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Author SHA1 Message Date
Thibault Saunier
2217dc11ee Merging gst-plugins-good 2021-09-24 16:13:37 -03:00
Thibault Saunier
2fd28195ca Move files from gst-plugins-base into the "subprojects/gst-plugins-base/" subdir 2021-09-24 16:13:26 -03:00
Thibault Saunier
d2822d09ea Merging gst-plugins-base 2021-09-24 16:13:17 -03:00
Thibault Saunier
6c364d9626 Move files from gstreamer into the "subprojects/gstreamer/" subdir 2021-09-24 16:13:07 -03:00
Tim-Philipp Müller
20bbeb5e37 Release 1.19.2 2021-09-23 01:33:41 +01:00
Tim-Philipp Müller
ce937bcb21 Release 1.19.2 2021-09-23 01:33:09 +01:00
Tim-Philipp Müller
b4ca58df76 Release 1.19.2 2021-09-23 01:32:33 +01:00
Tim-Philipp Müller
2d23bccf47 rtph263pdepay: flag keyframes on output buffers 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/1091>
 2021-09-22 14:03:57 +01:00
Sebastian Dröge
55ca21edae clocksync: Add some debug output to the clock waiting code 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/841>
 2021-09-22 12:04:44 +00:00
Tim-Philipp Müller
cc50672d29 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>
 2021-09-21 22:39:46 +01:00
Nicolas Dufresne
3607c617a5 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>
 2021-09-20 18:06:44 +00:00
Nicolas Dufresne
7b76c97de9 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>
 2021-09-20 18:06:44 +00:00
Sebastian Dröge
fea7f02a1d 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>
 2021-09-20 13:12:12 +03:00
Sebastian Dröge
f3601164d2 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>
 2021-09-20 13:06:38 +03:00
He Junyan
9289df4291 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>
 2021-09-19 22:43:11 +08:00
He Junyan
25dbae6e73 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>
 2021-09-19 22:43:06 +08:00
He Junyan
81cf9754e3 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>
 2021-09-19 20:41:59 +08:00
Havard Graff
e0811f890f 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>
 2021-09-16 17:00:14 +03:00
Olivier Crête
972184f434 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>
 2021-09-15 17:02:01 +00:00
Olivier Crête
f8f24a2619 rtphdrext: Update write() API to return a signed value 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/1059>
 2021-09-15 17:02:01 +00:00
Olivier Crête
fd93c1ac19 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>
 2021-09-15 16:35:09 +00:00
Olivier Crête
98f2a84a28 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>
 2021-09-15 15:35:43 +00:00
Olivier Crête
24fd80344d videorate: Drop incoming buffers that are outside of the segment 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-base/-/merge_requests/767>
 2021-09-15 15:35:43 +00:00
Olivier Crête
6f7922b4db 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>
 2021-09-15 15:35:43 +00:00
Olivier Crête
a76f38b2c7 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>
 2021-09-15 15:35:43 +00:00
Olivier Crête
75b4809ebc 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>
 2021-09-15 15:35:43 +00:00
Seungha Yang
4576abde67 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>
 2021-09-14 17:55:13 +09:00
Mathieu Duponchelle
c6acee201e 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>
 2021-09-13 14:16:06 +00:00
Mathieu Duponchelle
67eb70bb9c 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>
 2021-09-13 14:16:06 +00:00
Vivienne Watermeier
44bfc00884 flv: fix seqnum handling for seeks 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/1078>
 2021-09-13 12:30:30 +00:00
Matthew Waters
f441c72e5a 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>
 2021-09-13 09:42:15 +00:00
Sebastian Dröge
3592bf7726 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>
 2021-09-13 10:05:18 +03:00
Philippe Normand
732b352df6 docs: Update cache 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/1081>
 2021-09-12 12:18:32 +01:00
Philippe Normand
a55dafe341 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>
 2021-09-12 10:20:05 +01:00
Matthew Waters
f6aea043f9 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>
 2021-09-09 07:29:37 +00:00
Tobias Ronge
3ec9795a28 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>
 2021-09-09 06:45:04 +00:00
Andika Triwidada
b6147e6037 add missing space 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/884>
 2021-09-09 04:08:22 +00:00
Ludvig Rappe
92338e3d80 pbutils: Add mjpg to MIME codecs 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-base/-/merge_requests/1270>
 2021-09-07 14:49:52 +00:00
Seungha Yang
2c69544d0c 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>
 2021-09-07 12:15:34 +00:00
Jan Schmidt
5eba408071 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>
 2021-09-06 01:43:57 +10:00
Mathieu Duponchelle
f5cdb2d002 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>
 2021-09-03 13:46:19 +00:00
Philippe Normand
8ad1ea0297 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>
 2021-09-02 17:29:51 +00:00
Sebastian Dröge
fedd6c2a28 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>
 2021-09-02 12:07:52 +00:00
Philippe Normand
ab6cb4c2c7 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>
 2021-08-31 11:15:24 +00:00
Matthew Waters
e43bbaf3d9 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>
 2021-08-31 06:09:47 +00:00
Olivier Crête
aa3d2c3369 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>
 2021-08-30 17:01:15 +00:00
Olivier Crête
f70ccd6d86 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>
 2021-08-30 17:01:15 +00:00
Olivier Crête
23d07f3c7b 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>
 2021-08-30 17:01:15 +00:00
Olivier Crête
9ff052d5be 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>
 2021-08-30 17:01:15 +00:00
Ludvig Rappe
75c44583ee pbutils: Add function to convert caps to MIME codec 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-base/-/merge_requests/1265>
 2021-08-30 08:49:33 +00:00