webrtc_sendrecv.py: Fix PEP8 warnings in CI lint

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3816>
This commit is contained in:
Nirbheek Chauhan 2023-01-18 07:38:51 +05:30 committed by GStreamer Marge Bot
parent dff9f5151b
commit 26ee3d83fb

View file

@ -6,6 +6,10 @@
# 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
@ -17,11 +21,8 @@ import argparse
import gi
gi.require_version('Gst', '1.0')
from gi.repository import Gst
gi.require_version('GstWebRTC', '1.0')
from gi.repository import GstWebRTC
gi.require_version('GstSdp', '1.0')
from gi.repository import GstSdp
# Ensure that gst-python is installed
try:
@ -53,8 +54,6 @@ PIPELINE_DESC = {
'VP8': PIPELINE_DESC_VP8,
}
from websockets.version import version as wsv
def print_status(msg):
print(f'--- {msg}')
@ -159,7 +158,7 @@ class WebRTCClient:
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()
@ -235,7 +234,7 @@ class WebRTCClient:
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()
@ -244,7 +243,7 @@ class WebRTCClient:
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)
@ -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')
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'])
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:
assert(self.webrtc)
assert self.webrtc
ice = msg['ice']
candidate = ice['candidate']
sdpmlineindex = ice['sdpMLineIndex']