Sebastian Dröge [Wed, 16 Dec 2020 16:32:25 +0000 (18:32 +0200)]
decklinkaudiosrc: Fix duration of the first audio frame after each discont
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1886>
Biswapriyo Nath [Tue, 15 Dec 2020 18:58:08 +0000 (00:28 +0530)]
mediafoundation: Fix redefinition of variables.
Remove duplicate GstMFDevice and GstMFDeviceProvider declaration.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1884>
Jan Schmidt [Wed, 16 Dec 2020 17:41:18 +0000 (04:41 +1100)]
audiobuffersplit: Calculate the correct size for fixed size buffers
Fix the output-buffer-size property to do what it says by calculating
the correct audio buffer size for that target size, rounded down to
the nearest whole number of samples.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1887>
Sebastian Dröge [Thu, 10 Dec 2020 10:35:07 +0000 (12:35 +0200)]
decklink: Implement GstBaseSrc::get_caps() to return more constrained caps
Instead of the template caps we can return a subset of them based on the
selected properties.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1868>
Seungha Yang [Thu, 29 Oct 2020 17:21:11 +0000 (02:21 +0900)]
wasapi2: Ensure unmute when opening audio client
ISimpleAudioVolume::SetMute() status seems to be preserved even
after process is terminated. In order to start audio client with
unmuted state, always disable mute when opening audio client.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1731>
Edward Hervey [Mon, 14 Dec 2020 15:12:22 +0000 (16:12 +0100)]
tsparse: Don't use non-object for debugging statement
Use the pad instead
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1880>
Edward Hervey [Mon, 14 Dec 2020 09:56:39 +0000 (10:56 +0100)]
examples/ts-parser: Use the section type for descriptor identification
Some descriptors can only be present in some section
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1880>
Edward Hervey [Mon, 14 Dec 2020 09:56:02 +0000 (10:56 +0100)]
examples/ts-parser: Try more descriptor/stream types
These were added recently
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1880>
Edward Hervey [Wed, 9 Dec 2020 08:14:12 +0000 (09:14 +0100)]
mpegts: Don't add non-padded streams to collection on updates
When carrying over existing GstStream to a new GstStreamCollection we need to
check whether they *actually* were being used in the previous collection.
This avoids adding unknown streams (metadata, PSI, etc...) to the collection on
updates.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1880>
Edward Hervey [Sun, 22 Nov 2020 17:48:08 +0000 (18:48 +0100)]
mpegts: Add support for SIT sections
Selection Information Tables (EN 300 468)
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1852>
Edward Hervey [Mon, 14 Dec 2020 09:50:02 +0000 (10:50 +0100)]
mpegts: Update documentation
* Split up into appropriate individual header files
* Document more sections and structures
* Add well-known list of registration id
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1879>
Thibault Saunier [Thu, 10 Dec 2020 19:29:31 +0000 (16:29 -0300)]
player/transcoder: Use bus signal watch
Instead of implementing exactly the same thing ourself but making
`GstBus` not know that it is the case.
Since we are *sure* that the bus can't have been access at the point
where we add the watch we are guaranteed that the current thread
maincontext is going to be used.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1870>
Lim Siew Hoon [Thu, 10 Dec 2020 07:37:14 +0000 (15:37 +0800)]
intervideosrc: fix negotiation of interlaced caps
In 1.0 the field in caps is called "interlace-mode", not "interlaced".
Fixes #1480
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1869>
Arun Raghavan [Sat, 12 Dec 2020 02:45:25 +0000 (21:45 -0500)]
openaptx: Drop lib prefix from option name for consistency
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1876>
Igor Kovalenko [Fri, 11 Dec 2020 08:45:06 +0000 (08:45 +0000)]
openaptx: add aptX and aptX-HD codecs using libopenaptx
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1871>
Philippe Normand [Mon, 19 Oct 2020 13:56:43 +0000 (14:56 +0100)]
wpe: Emit load-progress messages
The estimated-load-progress value can be used on application side to display a
progress bar for instance.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1710>
Vivia Nikolaidou [Tue, 8 Dec 2020 14:46:42 +0000 (16:46 +0200)]
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>
Matthew Waters [Fri, 4 Dec 2020 06:02:00 +0000 (17:02 +1100)]
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>
Jan Alexander Steffens (heftig) [Mon, 7 Dec 2020 13:54:28 +0000 (14:54 +0100)]
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
d2d00e07acc2b1ab1ae5a728ef5dc33c9dee7869.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1861>
Sebastian Dröge [Wed, 25 Nov 2020 14:24:25 +0000 (16:24 +0200)]
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>
Sebastian Dröge [Wed, 25 Nov 2020 12:54:09 +0000 (14:54 +0200)]
ccconverter: Refactor code to only retrieve the timecode meta once
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1833>
Víctor Manuel Jáquez Leal [Sun, 6 Dec 2020 17:03:47 +0000 (18:03 +0100)]
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>
Marc Leeman [Fri, 3 Jul 2020 10:25:31 +0000 (12:25 +0200)]
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>
Seungha Yang [Sat, 21 Nov 2020 19:39:57 +0000 (04:39 +0900)]
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>
Seungha Yang [Thu, 3 Dec 2020 18:40:17 +0000 (03:40 +0900)]
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>
Mathieu Duponchelle [Thu, 3 Dec 2020 16:13:15 +0000 (17:13 +0100)]
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>
Edward Hervey [Wed, 2 Dec 2020 10:29:08 +0000 (11:29 +0100)]
opencv: Expose retinex parameters
Makes the plugin a tad more useful :)
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1845>
Marius Vlad [Mon, 12 Oct 2020 11:12:07 +0000 (14:12 +0300)]
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>
Edward Hervey [Thu, 3 Dec 2020 13:12:06 +0000 (14:12 +0100)]
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>
Arun Raghavan [Thu, 3 Dec 2020 11:55:00 +0000 (06:55 -0500)]
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>
Edward Hervey [Wed, 25 Nov 2020 16:59:54 +0000 (17:59 +0100)]
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>
Edward Hervey [Thu, 5 Nov 2020 11:48:27 +0000 (13:48 +0200)]
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>
Sebastian Dröge [Wed, 2 Dec 2020 07:39:45 +0000 (09:39 +0200)]
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>
Thibault Saunier [Fri, 20 Nov 2020 14:29:46 +0000 (11:29 -0300)]
transcodebin: Minor error message enhancement
Thibault Saunier [Fri, 20 Nov 2020 01:56:46 +0000 (22:56 -0300)]
transcodebin: Unlock while setting decodebin caps
Otherwise it will deadlock recursing up to notify parent object property changes
Thibault Saunier [Thu, 19 Nov 2020 21:31:34 +0000 (18:31 -0300)]
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.
Thibault Saunier [Thu, 19 Nov 2020 21:39:33 +0000 (18:39 -0300)]
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.
Thibault Saunier [Thu, 19 Nov 2020 21:29:15 +0000 (18:29 -0300)]
uritranscodebin: Add `setup-source` and `element-setup` signals
The same way as playbinX does it as it is often quite useful
Thibault Saunier [Thu, 19 Nov 2020 20:55:10 +0000 (17:55 -0300)]
transcode: Port to encodebin2
This allows supporting muxing sinks like hlssink2 or splitmux
Thibault Saunier [Thu, 19 Nov 2020 20:55:10 +0000 (17:55 -0300)]
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.
Thibault Saunier [Thu, 19 Nov 2020 20:54:28 +0000 (17:54 -0300)]
transcoder: Minor refactoring to output better debug logs
Thibault Saunier [Thu, 19 Nov 2020 20:51:56 +0000 (17:51 -0300)]
hlssink2: Mark as Muxer
The way it is usable by encodebin2. This is what splitmux does already.
Víctor Manuel Jáquez Leal [Mon, 30 Nov 2020 16:12:14 +0000 (17:12 +0100)]
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>
Víctor Manuel Jáquez Leal [Mon, 30 Nov 2020 14:01:01 +0000 (15:01 +0100)]
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>
He Junyan [Thu, 26 Nov 2020 06:04:31 +0000 (14:04 +0800)]
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>
Thibault Saunier [Sat, 21 Nov 2020 22:00:02 +0000 (19:00 -0300)]
qroverlay: Reuse the same OverlayComposition object when possible
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1829>
Thibault Saunier [Fri, 20 Nov 2020 14:28:25 +0000 (11:28 -0300)]
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>
Seungha Yang [Wed, 25 Nov 2020 20:55:29 +0000 (05:55 +0900)]
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>
Marijn Suijten [Wed, 25 Nov 2020 16:52:42 +0000 (17:52 +0100)]
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
Philippe Normand [Thu, 5 Nov 2020 17:14:22 +0000 (17:14 +0000)]
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>
Víctor Manuel Jáquez Leal [Mon, 23 Nov 2020 19:44:27 +0000 (20:44 +0100)]
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>
Víctor Manuel Jáquez Leal [Tue, 17 Nov 2020 13:53:05 +0000 (14:53 +0100)]
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>
Víctor Manuel Jáquez Leal [Tue, 17 Nov 2020 12:18:37 +0000 (13:18 +0100)]
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>
Víctor Manuel Jáquez Leal [Tue, 17 Nov 2020 12:17:03 +0000 (13:17 +0100)]
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>
Víctor Manuel Jáquez Leal [Mon, 23 Nov 2020 16:01:52 +0000 (17:01 +0100)]
va: allocator: dmabuf: initialize cond
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1815>
Olivier Crête [Fri, 20 Nov 2020 22:32:44 +0000 (17:32 -0500)]
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>
Olivier Crête [Fri, 16 Oct 2020 01:23:08 +0000 (21:23 -0400)]
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>
Olivier Crête [Thu, 15 Oct 2020 23:36:45 +0000 (19:36 -0400)]
webrtcbin: Implement getting stats for a specific pad
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1766>
Olivier Crête [Sat, 10 Oct 2020 22:21:19 +0000 (18:21 -0400)]
webrtcstats: Also return the raw rtpsource stats for more information
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1766>
Olivier Crête [Sat, 10 Oct 2020 00:59:58 +0000 (20:59 -0400)]
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>
Olivier Crête [Sat, 10 Oct 2020 00:45:10 +0000 (20:45 -0400)]
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>
Olivier Crête [Sat, 10 Oct 2020 00:27:40 +0000 (20:27 -0400)]
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>
Olivier Crête [Fri, 9 Oct 2020 22:45:57 +0000 (18:45 -0400)]
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>
Olivier Crête [Fri, 9 Oct 2020 23:59:18 +0000 (19:59 -0400)]
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>
Olivier Crête [Fri, 9 Oct 2020 23:38:15 +0000 (19:38 -0400)]
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>
Olivier Crête [Thu, 8 Oct 2020 21:11:30 +0000 (17:11 -0400)]
webrtcstats: Remove redundant lines
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1766>
Olivier Crête [Wed, 4 Nov 2020 22:06:02 +0000 (17:06 -0500)]
webrtc: Also remove rtcp_transport from the structure
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1765>
Olivier Crête [Tue, 3 Nov 2020 00:55:46 +0000 (19:55 -0500)]
webrtc: Remove APIs to set transport on sender/receiver
They're not not used ever.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1765>
Olivier Crête [Tue, 3 Nov 2020 00:49:55 +0000 (19:49 -0500)]
webrtc: Remove non rtcp-mux code
RTCP mux is now always required by the WebRTC spec
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1765>
Julian Bouzas [Fri, 20 Nov 2020 15:01:03 +0000 (15:01 +0000)]
nvcodec: Assume 25fps if framerate is invalid when calculating latency
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1826>
Víctor Manuel Jáquez Leal [Fri, 20 Nov 2020 21:26:14 +0000 (22:26 +0100)]
codecs: h264decoder: fix memory leak
gst_h264_dbp_get_picture_all() returns a full transfer of the GArray, which
needs be unrefed. But it is not unrefed in
gst_h264_decoder_find_first_field_picture() leaking it.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1827>
Edward Hervey [Fri, 20 Nov 2020 15:07:36 +0000 (16:07 +0100)]
mpegts: Documentation fixes
gtk-doc was complaining :)
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1825>
Tim-Philipp Müller [Fri, 20 Nov 2020 13:24:24 +0000 (13:24 +0000)]
qroverlay: unset executable flag on source files
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1824>
Tim-Philipp Müller [Fri, 20 Nov 2020 13:22:48 +0000 (13:22 +0000)]
qroverlay: fix auto detection of json-glib for plugin
Only want to check for json-glib when libqrencode was found,
but also it shouldn't be required but depend on the option.
Fixes #1465
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1824>
Seungha Yang [Thu, 19 Nov 2020 12:15:25 +0000 (21:15 +0900)]
d3d11: Add support for packed 4:2:2 and 4:4:4 10bits formats
Add support for Y210 and Y410 formats which are commonly used format
for en/decoders on Windows. Note that those formats cannot be used for
render target (output) of shader.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1821>
Olivier Crête [Fri, 2 Oct 2020 22:47:16 +0000 (18:47 -0400)]
openh264dec: Accept constrained-high and progressive-high profiles
They're just subsets of the high profile.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1634>
Olivier Crête [Fri, 2 Oct 2020 22:47:06 +0000 (18:47 -0400)]
d3d11h264dec: Accept constrained-high and progressive-high profiles
They're just subsets of the high profile.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1634>
Olivier Crête [Fri, 2 Oct 2020 22:46:56 +0000 (18:46 -0400)]
msdkh264dec: Accept constrained-high and progressive-high profiles
They're just subsets of the high profile.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1634>
Olivier Crête [Tue, 22 Sep 2020 19:42:37 +0000 (15:42 -0400)]
nvdec: Accept progressive-high and contrained-high profiles
They're subsets of the high profiles with no interlacing and
no B-frames for constrained
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1634>
He Junyan [Mon, 28 Sep 2020 05:33:00 +0000 (13:33 +0800)]
codecparsers: av1: add the set_operating_point() API.
The av1 can support multi layers when scalability is enabled. We
need an API to set the operating point and filter the OBUs just
belonging to some layers(the layers are specified by the operating
point).
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1464>
He Junyan [Fri, 9 Oct 2020 08:13:28 +0000 (16:13 +0800)]
codecparsers: av1: Add an API to reset the annex_b state only.
In practice, we encounter streams that have one or more temporal units
error. When that kind of error temporal units is in annex b format, the
whole temporal unit should be discarded.
But the temporal units before it are correct and can be used. More
important, because of the error temporal unit, the parser is in a wrong
state and all later temporal unit are also parsed uncorrectly.
We need to add this API to reset the annex_b state only when we meet
some temporal unit error.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1464>
He Junyan [Fri, 9 Oct 2020 08:01:35 +0000 (16:01 +0800)]
codecparsers: av1: clean the seen_frame_header in parse_tile_group().
The current seen_frame_header is not cleaned correctly. According
to the spec, it should be cleaned when tiles are parsed completely.
Also delete a verbose seen_frame_header init in reset_state().
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1464>
He Junyan [Tue, 29 Sep 2020 05:15:37 +0000 (13:15 +0800)]
codecparsers: av1: fix a typo in parse_metadata_scalability
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1464>
He Junyan [Mon, 28 Sep 2020 10:22:08 +0000 (18:22 +0800)]
codecparsers: av1: Do not assert in identify_one_obu when check annex b size.
Some buggy stream just writes the wrong temporal unit and frame size in
the stream. We should return failure rather than assert to abort.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1464>
He Junyan [Tue, 22 Sep 2020 11:16:30 +0000 (19:16 +0800)]
codecparsers: av1: Add unknow AV1 profile define for saint check.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1464>
He Junyan [Fri, 24 Jul 2020 06:54:37 +0000 (14:54 +0800)]
codecparsers: av1: Improve the parse_tile_info.
1. store more tile info when parse tile group.
The column, row, tile offset and tile data size are all useful for
decoder process, especially for HW kind decoder such as VAAPI dec.
Also fix the tile group skip size for each tile data.
2. No min_inner_tile_width requirement in newest spec.
3. Calculate the sbs of each tile for both uniform tile and non-uniformi
tile.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1464>
He Junyan [Tue, 28 Jul 2020 09:25:44 +0000 (17:25 +0800)]
codecparsers: av1: Fix a tile info read typo in frame header.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1464>
He Junyan [Tue, 25 Aug 2020 11:44:48 +0000 (19:44 +0800)]
codecparsers: av1: Fix a typo when get value of segmentation params.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1464>
He Junyan [Tue, 25 Aug 2020 08:33:26 +0000 (16:33 +0800)]
codecparsers: av1: add valid check for global motion params.
The global motion params and its matrix values need to be verified
before we use them. If it is invalid, we should notify the decoder
that it should not be used.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1464>
He Junyan [Tue, 25 Aug 2020 07:25:56 +0000 (15:25 +0800)]
codecparsers: av1: uint8 range is not enough for av1_bitstreamfn_ns
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1464>
He Junyan [Tue, 25 Aug 2020 07:25:06 +0000 (15:25 +0800)]
codecparsers: av1: delete duplicated GST_AV1_GM_ABS_ALPHA_BITS define.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1464>
He Junyan [Thu, 27 Aug 2020 13:33:14 +0000 (21:33 +0800)]
test: av1parser: update the test result because of bug fixing.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1464>
He Junyan [Mon, 24 Aug 2020 07:29:56 +0000 (15:29 +0800)]
codecparsers: av1: Improve the loop filter setting.
1. loop_filter_ref_deltas should be int because it needs to compare
with 0.
2. Move the loop filter init logic to setup_past_independence() and
load_previous(), which make it more precise with the spec.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1464>
He Junyan [Fri, 14 Aug 2020 06:40:49 +0000 (14:40 +0800)]
codecparsers: av1: Fix a error report for metadata obu.
The metadata OBUs, for example, ITUT_T35 has an undefined payload such
as itu_t_t35_payload_bytes field in AV1 spec, which may cause the failure
of parsing the trailings bits. We can give a warning and ignore this kind
of errors.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1464>
He Junyan [Tue, 28 Jul 2020 07:06:04 +0000 (15:06 +0800)]
codecparsers: av1: Fix a level index bug in sequence.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1464>
He Junyan [Fri, 24 Jul 2020 04:49:10 +0000 (12:49 +0800)]
codecparsers: av1: all ref idx should be gint8.
All the ref index need to compare with 0 in reference index decision
algorithm. We also need to init them to -1.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1464>
Seungha Yang [Sat, 14 Nov 2020 09:48:05 +0000 (18:48 +0900)]
va: h264dec: Add support for interlaced stream
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1812>
Víctor Manuel Jáquez Leal [Mon, 16 Nov 2020 15:29:04 +0000 (16:29 +0100)]
va: h264dec: implement gst_va_h264_dec_new_field_picture()
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1812>
Seungha Yang [Sat, 14 Nov 2020 11:46:30 +0000 (20:46 +0900)]
va: h264dec: Fix picture_height_in_mbs_minus1
Fix for interlaced stream (when sps->frame_mbs_only_flag is equal to one)
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1812>
Víctor Manuel Jáquez Leal [Mon, 16 Nov 2020 15:29:46 +0000 (16:29 +0100)]
va: h264dec: admit baseline if stream obeys A.2
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1812>