Commit graph

580 commits

Author SHA1 Message Date
Adrian Fiergolski
d8f449ccda avtp: cvf: extract AVTP VF depayload base class
Extract a part which could be common with the AVTP RVF depayload plugin to a separate class.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1335>
2023-02-02 19:15:38 +01:00
Adrian Fiergolski
4f2fde0163 avtp: rvf: add AVTP RVF payload support
Add AVTP Raw Video Format payload support. The element supports only GRAY16_LE
input format, so:
- active pixels (no vertical blanking),
- progressive mode,
- 8 and 16-bit pixel depth,
- mono pixel format,
- grayscale colorspace.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1335>
2023-02-02 19:15:33 +01:00
Adrian Fiergolski
8702a1fa67 avtp: cvf: extract AVTP VF payload base class
Extract a part which could be common with the AVTP RVF payload plugin to a separate class.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1335>
2023-02-02 19:15:29 +01:00
Sebastian Dröge
b12c66042b aom: Include stream-format and alignment in the AV1 caps
The decoder does not work with arbitrary alignment and annexb stream
format and the encoder can give the information that it outputs
obu-stream/tu to downstream.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3862>
2023-02-01 19:04:32 +00:00
Benjamin Gaignard
16ad80179b codec2json: Add av12json element
This element convert AV1 frame header into human readable
json data.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3734>
2023-01-30 19:46:55 +00:00
Benjamin Gaignard
fd588a50e4 codec2json: Add vp82json element
This element convert vp8 frame header into human readable
json data.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3734>
2023-01-30 19:46:55 +00:00
Sebastian Dröge
aba0b0e90a gstreamer: Use stream-format=obu-stream alignment=tu in the caps
There is no byte-stream/au format for AV1 but only for H264, and the
encoder actually outputs obu-stream/tu instead of the annexb
stream-format that is similar to H264 byte-stream format.

Without this the encoder can't be used with elements that require a
specific AV1 stream-format, e.g. the MP4 or Matroska/WebM muxer.
2023-01-26 01:46:46 +00:00
Nirbheek Chauhan
cc3078d819 meson: Add a wrap file for libsrt2p
And allow fallback to it.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3708>
2023-01-25 11:38:52 +00:00
Tim-Philipp Müller
82cd540d08 svthevcenc: drop use of GSlice
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3695>
2023-01-24 15:25:07 +00:00
Tim-Philipp Müller
93fc68c5ba x265: drop use of GSlice
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3695>
2023-01-24 15:25:07 +00:00
Tim-Philipp Müller
bfc2fab9ea wpe: drop use of GSlice
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3695>
2023-01-24 15:25:07 +00:00
Tim-Philipp Müller
664d83de99 ttml: drop use of GSlice
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3695>
2023-01-24 15:25:07 +00:00
Tim-Philipp Müller
c095a1d620 srtp: drop use of GSlice
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3695>
2023-01-24 15:25:07 +00:00
Tim-Philipp Müller
4994c730c8 resindvd: drop use of GSlice
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3695>
2023-01-24 15:25:07 +00:00
Tim-Philipp Müller
d1fe992f1f kate: drop use of GSlice
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3695>
2023-01-24 15:25:07 +00:00
Tim-Philipp Müller
c1759353c1 openjpeg: drop use of GSlice
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3695>
2023-01-24 15:25:07 +00:00
Jonas Danielsson
8eeaeab6af wpe: Add 'run-javascript' action signal
Introduce way of running a script in the context of the internal
webView.

Fixes #1722

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3753>
2023-01-20 10:58:31 +00:00
Sebastian Dröge
80e364876b gstreamer: Fix code style by running clang-format 2023-01-12 21:38:51 +02:00
Sebastian Dröge
7652026f0d gstreamer: Don't leak all video frames 2023-01-12 21:38:51 +02:00
Sebastian Dröge
ff911c76c0 gstreamer: Don't explicitly drop frames on stop()
This is already handled by the base class.
2023-01-12 01:06:38 +00:00
Sebastian Dröge
f56e8b2fad gstreamer: Remove unused variable 2023-01-12 01:06:38 +00:00
Sebastian Dröge
040c92d8b0 gstreamer: Stop outputting frames if pushing one has caused an error 2023-01-12 01:06:38 +00:00
Sebastian Dröge
5f03d9c4d1 gstreamer: Don't set bogus LIVE flag on output buffers 2023-01-12 01:06:38 +00:00
Sebastian Dröge
85b9c8e103 gstreamer: Allocate output buffers via the encoder
This makes sure the correct allocator and configuration is used.
2023-01-12 01:06:38 +00:00
Sebastian Dröge
30e0c1e4fd gstreamer: Negotiate the encoder immediately after setting the format 2023-01-12 01:06:38 +00:00
Sebastian Dröge
d9efa54783 gstreamer: Fix output state reference leak 2023-01-12 01:06:38 +00:00
Sebastian Dröge
b498bdb765 gstreamer: Add missing property setter/getter for lookahead property 2023-01-12 01:06:38 +00:00
Sebastian Dröge
13fa6d387d gstreamer: Fix reference leak of the input state if the caps are changing
Also remove misleading comment: reconfiguration was already handled by
the following code.
2023-01-12 01:06:38 +00:00
Sebastian Dröge
9128978042 gstreamer: The encoder has no maximum latency
It will buffer as much as it needs to.
2023-01-12 01:06:38 +00:00
Sebastian Dröge
785ba05bca gstreamer: Announce support for video meta on the input side
This allows handling input buffers with non-default strides, which was
already handled fine by the element code.

Without this, potentially expensive conversion was needed.
2023-01-12 01:06:38 +00:00
Sebastian Dröge
c5b166491c gstreamer: Don't override various virtual methods unnecessarily
There was no custom behaviour in there.
2023-01-12 01:06:38 +00:00
Sebastian Dröge
158c1f6602 gstreamer: Don't unnecessarily override decide_allocation()
This avoids more optimal output buffer allocation.
2023-01-12 01:06:38 +00:00
Sebastian Dröge
3b3e862580 gstreamer: Don't set a DTS and remove non-working DTS hack
The previous hack would create bogus DTS that confused other elements.

Fixes https://gitlab.com/AOMediaCodec/SVT-AV1/-/issues/1915
2023-01-12 01:06:38 +00:00
Sebastian Dröge
a8c6eb0606 gstreamer: Don't use private data but simply always get the oldest frame
The private data is not copied over for SVT AV1 encoder so this code
path would've never worked.

Instead of relying on the PTS, which is not required to be unique or
existing at all, we always take the oldest frame as AV1 has no frame
reordering / B frames.
2023-01-12 01:06:38 +00:00
Philippe Normand
f532ea6627 av1enc: Add property for controlling max distance between 2 keyframes
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2743>
2023-01-08 18:51:55 +00:00
Yatin Mann
59529ae918 aom: av1enc: Expose more properties
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2743>
2023-01-08 18:51:54 +00:00
yatinmaan
5cb04de96a aom: av1enc: Remove redundant enum variants from header
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2743>
2023-01-08 18:51:54 +00:00
Yatin Mann
cbc7334d93 aom: av1enc: Ensure that input pts is strictly increasing
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2743>
2023-01-08 18:51:54 +00:00
Yatin Mann
cfcd2aac67 aom: av1enc: Fix pts unit
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2743>
2023-01-08 18:51:54 +00:00
ekwange
beccaf31ef dfbvideosink: Fix compile error
Fix some compile errors

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3670>
2023-01-08 03:49:03 +00:00
Olivier Crête
f45cfe0d53 srt: Avoid crash on unknown option
Use the correct field that is null instead of the struct value which
never is.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3672>
2023-01-04 02:45:51 +00:00
Xavier Claessens
5ff5f9fd5b qroverlay: Add qrcode-case-sensitive property
This allows to encode case sensitive strings, like wifi SSID/password.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3662>
2023-01-03 19:40:20 +00:00
مهدي شينون (Mehdi Chinoune)
8d5ac30955 meson: Accept latest version of opencv 4.x
We've been bumping along the maximum opencv 4.x version for years,
just accept all opencv versions till someone reports breakage.

Fixes https://gitlab.freedesktop.org/gstreamer/gstreamer/-/issues/1680

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3664>
2022-12-30 17:10:12 +00:00
Philippe Normand
72884f141c webrtcbin: Support for setting kind attribute on RTCRtpStreamStats
The attribute maps the `kind` property of the associated transceiver.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3630>
2022-12-22 21:35:51 +00:00
Thibault Saunier
f7b342f1dd base:navigation: Cleanup navigation key modifiers enum
We were exposing the 'ALT' modifier as if we were guaranteeing its
accuracy but truth is we were only exposing configuration dependent
values.

Make the API simpler for now, the same way as Gtk3 was exposing it, and
when we have time to guarantee more values by making them take backends
configuration into account, we will expose those values in a accurate
way.

Fixes https://gitlab.freedesktop.org/gstreamer/gstreamer/-/issues/1402

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3565>
2022-12-15 16:47:13 +00:00
Matthew Waters
993bc8fc01 webrtc: implement support for msid values
Local msid values are taken from sink pad property, or fallback to the
previously used cname.

The remote msid values are exposed on the relevant src pads.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3106>
2022-12-14 12:23:32 +11:00
Stéphane Cerveau
7cfc3130a7 zxing: update to 1.4.0 tag
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3450>
2022-12-11 15:52:08 +00:00
Daniel Morin
855f84c558 onnx: Update to OnnxRT >= 1.13.1 API
- Replace deprecated methods
- Add a check on ORT version we are compatible with.
- Add clarification to the example given.
- Add the url to retrieve the model mentioned in the example.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3388>
2022-11-22 22:36:34 +00:00
Seungha Yang
6c007b8936 av1dec: Demote rank to secondary
cerbero does not build this plugin for now, and there's altanative
dav1ddec

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3287>
2022-11-22 17:48:25 +00:00
Jan Schmidt
dfb5e3365e webrtcbin: Remove queue after rtpfunnel
The original BUNDLE support commit placed a queue after the
rtpfunnel that combines streams, but I don't see a good reason for
it. It has default settings, so if network output is slow might
accidentally store up to 1 second of pending data, increasing
latency.

Remove it in favour of doing any necessary buffering before
webrtcbin. If it turns out that there is a reason for it to
exist, the limits should probably be configurable and small.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3437>
2022-11-19 10:31:50 +00:00
Jan Schmidt
5fa4f0562c webrtcbin: Fix a typo in debug log
transceiever -> transceiver

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3434>
2022-11-19 13:12:58 +11:00
Johan Sternerup
e708543039 webrtcbin: Add settings for HTTP proxy
Pass this to libnice which has a simple HTTP 1.0 proxy with basic
authentication only.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2867>
2022-11-18 15:00:58 +00:00
Enrique Ocaña González
a2990020b2 hlsdemux: Expose EXT-X-PROGRAM-DATE-TIME as tags.
This allows an application to use timestamps associated
with fragments.

Patch by: Thomas Bluemel <tbluemel@control4.com>

See: https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/issues/195
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1417>
2022-11-17 22:11:12 +00:00
Rafał Dzięgiel
30c2bdad61 mpdparser: Fix missing baseURL query
When no initializationURL or mediaURL, return baseURL that also
contains original URI query if available. This fixes a problem
where URI query was being omitted in the HTTP requests.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1147>
2022-11-14 23:45:53 +00:00
Rafał Dzięgiel
548bbc3147 mpdparser: Be consistent about returning duplicated URL
Instead of returning a "const gchar" or a "gchar" that should not be freed, always
return a duplicated string as those functions were used together with g_strdup anyway.

This is needed to prepare support for returning modified strings in next commit.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1147>
2022-11-14 23:45:53 +00:00
Rafał Dzięgiel
0d79dbedf3 mpdparser: Return correct mediaURL value
This fixes a problem where get_mediaURL was returning NULL when segmentURL
was unavailable instead of baseURL as a fallback.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1147>
2022-11-14 23:45:53 +00:00
Jan Schmidt
452890093d aesdec: Fix padding removal for per-buffer-padding=FALSE
When per-buffer-padding is FALSE, the OpenSSL context needs
to be told to remove any padding at the end of the ciphertext

Fixes https://gitlab.freedesktop.org/gstreamer/gstreamer/-/issues/1243

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3406>
2022-11-15 00:13:15 +11:00
Matthew Waters
5ca3988420 webrtc/datachannel: handle error messages from appsrc/sink
Fixes a possible race where closing a data channel may produce e.g.
not-linked errors.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3381>
2022-11-11 10:13:27 +00:00
Matthew Waters
a34e380e2e sctpdec: fix stream reset (src pad removal) if no data is ever received
If we don't receive any data from usrsctp, then there will be no src pad
for the stream id and the stream reset will fail to remove the relevant
src pad.  Workaround by first attempting to add the relevant src pad, then
almost immediately removing it.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3381>
2022-11-11 10:13:27 +00:00
Guillaume Desmottes
9eee5adb24 gssink: add 'content-type' property
Useful when one wants to upload a video as `video/mp4` instead of
'video/quicktime` for example.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3371>
2022-11-10 09:53:29 +00:00
Matthew Waters
e2ff6b61ce cccombiner: initial implementation of using CCBuffer helper
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3211>
2022-11-10 00:52:14 +00:00
Matthew Waters
088597b430 closedcaption: move CC buffering to helper object
Move most of the interesting code from ccconverter to this new helper
object.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3211>
2022-11-10 00:52:14 +00:00
Matthew Waters
06a20f9243 closedcaption: move cdp->cc_data into shared location
So it can be used by both ccconverter and cccombiner

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3211>
2022-11-10 00:52:14 +00:00
Matthew Waters
fde92ec43f closedcaption: move cc_data->cdp to shared file
Used by both ccconverter and cccombiner

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3211>
2022-11-10 00:52:14 +00:00
Matthew Waters
9f1b54f6ee ccconverter: avoid different indent versions indenting !! differently.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3211>
2022-11-10 00:52:14 +00:00
Matthew Waters
067185e7da closedcaption: move cdp framerate table to common file
shared by both cccombiner and ccconverter

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3211>
2022-11-10 00:52:14 +00:00
Matthew Waters
5dd199f7e8 cccombiner: don't assume a single cea608 data packet per buffer
e.g. 24fps can have up to 3 and would include either two field0 or
field1 cea608 data.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3211>
2022-11-10 00:52:14 +00:00
Matthew Waters
741cfd18b5 ccconverter: drop data when overflow on extracting cea608 from cc_data
If the buffer overflows, then drop rather than causing a failure and
fropping the output buffer indefinitely.  This may have caused downstream to
be waiting for data the will never arrive.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3211>
2022-11-10 00:52:14 +00:00
Matthew Waters
542060fea7 ccconverter: fix framerate passthrough with malformed input
If an input is malformed (only produces cea608 field 1 cc_data) then
when in passthrough we would effectively be dropping every second cea608
on output as we would not store any unused cea608 data.

Fix by having all code paths go through the framerate conversion code
which will store and retrieve any relevant data across buffers.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3211>
2022-11-10 00:52:14 +00:00
Jan Alexander Steffens (heftig)
28628a67e5 srt: Add a property to disable automatic reconnect
This adds a new boolean property `auto-reconnect`, defaulting to `true`.

Setting it to `false` makes the elements (in caller mode) immediately
report an error to the application instead of trying to reconnect.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3326>
2022-11-07 22:23:02 +00:00
Edward Hervey
a100f36b69 webrtcbin: Don't duplicate enum string values
Some were leaked when debugging was enabled. Instead just directly use the
static strings as-is.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3347>
2022-11-07 11:21:00 +00:00
Edward Hervey
f4d0537b3e lv2: Don't leak plugin information on registration
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3319>
2022-11-04 17:59:21 +00:00
Edward Hervey
685a4aaaa7 ladspa: Don't leak plugin information on registration
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3319>
2022-11-04 17:59:21 +00:00
Edward Hervey
8ca2a2a230 fdkaacenc: Properly terminate GEnumValue table
It should be terminated with a NULL entry, otherwise we just stray into the
realms of cryptographic libraries^W^W random memory usage.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3319>
2022-11-04 17:59:21 +00:00
Jan Alexander Steffens (heftig)
424b331afc srt: Remove callers for which srt_bstats fails
This keeps them from accumulating in the element and in the stats while
the sink is not being fed, as long as we at least periodically grab
stats.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3156>
2022-11-04 13:07:34 +00:00
Jan Alexander Steffens (heftig)
d575a41145 srt: Use simpler list operations for callers
Avoid `g_list_append` and `g_list_remove` (which have to scan the list)
and replace them with `g_list_prepend` and `g_list_delete_link`.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3156>
2022-11-04 13:07:34 +00:00
Jan Alexander Steffens (heftig)
3c23c16f40 srt: Clean up poll/sock lifecycle
Make sure `srtobject->poll_id` is never invalid as long as `srtobject`
exists. Only remove our caller socket from it when the socket becomes
invalid.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3156>
2022-11-04 13:07:34 +00:00
Jan Alexander Steffens (heftig)
4e05100e8c srt: Clean up error handling
- Make the srt_epoll_wait loops more uniform.

- Error only via GError when possible; let the element send the error
  message. Avoids a second error message.

- Return 0 when cancelled. Avoids an error message from the element.

- Don't send an error message from send_headers when we're a server
  sink.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3156>
2022-11-04 13:07:34 +00:00
Jan Alexander Steffens (heftig)
a3cc5cf257 srt: Simplify socket stats
Don't hide stats depending on whether we're a sending or receiving
socket. While we're here, add some more debug logs.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3156>
2022-11-04 13:07:34 +00:00
Jan Alexander Steffens (heftig)
b6974b6afc srt: Replace stats accumulation with naive byte counting
srt_bstats cannot be used to get the stats of closed connections, so the
best we can do is keep the running count ourselves.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3156>
2022-11-04 13:07:34 +00:00
Guillaume Desmottes
a92f41e0c7 wpe: fix wpevideosrc gst-play example
wpe:// no longer works since 1.20, see wpesrc examples.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3306>
2022-11-02 00:21:21 +00:00
Sanchayan Maity
da52bedbff fdkaacenc: Update documentation to clarify bitrate and peak-bitrate
bitrate property is only applicable for constant bitrate and
peak-bitrate is only applicable for variable bitrate. Clarify
the same.
2022-10-30 16:54:51 +05:30
Sanchayan Maity
f0ceb9ea4f fdkaacenc: Add support for setting bitrate mode 2022-10-30 16:54:51 +05:30
Sanchayan Maity
595dd7a1ed fdkaacenc: Add support for setting peak bitrate 2022-10-29 16:04:42 +05:30
Sanchayan Maity
734593ccab fdkaacenc: Add support for enabling afterburner
This is an additional quality parameter. In the default configuration this
quality switch is deactivated because it would cause a workload increase
which might be significant. If workload is not an issue in the application
it can be recommended to activate this feature.
2022-10-29 15:57:52 +05:30
Sanchayan Maity
a63d8ee720 fdkaacdec: Do not report decoding error for flush request
A flush request is done when set_format is called to empty internal bit
buffer maintained by fdk-aac. When this happens, during the explicit
call to handle_buffer, decodeFrame does not return a AAC_DEC_OK. This
gets reported as a decoding error while no decoding error in fact took
place. Since this can be confusing, just return a GST_FLOW_OK and log
that an explicit flush was requested.
2022-10-29 10:47:16 +05:30
Thibault Saunier
4f991a55af adaptivedemux: Minor typo fix
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3159>
2022-10-27 19:45:44 +00:00
Thibault Saunier
8a9821e805 dash: Fix computing repeat_index when seeking in stream with a start !=0 on the first fragment
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3159>
2022-10-27 19:45:44 +00:00
Tim-Philipp Müller
d7e2aff994 fdkaacenc: fix output caps in case of implicit signaling and HE-AAC
Need to put the actual profile in the output caps otherwise any
capsfilter after the encoder that was used to force the output
profile will fail, such as

  fdkaacenc ! audio/mpeg,stream-format=adts,profile=he-aac-v1 ! ..

because we put profile=lc in there to match the profile signaled
in the ADTS header. This is expressed through the base-profile=lc
in the GStreamer caps though, the profile needs to carry the
'real' profile.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1785>
2022-10-25 00:13:04 +00:00
Tim-Philipp Müller
24645e35c5 fdkaacenc: don't set base-profile=lc for non-backwards compatible output
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1785>
2022-10-25 00:13:04 +00:00
Tim-Philipp Müller
31c04f87e3 fdkaacenc: rename profile=sbr|ps to profile=he-aac-v1|he-aac-v2
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1785>
2022-10-25 00:13:04 +00:00
Piotrek Brzeziński
d8b1ff4668 fdkaacenc: add support for AAC-LD
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1785>
2022-10-25 00:13:04 +00:00
Piotrek Brzeziński
8cda666cb0 fdkaacenc: add support for HE-AACv1 and HE-AACv2
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1785>
2022-10-25 00:13:04 +00:00
Matthew Waters
0077d13304 webrtcbin: configure rtpulpfecdec passthrough property
This allows downstream (payloaders mostly) to be able to correctly
detect actual packet loss from rtp sequence numbers.

See
https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/issues/581
for background.

Fixes https://gitlab.freedesktop.org/gstreamer/gstreamer/-/issues/1407

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3212>
2022-10-23 23:44:07 +00:00
Matthew Waters
a633f5d287 webrtcbin: also add rtcp-fb ccm fir for video mlines by default
In addition to the 'nack pli' already added.  Both are supported by
rtpbin/rtpsession by default already.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3235>
2022-10-21 01:02:34 +00:00
Sangchul Lee
0f05be382b webrtcbin: Improve documentation of 'turn-server' property
Description about how to set time-limited credentials is added.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3229>
2022-10-20 15:30:07 +00:00
Fabian Orccon
50c6c54675 srtp: Fix test skipping when plugin option is disabled
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3200>
2022-10-18 22:12:41 +00:00
Johan Sternerup
44eea7bd8a 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>
2022-10-11 09:36:13 +00:00
Xavier Claessens
56eb44c502 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>
2022-10-07 07:56:21 -04:00
Sangchul Lee
93b896eb4e webrtcbin: Fix pointer dereference before null check
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3129>
2022-10-06 16:46:33 +00:00
Johan Sternerup
212c09a70e 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>
2022-10-05 11:08:30 +00:00
Devin Anderson
31831eb47e voamrwbenc: Fix truncation of audio data at end-of-stream when audio data
doesn't align on 20 millisecond frame size.

The AMR-WB codec imposes a fixed 20 millisecond frame size.  In its current
form, the `voamrwbenc` plugin deals with this limitation by discarding any
audio at the end of the stream that falls short of 20 milliseconds.  This patch
keeps the audio data, and appends silence to the end to preserve frame size
alignment.

The patch also adds tests to check for the updated behavior.  I noticed that
tests weren't being built, so I changed the build to allow for building the
tests when the `tests` and `voamrwbenc` options are set.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3027>
2022-09-16 00:14:58 +00:00
Mathieu Duponchelle
b454ec972f webrtcbin: fix picking available payload types
When picking an available payload type, we need to pick one that is
available across all media.

The previous code, when multiple media were present, looked at the first one,
noticed it had pt 96 as the media pt, then simply looked at the next media,
noticed it didn't, and decided 96 was available.

Instead, check if the pt is used by any of the media, if it is, decide
it is not available and go to the next pt. I'm fairly sure that was the
original intent.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2984>
2022-09-07 03:22:34 +00:00
Jordan Petridis
a7f9c97454 fluidsynth: correctly version guard methods
We bumped the minimum version to 2.1 but the api we used
wasn't introduced till version 2.2 of fluidsynth

Follow-up to gstreamer/gstreamer!2718

https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2718

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2835>
2022-09-05 17:48:27 +00:00
Jan Schmidt
4e25c519de dashdemux: Preserve current representation on live manifest updates
When updating a manifest during live playback, preserve the current
representation for each stream.

During update_fragment_info, if the current representation changed
because it couldn't be matched, trigger a caps change and new
header download.

This reverts commit e0e1db212f
and reapplies "dashdemux: Fix issue when manifest update sets slow start
without passing necessary header & caps changes downstream" with
changes.

Fixes https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/issues/507
Fixes https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/issues/1729

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2920>
2022-09-05 16:07:00 +00:00
Olivier Crête
4b3b234f72 webrtcbin: Allow locked mlines with no caps, as the last ones
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2439>
2022-09-02 11:52:58 +02:00
Olivier Crête
0930c467d4 webrtcbin: Reject creating an offer if a locked mline has no caps
This avoids getting in a bunch of corner cases. We'd have to insert
a "rejected" line from the start as a place-holder to get around this,
but the rest of the code just becomes more complicated, so just
disallow it for now.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2439>
2022-09-02 11:52:58 +02:00
Olivier Crête
3503599e0a webrtcbin: Store pending mid to make create-offer idempotent
If the mid is not stored in the transceiver, but it is stored in
last_offer, then a further create-offer call will just ignore that
transceiver.

Also include unit test for ensure it doesn't regress.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2439>
2022-09-02 11:52:58 +02:00
Thibault Saunier
6a4425e46a meson: Call pkgconfig.generate in the loop where we declare plugins dependencies
Removing some copy pasted code

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2970>
2022-09-01 21:17:35 +00:00
Robert Rosengren
ab9ce0500a curlbasesink: gst_curl_base_sink_transfer_thread_close is internal
gst_curl_base_sink_transfer_thread_close is moved from external header
to be static function, as it has no users.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2732>
2022-08-29 09:40:23 +00:00
Robert Rosengren
8677d573b7 curlhttpsink: Only set MIME as content-type if not set by property
Setting the content-type property shall override internally detected MIME
types, to make it possible to do as following example (where audio/basic to be
used prior to audio/x-mulaw):

gst-launch-1.0 ... ! mulawenc ! audio/x-mulaw,rate=8000,channels=1 !
  curlhttpsink location=<url> content-type=audio/basic

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2732>
2022-08-29 09:40:23 +00:00
Philippe Normand
0151d621af openh264: Register debug categories earlier
Otherwise the GST_ERROR message logged in case of ABI mismatch would be done on
an uninitialized category.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2918>
2022-08-22 13:34:33 +00:00
Philippe Normand
cfd3bd4850 openh264enc: Fix constrained-high encoding
constrained-high is high without B-frames, there is no EProfileIdc for this, so
assume high instead of hitting an assert down the line.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2919>
2022-08-20 16:57:27 +01:00
Philippe Normand
90d46c1748 wpesrc: Switch URI handler to web+... protocols
The web://http:// URIs were not compliant with RFC 3986. Using web+http://
allows us to use the GstUri parser to pass down a valid URI to `wpevideosrc`.

Corresponding change for the CEF source element:
8d499495dd

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2856>
2022-08-10 15:10:26 +00:00
Robert Mader
e93773bda7 waylandsink: Logging code style updates
For better readability of debug messages and to keep similar code
in sync with `GstGtkWaylandsink`.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2694>
2022-08-03 14:25:17 +00:00
Robert Mader
062638a639 waylandsink: Rename occurrences of GstWaylandSink to 'self'
Rename all occurrences to `self`, making it consintent with `GstWl*`
and `GstGtkWaylandsink`.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2694>
2022-08-03 14:25:17 +00:00
George Kiagiadakis
7e18fc1b1f Add new gtkwaylandsink element
This is based on gtksink, but similar to waylandsink uses Wayland APIs
directly instead of rendering with Gtk/Cairo primitives.

Note that the long term plan is to move this into the existing extension
in `-good`, which requires the Wayland library to move the as well.

For this reason several files like `gstgtkutils.*` and `gtkgstbasewidget.*`
are straight copies and should be kept in sync.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1515>
2022-08-02 16:34:13 +00:00
Philippe Normand
10eaae1243 dtls: Properly name encoder/decoder logging categories
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2820>
2022-08-01 09:02:03 +00:00
Philippe Normand
7c3f73ec2e dtls: Make agent and connection GstObjects
Facilitates debug logs interpretation of GST_DEBUG_OBJECT() calls.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2820>
2022-08-01 09:02:03 +00:00
Nirbheek Chauhan
b2d22c0f00 meson: Don't pass -Werror to vendored code
Do it the correct way with libusrsctp -- override the option so that
it's done in a compiler-agnostic and future-proof way.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2494>
2022-07-30 11:27:12 +00:00
Nirbheek Chauhan
11ecda9d73 dtls: Disable OpenSSL 3.0 deprecation warnings for now
Fedora 36 ships with OpenSSL 3.0, which deprecates all low-level APIs,
so this code needs to be rewritten. There is no easy fix in the
porting guide, and it recommends disabling the warnings if you can't
use the high-level API.

https://wiki.openssl.org/index.php/OpenSSL_3.0#Upgrading_to_OpenSSL_3.0_from_OpenSSL_1.1.1

Here's the replacement API:

https://www.openssl.org/docs/man3.0/man7/migration_guide.html#Deprecated-low-level-object-creation

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2494>
2022-07-30 11:27:12 +00:00
U. Artie Eoff
e3e98da727 meson: webrtc: ensure definition of libgstwebrtcnice_dep
... and skip if it's disabled.

Fixes #1344

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2797>
2022-07-26 17:39:52 -04:00
yatinmaan
2c1e61ea16 webrtc: Split WebRTCICE into base classes and implementation.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2398>
2022-07-26 13:51:11 +00:00
Jordan Petridis
3a20a4564f openmpt: update from now deprecated api
https://lib.openmpt.org/doc/classopenmpt_1_1module.html#ab2695af0baa274054f5687741fa7c05b

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2721>
2022-07-21 07:52:37 +00:00
Thibault Saunier
073df3d820 webrtcbin: Add a signal to plug bandwidth estimator elements
We need GStreamer elements to do the bandwidth estimation as this way
they can also control the pacing of the transmission flow as specified
 in the [GCC] algorithm for example.

Bandwidth estimator element are placed right before the "RTPSession" as
an "rtp-aux-sender" element. This way they can use the "Transport-wide
Congestion Control" RTCP feedback messages through the "RTPTwcc" custom
events that are sent by the rtpsession.

Applications are responsible to react to the bandwidth estimator element
and set the encoder target bitrate etc... which means that we can not
pass an estimator as an element factory, so a signal as been chosen
instead.

[GCC]: https://datatracker.ietf.org/doc/html/draft-ietf-rmcat-gcc-02

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2562>
2022-07-12 20:40:55 +00:00
Christopher Degawa
8478600596 gst: init metadata to null
Signed-off-by: Christopher Degawa <christopher.degawa@intel.com>
2022-07-09 15:19:11 -05:00
Jordan Petridis
3385ea3481 fluiddec: Remove workaround for version 1.1.9
We require >= 2.1 version since the previous commit

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2718>
2022-07-09 14:19:11 +00:00
Jordan Petridis
2fa6ec8733 fluidsynth: update from now deprecated api
fluid_synth_set_chorus_on and fluid_synth_set_reverb_on were
deprecated in favor of new funtions where you can also specify
the fx_group the effect would apply.

The behavior of the set_* variants was to apply to all groups
so we pass -1 to the new functions as per documentation.

https://www.fluidsynth.org/api/group__chorus__effect.html#ga3c48310eecdca9cd338799d19f19c32d

and

https://www.fluidsynth.org/api/group__reverb__effect.html#gacb7917564c988cf54f2e35189b509c8e

and the introduction of the change:

https://github.com/FluidSynth/fluidsynth/pull/673

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2718>
2022-07-09 14:19:11 +00:00
Matthew Waters
6066e913ee webrtc: implement support for asynchronous host resolution
Doesn't block anymore if a mdns host resolution takes multiple seconds
to complete in e.g. stun/turn/ice candidate usage.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1961>
2022-07-05 03:20:57 +00:00
Sebastian Dröge
a54eddad3a webrtcbin: Reject caps that are not valid for creating an SDP media.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2689>
2022-06-30 09:28:27 +00:00
Tim-Philipp Müller
afc94046ba dv, opusparse: fix duplicate symbols in static build
Fixes https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/issues/1262

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2670>
2022-06-28 01:29:06 +01:00
Robert Mader
6aa0b0cae2 gstwaylandsink: Add rotate-method property
Similar to and inspired by glimagesink and gtkglsink.

Using the Wayland buffer transform API allows to offload
rotate operations to the Wayland compositor. This can have
several advantages:
 - The Wayland compositor may be able to use hardware plane
   capabilities to do the rotation.
 - In case of pre-rotated content on rotated outputs the
   rotations may equal out, potentially allowing the
   compositor to use hardware planes even if they don't
   support rotate operations.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2543>
2022-06-20 18:30:56 +00:00
Mathieu Duponchelle
f10e2eb88f cccombiner: expose output-padding property
When schedule=true and output-padding=false, cccombiner will not
inject padding in the output closed caption meta stream.

The property has no effect when schedule=false.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1621>
2022-06-17 14:11:46 +00:00
Olivier Crête
c4971a456e webrtcbin: Limit sink query to sink pads
This allows the reception of streams that don't exactly match
the codec preferences. In particular, the ssrc in the codec preferences
is local sender SSRC, the other side is expected to send a different SSRC.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2615>
2022-06-17 08:08:43 +00:00
Stéphane Cerveau
19972b8153 srtsrc: add "keep-listening" property to avoid EOS on disconnect
The property 'keep-listening' avoids EOS
when the remote client disconnects.

It can be useful to a keep a pipeline alive
when the srt connection drops remotely.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/967>
2022-06-15 20:35:14 +00:00
Stéphane Cerveau
eb1f21b484 srtsrc: remove dead code
Remove code useless since
132e3a1af9

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/967>
2022-06-15 20:35:14 +00:00
Matthew Waters
9df7a21ec9 vulkan: add vulkan overlay compositor element
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2470>
2022-06-14 03:34:06 +00:00
Matthew Waters
81e601ccaa vulkan: move element register definition to relevant element headers
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2470>
2022-06-14 03:34:05 +00:00
Tim-Philipp Müller
9d9e59622f Bump GLib requirement to >= 2.62
Can't require 2.64 yet because of
https://gitlab.freedesktop.org/gstreamer/cerbero/-/issues/323

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2568>
2022-06-10 06:01:41 +00:00
Philippe Normand
c287711418 webrtcbin: Add a prepare-data-channel GObject signal
This new signal allows data-channel consumers to configure signal handlers on a
newly created data-channel, before any data or state change has been notified.

The webrtcin unit-tests were refactored to make use of this new signal.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2427>
2022-06-07 11:29:33 +00:00
Philippe Normand
779ca38229 webrtcdatachannel: Chain to parent class constructed
And add a debug log statement.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2427>
2022-06-07 11:29:32 +00:00
Robert Mader
eb915b662a gstwaylandsink: Add support for the "render-rectangle" property
We already implement the `set_render_rectangle` videooverlay interface,
thus install the videooverlay property accordingly.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2479>
2022-06-06 14:36:39 +02:00
Robert Mader
8c3e33d494 gstwayland: Move reusable parts of the waylandsink into a library
In preparation for the new element `GstGtkWaylandSink`, move reusable
parts out of `GstWaylandSink` into the already exisiting but very
barebone library.

Notable changes include:
 - the `GstWaylandVideo` interface was dropped
 - support for `wl-shell` was dropped
 - lots of renaming in order to match established naming patterns
 - lots of code modernisations, reducing boilerplate
 - members were made private wherever possible

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2479>
2022-06-06 14:36:39 +02:00
Jan Alexander Steffens (heftig)
d86ad30be2 opencv: Allow building against 4.6.x
Replace the broken version checks with one modeled after
`GLIB_CHECK_VERSION`.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2557>
2022-06-06 00:30:15 +02:00
Olivier Crête
9fe2e1c5eb webrtcbin: Reject answers that don't contain the same number of m-line as offer
Otherwise, it segfaults later. Also add test to validate this.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2526>
2022-06-03 20:28:19 +00:00
Cidana-Developers
e2a6d5f76d refine reference scaling
1. modify codes by review suggestion
2. clean-up macros
2022-05-26 23:29:51 +00:00
Cidana-Developers
b223764832 add random access for reference scaling
1. add random access configuration for reference scaling fixed and random mode
2. add e2e tests for random access configuration of reference scaling
2022-05-26 23:29:51 +00:00
Tim-Philipp Müller
962dc37d4f webrtc: fix build with older libnice versions
1) check for right macro name when checking for NICE_VERSION_CHECK

2) if libnice version is 0.1.18.1 this should not satisfy
   a NICE_VERSION_CHECK(0,1,19).

Fixes build with libnice 0.1.18.1 subproject checkout.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2499>
2022-05-26 18:17:49 +00:00
Philippe Normand
eefd793011 webrtc: Use new libnice API to get the candidate relay address
Corresponding libnice API added in:
https://gitlab.freedesktop.org/libnice/libnice/-/merge_requests/229 (0.1.19)
https://gitlab.freedesktop.org/libnice/libnice/-/merge_requests/232 (0.1.20)

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1998>
2022-05-26 10:54:59 +00:00
Philippe Normand
08021caa73 webrtc: Ensure the NICE_CHECK_VERSION macro is available
This macro was introduced in libnice 0.1.19.1, so until we bump our libnice
dependency to 0.1.20 we have to vendor the macro.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1998>
2022-05-26 10:54:59 +00:00
Philippe Normand
c19319c777 webrtc: Refactor ICECandidateStats freeing logic to a dedicated function
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1998>
2022-05-26 10:54:59 +00:00
Philippe Normand
dce8a7750d webrtcbin: Document IceCandidateStats and RTCIceCandidatePairStats
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1998>
2022-05-26 10:54:59 +00:00
Sherrill Lin
3e7fb83393 webrtcstats: Improve selected candidate pair stats by adding ICE candidate info
The implementation follows w3.org specs:
* https://www.w3.org/TR/webrtc-stats/#icecandidate-dict*
* https://www.w3.org/TR/webrtc-stats/#candidatepair-dict*

Corresponding unit tests are also added.

Rebased and updated from
https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1462

Fixes #1207

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1998>
2022-05-26 10:54:59 +00:00
Matthew Waters
be2dfd0c36 webrtcbin: reuese the same fec/rtx/red payload types for the same media payload
WHen bundling, if multiple medias are used with the same media payload, then
each of the fec/rtx/red additions would add a distinct payload.  This could
very easily overflow the available payload space.

Instead, track the relationship between the media payload value and
the relevant fec/rtx/red payload values and reuse them whenever
necessary, even when bundling.

e.g.

...
a=group:BUNDLE video0 video1
m=video 9 UDP/SAVPF 96 97
a=mid:video0
a=rtpmap:96 VP8/90000
a=rtpmap:97 rtx/90000
a=fmtp:97 apt=96
...
m=video 9 UDP/SAVPF 96 97
a=mid:video1
a=rtpmap:96 VP8/90000
a=rtpmap:97 rtx/90000
a=fmtp:97 apt=96
...

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2474>
2022-05-24 10:21:11 +00:00
Philippe Normand
556ee45bfa datachannel: Notify low buffered amount according to spec
Quoting
https://www.w3.org/TR/webrtc/#dom-rtcdatachannel-bufferedamountlowthreshold

The bufferedAmountLowThreshold attribute sets the threshold at which the
bufferedAmount is considered to be low. When the bufferedAmount decreases from
above this threshold to **equal** or below it, the bufferedamountlow event fires.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2448>
2022-05-19 05:52:51 +00:00
Ludvig Rappe
26263c194e webrtc: Fix memory leak in icestream
Since both g_value_set_object() and g_weak_ref_get() takes a reference
there will be two new references to the GstWebRTCICE object when there
should be only one. g_value_take_object() has the same functionality as
g_value_set_object() but does not take a reference.

Without this change, the GstWebRTCICE object will be leaked.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2333>
2022-04-29 21:52:43 +00:00
Thibault Saunier
4fd3886f5d qroverlay: Reset data_changed after we use the info
It was never reset so it was always TRUE once the data was changed!

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2299>
2022-04-27 15:09:47 +00:00
Thibault Saunier
1b31a2af45 qroverlay: Add a GstQROverlay meta
See documentation

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2299>
2022-04-27 15:09:47 +00:00
Stéphane Cerveau
c77d07752a srtpdec: add counts in stats
In order to count the buffers which have been received and dropped for
decryption reason, add a stats to track it.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2027>
2022-04-25 13:57:42 +00:00
Stéphane Cerveau
9d6a7dbdf3 rvsg: fix cairo include
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2276>
2022-04-23 00:00:23 +00:00
Sangchul Lee
c5b1eecb69 webrtcbin: Avoid access of freed memory
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2256>
2022-04-22 14:45:05 +00:00
Wonchul Lee
150db81287 dashsink: Unlock when failed to get content
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2242>
2022-04-20 09:07:29 +00:00
Xavier Claessens
b99ecc78ca Replace gst-i18n-*.h with gi18n-lib.h
GLib guarantees libintl is always present, using proxy-libintl as
last resort. There is no need to mock gettex API any more.

This fix static build on Windows because G_INTL_STATIC_COMPILATION must
be defined before including libintl.h, and glib does it for us as part
as including glib.h.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2028>
2022-04-19 18:01:06 +00:00
Olivier Crête
2771490992 wpevideosrc: Give WebKit the keyboard, touch and pointer modifiers
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2156>
2022-04-12 11:52:34 +00:00
Olivier Crête
41967e503c wpesrc: Convert from utf32 to support other keys
This makes all of the non-letter keys work.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2156>
2022-04-12 11:52:34 +00:00
Olivier Crête
3ca61ae0d3 wpesrc: Initialize key event to 0
Otherwise, WebKit sees random modifiers

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2156>
2022-04-12 11:52:34 +00:00
Johan Sternerup
1842ffc906 webrtc: Improve robustness of nice agent signal handlers
NiceAgent and it's associated thread is alive for as long as
GstWebRTCICE is alive so make sure any signal handlers connected to
NiceAgent do not access data that is deleted earlier.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2073>
2022-04-04 02:10:35 +00:00
anaghdin
02258ba529 - In lowdelay VBR is not supported: forces CBR, In RA, CBR is not supported, forces VBR
- Limit the max bit rate and target bitrate to 100,000 kbps
- Remove frame_rate from API. Inside library frame_rate is always in Q16 format
- Fix the seg fault with 2 PASS and max bit rate
- Remove frame_rate from CI and gstreamer
2022-04-01 23:47:38 +00:00
Xavier Claessens
b004464ac6 Remove glib and gobject dependencies everywhere
They are part of gst_dep already and we have to make sure to always have
gst_dep. The order in dependencies matters, because it is also the order
in which Meson will set -I args. We want gstreamer's config.h to take
precedence over glib's private config.h when it's a subproject.

While at it, remove useless fallback args for gmodule/gio dependencies,
only gstreamer core needs it.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2031>
2022-04-01 16:32:17 +00:00
Xavier Claessens
f270f9e974 Fix cross build with mingw32
At least on Ubuntu 20.04 the x86_64-w64-mingw32-gcc toolchain defaults
to WinXP. We require at least Vista for FILE_STANDARD_INFO.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2022>
2022-04-01 15:52:28 +00:00
Sangchul Lee
a801d6dd63 webrtcstats: Unify 'packets-lost' data type to int64
Previously, 'packets-lost' member of RTCReceivedRtpStreamStats had
a value of G_TYPE_INT from rtpsource or a value of G_TYPE_UINT64
from rtpjitterbuffer.
Because of the negative value of estimated amount of packets lost
in rtpsource as well as the description in
https://www.w3.org/TR/webrtc-stats/#dom-rtcreceivedrtpstreamstats
it is fixed to set this value with G_TYPE_INT64.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2049>
2022-03-31 05:37:39 +00:00
Matthew Waters
041eee6c2e webrtc: produce stats for all relevant streams
Instead of only using the last ssrc that was pushed into a sink pad.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1664>
2022-03-29 23:55:41 +00:00
Matthew Waters
04de1a161f webrtc: avoid different versions of gnu-indent always wanting to change !!
Add some sneaky parenthesis to avoid always having to use git commit -n
or revert out hunk of the change.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1664>
2022-03-29 23:55:41 +00:00
Matthew Waters
5bfe36746a webrtc: implement initial simulcast fec/rtx usage
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1664>
2022-03-29 23:55:41 +00:00
Matthew Waters
5741ee38e0 webrtc/datachannel: fix use-after-free in sctp state notification
g_signal_disconnect*() doesn't stop any existing callbacks from running
which means that if the notify::state callback is in progress in one
thread and the data channel object is finalize()ed in another thread,
then there could be a use-after-free trying lock the data channel
object.

We can't reasonably use a GWeakRef as we don't have a 'parent' object to
free the GWeakRef after the data channel is finalized.  This is also
complicated by the fact that the application can hold a reference to the
data channel object that would live beyond the lifetime of webrtcbin
itself.

We solve this by implementing a ghetto weak-ref solution internally with
a list of outstanding data channels.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1664>
2022-03-29 23:55:40 +00:00
Matthew Waters
2377f8b3f2 webrtcbin: initial support for sending and receiving simulcast streams
Input (sink pads) is the already-ssrc-muxed stream with the relevant rtp
sdes header extensions already applied:
  - mid
  - stream-id
  - repaired-stream-id

Output (src pads) have the pads separated into individual ssrc's as
that's what rtpbin gives us.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1664>
2022-03-29 23:55:40 +00:00
Matthew Waters
699739c130 webrtcbin: support multiple received streams for a single mline
Each rtpbin exposed recv_src pad is now exposed as webrtcbin src_%u pad
now with no meaining applied to the value of %u.  Previously this used
to mean the mline in the SDP.  If this is is still required, then the
transceiver can be retrieved from the pad and the "mlineindex" property
from the transciever.  The "mid" is also retrievable from the
transceiver.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1664>
2022-03-29 23:55:40 +00:00
Matthew Waters
e28c45fd05 webrtc: explicitly error out in a couple of renegotiation cases
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1664>
2022-03-29 23:55:40 +00:00
Matthew Waters
318a639e43 webrtc/transportstream: add debug category
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1664>
2022-03-29 23:55:40 +00:00
Matthew Waters
2aeca9ed84 webrtcbin: don't name src pads based on the mline specifically anymore
Naming based on the mline doesn't really work with e.g. simulcast
scenarios.

It is entirely possible to retrieve the transceiver and then the mline
from that if that is so required.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1664>
2022-03-29 23:55:40 +00:00
Matthew Waters
cda81bdb1e webrtcbin: improve some debugging output
- Put human readable names into debug strings.
- Demote some frequent rtpbin signal logging
- Don't use GST_PTR_FORMAT in g_set_error()

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1664>
2022-03-29 23:55:40 +00:00
Matthew Waters
533d4937fe webrtcbin: add a specific find_transceiver_by_mid function
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1664>
2022-03-29 23:55:40 +00:00
Matthew Waters
79d58200c9 webrtcbin: explicitly use a variable for the rtp session idx
Slightly clearer in meaning.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1664>
2022-03-29 23:55:40 +00:00
Matthew Waters
9a758d78a9 webrtcbin: support using an a=mid value from the sink/transceiver caps
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1664>
2022-03-29 23:55:40 +00:00
Matthew Waters
fc28db57ae resindvd: silence unused-but-set warning
../ext/resindvd/gstpesfilter.c:117:11: error: variable 'STD_buffer_size_bound' set but not used [-Werror,-Wunused-but-set-variable]
  guint16 STD_buffer_size_bound;
          ^

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2046>
2022-03-28 10:30:23 +00:00
Acky Xu
86c308549f Prevent gstreamer from reinitializing svt default values 2022-03-26 02:28:16 +00:00
Matthew Waters
2e69886a02 ccconverter: ensure correct ordering of cea608 across output buffers
e.g. if a 60fps output is configured, we can only produce a single field
of cea608 that must alternate between field 1 and field 2.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2019>
2022-03-26 00:00:36 +00:00
Matthew Waters
6977119f99 ccconverter: ignore padding cea608 data even if marked as 'valid'
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2019>
2022-03-26 00:00:36 +00:00
Thibault Saunier
25819c41fb navigation: Add support for key Modifiers in all relevant events
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2010>
2022-03-25 15:16:03 +00:00
Chun-wei Fan
b9f29bfc39 openexr: Specify modules when finding OpenEXR.
Specify modules to look for OpenEXR when CMake is used, as we may have
CMake config files instead of pkg-config files that result from building
OpenEXR, which may be built with CMake which is typically the case on Visual
Studio builds.

In this case, Meson does seem to find the 'OpenEXR' package with CMake
after trying pkg-config, but does not consider it enough without the
'modules:' argument.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2014>
2022-03-25 07:45:54 +00:00
Sangchul Lee
952c1194f3 webrtcbin: Update documentation of 'get-stats' action signal
Some stats fields are updated according to the current implementation.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2005>
2022-03-25 07:01:40 +00:00
Mathieu Duponchelle
29de0e8e1d Revert "webrtcbin: fix msid line and allow customization"
This reverts commit 3cad3455377d5a22faa138d9df840257059776c8.

That commit was breaking the association between an audio and
a video track in the standard case.

In practice, to support carrying separate MediaStream, we are
going a way to map what MediaStreamTrack belong to what MediaStream,
but that will require some thinking about the API.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2023>
2022-03-25 00:31:58 +01:00
Mathieu Duponchelle
06fec40f45 webrtcbin: fix msid line and allow customization
From https://datatracker.ietf.org/doc/html/draft-ietf-mmusic-msid-16:

> Multiple media descriptions with the same value for msid-id and
> msid-appdata are not permitted.

Our previous implementation of simply using the CNAME as the msid
identifier and the name of the transceiver as the msid appdata was
misguided and incorrect, and created issues when bundling multiple
video streams together: the ontrack event was emitted with the same
streams for the two bundled medias, at least in Firefox.

Instead, use the transceiver name as the identifier, and expose
a msid-appdata property on transceivers to allow for further
customization by the application. When the property is not set,
msid-appdata can be left empty as it is specified as optional.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2003>
2022-03-24 16:43:29 +00:00
Thibault Saunier
fe16900dff wpe: Mark first audio buffer as discont
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1492>
2022-03-24 00:01:20 +00:00
Thibault Saunier
a14e36fde4 wpe: Use about:blank as default URL to support only using load-bytes
WebKit is not going to render anything until a URI is set, leading to a
WPE posting a `WPE View did not render a buffer` error message. To avoid
requiring the user to know it if they only want to use
`wpesrc::load-bytes` we can just use `about:blank` as default and
everything will work as users would expect.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1492>
2022-03-24 00:01:20 +00:00
Seungha Yang
454e8f58a8 aom: av1enc: Specify Temporal Unit alignment
Encoded bitstream consists of leading Temporal delimiter OBU
with frame, that's Temporal Unit alignment.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1471>
2022-03-23 19:16:25 +00:00
Vivienne Watermeier
e6b187032b wpevideosrc: Add touch event support
Dispatches a list of active touch events to the wpe view on each
received TOUCH_FRAME event. Touch inputs currently only move the cursor,
since wpe doesn't seem to support clicking/scrolling or zooming with
touch input.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1633>
2022-03-23 13:14:52 +00:00
Vivienne Watermeier
6c2f6c3bd4 all: Use new navigation interface and API
Use and implement the new navigation interface in all relevant sink elements,
and use API functions everywhere instead of directy accessing the event structure.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1633>
2022-03-23 13:14:52 +00:00
Nirbheek Chauhan
8819350b74 openexr: Fix some warnings
```
../subprojects/gst-plugins-bad/ext/openexr/gstopenexrdec.cpp:46:24: warning: ‘Imf_3_1::Int64’ is deprecated: use uint64_t [-Wdeprecated-declarations]
   46 |   virtual Int64 tellg ();
      |                        ^
In file included from ../subprojects/gst-plugins-bad/ext/openexr/gstopenexrdec.cpp:32:
/usr/include/OpenEXR/ImfInt64.h:23:32: note: declared here
   23 | typedef IMATH_NAMESPACE::Int64 Int64;
      |                                ^~~~~
../subprojects/gst-plugins-bad/ext/openexr/gstopenexrdec.cpp:47:32: warning: ‘Imf_3_1::Int64’ is deprecated: use uint64_t [-Wdeprecated-declarations]
   47 |   virtual void seekg (Int64 pos);
      |                                ^
In file included from ../subprojects/gst-plugins-bad/ext/openexr/gstopenexrdec.cpp:32:
/usr/include/OpenEXR/ImfInt64.h:23:32: note: declared here
   23 | typedef IMATH_NAMESPACE::Int64 Int64;
      |                                ^~~~~
../subprojects/gst-plugins-bad/ext/openexr/gstopenexrdec.cpp:67:26: warning: ‘Imf_3_1::Int64’ is deprecated: use uint64_t [-Wdeprecated-declarations]
   67 | Int64 MemIStream::tellg ()
      |                          ^
In file included from ../subprojects/gst-plugins-bad/ext/openexr/gstopenexrdec.cpp:32:
/usr/include/OpenEXR/ImfInt64.h:23:32: note: declared here
   23 | typedef IMATH_NAMESPACE::Int64 Int64;
      |                                ^~~~~
../subprojects/gst-plugins-bad/ext/openexr/gstopenexrdec.cpp:73:29: warning: ‘Imf_3_1::Int64’ is deprecated: use uint64_t [-Wdeprecated-declarations]
   73 | MemIStream::seekg (Int64 pos)
      |                             ^
In file included from ../subprojects/gst-plugins-bad/ext/openexr/gstopenexrdec.cpp:32:
/usr/include/OpenEXR/ImfInt64.h:23:32: note: declared here
   23 | typedef IMATH_NAMESPACE::Int64 Int64;
      |                                ^~~~~
```

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1977>
2022-03-18 22:49:16 +00:00
Nirbheek Chauhan
253ee75a72 webrtcbin: Warn when offer didn't intersect with transceiver caps
We were silently falling back to creating a recvonly offer if the caps
didn't intersect.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1864>
2022-03-18 08:16:46 +00:00
Matthew Waters
098ff9a453 ccconverter: drop data with a warning if scratch buffers overflow
Instead of asserting which could bring down the entire application.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1972>
2022-03-17 21:46:44 +11:00
Philippe Normand
3e3ba1772c wpe: Reintroduce persistent WebContext
A WebContext leak was introduced in MR
https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/2252.
If we wanted one WebContext per WebView we should also unref the
WebKitWebContext when destroying the WebView.

This patch reintroduces the persistent WebContext, initially part of
https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1484.

Fixes #1084

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1933>
2022-03-16 09:07:21 +00:00
Mathieu Duponchelle
30d028317b webrtcbin: fix deadlock when setting up FEC encoder
We bind transceivers' fec_percentage property to the FEC encoder
percentage property, and with the binding bidirectional a deadlock
was introduced by the latest changes from !1762:

We take hold of the transceiver's object lock, then add the binding
and set the property to its initial value on the encoder, which causes
set_property to deadlock in the transceiver when the binding kicks in.

Changing the binding type to DEFAULT (source to target) is enough
to address the deadlock and still serves the original intent.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1967>
2022-03-16 06:06:39 +00:00
Sangchul Lee
2f7c843f2b webrtcbin: Check data channel transport for notifying 'ice-gathering-state'
Previously, it did not care about data channel's. It is fixed by adding
some conditions.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1957>
2022-03-16 03:31:08 +00:00
Matthew Waters
ccd1b76625 webrtcbin: fix ulpfecenc passthrough pt
ulpfecenc uses a value of pt=255 for passthrough.

Fixes: https://gitlab.freedesktop.org/gstreamer/gstreamer/-/issues/1075
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1914>
2022-03-10 16:20:03 +00:00
Matthew Waters
b7d0ddd1a4 webrtc: support renegotiating adding/removing RTX
We need to always add the RTX/RED/ULPFEC elements as rtpbin will only
call us once to request aux/fec senders/receivers.

We also need to regenerate the media section of the SDP instead of
blindly copying from the previous offer.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1762>
2022-03-04 19:21:59 +11:00
Sebastian Fricke
0b6bbce012 Remove the uninstalled term
Remove the symbolic link `gst-uninstalled` which points to `gst-env`.
The `uninstalled` is the old name and the project should stick to a
single name for the procedure.
Remove the term from all the files, exceptions are variables from
dependencies like `uninstalled_variables` from pkgconfig and
`meson-uninstalled`.
Adjust mentions of the script in the documentation and README.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1743>
2022-03-01 11:33:10 +00:00
Guillaume Desmottes
1f02f24828 gs: look for google_cloud_cpp_storage.pc
storage_client.pc was legacy and has been removed:
df6fa3611c (diff-bc35ad7c2fe631fd5578a06092412dba81c7ddd27bb25df7e17bb13771799afcL743)

No need to keep looking for storage_client.pc as a fallback as 1.25.0,
our minimum version, already ships google_cloud_cpp_storage.pc

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1815>
2022-03-01 08:10:39 +00:00
Sanchayan Maity
7c9a315578 ldac: Set eqmid in caps
We set the eqmid in caps to be usable downstream by rtpldacpay for
knowing the frame count.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1797>
2022-02-26 17:05:22 +05:30
Xavier Claessens
3d8372cc50 devenv: Add some missing GStreamer specific env variables
This should make "meson devenv" closer to what "gst-env.py" sets.

- GST_VALIDATE_SCENARIOS_PATH
- GST_VALIDATE_APPS_DIR
- GST_OMX_CONFIG_DIR
- GST_ENCODING_TARGET_PATH
- GST_PRESET_PATH
- GST_PLUGIN_SCANNER
- GST_PTP_HELPER
- _GI_OVERRIDES_PATH

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1768>
2022-02-25 20:35:26 +00:00
Jan Alexander Steffens (heftig)
e10bd02e1d fdkaacdec: Support arbitrary channel configs
Try to match the config to GStreamer positions. If something doesn't
fit, fall back to a set of unpositioned channels.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1561>
2022-02-25 18:20:52 +00:00
Jan Alexander Steffens (heftig)
d4b4ffc944 fdkaacdec: Use predefined channel layouts
This limits the decoder to the layouts predefined for the encoder
(including the MPEG standard layouts) but greatly simplifies the
implementation.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1561>
2022-02-25 18:20:52 +00:00
Nicolas Dufresne
bab9041c4b Port plugins to gst_video_format_info_extrapolate_stride()
This reduces code duplication and simplify addition of new
pixel formats into related plugins.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1567>
2022-02-20 22:32:55 +00:00
Jan Alexander Steffens (heftig)
10904e5580 wpe: Clean up build script
Use feature.require to check for gstgl and exit early if 'wpe' is
disabled (don't even check for wpe-webkit-1.1).

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1668>
2022-02-20 14:34:12 +00:00
Martin Reboredo
717009f8f5 vulkanshaderspv: SPIRV based filter
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1197>
2022-02-19 13:55:32 -03:00
Wojciech Kapsa
f8c0485af4 SVT-AV1 0.9 speed updated. 2022-02-18 07:01:57 +00:00
Tim Mooney
97720dabe0 meson: check for libsocket and libnsl
If present, add '-lsocket' and '-lnsl' to network_deps.

ext/curl/meson.build: add network_deps to dependencies
gst/festival/meson.build: same
sys/shm/meson.build: same

Fixes linking issues on Illumos distros.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1525>
2022-02-17 18:44:49 +00:00
Jan Alexander Steffens (heftig)
acd0300485 openaptx: Support libfreeaptx
[libfreeaptx][1] is a fork of libopenapt 0.2.0, used by pipewire.

[1]: https://github.com/iamthehorker/libfreeaptx

Fixes: https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/issues/1642
Closes: https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/issues/1589
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1667>
2022-02-15 08:18:44 +00:00
Sangchul Lee
dcff37722d webrtcice: Fix memory leaks in gst_webrtc_ice_add_candidate()
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1646>
2022-02-09 09:00:25 +00:00
Stéphane Cerveau
0600acd715 dashsink: doc cleanup
Remove max-files mention in the command line test
Fix some typos
Use mpegtsdemux instead of tsdemux in the pipeline description

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1624>
2022-02-02 10:21:08 +01:00
Jan Alexander Steffens (heftig)
16dc8f8442 wpe: Support wpe-webkit-1.1
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1522>
2022-01-31 08:31:34 +00:00
Philippe Normand
8e4cce6cd3 wpe: Install WebExtension in pkglibdir
The uninstalled WebExtension takes precedence over the installed one, so that
audio support works in local developer builds as well.

Fixes #975

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1602>
2022-01-31 00:54:10 +00:00
Philippe Normand
4254920b72 webrtc: Expose RTCError enum
The error codes not complying with the spec are now notified with the
GST_WEBRTC_ERROR_INTERNAL_FAILURE code.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1485>
2022-01-29 14:42:22 +00:00
Jakub Adam
bea8cba5e6 webrtcbin: Chain up to parent constructed method
Failing to do so makes GstWebRTCBin invisible to the leaks tracer.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1587>
2022-01-27 17:43:18 +00:00
Sangchul Lee
5cedf017f5 webrtc: Fix memory leaks
Redundant condition and unreachable codes are also removed.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1544>
2022-01-22 11:21:18 +00:00
Robert Mader
e7c9960783 waylandsink: Ensure correct mapping of area_surface
If the `area_surface` got unmapped when changing to the `READY` or
`NULL` state, we currently don't remap it when playback resumes and
`wp_viewporter` is supported. Without `wp_viewporter` we do remap
it, but rather unintentionally and also when not wanted.

On Weston this has not been a big problem as it so far wrongly maps
subsurfaces of unmapped surfaces anyway - i.e. only the black
background was missing on resume. On other compositors and future
Weston this prevents the `video_surface` to get remapped.

Shuffle things around to ensure `area_surface` is mapped in the
right situations and do some minor cleanup.

See also https://gitlab.freedesktop.org/wayland/weston/-/issues/426

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1483>
2022-01-17 13:17:57 +00:00
hassount
a85ab85bf8 Fix warning typos and Gstreamer ci test 2022-01-15 06:30:46 +00:00
Robert Mader
f0b04f1ef1 waylandsink: Use wl_surface_damage_buffer() instead of wl_surface_damage()
The later, doing damage in surface coordinates instead of buffer
coordinates, has been deprecated. The reason for that is that it
is more prone to bugs, both on the client and the compositor side,
especially when paired with buffer scale, `wp_viewporter` or
buffer transforms.

Unfortunately, on Weston this risks running into
https://gitlab.freedesktop.org/wayland/weston/-/issues/446
(which causes trouble for several other projects as well). However,
that bug only affects cases where we run in sync mode, i.e. only
during resizes. In practise I haven't been able to observe the
issue.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1446>
2022-01-13 19:39:59 +00:00
Robert Mader
1249362f96 waylandsink: Use G_MAXINT32 for surface damage
Each time we call `wl_surface_damage()` we want to do full surface
damage. Like Mesa, just use `G_MAXINT32` to ensure we always do
full damage, reducing the need to track the right dimensions.

`window->video_rectangle` is now unused, but we keep it around for
now as we may need it again in the future.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1446>
2022-01-13 19:39:59 +00:00
Robert Mader
3bbd091bb4 waylandsink: Only call wl_surface_damage() when buffer content changed
From the spec:
> This request is used to describe the regions where the pending
> buffer is different from the current surface contents

We currently also call `wl_surface_damage()` on surfaces without
new or still compositor-hold buffers, e.g. when resizing the window.
In that case we call it on `area_surface_wrapper`, even though it
gets resized via `wp_viewport_set_destination()`, in which case
the compositor is in charge of repainting the area on screen.

Doing so is currently not forbidden by the spec, however it might
be in the future, see
https://gitlab.freedesktop.org/wayland/wayland/-/issues/267

Thus lets stay close to the spec and only call `wl_surface_damage()`
when we just attached a buffer.

Right now this prevents runtime assertions in Mutter.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1446>
2022-01-13 19:39:59 +00:00
Robert Mader
b03c7edfcf waylandsink: Simplify input region handling
We only need to unset the input region for the area surface when
we don't have our own toplevel surface. By default, the input region
covers the whole surface, thus no need to change it on resize.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1446>
2022-01-13 19:39:59 +00:00
Robert Mader
1e2bc68171 waylandsink: Use G_MAXINT32 for opaque regions
`gst_wl_window_set_opaque` does not get called on window resizes,
potentially leaving opaque regions too small.
According to the spec opaque regions can be bigger than the surface
size - parts that fall outside of the surface will get ignored.
Thus we can can simply use `G_MAXINT32` and be sure that the whole
surfaces will always be covered.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1446>
2022-01-13 19:39:59 +00:00
Dave Piché
574cbbf0b5 webrtc: fix log error message in function gst_webrtc_bin_set_local_description
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1511>
2022-01-13 15:11:35 +00:00
Mathieu Duponchelle
d8c8737e71 cccombiner: fix s334-1a scheduling
The previous code was mistakenly trying to compute a cc_type out
of the first byte in the byte triplet, whereas it is to be interpreted
as:

> Bit b7 of the LINE value is the field number (0 for field 2; 1 for field 1).
> Bits b6 and b5 are 0. Bits b4-b0 form a 5-bit unsigned integer which
> represents the offset

The same mistake was made when creating padding packets.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1496>
2022-01-12 14:34:22 +00:00
Mathieu Duponchelle
6861ea8fe1 cccombiner: merge buffers for both fields with caption type s334-1a
Other elements such as line21encoder expect both fields to be present
in the same meta, not one meta per field.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1496>
2022-01-12 14:34:22 +00:00
Nirbheek Chauhan
1be6d6ccf5 meson: Add explicit check: kwarg to all run_command() calls
This is required since Meson 0.61.0, and causes a warning to be
emitted otherwise:

2c079d855e
https://github.com/mesonbuild/meson/issues/9300

This exposed a bunch of places where we had broken run_command()
calls, unnecessary run_command() calls, and places where check: true
should be used.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1507>
2022-01-09 18:12:47 +05:30
Christopher Degawa
b1d167ec22 rename unrestricted motion vector to restricted motion vector
Signed-off-by: Christopher Degawa <christopher.degawa@intel.com>
2022-01-06 16:46:53 -06:00
Christopher Degawa
a2ec26a8ca rename disable-dlf to enable-dlf
Signed-off-by: Christopher Degawa <christopher.degawa@intel.com>
2022-01-06 16:46:53 -06:00
Christopher Degawa
571897ef0c rename --tf-level to --enable-tf
Also does the renaming for the API and the config file option along
with changing the option to a EbBool

Signed-off-by: Christopher Degawa <christopher.degawa@intel.com>
2022-01-05 14:35:58 -06:00
Rafał Dzięgiel
8889b6351d assrender: Support RFC8081 mime types
Old "application/*" are now as per RFC8081 deprecated in favor of
new "font/*" mime types. Some new encoders are already using the
updated mime types. We need to also add them to the support list
in order for assrender to correctly identify them as fonts.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1481>
2022-01-03 06:42:23 +00:00
Rafał Dzięgiel
a2719d79ff assrender: Handle ".ttc" attachment extension
TTC stands for "TrueType Collection" file. We can pass it
into libass as any other attachment. Add it to the supported
extensions list, so the fonts it contains will be used.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1481>
2022-01-03 06:42:23 +00:00
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
Christopher Degawa
4bd83fb3c0 gstreamer: remove useless if 0
Signed-off-by: Christopher Degawa <christopher.degawa@intel.com>
2021-12-29 12:51:18 -06:00
hassount
4d99d86f99 Cleanup some API signals
API signaled (re)moved:

- intra_angle_delta
- palette_level
- 16-bit-pipeline
- sb_sz
- super_block_size
- partition_depth
- warp motion
- global motion
- self guided filter level
- weiner filter level
- inter intra compound
- paeth
- smooth
- mrp level
- spatial sse
- over_bndry_blk
- new_nearest_comb_inject
- nsq_table
- frame_end_cdf_update
- pred_me
- bipred_3x3_inject
- compound_level
- set_chroma_mode
- disable_cfl_flag
- obmc_level
- rdoq_level
- filter_intra_level
- enable_intra_edge_filter
- pic_based_rate_est
- unused me parameters
- unused me variables
- intrabc_mode
- enable_hbd_mode_decision
- enable_qp_scaling_flag
- ten_bit_format
- enable_adaptive_mini_gop
- max_heirachical_level
- speed_control_flag

Move unnecessary definitions to EbDefinitions.h
2021-12-29 12:51:12 -06:00
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
hassount
775f8fca11 Cleanup unused tokens and apis - Fix Lookahead distance documentation to be capped at 120 2021-12-17 23:09:45 -08:00
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
Tim-Philipp Müller
d4bdbeaa0d meson: add 'gpl' option and only build plugins with (A)GPL deps if explicitly enabled
Require explicit opt-in to build plugins with (A)GPL dependencies.

Keep ugly/bad options on 'auto' for now so cerbero doesn't fail.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1084>
2021-10-18 18:03:19 +01:00
Rob Agar
66a24023c0 webrtcbin: fix prevention of webrtcbin deletion due to ref held by probe callback
https://gitlab.freedesktop.org/gstreamer/gstreamer/-/issues/810

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1150>
2021-10-18 10:42:12 +01:00
Tim-Philipp Müller
e7c8b23b71 ofa: remove ofa audio fingerprinting plugin
I think the MusicIP database for this has been defunct for years,
so I can't imagine this plugin is particularly useful or still
used by anyone.

See https://musicbrainz.org/doc/Fingerprinting#PUID

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1153>
2021-10-17 10:17:00 +00:00
Thibault Saunier
a75382eaff meson: Mark newly fdkaac/ogg/vorbis as allow fallback
This way when the dep is `auto` we will fallback if the system
dependency is not available.

And use https to get libvorbis

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1171>
2021-10-16 09:52:23 -03:00
Thibault Saunier
0a781424e6 meson:avtp: Error out if sock_txtime is not present and avtp is enabled
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1142>
2021-10-14 01:01:54 +00:00
Stéphane Cerveau
2200a3dbcd ttml: fix log init
The log system should be init before calling a log

Fix regression after:

https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/2038

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1112>
2021-10-11 15:23:08 +02:00