Gilbok Lee [Tue, 25 Jan 2022 04:22:57 +0000 (13:22 +0900)]
Merge branch 'move_subdir_good' into tizen_gst_1.19.2_mono
Change-Id: I0cbd209497c9504ce21bc63b98d792a227f85ab8
Gilbok Lee [Tue, 25 Jan 2022 04:22:44 +0000 (13:22 +0900)]
Merge remote-tracking branch 'gst-plugins-good/tizen_gst_1.19.2' into tizen_gst_1.19.2_mono
Change-Id: Ic075d639fe00c24a6534f07f396f02f6fbbdbb0e
Gilbok Lee [Tue, 25 Jan 2022 04:22:30 +0000 (13:22 +0900)]
Merge branch 'move_subdir_base' into tizen_gst_1.19.2_mono
Change-Id: I86f250af954c85db7a92face2c7e79f53673787f
Gilbok Lee [Tue, 25 Jan 2022 04:21:51 +0000 (13:21 +0900)]
Merge remote-tracking branch 'gst-plugins-base/tizen_gst_1.19.2' into tizen_gst_1.19.2_mono
Change-Id: I8019d28ceb5b9b25c316f777dce7787fda344556
Gilbok Lee [Tue, 25 Jan 2022 04:21:26 +0000 (13:21 +0900)]
Merge branch 'move_subdir' into tizen_gst_1.19.2_mono
Change-Id: I6712b0965ca4ce5eb157ac1c26fd33b419498818
Hyunil [Mon, 24 Jan 2022 06:37:54 +0000 (15:37 +0900)]
Rtsp: Set start position to Range general-header for PLAY, RESUME and seek for Player
Change-Id: Iec86b75ce50981eb843d306cfffe52b498df9506
Signed-off-by: Hyunil <hyunil46.park@samsung.com>
Sangchul Lee [Mon, 24 Jan 2022 06:33:25 +0000 (15:33 +0900)]
gstinfo: Change definition name for enabling dlog
It is changed according to the meson.build option for Tizen.
Change-Id: I36b0cdbf76b3eab88bf271ca03965093923167c2
Signed-off-by: Sangchul Lee <sc11.lee@samsung.com>
Gilbok Lee [Mon, 17 Jan 2022 02:07:07 +0000 (11:07 +0900)]
Send seek event to baseparse when aacparse seek failed in push mode
Change-Id: Ifcbb545a3d68c110a5442db216ec23ead9a9ca26
Gilbok Lee [Thu, 20 Jan 2022 06:26:12 +0000 (15:26 +0900)]
Move the defines from spec to meson.build
Change-Id: Icd30589c3f5318806b4f3708abc75ee0398b3a94
Gilbok Lee [Thu, 20 Jan 2022 05:30:54 +0000 (14:30 +0900)]
Move the defines from spec to meson.build
- Change file permission 755 to 644
- Remove white space
Change-Id: I4d27826eed80c5698dd30f0b989429af36cf4e9f
Gilbok Lee [Thu, 20 Jan 2022 01:15:56 +0000 (10:15 +0900)]
Move the defines from spec to meson.build
Change-Id: Ic7483b9b364186652edb075b6681a6992185bde2
Eunhye Choi [Mon, 17 Jan 2022 18:37:57 +0000 (03:37 +0900)]
Merge branch 'tizen' into tizen_gst_1.19.2
Change-Id: I1eb8ccf3e811415df3c7828ebafc22d5dabe6a7f
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
Eunhye Choi [Thu, 13 Jan 2022 14:09:08 +0000 (23:09 +0900)]
fix tv profile option bug
Change-Id: Ibdb6db6df54241508910f4345d434534f9719f22
Eunhye Choi [Wed, 12 Jan 2022 20:57:38 +0000 (05:57 +0900)]
apply meson option and fix build error
- apply meson option
- fix packaging error
- .gbs.conf will be applied after
Change-Id: Ic53c05c010f88e43500fbd09334f0d12f1a38aa5
Gilbok Lee [Wed, 12 Jan 2022 05:31:00 +0000 (14:31 +0900)]
Merge remote-tracking branch 'upstream/master' into tizen
Change-Id: If3cf3c18b851741dcaec1b5a01796b7ba12242bc
Inki Dae [Fri, 17 Dec 2021 02:57:50 +0000 (11:57 +0900)]
ext/cairo: pack gstcairo plugin in default
Native applications which use GStreamer API need gst cairo plugin
library to draw graphic primitives on Cairo surface. So pack gst cario
plugin library in default.
Change-Id: I152eb1407448794892942d9c77ebb1dce4f7d78a
Signed-off-by: Inki Dae <inki.dae@samsung.com>
Doug Nazar [Fri, 23 Apr 2021 16:12:58 +0000 (12:12 -0400)]
Use g_memdup2() where available and add fallback for older GLib versions
glib 2.68 deprecates g_memdup(). Replace with g_memdup2() and
add fallback if compiling against older versions, since we
want to avoid deprecation warnings.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/803>
Change-Id: I1adcb816a0cc48003200b46c08d2d58985a79b59
Signed-off-by: Jeongmo Yang <jm80.yang@samsung.com>
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
Sangchul Lee [Mon, 22 Nov 2021 04:49:39 +0000 (13:49 +0900)]
pulsesink: Revise condition to set mute in the initial stage
If gst_pulseringbuffer_acquire() is called after _release() due
to any reason (e.g. caps changes), mute was not applied properly
with the current value. It is now fixed.
Note that the condition is slightly changed from upstream codes
especially on the mute_set variable.
Change-Id: I7b81160d12f30fbf1e872212b051adafeb2c50aa
Signed-off-by: Sangchul Lee <sc11.lee@samsung.com>
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
Gilbok Lee [Wed, 3 Nov 2021 06:30:32 +0000 (15:30 +0900)]
qtdemux: Determine duration with reference to track header duration
- The maximum value of the duration of each tkhd is decided with the total duration.
Change-Id: I8a88cd63ed58ce6677b70943d71ee5df8bfa2013
Haesu Gwon [Mon, 1 Nov 2021 08:28:54 +0000 (17:28 +0900)]
[effectv] Enable effectv for Media Editing FW
Change-Id: I8f0920bb96f3b93eb60f61052a54d5a0c80414ea
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
Thibault Saunier [Fri, 24 Sep 2021 19:13:50 +0000 (16:13 -0300)]
Move files from gst-plugins-good into the "subprojects/gst-plugins-good/" subdir
Thibault Saunier [Fri, 24 Sep 2021 19:13:37 +0000 (16:13 -0300)]
Merging gst-plugins-good
Thibault Saunier [Fri, 24 Sep 2021 19:13:26 +0000 (16:13 -0300)]
Move files from gst-plugins-base into the "subprojects/gst-plugins-base/" subdir
Thibault Saunier [Fri, 24 Sep 2021 19:13:17 +0000 (16:13 -0300)]
Merging gst-plugins-base
Thibault Saunier [Fri, 24 Sep 2021 19:13:07 +0000 (16:13 -0300)]
Move files from gstreamer into the "subprojects/gstreamer/" subdir
Tim-Philipp Müller [Thu, 23 Sep 2021 00:33:39 +0000 (01:33 +0100)]
Release 1.19.2
Tim-Philipp Müller [Thu, 23 Sep 2021 00:33:08 +0000 (01:33 +0100)]
Release 1.19.2
Tim-Philipp Müller [Thu, 23 Sep 2021 00:32:32 +0000 (01:32 +0100)]
Release 1.19.2
Tim-Philipp Müller [Wed, 22 Sep 2021 13:03:57 +0000 (14:03 +0100)]
rtph263pdepay: flag keyframes on output buffers
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/1091>
Sebastian Dröge [Wed, 23 Jun 2021 13:41:20 +0000 (16:41 +0300)]
clocksync: Add some debug output to the clock waiting code
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/841>
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 [Mon, 20 Sep 2021 10:12:12 +0000 (13:12 +0300)]
gst: Initialize optional event/message fields when parsing
These might not exist inside the structure and then we would potentially
keep around uninitialized memory from the caller in the out parameter.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/887>
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>
He Junyan [Sun, 19 Sep 2021 13:01:21 +0000 (21:01 +0800)]
test: bitwriter: Add a test for reset_and_get_data when not byte unaligned.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/886>
He Junyan [Sun, 19 Sep 2021 14:39:09 +0000 (22:39 +0800)]
bitwriter: Fix a memory leak in reset_and_get_buffer.
We should record the ownership of the data before we reset the bitwriter.
Or we will always dup the buffer data and leak the memory.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/886>
He Junyan [Sat, 18 Sep 2021 16:19:43 +0000 (00:19 +0800)]
bitwriter: Fix the trailing bits lost when getting its data.
In reset_and_get_data and reset_and_get_buffer, it fails to include
the trailing bits less than 8. So, when the bit_size is not byte
aligned, the trailing bits are lost in the return buffer.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/886>
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 23:47:40 +0000 (19:47 -0400)]
rtphdrhext-twcc: Return failure on map failure
This feels like exactly like a case that should fail.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/1059>
Olivier Crête [Wed, 18 Aug 2021 23:46:25 +0000 (19:46 -0400)]
rtphdrext: Update write() API to return a signed value
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/1059>
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>
Seungha Yang [Tue, 14 Sep 2021 08:26:27 +0000 (17:26 +0900)]
qtdemux: Try to build AAC codec-data whenever it's possible
AAC codec_data is a just collection of AAC profile, samplerate, and
channels. We can know samplerate and channels from parsed
SampleEntry data. Although the AAC profile is unknown there,
let's assume it as AAC-LC like we've been doing for the version 1
atom.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/1082>
Mathieu Duponchelle [Thu, 9 Sep 2021 23:43:18 +0000 (01:43 +0200)]
multiqueue: fix obsolete comment re initial flow status
The initial single queue srcresult is OK, it hasn't been
NOT_LINKED since 2007.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/885>
Mathieu Duponchelle [Thu, 9 Sep 2021 18:25:25 +0000 (20:25 +0200)]
multiqueue: never consider a queue that is not waiting
.. when computing the high id.
After a flush for instance, sq->srcresult is reset to OK,
yet it doesn't make sense to pick a non-existing position
id as the high id when a queue doesn't contain any items
in that situation either.
It is in any case completely OK to let the not-linked stream
get consumed without throttling at this stage, as any
first packet arriving on other single queues will get assigned
a higher position id.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/885>
Vivienne Watermeier [Tue, 7 Sep 2021 20:23:01 +0000 (22:23 +0200)]
flv: fix seqnum handling for seeks
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/1078>
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>
Matthew Waters [Mon, 18 Jan 2021 05:06:27 +0000 (16:06 +1100)]
isomp4: also allow muxing different h264/5 profiles/levels/etc
All of that is advertised through the codec_data itself so can change
just fine within isomp4.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/1071>
Sebastian Dröge [Sat, 11 Sep 2021 06:24:35 +0000 (09:24 +0300)]
matroska: Add support for muxing/demuxing ffv1
Previously only demuxing when stored via the RIFF/AVI mapping was
supported.
See https://github.com/FFmpeg/FFV1/blob/master/ffv1.md#matroska-file-format
Fixes https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/issues/923
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/1080>
Philippe Normand [Sun, 12 Sep 2021 11:18:32 +0000 (12:18 +0100)]
docs: Update cache
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/1081>
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>
Andika Triwidada [Thu, 9 Sep 2021 04:08:22 +0000 (04:08 +0000)]
add missing space
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/884>
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>
Seungha Yang [Tue, 3 Aug 2021 10:12:11 +0000 (19:12 +0900)]
jpegdec: Fix crash when interlaced field height is not DCT block size aligned
In case of interlaced JPEG file, we are doubling stride.
The scratch scan line should take account of it as well.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/1042>
Jan Schmidt [Sun, 5 Sep 2021 15:43:57 +0000 (01:43 +1000)]
multiqueue: Use running time of gap events for wakeups.
Use gap events to update the next_time of a queue the same
as buffers or segment events. Fixes problems where a group
consisting only of sparse streams primarily driven by
gap events would stall with a full multiqueue because
unlinked streams in the group were not being woken to
push data.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/879>
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>
Sebastian Dröge [Thu, 2 Sep 2021 05:38:54 +0000 (08:38 +0300)]
avidemux: Also detect 0x000001 as H264 byte-stream start code in codec_data
This works around some AVI files storing byte-stream data in the
codec_data. The previous workaround was only checking for
0x00000001 (4 bytes) instead of 0x000001 (3 bytes).
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/1072>
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>
Philippe Normand [Tue, 31 Aug 2021 10:05:16 +0000 (11:05 +0100)]
qt: Fix build for Qt 5.9
The QQuickItem::size() method was introduced in 5.10, so use direct width() and
height() access instead.
Fixes #908
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/1069>
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>
Olivier Crête [Thu, 19 Aug 2021 14:32:27 +0000 (10:32 -0400)]
rtphdrext-rfc6464: Add test for inserting in payloader using the API
This makes it clearer how to use the plugin in an API driven application.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/1058>
Olivier Crête [Wed, 18 Aug 2021 23:36:07 +0000 (19:36 -0400)]
rtphdrext-rfc6464: Put max level if the audio is beyond it
Otherwise, it just fails to add the extension, which makes no
sense. And our level element produces levels higher than 127 in some
cases.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/1058>
Olivier Crête [Wed, 18 Aug 2021 23:35:36 +0000 (19:35 -0400)]
rtphdrext-rfc6464: Add example pipeline
This makes it a bit easier to understand how to use it in an application.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/1058>
Olivier Crête [Wed, 18 Aug 2021 23:07:18 +0000 (19:07 -0400)]
rtphdrext-rfc6464: Add test for inserting it based on caps
Tests adding the extension based on the caps.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/1058>
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>
Edward Hervey [Fri, 27 Aug 2021 12:32:45 +0000 (14:32 +0200)]
qtdemux: Force stream-start push when re-using EOS'd streams
When re-using streams, we *do* need to push a `stream-start` event downstream if
we previously were EOS'd. Failure to do that would never remove the EOS status
on all downstream elements and cause weird issues.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/1067>
Brad Smith [Fri, 27 Aug 2021 06:05:45 +0000 (02:05 -0400)]
deinterlace: Use proper ASM output format for *BSD OS
FreeBSD/NetBSD/OpenBSD amd64 use the ELF binary format.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/1066>
Matthew Waters [Fri, 27 Aug 2021 03:51:07 +0000 (13:51 +1000)]
element: NULL the lists of contexts in dispose()
If dispose() is called more than once, we may double unref the list of
GstContext's.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/875>
Matthew Waters [Fri, 27 Aug 2021 03:30:57 +0000 (13:30 +1000)]
qmlgl: don't critical on input events before input format has been set
Accessing the unset GstVideoInfo would result in criticals
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/1065>
Sebastian Dröge [Wed, 25 Aug 2021 08:53:58 +0000 (11:53 +0300)]
docs: Add `Since` marker to "twcc-feedback-interval" property
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/927>
Havard Graff [Wed, 25 Aug 2021 08:33:24 +0000 (10:33 +0200)]
docs: update with "twcc-feedback-interval"
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/927>
Tulio Beloqui [Tue, 13 Apr 2021 14:19:22 +0000 (16:19 +0200)]
rtptwcc: changes to use rtp buffer arrival time and current time.
For TWCC we are more interested to track the arrival time (receive side)
and the current time (sender side) of the buffers rather than the
running time.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/927>
Knut Inge Hvidsten [Fri, 26 Mar 2021 10:57:42 +0000 (11:57 +0100)]
rtptwcc: add payloadtype to RTPTWCCPacket
The consumer of the stats can then separate between different media-types,
and do individual stats for each of them.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/927>
Havard Graff [Fri, 19 Mar 2021 17:19:43 +0000 (18:19 +0100)]
rtptwcc: make enabling TWCC sticky
Meaning that if a caps comes along that does NOT have TWCC in it,
this does not turn of TWCC for the rest, as this is in fact
completely allowed. (To have some payload-types not containing TWCC
seqnums).
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/927>
Havard Graff [Tue, 23 Feb 2021 08:44:05 +0000 (09:44 +0100)]
rtptwcc: move TWCC-logic over to the TWCC-manager
Prevent cluttering up the rtpsession, and keeping things localized.
Also write TWCC-seqnums for *all* streams in the session if configured by
caps.
A while back WebRTC was not doing TWCC for audio, basically breaking the
whole idea of a "transport-wide seqnuencenumber" applying for all bundled
streams. However, they have since fixed this, and now it no longers
makes sense to be able to single out certain payloadtypes for
use with TWCC, rather just including them all.
This also makes using RTX, RED, FEC etc much simpler, as it will apply
to them all as they enter the rtpsession.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/927>
Havard Graff [Tue, 23 Feb 2021 08:50:04 +0000 (09:50 +0100)]
rtptwcc: fix warning
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/927>
Tulio Beloqui [Thu, 11 Feb 2021 14:17:16 +0000 (15:17 +0100)]
rtptwcc: fixes and optimizations around run-length chunks
Co-authored-by: Havard Graff <havard.graff@gmail.com>
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/927>
Havard Graff [Fri, 18 Dec 2020 13:01:23 +0000 (14:01 +0100)]
rtptwcc: fix seqnum-wrap
Using the proper API to do this is obviously an improvement, and
adding a test for the case of a packet-loss when the seqnum wrap
is also a good idea.
Co-authored-by: Tulio Beloqui <tulio.beloqui@pexip.com>
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/927>
Tulio Beloqui [Fri, 18 Dec 2020 12:06:35 +0000 (13:06 +0100)]
rtptwcc: fixed feedback packet count overflow that allowed late
packets to be processed
Co-authored-by: Havard Graff <havard.graff@gmail.com>
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/927>
Tulio Beloqui [Wed, 16 Dec 2020 15:31:18 +0000 (16:31 +0100)]
rtptwcc: fixed parsing of old sequence number
Co-authored-by: Havard Graff <havard.graff@gmail.com>
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/927>
Tulio Beloqui [Wed, 16 Dec 2020 15:16:09 +0000 (16:16 +0100)]
rtptwcc: fixed guint8 overflow of feedback packet count
Co-authored-by: Havard Graff <havard.graff@gmail.com>
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/927>
Havard Graff [Thu, 19 Nov 2020 22:50:23 +0000 (23:50 +0100)]
rtptwcc: add feedback-interval
To allow RTCP TWCC reports to be scheduled on a timer instead of per
marker-bit.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/927>
Havard Graff [Fri, 20 Aug 2021 09:54:01 +0000 (11:54 +0200)]
rtptwcc: remove _set_send_packet_ts
Not in use.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/927>
Havard Graff [Mon, 16 Nov 2020 23:45:02 +0000 (00:45 +0100)]
rtptwcc: make twcc-tests more deterministic
They were a bit racy.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/927>
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>