Sebastian Dröge [Sat, 15 Oct 2022 08:38:02 +0000 (11:38 +0300)]
validate: Add/fix various annotations
And fix monitor constructors to correctly check for valid arguments.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3194>
Sebastian Dröge [Fri, 14 Oct 2022 21:19:53 +0000 (00:19 +0300)]
pbutils: Add/fix various annotations
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3194>
Sebastian Dröge [Fri, 14 Oct 2022 21:00:03 +0000 (00:00 +0300)]
rtp: Add/fix various annotations
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3194>
Sebastian Dröge [Fri, 14 Oct 2022 20:52:40 +0000 (23:52 +0300)]
rtsp: Add/fix various annotations
Also initialize out parameters so that they don't have an undefined
value in case of errors.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3194>
Sebastian Dröge [Fri, 14 Oct 2022 20:48:09 +0000 (23:48 +0300)]
controller: Add/fix various annotations
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3194>
Sebastian Dröge [Fri, 14 Oct 2022 19:08:07 +0000 (22:08 +0300)]
app: Add/fix various annotations
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3194>
Sebastian Dröge [Fri, 14 Oct 2022 19:04:00 +0000 (22:04 +0300)]
audio: Add/fix various annotations
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3194>
Sebastian Dröge [Fri, 14 Oct 2022 18:59:43 +0000 (21:59 +0300)]
video: Add/fix various annotations
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3194>
Sebastian Dröge [Fri, 14 Oct 2022 18:51:17 +0000 (21:51 +0300)]
webrtc: Add/fix various annotations
And mark string parameters as const.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3194>
Edward Hervey [Mon, 17 Oct 2022 11:01:53 +0000 (13:01 +0200)]
urisourcebin: Fix usage of raw and non-raw source provider
The computation in analyze_source was wrong, and would state that the element
has "all raw source pads" if it had at least one.
Fixes https://gitlab.freedesktop.org/gstreamer/gstreamer/-/issues/1029
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3199>
Sebastian Dröge [Mon, 17 Oct 2022 15:38:43 +0000 (18:38 +0300)]
rtpbuffer: Initialize extended timestamp to the first wraparound period
This allows correct handling of wrapping around backwards during the
first wraparound period and avoids the infamous "Cannot unwrap, any
wrapping took place yet" error message.
It allows makes sure that for actual timestamp jumps a valid value is
returned instead of 0, which then allows the caller to handle it
properly. Not having this can have the caller see the same timestamp (0)
for a very long time, which for example can cause rtpjitterbuffer to
output the same timestamp for a very long time.
Fixes https://gitlab.freedesktop.org/gstreamer/gstreamer/-/issues/1500
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3202>
Matthew Waters [Tue, 18 Oct 2022 01:26:21 +0000 (12:26 +1100)]
examples/webrtc: don't use factory_make_full() for enums
They are not currently translated into their respective enum values and
will produce an error.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3210>
Seungha Yang [Sun, 16 Oct 2022 15:40:46 +0000 (00:40 +0900)]
wasapi2: Add support for process loopback capture
Adding loopback capture mode for specified PID.
Note that this feature requires Windows 10 build 20348
(Windows 11/Windows Server 2022 or later),
and any process loopback related properties will not be exposed
if OS does not support it.
Example launch lines:
* wasapi2src loopback-mode=include-process-tree loopback-target-pid=<PID>
Captures audio generated by an application (specified by PID)
and its child process
* wasapi2src loopback-mode=exclude-process-tree loopback-target-pid=<PID>
Captures desktop audio excluding PID and its child process
Fixes: https://gitlab.freedesktop.org/gstreamer/gstreamer/-/issues/1278
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3195>
Nirbheek Chauhan [Mon, 17 Oct 2022 20:54:48 +0000 (02:24 +0530)]
ci: Run windows jobs when win-* binary subprojects are updated
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3209>
Nirbheek Chauhan [Mon, 17 Oct 2022 20:51:07 +0000 (02:21 +0530)]
meson: Update flex, bison, and nasm
Latest flex is 2.6.4, bison is 3.8.2, nasm is 2.15.04
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3209>
Sam Van Den Berge [Mon, 17 Oct 2022 08:39:18 +0000 (10:39 +0200)]
examples/webrtc/signalling: Fix compatibility with Python 3.10
Fix asyncio throwing a deprecation warning when using
asyncio.get_event_loop().
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3196>
Arun Raghavan [Fri, 14 Oct 2022 20:21:07 +0000 (16:21 -0400)]
rtmp2sink: Correctly return GST_FLOW_ERROR on error
If there is an error while connecting, the streaming task will be stopped, and
is_running() will be false, causing a GST_FLOW_FLUSHING to be returned. Instead,
we perform the error check (!self->connection) first, to return an error if
that's what occured.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3189>
Edward Hervey [Fri, 14 Oct 2022 14:12:45 +0000 (16:12 +0200)]
gst-play: Don't leak the stream collection
We are given a reference to the collection when parsing it from the
message. Just store it (instead of referencing it again).
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3186>
Mathieu Duponchelle [Fri, 14 Oct 2022 16:51:43 +0000 (18:51 +0200)]
webrtc/nice: fix small leak of split strings
g_strfreev previously stopped at our manual NULL-termination. Fix by
restoring the pointer after joining.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3188>
Nirbheek Chauhan [Thu, 13 Oct 2022 20:40:35 +0000 (02:10 +0530)]
gst-docs: Fix typo in hotdoc kwarg
The hotdoc module passes unknown keyword arguments as arguments to
hotdoc, and the fatal warnings argument is --fatal-warnings.
https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2972#note_1586361
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3178>
Devin Anderson [Fri, 14 Oct 2022 01:23:04 +0000 (01:23 +0000)]
wavparse: Avoid occasional crash due to referencing freed buffer.
We've seen occasional crashes in the `wavparse` module associated with
referencing a buffer in `gst_wavparse_chain` that's already been freed. The
reference is stolen when the buffer is transferred to the adapter with
`gst_adapter_push` and, IIUC, assuming the source doesn't hold a reference to
the buffer, the buffer could be freed during interaction with the adapter in
`gst_wavparse_stream_headers`.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3179>
Edward Hervey [Fri, 14 Oct 2022 06:53:18 +0000 (08:53 +0200)]
rtsp-client: Remove duplicate documentation
Confuses the documentation builder, since it's documented twice it complains
about a missing "Since:" marker whereas it's present in the documentation
comment further down
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3180>
Julian Bouzas [Thu, 13 Oct 2022 15:21:42 +0000 (11:21 -0400)]
riff: Mark jpeg as parsed
This is needed so that autoplugging works with avidemux and JPEG decoders that
need parsed sink caps (eg rockchip 'mppjpegdec' decoder). It also works fine
with 'jpegdec' decoder regardless.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3175>
Devin Anderson [Thu, 13 Oct 2022 00:20:45 +0000 (00:20 +0000)]
wavparse: Fix crash that occurs in push mode when header chunks are corrupted
in certain ways.
In the case that a test is provided for, the size of the `fmt ` chunk is
changed from 16 bytes to 18 bytes (bytes 17 - 20 below):
```
$ hexdump -C corruptheadertestsrc.wav
00000000 52 49 46 46 e4 fd 00 00 57 41 56 45 66 6d 74 20 |RIFF....WAVEfmt |
00000010 12 00 00 00 01 00 01 00 80 3e 00 00 00 7d 00 00 |.........>...}..|
00000020 02 00 10 00 64 61 74 61 |....data|
00000028
```
(Note that the original file is much larger. This was the smallest sub-file
I could find that would generate the crash.)
Note that, while the same issue doesn't cause a crash in pull mode, there's a
different issue in that the file is processed successfully as if it was a .wav
file with zero samples.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3173>
He Junyan [Wed, 12 Oct 2022 08:11:47 +0000 (16:11 +0800)]
vp9parse: The show_existing_frame buffer should not be decode only.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3155>
He Junyan [Tue, 11 Oct 2022 08:17:26 +0000 (16:17 +0800)]
vp9parse: Correct the pts for frames inside a super frame.
When the alignment is "FRAME" and the parse is likely connecting to
a decoder, the current PTS setting for VP9 frames inside a super
frame is not very correct.
For example, the super frame may begin with non-displayed frames and
end with a displayed frame. The current way will assign the PTS to
the first non-displayed frame, which is a decode-only frame and the
PTS will be discarded in the video decoder. While the last displayed
frame has invalid PTS, and so the video decoder needs to guess its
PTS based on the frame rate and previous frame's PTS. This is not a
decent and robust way. And more important, when the previous frames
provide DTS, the video decoder will also guess the PTS based on the
previous frames' DTS and trigger the warning like:
gstvideodecoder.c:3147:gst_video_decoder_prepare_finish_frame: \
<vavp9dec0> decreasing timestame
It sets the reordered_output and makes the decoder in free run mode.
We should correct the PTS for a super frame, let the non-displayed
frames have no PTS while set the correct PTS to the displayed one.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3155>
Piotr Brzeziński [Fri, 7 Oct 2022 22:29:03 +0000 (00:29 +0200)]
avfvideosrc: Allow specifying crop coordinates during screen capture
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3153>
Linus Svensson [Fri, 19 Aug 2022 14:16:26 +0000 (16:16 +0200)]
rtsp-server: Free client if no connection could be created
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3164>
Edward Hervey [Wed, 12 Oct 2022 09:12:50 +0000 (11:12 +0200)]
mxfdemux: Add support for Canon XF-HEVC
Fixes https://gitlab.freedesktop.org/gstreamer/gstreamer/-/issues/1495
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3163>
Edward Hervey [Wed, 12 Oct 2022 09:12:14 +0000 (11:12 +0200)]
mxfdemux: Don't leak index table segments on failures
The segment was freed ... but not the contents :)
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3163>
Peter Stensson [Tue, 11 Oct 2022 12:55:48 +0000 (14:55 +0200)]
rtsp-server: Add since marker for adjust_error_code
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3157>
Xavier Claessens [Tue, 11 Oct 2022 18:53:40 +0000 (14:53 -0400)]
meson: Update libsoup.wrap
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3162>
Thibault Saunier [Fri, 7 Oct 2022 14:46:29 +0000 (11:46 -0300)]
videorate: Do not close segment when getting a same segment twice
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3059>
Thibault Saunier [Wed, 14 Sep 2022 19:39:48 +0000 (16:39 -0300)]
videorate: Handle closing segment on EOS right after caps event
The scenario is what we try in the tests:
- we have a segment with .stop set
- some frame(s) flow
- we get a CAPS event
- we get an EOS (before getting buffers after the CAPS event)
in that case, without that patch, the segment is not properly closed
which is not correct. In this patch we keep track of previous caps until
a new buffer arrives, this way in that situation we set previous caps
again, and close the segment with the previous buffer.
Fixes: https://gitlab.freedesktop.org/gstreamer/gstreamer/-/issues/1352
in this specific case
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3059>
Thibault Saunier [Wed, 21 Sep 2022 14:44:59 +0000 (11:44 -0300)]
validate: Plug some leaks
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3059>
Edward Hervey [Tue, 11 Oct 2022 13:00:37 +0000 (15:00 +0200)]
oss4: Fix debug category initialization
Fixes https://gitlab.freedesktop.org/gstreamer/gstreamer/-/issues/1456
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3158>
Sebastian Dröge [Tue, 11 Oct 2022 10:09:20 +0000 (13:09 +0300)]
webrtc: Move GST_WEBRTC_ERROR_TYPE_ERROR at the end of the enum to keep ABI compatibility
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3154>
Sangchul Lee [Thu, 6 Oct 2022 11:37:45 +0000 (20:37 +0900)]
tests/webrtc: Add test for 'add-turn-server' action signal
It just checks return value of the action signal.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3131>
Johan Sternerup [Thu, 10 Dec 2020 15:25:26 +0000 (16:25 +0100)]
sctpenc: Prohibit sending of interleaved message parts
Apparently we cannot start sending messages from another datachannel
before the previous message was completely sent. usrsctplib will
complain about being locked on another stream id and set
errno=EINVAL.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2454>
Nicolas Dufresne [Mon, 22 Aug 2022 20:33:23 +0000 (16:33 -0400)]
opengl: Fix usage of eglCreate/DestroyImage
The implementation was inconsistent between create and destroy. EGLImage
creation and destruction is requires for EGL 1.5 and up, while
otherwise the KHR version is only available if EGL_KHR_image_base
feature is set. Not doing these check may lead to getting a function
pointer to a stub, which is notably the case when using apitrace.
Fixes #1389
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2925>
Peter Stensson [Tue, 21 Jun 2022 07:51:55 +0000 (09:51 +0200)]
rtsp-server: Add support for adjusting request response on pipeline errors
The idea is to give the application the possibility to adjust the error
code when responding to a request. For that purpose the pipeline's bus
messages are emitted to subscribers through a signal handle-message.
The subscribers can then check those messages for errors and adjust
the response error code by overriding the virtual method
adjust_error_code().
Fixes #1294
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2972>
Mathieu Duponchelle [Fri, 7 Oct 2022 23:03:13 +0000 (01:03 +0200)]
splitmuxsrc: don't queue data on unlinked pads
Once a pad has returned NOT_LINKED, the part reader shouldn't let its
corresponding data queue run full and eventually (after 20 seconds)
stall playback.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3145>
Sebastian Dröge [Mon, 3 Oct 2022 17:28:47 +0000 (20:28 +0300)]
rtpsource: Don't do probation for RTX sources
Disable probation for RTX sources as packets will arrive very
irregularly and waiting for a second packet usually exceeds the deadline
of the retransmission.
Fixes https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/issues/181
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3112>
Sebastian Dröge [Mon, 3 Oct 2022 16:58:38 +0000 (19:58 +0300)]
rtp: examples: client-rtpaux: Provide correct caps by payload type and RTX pt map by session
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3112>
George Kiagiadakis [Fri, 25 Jan 2019 22:04:50 +0000 (17:04 -0500)]
tests/check/rtpsession: extend test_internal_sources_timeout
to verify that rtx SSRCs do not BYE after timeout
See https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/issues/360
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3112>
Sebastian Dröge [Mon, 3 Oct 2022 16:12:55 +0000 (19:12 +0300)]
rtpsession: Remember the corresponding media SSRC for RTX sources
This allows timing out the RTX source and sending BYE for it when the
actual media source belonging to it is timed out.
This change only applies to sending sources from this session.
Fixes https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/issues/360
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3112>
Sebastian Dröge [Mon, 3 Oct 2022 16:20:14 +0000 (19:20 +0300)]
rtpsource: Rename rtp_source_update_caps to rtp_source_update_send_caps
To make it clear that this is only used for sending RTP sources.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3112>
Sebastian Dröge [Mon, 3 Oct 2022 10:48:36 +0000 (13:48 +0300)]
rtpsession: Rename gst_rtp_session_sink_setcaps to gst_rtp_session_setcaps_recv_rtp
to make it clearer that this is for setting receiver caps and to make it
more consistent with gst_rtp_session_setcaps_send_rtp.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3112>
Aleksandr Slobodeniuk [Thu, 6 Oct 2022 14:23:56 +0000 (16:23 +0200)]
decodebin3: allow to call "dispose" multiple times
https://docs.gtk.org/gobject/concepts.html#reference-counts-and-cycles
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3135>
Wojciech Kapsa [Fri, 7 Oct 2022 06:30:05 +0000 (08:30 +0200)]
decklink: reset calculation of gst_decklink_video_src_update_time_mapping on no_signal. When the HDMI cable was disconnected for a long time, the calculation took too much time. SDI cable works fine.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3132>
Nirbheek Chauhan [Fri, 7 Oct 2022 09:52:36 +0000 (15:22 +0530)]
ci: Fix website regen on push
Don't make the integrate stage manual, we need it to regen the website
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3134>
Guillaume Desmottes [Fri, 7 Oct 2022 12:39:47 +0000 (14:39 +0200)]
aggregator: fix input buffering
We need to be able to buffer at least the aggregator latency +
upstream latency, which is the value used to compute the aggregator
deadline.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3138>
Xavier Claessens [Fri, 7 Oct 2022 11:56:21 +0000 (07:56 -0400)]
Meson: Fix libxml2 fallback
The variable xml2lib_dep does not exist. The correct name is already in
the wrap file.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3136>
Sebastian Dröge [Thu, 6 Oct 2022 12:02:22 +0000 (15:02 +0300)]
rtspsrc: Retry SETUP with non-compliant URL resolution on "Bad Request" and "Not found"
Various RTSP servers/cameras assume base and control URL to be simply
appended instead of being resolved according to the relative URL
resolution algorithm as mandated by the RTSP specification.
To work around this, try using such a non-compliant control URL if the
server didn't like the URL used in the first SETUP request.
Fixes https://gitlab.freedesktop.org/gstreamer/gstreamer/-/issues/1447
Fixes https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/issues/922
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3127>
Sangchul Lee [Thu, 22 Sep 2022 13:39:31 +0000 (22:39 +0900)]
webrtcbin: Fix pointer dereference before null check
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3129>
Wojciech Kapsa [Mon, 26 Sep 2022 11:57:15 +0000 (13:57 +0200)]
decklink: Add new persistent-id property and sort devices by persistent ID
The order of the devices iterator from the SDK is undefined and can
randomly change.
Keep the device-number property for backwards compatibility and
simplicity but prefer the persistent-id property and also use it for the
device provider implementation.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3078>
Aleksandr Slobodeniuk [Thu, 6 Oct 2022 11:03:26 +0000 (13:03 +0200)]
decodebin3: fix mutex leaks
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3126>
Thibault Saunier [Tue, 4 Oct 2022 22:19:57 +0000 (19:19 -0300)]
transcodebin: Implement support for upstream stream selection
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3128>
Thibault Saunier [Mon, 3 Oct 2022 18:20:53 +0000 (15:20 -0300)]
decodebin3: Do not try to plug a decoder on raw formats
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3123>
Seungha Yang [Wed, 5 Oct 2022 15:00:59 +0000 (00:00 +0900)]
videosink: Don't return unknown end-time from get_times()
... in case of reverse playback. Otherwise basesink will not
wait for clock
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3121>
Mathieu Duponchelle [Tue, 4 Oct 2022 19:27:36 +0000 (21:27 +0200)]
avauddec: address regression with WMA files ..
By outputting lead-in samples that FFmpeg now would like us to ignore,
and discarding trailing samples that it would now like us to output.
See https://gitlab.freedesktop.org/gstreamer/gstreamer/-/issues/1474
Fixes: https://gitlab.freedesktop.org/gstreamer/gstreamer/-/issues/1348
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3117>
Thibault Saunier [Wed, 5 Oct 2022 15:57:02 +0000 (12:57 -0300)]
validate:launcher: Cleanup test uuid when copying it
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3122>
Thibault Saunier [Tue, 4 Oct 2022 22:17:15 +0000 (19:17 -0300)]
validate: launcher: Add a argument to avoid rereuning flaky tests
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3122>
Thibault Saunier [Tue, 4 Oct 2022 22:16:44 +0000 (19:16 -0300)]
validate: launcher: Keep variable framerate from input when possible
But disable it if forcing a framerate for some reason
Fixing our support for variable framerate in the encoding profile
serialization format.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3122>
Thibault Saunier [Tue, 4 Oct 2022 22:16:04 +0000 (19:16 -0300)]
validate: launcher: Fix some pep8 issues
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3122>
Thibault Saunier [Tue, 4 Oct 2022 22:14:49 +0000 (19:14 -0300)]
validate: Handle testfiles that need an HTTP server
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3122>
Thibault Saunier [Tue, 27 Sep 2022 19:35:31 +0000 (16:35 -0300)]
validate:ssim: Show original/nok/diff images on error images
This way the user can easily see how off the image is very easily
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3122>
Thibault Saunier [Tue, 27 Sep 2022 15:51:33 +0000 (12:51 -0300)]
validate:scenario: Fix setting iterating variable from arrays
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3122>
Nirbheek Chauhan [Wed, 5 Oct 2022 00:08:54 +0000 (05:38 +0530)]
d3d11: Fix debug build on UWP
GstDXGIGetDebugInterface() is unused when targeting UWP. We directly
call DXGIGetDebugInterface1() in that case.
Fixes build failure:
../gst-libs/gst/d3d11/gstd3d11device.cpp(271): error C2440: '=': cannot convert from 'HRESULT (__cdecl *)(UINT,const IID &,void **)' to 'DXGIGetDebugInterface_t'
../gst-libs/gst/d3d11/gstd3d11device.cpp(271): note: This conversion requires a reinterpret_cast, a C-style cast or function-style cast
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3118>
Sangchul Lee [Wed, 5 Oct 2022 06:59:03 +0000 (15:59 +0900)]
webrtc/nice: Make sure to return NULL when validating turn server fails
It affects 'add-turn-server' signal action and 'turn-server' property
of webrtcbin.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3120>
Johan Sternerup [Mon, 21 Mar 2022 09:29:21 +0000 (10:29 +0100)]
webrtc: return error when sending on non-open datachannel
According to W3C
specification (https://w3c.github.io/webrtc-pc/#datachannel-send) we
should return InvalidStateError exception when trying to send when the
channel is not open. In the world of C/glib/gstreamer we don't have
exceptions but have to rely on gboolean/GError instead. Introducing
these calls for a change in function signature of the action signals
used to send data on the datachannel. Changing the signature of the
existing "send-string" and "send-data" signals would mean an immediate
breaking change so instead we deprecate them. Furthermore, there is no
way to express GError** as an argument to an action signal in a way
that fits language bindings (pointer-to-pointer simply does not work)
and we have to use regular functions instead.
Therefore we introduce gst_webrtc_data_channel_send_data_full() and
gst_webrtc_data_channel_send_string_full() while deprecating the old
functions and corresponding signals.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1958>
Haihao Xiang [Fri, 2 Jul 2021 08:22:04 +0000 (16:22 +0800)]
eglimage: fix Y412_LE DMABuf import support
This fixed the error below:
GST_GL_PLATFORM=egl gst-launch-1.0 videotestsrc ! msdkvpp !
"video/x-raw(memory:DMABuf),format=Y412_LE" ! glimagesink
0:00:00.
063737691 576978 0x557c11266920 ERROR gleglimage
gsteglimage.c:577:_drm_rgba_fourcc_from_info: Unsupported format for
DMABuf.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1389>
Haihao Xiang [Fri, 2 Jul 2021 08:01:19 +0000 (16:01 +0800)]
eglimage: add Y410 DMABuf import support
This fixed the error below:
GST_GL_PLATFORM=egl gst-launch-1.0 videotestsrc ! msdkvpp ! \
"video/x-raw(memory:DMABuf),format=Y410" ! glimagesink
0:00:00.
040553883 576732 0x55be9b35f920 ERROR gleglimage
gsteglimage.c:577:_drm_rgba_fourcc_from_info: Unsupported format for
DMABuf.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1389>
Haihao Xiang [Fri, 2 Jul 2021 12:31:22 +0000 (20:31 +0800)]
eglimage: add Y212_LE / Y212_BE DMABuf import support
This fixed the error below:
GST_GL_PLATFORM=egl gst-launch-1.0 videotestsrc ! msdkvpp \
! "video/x-raw(memory:DMABuf),format=Y212_LE" ! glimagesink
0:00:00.
148086281 575148 0x562a6d36d920 ERROR gleglimage
gsteglimage.c:555:_drm_rgba_fourcc_from_info: Unsupported format for
DMABuf.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1389>
Haihao Xiang [Fri, 2 Jul 2021 07:37:19 +0000 (15:37 +0800)]
eglimage: add Y210 DMABuf import support
This fixed the error below:
GST_GL_PLATFORM=egl gst-launch-1.0 videotestsrc ! msdkvpp ! \
"video/x-raw(memory:DMABuf),format=Y210" ! glimagesink
0:00:00.
039881627 571365 0x56245eba6920 ERROR gleglimage
gsteglimage.c:549:_drm_rgba_fourcc_from_info: Unsupported format for
DMABuf
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1389>
Jan Schmidt [Fri, 30 Sep 2022 18:58:04 +0000 (04:58 +1000)]
gstbin: Use g_queue_clear_full()
Use g_queue_clear_full() to release the child list
instead of iterating over the list twice.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3101>
Jan Schmidt [Fri, 30 Sep 2022 18:55:49 +0000 (04:55 +1000)]
gstbin: Fix a potential leak in gst_bin_do_deep_add_remove()
If a child element of a bin is unparented while
gst_bin_do_deep_add_remove() is iterating the children,
don't leak a ref to it.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3101>
Jan Schmidt [Fri, 30 Sep 2022 18:54:20 +0000 (04:54 +1000)]
gstiterator: Don't use gst_object_unref() for a GObject
The owner of a GstIterator is a plain GObject. Don't unref it
with gst_object_unref(), or it will be logged in tracer
info incorrectly.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3101>
Stéphane Cerveau [Fri, 23 Sep 2022 15:18:35 +0000 (17:18 +0200)]
h265parse: fix typo in member of GstH265SPS
Rename sps_extnsion_params to sps_extension_params
Fix comment about vui_parameters_present_flag
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3072>
Tim-Philipp Müller [Tue, 4 Oct 2022 02:57:31 +0000 (03:57 +0100)]
Back to development
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3115>
Tim-Philipp Müller [Tue, 4 Oct 2022 00:14:01 +0000 (01:14 +0100)]
Release 1.21.1
Tim-Philipp Müller [Tue, 4 Oct 2022 00:13:59 +0000 (01:13 +0100)]
Update ChangeLogs for 1.21.1
Jan Schmidt [Fri, 30 Sep 2022 16:33:49 +0000 (02:33 +1000)]
playsink: Hold a reference to the soft volume element
Always hold a reference to the soft volume element
provided by the playsinkaudioconvert bin helper, the
same as when volume is provided by a sink element,
or the soft volume element gets unreffed too soon.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3108>
Jan Schmidt [Fri, 30 Sep 2022 18:51:21 +0000 (04:51 +1000)]
xvimagesink: Don't leak XvShmImage
If allocating an XvShmImage generates an X error, but
still returns some allocated memory, make sure to free it
so it doesn't leak.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3100>
Jan Schmidt [Fri, 30 Sep 2022 18:49:16 +0000 (04:49 +1000)]
xvimagesink: Zero initialize mask array
Clear the stack array before setting bits to pass to
XISelectEvents(). Fixes spurious crash from an X error:
X Error of failed request: BadValue (integer parameter out of range for operation)
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3100>
Jan Schmidt [Fri, 30 Sep 2022 18:47:31 +0000 (04:47 +1000)]
xvimagesink: Don't leak temporary
Use a stack variable instead of a temporarily
malloced variable that wasn't being freed
properly. Fixes a small leak of a GstXvTouchDevice
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3100>
Jan Schmidt [Fri, 30 Sep 2022 18:41:06 +0000 (04:41 +1000)]
decodebin3: Make sure event is writable before modifying
Make sure we're operating on a private copy of an event when
modifying it.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3107>
Jan Schmidt [Fri, 30 Sep 2022 18:40:09 +0000 (04:40 +1000)]
urisourcebin: Make sure event is writable before modifying.
Make sure we're operating on a private copy of an event before
modifying it.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3107>
Jan Schmidt [Fri, 30 Sep 2022 18:37:05 +0000 (04:37 +1000)]
decodebin3: Don't lose a ref on EOS event
Make sure not to give away the ref on the final EOS
event for which the probe handler is returning GST_PAD_PROBE_REMOVE
when pushing the event manually.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3107>
Edward Hervey [Mon, 3 Oct 2022 09:16:25 +0000 (11:16 +0200)]
queue2: Hold the lock when modifying sinkresult
As it's done elsewhere. Avoids a potential race of the field being modified in
the meantime.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3110>
Edward Hervey [Wed, 27 Jul 2022 09:19:50 +0000 (11:19 +0200)]
qtdemux: Don't stop task when resetting
This is a regression that was introduced in
cca2f555d14b5751f7f9d466b66127544dad5138 (yes, 9 years ago).
The only place where a demuxer streaming thread should be stopped is when the
sinkpad is deactivated from pull mode (i.e. PAUSED->READY).
Attempting to stop the task in this function would cause this to happen when a
FLUSH_STOP or STREAM_START event is received... which can cause deadlocks.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3109>
Tim-Philipp Müller [Thu, 29 Sep 2022 11:42:21 +0000 (12:42 +0100)]
avauddec: fix unnecessary reconfiguration if the audio layout isn't specified
It would constantly want to renegotiate (and spam the debug log) even
though the channel layout hasn't actually changed. We use the same
fallback in gst_ffmpegauddec_negotiate() already.
This happens with WMA files for example.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3103>
Mathieu Duponchelle [Fri, 30 Sep 2022 16:57:01 +0000 (18:57 +0200)]
splitmuxsrc: don't consider unlinked pads when deactivating part
If splitmuxsrc exposes multiple pads, but only one is linked, part pads
will never see an EOS event. This shouldn't prevent the part from being
eventually deactivated.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3099>
Tim-Philipp Müller [Thu, 29 Sep 2022 23:10:25 +0000 (00:10 +0100)]
pbutils: descriptions: add meta/x-klv
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3094>
Tim-Philipp Müller [Thu, 29 Sep 2022 23:07:07 +0000 (00:07 +0100)]
pbutils: descriptions: fix gst_pb_utils_get_caps_description_flags()
And add a little unit test.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3094>
James Cowgill [Thu, 29 Sep 2022 08:39:15 +0000 (09:39 +0100)]
xmptag: Call gst_tag_register_musicbrainz_tags during init
We need to call this to register the MusixBrainz tags before we use
them in an XMP schema.
Fixes this critical when attempting to run jpegparse on a JPEG
containing MusicBrainz XMP tags:
GStreamer-CRITICAL **: 20:41:07.885: gst_tag_get_type: assertion 'info != NULL' failed
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3092>
Sebastian Dröge [Thu, 29 Sep 2022 11:36:38 +0000 (14:36 +0300)]
test-effect-switch: Fix some memory leaks and make effect element ownership clearer
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3091>
Sebastian Dröge [Thu, 29 Sep 2022 11:34:31 +0000 (14:34 +0300)]
gaussblur: Fix memory leak when setting caps multiple times
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3091>
Christopher Obbard [Tue, 27 Sep 2022 17:17:24 +0000 (18:17 +0100)]
v4l2codecs: Warn user when no media devices are found
Currently if the user is not able to access the devices under /dev/media*,
either due to no media devices present on the system or simply no permission
to access the device, v4l2codecs initialises with no features or debug messages.
Since calling `GST_DEBUG="v4l2*:7" gst-inspect-1.0 v4l2codecs` is a typical way
to diagnose why element(s) failed to enumerate, we should be more verbose here
when the user is not able to access any /dev/media* device. So print a simple
debug message in this case to aid debugging.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3088>
Philipp Zabel [Wed, 28 Mar 2018 15:54:15 +0000 (17:54 +0200)]
buffer: drop parent meta in deep copy/foreach_metadata
The purpose of a deep buffer copy is to be able to release the source
buffer and all its dependencies. Attaching the parent buffer meta to
the newly created deep copy needlessly keeps holding a reference to the
parent buffer.
The issue this solves is the fact you need to allocate more
buffers, as you have free buffers being held for no reason. In the good
cases it will use more memory, in the bad case it will stall your
pipeline (since codecs often need a minimum number of buffers to
actually work).
Fixes #283
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2928>