Commit graph

64 commits

Author SHA1 Message Date
Philippe Normand f0e6959bba webrtcdatachannel: Notify buffered-amount property updates
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1484>
2022-01-02 10:18:35 +00:00
Philippe Normand 43856a0735 webrtcstats: Fix null pointer dereference
If there is no jitterbuffer stats we should not attempt to store them in the
global stats structure.

Also add a g_return_if_fail in _gst_structure_take_structure() about this
because it is a programmer error to pass an invalid pointer address there.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1479>
2021-12-29 15:55:57 +00:00
Olivier Crête 818a185b5d webrtcstats: Fall back to last packet ssrc if caps dont provide it
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1448>
2021-12-23 23:48:17 -05:00
Olivier Crête 4e32d6bf3e webrtcstats: Use our own caps instead of the sticky event
The sticky event seems to get cleared sometimes.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1448>
2021-12-23 23:48:17 -05:00
Olivier Crête 29befed685 webrtcbin: Store the ssrc of the last received packet
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1448>
2021-12-23 23:48:17 -05:00
Olivier Crête fc7e7f5ccc webrtc stats: Remove duplicate structure get
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1448>
2021-12-23 23:48:17 -05:00
Olivier Crête f35435f1f7 webrtc stats: Add more details about codecs into the stats
This makes the output a little closer to what the upstream stats are.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1448>
2021-12-23 23:48:17 -05:00
Seungha Yang 796007f75d av1enc: Update for newly designed AV1 profile signalling
Accept named AV1 profiles (i.e., main, high, and professional)
as well

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1456>
2021-12-21 22:20:34 +09:00
Mathieu Duponchelle abd61732bf webrtcbin: bind transceiver's fec-percentage to encoder percentage
Allows for dynamic control of the applied FEC overhead

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1429>
2021-12-14 17:34:53 +00:00
Mathieu Duponchelle 06893b8b5e webrtcbin: fix ulpfec / red for the BUNDLE case
* Add fec / red encoders as direct children of webrtcbin, instead
  of providing them to rtpbin through the request-fec-encoder signal.

  That is because they need to be placed before the rtpfunnel, which
  is placed upstream of rtpbin.

* Update configuration of red decoders to set a list of RED payloads
  on them, instead of setting the pt property.

  That is because there may be one RED pt per media in the same session.

* Connect to request-fec-decoder-full instead of request-fec-decoder,
  in order to instantiate FEC decoders according to the payload type
  of the stream.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1429>
2021-12-14 17:34:53 +00:00
Thibault Saunier d82efb47aa pitch: Specify layout as required for negotiation
There are cases where it might negotiate 'non-interleaved' while it
is wrong.

```
gst-launch-1.0 audiotestsrc !  "audio/x-raw, format=(string)F32LE, layout=(string)non-interleaved" ! audioconvert ! audioresample ! pitch tempo=1.2 ! audioconvert ! "audio/x-raw,format=S16LE" ! fakesink

Setting pipeline to PAUSED ...
Pipeline is PREROLLING ...
(gst-launch-1.0:3029628): GStreamer-Audio-CRITICAL **: 11:42:22.477: gst_audio_buffer_map: assertion '(!meta && info->layout == GST_AUDIO_LAYOUT_INTERLEAVED) || (meta && info->layout == meta->info.layout)' failed
ERROR: from element /GstPipeline:pipeline0/GstAudioConvert:audioconvert1: The stream is in the wrong format.
Additional debug info:
../subprojects/gst-plugins-base/gst/audioconvert/gstaudioconvert.c(876): gst_audio_convert_transform (): /GstPipeline:pipeline0/GstAudioConvert:audioconvert1:
failed to map input buffer
ERROR: pipeline doesn't want to preroll.
ERROR: from element /GstPipeline:pipeline0/GstAudioTestSrc:audiotestsrc0: Internal data stream error.
Setting pipeline to NULL ...
Additional debug info:
../subprojects/gstreamer/libs/gst/base/gstbasesrc.c(3127): gst_base_src_loop (): /GstPipeline:pipeline0/GstAudioTestSrc:audiotestsrc0:
streaming stopped, reason error (-5)
ERROR: pipeline doesn't want to preroll.
Freeing pipeline ...
```

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1441>
2021-12-11 19:09:09 -03:00
Philippe Normand 86719e25a4 wpevideosrc: Use basesrc event vfunc
Allows for basic default handling from the base class.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1422>
2021-12-07 11:43:26 +00:00
Tim-Philipp Müller 26169cee0e teletextdec: fix minor string leak
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1416>
2021-12-06 13:07:37 +00:00
Mathieu Duponchelle e90859f4d8 webrtcbin: deduplicate extmaps
When an extmap is defined twice for the same ID, firefox complains and
errors out (chrome is smart enough to accept strict duplicates).

To work around this, we deduplicate extmap attributes, and also error
out when a different extmap is defined for the same ID.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1383>
2021-11-25 18:38:22 +00:00
Seungha Yang 2a17618dcc openjpegenc: Fix build warning
Compiling C object subprojects/gst-plugins-bad/ext/openjpeg/gstopenjpeg.dll.p/gstopenjpegenc.c.obj
../subprojects/gst-plugins-bad/ext/openjpeg/gstopenjpegenc.c(416):
  warning C4133: '=': incompatible types - from 'GstFlowReturn (__cdecl *)(GstVideoEncoder *,GstVideoCodecFrame *)' to
  'gboolean (__cdecl *)(GstVideoEncoder *,GstVideoCodecFrame *)'

../subprojects/gst-plugins-bad/ext/openjpeg/gstopenjpegenc.c(418):
  warning C4133: '=': incompatible types - from 'GstFlowReturn (__cdecl *)(GstVideoEncoder *,GstVideoCodecFrame *)' to
  'gboolean (__cdecl *)(GstVideoEncoder *,GstVideoCodecFrame *)'

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1378>
2021-11-24 13:11:23 +00:00
Guillaume Desmottes d67a63a298 gssink: add metadata property
This property can be used to set metadata on the storage object.

Similar API has been added to the S3 sink already, see
https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/613

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1377>
2021-11-23 16:00:53 +01:00
Philippe Normand a6fd767025 wpevideosrc: Fix frame stuttering in GL rendering path
Make sure the EGLImage we're rendering to the GL memory stays alive long enough,
until the the GL memory has been destroyed.

This change fixes tearing and black flashes artefacts that were happening
because the EGLImage was sometimes destroyed before the sink actually rendered
the associated texture.

Fixes #889

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1354>
2021-11-16 21:55:41 +00:00
Philippe Normand 053dd564a1 wpevideosrc: Run through gst-indent
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1354>
2021-11-16 21:55:41 +00:00
Tim-Philipp Müller 972615cf22 docs: fix unnecessary ampersand, < and > escaping in code blocks
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1340>
2021-11-12 11:39:19 +00:00
Timo Wischer 8e7ce64a6e avtp: crf: Process also local CRF streams
Without this patch locally generated CRF streams will be ignored.
Therefore the same network interface could not be CRF talker and
CRF listener.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1074>
2021-11-10 16:53:04 +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
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
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 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
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
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
Tim-Philipp Müller 1f560af76b dtls: don't use deprecated g_binding_get_source() with newer GLib versions
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1279>
2021-10-30 00:52:42 +01:00
Heiko Becker b83e85ab67 neon: Allow building against neon 0.32.x
No API/ABI changes: https://github.com/notroj/neon/blob/0.32.0/NEWS#L3

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1267>
2021-10-29 00:14:53 +00:00
Mathieu Duponchelle 303c8025c6 webrtcbin: fix check_negotiation computing on caps event
It seems logical that check_negotiation be true if received_caps
is *not* equal to the new caps.

Also clean up handling of transceivers' ssrc events, as this
patch triggered a leaky code path.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1233>
2021-10-28 19:05:59 +00:00
Mathieu Duponchelle be0b5c54fd webrtcbin: connect input stream when receiving caps
.. if a current direction has already been set

When `webrtcbin` has created an offer based on codec_preferences,
it might not have received caps on its sinkpads by the time a
remote description is set, in which case we want to connect the
input stream upon actual reception of the caps instead.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1233>
2021-10-28 19:05:59 +00:00
Mathieu Duponchelle a9506f20d3 webrtcbin: consider pads with trans->codec_preferences ready
.. when determining whether we can emit on-negotiation-needed

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1233>
2021-10-28 19:05:59 +00:00
Rob Agar 641b319fd6 webrtcbin: Also check data channel transport when collating connection state
https://gitlab.freedesktop.org/gstreamer/gstreamer/-/issues/838

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1224>
2021-10-28 05:05:44 +00:00
Timo Wischer 20b87e39e9 avtpsrc: Retry receive with same buffer size
Without this patch in case of a retry recv() will be called with a
negative size argument.

Signed-off-by: Timo Wischer <timo.wischer@de.bosch.com>
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1078>
2021-10-26 22:46:46 +00:00
Mathieu Duponchelle e6f39394f5 cccombiner: fix default value when installing schedule property
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1252>
2021-10-26 14:48:13 +00:00
Mathieu Duponchelle e730bdaa8e cccombiner: fix emission of selected-samples in one case
Detected while reading the code, cccombiner must set
self->current_video_buffer to NULL *after* emitting selected-samples
in order for the application to get a useful return when peeking
the next video sample.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1252>
2021-10-26 14:48:13 +00:00
Mathieu Duponchelle fa1805d531 cccombiner: stop attaching caption buffers when caption pad has gone EOS
When schedule is true (as is the case by default), we insert padding
when no caption data is present in the schedule queue, and previously
weren't checking whether the caption pad had gone EOS, leading to
infinite scheduling of padding after EOS on the caption pad.

Rectify that by adding a "drain" parameter to dequeue_caption()

In addition, update the captions_and_eos test to push valid cc_data
in: without this cccombiner was attaching padding buffers it had
generated itself, and with that patch would now stop attaching
said padding to the second buffer. By pushing valid, non-padding
cc_data we ensure a caption buffer is indeed attached to the first
and second video buffers.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1252>
2021-10-26 14:48:13 +00:00
Mats Lindestam 9b3f040d3c curlsftpsink: Add support for sha256 fingerprint
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1193>
2021-10-26 12:18:40 +00:00
Floris 6d447acef5 gs: update README to use fixed versions
Use specific versions, instead of relying on 'master'.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1246>
2021-10-25 18:34:16 +00:00
Floris 14368c6457 gssrc: use default blocksize
The blocksize is set to 3 * 1024 * 1024 / 2, which is the default download_size of Google-Cloud-CPP.

Fixes https://gitlab.freedesktop.org/gstreamer/gstreamer/-/issues/846

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1246>
2021-10-25 18:34:16 +00:00
Rafał Dzięgiel cfbd820153 assrender: Add "application/vnd.ms-opentype" mimetype detection
The "application/vnd.ms-opentype" mimetype is commonly used in many fonts attached in the matroska videos.
Assrender should treat it as compatible without the need of parsing the file extension.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1207>
2021-10-22 23:41:26 +00:00
Stéphane Cerveau 22f502d864 wrap: libopenjp2: use patch version 7
Add support for win32 build
Disable the binary to avoid the thirdparty
dependency to be checked.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1229>
2021-10-22 19:47:27 +00:00
Rafał Dzięgiel 18b5806ab9 assrender: Do not iterate over mimetypes without filename
No point spending time on iterating and comparing strings if we
are going to reject the value due to missing filename anyway.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1206>
2021-10-21 22:21:40 +00:00
Rafał Dzięgiel e6ff71f8e7 assrender: Fix mimetype detection
Previously gst_structure_has_name was used to get a string to compare with supported mimetypes.
This is incorrect as above function returns a user defined structure name which is
not the structure mimetype value.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1206>
2021-10-21 22:21:40 +00:00
Tim-Philipp Müller 9a4651e387 mms: remove mmssrc plugin
Doubtful that anyone still needs that or there are even
any streams left out there.

MMS was deprecated in 2003 (in favour of RTSP) and support for
it was dropped with Microsoft Media Services 2008.

https://en.wikipedia.org/wiki/Microsoft_Media_Server
https://sdp.ppona.com/news2008.html

Fixes https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/issues/821

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1211>
2021-10-20 17:05:34 +00:00
Tim-Philipp Müller f1bb2c76c6 meson: update for dep.get_pkgconfig_variable() deprecation
... in favour of dep.get_variable('foo', ..) which in some
cases allows for further cleanups in future since we can
extract variables from pkg-config dependencies as well as
internal dependencies using this mechanism.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1183>
2021-10-20 11:20:44 +00:00
Rafał Dzięgiel c016bed943 assrender: fix smooth scaling by disabling hinting
When ass hinting value is set to anything other than NONE,
subtitles cannot use smooth scaling, thus all animations will jitter.

The libass author warns about possibility of breaking some scripts when it is enabled,
so lets do what is recommended and disable it to get the smooth scaling working.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1201>
2021-10-20 09:57:43 +00:00
Jan Alexander Steffens (heftig) df403df2ca srt: Plug leak of headers
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1040>
2021-10-19 07:19:41 +00:00
Antonio Ospite 7f001b6311 aes: specify the required OpenSSL version
The code in the aes elements assumes OpenSSL >= 1.1.0:

  - implicit library initialization;
  - version retrieved with OpenSSL_version(OPENSSL_VERSION);

and it fails to build with older versions.

Specify the required OpenSSL version explicitly in meson.build so that
the elements are excluded on older systems (e.g. Ubuntu 16.04) and the
rest of GStreamer can still build.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1067>
2021-10-18 23:25:50 +00:00