Commit graph

27488 commits

Author SHA1 Message Date
Seungha Yang
c3b26de1f2 av1decoder: Store display resolution for duplicated picture
Target display resolution might be required by subclass implementation

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/2365>
2021-06-29 07:12:27 +00:00
Seungha Yang
7a1effc499 av1decoder: Fix debug typo
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/2365>
2021-06-29 07:12:27 +00:00
Seungha Yang
6dbb9c705a av1parser: Fix tile size calculation
Remaining size should exclude already read "tile size bits".
And see also "5.11.1. General tile group OBU syntax"

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/2365>
2021-06-29 07:12:27 +00:00
Olivier Crête
e548916d85 webrtc receivebin: Drop serialized queries before receive queue
If they're not dropped, they can be blocked in the queue even if it is
leaky in the case where there is a buffer being pushed downstream. Since
in webrtc, it's unlikely that there will be a special allocator to
receive RTP packets, there is almost no downside to just ignoring the
queries.

Also drop queries if they get caught in the pad probe after the queue.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/2363>
2021-06-29 00:42:20 -04:00
Olivier Crête
543fcb93a4 webrtc receivebin: Only set queue to leaky when the pad is blocked
When the pad is no longer blocked, remove the leakyness to make sure
everything gets into the jitterbuffer.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/2363>
2021-06-29 00:42:20 -04:00
Olivier Crête
a07e52528c webrtc receivebin: Don't unblock pad until sender is unblocked
As ther OpenSSL session is created when the receiver goes into
playing, we have to wait for the ICE session to be connected before we
can start delivering packets to the DTLS element.

Fixes #1599

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/2363>
2021-06-29 00:42:20 -04:00
Sebastian Dröge
0e559fc2f3 webrtcbin: Sync to the clock per stream and not per bundle
By using the clocksync inside the dtlssrtpenc, all streams inside a
bundled are synchronized together. This will cause problems if their
buffers are not already arriving synchronized: clocksync would wait for
a buffer on one stream and then buffers from the other stream(s) with
lower timestamps would all be sent out too late.

Placing the clocksync before the rtpbin and rtpfunnel synchronizes each
stream individually and they will be send out more smoothly as a result.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/2355>
2021-06-28 16:38:33 +00:00
Olivier Crête
ee0124cb36 webrtc: Remove the webrtc-priv.h header from public headers
And this time for real, also import it in a couple more places
inside the webrtc element to make it build.

Fixes #1607

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/2359>
2021-06-28 16:06:59 +00:00
He Junyan
a2d5223473 va: change AV1 GstVideoAlignment setting to left-top corner.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/2298>
2021-06-28 15:16:39 +00:00
He Junyan
abf6c51e83 va: h264dec: Set the GstVideoAlignment correctly.
We should set GstVideoAlignment based on the sequence's crop information.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/2298>
2021-06-28 15:16:39 +00:00
He Junyan
027726d6c8 va: h265dec: Set the GstVideoAlignment correctly.
We should set GstVideoAlignment based on the conformance window info.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/2298>
2021-06-28 15:16:39 +00:00
He Junyan
49cd009778 va: pool: Add VideoCropMeta to the buffer if crop_top/left > 0.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/2298>
2021-06-28 15:16:39 +00:00
He Junyan
85c56c1f07 va: basedec: Copy the frames into other_pool if needed.
If decoder's crop_top/left value > 0 and the downstream does not
support the VideoCropMeta, we need to manually copy the frames
into the other_pool and output it.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/2298>
2021-06-28 15:16:39 +00:00
He Junyan
55302c9705 va: basedec: Setup the other_pool to copy output if crop_left/top.
If the decoder has crop_top/left value > 0(e.g. the conformance
window in the H265). Which means that the real output picture
locates in the middle of the decoded buffer. If the downstream can
support VideoCropMeta, a VideoCropMeta is added to notify the
real picture's coordinate and size. But if not, we need to copy
it manually and the other_pool is needed. We always assume that
decoded picture starts from top-left corner, and so there is no
need to do this if crop_bottom/right value > 0.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/2298>
2021-06-28 15:16:39 +00:00
He Junyan
c03350e234 va: No need to set the alignment for VideoMeta
The base va decoder's video_align is just used for calculation the
real decoded buffer's width and height. It does not have meaning
for the VideoMeta, because it does not align to the real picture
in the output buffer. We will use VideoCropMeta to replace it later.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/2298>
2021-06-28 15:16:39 +00:00
He Junyan
98cf9ce6f5 va: Delete the useless align expand in va_pool_set_config().
The base va decoder's video_align is just used for calculation the
real decoded buffer's width and height. While the gst_video_info_align
just calculate the offset and stride based on the video_align. But
all the offsets and strides are overwritten in gst_va_dmabuf_allocator_try
or gst_va_allocator_try, which make that calculation useless.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/2298>
2021-06-28 15:16:39 +00:00
Sebastian Dröge
03d3e0fe73 webrtc: Re-add WebRTC object docs to the public headers
So they end up in the generated documentation and the Since markers
appear in the .gir files too.

Also remove wrong "Since: 1.16" markers for some objects that were
available since 1.14.0 already.

Fixes https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/issues/1609

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/2366>
2021-06-28 14:45:37 +00:00
Sebastian Dröge
096a7f1ac0 webrtcbin: Set transceiver kind and codec preferences immediately when creating it
Otherwise the on-new-transceiver signal will always be emitted with kind
set to UNKNOWN and no codec preferences although both are often known at
this point already.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/2360>
2021-06-25 14:35:43 +03:00
Sebastian Dröge
dcc49f846b webrtcbin: Add a test for setting codec preferences as part of "on-new-transceiver" when setting the remote offer
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/2310>
2021-06-25 09:45:24 +00:00
Sebastian Dröge
348d4229e7 webrtc: Use fail_unless_equals_string() for string assertions
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/2310>
2021-06-25 09:45:24 +00:00
Sebastian Dröge
7ee8f4539e webrtcbin: Store newly created transceivers when creating an answer also in the seen transceivers list
Otherwise it might be used a second time for another media afterwards.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/2310>
2021-06-25 09:45:24 +00:00
Sebastian Dröge
4efdb40f43 webrtcbin: When creating a new transceiver as part of creating the answer also take its codec preferences into account
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/2310>
2021-06-25 09:45:24 +00:00
Sebastian Dröge
b7951fb897 webrtcbin: Fix a couple of caps leaks of the offer caps
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/2310>
2021-06-25 09:45:24 +00:00
Philippe Normand
0f492a39c9 webrtcbin: Stop transceivers update after first SDP error on data channel
When invalid SDP is supplied, _update_data_channel_from_sdp_media() sets the
GError, so it is invalid to continue any further SDP processing, we have to exit
early when the first error is raised.

This change is similar to the one applied in
064428cb34.
See also #1595

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/2356>
2021-06-25 05:12:37 +00:00
Olivier Crête
71052f0321 webrtcbin test: Fix race in new test
Pull a buffer from a sink to make sure that the caps are already
set before trying to update them.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/2348>
2021-06-24 09:41:09 +00:00
Mengkejiergeli Ba
85c17f6c23 msdk: fix qp range for vp9enc
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/2349>
2021-06-24 01:30:18 +00:00
Sebastian Dröge
1d4ecd0bde avwait: Don't consider it a segment change if the segment is the same except for the position
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/2319>
2021-06-23 16:03:38 +00:00
Seungha Yang
7c94b9c4b0 d3d11: Add support for GRAY and more YUV formats
By this commit, following formats will be newly supported by d3d11 elements

* Y444_{8, 12, 16}LE formats:
  Similar to other planar formats. Such Y444 variants are not supported
  by Direct3D11 natively, but we can simply map each plane by
  using R8 and/or R16 texture.
* P012_LE:
  It is not different from P016_LE, but defining P012 and P016 separately
  for more explicit signalling. Note that DXVA uses P016 texture
  for 12bits encoded bitstreams.
* GRAY:
  This format is required for some codecs (e.g., AV1) if monochrome
  is supported
* 4:2:0 planar 12bits (I420_12LE) and 4:2:2 planar 8, 10, 12bits
  formats (Y42B, I422_10LE, and I422_12LE)

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/2346>
2021-06-23 15:35:36 +00:00
Sebastian Dröge
52a0c36598 tsmux: When selecting random PIDs, name the pads according to those PIDs
Some elements will make use of the automatically generated names to
create new pads in future muxer instances, for example splitmuxsink.

Previously we would've created a pad with a random pid that would become
"sink_0", and then on a new muxer instance a pad "sink_0" and tsmux
would've then failed because 0 is not a valid PID.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/2318>
2021-06-23 14:59:43 +00:00
Seungha Yang
ba26a5aea8 mfvideoenc: Enhance B-frame timestamp handling
When B-frame is enabled, encoder seems to adjust PTS of encoded sample
by using frame duration.

For instance, one observed timestamp pattern by using B-frame enabled
and 30fps stream is:
* Frame-1: MF pts 0:00.033333300 MF dts 0:00.000000000
* Frame-2: MF pts 0:00.133333300 MF dts 0:00.033333300
* Frame-3: MF pts 0:00.066666600 MF dts 0:00.066666600
* Frame-4: MF pts 0:00.099999900 MF dts 0:00.100000000

We can notice that the amount of PTS shift is frame duration and
Frame-4 exhibits PTS < DTS.

To compensate shifted timestamp, we should
calculate the timestamp offset and re-calculate DTS correspondingly.
Otherwise, total timeline of output stream will be shifted, and that
can cause time sync issue.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/2354>
2021-06-23 14:12:22 +00:00
Sebastian Dröge
127ade39cf tsmux: Recheck existing pad PIDs when requesting a new pad with a random pid
Previously pads might have been requested already (e.g. in NULL state),
then reset was called (e.g. because changing state) and then a new pad
was requested. Resetting is re-creating the internal muxer object and as
such resetting the pid counter, so the next requested pad would get the
same pid as the first requested pad which then leads to collisions.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/2317>
2021-06-23 05:44:48 +00:00
Seungha Yang
569910a5ac mfh264enc, mfh265enc: Set profile string to src caps
Set configured profile to src caps so that downstream can figure
out selected profile.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/2347>
2021-06-22 15:20:05 +00:00
Olivier Crête
a931e31141 webrtc lib: Make the datachannel struct private
This will prevent any unsafe access.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/2241>
2021-06-21 20:53:09 +00:00
Olivier Crête
dd2da6f2b4 webrtc lib: Make the DTLSTransport struct private
This will prevent any unsafe access.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/2241>
2021-06-21 20:53:09 +00:00
Olivier Crête
a0813c5bd2 webrtc lib: Make the icetransport struct private
This will prevent any unsafe access.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/2241>
2021-06-21 20:53:09 +00:00
Olivier Crête
5233c349e7 webrtc lib: Make the rtpreceiver struct private
This will prevent any unsafe access.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/2241>
2021-06-21 20:53:09 +00:00
Olivier Crête
a6593753a5 webrtc lib: Make the rtpsender struct private
This will prevent any unsafe access.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/2241>
2021-06-21 20:53:09 +00:00
Olivier Crête
b5f2de3124 webrtc lib: Make the transceiver struct private
This will prevent any unsafe access.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/2241>
2021-06-21 20:53:09 +00:00
Mathieu Duponchelle
08323f382c x265enc: add negative DTS support
Use the same set_min_pts approach as x264enc.

Fixes: https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/issues/304
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/2340>
2021-06-21 10:31:21 +00:00
Seungha Yang
ba83f29cec decklinkaudiosrc: Don't assume that stream time is always valid
As per SDK doc, IDeckLinkInputCallback::VideoInputFrameArrived
method might not provide video frame and it can be null.
In that case, given stream_time can be invalid.
So, we should not try to convert GST_CLOCK_TIME_NONE
by using gst_clock_adjust_with_calibration()

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/2337>
2021-06-21 09:58:46 +00:00
Thibault Saunier
d7c716d562 transcoder: Fix usage of g_error_propagate
In the error callback we were propagating an error we were not owning
which is incorrect use of the API.

Also we were clearing a GError we already propagated which is wrong
as propagating gives ownership away.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/2325>
2021-06-21 09:18:32 +00:00
Thibault Saunier
d85368eada transcoder: Add a missing object unlocking
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/2325>
2021-06-21 09:18:32 +00:00
Stéphane Cerveau
f30e74bb20 faad: fix typo in element documentation
seealso is now see_also

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/2323>
2021-06-21 07:35:26 +00:00
Víctor Manuel Jáquez Leal
046e92c503 tests: msdkh264dec: Run test only if factory is available.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/2336>
2021-06-21 06:30:07 +00:00
Víctor Manuel Jáquez Leal
78f4777a82 msdk: Demote error log message to warning.
It is not an error that the available hardware doesn't support VA-API/MSDK. Just
none plugin features will be registered.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/2336>
2021-06-21 06:30:07 +00:00
Seungha Yang
2c14a7253b libs: d3d11: Port to C++
In general, C++ COM APIs are slightly less verbose and more readable
than C APIs. And C++ supports some helper methods
(smart pointer and C++ only macros for example) which are not allowed for C.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/2343>
2021-06-20 20:10:24 +09:00
U. Artie Eoff
300f25ae36 msdk: declare external dependencies
Track kernel and VA driver dependencies so gstreamer
will re-inspect the plugin if any of them change.

Also, do not blacklist the plugin if !msdk_is_available
since it could be a transient issue caused by one or
more external dependency issues (e.g. wrong/missing
driver specified, but corrected by user later on).

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/2335>
2021-06-17 02:56:45 +00:00
Seungha Yang
058957fc22 h264parse,h265parse: Push parameter set NAL units again per segment-done
Some decoder implementations might drain out internal buffers and
reset its status on segment-done event. So, in case that
upstream stream-format is packetized but downstream supports only
byte-format, required codec-data might not be forwarded toward
downstream if such parameter set NAL units don't exist in inband
bitstream. Therefore, parse elements should re-send parameter set NAL
units like the case of flush event.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/2334>
2021-06-16 18:00:39 +00:00
Nicolas Dufresne
2696bcd9e2 vp8decoder: Drain the output queue on EOS/finish
The finish() virtual method was flushing the queue, instead push the
remaining buffers. It is not required to reset in finish() unlike
drain(). This a regression causing last frame to always be lost.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/2333>
2021-06-16 17:34:54 +00:00
Nicolas Dufresne
e5e2b6a652 v4l2slvp8dec: Only ask for output delay once per negotiation
While it's technically possible to change it per frame, asking for
that every frame is not very useful. This mimic H264 decoder better.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/2333>
2021-06-16 17:34:54 +00:00