Commit graph

19 commits

Author SHA1 Message Date
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
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
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
Thibault Saunier
41ed155bdf Move files from gst-examples into the "subprojects/gst-examples/" subdir 2021-09-24 16:15:58 -03:00