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Change GstSdp.sdp_message_parse_buffer to GstSdp.SDPMessage.new_from_text in examples

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Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3477>
This commit is contained in:
Seungmin Kim 2022-11-28 14:54:27 +00:00 committed by GStreamer Marge Bot
parent 412eaf3526
commit 0db1ff532d
4 changed files with 5 additions and 13 deletions
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5
subprojects/gst-examples/webrtc/check/validate/client.py
View file

@ -88,10 +88,7 @@ class WebRTCBinObserver(WebRTCObserver):
return new_state
def _deepcopy_session_description(self, desc):
# XXX: passing 'offer' to both a promise and an action signal without
# a deepcopy will segfault...
new_sdp = GstSdp.SDPMessage.new()[1]
GstSdp.sdp_message_parse_buffer(bytes(desc.sdp.as_text().encode()), new_sdp)
_, new_sdp = GstSdp.SDPMessage.new_from_text(desc.sdp.as_text())
return GstWebRTC.WebRTCSessionDescription.new(desc.type, new_sdp)
def _on_offer_created(self, promise, element):

3
subprojects/gst-examples/webrtc/check/validate/webrtc_validate.py
View file

@ -170,8 +170,7 @@ class WebRTCApplication(object):
def have_json(msg):
if 'sdp' in msg:
sdp = msg['sdp']
res, sdpmsg = GstSdp.SDPMessage.new()
GstSdp.sdp_message_parse_buffer(bytes(sdp['sdp'].encode()), sdpmsg)
res, sdpmsg = GstSdp.SDPMessage.new_from_text(sdp['sdp'])
sdptype = GstWebRTC.WebRTCSDPType.ANSWER if sdp['type'] == 'answer' else GstWebRTC.WebRTCSDPType.OFFER
desc = GstWebRTC.WebRTCSessionDescription.new(sdptype, sdpmsg)
self.client.set_remote_description(desc)

4
subprojects/gst-examples/webrtc/janus/janusvideoroom.py
View file

@ -358,9 +358,7 @@ class WebRTCClient:
sdp = msg['sdp']
assert(msg['type'] == 'answer')
print ('Received answer:\n%s' % sdp)
res, sdpmsg = GstSdp.SDPMessage.new()
GstSdp.sdp_message_parse_buffer(bytes(sdp.encode()), sdpmsg)
res, sdpmsg = GstSdp.SDPMessage.new_from_text(sdp)
answer = GstWebRTC.WebRTCSessionDescription.new(GstWebRTC.WebRTCSDPType.ANSWER, sdpmsg)
promise = Gst.Promise.new()
self.webrtc.emit('set-remote-description', answer, promise)

6
subprojects/gst-examples/webrtc/sendrecv/gst/webrtc_sendrecv.py
View file

@ -224,16 +224,14 @@ class WebRTCClient:
sdp = msg['sdp']['sdp']
if msg['sdp']['type'] == 'answer':
print_status('Received answer:\n%s' % sdp)
res, sdpmsg = GstSdp.SDPMessage.new()
GstSdp.sdp_message_parse_buffer(bytes(sdp.encode()), sdpmsg)
res, sdpmsg = GstSdp.SDPMessage.new_from_text(sdp)
answer = GstWebRTC.WebRTCSessionDescription.new(GstWebRTC.WebRTCSDPType.ANSWER, sdpmsg)
promise = Gst.Promise.new()
self.webrtc.emit('set-remote-description', answer, promise)
promise.interrupt() # we don't care about the result, discard it
else:
print_status('Received offer:\n%s' % sdp)
res, sdpmsg = GstSdp.SDPMessage.new()
GstSdp.sdp_message_parse_buffer(bytes(sdp.encode()), sdpmsg)
res, sdpmsg = GstSdp.SDPMessage.new_from_text(sdp)
if not self.webrtc:
print_status('Incoming call: received an offer, creating pipeline')