webrtc_sendrecv.py: Fix PEP8 warnings in CI lint

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3816>

diff --git a/subprojects/gst-examples/webrtc/sendrecv/gst/webrtc_sendrecv.py b/subprojects/gst-examples/webrtc/sendrecv/gst/webrtc_sendrecv.py
index 5f72485..6d5cb03 100755 (executable)
--- a/subprojects/gst-examples/webrtc/sendrecv/gst/webrtc_sendrecv.py
+++ b/subprojects/gst-examples/webrtc/sendrecv/gst/webrtc_sendrecv.py
@@ -6,6 +6,10 @@
 # Demo gstreamer app for negotiating and streaming a sendrecv webrtc stream
 # with a browser JS app, implemented in Python.
 
 # Demo gstreamer app for negotiating and streaming a sendrecv webrtc stream
 # with a browser JS app, implemented in Python.
 

+from websockets.version import version as wsv
+from gi.repository import GstSdp
+from gi.repository import GstWebRTC
+from gi.repository import Gst

 import random
 import ssl
 import websockets
 import random
 import ssl
 import websockets


@@ -17,11 +21,8 @@ import argparse
 
 import gi
 gi.require_version('Gst', '1.0')
 
 import gi
 gi.require_version('Gst', '1.0')

-from gi.repository import Gst

 gi.require_version('GstWebRTC', '1.0')
 gi.require_version('GstWebRTC', '1.0')

-from gi.repository import GstWebRTC

 gi.require_version('GstSdp', '1.0')
 gi.require_version('GstSdp', '1.0')

-from gi.repository import GstSdp

 
 # Ensure that gst-python is installed
 try:
 
 # Ensure that gst-python is installed
 try:


@@ -53,8 +54,6 @@ PIPELINE_DESC = {
     'VP8': PIPELINE_DESC_VP8,
 }
 
     'VP8': PIPELINE_DESC_VP8,
 }
 

-from websockets.version import version as wsv
-

 
 def print_status(msg):
     print(f'--- {msg}')
 
 def print_status(msg):
     print(f'--- {msg}')


@@ -159,7 +158,7 @@ class WebRTCClient:
         self.send_soon(msg)
 
     def on_offer_created(self, promise, _, __):
         self.send_soon(msg)
 
     def on_offer_created(self, promise, _, __):

-        assert(promise.wait() == Gst.PromiseResult.REPLIED)
+        assert promise.wait() == Gst.PromiseResult.REPLIED

         reply = promise.get_reply()
         offer = reply['offer']
         promise = Gst.Promise.new()
         reply = promise.get_reply()
         offer = reply['offer']
         promise = Gst.Promise.new()


@@ -235,7 +234,7 @@ class WebRTCClient:
         self.pipe.set_state(Gst.State.PLAYING)
 
     def on_answer_created(self, promise, _, __):
         self.pipe.set_state(Gst.State.PLAYING)
 
     def on_answer_created(self, promise, _, __):

-        assert(promise.wait() == Gst.PromiseResult.REPLIED)
+        assert promise.wait() == Gst.PromiseResult.REPLIED

         reply = promise.get_reply()
         answer = reply['answer']
         promise = Gst.Promise.new()
         reply = promise.get_reply()
         answer = reply['answer']
         promise = Gst.Promise.new()


@@ -244,7 +243,7 @@ class WebRTCClient:
         self.send_sdp(answer)
 
     def on_offer_set(self, promise, _, __):
         self.send_sdp(answer)
 
     def on_offer_set(self, promise, _, __):

-        assert(promise.wait() == Gst.PromiseResult.REPLIED)
+        assert promise.wait() == Gst.PromiseResult.REPLIED

         promise = Gst.Promise.new_with_change_func(self.on_answer_created, None, None)
         self.webrtc.emit('create-answer', None, promise)
 
         promise = Gst.Promise.new_with_change_func(self.on_answer_created, None, None)
         self.webrtc.emit('create-answer', None, promise)
 


@@ -270,17 +269,17 @@ class WebRTCClient:
                 if not self.webrtc:
                     print_status('Incoming call: received an offer, creating pipeline')
                     pts = get_payload_types(sdpmsg, video_encoding=self.video_encoding, audio_encoding='OPUS')
                 if not self.webrtc:
                     print_status('Incoming call: received an offer, creating pipeline')
                     pts = get_payload_types(sdpmsg, video_encoding=self.video_encoding, audio_encoding='OPUS')

-                    assert(self.video_encoding in pts)
-                    assert('OPUS' in pts)
+                    assert self.video_encoding in pts
+                    assert 'OPUS' in pts

                     self.start_pipeline(create_offer=False, video_pt=pts[self.video_encoding], audio_pt=pts['OPUS'])
 
                     self.start_pipeline(create_offer=False, video_pt=pts[self.video_encoding], audio_pt=pts['OPUS'])
 

-                assert(self.webrtc)
+                assert self.webrtc

 
                 offer = GstWebRTC.WebRTCSessionDescription.new(GstWebRTC.WebRTCSDPType.OFFER, sdpmsg)
                 promise = Gst.Promise.new_with_change_func(self.on_offer_set, None, None)
                 self.webrtc.emit('set-remote-description', offer, promise)
         elif 'ice' in msg:
 
                 offer = GstWebRTC.WebRTCSessionDescription.new(GstWebRTC.WebRTCSDPType.OFFER, sdpmsg)
                 promise = Gst.Promise.new_with_change_func(self.on_offer_set, None, None)
                 self.webrtc.emit('set-remote-description', offer, promise)
         elif 'ice' in msg:

-            assert(self.webrtc)
+            assert self.webrtc

             ice = msg['ice']
             candidate = ice['candidate']
             sdpmlineindex = ice['sdpMLineIndex']
             ice = msg['ice']
             candidate = ice['candidate']
             sdpmlineindex = ice['sdpMLineIndex']