mirror of
https://gitlab.freedesktop.org/gstreamer/gstreamer.git
synced 2024-12-20 23:36:38 +00:00
4ae903d383
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>
336 lines
13 KiB
Python
Executable file
336 lines
13 KiB
Python
Executable file
#!/usr/bin/env python3
|
|
#
|
|
# Copyright (C) 2018 Matthew Waters <matthew@centricular.com>
|
|
# 2022 Nirbheek Chauhan <nirbheek@centricular.com>
|
|
#
|
|
# Demo gstreamer app for negotiating and streaming a sendrecv webrtc stream
|
|
# with a browser JS app, implemented in Python.
|
|
|
|
import random
|
|
import ssl
|
|
import websockets
|
|
import asyncio
|
|
import os
|
|
import sys
|
|
import json
|
|
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:
|
|
from gi.overrides import Gst as _
|
|
except ImportError:
|
|
print('gstreamer-python binding overrides aren\'t available, please install them')
|
|
raise
|
|
|
|
# These properties all mirror the ones in webrtc-sendrecv.c, see there for explanations
|
|
PIPELINE_DESC = '''
|
|
webrtcbin name=sendrecv bundle-policy=max-bundle stun-server=stun://stun.l.google.com:19302
|
|
videotestsrc is-live=true pattern=ball ! videoconvert ! queue ! \
|
|
vp8enc deadline=1 keyframe-max-dist=2000 ! rtpvp8pay picture-id-mode=15-bit !
|
|
queue ! application/x-rtp,media=video,encoding-name=VP8,payload={vp8_pt} ! sendrecv.
|
|
audiotestsrc is-live=true wave=red-noise ! audioconvert ! audioresample ! queue ! opusenc ! rtpopuspay !
|
|
queue ! application/x-rtp,media=audio,encoding-name=OPUS,payload={opus_pt} ! sendrecv.
|
|
'''
|
|
|
|
from websockets.version import version as wsv
|
|
|
|
|
|
def print_status(msg):
|
|
print(f'--- {msg}')
|
|
|
|
|
|
def print_error(msg):
|
|
print(f'!!! {msg}', file=sys.stderr)
|
|
|
|
|
|
def get_payload_types(sdpmsg, video_encoding, audio_encoding):
|
|
'''
|
|
Find the payload types for the specified video and audio encoding.
|
|
|
|
Very simplistically finds the first payload type matching the encoding
|
|
name. More complex applications will want to match caps on
|
|
profile-level-id, packetization-mode, etc.
|
|
'''
|
|
video_pt = None
|
|
audio_pt = None
|
|
for i in range(0, sdpmsg.medias_len()):
|
|
media = sdpmsg.get_media(i)
|
|
for j in range(0, media.formats_len()):
|
|
fmt = media.get_format(j)
|
|
if fmt == 'webrtc-datachannel':
|
|
continue
|
|
pt = int(fmt)
|
|
caps = media.get_caps_from_media(pt)
|
|
s = caps.get_structure(0)
|
|
encoding_name = s['encoding-name']
|
|
if video_pt is None and encoding_name == video_encoding:
|
|
video_pt = pt
|
|
elif audio_pt is None and encoding_name == audio_encoding:
|
|
audio_pt = pt
|
|
return {video_encoding: video_pt, audio_encoding: audio_pt}
|
|
|
|
|
|
class WebRTCClient:
|
|
def __init__(self, loop, our_id, peer_id, server, remote_is_offerer):
|
|
self.conn = None
|
|
self.pipe = None
|
|
self.webrtc = None
|
|
self.event_loop = loop
|
|
self.server = server
|
|
# An optional user-specified ID we can use to register
|
|
self.our_id = our_id
|
|
# The actual ID we used to register
|
|
self.id_ = None
|
|
# An optional peer ID we should connect to
|
|
self.peer_id = peer_id
|
|
# Whether we will send the offer or the remote peer will
|
|
self.remote_is_offerer = remote_is_offerer
|
|
|
|
async def send(self, msg):
|
|
assert self.conn
|
|
print(f'>>> {msg}')
|
|
await self.conn.send(msg)
|
|
|
|
async def connect(self):
|
|
self.conn = await websockets.connect(self.server)
|
|
if self.our_id is None:
|
|
self.id_ = str(random.randrange(10, 10000))
|
|
else:
|
|
self.id_ = self.our_id
|
|
await self.send(f'HELLO {self.id_}')
|
|
|
|
async def setup_call(self):
|
|
assert self.peer_id
|
|
await self.send(f'SESSION {self.peer_id}')
|
|
|
|
def send_soon(self, msg):
|
|
asyncio.run_coroutine_threadsafe(self.send(msg), self.event_loop)
|
|
|
|
def send_sdp(self, offer):
|
|
text = offer.sdp.as_text()
|
|
if offer.type == GstWebRTC.WebRTCSDPType.OFFER:
|
|
print_status('Sending offer:\n%s' % text)
|
|
msg = json.dumps({'sdp': {'type': 'offer', 'sdp': text}})
|
|
elif offer.type == GstWebRTC.WebRTCSDPType.ANSWER:
|
|
print_status('Sending answer:\n%s' % text)
|
|
msg = json.dumps({'sdp': {'type': 'answer', 'sdp': text}})
|
|
else:
|
|
raise AssertionError(offer.type)
|
|
self.send_soon(msg)
|
|
|
|
def on_offer_created(self, promise, _, __):
|
|
assert(promise.wait() == Gst.PromiseResult.REPLIED)
|
|
reply = promise.get_reply()
|
|
offer = reply['offer']
|
|
promise = Gst.Promise.new()
|
|
print_status('Offer created, setting local description')
|
|
self.webrtc.emit('set-local-description', offer, promise)
|
|
promise.interrupt() # we don't care about the result, discard it
|
|
self.send_sdp(offer)
|
|
|
|
def on_negotiation_needed(self, _, create_offer):
|
|
if create_offer:
|
|
print_status('Call was connected: creating offer')
|
|
promise = Gst.Promise.new_with_change_func(self.on_offer_created, None, None)
|
|
self.webrtc.emit('create-offer', None, promise)
|
|
|
|
def send_ice_candidate_message(self, _, mlineindex, candidate):
|
|
icemsg = json.dumps({'ice': {'candidate': candidate, 'sdpMLineIndex': mlineindex}})
|
|
self.send_soon(icemsg)
|
|
|
|
def on_incoming_decodebin_stream(self, _, pad):
|
|
if not pad.has_current_caps():
|
|
print_error(pad, 'has no caps, ignoring')
|
|
return
|
|
|
|
caps = pad.get_current_caps()
|
|
assert (len(caps))
|
|
s = caps[0]
|
|
name = s.get_name()
|
|
if name.startswith('video'):
|
|
q = Gst.ElementFactory.make('queue')
|
|
conv = Gst.ElementFactory.make('videoconvert')
|
|
sink = Gst.ElementFactory.make('autovideosink')
|
|
self.pipe.add(q, conv, sink)
|
|
self.pipe.sync_children_states()
|
|
pad.link(q.get_static_pad('sink'))
|
|
q.link(conv)
|
|
conv.link(sink)
|
|
elif name.startswith('audio'):
|
|
q = Gst.ElementFactory.make('queue')
|
|
conv = Gst.ElementFactory.make('audioconvert')
|
|
resample = Gst.ElementFactory.make('audioresample')
|
|
sink = Gst.ElementFactory.make('autoaudiosink')
|
|
self.pipe.add(q, conv, resample, sink)
|
|
self.pipe.sync_children_states()
|
|
pad.link(q.get_static_pad('sink'))
|
|
q.link(conv)
|
|
conv.link(resample)
|
|
resample.link(sink)
|
|
|
|
def on_ice_gathering_state_notify(self, pspec, _):
|
|
state = self.webrtc.get_property('ice-gathering-state')
|
|
print_status(f'ICE gathering state changed to {state}')
|
|
|
|
def on_incoming_stream(self, _, pad):
|
|
if pad.direction != Gst.PadDirection.SRC:
|
|
return
|
|
|
|
decodebin = Gst.ElementFactory.make('decodebin')
|
|
decodebin.connect('pad-added', self.on_incoming_decodebin_stream)
|
|
self.pipe.add(decodebin)
|
|
decodebin.sync_state_with_parent()
|
|
pad.link(decodebin.get_static_pad('sink'))
|
|
|
|
def start_pipeline(self, create_offer=True, opus_pt=96, vp8_pt=97):
|
|
print_status(f'Creating pipeline, create_offer: {create_offer}')
|
|
self.pipe = Gst.parse_launch(PIPELINE_DESC.format(vp8_pt=vp8_pt, opus_pt=opus_pt))
|
|
self.webrtc = self.pipe.get_by_name('sendrecv')
|
|
self.webrtc.connect('on-negotiation-needed', self.on_negotiation_needed, create_offer)
|
|
self.webrtc.connect('on-ice-candidate', self.send_ice_candidate_message)
|
|
self.webrtc.connect('notify::ice-gathering-state', self.on_ice_gathering_state_notify)
|
|
self.webrtc.connect('pad-added', self.on_incoming_stream)
|
|
self.pipe.set_state(Gst.State.PLAYING)
|
|
|
|
def on_answer_created(self, promise, _, __):
|
|
assert(promise.wait() == Gst.PromiseResult.REPLIED)
|
|
reply = promise.get_reply()
|
|
answer = reply['answer']
|
|
promise = Gst.Promise.new()
|
|
self.webrtc.emit('set-local-description', answer, promise)
|
|
promise.interrupt() # we don't care about the result, discard it
|
|
self.send_sdp(answer)
|
|
|
|
def on_offer_set(self, promise, _, __):
|
|
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)
|
|
|
|
def handle_json(self, message):
|
|
try:
|
|
msg = json.loads(message)
|
|
except json.decoder.JSONDecoderError:
|
|
print_error('Failed to parse JSON message, this might be a bug')
|
|
raise
|
|
if 'sdp' in msg:
|
|
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)
|
|
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)
|
|
|
|
if not self.webrtc:
|
|
print_status('Incoming call: received an offer, creating pipeline')
|
|
pts = get_payload_types(sdpmsg, video_encoding='VP8', audio_encoding='OPUS')
|
|
assert('VP8' in pts)
|
|
assert('OPUS' in pts)
|
|
self.start_pipeline(create_offer=False, vp8_pt=pts['VP8'], opus_pt=pts['OPUS'])
|
|
|
|
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)
|
|
ice = msg['ice']
|
|
candidate = ice['candidate']
|
|
sdpmlineindex = ice['sdpMLineIndex']
|
|
self.webrtc.emit('add-ice-candidate', sdpmlineindex, candidate)
|
|
else:
|
|
print_error('Unknown JSON message')
|
|
|
|
def close_pipeline(self):
|
|
if self.pipe:
|
|
self.pipe.set_state(Gst.State.NULL)
|
|
self.pipe = None
|
|
self.webrtc = None
|
|
|
|
async def loop(self):
|
|
assert self.conn
|
|
async for message in self.conn:
|
|
print(f'<<< {message}')
|
|
if message == 'HELLO':
|
|
assert self.id_
|
|
# If a peer ID is specified, we want to connect to it. If not,
|
|
# we wait for an incoming call.
|
|
if not self.peer_id:
|
|
print_status(f'Waiting for incoming call: ID is {self.id_}')
|
|
else:
|
|
if self.remote_is_offerer:
|
|
print_status('Have peer ID: initiating call (will request remote peer to create offer)')
|
|
else:
|
|
print_status('Have peer ID: initiating call (will create offer)')
|
|
await self.setup_call()
|
|
elif message == 'SESSION_OK':
|
|
if self.remote_is_offerer:
|
|
# We are initiating the call, but we want the remote peer to create the offer
|
|
print_status('Call was connected: requesting remote peer for offer')
|
|
await self.send('OFFER_REQUEST')
|
|
else:
|
|
self.start_pipeline()
|
|
elif message == 'OFFER_REQUEST':
|
|
print_status('Incoming call: we have been asked to create the offer')
|
|
self.start_pipeline()
|
|
elif message.startswith('ERROR'):
|
|
print_error(message)
|
|
self.close_pipeline()
|
|
return 1
|
|
else:
|
|
self.handle_json(message)
|
|
self.close_pipeline()
|
|
return 0
|
|
|
|
async def stop(self):
|
|
if self.conn:
|
|
await self.conn.close()
|
|
self.conn = None
|
|
|
|
|
|
def check_plugins():
|
|
needed = ["opus", "vpx", "nice", "webrtc", "dtls", "srtp", "rtp",
|
|
"rtpmanager", "videotestsrc", "audiotestsrc"]
|
|
missing = list(filter(lambda p: Gst.Registry.get().find_plugin(p) is None, needed))
|
|
if len(missing):
|
|
print_error('Missing gstreamer plugins:', missing)
|
|
return False
|
|
return True
|
|
|
|
|
|
if __name__ == '__main__':
|
|
Gst.init(None)
|
|
if not check_plugins():
|
|
sys.exit(1)
|
|
parser = argparse.ArgumentParser()
|
|
parser.add_argument('--peer-id', help='String ID of the peer to connect to')
|
|
parser.add_argument('--our-id', help='String ID that the peer can use to connect to us')
|
|
parser.add_argument('--server', default='wss://webrtc.nirbheek.in:8443',
|
|
help='Signalling server to connect to, eg "wss://127.0.0.1:8443"')
|
|
parser.add_argument('--remote-offerer', default=False, action='store_true',
|
|
dest='remote_is_offerer',
|
|
help='Request that the peer generate the offer and we\'ll answer')
|
|
args = parser.parse_args()
|
|
if not args.peer_id and not args.our_id:
|
|
print('You must pass either --peer-id or --our-id')
|
|
sys.exit(1)
|
|
loop = asyncio.new_event_loop()
|
|
c = WebRTCClient(loop, args.our_id, args.peer_id, args.server, args.remote_is_offerer)
|
|
loop.run_until_complete(c.connect())
|
|
res = loop.run_until_complete(c.loop())
|
|
sys.exit(res)
|