Gilbok Lee [Mon, 17 Jan 2022 05:10:42 +0000 (14:10 +0900)]
Merge branch 'tizen' into tizen_gst_1.19.2
Change-Id: I904018a26020868a46b46717571e6786b0362697
Gilbok Lee [Wed, 1 Dec 2021 00:34:36 +0000 (09:34 +0900)]
subparse: Add drop-out-of-segment property
- When property 'drop-out-of-segment' is set to false,
subparser will not drop the buffer even
the start time of the parsed buffer is out of the segment.
Change-Id: Ia7d84ad26c2e93cda46778c86d35bd67442e907b
Gilbok Lee [Wed, 24 Nov 2021 10:28:03 +0000 (19:28 +0900)]
subparse: Calcurate buffer pts using timestamp map for HLS webvtt
- parsing error occurs due to out of segment
Change-Id: Ib7945d1d3e64ed2568df94c77436444117fb9ea5
Gilbok Lee [Wed, 24 Nov 2021 00:33:17 +0000 (09:33 +0900)]
subparse: Send custom event for fragment_timestamp
- If there is no buffer in case of discontinuous,
do not send reference timestamp.
- related commit:
d4e6aa89f86efbc9cc665f2ee123a33015f1449a
'subparse: Add reference timestamp meta in GstBuffer for HLS webvtt' commit
Change-Id: Id98697ba6db1dc94b4ce4f753670524f6fcf506e
Gilbok Lee [Sat, 20 Nov 2021 06:39:08 +0000 (15:39 +0900)]
subparse: Add reference timestamp meta in GstBuffer for HLS webvtt
- When discontinuous buffer come in during HLS,
send the input buffer pts to the reference timestamp meta data
- The reference timetamp meta data is added with the input buffer pts
to synchronize with the mpeg ts stream
Change-Id: I5ff5b9523b44323f1d6aa37133e5341505d4ce55
Eunhye Choi [Tue, 28 Sep 2021 07:08:28 +0000 (16:08 +0900)]
decodebin3: Avoid overriding explicit user selection
In case the user set a list of streams to select or answer explicitly
to all 'select-stream' event, we should respect the choice and not
try to add a stream per type.
related upstream commit :
b41b87522f59355bb21c001e9e2df96dc6956928
c9c93339fbd2d37f1ddfd054f7f9e26bce6df743
40fde5fcad0bcdb5429d7bf573690cfe55fc79c8
Change-Id: I63b75bb02fbe40392ae3edbf83a9830d7b606437
Tim-Philipp Müller [Thu, 23 Sep 2021 00:33:08 +0000 (01:33 +0100)]
Release 1.19.2
Tim-Philipp Müller [Tue, 21 Sep 2021 21:39:46 +0000 (22:39 +0100)]
pbutils: codec-utils: fix g-ir-scanner warning
Warning: GstPbutils: gst_codec_utils_h264_get_profile_flags_level:
unknown parameter 'codec_data' in documentation comment, should be 'codecs_data
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-base/-/merge_requests/1279>
Nicolas Dufresne [Tue, 24 Aug 2021 19:27:32 +0000 (15:27 -0400)]
alsasink: Allow stop() function to happen during failing writes
In ALSA, there is possible temporary failures that may require a retry,
though in certain situation, this may leak to the write() function
holding on a lock forever preventing the pipeline from going to pause
or stop. Fix this by shortly dropping the lock between retries.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-base/-/merge_requests/1261>
Nicolas Dufresne [Tue, 24 Aug 2021 19:26:12 +0000 (15:26 -0400)]
alsasink: Improve logging in write() function
This moves the "written X frames" lower so that we don't trace
confusing negative values on errors and add the error code in the
"Write error" log.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-base/-/merge_requests/1261>
Sebastian Dröge [Fri, 10 Sep 2021 12:10:46 +0000 (15:10 +0300)]
videodecoder: Add properties to automatically request sync points and vfunc to allow subclasses to handle packet loss / missing data
Subclasses could use the new vfunc to activate packet loss concealment,
for example.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-base/-/merge_requests/1274>
Havard Graff [Fri, 10 Sep 2021 14:12:51 +0000 (16:12 +0200)]
videodecoder: Fix min-force-key-unit-interval logic and logging
The new keyframe is needed when the deadline of the buffer has exeeded
the waiting time, not while it is within it.
Also, since we look at the deadline of the frame, log that instead of PTS.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-base/-/merge_requests/1278>
Olivier Crête [Wed, 18 Aug 2021 23:40:55 +0000 (19:40 -0400)]
rtphdrext: Make write function return a signed value
Since the return value is documented to possibly be smaller than 0,
then it needs to be signed.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-base/-/merge_requests/1258>
Olivier Crête [Wed, 16 Jun 2021 19:07:13 +0000 (15:07 -0400)]
videorate: Add unit test for closing a segment and opening a separate one
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-base/-/merge_requests/767>
Olivier Crête [Wed, 16 Jun 2021 19:06:57 +0000 (15:06 -0400)]
videorate: Drop incoming buffers that are outside of the segment
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-base/-/merge_requests/767>
Olivier Crête [Fri, 24 Jul 2020 21:41:57 +0000 (17:41 -0400)]
videorate: Only "close" the segment if it is discontinous
Otherwise, it will drop valid buffers on a simple segment update
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-base/-/merge_requests/767>
Olivier Crête [Fri, 24 Jul 2020 21:38:58 +0000 (17:38 -0400)]
videorate: Add test for segment update
Continue as-is on segment update.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-base/-/merge_requests/767>
Olivier Crête [Fri, 24 Jul 2020 20:35:04 +0000 (16:35 -0400)]
videorate: Update the base time on segment updates
Dropping it to 0 makes videorate push buffers from timestamp 0 again.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-base/-/merge_requests/767>
Jeongmo Yang [Mon, 13 Sep 2021 11:16:35 +0000 (20:16 +0900)]
cameracontrol: Add new interface for extra preview GOP interval
[Version] 1.16.2-18
[Issue Type] New feature
Change-Id: I5bff2e0433c87bc2daf2949804719cd41c1b48f2
Signed-off-by: Jeongmo Yang <jm80.yang@samsung.com>
Philippe Normand [Sun, 12 Sep 2021 09:07:49 +0000 (10:07 +0100)]
discoverer: Prevent stream tags from leaking in global tags
The PrivateStream should keep track of stream tags only. Likewise, the
GstDiscovererInfo should keep track of global tags only.
This patch fixes the issue where the discoverer would report duplicated tag
titles, especially for Matroska media files. The Matroska demuxer emits
correctly-scoped tags, but downstream was making no distinction of them.
Fixes #598, #836, https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/issues/827
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-base/-/merge_requests/1275>
Matthew Waters [Thu, 9 Sep 2021 05:44:55 +0000 (15:44 +1000)]
gl/buffer_storage: re-enable GL_ARB_buffer_storage
The extension version doesn't have the ARB suffix.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-base/-/merge_requests/1273>
Tobias Ronge [Tue, 7 Sep 2021 11:55:08 +0000 (13:55 +0200)]
rtspconnection: Only reset timeout when socket is unused
After sending or retrieving data, gstrtspconnection resets the socket's
timeout to 0 (infinite). This could cause problems if sending and
receiving at the same time. For example, if RTCP data is sent from the
streaming thread while gstrtspsrc is already retrieving data.
With this patch, timeout is only reset to 0 if there is no other
thread using the socket.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-base/-/merge_requests/1260>
Ludvig Rappe [Thu, 2 Sep 2021 09:55:09 +0000 (11:55 +0200)]
pbutils: Add mjpg to MIME codecs
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-base/-/merge_requests/1270>
Mathieu Duponchelle [Sun, 1 Aug 2021 16:20:06 +0000 (18:20 +0200)]
decodebin3: fix unblocking on input gap events
Initial gap events should not be discarded on the input streams,
but instead cause unblocking just as buffers do.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-base/-/merge_requests/1239>
Philippe Normand [Thu, 24 Jun 2021 15:00:03 +0000 (16:00 +0100)]
parsebin: Guess subtitle/ caps as text streams
The subtitles in ogg/kate are identified using subtitle/ caps names.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-base/-/merge_requests/1213>
Jeongmo Yang [Wed, 1 Sep 2021 11:54:26 +0000 (20:54 +0900)]
cameracontrol: Add new interface for extra preview bitrate
[Version] 1.16.2-17
[Issue Type] New feature
Change-Id: Iaace8d2d4814cb782809daf87905d4ec946664f8
Signed-off-by: Jeongmo Yang <jm80.yang@samsung.com>
Matthew Waters [Tue, 31 Aug 2021 05:31:23 +0000 (15:31 +1000)]
rtp: add some additional rtcp sdes values
Matches the current list at
https://www.iana.org/assignments/rtp-parameters/rtp-parameters.xhtml#rtp-parameters-5
as of 2021-September.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-base/-/merge_requests/1267>
Ludvig Rappe [Wed, 25 Aug 2021 15:03:49 +0000 (17:03 +0200)]
pbutils: Add function to convert caps to MIME codec
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-base/-/merge_requests/1265>
Ludvig Rappe [Wed, 25 Aug 2021 15:01:19 +0000 (17:01 +0200)]
pbutils: Add function for parsing H.264 extradata
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-base/-/merge_requests/1265>
Olivier Blin [Tue, 24 Aug 2021 16:14:22 +0000 (18:14 +0200)]
eglimage: fix redefinition of EGLuint64KHR
It is already defined in gst/gl/egl/gstegl.h
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-base/-/merge_requests/1262>
Seungha Yang [Thu, 5 Aug 2021 10:59:38 +0000 (19:59 +0900)]
video-converter: Add support for A420 to RGB fast path
Add fast path for A420 -> RGB format conversion
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-base/-/merge_requests/1245>
Seungha Yang [Mon, 19 Jul 2021 13:11:41 +0000 (22:11 +0900)]
compositor: Fix crash while drawing background and/or blending for subsampled YUV
Fix crash caused by out-of-bounds memory accesses when drawing
background and/or blending. This fix is conceptually identical to the
approach as the commit of
8ff5079e5eef37b9bd5b212350f0cefbd9546b1b
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-base/-/merge_requests/1229>
Mathieu Duponchelle [Sat, 14 Aug 2021 23:27:39 +0000 (01:27 +0200)]
encoding-profile: ignore more encoding private fields
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-base/-/merge_requests/1249>
Thibault Saunier [Tue, 10 Aug 2021 01:24:34 +0000 (21:24 -0400)]
smartencoder: Respect user `stream-format` when specified
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-base/-/merge_requests/1249>
Mathieu Duponchelle [Tue, 10 Aug 2021 00:09:09 +0000 (02:09 +0200)]
smartencoder: clean up and extend accepted formats
* Add support for H265
* Don't overwrite original codec_data / streamheader in the output
caps, but instead allow them to change and send them to the
combiner at the right moment: encoder caps, reencoded GOP,
original caps, original GOP(s), and potentially encoder caps
and rencoded last GOP.
* For H264 / H265, force usage of a format with inband SPS / PPS
(avc3 / hev1), this is cleaner than misadvertising avc1, hvc1 and
some muxers like mp4mux will actually advertise both differently.
Unfortunately, while mp4 supports updating the codec_data and using
avc1 with no in-band SPS / PPS updates, it turns out some decoders
(eg chrome / firefox) don't handle this particularly well and stop
decoding after the reencoded GOP. We could expose a switch to
force usage of avc1 / hvc1 nevertheless, but for now stick to
requiring that the parser output SPS / PPS in-band with
config-interval=-1 (that has not changed)
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-base/-/merge_requests/1249>
Seungha Yang [Tue, 3 Aug 2021 15:51:24 +0000 (00:51 +0900)]
compositor: Add "max-threads" property
Adding new property for user to be able to set expected the maximum
number of blend task threads. This can be useful in case that user
wants to restrict the number of parallel task runners for system
resource management or debugging/development purpose.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-base/-/merge_requests/1242>
Sebastian Dröge [Thu, 1 Jul 2021 09:41:11 +0000 (12:41 +0300)]
pbutils: Expose functions for getting a file extension for caps and flags for describing the format of the caps
This information was available internally already but not available from
the outside.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-base/-/merge_requests/1221>
Sebastian Dröge [Wed, 4 Aug 2021 07:06:02 +0000 (10:06 +0300)]
playbin/uridecodebin: Emit source-setup signal early before doing the scheduling query
Some elements will require the source to be set up properly before the
scheduling query returns useful results, e.g. appsrc and giostreamsrc.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-base/-/merge_requests/1241>
Devarsh Thakkar [Thu, 10 Jun 2021 07:55:23 +0000 (00:55 -0700)]
ext: alsa: Fix fallback paths for setting buffer and period times
Below fallback paths were introduced in
https://github.com/GStreamer/gst-plugins-base/commit/
9759810d8206b5f1aa199f98599caec3630a1813
if setting period time after buffer time failed :
1) Set period time and then buffer time if it doesn't work
2) Set only buffer time
3) Set only period time
These all were not functioning properly since they were using old
copy of snd_pcm_hw_params_t which already had some fields set
as per previous try and this was causing issues as driver was
referring to that old value while trying to set them again in
fallback paths.
So now we always use the initial copy of snd_pcm_hw_params_t
for every fallback and same is also being done at
https://gitlab.freedesktop.org/pulseaudio/pulseaudio/-/commit/
557c4295107dc7374c850b0bd5331dd35e8fdd0f
Also we change the sequence to set period time earlier than
buffer time since period bytes being the smaller unit, most of the times
if underlying alsa device has a dependency then it is of period bytes
to be a multiple of some value (as per underlying DMA constraint)
and rest of the parameters like buffer bytes need to be adjusted
as per period bytes.
The same sequence is also followed in alsa-utils at
https://github.com/alsa-project/alsa-utils/commit/
9b621eeac4d55c4e881f093be5b163ca07d01b63
Fix 2) and 3) scenarios by returning success if the exclusive setting is passed
and not doing any further setting for buffer time or period time.
Add new fallback path of not setting any buffer time and period time
if all above fallback paths fail. The same is also being
followed at aforementioned pulseaudio commit.
In case of alsasink, remove the retry goto label, since it is not
required anymore as fallback paths take care of setting default
values if driver is not accepting any of the fallback paths.
Use separate label for exit to free params structs and return err
code. This also fixes leak in no_rate goto path in alsasink
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-base/-/merge_requests/1212>
Jakub Adam [Tue, 25 May 2021 19:16:48 +0000 (21:16 +0200)]
videoencoder: pass upstream HDR information through codec state
Don't copy HDR metadata from sink pad, because its caps may not have
been set yet if GstVideoEncoder::negotiate is called from
GstVideoEncoder::set_format, as e.g. vpx encoder does.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-base/-/merge_requests/1175>
Jakub Adam [Tue, 25 May 2021 19:15:53 +0000 (21:15 +0200)]
videoutils: add HDR metadata fields to GstVideoCodecState
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-base/-/merge_requests/1175>
Sebastian Dröge [Mon, 16 Aug 2021 07:19:07 +0000 (10:19 +0300)]
video-overlay-composition: Allow empty overlay compositions
Allowing to pass NULL to the constructor removes the need to
special-case the first rectangle in calling code and generally
simplifies application code.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-base/-/merge_requests/1256>
Per Förlin [Tue, 1 Jun 2021 13:27:31 +0000 (15:27 +0200)]
gstrtspconnection: Add support to ignore x-server header reply
When connecting to an RTSP server in tunnled mode (HTTP) the server
usually replies with a x-server header. This contains the address
of the intended streaming server. However some servers return an
"invalid" address. Here follows two examples when it might happen.
1. A server use Apache combined with a separate RTSP process to handle
Https request on port 443. In this case Apache handle TLS and
connects to the local RTSP server, which results in a local
address 127.0.0.1 or ::1 in the x-server reply. This address is
returned to the actual RTSP client in the x-server header.
The client will receive this address and try to connect to it
and fail.
2. The client use a ipv6 link local address with a specified scope id
fe80::aaaa:bbbb:cccc:dddd%eth0 and connects via Http on port 80.
The RTSP server receives the connection and returns the address
in the x-server header. The client will receive this address and
try to connect to it "as is" without the scope id and fail.
In the case of streaming data from RTSP servers like 1. and 2. it's
useful to have the option to simply ignore the x-server header reply
and continue using the original address.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-base/-/merge_requests/1192>
Nirbheek Chauhan [Fri, 13 Aug 2021 14:05:23 +0000 (19:35 +0530)]
sdp: Avoid using g_memdup() since it is deprecated
g_memdup() is deprecated since GLib 2.68 and we want to avoid
deprecation warnings with recent versions of GLib. Instead of using
g_memdup2(), we can simply use the new gst_buffer_new_memdup() added
in 1.19.x
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-base/-/merge_requests/1254>
Matthew Waters [Fri, 4 Jun 2021 08:32:07 +0000 (18:32 +1000)]
glbuffer: support persistent buffer mappings
Requires OpenGL 4.4 or EXT_buffer_storage
Current mesa exposes GL_ARB_buffer_storage when retrieving the relevant
functions returns no-ops and causes failures.
Improves throughput of uploads by roughly 30%-60% and download throughput by
roughly 10-30% across depending on the exact scenario and hardware.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-base/-/merge_requests/1191>
Seungha Yang [Fri, 30 Jul 2021 14:57:20 +0000 (23:57 +0900)]
examples: win32-videooverlay: Add support for testing gst_video_overlay_set_render_rectangle
Add keyboard handler to test gst_video_overlay_set_render_rectangle()
API for Windows video elements
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-base/-/merge_requests/1235>
Seungha Yang [Fri, 30 Jul 2021 14:04:57 +0000 (23:04 +0900)]
examples: win32-videooverlay: Use d3d11videosink by default
d3d11videosink was promoted to have primary rank and
it's recommended videosink element on Windows
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-base/-/merge_requests/1235>
Seungha Yang [Thu, 8 Jul 2021 08:47:28 +0000 (17:47 +0900)]
tests: appsink: Add reverse stepping test case
To demonstrate reverse stepping issue of
https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/848
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-base/-/merge_requests/1223>
Gilbok Lee [Fri, 30 Jul 2021 00:18:59 +0000 (00:18 +0000)]
Merge "subparse: Remove gst_event_unref when not support byte seek" into tizen
Gilbok Lee [Thu, 29 Jul 2021 23:45:57 +0000 (08:45 +0900)]
subparse: Remove gst_event_unref when not support byte seek
Change-Id: I5b1ab17ef331a4a712d02cbb6969dd5bbc09cd0e
Seungha Yang [Thu, 3 Jun 2021 10:15:22 +0000 (19:15 +0900)]
examples: win32-videooverlay: Add test option for threading scenario
Add an option to test the case where window thread and pipeline handling
thread are different. Mainly to test the HWND leak fixed by
https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/2302
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-base/-/merge_requests/1188>
Matthew Waters [Tue, 27 Jul 2021 07:44:02 +0000 (17:44 +1000)]
rtpbasedepayload: remove object locking an extension
Doing that is fraught with danger of deadlocks and is not conceptually
part of the API contract. The object lock is generally intended for
internal-object-use only.
If another lock is needed, that should be added separately.
This lock was erronously added as part of:
https://gitlab.freedesktop.org/gstreamer/gst-plugins-base/-/merge_requests/1118
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-base/-/merge_requests/1233>
Matthew Waters [Tue, 27 Jul 2021 03:30:56 +0000 (13:30 +1000)]
gldownload: use the GstGLSyncMeta in all cases
fixes qmlglsrc ! gldownload ! videoconvert in some cases.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-base/-/merge_requests/1232>
Nicolas Dufresne [Thu, 22 Jul 2021 21:11:26 +0000 (17:11 -0400)]
glcontext: egl: Stop comparing native surface pointer
This was noticed with wayland, sometimes the newly create native
handle can have the same pointer (even though its new). This lead
to unwanted errors or crash.
Fixes #927
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-base/-/merge_requests/1231>
Nicolas Dufresne [Thu, 22 Jul 2021 21:02:51 +0000 (17:02 -0400)]
glwindow: Add "window-handle-changed" signal
This allow other objects to clear any wrapper object that depends
on the previous handle, and properly re-create the new wrappers without
having to resort into doing pointer comparison.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-base/-/merge_requests/1231>
Nicolas Dufresne [Fri, 23 Jul 2021 17:39:34 +0000 (13:39 -0400)]
Revert "glwindow: wayland: Skip redoing surfaces if window haven't changed"
This reverts commit
aba6bd7822f4c0f572765bfaada76f454a594317.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-base/-/merge_requests/1231>
Guillaume Desmottes [Mon, 22 Feb 2021 12:17:18 +0000 (13:17 +0100)]
appsrc: serialize custom events with buffers flow
Application may want to inject events to the pipeline and keep them
synchronized with the buffers flow.
Fix #247
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-base/-/merge_requests/1046>
Guillaume Desmottes [Fri, 19 Feb 2021 12:32:48 +0000 (13:32 +0100)]
appsink: add API to catch events
There is currently no way for users to receive incoming events from
appsink while keeping them properly serialized with the buffers flow.
This can be especially useful when application is injecting custom
downstream events into the pipeline and needs to know when they reached
appsink.
Solving this by adding a new signal notifying about new incoming events
and a set of action signals and method to pull those events.
The API is actually pulling the samples and events all together as they
are actually fetched from the same queue.
Having a specific API to pull only events would have the side effect of
discarding samples (and pulling samples would discard events) making
this API not convenient for users.
Partially fix #247
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-base/-/merge_requests/1046>
Guillaume Desmottes [Fri, 19 Feb 2021 13:45:08 +0000 (14:45 +0100)]
appsink: factor out dequeue_object()
No semantic change, will be used to implement new event API.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-base/-/merge_requests/1046>
Nicolas Dufresne [Tue, 20 Jul 2021 13:37:58 +0000 (09:37 -0400)]
glwindow: wayland: Skip redoing surfaces if window haven't changed
The problem is that EGLNativeWindowSurface and wl_egl_surface are the
same object underneath, so we must recreate both together. As an
optimization, the EGLNativeWindowSurface wrapper is only re-created
if the window_handle changed.
On Mesa, this would cause crash, which will be fixed by:
https://gitlab.freedesktop.org/mesa/mesa/-/merge_requests/11979
And will lead to proper errors in the future or on other GL stack. This
issue was encounter using a permanent GstGLDisplay after cycling one of
multiple independent pipelines through NULL state.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-base/-/merge_requests/1230>
Nicolas Dufresne [Tue, 20 Jul 2021 13:36:22 +0000 (09:36 -0400)]
glwindow: wayland: Remove redundant create_surfaces call
The surfaces will be created in _roundtrip_async, so no need to call
this early. This should cause no functional difference.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-base/-/merge_requests/1230>
Hyunil [Wed, 21 Jul 2021 06:37:25 +0000 (15:37 +0900)]
GstVideoOverlay: Add gst_video_overlay_set_wl_window_exported_shell_handle
- new interface for a shell handle exported by wayland window for synchronization between UI and video
[Version] 1.16.2-16
[Issue Type] New feature
Change-Id: I04d6244062892712c34278343eb9926c25483c07
Signed-off-by: Hyunil <hyunil46.park@samsung.com>
Michael Olbrich [Fri, 11 Jun 2021 07:02:29 +0000 (09:02 +0200)]
decodebin3: improve decoder selection
Currently the decoder selection is very naive: The type with the highest
rank that matches the current caps is used. This works well for software
decoders. The exact supported caps are always known and the static caps are
defined accordingly.
With hardware decoders, e.g. vaapi, the situation is different. The decoder
may reject the caps later during a caps query. At that point, a new decoder
is created. However, the same type is chosen an after several tries,
decodebin fails.
To avoid this, do the caps query while adding the decoder and try again
with other decoder types if the query fails:
1. create the decoder from the next matching type
2. add and link the decoder
3. change the decoder state to READY
4. do the caps query
if it fails then remove the decoder again and go back to 1.
5. expose the source pad
6. sync the decoder state with the parent.
This way, the decoder is already part of the pipeline when the state change
to READY happens. So context handling should work as before.
Exposing the source pad after the query was successful is important:
Otherwise the thread from the decoder source pad may block in a blocked pad
downstream in the playsink waiting for other pads to be ready.
The thread now blocks trying to set the state back to NULL while holding
the SELECTION_LOCK. Other streams may block on the SELECTION_LOCK and the
playsink never unblocks the pad. The result is a deadlock.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-base/-/merge_requests/1201>
Hosang Lee [Wed, 17 Jun 2020 00:03:51 +0000 (09:03 +0900)]
subparse: lower text buffer threshold
It is possible for subtitle files to have a string length less than 30.
WebVTT for example may contain only the 'WEBVTT' string in the file
without any cues.
As an example in hls streams, since WEBVTT files can be segmented
like video/audio, some subtitle segments may only contain just the
header string.
Change-Id: I42dd0497852550d7cd6dd21a485856f06af55d71
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-base/-/merge_requests/708>
Mathieu Duponchelle [Fri, 20 Mar 2020 18:09:17 +0000 (19:09 +0100)]
subparse: convert from pango-markup to utf8 ..
when downstream requires it
Change-Id: I855f284401d8f5abbc4a1b1351f541c9883c92e4
Mathieu Duponchelle [Wed, 11 Mar 2020 00:01:34 +0000 (01:01 +0100)]
subparse: accept WebVTT timestamps without an hour component
https://www.w3.org/TR/webvtt1/#webvtt-timestamp
Change-Id: I5e87fdcda7f150c12fc08857ce04c378001321bd
mm:ss,000 is a valid WebVTT timestamp
Nicolas Dufresne [Wed, 14 Jul 2021 20:09:41 +0000 (16:09 -0400)]
gl: x11: Issue XSync to close our top level window
This is similar action as when the window handle is modified, we now issue
XSync whenever we destroy our internal window. This ensure that the window is
properly closed before the connecgtion is dropped.
Fixes #815
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-base/-/merge_requests/1226>
Nicolas Dufresne [Wed, 14 Jul 2021 15:43:10 +0000 (11:43 -0400)]
gl: wayland: Fix hinding the window on close()
When the window is called, we properly destroy all surfaces, which effectively
will unmap that surface and should make it disapear on screen, but we also
destroy the wl_source, a GSource that is resposibble of dispatching and executing
messages to/from the Wayland server.
As a side effect, the server never gets the message and the surfaces are
"leaked" on the server. We fix this using wl_display_flush() before destroying
the wl_source.
Fixes #815
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-base/-/merge_requests/1226>
Nicolas Dufresne [Thu, 15 Jul 2021 15:09:35 +0000 (11:09 -0400)]
tests: example: Add missing glx_dep when building sdlshare
Might be realted to some recent Mesa cleanup, but GLX is not longer visible
through libOpenGL, so add the missing deps now.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-base/-/merge_requests/1227>
Matthew Waters [Mon, 17 May 2021 02:04:50 +0000 (12:04 +1000)]
examples/qt/textureshare: add explicit dep on glx_dep
Fixes linking:
/usr/bin/ld: subprojects/gst-plugins-base/tests/examples/gl/qt/qglwtextureshare/qglwtextureshare.p/qglrenderer.cpp.o: undefined reference to symbol 'glXGetCurrentContext'
/usr/bin/ld: /usr/lib64/libGLX.so.0: error adding symbols: DSO missing from command line
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-base/-/merge_requests/1143>
Haelwenn (lanodan) Monnier [Fri, 14 May 2021 12:10:55 +0000 (14:10 +0200)]
gl: Try GLVND 'opengl' and 'glx' first
This fixes targetting desktop OpenGL without libGL.so
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-base/-/merge_requests/1143>
Sebastian Dröge [Mon, 12 Jul 2021 06:37:24 +0000 (09:37 +0300)]
audioaggregator: Only post QoS messages if the property is enabled
Previously one of the branches did not check for the property value. To
avoid this in the future, check inside the QoS calculation function
instead.
As a side effect this now always prints the debug messages into the logs
when samples are dropped, which is useful information even without the
QoS messages.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-base/-/merge_requests/1224>
Sebastian Dröge [Fri, 9 Jul 2021 06:49:15 +0000 (09:49 +0300)]
audioaggregator: Resync on the next buffer when dropping a buffer on discont resyncing
If a buffer is dropped during resyncing on a discont because either its
end offset is already before the current output offset of the
aggregator or because it fully overlaps with the part of the current
output buffer that was already filled, then don't just assume that the
next buffer is going to start at exactly the expected offset. It might
still require some more dropping of samples.
This caused the input to be mixed with an offset to its actual position
in the output stream, causing additional latency and wrong
synchronization between the different input streams.
Instead consider each buffer after a discont as a discont until the
aggregator actually resynced and starts mixing samples from the input
again.
Also update the start output offset of a new input buffer if samples
have to be dropped at the beginning. Otherwise it might be mixed too
early into the output and overwrite part of the output buffer that
already took samples from this input into account.
Fixes https://gitlab.freedesktop.org/gstreamer/gst-plugins-base/-/issues/912
which is a regression introduced by https://gitlab.freedesktop.org/gstreamer/gst-plugins-base/-/merge_requests/1180/
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-base/-/merge_requests/1224>
Olivier Crête [Wed, 26 May 2021 22:20:02 +0000 (18:20 -0400)]
audiomixer: Add test for QoS message posting
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-base/-/merge_requests/1209>
Olivier Crête [Wed, 26 May 2021 14:38:18 +0000 (10:38 -0400)]
audio aggregator: Post QoS message when dropping audio
Post a QoS message every time some audio samples are dropped.
Also print log messages to make it easier to debug
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-base/-/merge_requests/1209>
Olivier Crête [Tue, 25 May 2021 22:05:05 +0000 (18:05 -0400)]
audio aggregator: Count samples that are dropped or processed
Keep a count of samples that are dropped or processed as statistics
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-base/-/merge_requests/1209>
Olivier Crête [Fri, 21 May 2021 20:16:50 +0000 (16:16 -0400)]
audio aggregator: Add QoS property to pad
Add a property to emit a QoS message whenever any data is dropped.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-base/-/merge_requests/1209>
Olivier Crête [Fri, 21 May 2021 20:10:06 +0000 (16:10 -0400)]
audio aggregator: Rename property enum to match class name
Add "CONVERT" into the property enum as we're going to add an
enum specifically for the base pad.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-base/-/merge_requests/1209>
Stéphane Cerveau [Tue, 24 Sep 2019 15:14:10 +0000 (17:14 +0200)]
videodecoder: add API to receive subframes
A video decoder can now receive subframes and start decoding
instead of waiting for the full frame to be complete.
Subframe support will reduce latency as described in the
video encoder base class.
A unit test illustrating this API is available in
tests/check/libs/videodecoder.c.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-base/-/merge_requests/454>
Seungha Yang [Sat, 3 Jul 2021 10:36:06 +0000 (19:36 +0900)]
gl/context/wgl: Add missing NULL init
The value of uninitialized local variable is varying depending
on compiler and not guaranteed to be NULL initialized.
That results in pointing random address instead of expected function pointer.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-base/-/merge_requests/1222>
Francisco Javier Velázquez-García [Fri, 12 Mar 2021 12:55:38 +0000 (13:55 +0100)]
videotestsrc: Add SMPTE75 RP-219 color bars conformant
Implement 8-bit values of SMPTE RP 2019-1:2014. The bar widths and
heights are the result of fractions as integers. The remainders of
widths are distributed in a way that they match the values in Table
C.1 (a) in the specification.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-base/-/merge_requests/1063>
Jan Alexander Steffens (heftig) [Fri, 12 Mar 2021 19:58:40 +0000 (20:58 +0100)]
videotestsrc: Add a start parameter to _blend_line
Makes it easier to paint part of a line.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-base/-/merge_requests/1063>
Jan Alexander Steffens (heftig) [Fri, 12 Mar 2021 19:57:13 +0000 (20:57 +0100)]
videotestsrc: Keep tmpline unchanged in_convert_tmpline
This will allow us to repeatedly
call it to render subsequent lines.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-base/-/merge_requests/1063>
Gilbok Lee [Thu, 24 Jun 2021 23:41:16 +0000 (08:41 +0900)]
subparse: Send timestamp map custom event for HLS webvtt
Change-Id: I19efb35add97820252d4a0cb8bab60f9d58e5002
Michael de Gans [Tue, 15 Jun 2021 20:22:55 +0000 (13:22 -0700)]
appsink: fix incorrect return nullability
This commit fixes the annoations for return nullability on several
GstAppSink functions. This was causing bindings to be generated
incorrectly.
Fixes #914
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-base/-/merge_requests/1203>
Seungha Yang [Wed, 19 May 2021 07:22:46 +0000 (16:22 +0900)]
compositor: Add scaling policy to support PAR-aware scaling
Adding "sizing-policy" property for user to be able to specify
scaling policy (aspect-ratio for example).
At the moment, supported mode is only keep-aspect-ratio, but we might
be able to add more policies such as cropping, etc.
Fixes: https://gitlab.freedesktop.org/gstreamer/gst-plugins-base/-/issues/696
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-base/-/merge_requests/1156>
Seungha Yang [Wed, 19 May 2021 11:11:15 +0000 (20:11 +0900)]
video: Deprecate gst_video_sink_center_rect()
... and add gst_video_center_rect() method as a replacement.
The method is useful for outside of videosink subclasses as well
but the old naming might be able to mislead people.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-base/-/merge_requests/1156>
Jakub Adam [Mon, 24 May 2021 17:11:51 +0000 (19:11 +0200)]
rtpbasepayload: don't write empty extension header
When some header extensions are present but none decides to write any
data to the currently processed RTP buffer, remove the extension data
section.
Resulting RTP buffer wasn't formatted correctly.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-base/-/merge_requests/1173>
Jakub Adam [Mon, 24 May 2021 17:02:42 +0000 (19:02 +0200)]
rtpbuffer: Add gst_rtp_buffer_remove_extension_data()
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-base/-/merge_requests/1173>
Jakub Adam [Mon, 24 May 2021 17:01:24 +0000 (19:01 +0200)]
rtpbasepayload: map RTP buffer READWRITE when setting headers
GstRTPHeaderExtension::write can map the RTP buffer for reading. If that
happens on a buffer that is already mapped WRITE-only by the payloader,
the payloader's mapping gets invalidated (GstRTPBuffer::map will point
to a different instance of GstMemory).
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-base/-/merge_requests/1173>
Olivier Crête [Thu, 24 Jun 2021 18:56:11 +0000 (14:56 -0400)]
rtphdrext: Make all fields private
The presence of a method and a field with the same name confuses the C#
binding generator. As there are accessor functions for all the fields,
let's just make them private.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-base/-/merge_requests/1218>
Olivier Crête [Sat, 26 Jun 2021 16:50:58 +0000 (12:50 -0400)]
gst: don't use volatile to mean atomic
volatile is not sufficient to provide atomic guarantees and real atomics
should be used instead. GCC 11 has started warning about using volatile
with atomic operations.
https://gitlab.gnome.org/GNOME/glib/-/merge_requests/1719
Discovered in gst-plugins-good#868
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-base/-/merge_requests/1219>
Jan Schmidt [Fri, 25 Jun 2021 13:42:34 +0000 (23:42 +1000)]
video-converter: Set up matrix tables only once.
When configuring a multi-thread converter, only allocate the
shared colour conversion matrices once for the first thread,
to avoid allocating multiple times and leaking memory.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-base/-/merge_requests/1216>
Jan Alexander Steffens (heftig) [Thu, 6 May 2021 17:01:41 +0000 (19:01 +0200)]
video-converter: Set up gamma tables only once
When the video converter is using multiple threads, the gamma tables
were created multiple times, leaking the tables set up for the previous
thread.
Only calculate the tables once.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-base/-/merge_requests/1140>
Jan Alexander Steffens (heftig) [Thu, 6 May 2021 16:22:45 +0000 (18:22 +0200)]
audio-converter: Free config when gst_audio_converter_new fails
The config got leaked when parameter validation fails.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-base/-/merge_requests/1140>
Seungha Yang [Fri, 25 Jun 2021 06:24:21 +0000 (15:24 +0900)]
glprototypes: Add GST_GL_API_OPENGL to available version of sync
Make sync APIs usable if supported, even when GST_GL_API_OPENGL3 is
not selected
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-base/-/merge_requests/1215>
Per Förlin [Fri, 23 Apr 2021 16:03:20 +0000 (18:03 +0200)]
gstrtspconnection: Add IPv6 support for tunneled mode
An IPv6 address must be specified within [] brackets.
Add brackets for IPv6 address used for tunneled mode,
for non-tunneled this is already supported.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-base/-/merge_requests/1145>
Nicolas Dufresne [Wed, 16 Jun 2021 18:49:14 +0000 (14:49 -0400)]
videodecoder: Call drain() rather then finish() on segment-done
The finish() virtual function documentation state that "Sub-classes can refuse
to decode new data after." Though, it is very common to issue a non-flushing
seek after that event in gapless playback uses case. This fixes potential
stalls with code using segment seeks, by using drain() virtual funciton
instead.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-base/-/merge_requests/1206>
Matthew Waters [Tue, 8 Jun 2021 04:55:36 +0000 (14:55 +1000)]
oggdemux: fix a race in push mode when performing the duration seek
There may be two or more threads involved here however the important
interaction is the use of ogg->seeK_event_drop_till value that was only
set in the push-mode seek-event thread and could race with upstream
sending e.g. and EOS (or data).
Scenario is this:
1. oggdemux performs a seek to near the end of the file to try and find
the duration. ogg->push_state is set to PUSH_DURATION.
2. Seek is picked up by the dedicated seek event thread and sets
ogg->seek_event_drop_till to the seek event's seqnum.
3. Most operations are blocked or dropped waiting on the duration to
be determined and processing continues until a duration is found.
4. Two branching options for how this ultimately plays out
4a. The source is too fast and we receive an EOS event which is dropped
because ogg->push_state == PUSH_DURATION. In this case everything
works.
4b. We hit our 'almost at the end' check in
gst_ogg_pad_handle_push_mode_state() and attempt to seek back to the
beginning (or to a user-provided seek). This seek is marshalled to
the seek event thread without setting ogg->seek_event_drop_till but
with change ogg->push_state = PUSH_PLAYING. If an EOS event or
e.g. buffers arrive from upstream before the seek event thread has
picked up the seek event, then the EOS/data is processed as if it
came as a result of the seek event. This is the case that fails.
The fix is two-fold:
1. Preemptively set ogg->seek_event_drop_till when setting the seek
event so that data and other events can be dropped correctly.
2. In addition to dropping and EOS events while ogg->push_state ==
PUSH_DURATION, also drop any EOS events that are received before the
seek event has been processed by also tracking the seqnum of the seek.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-base/-/merge_requests/1196>
Sergei Kovalev [Mon, 21 Jun 2021 14:06:14 +0000 (14:06 +0000)]
audiobasesink: Fix of double lock release
Add missing "return;" which prevents double lock release.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-base/-/merge_requests/1208>