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141 commits

Author SHA1 Message Date
Sebastian Dröge
4e86c77270 examples: webrtc: rust: Update dependencies 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3758>
 2023-01-20 11:36:57 +00:00
Sebastian Dröge
f45136827b examples: webrtc: multiparty-sendrecv: rust: Remove unnecessary macro recursion limit annotation 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3758>
 2023-01-20 11:36:57 +00:00
Sebastian Dröge
bf4a3c89cd examples: webrtc: sendrecv: rust: Implement OFFER_REQUEST handling 
Allow requesting an offer from the peer if we're joining a call with a
peer, and allow the peer to request an offer from us if waiting for an
incoming call.

This implements all 4 variants the protocol allows for.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3758>
 2023-01-20 11:36:57 +00:00
Sebastian Dröge
638465908e examples: webrtc: sendrecv: rust: Allow providing our ID via the commandline 
Otherwise it continues to use a random ID as before.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3758>
 2023-01-20 11:36:57 +00:00
Sebastian Dröge
541c637910 examples: webrtc: sendrecv: rust: Implement TWCC support in both directions 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3758>
 2023-01-20 11:36:57 +00:00
Sebastian Dröge
6541dccaea examples: webrtc: rust: Set keyframe-max-dist=2000 and picture-id-mode=15-bit for VP8 and perfect-timestamps=true for audio 
This makes it in sync with the C sendrecv and generally behaves better.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3758>
 2023-01-20 11:36:57 +00:00
Sebastian Dröge
083b9f2a6e examples: webrtc: sendrecv: rust: Use the correct payload types if the remote is the offerer 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3758>
 2023-01-20 11:36:57 +00:00
Sebastian Dröge
ac1d10f80c gst-examples: Update Rust dependencies 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3750>
 2023-01-19 10:40:32 +02:00
Tim-Philipp Müller
a9ec35b1ca Release 1.21.90 2023-01-13 19:08:48 +00:00
Sebastian Dröge
085e6c036a android: Update minimum SDK version to Android 21 
Otherwise we can't bump the minimum version of the cerbero build without
it breaking linking of the applications.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3717>
 2023-01-12 20:11:14 +00:00
Olivier Crête
b7c0e8bc84 webrtc examples: Force regular non-MULTIOPUS 
Using MULTIOPUS breaks with most browsers

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3675>
 2023-01-04 12:02:25 +00:00
Olivier Crête
c7bc6bc064 webrtc-unidirectional: Avoid critical 
Don't unref the parameter passed to a signal, it's always owned by
the caller. Fixes a GLib critical.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3675>
 2023-01-04 12:02:25 +00:00
Sebastian Dröge
c739fcbe41 examples: webrtc: Add handling of the LATENCY messages to the Rust examples 
Without this the configured latency on the pipeline will be wrong.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3609>
 2022-12-20 13:10:27 +02:00
Sebastian Dröge
284d22437e examples: webrtc: Update dependencies 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3609>
 2022-12-20 13:06:43 +02:00
Sebastian Dröge
ec6290d63f examples: webrtc: Remove the bus watch at the end 
Otherwise a file descriptor will be leaked.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3609>
 2022-12-20 13:03:44 +02:00
Sebastian Dröge
1f4f338d85 examples: webrtc: Add handling of the LATENCY messages to the C examples 
Without this the configured latency on the pipeline will be wrong.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3609>
 2022-12-20 13:03:15 +02:00
Sebastian Dröge
d10981f7b9 examples: webrtc: Add bus handling to the Android and C sendrecv examples 
Without a bus, messages will just pile up and errors are not handled at
all. Also without handling the LATENCY messages the latency configured
on the pipeline will be wrong.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3609>
 2022-12-20 13:02:08 +02:00
Seungmin Kim
0db1ff532d Change GstSdp.sdp_message_parse_buffer to GstSdp.SDPMessage.new_from_text in examples 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3477>
 2022-12-16 10:40:41 +00:00
Nirbheek Chauhan
7fd8e4001c webrtc/signalling: Give a helpful error when starting a double-session 
If the peer is already in a session and tries to start a new one, give
them a helpful error.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2460>
 2022-12-12 15:08:23 +00:00
byran77
1e5abde7b1 gst-examples: webrtc: signalling: simple-server Fix condition when calling a busy peer 
When a session request is coming in, ERROR occurs when the callee is busy.
But peer_status is the status of the caller, which is of course None when
calling someone, while self.peers[callee_id][2] is that of the callee.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2460>
 2022-12-12 15:08:23 +00:00
Guillaume Desmottes
cbab7ffefb examples: webrtc: fix unidirectional pipeline 
'autoaudiosrc' does not have a 'is-live' property.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3550>
 2022-12-09 13:49:44 +01:00
Guillaume Desmottes
ebfbdf9076 examples: webrtc: fix plugins check 
`videoconvert` and `videoscale` are now part of the `videoconvertscale`
plugin, see d11f13f476

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3529>
 2022-12-05 17:04:57 +00:00
Tim-Philipp Müller
1f65d7cc5c Back to development 2022-12-05 02:29:08 +00:00
Tim-Philipp Müller
fd6a3948c6 Release 1.21.3 2022-12-05 01:28:21 +00:00
Jan Schmidt
8177588250 examples/sendrecv: Remove extra unref of webrtcbin 
The code now constructs webrtcbin with a floating ref and then
gives it to the pipeline. The extra unref is one too many.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3436>
 2022-11-19 19:51:54 +11:00
Jan Schmidt
f2ae481a69 examples/webrtc: Configure payload types 
MR 2398 broke the webrtc sendrecv example
by not configuring the payload types, so both audio and video streams
get sent on payload 96.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3434>
 2022-11-19 13:12:58 +11:00
Tim-Philipp Müller
db450689db Back to development 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3358>
 2022-11-08 02:08:08 +00:00
Tim-Philipp Müller
3e29ac35c4 Release 1.21.2 2022-11-07 23:54:03 +00:00
Nicolas Dufresne
4fb9f2a2b4 meson: Fix path for webrtc validate tests 
This fixes a crash when trying to run gst-validate-launcher from inside
the meson devenv. The error was:

  ModuleNotFoundError: No module named 'observer'

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3273>
 2022-10-26 18:16:25 +00:00
Patrick Griffis
2a59e8af97 webrtc: Fix double free in webrtc-recvonly-h264 demo 
The "message" signal does not transfer ownership of the GBytes passed
to it so calling g_bytes_unref() on it is incorrect.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3257>
 2022-10-24 22:16:44 +00:00
Sebastian Dröge
7193a601b3 examples: webrtc: Update to gstreamer-rs 0.19 release 
Also update the macOS workaround for gstreamer-gl requiring a
`NSRunLoop` / `NSApp` on the main thread, and update from strucopt to
clap 4.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3255>
 2022-10-24 11:50:09 +00:00
Patrick Griffis
d0e2b31470 webrtc: Fix critical in webrtc-recvonly-h264 example 
This signal only takes 2 properties yet a third was passed.
This would cause a critical in GLib.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3252>
 2022-10-23 22:51:28 +00:00
Sam Van Den Berge
094b251901 examples: webrtc: mp-sendrecv: add bus handler 
Without this bus handler, messages posted to the bus will keep a ref to
their source elements, preventing them from being disposed.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3219>
 2022-10-19 00:51:44 +00:00
Sam Van Den Berge
93ed51cbb2 examples: webrtc: mp-sendrecv: set element states to NULL after removing them from pipeline 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3219>
 2022-10-19 00:51:44 +00:00
Sam Van Den Berge
17c111d2b9 examples: webrtc: mp-sendrecv: remove wrong gst_object_unrefs 
In !2958 some gst_object_unrefs were added. However these two don't
belong there because ownership is transfered due to the gst_bin_add_many
call a bit above.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3219>
 2022-10-19 00:51:44 +00:00
Sebastian Dröge
64c376b5b2 webrtc: Add/fix various annotations 
And mark string parameters as const.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3194>
 2022-10-18 08:56:58 +00:00
Matthew Waters
d586c2cc28 examples/webrtc: don't use factory_make_full() for enums 
They are not currently translated into their respective enum values and
will produce an error.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3210>
 2022-10-18 01:30:37 +00:00
Sam Van Den Berge
07d8e53aac examples/webrtc/signalling: Fix compatibility with Python 3.10 
Fix asyncio throwing a deprecation warning when using
asyncio.get_event_loop().

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3196>
 2022-10-17 11:46:51 +02:00
Tim-Philipp Müller
99bbb1ef35 Back to development 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3115>
 2022-10-04 03:57:31 +01:00
Tim-Philipp Müller
9820e58be6 Release 1.21.1 2022-10-04 01:18:20 +01:00
Nirbheek Chauhan
6a3319c8f2 examples: Support multiple video streams in JS webrtc sendrecv 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3079>
 2022-09-27 19:48:56 +00:00
Stéphane Cerveau
0c96e838e8 docs: update to mono repo locations 
Some links/repos in the documentation were still pointing to old
repositories, change to mono repository

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2982>
 2022-09-06 14:20:49 +02:00
Nirbheek Chauhan
a5ef7e2781 examples: Disable bitcode support in iOS examples 
For explanation, see:

https://gitlab.freedesktop.org/gstreamer/cerbero/-/merge_requests/879

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2978>
 2022-09-03 18:04:34 +05:30
Sebastian Dröge
ad6ba10ae3 examples: webrtc: mp-sendrecv: Add missing semicolon 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2961>
 2022-08-31 10:57:39 +03:00
Alireza Miryazdi
eab9383812 examples/webrtc: add some missing unrefs 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2958>
 2022-08-31 05:07:52 +04:30
Seungha Yang
356a0ce8f4 examples: gtk: Fix MSVC build 
gtk-play.c(370): error C2375: 'rewind_button_clicked_cb': redefinition;
different linkage

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2871>
 2022-08-17 13:31:55 +00:00
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
Matthew Waters
b06a97c429 examples/webrtc/signalling: Fix compatibility with Python 3.10 
- ssl module requires an explicit TLS_SERVER role
- asyncio throws a deprecation warning when using
  asyncio.get_event_loop().  Remove custom event loop handling entirely
- No need to keep the websocket server in a member variable, can use
  a future to signal exit case along with the async with context manager
  of websockets.serve()

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2698>
 2022-07-04 03:17:15 +00:00
Stéphane Cerveau
a5cd1adc97 gst-examples: continue if webrtc deps are not satisfied 
The WebRTC examples are disabled if one dependency is
not satisfied, especially libsoup.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2261>
 2022-05-14 09:49:33 +00:00
Stéphane Cerveau
5c46099b8d gst-examples: change include for gtk in gtk-play 
The gdk.h is not necessary but the gtk.h yes for
the GTK_CHECK_VERSION macro.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2261>
 2022-05-14 09:49:33 +00:00
Thibault Saunier
f90916f4bb docs:examples: Fix building iOS examples 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2264>
 2022-04-21 09:18:56 -04:00
Stéphane Cerveau
5020a6e83a gst-examples: add -xobjective-c for darwin 
To avoid error when including #include <AppKit/AppKit.h>
the  build should include the c args xobjective-c

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2095>
 2022-04-04 16:27:45 +00:00
Zhao Zhili
dd27850a9a examples: fix build on macOS with gtk+-quartz-3.0 
gdk_quartz_window_get_nsview is not declared in the header file now:
error: implicit declaration of function 'gdk_quartz_window_get_nsview'
is invalid in C99 [-Werror,-Wimplicit-function-declaration]

fixes #979

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2095>
 2022-04-04 16:27:45 +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
Matthew Waters
4a2e0c1f4c examples/player/android: add missing dummy.cpp 
allows libc++_shared.so to be placed in the application

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2043>
 2022-03-28 14:38:24 +11:00
Matthew Waters
5d2d03892d examples/player/android: update for android changes 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2041>
 2022-03-28 02:32:41 +00:00
Sebastian Dröge
d2ecce5862 webrtc: Update dependencies of the Rust examples 
And also clean up code a bit while updating to new APIs.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2016>
 2022-03-24 12:05:29 +02:00
Nirbheek Chauhan
4ae903d383 webrtc_sendrecv.py: Link pads instead of elements 
This was not a problem here because even if we end up accidentally
linking to the wrong pad, things will work out eventually as long as
one pad-added is emitted for each pad that is added.

But it will be a huge problem if someone copies this code and changes
something that requires different handling for different sorts of
pads. The resultant code will be racy. Let's not do this, it's a bad
example.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2008>
 2022-03-23 21:04:39 +00:00
Nirbheek Chauhan
0007fa38e0 webrtc-sendrecv: Fix create-answer caps negotiation 
We need to parse the payload type map provided by the offer SDP and
set those values on the payloader, otherwise webrtcbin will create
a recvonly answer SDP and we won't send anything to the browser.

Fixed it for both C and Python sendrecv examples.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1864>
 2022-03-18 08:16:46 +00:00
Nirbheek Chauhan
3c0d582b7c webrtc_sendrecv.py: Add picture-id-mode to rtpvp8pay 
This doesn't just make TWCC stats perform better, it also fixes
stuttery video playback in Chrome.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1864>
 2022-03-18 08:16:46 +00:00
Nirbheek Chauhan
e0378f9913 webrtc_sendrecv.py: Print an error on unknown JSON message 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1864>
 2022-03-18 08:16:46 +00:00
Nirbheek Chauhan
90da0e1d1e webrtc_sendrecv.py: Add missing copyright headers 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1864>
 2022-03-18 08:16:46 +00:00
Nirbheek Chauhan
583408c312 webrtc_sendrecv.py: Implement all negotiation modes 
Earlier, the example only supported one negotiation mode:
* Browser client is running, gstreamer starts a call and sends offer

Now these three modes are also supported:
* Browser client is running, gstreamer starts a call and sends an
  offer request
* gstreamer connects and waits for browser client to start a call and
  send an offer
* gstreamer connects and waits for browser client to start a call and
  send an offer request

The following features are still missing:
* Data channel support
* TWCC support + stats logging

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1864>
 2022-03-18 08:16:46 +00:00
Nirbheek Chauhan
0b1438cc97 webrtc_sendrecv.py: Make it executable 
Why wasn't it already. Tired of typing 'python webrtc_sendrecv.py'.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1864>
 2022-03-18 08:16:45 +00:00
Nirbheek Chauhan
2294356d9b webrtc_sendrecv.py: Fix event loop usage for messages 
Instead of creating a new loop, we should just be fetching the running
loop, then doing a blocking network call inside the callback, schedule
it on the event loop. This is what the C example does too.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1864>
 2022-03-18 08:16:45 +00:00
Nirbheek Chauhan
ee4ca699a0 webrtc_sendrecv.py: Use default arg for server URL 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1864>
 2022-03-18 08:16:45 +00:00
Nirbheek Chauhan
4de365b31c webrtc_sendrecv.py: Sync element props with C version 
Also add indentation to make it easier to read

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1821>
 2022-03-01 16:33:28 +00:00
Nirbheek Chauhan
5ca5a83e75 webrtc_sendrecv.py: Ensure that gst-python overrides are installed 
Otherwise fetching of the offer will fail with a cryptic error:

```
Traceback (most recent call last):
  File "/../gstreamer/subprojects/gst-examples/webrtc/sendrecv/gst/webrtc_sendrecv.py", line 56, in on_offer_created
    offer = reply['offer']
TypeError: 'Structure' object is not subscriptable
```

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1821>
 2022-03-01 16:33:28 +00:00
Nirbheek Chauhan
e9a02a7380 webrtc_sendrecv.py: Don't try to set state on a None pipe 
```
ERROR peer '5762' not found
Traceback (most recent call last):
  File "/../gstreamer/subprojects/gst-examples/webrtc/sendrecv/gst/webrtc_sendrecv.py", line 190, in <module>
    res = loop.run_until_complete(c.loop())
  File "/usr/lib64/python3.10/asyncio/base_events.py", line 641, in run_until_complete
    return future.result()
  File "/../gstreamer/subprojects/gst-examples/webrtc/sendrecv/gst/webrtc_sendrecv.py", line 155, in loop
    self.close_pipeline()
  File "/../gstreamer/subprojects/gst-examples/webrtc/sendrecv/gst/webrtc_sendrecv.py", line 142, in close_pipeline
    self.pipe.set_state(Gst.State.NULL)
AttributeError: 'NoneType' object has no attribute 'set_state'
```

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1821>
 2022-03-01 16:33:28 +00:00
Nirbheek Chauhan
78f8505b9a webrtc_sendrecv.py: Fix SSLError when connecting to websocket server 
```
  File "/../gstreamer/subprojects/gst-examples/webrtc/sendrecv/gst/webrtc_sendrecv.py", line 189, in <module>
    loop.run_until_complete(c.connect())
  File "/usr/lib64/python3.10/asyncio/base_events.py", line 641, in run_until_complete
    return future.result()
  File "/../gstreamer/subprojects/gst-examples/webrtc/sendrecv/gst/webrtc_sendrecv.py", line 40, in connect
    self.conn = await websockets.connect(self.server, ssl=sslctx)
  File "/home/nirbheek/.local/lib/python3.10/site-packages/websockets/legacy/client.py", line 650, in __await_impl_timeout__
    return await asyncio.wait_for(self.__await_impl__(), self.open_timeout)
  File "/usr/lib64/python3.10/asyncio/tasks.py", line 445, in wait_for
    return fut.result()
  File "/home/nirbheek/.local/lib/python3.10/site-packages/websockets/legacy/client.py", line 654, in __await_impl__
    transport, protocol = await self._create_connection()
  File "/usr/lib64/python3.10/asyncio/base_events.py", line 1080, in create_connection
    transport, protocol = await self._create_connection_transport(
  File "/usr/lib64/python3.10/asyncio/base_events.py", line 1110, in _create_connection_transport
    await waiter
  File "/usr/lib64/python3.10/asyncio/sslproto.py", line 631, in _on_handshake_complete
    raise handshake_exc
  File "/usr/lib64/python3.10/asyncio/sslproto.py", line 676, in _process_write_backlog
    ssldata = self._sslpipe.do_handshake(
  File "/usr/lib64/python3.10/asyncio/sslproto.py", line 116, in do_handshake
    self._sslobj = self._context.wrap_bio(
  File "/usr/lib64/python3.10/ssl.py", line 526, in wrap_bio
    return self.sslobject_class._create(
  File "/usr/lib64/python3.10/ssl.py", line 865, in _create
    sslobj = context._wrap_bio(
ssl.SSLError: Cannot create a client socket with a PROTOCOL_TLS_SERVER context (_ssl.c:801)
```

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1821>
 2022-03-01 16:33:28 +00:00
Nirbheek Chauhan
e453e43e5a webrtc_sendrecv.py: Fix deprecation warning with Python 3.10 
asyncio.get_event_loop() will not implicitly create a new event loop
in a future version of Python, so we need to do that explicitly.

```
webrtc_sendrecv.py:188: DeprecationWarning: There is no current event loop
  loop = asyncio.get_event_loop()
```

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1821>
 2022-03-01 16:33:28 +00:00
Nirbheek Chauhan
4c2fd7f104 webrtc_sendrecv.py: Fix styling errors 
These are now enforced by the pre-commit python style hook.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1821>
 2022-03-01 16:33:28 +00:00
Nirbheek Chauhan
d6799b069a webrtc: Update Makefile for building webrtc-sendrecv 
This now needs the RTP library.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1821>
 2022-03-01 16:33:28 +00:00
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
Matthew Waters
2f9ad38118 examples/docs: update android applications to the latest relevant sdk's and build tools 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1622>
 2022-02-07 06:51:58 +00:00
Tim-Philipp Müller
701ed92d27 Back to development 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1635>
 2022-02-04 22:59:41 +00:00
Tim-Philipp Müller
f0b045a69b Release 1.20.0 2022-02-03 20:03:15 +00:00
Tim-Philipp Müller
31b5243e1d Release 1.19.90 2022-01-28 14:28:42 +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
Olivier Crête
27808444ea webrtc janus rust: Update extra dependencies 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1332>
 2021-11-10 16:13:38 +00:00
Olivier Crête
5ab09323cd webrtc multiparty rust: Upgrade all other deps 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1332>
 2021-11-10 16:13:38 +00:00
Olivier Crête
c592f75cdd webrtc sendrecv rust: Upgrade all other deps 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1332>
 2021-11-10 16:13:38 +00:00
Olivier Crête
26e8624c9b webrtc multiparty rust: Port to bindings 0.17 version 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1332>
 2021-11-10 16:13:38 +00:00
Olivier Crête
93032b2ecc webrtc sendrecv rust: Port to bindings 0.17 version 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1332>
 2021-11-10 16:13:38 +00:00
Tim-Philipp Müller
a7b376011b Back to development 2021-11-03 19:31:23 +00:00
Tim-Philipp Müller
f513c289b0 Release 1.19.3 2021-11-03 15:43:43 +00:00
Tim-Philipp Müller
8dfab0b08c meson: update for meson.build_root() and .build_source() deprecation 
-> use meson.project_build_root() or .global_build_root() instead.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1183>
 2021-10-20 11:20:44 +00:00
Tim-Philipp Müller
c95a7ab1ca docs: remove mention of mms from docs and mmssrc from tutorials 
Prerequisite for our removing it from the cerbero build, see
https://gitlab.freedesktop.org/gstreamer/cerbero/-/merge_requests/760

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1199>
 2021-10-20 00:40:22 +00:00
Tim-Philipp Müller
f1a169f39d Back to development 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/925>
 2021-09-26 01:07:02 +01:00
Thibault Saunier
c6b9c81fdd ci: Remove now useless .gitlab-ci.yml files 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/891>
 2021-09-24 16:21:18 -03:00
Thibault Saunier
41ed155bdf Move files from gst-examples into the "subprojects/gst-examples/" subdir 2021-09-24 16:15:58 -03:00