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Author SHA1 Message Date
Jan Alexander Steffens (heftig)
4eeff95f92 srtsrc: Prevent delay from being negative 
`delay` should be a GstClockTimeDiff since SRT time is int64_t.

All values are in local time so we should never see a srctime that's in
the future. If we do, clamp the delay to 0 and warn about it.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1674>
 2020-10-12 12:58:22 +00:00
Jan Alexander Steffens (heftig)
ec11ad9d55 srtsrc: Don't calculate a delay if the srctime is 0 
A zero srctime is a missing srctime. Apparently this can happen when
["the connection is not between SRT peers or if Timestamp-Based Packet
Delivery mode (TSBPDMODE) is not enabled"][1] so it may not apply to us,
but it's best to be defensive.

[1]: https://github.com/Haivision/srt/blob/v1.4.2/docs/API.md#sending-and-receiving

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1674>
 2020-10-12 12:58:22 +00:00
Jan Alexander Steffens (heftig)
6b2fcb52e5 srtsrc: Defend against missing clock 
If we don't have a clock, stop the source instead of asserting in
gst_clock_get_time. This can happen when the element is removed from the
pipeline while it's playing.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1674>
 2020-10-12 12:58:22 +00:00
Olivier Crête
8a0d1d85cf dtlsconnection: Ignore OpenSSL system call errors 
OpenSSL shouldn't be making real system calls, so we can safely
ignore syscall errors. System interactions should happen through
our BIO. So especially don't look at the system's errno, as it
should be meaningless.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1656>
 2020-10-10 15:34:21 +00:00
Jan Alexander Steffens (heftig)
c6eeead1e4 srt: Consume the error from gst_srt_object_write 
Instead of leaking it.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1668>
 2020-10-09 07:47:47 +00:00
Jan Alexander Steffens (heftig)
2a7fa67693 srt: Check socket state before retrieving payload size 
The connection might be broken, which we should detect instead of just
aborting the write.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1669>
 2020-10-09 07:12:04 +00:00
Jakub Adam
6f2f15b5fb x265enc: fix deadlock on reconfig 
Don't attempt to obtain encoder lock that is already held by
gst_x265_enc_encode_frame().

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1671>
 2020-10-09 06:39:36 +00:00
Edward Hervey
dd11e91c3b srtsrc: Fix timestamping 
SRT provides the original timestamp of a packet (with drift/skew corrected for
local clock), which is what should be used for timestamping the outgoing
buffers. This ensures that we output the packets with the same timestamp (and by
extension rate) as the original feed.

Also detect if packets were dropped (by checking the sequence number) and
properly set DISCONT flag on the outgoing buffer.

Finally answer the latency queries

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1658>
 2020-10-08 21:12:17 +00:00
Sebastian Dröge
cc7e98816f Revert "webrtc: Save the media kind in the transceiver" 
This reverts commit f54d8e9945.

It breaks the CI until the C# bindings are fixed.
 2020-10-08 18:53:12 +03:00
Sebastian Dröge
849839ba97 Revert "rtptransceiver: Store the SSRC of the current stream" 
This reverts commit d1da271f25.

It breaks the CI until the C# bindings are fixed.
 2020-10-08 18:53:07 +03:00
Sebastian Dröge
e65a8cbcf1 Revert "webrtcbin: Remove unused function" 
This reverts commit 39723dbe93.

It breaks the CI until the C# bindings are fixed.
 2020-10-08 18:53:04 +03:00
Sebastian Dröge
b565a7ef66 Revert "webrtc: Set the DSCP markings based on the priority" 
This reverts commit 8ba08598bb.

It breaks the CI until the C# bindings are fixed.
 2020-10-08 18:53:00 +03:00
Olivier Crête
8ba08598bb webrtc: Set the DSCP markings based on the priority 
This matches how the WebRTC javascript API works and the Chrome implementation.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1425>
 2020-10-06 16:49:08 -04:00
Olivier Crête
39723dbe93 webrtcbin: Remove unused function 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1425>
 2020-10-06 16:49:08 -04:00
Olivier Crête
d1da271f25 rtptransceiver: Store the SSRC of the current stream 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1425>
 2020-10-06 16:49:08 -04:00
Olivier Crête
f54d8e9945 webrtc: Save the media kind in the transceiver 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1425>
 2020-10-06 16:49:08 -04:00
Jan Alexander Steffens (heftig)
92dc2f4192 srt: Remove unused sa_family tracking 
Now that SRT no longer needs the family when creating the socket, this
code has become useless.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1659>
 2020-10-06 13:56:32 +02:00
Niklas Hambüchen
13c8bda531 srt: Move off deprecated srt_socket(). 
See 73ee1e1a3e/docs/API-functions.md (srt_socket)

`srt_create_socket()` was added in
4b897ba92d
and srt `v1.3.0` is the first release that has it.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1659>
 2020-10-06 13:56:32 +02:00
Jan Alexander Steffens (heftig)
4e26b447f6 srt: Register a log handler 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1659>
 2020-10-06 12:39:04 +02:00
Jan Alexander Steffens (heftig)
936f422764 srt: Avoid removing invalid sockets from the polls 
This would provoke error messages from SRT.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1659>
 2020-10-06 12:39:00 +02:00
Jan Alexander Steffens (heftig)
fda4cfd15e srt: Fix use of srt_startup 
`srt_startup` can also return 1 if it was successful. Avoid warning in
this case.

Avoid a race when checking whether we need to call it at all.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1659>
 2020-10-06 12:38:57 +02:00
Jan Alexander Steffens (heftig)
6b8c4a5f34 srt: Fix parameter types used for socket options 
The [SRT documentation][1] specifies exact types for the socket options.
Make sure we match these.

This reverts the linger workaround in commit 84f8dbd932
and extends srt_constant_params to support other types than int.

[1]: https://github.com/Haivision/srt/blob/master/docs/APISocketOptions.md

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1659>
 2020-10-06 12:36:40 +02:00
Lars Lundqvist
9ded00bcf0 curlbasesink: Add curl seek callback 
Adding functionality to handle SEEK_SET enables rewinding of sent data.
In the HTTP case, this happens after an HTTP 401 has been received from
the other end. This will result in the sent data being resent.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1616>
 2020-10-01 10:40:14 +00:00
Matthew Waters
b003387526 wpesrc: fix some caps leaks using the non-GL output 
Always chain up to the parent _stop() implementation as it unrefs some
caps (among other things).

Fixes: https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/issues/1409
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1618>
 2020-09-30 12:10:44 +00:00
Hosang Lee
f7a8ece5ef smoothstreaming: clear live adapter on seek 
In live streaming, buffers sent by souphttpsrc are pushed to the live
adapter. The buffers in the adapter are sent out of mssdemux when it
is greater than 4096 bytes.

Occasionally, when seeking in live streams, if seek occurs just
after the last data chunk was received, and if this data chunk is
smaller than 4096 bytes, it will be kept in the live adapter.
This remaining data in the live adapter will be erroneously prepended
to the new data that is downloaded after seek and pushed out.
When qtdemux receives this data, since it does not start with
a moof box, it is impossible to demux the fragment, and bogus
size error will occur.

Clear out the live adapter on seek so that no unnecessary remaining
data is pushed out together with the new fragment after seeking.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1345>
 2020-09-30 11:48:02 +00:00
Ederson de Souza
8335039ecd tests/avtp: Fix coverity issues 
Fixes sign extension issues, unchecked return values and some constant
expression results.

CID: 1465073, 1465074, 1465075, 1465076, 1465077
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1398>
 2020-09-28 18:40:43 +00:00
Ederson de Souza
38d3360edb avtp: Change "%lu" for G_GUINT64_FORMAT 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1398>
 2020-09-28 18:40:43 +00:00
raghavendra
84f8dbd932 srtobject: typecast SRTO_LINGER to linger 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1615>
 2020-09-25 22:00:26 +05:30
Philippe Normand
2e8927ce93 wpe: Plug event leak 
Handled events don't go through the default pad event handler, so they need to
be unreffed in this case.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1568>
 2020-09-21 16:39:57 +00:00
Jan Schmidt
6fc7455881 wpesrc: Don't crash if WPE doesn't generate a buffer. 
On creating a 2nd wpesrc in a new pipeline in an app that already
has a runnig wpesrc, WPE sometimes doesn't return a buffer on request,
leading to a crash. This commit fixes the crash, but not the underlying
failure - a 2nd wpesrc can still error out instead.

Partially fixes https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/issues/1386

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1568>
 2020-09-21 16:39:57 +00:00
Philippe Normand
c3659cd611 wpe: Plug SHM buffer leaks 
Fixes #1409

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1568>
 2020-09-21 16:39:57 +00:00
Philippe Normand
8ef30a9ce5 wpe: Move webview load waiting to WPEView 
As waiting for the load to be finished is specific to the WebView, it should be
done from our WPEView, not from the WPEContextThread. This fixes issues where
multiple wpesrc elements are created in sequence. Without this patch the first
view might receive erroneous buffer notifications.

Fixes #1386

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1568>
 2020-09-21 16:39:57 +00:00
Philippe Normand
b707454a5a wpe: Use proper callback for TLS errors signal handling 
The load-failed and load-failed-with-tls-errors signals expect distinct callback
signatures.

Fixes #1388

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1566>
 2020-09-21 14:11:15 +00:00
Olivier Crête
825a79f01f webrtcbin: Accept end-of-candidate pass it to libnice 
libnice now supports the concept of end-of-candidate, so use the API
for it. This also means that if you don't do that, the webrtcbin will
never declared the connection as failed.

This requires bumping the dependency to libnice 0.1.16

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1139>
 2020-09-18 18:40:58 -04:00
Olivier Crête
63f06d16db webrtcbin: Merge the RTX SSRCs from all transceivers when bundling 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1545>
 2020-09-18 14:20:03 +00:00
Marian Cichy
c145798876 avtp: avtpaafdepay: fix crash when building caps 
gst_caps_new_simple gets wrong types for rate and channel which
may lead to a crash.

As 64-bit values for rate, depth, format, channels does not
make much sense and since any other functionality in gstreamer
expects G_TYPE_INT for channels and rate, we should stick to that

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1576>
 2020-09-17 08:32:07 +00:00
Emmanuel Gil Peyrot
f97b718b4c waylandsink: Use memfd_create() when available 
This (so-far) Linux- and FreeBSD-only API lets users create file
descriptors purely in memory, without any backing file on the filesystem
and the race condition which could ensue when unlink()ing it.

It also allows seals to be placed on the file, ensuring to every other
process that we won’t be allowed to shrink the contents, potentially
causing a SIGBUS when they try reading it.

This patch is best viewed with the -w option of git log -p.

It is an almost exact copy of Wayland commit
6908c8c85a2e33e5654f64a55cd4f847bf385cae, see
https://gitlab.freedesktop.org/wayland/wayland/merge_requests/4

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1577>
 2020-09-15 19:17:12 +00:00
Matthew Waters
e2d88f0569 webrtc: propagate more errors through the promise 
Return errors on promises when things fail where available.

Things like parsing errors, invalid states, missing fields, unsupported
transitions, etc.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1565>
 2020-09-14 04:04:29 +00:00
Adam Williamson
52ef192526 opencv: set opencv_dep when option is disabled (#1406) 
The examples build file checks opencv_dep, so it still needs to
be set even if the option is disabled.

Fixes https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/issues/1406

Signed-off-by: Adam Williamson <awilliam@redhat.com>
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1570>
 2020-09-11 07:16:21 +00:00
Mathieu Duponchelle
c096d30f6b openh264dec: port to new request_sync_point() API 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1571>
 2020-09-10 23:44:50 +02:00
Mathieu Duponchelle
c58357fb66 line21enc: add remove-caption-meta property 
Similar to #GstCCExtractor:remove-caption-meta

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1554>
 2020-09-09 22:11:28 +02:00
Mathieu Duponchelle
c07e2a89ba line21enc: heavily constrain video height 
We can only determine a correct placement for the CC line
with:

* height == 525 (standard NTSC, line 21 / 22)
* height == 486 (NTSC usable lines + 6 lines for VBI, line 1 / 2)

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1554>
 2020-09-09 19:38:58 +02:00
Mathieu Duponchelle
1d416750d1 line21enc: add support for CDP closed caption meta 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1554>
 2020-09-09 19:12:11 +02:00
Jan Alexander Steffens (heftig)
3f9a7e5c73 hlssink2: Actually release splitmuxsink's pads 
It was looking at the "outer" peer of the ghost pad, not the "inner"
peer (the target).

It provided the wrong pad to gst_element_release_request_pad.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1551>
 2020-09-09 01:06:21 +00:00
Sebastian Dröge
64039cdf84 gst: Update for gst_video_transfer_function_*() function renaming 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1557>
 2020-09-07 12:14:47 +03:00
Nirbheek Chauhan
16d84a2816 webrtc: Clean up the userinfo unescaping code 
Continuation from 04fd705906. This is
easier to understand and also avoids two copies.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1547>
 2020-08-30 09:53:42 +00:00
Jonathan Matthew
2b024ec1b4 modplug: avoid division by zero 
Under some conditions, GetMaxPosition() returns zero, which should cause
position queries to fail rather than crash.
 2020-08-28 08:10:04 +10:00
trilene
04fd705906 webrtc: Unescape turnserver user and password 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1530>
 2020-08-26 23:37:17 +01:00
Tim-Philipp Müller
b48702e4bc sctp: usrsctp: increase DIAG_MSG_LEN to accomodate longer file path 
Fixes "‘%s’ directive output truncated writing XX bytes into
a region of size NN [-Wformat-truncation=]" compiler warnings.

https://github.com/sctplab/usrsctp/pull/521

Fixes #1389

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1540>
 2020-08-26 00:00:24 +01:00
Philippe Normand
2caa3e0230 wpe: skip glbasesrc decide_allocation when non-GL caps are negotiated 
Checking for GL caps features in gl_start() was done too late in case the parent
class fails to setup a working GL context. The element now determines if GL
support should be enabled during the decide-allocation query handling.

Additionally, when no GL context was found, we need to handle the element
cleanup because in that situation glbasesrc won't call gl_stop.

Fixes https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/issues/1376

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1532>
 2020-08-24 20:59:50 +00:00
Matthew Waters
e15a8fcbdd webrtc/datachannel: clear the error after use 
Fixes a memory leak

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1535>
 2020-08-24 17:02:35 +10:00
Matthew Waters
7489addc0a webrtc/datachannel: free previous protocol/label fields 
Fixes a memory leak

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1535>
 2020-08-24 17:02:35 +10:00
Matthew Waters
e5a2e3ac4c sctpdec: unref after retrieving the static pad template 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1535>
 2020-08-24 17:02:35 +10:00
Matthew Waters
9011539940 webrtc/ice: resolve .local candidates internally 
Requires the system's DNS resolver to support mdns resolution.

Fixes interoperablity with recent versions of chrome/firefox that
produce .local address in for local candidates.

Fixes https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/issues/1139
 2020-08-20 13:01:17 +10:00
J. Kim
8e9f8c7f2c srtobject: set error when canceled waiting for a caller 
To propagate error, this commit sets a reason. Otherwise, the function
caller should check if `error` is NULL when the return value is not normal.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1522>
 2020-08-19 12:01:37 +00:00
J. Kim
ebdb3447ce srtobject: fix typo, s/errorj/error 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1522>
 2020-08-19 11:31:41 +00:00
Vivia Nikolaidou
dc58065dfb fdkaacenc: Implement flush function 
The internal fdk encoder always produces 1024 bytes even with no input,
so special care should be taken to not drain it twice.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1515>
 2020-08-17 23:39:39 +03:00
Jan Alexander Steffens (heftig)
76c171509e fdkaacenc: Refactor layout selection code 
No functional change.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1359>
 2020-08-17 10:19:56 +02:00
Jan Alexander Steffens (heftig)
5e68124981 fdkaacenc: Move channel layouts to gstfdkaac.c 
In preparation of sharing them with the decoder. Iteration of the
channel layouts needs to be changed to use a sentinel element.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1359>
 2020-08-17 08:07:00 +00:00
Matthew Waters
2d31aba78d vulkan: docs annotation updates 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1506>
 2020-08-15 02:55:30 +00:00
Philippe Normand
314a8c023f wpe: WebView and WebContext handling fixes 
The WPEThreaded view is now split in 2 classes:
- WPEContextThread handles the persistent WebKit thread, where all WebKit API
calls should be handled.
- WPEView: is created from the WPEContextThread. It handles the WebView and
maintains the public interface on which wpesrc relies. This is the facade for
the WebView, basically. It takes care of dispatching API calls into the context
thread.

With these fixes it is now possible to create (and reuse) mutlple wpesrc
elements during the application lifetime.

Fixes #1372

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1484>
 2020-08-14 09:41:56 +00:00
Sebastian Dröge
7ef393d5ff sctp: fix build with GST_DISABLE_GST_DEBUG 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1465>
 2020-08-14 01:48:33 +01:00
Tim-Philipp Müller
80a0da9698 sctp: hook up internal copy of libusrsctp to build 
Add option 'sctp-internal-usrsctp' so people can choose
to build againts the distro version instead.

Fixes https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/issues/870

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1465>
 2020-08-14 01:33:28 +01:00
Tim-Philipp Müller
f4538e24b6 sctp: import internal copy of usrsctp library 
There are problems with global shared state and no API stability
guarantees, and we can't rely on distros shipping the fixes we
need. Both firefox and Chrome bundle their own copies too.

Imported from https://github.com/sctplab/usrsctp,
commit 547d3b46c64876c0336b9eef297fda58dbe1adaf
Date: Thu Jul 23 21:49:32 2020 +0200

Fixes https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/issues/870

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1465>
 2020-08-14 01:32:45 +01:00
Seungha Yang
91f9490529 cccombiner: Correct sink_query chain up and fix caps leaks 
Don't chain up to src_query() from sink_query() method, and
returned caps by gst_static_pad_template_get_caps() needs to be
cleared.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1513>
 2020-08-13 20:42:51 +09:00
Hosang Lee
d9dda36e02 smoothstreaming: start closer to the edge in live streams 
It is more appropriate to start closer to the live edge in
live streams. Some live streams maintain a large dvr window
(over few hours in some cases), so starting from the first
fragment will be too far away from the live edge.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1346>
 2020-08-10 16:13:30 +09:00
Sebastian Dröge
1d1b3eb8b4 cccombiner: Update for additional info parameter to the "samples-selected" signal 
See https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/590

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1498>
 2020-08-07 17:53:14 +00:00
Mathieu Duponchelle
93a54093ec mpeg2enc: add disable-encode-retries property 
MJPEG Tools may reencode pictures in a second pass to stick
closer to the target bitrate. This can result in slower than
real-time encoding for full HD content in certain situations,
as entire GOPs need reencoding when the reference picture is
reencoded.

See https://sourceforge.net/p/mjpeg/bugs/141/ for background

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1491>
 2020-08-06 17:13:03 +00:00
Mathieu Duponchelle
674ad01016 mpeg2enc: report a latency 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1491>
 2020-08-06 17:13:03 +00:00
Mathieu Duponchelle
2dfceac9fc mpeg2enc: finalize GstVideoEncoder port 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1491>
 2020-08-06 17:13:03 +00:00
Tim-Philipp Müller
c9d10e2277 mpeg2enc: store video encoder instance directly in stream writer class 
Instead of storing the pad and then only using it to get the
element.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1491>
 2020-08-06 17:13:03 +00:00
Tim-Philipp Müller
8a745529c7 mpeg2enc: remove unused streamwriter member 'buf' 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1491>
 2020-08-06 17:13:03 +00:00
Tim-Philipp Müller
0c28c406cc mpeg2enc: remove some unused code 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1491>
 2020-08-06 17:13:03 +00:00
Tim-Philipp Müller
7f6eb54d42 mpeg2enc: remove code paths for older mjpegtools versions 
Gets rid of lots of code paths that no one has built,
used or tested for ages, and makes code more maintainable.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1491>
 2020-08-06 17:13:03 +00:00
Alban Browaeys
79d90b4fd2 mpeg2enc: initial port to GstVideoEncoder base class 
https://bugzilla.gnome.org/show_bug.cgi?id=685414

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1491>
 2020-08-06 17:13:03 +00:00
Sebastian Dröge
e70ec38000 srt: Add support for using hostnames instead of IP addresses 
If an address can't be parsed as IP address, try resolving it via
GResolver instead. SRT URIs more often than not contain hostnames and
without trying to resolve them we won't be able to handle such URIs.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1493>
 2020-08-06 07:29:14 +00:00
Mathieu Duponchelle
1522e40397 cccombiner: update to new samples selection API 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1497>
 2020-08-05 18:16:32 +02:00
Jordan Petridis
cee211123a opencv: compile with -Wno-format-nonliteral 
opencv plugin is pulling a header which makses clang++ 10
complain a lot and blocks -werror.

```
/usr/include/opencv4/opencv2/flann/logger.h:83:36: error: format string is not a string literal [-Werror,-Wformat-nonliteral]
        int ret = vfprintf(stream, fmt, arglist);
                                   ^~~
```

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1494>
 2020-08-05 12:17:06 +00:00
Jordan Petridis
92f9567737 gstlv2utils.c: avoid implicit float to int conversion 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1487>
 2020-08-04 11:37:52 +00:00
Jordan Petridis
5705301ed5 gstladspautils.c: avoid implicit float to int conversion 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1487>
 2020-08-04 11:37:52 +00:00
Francisco Javier Velázquez-García
97b5951d25 srtobject: Add support for IPv6 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1477>
 2020-08-03 21:46:04 +00:00
Francisco Javier Velázquez-García
1ba379ded0 srtobject: Reset parameters before setting URI 
This makes `gst_srt_object_validate_parameters` work properly since
`localaddress` and `localport` will be missing if the URL did not
provide them.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1477>
 2020-08-03 21:46:04 +00:00
Francisco Javier Velázquez-García
096c60f9c5 srtobject: Simplify gst_srt_object_set_*_value 
This fixes `gst_srt_object_set_string_value` in particular because the
value might not be a static string.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1477>
 2020-08-03 21:46:04 +00:00
Francisco Javier Velázquez-García
1a8e2cf981 srtobject: Store passphrase like other parameters 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1477>
 2020-08-03 21:46:04 +00:00
Nirbheek Chauhan
d4ca8820e7 webrtc, rtmp2: Warn if the user or password aren't escaped 
If the user/pass aren't escaped, the userinfo will be ambiguous and we
won't know where to split. We will accidentally get it right if the :
belongs in the password.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1481>
 2020-08-03 18:12:50 +00:00
Nirbheek Chauhan
827afa206d webrtc, rtmp2: Fix parsing of userinfo in URI strings 
While parsing the string, `gst_uri_from_string()` also unescapes the
userinfo. This is bad if your username contains a `:` character, since
we will then split the userinfo at the wrong location when parsing it.

To fix this, we can use the new `gst_uri_from_string_escaped()` API
that was added in
https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/583

Fixes https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/issues/831

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1481>
 2020-08-03 18:12:50 +00:00
Sebastian Dröge
6e412d42c7 hlssink2: Don't assert if we don't have a current location when receiving the fragment-closed message 
This can happen if the application did not provide an output stream for
the fragment and didn't handle the error message before splitmuxsink
decided to consider the fragment closed.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1469>
 2020-08-03 11:23:36 +00:00
Nicola Murino
8544f3928e opencv: allow compilation against 4.4.x 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1482>
 2020-08-01 17:38:42 +00:00
Mathieu Duponchelle
265128e7f7 cccombiner: implement samples selection API 
Call gst_aggregator_selected_samples() after identifying the
caption buffers that will be added as a meta on the next video
buffer.

Implement GstAggregator.peek_next_sample.

Add an example that demonstrates usage of the new API in
combination with the existing buffer-consumed signal.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1390>
 2020-07-30 23:10:33 +00:00
Mathieu Duponchelle
480ede1aa7 wpesrc: timestamp buffers when working with SHM buffers 
GLBaseSrc::fill() will take care of that when dealing with
images, but as we don't chain up when dealing with SHM buffers
this needs to be done in order for GLBaseSrc::get_times() to
work appropriately.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1476>
 2020-07-30 08:09:47 +00:00
Mathieu Duponchelle
ad57b8040f wpe: fix ready signalling 
Receiving the WEBKIT_LOAD_COMMITTED event doesn't actually
mean we have committed an SHM buffer / image yet.

As this is the condition we are interested in, check it
instead.

Also wrap g_cond_wait in a loop for extra correctness points.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1476>
 2020-07-30 08:09:47 +00:00
Damian Hobson-Garcia
deefedd002 waylandsink: Update stale GstBuffer references in wayland buffer cache 
"waylandsink: use GstMemory instead of GstBuffer for cache lookup"
changes the cache key to GstMemory, but the cached data still needs
a pointer to the GstBuffer to control the buffer lifecycle.
If the GstMemory used as the cache key moves from one GstBuffer to
another, the pointer in the cached data will be out-of-date.

Update the current GstBuffer pointer for each frame so that it always
represents the currently in use (from attach to release) GstBuffer
for each wl_buffer.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1473>
 2020-07-28 11:35:53 +00:00
Tim-Philipp Müller
798249dc9f directfb: suppress compiler warning from directfb headers 
On debian sid, directfb 1.7.7

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1467>
 2020-07-25 19:47:43 +01:00
Thibault Saunier
7db147e9aa iqa: Add a 'mode' property 
This property currently only supports a 'strict' that checks that
all the input streams have the exact same number of frames.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1424>
 2020-07-23 17:14:08 +00:00
Thibault Saunier
0349f032bf iqa: Implement child proxy 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1424>
 2020-07-23 17:14:08 +00:00
Matthew Waters
597c1b4ec6 webrtc: remove private properties/signals from the now public ice object 
We don't want to expose all of the webrtcbin internals to the world.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1444>
 2020-07-20 15:56:20 +10:00
Ederson de Souza
51d5aee94d avtp: Update documentation 
- Mention that a new capability is required by "avtpsink" element;
 - Use "clockselect" element to change pipeline clock, instead of a
   gst-launch option that never saw the light of day.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1443>
 2020-07-16 13:32:56 -07:00
Damian Hobson-Garcia
e6944da134 waylandsink: use GstMemory instead of GstBuffer for cache lookup 
The GstMemory objects contained in a GstBuffer could be replaced
by an upstream element, which would break the association beteen
the GstBuffer and the wayland wl_buffer, make the cache lookup
results incorrect.
This patch changes the cache lookup to use the first GstMemory
in a buffer instead.  For multi-plane buffers, this assumes that
all of the GstMemory(s) will always be moved together as a set,
and that the same (first) GstMemory isn't used with different
combinations of other GstMemory(s).

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1401>
 2020-07-15 14:35:06 +00:00
Damian Hobson-Garcia
ff5f264045 waylandsink: Keep per display wayland buffer caches 
Instead of attaching a single wayland wl_buffer to each GStBuffer as qdata,
keep a separate cache for each display.
A unique wl_buffer and associated metadata is created for each display.
This allows for sharing of GstBuffer objects between multiple
displays, such as when using tee elements.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1401>
 2020-07-15 14:35:06 +00:00
Tim-Philipp Müller
395ecb3d2f avtp: rename tstamp-mode to timestamp-mode 
I thnk w cn spre the xtra lttrs.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1397>
 2020-07-11 00:14:44 +01:00
Tim-Philipp Müller
92456967d0 opencv: suppress warnings about non-existent include dirs 
Looks like opencv4 ships with a broken .pc file.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1427>
 2020-07-10 14:57:44 +01:00
Vivia Nikolaidou
bf38898af8 cccombiner: Update segment according to video sink pad 
Otherwise the following pipeline would preroll after 1000 hours:
gst-launch-1.0 videotestsrc ! x264enc ! cccombiner ! fakesink silent=0 sync=1 -v

Fixes #1355

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1419>
 2020-07-08 17:11:38 +00:00
Tim-Philipp Müller
f3fdd76683 rtmp, transcodebin: fix i18n header includes 
Fixes #1351

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1416>
 2020-07-07 19:55:00 +01:00
Matthew Waters
ebd1b2c929 ccconverter: write the cdp timecode data correctly 
We were mixing up the tens part with the unit parts all over the place.

e.g. 12 seconds would be encoded as 0x21 instead of the correct 0x12

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1400>
 2020-07-03 06:54:46 +00:00
Matthew Waters
327a79e982 ccconverter: output warning log if parsing a cdp packet fails 
Simplifies figuring out why there may be no output from ccconverter with
a cdp input.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1400>
 2020-07-03 06:54:46 +00:00
Matthew Waters
6fa4a8c3c3 ccconverter: fix cdp timecode parsing 
The first reserved bits are in the most significant bit.

i.e. 0xc0 vs 0x0c

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1400>
 2020-07-03 06:54:46 +00:00
Ederson de Souza
f8bf84307f avtp: Use g_strerror instead of strerror 
It should avoid some implicit declaration errors (and be utf-8 friendly).

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1404>
 2020-07-03 04:04:39 +00:00
Mathieu Duponchelle
b33f10e7e2 docs: remove gst prefix from plugin titles 2020-07-02 18:10:21 +02:00
Philippe Normand
db0ab58e14 wpe: Set documentation caps 
As the caps template can vary depending on the WPEBackend-FDO version
found at build time, set a fixed template for the generate documentation.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1392>
 2020-07-01 20:31:42 +00:00
Matthew Waters
c6c4d42c4a ccconverter: fail negotiation when framerate conversion is not possible 
Converting between anything but cdp will fail at converting
framerates and negotiation should reflect that.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1393>
 2020-07-01 19:33:56 +00:00
Matthew Waters
4f334234c8 ccconverter: fix missing output framerate on the caps 
A pipeline like this:

closedcaption/x-cea-708,format=cdp,framerate=30000/1001 ! ccconverter ! closedcaption/x-cea-708,format=cc_data

would produce a critical/assert:

GStreamer-CRITICAL **: 14:21:11.509: gst_util_fraction_multiply: assertion 'a_d != 0' failed

because there would be no framerate field on ccconverter's output.

Fixed by always fixating a framerate if the input has a framerate.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1393>
 2020-07-01 19:33:56 +00:00
Tim-Philipp Müller
c229127b43 avtp: documentation fixes 
Unclear why hotdoc wants 'gstavtp' as the plugin name here,
that's just wrong.

Add since marker and mark private subclasses as plugin API
so hotdoc knows they belong to the plugin and aren't external.

Fix GstAvtpAafTstampMode get_type() function.
 2020-07-01 18:41:25 +01:00
Olivier Crête
cceca1ffe8 webrtcbin: Expose "latency" property 
This property sets the latency both on the rtpbin/rtpjittbuffer, but
also on the RTPStorage elements currently used by the FEC decoder.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1367>
 2020-06-29 22:45:31 -04:00
Michael Olbrich
76411205c8 wlvideoformat: fix typo in the format list 
DRM_FORMAT_ARGB8888 was actually used twice in the list for different SHM /
Gstreamer formats. In this case DRM_FORMAT_ABGR8888 is the correct format.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1382>
 2020-06-28 18:55:02 +02:00
Sebastian Dröge
3864c9f97f gstdtlsconnection: Propagate errors from key export to the caller 
Otherwise the DTLS connection silently does nothing instead of reporting
an error via the elements.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1156>
 2020-06-26 10:20:04 +03:00
Miguel Paris
3dd2bbf23c dtlsconnection: do not set keys_exported flag if actually not exported 
keys_exported flag should be set only if keys are actually exported.
For that the next conditions are needed:
  1 - SSL_export_keying_material on success
  2 - SSL_get_selected_srtp_profile returns a valid profile
  3 - The profile ID is SRTP_AES128_CM_SHA1_80 or SRTP_AES128_CM_SHA1_32

Also don't crash if NULL is returned as profile.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1156>
 2020-06-26 10:19:28 +03:00
Sebastian Dröge
ed3417219a ccextractor: Push a GAP event if we have a caption pad but a video buffer did not contain any captions 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1371>
 2020-06-25 08:18:37 +00:00
Sebastian Dröge
f694956511 ccextractor: Add property to remove caption meta from the outgoing video buffers 
This is disabled by default to keep backwards compatibility.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1371>
 2020-06-25 08:18:37 +00:00
Mathieu Duponchelle
ad49ae42f7 docs: mark more types as plugin API 2020-06-23 12:10:19 -04:00
Mathieu Duponchelle
6baffc2931 docs: mark more types as plugin API 2020-06-23 12:10:17 -04:00
Sebastian Dröge
aa01e6ba22 webrtcbin: Don't call gst_ghost_pad_construct() anymore 
It's deprecated, unneeded and doesn't do anything anymore.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1360>
 2020-06-22 17:01:34 +00:00
Matthew Waters
0f41c0f000 webrtc: fix ice control mode when we offer initially 
An initial offer means we have a local description not a remote
description.

Fixes https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/issues/1332

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1358>
 2020-06-22 12:17:09 +00:00
Vivia Nikolaidou
41950f2aba fdkaacenc: Add missing SURROUND mappings 
SURROUND is more to spec according to the FIXME comments, so add this.

Also add SIDE for 5 and 5.1 because of ffmpeg compatibility, because the
following pipeline downmixes to mono otherwise:

gst-launch-1.0 audiotestsrc num-buffers=1 ! audio/x-raw, channels=6 !
avenc_ac3 ! avdec_ac3 ! audioconvert ! fdkaacenc ! fakesink -v

Fixes #1327

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1352>
 2020-06-22 07:14:20 +00:00
Tim-Philipp Müller
a8ce8db982 srt: add "empty" subclasses for deprecated srt{client,server}{src,sink} 
The doc system gets confused when we register the exact same
class as multiple elements, so make a subclass for each.

Also wrap registration of deprecated elements with #ifndef GST_REMOVE_DEPRECATED.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1354>
 2020-06-19 17:20:02 +01:00
Tim-Philipp Müller
8e93ae65e8 x265: ignore tune property when diffing generated docs 
Unfortunately it means those tune enums don't show up in
the docs then, but if that's how it's gotta be..

(Problem at hand is that on Tim's machine x265enc gets an
tune=animation and on the CI machine this doesn't show up.)

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1354>
 2020-06-19 15:41:51 +01:00
Tim-Philipp Müller
29026b1c27 Mark more plugin GTypes as plugin API 
To appease the CI gods.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1354>
 2020-06-19 13:05:38 +01:00
Hosang Lee
e04be18c49 mssdemux: ignore unrecognized stream 
Only create pads for steams with caps that can be recognized
from the fourcc.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1348>
 2020-06-17 06:48:18 +00:00
Matthew Waters
3cf0abddbb vulkan/shaders: add explicit license headers 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1338>
 2020-06-12 15:52:44 +10:00
Matthew Waters
1f44cb0519 vulkan/shaders: manually indent bin2array 
Looks much nicer with some semblance of code formatting

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1338>
 2020-06-12 15:52:44 +10:00
Haihua Hu
d77de13ca6 waylandsink: add wl_registry.global_remove listener 
when hotplug display, wayland client will call this listener
to notify client do clean up. Temporarily set a dummy function
here to avoid app abort

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1327>
 2020-06-09 19:54:24 +00:00
Thibault Saunier
ebababae03 srt: doc: Add missing gst_type_mark_as_plugin_api 2020-06-09 12:28:13 -04:00
Thibault Saunier
af26b95c29 docs: Mark lv2 runtime generated enums as plugins API types 2020-06-09 12:28:13 -04:00
Thibault Saunier
60fba5f380 docs: Add some more plugin API types 
And allow creating vulkan device object without specifying an instance
so it can be introspected.
 2020-06-09 12:28:13 -04:00
Thibault Saunier
3a98a37375 docs: Update plugins cache 2020-06-09 12:28:13 -04:00
Mathieu Duponchelle
a048ce81d4 plugins: uddate gst_type_mark_as_plugin_api() calls 2020-06-06 00:40:42 +02:00
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