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155 commits

Author SHA1 Message Date
Philippe Normand
906b90287c webrtcbin: Relay add-ice-candidate errors from Ice implementation to Application 
The `add_candidate` vfunc of the GstWebRTCICE interface gained a GstPromise
argument, which is an ABI break. We're not aware of any external user of this
interface yet so we think it's OK.

This change is useful in cases where the application needs to bubble up errors
from the underlying ICE agent, for instance when the agent was given an invalid
ICE candidate.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3960>
 2023-02-27 09:09:47 +00:00
Thibault Saunier
0f577533e6 examples: Add an option to disable tests 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3930>
 2023-02-10 12:59:55 +00:00
Sebastian Dröge
fc5bad5f75 examples: webrtc: rust: Fix a couple of minor clippy warnings 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3928>
 2023-02-10 11:43:00 +00:00
Sebastian Dröge
28ab612a88 examples: webrtc: rust: Update to gstreamer-rs 0.20 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3928>
 2023-02-10 11:43:00 +00:00
Nirbheek Chauhan
033a71e405 webrtc examples: Use webrtc.gstreamer.net 
Actually just a CNAME to webrtc.nirbheek.in for now, but it allows
replacement / hosting without my involvement, so reduces the bus
factor.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3802>
 2023-02-04 13:37:02 +00:00
Tim-Philipp Müller
06e9d78ade gst-examples: drop use of GSlice allocator 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3784>
 2023-02-03 17:48:09 +00:00
Matthew Waters
b134433e0b examples/webrtc-sendrecv: add some dot file dumps on async-done and error messages 
Just as a helpful thing if debugging is needed

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3823>
 2023-01-30 05:22:59 +00:00
Nirbheek Chauhan
32e8ff4e2a webrtc_sendrecv.py: Fix PEP8 warnings in CI lint 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3742>
 2023-01-25 16:53:17 +00:00
Nirbheek Chauhan
6a83602601 webrtc_sendrecv.py: Handle LATENCY messages 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3742>
 2023-01-25 16:53:17 +00:00
Nirbheek Chauhan
5500c228f6 webrtc_sendrecv.py: Add bus message handling 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3742>
 2023-01-25 16:53:17 +00:00
Nirbheek Chauhan
9b2404e76d webrtc_sendrecv.py: Add support for using H264 encoding 
Currently only works when we are creating the offer or the offer only
contains H264.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3742>
 2023-01-25 16:53:17 +00:00
Nirbheek Chauhan
6f99faa080 webrtc_sendrecv.py: Use sine wave for audio instead of red-noise 
Makes it easier to notice when there's packet loss or other audio
distortion.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3742>
 2023-01-25 16:53:17 +00:00
Tim-Philipp Müller
41c69372b5 Back to development 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3775>
 2023-01-23 23:04:53 +00:00
Tim-Philipp Müller
f13c65d977 Release 1.22.0 2023-01-23 19:41:07 +00:00
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