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webrtc_sendrecv.py: Fix PEP8 warnings in CI lint
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3742>
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parent
6a83602601
commit
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1 changed files with 11 additions and 12 deletions
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@ -6,6 +6,10 @@
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# Demo gstreamer app for negotiating and streaming a sendrecv webrtc stream
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# with a browser JS app, implemented in Python.
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from websockets.version import version as wsv
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from gi.repository import GstSdp
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from gi.repository import GstWebRTC
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from gi.repository import Gst
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import random
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import ssl
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import websockets
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@ -17,11 +21,8 @@ import argparse
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import gi
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gi.require_version('Gst', '1.0')
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from gi.repository import Gst
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gi.require_version('GstWebRTC', '1.0')
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from gi.repository import GstWebRTC
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gi.require_version('GstSdp', '1.0')
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from gi.repository import GstSdp
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# Ensure that gst-python is installed
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try:
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@ -53,8 +54,6 @@ PIPELINE_DESC = {
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'VP8': PIPELINE_DESC_VP8,
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}
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from websockets.version import version as wsv
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def print_status(msg):
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print(f'--- {msg}')
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@ -159,7 +158,7 @@ class WebRTCClient:
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self.send_soon(msg)
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def on_offer_created(self, promise, _, __):
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assert(promise.wait() == Gst.PromiseResult.REPLIED)
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assert promise.wait() == Gst.PromiseResult.REPLIED
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reply = promise.get_reply()
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offer = reply['offer']
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promise = Gst.Promise.new()
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@ -235,7 +234,7 @@ class WebRTCClient:
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self.pipe.set_state(Gst.State.PLAYING)
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def on_answer_created(self, promise, _, __):
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assert(promise.wait() == Gst.PromiseResult.REPLIED)
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assert promise.wait() == Gst.PromiseResult.REPLIED
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reply = promise.get_reply()
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answer = reply['answer']
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promise = Gst.Promise.new()
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@ -244,7 +243,7 @@ class WebRTCClient:
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self.send_sdp(answer)
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def on_offer_set(self, promise, _, __):
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assert(promise.wait() == Gst.PromiseResult.REPLIED)
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assert promise.wait() == Gst.PromiseResult.REPLIED
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promise = Gst.Promise.new_with_change_func(self.on_answer_created, None, None)
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self.webrtc.emit('create-answer', None, promise)
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@ -270,17 +269,17 @@ class WebRTCClient:
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if not self.webrtc:
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print_status('Incoming call: received an offer, creating pipeline')
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pts = get_payload_types(sdpmsg, video_encoding=self.video_encoding, audio_encoding='OPUS')
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assert(self.video_encoding in pts)
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assert('OPUS' in pts)
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assert self.video_encoding in pts
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assert 'OPUS' in pts
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self.start_pipeline(create_offer=False, video_pt=pts[self.video_encoding], audio_pt=pts['OPUS'])
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assert(self.webrtc)
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assert self.webrtc
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offer = GstWebRTC.WebRTCSessionDescription.new(GstWebRTC.WebRTCSDPType.OFFER, sdpmsg)
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promise = Gst.Promise.new_with_change_func(self.on_offer_set, None, None)
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self.webrtc.emit('set-remote-description', offer, promise)
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elif 'ice' in msg:
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assert(self.webrtc)
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assert self.webrtc
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ice = msg['ice']
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candidate = ice['candidate']
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sdpmlineindex = ice['sdpMLineIndex']
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