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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
Marc Leeman
0be59181d7 rtpmanagerbad: remove duplicate parent declaration 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1689>
 2020-10-12 13:56:50 +02:00
Tim-Philipp Müller
1ed969d276 rtmp2sink: fix since marker on new "stop-commands" property 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1687>
 2020-10-12 11:55:46 +01:00
Víctor Manuel Jáquez Leal
18444fd993 va: basedec: copy frames logic to decide_allocation() 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1673>
 2020-10-10 17:18:04 +00:00
Víctor Manuel Jáquez Leal
5658c4182a va: basedec: refactor the other video pool instantiation 
Just a code clean up

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1673>
 2020-10-10 17:18:04 +00:00
Víctor Manuel Jáquez Leal
a6398eca17 va: basedec: add gstvabasedec helper 
This is a helper for all decoders.

It is not an abstract subclass, just merely a helper that avoids code
duplication among the decoders.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1673>
 2020-10-10 17:18:04 +00:00
Víctor Manuel Jáquez Leal
a5dcb35ad6 va: vp8dec: add element documentation 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1673>
 2020-10-10 17:18:04 +00:00
Víctor Manuel Jáquez Leal
a0ba72ed50 va: h264dec: set video alignment definition earlier 
This patch renames need_cropping variable to need_videoalign which is clearer
with its function. And now GstVideoAlignment is part of GstVaH264Dec structure,
so it can be set earlier.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1673>
 2020-10-10 17:18:04 +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)
3ea6387f96 tests: svthevcenc: Fix test_encode_simple 
Pick the same I420 format the other test use. Without this, the source
picks AYUV64, which fails.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1573>
 2020-10-10 04:34:56 +00:00
Víctor Manuel Jáquez Leal
fbbf9c629a va: allocator: add _set_format() and _get_format() 
Since allocators keep an available memory queue to reuse, video format and usage
hint are now persistant while allocator's memories are around.

This patch adds _set_format() and _get_format() for both VA allocators.

_set_format() validates if given format can be used or reused. If no allocated
surface previously it creates a dummy one to fetch its offsets and
strides. Updated info is returned to callee.

GstVaPool uses _set_format() at config to verify the allocator capacity and to
get the surfaces offsets and strides, which are going to be used by the video
meta.

Allocator extracted caps are compared with caps from config and if they have
different strides or offsets, force_videometa is set.

A new bufferpool method gst_va_pool_requires_video_meta() is added return the
value of force_videometa. This value is checked in order to know if decoders
need to copy the surface if downstream doesn't announce video meta support.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1667>
 2020-10-09 16:34:57 +02:00
Víctor Manuel Jáquez Leal
482e93b4d8 va: caps: added gst_caps_is_raw() 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1667>
 2020-10-09 16:34:57 +02:00
Víctor Manuel Jáquez Leal
6a8fa67f42 va: pool: call parent's start() method 
Without preallocating buffers and memories a deadlock in pool allocator is
highly probably since it might hit the case were buffer is returned to the pool
but their memories are still hold by a copy downstream, without other
preallocated buffers available.

This kind of a hack, where buffer_reset() follow the normal path if it's called
from start().

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1667>
 2020-10-09 16:34:57 +02:00
Víctor Manuel Jáquez Leal
8f936baffe va: pool: fix log's object 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1667>
 2020-10-09 16:34:57 +02:00
Víctor Manuel Jáquez Leal
26ee5b1e4e va: allocator: remove noisy log message 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1667>
 2020-10-09 16:34:57 +02:00
Víctor Manuel Jáquez Leal
9254045b3e va: allocator: add a todo for dmabuf_memories_setup() 
It would be nice to add a surface pool for this type of surface allocation in
order to have a better control of them.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1667>
 2020-10-09 16:34:56 +02:00
Víctor Manuel Jáquez Leal
a299dff9d4 va: allocator: add a surface counter 
Every time a new surface is created the counter increases by one, and when it is
destroyed (or will be destroyed in case of GstVaAllocator), the counter is
decreased. Then, at allocator dispose, it is warning if the counter is not zero.

This counter will be also used to check if the allocator can change its
configuration if the counter is zero or can not.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1667>
 2020-10-09 16:34:56 +02:00
Víctor Manuel Jáquez Leal
30281af83f va: allocator: remove GstVideoInfo from GstVaBufferSurface 
Don't store it them anymore since it is related with the negotiated stream and
not the concrete buffer.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1667>
 2020-10-09 16:34:56 +02:00
Víctor Manuel Jáquez Leal
67eb0a9440 va: remove GstVideoInfo parameter from _get_surface() functions 
There shouldn't be need to retrieve GstVideoInfo per buffer or memory since it
is the same for all the negotiated stream.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1667>
 2020-10-09 16:34:56 +02:00
Víctor Manuel Jáquez Leal
895fe44154 va: vpp: don't fetch video info from buffer 
Instead of fetching video info from the buffer, use the already set ones.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1667>
 2020-10-09 16:34:56 +02:00
Víctor Manuel Jáquez Leal
64eb0f0ed2 va: dec, vpp: don't get buffer size from allocators 
Since buffer size is now ignored by bufferpool there's no need to get tha value
from the allocator.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1667>
 2020-10-09 16:34:56 +02:00
Víctor Manuel Jáquez Leal
e73b866fe0 va: pool: ignore size in config 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1667>
 2020-10-09 16:34:56 +02:00
Guillaume Desmottes
75dc98cc08 h265parse: set interlace-mode=interleaved on interlaced content 
interlace-mode=alternate is a special case of interlace-mode=interleaved
where the fields are split using two different buffers.

We should use the latter instead of the former to no break compat with
elements supporting only 'interleaved'.
Decoders producing alternate, such as OMX on the Zynq, should change the
interlace-mode on their output caps.

Fix https://gitlab.freedesktop.org/gstreamer/gst-plugins-base/-/issues/825

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1655>
 2020-10-09 10:19:52 +00:00
Jacek Tomaszewski
ca4a0273df Replace LGPL v2 with LGPL v2.1 in COPYING and remove COPYING.LIB 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1631>
 2020-10-09 08:48:17 +00:00
Jacek Tomaszewski
9cac8bb449 Replace GPL v2 with LGPL v2 in COPYING file 
Fixes #1422
https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/issues/1422

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1631>
 2020-10-09 08:48:17 +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
Sebastian Dröge
d3d73f61fa webrtc: Require gstreamer-sdp in the pkg-config file 
Some headers include API from it.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1660>
 2020-10-08 21:55:18 +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
Seungha Yang
20d9283e3d mfvideosrc: Use only the first video stream per device 
Non-first video stream might not be working with current
implementation. It could be non-video (e.g., photo source) and then
ReadSample() might be blocked forever.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1661>
 2020-10-08 20:43:58 +00:00
Seungha Yang
9279326d8a decklink: Update doc 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1665>
 2020-10-08 20:05:03 +00:00
Seungha Yang
b86e77e3a3 decklink: Update Windows headers with SDK 11.2 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1665>
 2020-10-08 20:05:03 +00:00
Seungha Yang
94a9a8f836 decklink: Update OSX headers with SDK 11.2 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1665>
 2020-10-08 20:05:03 +00:00
Tim
c6151f635f decklink: Updated DeckLink SDK to 11.2 to support DeckLink 8K Pro 
Updated Decklink SDK to version 11.2 in order to support newer cards like the Decklink 8K Pro.
This required to replace the duplex property by a profile property.

Profile values can be the following:
-  bmdProfileOneSubDeviceFullDuplex
-  bmdProfileOneSubDeviceHalfDuplex
-  bmdProfileTwoSubDevicesFullDuplex
-  bmdProfileTwoSubDevicesHalfDuplex
-  bmdProfileFourSubDevicesHalfDuplex

Fixes #987

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1665>
 2020-10-08 20:05:03 +00:00
Seungha Yang
9ecdfea7da mfvideosrc: Fix invalid memory access when outputting jpeg 
Don't access unknown-dangerous-nonsense address

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1662>
 2020-10-08 16:28:16 +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
a40d6f4994 Revert "rtpsender: Add API to set the priority" 
This reverts commit a8b287c764.

It breaks the CI until the C# bindings are fixed.
 2020-10-08 18:53:10 +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
Sebastian Dröge
f12265d9c5 Revert "webrtc: Document more objects" 
This reverts commit ad68c6b1eb.

It breaks the CI until the C# bindings are fixed.
 2020-10-08 18:52:50 +03:00
Sebastian Dröge
74a42c5ba8 Revert "webrtc: Add hotdoc style since tags" 
This reverts commit 63a5fa818c.

It breaks the CI until the C# bindings are fixed.
 2020-10-08 18:51:37 +03:00
Olivier Crête
63a5fa818c webrtc: Add hotdoc style since tags 
We're stuck having to add a separate comment for now.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1425>
 2020-10-06 16:52:48 -04:00
Olivier Crête
ad68c6b1eb webrtc: Document more objects 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1425>
 2020-10-06 16:49:08 -04:00
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