 13:08:46 +00:00
Philippe Normand
49560b4ba8 wpe: Mouse scroll events support 2020-03-23 13:08:46 +00:00
Philippe Normand
158a2b3fd1 webrtcdsp: Fix documentation markup 2020-03-15 12:44:31 +00:00
Philippe Normand
b36e36f74a openni2: Remove spurious gtk-doc markers 2020-03-15 10:47:02 +00:00
yychao
cb0e4bffea smoothstreaming: fix H264 CodecPrivateData parsing
Do not pass SPS nal_unit_type (0x67) into gst_h264_parse_sps()

Fixes #648
2020-03-10 12:55:05 +00:00
Sebastian Dröge
5a2053e0af webrtcbin: Use GPtrArrays or store items inline instead of using GArrays of pointers 2020-03-09 21:38:42 +02:00
Jan Schmidt
8274fcd311 webrtcbin: Prevent ICE gathering state reaching complete early
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.
2020-03-10 05:47:40 +11:00
Jan Schmidt
9410ef56b8 webrtc: Protect the pending ICE candidates array
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
2020-03-10 05:25:40 +11:00
Jan Schmidt
ad53de1da1 webrtc: Don't crash in ICE gathering
Fix a crash collating ICE gathering states if there are
unassociated transceivers in the list with no TransportStream
2020-03-04 23:06:52 +00:00
Jan Schmidt
905988c63f webrtc: Unblock transportreceivebin for send-only bundled streams
If there is any active mline in a bundle, we need to unblock
the transportreceivebin for DTLS setup and RTCP reception,
otherwise no data can ever start flowing.

Fixes https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/issues/1206
2020-03-04 10:15:19 +00:00
Jan Schmidt
cb48733ff3 webrtc: Remove RECEIVE_STATE_DROP from transportreceivebin
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
2020-03-04 10:15:19 +00:00
Matthew Waters
0f1ba5b2f2 dash: add build-dep on pbutils
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
2020-03-03 06:34:40 +00:00
Matthew Waters
d66743e482 vulkan/sink: implement GstNavigation support 2020-03-03 05:00:50 +00:00
Jan Schmidt
8e3472faee webrtc: Use the dtlssrtenc rtp-sync property
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
2020-02-27 12:30:32 +00:00
Jan Schmidt
0c72a41767 gstdtlsrtpenc: Add rtp-sync property
Add an rtp-sync property which synchronises RTP streams
to the pipeline clock before passing them to funnel for
merging with RTCP.

https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/issues/1212
2020-02-27 12:30:32 +00:00
Nirbheek Chauhan
a06ddd182d dash: Don't use sscanf + glib format modifiers
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
2020-02-27 09:42:33 +00:00
Sebastian Dröge
cc8b90967b dtls: Set a random serial number and issuer/subject in the self-signed certificates
This is also what Chrome and Firefox are doing, citing privacy concerns.
Also putting OpenWebRTC from Sweden as issuer/subject is rather
confusing.
2020-02-27 08:27:19 +00:00
Jan Schmidt
499be261cd webrtc: Configure transportsendbin latency internally
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
2020-02-21 13:42:05 +11:00
Jan Schmidt
96a407334d webrtc: Merge ICE candidates to local descriptions
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
2020-02-17 14:23:56 +00:00
Sebastian Dröge
f156ee1da4 webrtcbin: Block the source pads before dtlssrtpdec inside transportreceivebin
Otherwise dropped sticky events are not actually re-sent on the next
opportunity and we can end up with data-flow before stream-start/segment
events.
2020-02-12 16:54:42 +00:00
Sebastian Dröge
26a6b17593 sctp: Take some socket configurations from Firefox's datachannel code
- 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
2020-02-12 16:11:15 +00:00
Sebastian Dröge
c497370254 sctp: Start connection synchronously when starting the association
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.
2020-02-12 16:11:15 +00:00
Sebastian Dröge
4c5c6e68c6 sctp: Switch back to a non-recursive mutex and don't hold it while calling any usrsctp functions
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.
2020-02-12 16:11:15 +00:00
Philippe Normand
9ac798ae5e wpe: Add software rendering support support
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.
2020-02-11 16:47:53 +00:00
Jan Alexander Steffens (heftig)
e2cefdd6ff fluiddec: Move logging init into plugin_init
This is a nicer place to keep it. We also initialize it before touching
the drivers.

https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/merge_requests/1026
2020-02-11 12:10:50 +00:00
Jan Alexander Steffens (heftig)
9aa12399a8 fluiddec: Keep fluidsynth from probing audio drivers
It might cause problems and we don't need the drivers anyway. This also
avoids a bunch of stderr spam from the drivers.

https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/merge_requests/1026
2020-02-11 12:10:50 +00:00
Jan Alexander Steffens (heftig)
c35e80dc0e fluiddec: Avoid deprecated fluid_synth_set_sample_rate
This function is used to change the rate at runtime, which has issues:
https://github.com/FluidSynth/fluidsynth/issues/585

Use the settings key instead (which already defaults to 44100, but I did
test other rates).

https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/merge_requests/1026
2020-02-11 12:10:50 +00:00
Sebastian Dröge
4ffa6350e8 webrtc: In all blocking pad probes except for sink pads also handle serialized events
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
2020-02-11 00:49:51 +00:00
Sebastian Dröge
c16d4d2c33 webrtcbin: Add a blocking pad probe for the receivebin -> sctpdec connection
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.
2020-02-11 00:49:51 +00:00
Sebastian Dröge
f8fa71da27 webrtcbin/transportreceivebin: Use actual pad blocks instead of an additional GCond for blocking pads
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().
2020-02-11 00:49:51 +00:00
Sebastian Dröge
1ecb27f221 webrtc/transportsendbin: Clean up pad probe removal
We already have a helper function for this so just use it instead of
duplicating it.
2020-02-11 00:49:51 +00:00
Ederson de Souza
916966606b avtp: Build with clang
Minor non-conformity on AVTP code made it not compile with clang.
2020-02-07 21:53:57 +00:00
Ederson de Souza
f1976e0de5 avtp: Plug several leaks
After finally running tests with valgrind enabled, some leaks were found
- both on code and on tests themselves. This patch plugs them all!
2020-02-07 21:53:57 +00:00
Ludvig Rappe
2d585f2b0b gstcurlhttpsink: Update HTTP header for curl 7.66
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.
2020-02-07 13:24:53 +00:00
Tim-Philipp Müller
dbb0e71e70 ladspa: only multiply bounded rate properties by sample rate
We don't want to accidentally multiply G_MAXFLOAT or -GMAXFLOAT
with the sample rate.
2020-02-06 10:15:12 +00:00
Tim-Philipp Müller
ffd3e189de ladspa: fix unbounded integer properties
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
2020-02-06 10:15:12 +00:00
Andre Guedes
352bf28a35 avtpsink: Implement synchronization mechanism
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.
2020-02-05 22:28:12 +00:00
Andre Guedes
4f0dc8cf58 avtpsink: Prepare code to new synchronization mechanism
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.
2020-02-05 22:28:12 +00:00
Andre Guedes
cd03c48f88 avtpsink: Remove SOCK_NONBLOCK from avtpsink
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.
2020-02-05 22:28:12 +00:00
Andre Guedes
e74c807633 avtp: Refactor if_index code
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.
2020-02-05 22:28:12 +00:00
Stéphane Cerveau
4b72e8cad5 fdkaacdec: add support for mpegversion=2
Fix for #1199
2020-02-04 07:52:22 +00:00
Mathieu Duponchelle
f8eef0aba0 webrtcbin: fix blocking of receive bin
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.
2020-02-01 01:46:57 +01:00
Sebastian Dröge
af32ca45fa sctpassociation: Add missing return to prevent double unlock 2020-01-31 08:55:10 +02:00
Sebastian Dröge
e6c6b5ea29 sctpenc: Report errors when sending out data and the association is in error or disconnected state 2020-01-31 08:55:10 +02:00
Sebastian Dröge
6d22e80f30 sctp: Clean up association state handling and go into error/disconnected state in more circumstances 2020-01-31 08:55:10 +02:00
Sebastian Dröge
8612da865e sctpassociation: Use GStreamer logging system instead of g_warning() and g_log() 2020-01-31 08:55:10 +02:00
Sebastian Dröge
ddcfde36fa sctp: Add more logging to the encoder/decoder elements on data processing
And convert g_warning()s into normal log output instead.
2020-01-31 08:55:10 +02:00
Sebastian Dröge
db16265d86 sctpenc: Correctly log/handle errors and handle short writes 2020-01-31 08:55:10 +02:00
Sebastian Dröge
e9df80b235 sctp: Constify buffers in callbacks and functions
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().
2020-01-31 08:54:49 +02:00
Sebastian Dröge
fa0a233fa7 sctp: Make receive/packetout callbacks thread-safe 2020-01-30 16:07:48 +02:00
Sebastian Dröge
bff33f3b21 sctp: Add logging and missing cleanup on errors when creating pads 2020-01-30 16:00:33 +02:00
Sebastian Dröge
16ec86faf0 sctpenc: Use g_signal_emit() instead of g_signal_emit_by_name()
We have all the required information around so make use of it.
2020-01-30 15:59:12 +02:00
Sebastian Dröge
90e9f12880 sctpenc: Propagate downstream flow errors upstream
Fixes https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/issues/1180
2020-01-30 15:58:30 +02:00
Sebastian Dröge
1f9c1aa489 sctpdec: Use a flow combiner for the source pad flow returns and propagate errors upstream
Fixes https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/issues/1180
2020-01-30 15:56:36 +02:00
Guillermo Rodríguez
62ac77e620 waylandsink: Clear window when pipeline is stopped
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.
2020-01-28 13:22:36 +01:00
Sebastian Dröge
0478e2dc1a ccconverter: Fill remainder of the cc_data in CDP packets with empty packets
Instead of filling it completely with zeroes. Filling with zeroes is
considered invalid by various CC implementations.
2020-01-24 09:26:28 +00:00
Mathieu Duponchelle
7cc185bd86 webrtcbin: connect rtp funnel after updating ptmaps
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
2020-01-21 11:17:38 +00:00
Sebastian Dröge
0c39068c89 webrtcbin: Start datachannel SCTP elements only after the DTLS connection is established
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.
2020-01-19 11:16:34 +00:00
Sebastian Dröge
2798a80ebe webrtcbin: Add handling of unspecified peer-connection-state situation
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
2020-01-19 11:16:34 +00:00
Sebastian Dröge
4b73322333 webrtcbin: Return the old state if we ended up being in an unspecified situation
Previously we would've returned NEW, which is usually more wrong.
2020-01-19 11:16:34 +00:00
Sebastian Dröge
22869356db webrtcbin: Fix transitions for the peer connection state
They're now mapping exactly to what
  https://www.w3.org/TR/webrtc/#rtcpeerconnectionstate-enum
actually specifies.

Related to https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/issues/758
2020-01-19 11:16:34 +00:00
Sebastian Dröge
41175f4ebe webrtcbin: Fix transitions for the connection state
They're now mapping exactly to what
  https://www.w3.org/TR/webrtc/#dom-rtciceconnectionstate
actually specifies.

Related to https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/issues/758
2020-01-19 11:16:34 +00:00
Sebastian Dröge
7fcfb6c6c5 dtls: Keep track of the connection state and signal it through all the layers
This allows the application to keep track of the underlying DTLS
connection state and act accordingly.
2020-01-19 11:16:34 +00:00
Sebastian Dröge
d66aa872ca dtls: Handle errors/close_notify at all steps and propagate through the layers properly
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.
2020-01-19 11:16:34 +00:00
Sebastian Dröge
a132138f1c dtls: Propagate write errors backwards through dtlsenc/dtlsconnection 2020-01-19 11:16:34 +00:00
Sebastian Dröge
ee55dac8d4 dtls: Use a plain function pointer instead of a GClosure for the send callback
There's not point in using GClosure and going through all the
GValue/libffi infrastructure for each DTLS packet.
2020-01-19 11:16:34 +00:00
Sebastian Dröge
47ce34d32c webrtcbin: Don't consider transceivers without mid as inactive during ICE gathering state updates
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
2020-01-19 10:47:59 +00:00