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

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#!/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.
from websockets.version import version as wsv
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 # NOQA
gi.require_version('GstWebRTC', '1.0')
from gi.repository import GstWebRTC # NOQA
gi.require_version('GstSdp', '1.0')
from gi.repository import GstSdp # NOQA
# 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
WEBRTCBIN = 'webrtcbin name=sendrecv latency=0 \
stun-server=stun://stun.l.google.com:19302 \
turn-server=turn://gstreamer:IsGreatWhenYouCanGetItToWork@webrtc.nirbheek.in:3478'
PIPELINE_DESC_VP8 = WEBRTCBIN + '''
{vsrc} ! 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={video_pt} ! sendrecv.
{asrc} ! audioconvert ! audioresample ! queue ! opusenc perfect-timestamp=true ! rtpopuspay !
queue ! application/x-rtp,media=audio,encoding-name=OPUS,payload={audio_pt} ! sendrecv.
'''
PIPELINE_DESC_H264 = WEBRTCBIN + '''
{vsrc} ! videoconvert ! queue !
x264enc tune=zerolatency speed-preset=ultrafast key-int-max=30 intra-refresh=true !
rtph264pay aggregate-mode=zero-latency config-interval=-1 !
queue ! application/x-rtp,media=video,encoding-name=H264,payload={video_pt} ! sendrecv.
{asrc} ! audioconvert ! audioresample ! queue ! opusenc perfect-timestamp=true ! rtpopuspay !
queue ! application/x-rtp,media=audio,encoding-name=OPUS,payload={audio_pt} ! sendrecv.
'''
# Force I420 because dav1d bundled with Chrome doesn't support 10-bit choma/luma (I420_10LE)
PIPELINE_DESC_AV1 = WEBRTCBIN + '''
{vsrc} ! videoconvert ! queue !
video/x-raw,format=I420 ! svtav1enc preset=13 ! av1parse ! rtpav1pay !
queue ! application/x-rtp,media=video,encoding-name=AV1,payload={video_pt} ! sendrecv.
{asrc} ! audioconvert ! audioresample ! queue ! opusenc perfect-timestamp=true ! rtpopuspay !
queue ! application/x-rtp,media=audio,encoding-name=OPUS,payload={audio_pt} ! sendrecv.
'''
PIPELINE_DESC = {
'AV1': PIPELINE_DESC_AV1,
'H264': PIPELINE_DESC_H264,
'VP8': PIPELINE_DESC_VP8,
}
VSRC = {
'test': 'videotestsrc is-live=true pattern=ball',
'camera': 'autovideosrc ! video/x-raw,framerate=[25/1,30/1]',
}
ASRC = {
'test': 'audiotestsrc is-live=true',
'camera': 'autoaudiosrc',
}
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, video_encoding, source_type):
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
# Video encoding: VP8, H264, etc
self.video_encoding = video_encoding.upper()
# Audio and video source to use
self.asrc = ASRC[source_type]
self.vsrc = VSRC[source_type]
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 on_bus_poll_cb(self, bus):
def remove_bus_poll():
self.event_loop.remove_reader(bus.get_pollfd().fd)
self.event_loop.stop()
while bus.peek():
msg = bus.pop()
if msg.type == Gst.MessageType.ERROR:
err = msg.parse_error()
print("ERROR:", err.gerror, err.debug)
remove_bus_poll()
break
elif msg.type == Gst.MessageType.EOS:
remove_bus_poll()
break
elif msg.type == Gst.MessageType.LATENCY:
self.pipe.recalculate_latency()
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, audio_pt=96, video_pt=97):
print_status(f'Creating pipeline, create_offer: {create_offer}')
desc = PIPELINE_DESC[self.video_encoding].format(video_pt=video_pt,
audio_pt=audio_pt,
vsrc=self.vsrc,
asrc=self.asrc)
self.pipe = Gst.parse_launch(desc)
bus = self.pipe.get_bus()
self.event_loop.add_reader(bus.get_pollfd().fd, self.on_bus_poll_cb, bus)
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=self.video_encoding, audio_encoding='OPUS')
assert self.video_encoding in pts
assert 'OPUS' in pts
self.start_pipeline(create_offer=False, video_pt=pts[self.video_encoding], audio_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_plugin_features(source_type, video_encoding):
"""ensure we have all the plugins/features we need"""
needed = ['opusenc', 'nicesink', 'webrtcbin', 'dtlssrtpenc', 'srtpenc',
'rtpbin', 'rtpopuspay']
if source_type == 'camera':
needed += ['autoaudiosrc', 'autovideosrc']
else:
needed += ['audiotestsrc', 'videotestsrc']
if video_encoding == 'vp8':
needed += ['vp8enc', 'vp8dec']
elif video_encoding == 'h264':
needed += ['x264enc', 'h264parse']
elif video_encoding == 'av1':
needed += ['svtav1enc', 'av1parse']
missing = []
reg = Gst.Registry.get()
for fname in needed:
feature = reg.find_feature(fname, Gst.ElementFactory.__gtype__)
if not feature:
missing.append(fname)
if missing:
print("Missing gstreamer elements:", *missing)
return False
return True
if __name__ == '__main__':
Gst.init(None)
parser = argparse.ArgumentParser()
parser.add_argument('--video-encoding', default='vp8', nargs='?', choices=['vp8', 'h264', 'av1'],
help='Video encoding to negotiate')
parser.add_argument('--camera', default='test', const='camera', action='store_const',
dest='source_type',
help='Use an attached camera and mic instead of test sources')
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.gstreamer.net: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 check_plugin_features(args.source_type, args.video_encoding):
sys.exit(1)
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, args.video_encoding, args.source_type)
loop.run_until_complete(c.connect())
res = loop.run_until_complete(c.loop())
sys.exit(res)
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