Commit graph

26721 commits

Author SHA1 Message Date
Vivia Nikolaidou
82dcb27401 basetsmux: Don't send the capsheader if src pad has no caps
That means we're shutting down, so there's no point in the streamheader
being sent

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1864>
2020-12-09 13:14:40 +00:00
Matthew Waters
1f7515100c rtmp2/connection: pass the parent cancellable down to the connection
Otherwise, when rtpm2src cancels an inflight operation that has a queued
message stored, then the rtmp connection operation is not stopped.

If the cancellation occurs during rtmp connection start up, then
rtpm2src does not have any way of accessing the connection object as it
has not been returned yet.  As a result, rtpm2src will cancel, the
connection will still be processing things and the
GMainContext/GMainLoop associated with the outstanding operation will be
destroyed.  All outstanding operations and the rtmpconnection object will
therefore be leaked in this case.

Fixes: https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/issues/1425
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1862>
2020-12-08 23:43:02 +00:00
Jan Alexander Steffens (heftig)
470e6989d2 srt: Don't take object lock calling gst_srt_object_get_stats
This function takes the sock lock. This can result in a deadlock when
another thread holding the sock lock is trying to take the object lock.

Thread A (Holds object lock, wants sock lock):

    #2  gst_srt_object_get_stats at gst-plugins-bad/ext/srt/gstsrtobject.c:1753
    #3  gst_srt_object_get_property_helper at gst-plugins-bad/ext/srt/gstsrtobject.c:409
    #4  gst_srt_sink_get_property at gst-plugins-bad/ext/srt/gstsrtsink.c:95
    #5  g_object_get_property from libgobject-2.0.so.0

Thread B (Holds sock lock, wants object lock):

    #2  gst_element_post_message_default at gstreamer/gst/gstelement.c:2069
    #3  gst_element_post_message at gstreamer/gst/gstelement.c:2123
    #4  gst_element_message_full_with_details at gstreamer/gst/gstelement.c:2259
    #5  gst_element_message_full at gstreamer/gst/gstelement.c:2298
    #6  gst_srt_object_send_headers at gst-plugins-bad/ext/srt/gstsrtobject.c:1407
    #7  gst_srt_object_send_headers at gst-plugins-bad/ext/srt/gstsrtobject.c:1444
    #8  gst_srt_object_write_to_callers at gst-plugins-bad/ext/srt/gstsrtobject.c:1444
    #9  gst_srt_object_write at gst-plugins-bad/ext/srt/gstsrtobject.c:1598
    #10 gst_srt_sink_render at gst-plugins-bad/ext/srt/gstsrtsink.c:179

Fixes d2d00e07ac.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1861>
2020-12-07 17:59:09 +00:00
Sebastian Dröge
0243afcb9d ccconverter: Add property to specify which sections to include in CDP packets
Various software, including ffmpeg's Decklink support, fails parsing CDP
packets that contain anything but CC data in the CDP packets.

Based on this property, timecodes are not written into the CDP packets
even if they're present.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1833>
2020-12-07 19:23:42 +02:00
Sebastian Dröge
b6debae2c0 ccconverter: Refactor code to only retrieve the timecode meta once
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1833>
2020-12-07 09:40:52 +00:00
Víctor Manuel Jáquez Leal
34683c36de va: decode: fix display type
Instead of a pointer to GstVaDisplay it was used a VADisplay type, which in
certain platforms is the same, and the compiler didn't complain.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1860>
2020-12-06 18:03:47 +01:00
Marc Leeman
102c60f82c rtpmanagerbad: allow setting caps on rtpsrc
rtpsrc tries to do a lookup of the caps based on the encoding-name. For
not so standard encodings, the caps can be set, avoiding the lookup.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1406>
2020-12-04 14:51:38 +00:00
Seungha Yang
96831233a3 d3d11videosink: Add a property to support rendering statistics data on window
Add a new property "render-stats" to allow rendering statistics
data on window for debugging and/or development purpose.
Text rendering will be accelerated by GPU since this implementation
uses Direct2D/DirectWrite API and Direct3D inter-op for minimal overhead.
Specifically, text data will be rendered on swapchain backbuffer
directly without any copy/allocation of extra texture.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1830>
2020-12-04 05:22:14 +09:00
Seungha Yang
22990bb9ea d3d11: Protect ID3D11VideoContext with lock
Likewise d3d11 immediate context (i.e., ID3D11DeviceContext),
ID3D11VideoContext API is not thread safe. It must be protected therefore.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1856>
2020-12-04 03:40:17 +09:00
Mathieu Duponchelle
cc44634422 docs: don't exit the subdir when optional deps aren't found
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1854>
2020-12-03 16:29:59 +00:00
Edward Hervey
d137171f03 opencv: Expose retinex parameters
Makes the plugin a tad more useful :)

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1845>
2020-12-03 17:04:07 +01:00
Marius Vlad
aa68d03013 gst-libs/gst/wayland: Install "unstable" wayland header
Context creation and retrieval is required, the symbols are exported
with the header missing. Users most likely define GST_USE_UNSTABLE_API
so they're aware of the implications of using a header that might change
between releases.

Signed-off-by: Marius Vlad <marius.vlad@collabora.com>
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1688>
2020-12-03 14:54:47 +00:00
Edward Hervey
339ad46b93 hlsdemux: Use actual object for logging
i.e. the pad of the stream

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1853>
2020-12-03 14:31:17 +00:00
Arun Raghavan
81abc4c825 curl: Remove incorrect GST_DEBUG_OBJECT() calls
klass is not a GstObject, and these debugs print should likely not be
around anyway.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1851>
2020-12-03 13:31:38 +00:00
Edward Hervey
dddd0af9cd cuda: Fix lowest targetted architecture for CUDA >= 11.0
Fixes https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/issues/1469

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1835>
2020-12-03 12:23:44 +01:00
Edward Hervey
30ee21eae3 tsparse: Forward incoming timestamps
Ensure we properly forward the upstream PTS/DTS on the regular and program
source pads. All packets being processed will carry over the latest PTS/DTS (as
a reconstructed GstBuffer).

Fixes https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/issues/1419

And properly forward PTS/DTS for program pads (which wasn't the case before)

Original patch by Vivia Nikolaidou <vivia@ahiru.eu>

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1769>
2020-12-02 14:22:06 +00:00
Sebastian Dröge
2f3e245426 adaptivedemux: Don't log with non-GObject objects
Instead of using the streams, log with the pad of the streams.

https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/issues/1457

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1844>
2020-12-02 12:03:36 +00:00
Thibault Saunier
8eb0e637c7 transcodebin: Minor error message enhancement 2020-11-30 17:31:48 -03:00
Thibault Saunier
eb0d72f382 transcodebin: Unlock while setting decodebin caps
Otherwise it will deadlock recursing up to notify parent object property changes
2020-11-30 17:31:48 -03:00
Thibault Saunier
5ccaa595a9 transcodebin: Avoid plugin converter if filter handles ANY caps
For example identity or clocksync or this kind of elements can be
used with any data flow and we should not enforce decoding to row in
that case.
2020-11-30 17:31:48 -03:00
Thibault Saunier
878a196080 transcodebin: Add filter as soon as it is set
Instead of waiting so that we can simply use a clocksync element as
filter, otherwise we won't know the pipeline is live as it won't
return NO_PREROLL as one would expect in that case.

Adding it right away shouldn't create any issue, both ways are fine.
2020-11-30 17:31:48 -03:00
Thibault Saunier
530f694366 uritranscodebin: Add setup-source and element-setup signals
The same way as playbinX does it as it is often quite useful
2020-11-30 17:31:48 -03:00
Thibault Saunier
142e571c28 transcode: Port to encodebin2
This allows supporting muxing sinks like hlssink2 or splitmux
2020-11-30 17:31:48 -03:00
Thibault Saunier
b3544e24ba transcoder: Handle the case where several errors are posted
There were cases where the loop was already destroyed when we were
receiving the following message.
2020-11-30 15:16:01 -03:00
Thibault Saunier
9d890c152e transcoder: Minor refactoring to output better debug logs 2020-11-30 15:16:01 -03:00
Thibault Saunier
f1cf5d0683 hlssink2: Mark as Muxer
The way it is usable by encodebin2. This is what splitmux does already.
2020-11-30 15:16:01 -03:00
Víctor Manuel Jáquez Leal
ef62e6cfa2 va: decoder: Picture dups only holds GstBuffer
Also removes the warning log message at destroying buffers when picture free()

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1841>
2020-11-30 17:12:14 +01:00
Víctor Manuel Jáquez Leal
14c28415b9 va: Remove gst_va_decoder_destroy_buffers()
Since GstVaDecodePicture is destroyed completely with its free() function and
it's used as destroy notify by codecs picture, there's no need to call
gst_va_decoder_destroy_buffers() externally, since the codecs base classes
destroy the codec picture when it's required.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1841>
2020-11-30 16:53:25 +01:00
He Junyan
f5c7ada98e va: Destroy picture unreleased buffers when finalize.
The current way of GstVaDecodePicture's finalize will leak some
resource such as parameter buffers and slice data.
The current way deliberately leaves these resource releasing logic
to va decoder related function and trigger a warning if we free the
GstVaDecodePicture without releasing these resources.
But in practice, sometimes, you do not have the chance to release
these resource before picture is freed. For example, H264/Mpeg2
support multi slice NALs/Packets for one frame. It is possible that
we already succeed to parse and generate the first several slices
data by _decode_slice(), but then we get a wrong slice NAL/packet
and fail to parse it. We decide to discard the whole frame in the
decoder's base class, it just free the current picture and does not
trigger sub class's function again. In this kind of cases, we do
not have the chance to cleanup the resource, and the resource will
be leaked.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1841>
2020-11-30 13:03:11 +00:00
Thibault Saunier
d608636327 qroverlay: Reuse the same OverlayComposition object when possible
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1829>
2020-11-26 14:34:34 +00:00
Thibault Saunier
ad5f812c91 qroverlay: Rework basing it on overlaycomposition
The base class is now a bin which wraps the `overlaycomposition`
element and implements the `draw` signal.

This way we support all the video formats the GstVideoOverlayComposition
API supports and the blending code can be reused. It is also possible
to have the blending happen in the sinks now.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1829>
2020-11-26 14:34:34 +00:00
Seungha Yang
3e35a6f03f d3d11h264dec: Reconfigure decoder object on DPB size change
Even if resolution and/or bitdepth is not updated, required
DPB size can be changed per SPS update and it could be even
larger than previously configured size of DPB. If so, we need
to reconfigure DPB d3d11 texture pool again.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1839>
2020-11-26 08:52:49 +00:00
Marijn Suijten
dc90a3d3cf audio: Use new AudioFormatInfo::fill_silence function
The function is renamed to be properly associated with AudioFormatInfo
(its instance) instead of AudioFormat (an unrelated enum), see [1] for
the rename itself.

[1]: https://gitlab.freedesktop.org/gstreamer/gst-plugins-base/-/merge_requests/940
2020-11-26 10:06:42 +02:00
Philippe Normand
d4df91cd9b player: Fix get_current_subtitle_track annotation
As the info returned is a new object, the annotation should be transfer-full,
similarly to the get_current_{audio,video}_track() implementations.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1775>
2020-11-25 11:30:16 +00:00
Víctor Manuel Jáquez Leal
8e6ad8267f va: allocator: add a memory pool object helper
Since both allocators use a memory pool, with its mutex and cond, this patch
refactors it into a single internal object, implementing a generic GstMemory
pool.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1815>
2020-11-24 13:00:00 +01:00
Víctor Manuel Jáquez Leal
8c128ead6d va: pool, allocator: honor GST_BUFFER_POOL_ACQUIRE_FLAG_DONTWAIT
In order to honor GST_BUFFER_POOL_ACQUIRE_FLAG_DONTWAIT in VA pool, allocators'
wait_for_memory() has to be decoupled from their prepare_buffer() so it could be
called in pools' acquire_buffer() if the flag is not set.

wait_for_memory() functions are blocking so the received memories are assigned
to the fist requested buffer, if multithreaded calls. For this a new mutex were
added.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1815>
2020-11-24 12:44:24 +01:00
Víctor Manuel Jáquez Leal
8fc50891b1 va: allocator: broadcast when flushing
This patch handles when the bufferpool request a new buffer while
flushing.

Also fixes the usage of g_cond_wait(), which demands to be used
inside a loop to avoid spurious wakeups.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1815>
2020-11-24 12:44:24 +01:00
Víctor Manuel Jáquez Leal
72ab56c376 va: allocator: free allocator when a mem is held
An application, using for example appsink, can hold buffers from any
va allocator after setting the pipeline to NULL. We need to destroy
the allocator when that memory is unrefed.

This patch juggles a bit with the allocator reference count in
memories in order to achieve this:

1. When memory is created no alloc ref is modified
2. When memory is released, alloc ref is decreased
3. When memory is reassiged to a buffer, alloc ref is increased
4. When memory is flushed, alloc ref is increased becase it is going
   to be decreased in gst_memory_unref()

Also this patch moves the deallocation of member variables to
finalize() rather than dispose()

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1815>
2020-11-24 12:44:24 +01:00
Víctor Manuel Jáquez Leal
ba4442a29d va: allocator: dmabuf: initialize cond
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1815>
2020-11-24 12:44:24 +01:00
Olivier Crête
a801018ef1 webrtc: Make ssrc map into separate data structures
They now contain a weak reference and that could be freed later
causing strange crashes as GWeakRef are not movable.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1766>
2020-11-24 04:27:52 +00:00
Olivier Crête
1deb034e3d webrtcstats: Get the remote-inbound stats from the right RTPSource
This also means that we need to get the clock-rate from the codec instead
of from the RTPSource, as the remote one doesn't include a clock rate.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1766>
2020-11-24 04:27:52 +00:00
Olivier Crête
1c1661b54f webrtcbin: Implement getting stats for a specific pad
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1766>
2020-11-24 04:27:52 +00:00
Olivier Crête
23ea950351 webrtcstats: Also return the raw rtpsource stats for more information
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1766>
2020-11-24 04:27:52 +00:00
Olivier Crête
b895240241 webrtcstats: Avoid copy of GstStructure
Instead transfer the ownership to the new structure

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1766>
2020-11-24 04:27:52 +00:00
Olivier Crête
a46c6e3a97 webrtcstats: Remove receiver side when sending
Those are just invalid and just reflect what we sent. We'd need to parse the
RTCP XR packets from the other side to know more about those.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1766>
2020-11-24 04:27:52 +00:00
Olivier Crête
ba0dfa52d2 webrtcstats: Extract statistics from the rtpjitterbuffer
And expose them as standardised webrtc statistics

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1766>
2020-11-24 04:27:52 +00:00
Olivier Crête
fc0f6db856 webrtcbin: Store the rtpjitterbuffer instances to extract stats from them
Store them as web refs to avoid having to worry about freeing later and because
the new-jitterbuffer is on a different thread

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1766>
2020-11-24 04:27:52 +00:00
Olivier Crête
d9d7814182 webrtcstats: Document all RTP missing fields according to the latest spec
Just document all the missing fields and document which ones will never
be implemented because they depend on the codec or depayloader

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1766>
2020-11-24 04:27:52 +00:00
Olivier Crête
895ea210c2 webrtcstats: RTCP computed RTT is only available at sender
The receiver doesn't have the information to compute it.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1766>
2020-11-24 04:27:52 +00:00
Olivier Crête
a5c3331197 webrtcstats: Remove redundant lines
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1766>
2020-11-24 04:27:52 +00:00