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Author SHA1 Message Date
Olivier Crête
27808444ea webrtc janus rust: Update extra dependencies 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1332>
 2021-11-10 16:13:38 +00:00
Olivier Crête
5ab09323cd webrtc multiparty rust: Upgrade all other deps 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1332>
 2021-11-10 16:13:38 +00:00
Olivier Crête
c592f75cdd webrtc sendrecv rust: Upgrade all other deps 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1332>
 2021-11-10 16:13:38 +00:00
Olivier Crête
26e8624c9b webrtc multiparty rust: Port to bindings 0.17 version 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1332>
 2021-11-10 16:13:38 +00:00
Olivier Crête
93032b2ecc webrtc sendrecv rust: Port to bindings 0.17 version 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1332>
 2021-11-10 16:13:38 +00:00
Jiri Uncovsky
9abac91c96 glcontext/egl: add missing unref 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1328>
 2021-11-10 15:27:45 +00:00
Matthew Waters
71dd47516c rtpbin: separate out the two fec decoder locations 
The pipeline flow for receiving looks like this:

rtpsession ! rtpssrcdemux ! session_fec_decoder ! rtpjitterbuffer ! \
  rtpptdemux ! stream_fec_decoder ! ...

There are two places where a fec decoder could be placed.
1. As requested from the 'request-fec-decoder' signal: after rtpptdemux
   for each ssrc/pt produced
2. after rtpssrcdemux but before rtpjitterbuffer: added for the
   rtpst2022-1-fecenc/dec elements,

However, there was some cross-contamination of the elements involved and
the request-fec-decoder signal was also being used to request the fec
decoder for the session_fec_decoder which would then be cached and
re-used for subsequent fec decoder requests.  This would cause the same
element to be attempted to be linked to multiple elements in different
places in the pipeline.  This would fail and cause all kinds of havoc
usually resulting in a not-linked error being returned upstream and an
error message being posted by the source.

Fix by not using the request-fec-decoder signal for requesting the
session_fec_decoder and instead solely rely on the added properties for
that case.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1300>
 2021-11-10 10:38:26 +00:00
Jean Felder
bd91286a3b id3tag: Map GST_TAG_MUSICBRAINZ_RELEASETRACKID 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1331>
 2021-11-10 01:33:33 +00:00
Jean Felder
aaf72b9ff4 id3tag: Map GST_TAG_MUSICBRAINZ_RELEASEGROUPID 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1331>
 2021-11-10 01:33:33 +00:00
Jean Felder
b1c74609e8 id3tag: Remove trailing whitespace 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1331>
 2021-11-10 01:33:33 +00:00
Zhao, Gang
6cad2a7150 qtdemux: Fix can not demux Opus track made by qtmux 
Opus stream info is read from dOps box [1]. The offset of dOps box in Opus box is different in mp4a version 1 and 0 [2]. Calculate the offset of dOps box according to mp4a version.

[1] https://opus-codec.org/docs/opus_in_isobmff.html

[2] subprojects/gst-plugins-good/gst/isomp4/atoms.c:sample_entry_mp4a_copy_data:2146

Fixed: https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/issues/918
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1329>
 2021-11-09 17:57:49 +00:00
Ralf Sippl
0c9d9d90d9 docs: app-dev: events: seeking: use CLOCK_TIME_NONE instead of -1 and fix parameter names 
to match the parameter names in the gst_element_seek() declaration.

Closes https://gitlab.freedesktop.org/gstreamer/gst-docs/-/merge_requests/34/

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1324>
 2021-11-09 17:46:31 +00:00
wuchang li
7ac662f19d docs: installing-on-macos: flesh out instructions what to download 
Closes https://gitlab.freedesktop.org/gstreamer/gst-docs/-/merge_requests/106/

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1324>
 2021-11-09 17:46:31 +00:00
Tyler Compton
ebb61c5e24 plugin-development: basics-boilerplate: Remove unneeded meson.build edit step 
Closes https://gitlab.freedesktop.org/gstreamer/gst-docs/-/merge_requests/157/

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1324>
 2021-11-09 17:46:31 +00:00
Teh Yule Kim
c66b10b5cb docs: installing-on-windows: mention packages to download 
Closes https://gitlab.freedesktop.org/gstreamer/gst-docs/-/merge_requests/158/

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1324>
 2021-11-09 17:46:31 +00:00
Teh Yule Kim
bda72282dd docs: tutorials: add link to Rust version of the tutorials 
Closes https://gitlab.freedesktop.org/gstreamer/gst-docs/-/merge_requests/159/

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1324>
 2021-11-09 17:46:31 +00:00
fjmax
5c9fe1dc3c docs: tutorials: playback-3: flesh out build instructions 
Add information about how to compile this file. The code in this
tutorial also requires `gstreamer-audio-1.0`, so we cannot use
the commands from the previous tutorial.

Closes https://gitlab.freedesktop.org/gstreamer/gst-docs/-/merge_requests/164/

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1324>
 2021-11-09 17:46:31 +00:00
wngecn
bcff6abafd docs: plugin-dev: basics-boilerplate: fix typo in variable name 
Closes https://gitlab.freedesktop.org/gstreamer/gst-docs/-/merge_requests/166/

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1324>
 2021-11-09 17:46:31 +00:00
Timo Wischer
36006c61e9 avtpsrc: Use correct size for provided buffers 
Without this patch the following pipeline would send packets containing
garbage in the data section.
$ gst-launch-1.0 avtpsrc ! avtpsink

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1077>
 2021-11-09 16:59:10 +00:00
Guillaume Desmottes
9b809d4cc3 appsrc: log when segment changes 
We were logging when it does not change but not when it does, which is
confusing when reading logs.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1327>
 2021-11-09 16:19:05 +00:00
Timo Wischer
de95d3a1c4 avtp: crfsync: Warn when CRF package not yet received 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1075>
 2021-11-09 15:36:25 +01:00
Haihua Hu
a66124a79c v4l2bufferpool: set video alignment of video meta 
need apply video alignment info on video meta, downstream
element can get buffer alignment from video meta

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1326>
 2021-11-09 13:32:46 +00:00
Timo Wischer
214691b972 test: avtp: crf: Check for rounding errors 
on average period calculation.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1073>
 2021-11-09 10:59:00 +00:00
Timo Wischer
5a25eb61b7 avtp: crf: Use double for average period calculation 
to also support CRF intervals like every 1,333,333ns 64 events

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1073>
 2021-11-09 10:59:00 +00:00
Timo Wischer
6a576938ac tests: avtp: crf: Test for timestamp_interval > 1 
in case of CRF AVTPDUs with single CRF timestamp.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1076>
 2021-11-09 09:07:01 +01:00
Timo Wischer
5a9e9895ab avtp: crf: Properly handling one timestamp per PDU 
The average_period should always represent the time between two
events. The specification defines the event time as the time
between audio samples, video frame sync, video line sync, etc.
In case of one timestamp per PDU the timestamp_interval identifies
the amount of events between the timestamp of one PDU and the
timestamp of the next PDU.
As described in IEEE 1722-2016 chapter
"10.4.12 timestamp_interval field" timestamp_interval shall be
nonzero.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1076>
 2021-11-09 09:07:01 +01:00
Thomas Klausner
49449dfa41 shm: NetBSD build fix 
shm_unlink() and friends live in librt on NetBSD. Adapt build system.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1323>
 2021-11-08 21:14:42 +00:00
Sebastian Dröge
79c4275ed4 bin: Don't check twice for adding a bin to itself or removing it from itself 
This is already covered by a `g_return_val_if_fail()` in the calling
function.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1320>
 2021-11-08 20:31:44 +00:00
Sebastian Dröge
ddfd778c19 bin: Switch g_warning()s to GST_WARNING_OBJECT()s when adding/removing an element to a bin fails 
The failure conditions can be overidden by subclasses, and a boolean
return value is provided to the caller whether adding/removing the child
element has actually worked. The caller can then handle this
accordingly but flooding stderr with this is not very useful.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1320>
 2021-11-08 20:31:44 +00:00
Zhao, Gang
5ac1a12927 vpxdec: Skip check of key frame when open GOP 
Valid stream [1] might has no key frame.

Fixed: https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/issues/890

[1] https://storage.googleapis.com/downloads.webmproject.org/test_data/libvpx/vp90-2-16-intra-only.webm

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1316>
 2021-11-08 19:42:47 +00:00
Tim-Philipp Müller
fb4be45e6f gst-integration-tests: medias: update git module to latest commit 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1184>
 2021-11-08 18:59:58 +00:00
Joe Todd
a0df14f36b osxaudio: Increase max channels to 64 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1059>
 2021-11-08 18:09:52 +00:00
Nicolas Dufresne
d4bcff7dab gstinfo: Fix leak in generate_backtrace_trace 
Spotted by Laurent Pinchart.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/901>
 2021-11-08 17:22:06 +00:00
Martin Reboredo
2546cef4be aom: Set fixed_qp_offsets to a deactivated value 
aom only uses fixed_qp_offsets with the
Constant Quality (Q) Rate Control mode,
previously this was locking any usage
with another Rate Control mode.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1198>
 2021-11-08 16:42:17 +00:00
Thibault Saunier
a2460a8adc meson:ges: Fix typo adding ges gir to libs list 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1322>
 2021-11-08 11:10:11 -03:00
Nirbheek Chauhan
557d385d6c gst-ptp-helper: Do not disable multicast loopback 
Otherwise we cannot run gst-ptp-helper if the PTP master is on the
same device.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1306>
 2021-11-08 07:27:08 +00:00
Thibault Saunier
9413fb7ea2 tests:ges: Take TIMEOUT_FACTOR into account 
This env var is set in the CI so we grow the timeout as required
(when running in valgrind for example).

See https://gitlab.freedesktop.org/gstreamer/gstreamer/-/jobs/15449209

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1315>
 2021-11-05 17:00:25 +00:00
Antonio Ospite
7deee582d2 magicleap: update lumin_rt libraries names to the latest official version 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1314>
 2021-11-05 16:21:34 +00:00
He Junyan
0345188aaa codecs: h264dec: Calculate the latency by its bump mode. 
The current latency calculation just uses the num_reorder_frames,
which is not very precise. We should consider the bump mode of the
DPB, the faster it bumps, the lower latency we will have.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1046>
 2021-11-05 14:40:42 +00:00
Thibault Saunier
d0c5778669 doc: Update nvdec documentation 
Fixes https://gitlab.freedesktop.org/gstreamer/gstreamer/-/issues/870

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1309>
 2021-11-05 13:26:33 +00:00
Sanchayan Maity
3f49b21b80 imagefreeze: Only set caps if they do not match current caps 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1311>
 2021-11-05 15:34:47 +05:30
Seungha Yang
21d7bd09ce h264parse: Don't insert extra AUD if exists in bitstream already 
AUD nalu in packetized format is completely valid and therefore we should not
assume that we should insert AUD for packetized -> bytestream
conversion.

Fixes: https://gitlab.freedesktop.org/gstreamer/gstreamer/-/issues/862
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1296>
 2021-11-04 16:32:55 +00:00
Seungha Yang
aa1d97b0fb tests: h264parse: Add test for AUD insertion 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1296>
 2021-11-04 16:32:54 +00:00
Sebastian Dröge
d2f66106a8 v4l2: Support bools for setting extra-controls 
They're just mapped to 0 / 1 and can already be set as integers, but
being able to set them as booleans seems useful.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1308>
 2021-11-04 15:21:40 +00:00
Seungha Yang
8eee442715 wasapi2ringbuffer: Fix client object leak 
Check whether ringbuffer is holding client object already since
open_device() may be called multiple times

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1307>
 2021-11-04 12:41:32 +00:00
Sebastian Dröge
f9a97efbe1 webrtcbin: Clear errors from finding codec preferences before the next iteration 
The media is just skipped and the error is not propagated to the caller,
so keeping it around here would cause assertions a bit later when trying
to set a new error over the old one.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1291>
 2021-11-04 10:51:15 +00:00
Sebastian Dröge
30153f1591 webrtcbin: Move addition of attributes to the caps after making sure they're not empty or any 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1291>
 2021-11-04 10:51:15 +00:00
Sebastian Dröge
d628ccf0e5 webrtcbin: Don't require fixed caps when querying caps for a transceiver pad to match it with a media 
Upstream caps might for example be
  application/x-rtp,media=audio,encoding-name={OPUS, X-GST-OPUS-DRAFT-SPITTKA-00, multiopus}
and while that is not fixed caps it is enough to match it with a media.

Only caps structures that have the correct structure name and that have
the media and encoding-name field are preserved, but if both are present
then these caps are used as "codec preferences".

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1291>
 2021-11-04 10:51:15 +00:00
Marcin Kolny
a46ab2ced2 typefind: fix reading file extension from URI 
Currently reading extension relies on the fact that everything after the
last"." character is a file extension. Whereas that works fine for most
of the cases, it breaks when the URI contains a query part.

E.g.: `http://url.com/file.mp4?param=value` returns `mp4?param=value`
instead of `mp4`.

In this commit we use URI parser to read the path of the URI (in the example
above, that is `/file.mp4`) and read extension from that path.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1305>
 2021-11-04 09:30:31 +00:00
Tim-Philipp Müller
a7b376011b Back to development 2021-11-03 19:31:23 +00:00