gstreamer/subprojects/gst-examples/webrtc/sendrecv/gst/webrtc_sendrecv.py

#!/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 ! 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_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_from_text(sdp)

                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)