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https://gitlab.freedesktop.org/gstreamer/gstreamer.git
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0007fa38e0
We need to parse the payload type map provided by the offer SDP and set those values on the payloader, otherwise webrtcbin will create a recvonly answer SDP and we won't send anything to the browser. Fixed it for both C and Python sendrecv examples. Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1864>
336 lines
13 KiB
Python
Executable file
336 lines
13 KiB
Python
Executable file
#!/usr/bin/env python3
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#
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# Copyright (C) 2018 Matthew Waters <matthew@centricular.com>
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# 2022 Nirbheek Chauhan <nirbheek@centricular.com>
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#
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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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import random
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import ssl
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import websockets
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import asyncio
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import os
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import sys
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import json
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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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from gi.overrides import Gst as _
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except ImportError:
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print('gstreamer-python binding overrides aren\'t available, please install them')
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raise
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# These properties all mirror the ones in webrtc-sendrecv.c, see there for explanations
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PIPELINE_DESC = '''
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webrtcbin name=sendrecv bundle-policy=max-bundle stun-server=stun://stun.l.google.com:19302
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videotestsrc is-live=true pattern=ball ! videoconvert ! queue ! \
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vp8enc deadline=1 keyframe-max-dist=2000 ! rtpvp8pay picture-id-mode=15-bit !
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queue ! application/x-rtp,media=video,encoding-name=VP8,payload={vp8_pt} ! sendrecv.
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audiotestsrc is-live=true wave=red-noise ! audioconvert ! audioresample ! queue ! opusenc ! rtpopuspay !
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queue ! application/x-rtp,media=audio,encoding-name=OPUS,payload={opus_pt} ! sendrecv.
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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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def print_error(msg):
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print(f'!!! {msg}', file=sys.stderr)
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def get_payload_types(sdpmsg, video_encoding, audio_encoding):
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'''
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Find the payload types for the specified video and audio encoding.
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Very simplistically finds the first payload type matching the encoding
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name. More complex applications will want to match caps on
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profile-level-id, packetization-mode, etc.
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'''
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video_pt = None
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audio_pt = None
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for i in range(0, sdpmsg.medias_len()):
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media = sdpmsg.get_media(i)
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for j in range(0, media.formats_len()):
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fmt = media.get_format(j)
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if fmt == 'webrtc-datachannel':
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continue
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pt = int(fmt)
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caps = media.get_caps_from_media(pt)
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s = caps.get_structure(0)
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encoding_name = s['encoding-name']
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if video_pt is None and encoding_name == video_encoding:
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video_pt = pt
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elif audio_pt is None and encoding_name == audio_encoding:
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audio_pt = pt
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return {video_encoding: video_pt, audio_encoding: audio_pt}
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class WebRTCClient:
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def __init__(self, loop, our_id, peer_id, server, remote_is_offerer):
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self.conn = None
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self.pipe = None
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self.webrtc = None
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self.event_loop = loop
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self.server = server
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# An optional user-specified ID we can use to register
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self.our_id = our_id
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# The actual ID we used to register
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self.id_ = None
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# An optional peer ID we should connect to
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self.peer_id = peer_id
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# Whether we will send the offer or the remote peer will
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self.remote_is_offerer = remote_is_offerer
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async def send(self, msg):
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assert self.conn
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print(f'>>> {msg}')
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await self.conn.send(msg)
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async def connect(self):
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self.conn = await websockets.connect(self.server)
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if self.our_id is None:
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self.id_ = str(random.randrange(10, 10000))
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else:
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self.id_ = self.our_id
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await self.send(f'HELLO {self.id_}')
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async def setup_call(self):
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assert self.peer_id
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await self.send(f'SESSION {self.peer_id}')
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def send_soon(self, msg):
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asyncio.run_coroutine_threadsafe(self.send(msg), self.event_loop)
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def send_sdp(self, offer):
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text = offer.sdp.as_text()
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if offer.type == GstWebRTC.WebRTCSDPType.OFFER:
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print_status('Sending offer:\n%s' % text)
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msg = json.dumps({'sdp': {'type': 'offer', 'sdp': text}})
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elif offer.type == GstWebRTC.WebRTCSDPType.ANSWER:
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print_status('Sending answer:\n%s' % text)
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msg = json.dumps({'sdp': {'type': 'answer', 'sdp': text}})
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else:
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raise AssertionError(offer.type)
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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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reply = promise.get_reply()
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offer = reply['offer']
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promise = Gst.Promise.new()
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print_status('Offer created, setting local description')
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self.webrtc.emit('set-local-description', offer, promise)
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promise.interrupt() # we don't care about the result, discard it
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self.send_sdp(offer)
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def on_negotiation_needed(self, _, create_offer):
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if create_offer:
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print_status('Call was connected: creating offer')
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promise = Gst.Promise.new_with_change_func(self.on_offer_created, None, None)
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self.webrtc.emit('create-offer', None, promise)
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def send_ice_candidate_message(self, _, mlineindex, candidate):
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icemsg = json.dumps({'ice': {'candidate': candidate, 'sdpMLineIndex': mlineindex}})
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self.send_soon(icemsg)
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def on_incoming_decodebin_stream(self, _, pad):
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if not pad.has_current_caps():
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print_error(pad, 'has no caps, ignoring')
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return
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caps = pad.get_current_caps()
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assert (len(caps))
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s = caps[0]
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name = s.get_name()
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if name.startswith('video'):
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q = Gst.ElementFactory.make('queue')
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conv = Gst.ElementFactory.make('videoconvert')
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sink = Gst.ElementFactory.make('autovideosink')
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self.pipe.add(q, conv, sink)
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self.pipe.sync_children_states()
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pad.link(q.get_static_pad('sink'))
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q.link(conv)
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conv.link(sink)
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elif name.startswith('audio'):
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q = Gst.ElementFactory.make('queue')
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conv = Gst.ElementFactory.make('audioconvert')
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resample = Gst.ElementFactory.make('audioresample')
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sink = Gst.ElementFactory.make('autoaudiosink')
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self.pipe.add(q, conv, resample, sink)
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self.pipe.sync_children_states()
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pad.link(q.get_static_pad('sink'))
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q.link(conv)
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conv.link(resample)
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resample.link(sink)
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def on_ice_gathering_state_notify(self, pspec, _):
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state = self.webrtc.get_property('ice-gathering-state')
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print_status(f'ICE gathering state changed to {state}')
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def on_incoming_stream(self, _, pad):
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if pad.direction != Gst.PadDirection.SRC:
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return
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decodebin = Gst.ElementFactory.make('decodebin')
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decodebin.connect('pad-added', self.on_incoming_decodebin_stream)
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self.pipe.add(decodebin)
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decodebin.sync_state_with_parent()
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self.webrtc.link(decodebin)
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def start_pipeline(self, create_offer=True, opus_pt=96, vp8_pt=97):
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print_status(f'Creating pipeline, create_offer: {create_offer}')
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self.pipe = Gst.parse_launch(PIPELINE_DESC.format(vp8_pt=vp8_pt, opus_pt=opus_pt))
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self.webrtc = self.pipe.get_by_name('sendrecv')
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self.webrtc.connect('on-negotiation-needed', self.on_negotiation_needed, create_offer)
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self.webrtc.connect('on-ice-candidate', self.send_ice_candidate_message)
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self.webrtc.connect('notify::ice-gathering-state', self.on_ice_gathering_state_notify)
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self.webrtc.connect('pad-added', self.on_incoming_stream)
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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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reply = promise.get_reply()
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answer = reply['answer']
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promise = Gst.Promise.new()
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self.webrtc.emit('set-local-description', answer, promise)
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promise.interrupt() # we don't care about the result, discard it
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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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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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def handle_json(self, message):
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try:
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msg = json.loads(message)
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except json.decoder.JSONDecoderError:
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print_error('Failed to parse JSON message, this might be a bug')
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raise
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if 'sdp' in msg:
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sdp = msg['sdp']['sdp']
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if msg['sdp']['type'] == 'answer':
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print_status('Received answer:\n%s' % sdp)
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res, sdpmsg = GstSdp.SDPMessage.new()
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GstSdp.sdp_message_parse_buffer(bytes(sdp.encode()), sdpmsg)
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answer = GstWebRTC.WebRTCSessionDescription.new(GstWebRTC.WebRTCSDPType.ANSWER, sdpmsg)
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promise = Gst.Promise.new()
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self.webrtc.emit('set-remote-description', answer, promise)
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promise.interrupt() # we don't care about the result, discard it
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else:
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print_status('Received offer:\n%s' % sdp)
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res, sdpmsg = GstSdp.SDPMessage.new()
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GstSdp.sdp_message_parse_buffer(bytes(sdp.encode()), sdpmsg)
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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='VP8', audio_encoding='OPUS')
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assert('VP8' in pts)
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assert('OPUS' in pts)
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self.start_pipeline(create_offer=False, vp8_pt=pts['VP8'], opus_pt=pts['OPUS'])
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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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ice = msg['ice']
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candidate = ice['candidate']
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sdpmlineindex = ice['sdpMLineIndex']
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self.webrtc.emit('add-ice-candidate', sdpmlineindex, candidate)
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else:
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print_error('Unknown JSON message')
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def close_pipeline(self):
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if self.pipe:
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self.pipe.set_state(Gst.State.NULL)
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self.pipe = None
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self.webrtc = None
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async def loop(self):
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assert self.conn
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async for message in self.conn:
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print(f'<<< {message}')
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if message == 'HELLO':
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assert self.id_
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# If a peer ID is specified, we want to connect to it. If not,
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# we wait for an incoming call.
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if not self.peer_id:
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print_status(f'Waiting for incoming call: ID is {self.id_}')
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else:
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if self.remote_is_offerer:
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print_status('Have peer ID: initiating call (will request remote peer to create offer)')
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else:
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print_status('Have peer ID: initiating call (will create offer)')
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await self.setup_call()
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elif message == 'SESSION_OK':
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if self.remote_is_offerer:
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# We are initiating the call, but we want the remote peer to create the offer
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print_status('Call was connected: requesting remote peer for offer')
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await self.send('OFFER_REQUEST')
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else:
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self.start_pipeline()
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elif message == 'OFFER_REQUEST':
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print_status('Incoming call: we have been asked to create the offer')
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self.start_pipeline()
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elif message.startswith('ERROR'):
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print_error(message)
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self.close_pipeline()
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return 1
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else:
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self.handle_json(message)
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self.close_pipeline()
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return 0
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async def stop(self):
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if self.conn:
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await self.conn.close()
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self.conn = None
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def check_plugins():
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needed = ["opus", "vpx", "nice", "webrtc", "dtls", "srtp", "rtp",
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"rtpmanager", "videotestsrc", "audiotestsrc"]
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missing = list(filter(lambda p: Gst.Registry.get().find_plugin(p) is None, needed))
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if len(missing):
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print_error('Missing gstreamer plugins:', missing)
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return False
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return True
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if __name__ == '__main__':
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Gst.init(None)
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if not check_plugins():
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sys.exit(1)
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parser = argparse.ArgumentParser()
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parser.add_argument('--peer-id', help='String ID of the peer to connect to')
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parser.add_argument('--our-id', help='String ID that the peer can use to connect to us')
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parser.add_argument('--server', default='wss://webrtc.nirbheek.in:8443',
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help='Signalling server to connect to, eg "wss://127.0.0.1:8443"')
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parser.add_argument('--remote-offerer', default=False, action='store_true',
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dest='remote_is_offerer',
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help='Request that the peer generate the offer and we\'ll answer')
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args = parser.parse_args()
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if not args.peer_id and not args.our_id:
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print('You must pass either --peer-id or --our-id')
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sys.exit(1)
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loop = asyncio.new_event_loop()
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c = WebRTCClient(loop, args.our_id, args.peer_id, args.server, args.remote_is_offerer)
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loop.run_until_complete(c.connect())
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res = loop.run_until_complete(c.loop())
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sys.exit(res)
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