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451 commits

Author SHA1 Message Date
Sebastian Dröge
5bc92375c9 svtav1enc: Fix indentation 2023-02-03 22:14:18 +02:00
Sebastian Dröge
7890a1f8c7 svtav1: Integrate into the build system properly 2023-02-03 22:14:18 +02:00
Sebastian Dröge
b15efacf84 svtav1: Merge SVT-AV1 encoder into gst-plugins-bad 
This is based on d5e1e2a586020854733f6b0806064d0c900c88d2 from
https://gitlab.com/AOMediaCodec/SVT-AV1.
 2023-02-03 22:13:30 +02:00
Tim-Philipp Müller
d95d3e39af cc708overlay: bump pango requirement and drop no longer required locking 
Gets rid of GSlice allocation that's never freed.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3784>
 2023-02-03 17:48:10 +00:00
Sebastian Dröge
8aa376d541 gstreamer: Decide rate-control-mode based on the bitrate/cqp/crf settings 
And also keep the default encoder settings but simply override them with
our own values that we care about.

This mirrors the encoder configuration behaviour from ffmpeg.
 2023-02-03 12:48:56 +02:00
Sebastian Dröge
70c83bc59c gstreamer: Use GLib types instead of stdint.h types consistently 2023-02-03 12:48:56 +02:00
Sebastian Dröge
dd1db338df gstreamer: Fix double unref 
The ownership of the caps is passed to `gst_video_encoder_set_output_state()`.
 2023-02-03 12:48:56 +02:00
Sebastian Dröge
df064c6cc2 gstreamer: Configure colorimetry and HDR metadata if present 
This raises the minimum GStreamer requirement to 1.16 as used by the CI
and optionally makes use of 1.18 features, including HDR.
 2023-02-03 12:48:25 +02:00
Sebastian Dröge
aeee6f5b6a gstreamer: Set correct maximum width/height limits 2023-02-03 12:35:24 +02:00
Sebastian Dröge
65eb56e7ad gstreamer: Use correct 10-bit format on big endian systems 2023-02-03 12:35:24 +02:00
Sebastian Dröge
90fd191392 gstreamer: Set force_key_frames=true in CQP/CRF mode 
Other modes don't support that so keyframes can't be requested at
arbitrary times.
 2023-02-03 12:35:24 +02:00
Sebastian Dröge
f24643b48f gstreamer: Add support for setting arbitrary parameters via parameters-string property 2023-02-03 12:35:18 +02:00
Sebastian Dröge
d746164ba0 gstreamer: Fix naming of function name that was taken over from the SVT-HEVC encoder 2023-02-03 12:34:38 +02:00
Sebastian Dröge
ddb9a037e1 gstreamer: Don't overwrite application configuration on initialization and initialize with the default configuration 2023-02-03 12:34:38 +02:00
Sebastian Dröge
7b1b33aff2 gstreamer: Clean up property handling 
Use more correct types, defaults and clean up property names a bit.
This now matches the configuration provided by ffmpeg.
 2023-02-03 12:34:38 +02:00
Sebastian Dröge
78ee7e82d9 gstreamer: Mark all internal functions as static 2023-02-03 12:34:38 +02:00
Sebastian Dröge
10769e7fe6 gstreamer: Fix encoder and buffer state life cycle 
Allocate/deallocate the encoder in `open()`/`close()` and its buffers in
`start()` / `stop()`.

Also fail correctly if configuring the encoder fails.
 2023-02-03 12:34:38 +02:00
Sebastian Dröge
24d6027d2e gstreamer: Remove unused frame_count and dts_offset 2023-02-03 12:34:38 +02:00
Sebastian Dröge
2d250439f1 gstreamer: Fix debug category description 2023-02-03 12:34:38 +02:00
Adrian Fiergolski
79d2af5626 avtp: rvf: add missing since markers 
Add missing markers.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1335>
 2023-02-02 19:15:46 +01:00
Adrian Fiergolski
9f880b37fc avtp: rvf: add AVTP RVF de-payload support 
Add AVTP Raw Video Format de-payload support. The element supports only
GRAY16_LE output 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 19:15:42 +01:00
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