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>
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>
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>
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>
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>
```
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>
```
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>
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>
