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Author SHA1 Message Date
Thibault Saunier
af054f7015 qroverlay: Import from gst-qroverlay 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1730>
 2020-11-11 00:18:32 +00:00
Olivier Crête
da2bd55177 webrtc: Add properties to change the socket buffer sizes to ice object 
libnice doesn't touch the kernel buffer sizes. When dealing with RTP data,
it's generally advisable to increase them to avoid dropping packets locally.
This is especially important when running multiple higher bitrate streams at
the same time.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1366>
 2020-11-03 22:07:53 +00:00
Jan Schmidt
5f9897745b vkdeviceprovider: Avoid deadlock on physical device 
Don't hold the object lock on the vk physical device while
constructing a GstVulkanDevice around it, as
GstVulkanDevice can make calls on the physical device that
require the object lock.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1754>
 2020-11-03 04:28:00 +00:00
Randy Li
4e4c26af4b wlvideoformat: fix DMA format convertor 
In the most of case, this typo would work. But for
ARGB8888 and XRGB8888, which shm format is not based on fourcc,
which would never appear in format enumeration.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1751>
 2020-11-02 12:39:38 +00:00
Jan Schmidt
dbeb576531 sctp: Do downward state change logic after chaining up. 
Call the parent state_change function first when changing state
downward, to make sure that the element has stopped before cleaning
it up.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1741>
 2020-10-31 01:47:06 +00:00
Jan Schmidt
760592a29c dtls: Avoid bio_buffer assertion on shutdown. 
On shutdown, a previous iteration of dtsl_connection_process()
might be incomplete and leave a partial bio_buffer behind.

If the DTLS connection is already marked closed, drop out
of dtls_connection_process early without asserting.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1741>
 2020-10-31 01:47:06 +00:00
Jan Schmidt
af90778314 webrtc: Fix a race on shutdown. 
The main context can disappear in gst_webrtc_bin_enqueue_task()
between checking the is_closed flag and enqueueing a source on the
main context. Protect the main context with the object lock instead
of the PC lock, and hold a ref briefly to make sure it stays alive.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1741>
 2020-10-31 01:47:06 +00:00
Olivier Crête
80a56c25a6 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/1707>
 2020-10-30 16:24:40 -04:00
Olivier Crête
0fbbdc5734 rtptransceiver: Store the SSRC of the current stream 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1707>
 2020-10-30 16:23:10 -04:00
Olivier Crête
7be09a5f22 webrtc: Save the media kind in the transceiver 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1707>
 2020-10-30 16:23:10 -04:00
Olivier Crête
e172ca5be1 webrtcbin: Remove unused function 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1707>
 2020-10-30 16:23:10 -04:00
Chris Bass
d25be0d16e ttmlparse: Handle whitespace before XML declaration 
When ttmlparse is in, e.g., an MPEG-DASH pipeline, there may be
whitespace between successive TTML documents in ttmlparse's accumulated
input. As libxml2 will fail to parse documents that have whitespace
before the opening XML declaration, ensure that any preceding whitespace
is not passed to libxml2.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1539>
 2020-10-30 07:01:52 +00:00
Chris Bass
8df2314c23 ttmlparse: Ensure only single TTML doc parsed 
The parser handles only one TTML file at a time, therefore if there are
multiple TTML documets in the input, parse only the first.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1539>
 2020-10-30 07:01:52 +00:00
Guillaume Desmottes
bfb9071081 isac: add iSAC plugin 
Wrapper on the iSAC reference encoder and decoder from webrtc,
see https://en.wikipedia.org/wiki/Internet_Speech_Audio_Codec

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1124>
 2020-10-29 16:59:18 +01:00
Randy Li
6d8133e41e waylandsink: release frame callback when destroyed 
We would use a frame callback from the surface to indicate
that last buffer is rendered, but when we destroy the surface
and that callback is not back yet, it may cause the wayland event
queue crash.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1729>
 2020-10-29 12:09:01 +00:00
Nicola Murino
77f28ee3e7 opencv: allow compilation against 4.5.x 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1709>
 2020-10-27 10:53:27 +00:00
Philippe Normand
37b7809d18 wpe: Convert launch lines to markdown and move since tag 
Seems like the examples don't appear in the generated docs because the Since tag
was badly positioned in the doc blurb.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1706>
 2020-10-18 15:17:25 +00:00
Tim-Philipp Müller
3f8d33abed hlssink2: fix and flesh out docs 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1699>
 2020-10-16 13:37:18 +00:00
Stéphane Cerveau
cbd61e28b2 meson: update glib minimum version to 2.56 
In order to support the symbol g_enum_to_string in various
project using GStreamer ( gst-validate etc.), the glib minimum
version should be 2.56.0.

Remove compat code as glib requirement
is now > 2.56

Version used by Ubuntu 18.04 LTS

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1695>
 2020-10-16 09:16:34 +00:00
Vivia Nikolaidou
94e1623434 cameracalibrate: Improve gst-inspect documentation 
Thanks to @kazz_naka on Twitter

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1691>
 2020-10-13 17:21:59 +03:00
Matthew Waters
2f29a4cde6 wpesrc: add some debug logging around WPEView creation/destruction 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1663>
 2020-10-13 08:48:05 +00:00
Matthew Waters
da18a8d93d wpesrc: fix a memory leak of the bytes 
free the previous GBytes if load-bytes is called multiple times
before view creation.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1663>
 2020-10-13 08:48:05 +00:00
Matthew Waters
356fee4dd6 wpesrc: only create webview if not already created 
e.g. _decide_allocation() can be called multiple times throughout the
element's lifetime and we only want to create the view once rather than
overwriting.

Fixes a leak of the WPEView under certain circumstances.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1663>
 2020-10-13 08:48:05 +00:00
Matthew Waters
b84c8821de wpe: free a previous pending image/shm buffer 
Don't blindly overwrite a possibly previously set buffer.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1663>
 2020-10-13 08:48:05 +00:00
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