478 lines
19 KiB
Python
478 lines
19 KiB
Python
##
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## Copyright (C) 2006 Gajim Team
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##
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## Gajim is free software; you can redistribute it and/or modify
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## it under the terms of the GNU General Public License as published
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## by the Free Software Foundation; version 3 only.
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##
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## Gajim is distributed in the hope that it will be useful,
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## but WITHOUT ANY WARRANTY; without even the implied warranty of
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## MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
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## GNU General Public License for more details.
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##
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## You should have received a copy of the GNU General Public License
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## along with Gajim. If not, see <http://www.gnu.org/licenses/>.
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"""
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Handles Jingle RTP sessions (XEP 0167)
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"""
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import logging
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import socket
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import nbxmpp
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import gi
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from gi.repository import Farstream
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gi.require_version('Gst', '1.0')
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from gi.repository import Gst
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from gi.repository import GLib
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from common import gajim
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from common.jingle_transport import JingleTransportICEUDP
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from common.jingle_content import contents, JingleContent, JingleContentSetupException
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from common.connection_handlers_events import InformationEvent
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from common.jingle_session import FailedApplication
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from collections import deque
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log = logging.getLogger('gajim.c.jingle_rtp')
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class JingleRTPContent(JingleContent):
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def __init__(self, session, media, transport=None):
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if transport is None:
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transport = JingleTransportICEUDP(None)
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JingleContent.__init__(self, session, transport)
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self.media = media
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self._dtmf_running = False
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self.farstream_media = {
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'audio': Farstream.MediaType.AUDIO,
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'video': Farstream.MediaType.VIDEO}[media]
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self.pipeline = None
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self.src_bin = None
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self.stream_failed_once = False
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self.candidates_ready = False # True when local candidates are prepared
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# TODO
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self.conference = None
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self.funnel = None
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self.p2psession = None
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self.p2pstream = None
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self.callbacks['session-initiate'] += [self.__on_remote_codecs]
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self.callbacks['content-add'] += [self.__on_remote_codecs]
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self.callbacks['description-info'] += [self.__on_remote_codecs]
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self.callbacks['content-accept'] += [self.__on_remote_codecs]
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self.callbacks['session-accept'] += [self.__on_remote_codecs]
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self.callbacks['session-terminate'] += [self.__stop]
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self.callbacks['session-terminate-sent'] += [self.__stop]
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def setup_stream(self, on_src_pad_added):
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# pipeline and bus
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self.pipeline = Gst.Pipeline()
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bus = self.pipeline.get_bus()
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bus.add_signal_watch()
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bus.connect('message', self._on_gst_message)
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# conference
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self.conference = Gst.ElementFactory.make('fsrtpconference', None)
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self.pipeline.add(self.conference)
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self.funnel = None
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self.p2psession = self.conference.new_session(self.farstream_media)
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participant = self.conference.new_participant()
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# FIXME: Consider a workaround, here...
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# pidgin and telepathy-gabble don't follow the XEP, and it won't work
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# due to bad controlling-mode
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params = {'controlling-mode': self.session.weinitiate, 'debug': False}
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if gajim.config.get('use_stun_server'):
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stun_server = gajim.config.get('stun_server')
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if not stun_server and self.session.connection._stun_servers:
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stun_server = self.session.connection._stun_servers[0]['host']
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if stun_server:
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try:
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ip = socket.getaddrinfo(stun_server, 0, socket.AF_UNSPEC,
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socket.SOCK_STREAM)[0][4][0]
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except socket.gaierror as e:
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log.warning('Lookup of stun ip failed: %s', str(e))
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else:
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params['stun-ip'] = ip
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self.p2pstream = self.p2psession.new_stream(participant,
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Farstream.StreamDirection.BOTH)
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self.p2pstream.connect('src-pad-added', on_src_pad_added)
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self.p2pstream.set_transmitter_ht('nice', params)
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def is_ready(self):
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return JingleContent.is_ready(self) and self.candidates_ready
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def make_bin_from_config(self, config_key, pipeline, text):
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pipeline = pipeline % gajim.config.get(config_key)
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try:
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gst_bin = Gst.parse_bin_from_description(pipeline, True)
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return gst_bin
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except GLib.GError as e:
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gajim.nec.push_incoming_event(
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InformationEvent(
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None, conn=self.session.connection, level='error',
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pri_txt=_('%s configuration error') % text.capitalize(),
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sec_txt=_('Couldn’t setup %s. Check your configuration.\n\n'
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'Pipeline was:\n%s\n\nError was:\n%s') % (text, pipeline, str(e))))
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raise JingleContentSetupException
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def add_remote_candidates(self, candidates):
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JingleContent.add_remote_candidates(self, candidates)
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# FIXME: connectivity should not be etablished yet
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# Instead, it should be etablished after session-accept!
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if self.sent:
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self.p2pstream.add_remote_candidates(candidates)
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def batch_dtmf(self, events):
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"""
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Send several DTMF tones
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"""
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if self._dtmf_running:
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raise Exception("There is a DTMF batch already running")
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events = deque(events)
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self._dtmf_running = True
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self._start_dtmf(events.popleft())
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GLib.timeout_add(500, self._next_dtmf, events)
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def _next_dtmf(self, events):
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self._stop_dtmf()
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if events:
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self._start_dtmf(events.popleft())
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GLib.timeout_add(500, self._next_dtmf, events)
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else:
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self._dtmf_running = False
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def _start_dtmf(self, event):
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if event in ('*', '#'):
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event = {'*': Farstream.DTMFEvent.STAR,
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'#': Farstream.DTMFEvent.POUND}[event]
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else:
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event = int(event)
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self.p2psession.start_telephony_event(event, 2)
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def _stop_dtmf(self):
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self.p2psession.stop_telephony_event()
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def _fill_content(self, content):
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content.addChild(nbxmpp.NS_JINGLE_RTP + ' description',
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attrs={'media': self.media},
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payload=list(self.iter_codecs()))
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def _setup_funnel(self):
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self.funnel = Gst.ElementFactory.make('funnel', None)
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self.pipeline.add(self.funnel)
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self.funnel.link(self.sink)
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self.sink.set_state(Gst.State.PLAYING)
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self.funnel.set_state(Gst.State.PLAYING)
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def _on_src_pad_added(self, stream, pad, codec):
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if not self.funnel:
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self._setup_funnel()
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pad.link(self.funnel.get_request_pad('sink_%u'))
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def _on_gst_message(self, bus, message):
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if message.type == Gst.MessageType.ELEMENT:
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name = message.get_structure().get_name()
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log.debug('gst element message: %s: %s', name, message)
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if name == 'farstream-new-active-candidate-pair':
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pass
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elif name == 'farstream-recv-codecs-changed':
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pass
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elif name == 'farstream-codecs-changed':
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if self.sent and self.p2psession.props.codecs_without_config:
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self.send_description_info()
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if self.transport.remote_candidates:
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# those lines MUST be done after we get info on our
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# codecs
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self.p2pstream.add_remote_candidates(
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self.transport.remote_candidates)
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self.transport.remote_candidates = []
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self.p2pstream.set_property('direction',
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Farstream.StreamDirection.BOTH)
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elif name == 'farstream-local-candidates-prepared':
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self.candidates_ready = True
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if self.is_ready():
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self.session.on_session_state_changed(self)
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elif name == 'farstream-new-local-candidate':
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candidate = self.p2pstream.parse_new_local_candidate(message)[1]
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self.transport.candidates.append(candidate)
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if self.sent:
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# FIXME: Is this case even possible?
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self.send_candidate(candidate)
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elif name == 'farstream-component-state-changed':
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state = message.get_structure().get_value('state')
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if state == Farstream.StreamState.FAILED:
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reason = nbxmpp.Node('reason')
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reason.setTag('failed-transport')
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self.session.remove_content(self.creator, self.name, reason)
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elif name == 'farstream-error':
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log.error('Farstream error #%d!\nMessage: %s',
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message.get_structure().get_value('error-no'),
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message.get_structure().get_value('error-msg'))
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elif message.type == Gst.MessageType.ERROR:
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# TODO: Fix it to fallback to videotestsrc anytime an error occur,
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# or raise an error, Jingle way
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# or maybe one-sided stream?
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if not self.stream_failed_once:
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gajim.nec.push_incoming_event(
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InformationEvent(
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None, conn=self.session.connection, level='error',
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pri_txt=_('GStreamer error'),
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sec_txt=_('Error: %s\nDebug: %s' %
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(message.get_structure().get_value('gerror'),
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message.get_structure().get_value('debug')))))
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sink_pad = self.p2psession.get_property('sink-pad')
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# Remove old source
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self.src_bin.get_static_pad('src').unlink(sink_pad)
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self.src_bin.set_state(Gst.State.NULL)
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self.pipeline.remove(self.src_bin)
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if not self.stream_failed_once:
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# Add fallback source
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self.src_bin = self.get_fallback_src()
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self.pipeline.add(self.src_bin)
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self.src_bin.link(sink_pad)
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self.stream_failed_once = True
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else:
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reason = nbxmpp.Node('reason')
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reason.setTag('failed-application')
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self.session.remove_content(self.creator, self.name, reason)
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# Start playing again
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self.pipeline.set_state(Gst.State.PLAYING)
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@staticmethod
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def get_fallback_src():
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return Gst.ElementFactory.make('fakesrc', None)
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def on_negotiated(self):
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if self.accepted:
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if self.p2psession.get_property('codecs'):
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# those lines MUST be done after we get info on our codecs
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if self.transport.remote_candidates:
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self.p2pstream.add_remote_candidates(
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self.transport.remote_candidates)
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self.transport.remote_candidates = []
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# TODO: Farstream.StreamDirection.BOTH only if senders='both'
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# self.p2pstream.set_property('direction',
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# Farstream.StreamDirection.BOTH)
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JingleContent.on_negotiated(self)
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def __on_remote_codecs(self, stanza, content, error, action):
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"""
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Get peer codecs from what we get from peer
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"""
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codecs = []
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for codec in content.getTag('description').iterTags('payload-type'):
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if not codec['id'] or not codec['name'] or not codec['clockrate']:
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# ignore invalid payload-types
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continue
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c = Farstream.Codec.new(int(codec['id']), codec['name'],
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self.farstream_media, int(codec['clockrate']))
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if 'channels' in codec:
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c.channels = int(codec['channels'])
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else:
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c.channels = 1
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for p in codec.iterTags('parameter'):
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c.add_optional_parameter(p['name'], str(p['value']))
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codecs.append(c)
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if codecs:
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try:
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self.p2pstream.set_remote_codecs(codecs)
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except GLib.Error:
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raise FailedApplication
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def iter_codecs(self):
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codecs = self.p2psession.props.codecs_without_config
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for codec in codecs:
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attrs = {
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'name': codec.encoding_name,
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'id': codec.id,
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'channels': codec.channels
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}
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if codec.clock_rate:
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attrs['clockrate'] = codec.clock_rate
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if codec.optional_params:
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payload = [nbxmpp.Node('parameter',
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{'name': p.name, 'value': p.value})
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for p in codec.optional_params]
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else:
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payload = []
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yield nbxmpp.Node('payload-type', attrs, payload)
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def __stop(self, *things):
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self.pipeline.set_state(Gst.State.NULL)
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def __del__(self):
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self.__stop()
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def destroy(self):
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JingleContent.destroy(self)
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self.p2pstream.disconnect_by_func(self._on_src_pad_added)
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self.pipeline.get_bus().disconnect_by_func(self._on_gst_message)
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class JingleAudio(JingleRTPContent):
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"""
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Jingle VoIP sessions consist of audio content transported over an ICE UDP
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protocol
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"""
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def __init__(self, session, transport=None):
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JingleRTPContent.__init__(self, session, 'audio', transport)
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self.setup_stream()
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def set_mic_volume(self, vol):
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"""
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vol must be between 0 ans 1
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"""
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self.mic_volume.set_property('volume', vol)
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def set_out_volume(self, vol):
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"""
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vol must be between 0 ans 1
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"""
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self.out_volume.set_property('volume', vol)
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def setup_stream(self):
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JingleRTPContent.setup_stream(self, self._on_src_pad_added)
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# Configure SPEEX
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# Workaround for psi (not needed since rev
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# 147aedcea39b43402fe64c533d1866a25449888a):
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# place 16kHz before 8kHz, as buggy psi versions will take in
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# account only the first codec
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codecs = [
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Farstream.Codec.new(Farstream.CODEC_ID_ANY, 'SPEEX',
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Farstream.MediaType.AUDIO, 16000),
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Farstream.Codec.new(Farstream.CODEC_ID_ANY, 'SPEEX',
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Farstream.MediaType.AUDIO, 8000)]
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self.p2psession.set_codec_preferences(codecs)
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# the local parts
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# TODO: Add queues?
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self.src_bin = self.make_bin_from_config('audio_input_device',
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'%s ! audioconvert',
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_("audio input"))
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self.sink = self.make_bin_from_config('audio_output_device',
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'audioconvert ! volume name=gajim_out_vol ! %s',
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_("audio output"))
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self.mic_volume = self.src_bin.get_by_name('gajim_vol')
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self.out_volume = self.sink.get_by_name('gajim_out_vol')
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# link gst elements
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self.pipeline.add(self.sink)
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self.pipeline.add(self.src_bin)
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self.src_bin.get_static_pad('src').link(self.p2psession.get_property(
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'sink-pad'))
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# The following is needed for farstream to process ICE requests:
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self.pipeline.set_state(Gst.State.PLAYING)
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class JingleVideo(JingleRTPContent):
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def __init__(self, session, transport=None, in_xid=0, out_xid=0):
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JingleRTPContent.__init__(self, session, 'video', transport)
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self.in_xid = in_xid
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self.out_xid = out_xid
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self.out_xid_set = False
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self.setup_stream()
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def setup_stream(self):
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# TODO: Everything is not working properly:
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# sometimes, one window won't show up,
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# sometimes it'll freeze...
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JingleRTPContent.setup_stream(self, self._on_src_pad_added)
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bus = self.pipeline.get_bus()
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bus.enable_sync_message_emission()
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bus.connect('sync-message::element', self._on_sync_message)
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# the local parts
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if gajim.config.get('video_framerate'):
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framerate = 'videorate ! video/x-raw,framerate=%s ! ' % \
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gajim.config.get('video_framerate')
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else:
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framerate = ''
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try:
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w, h = gajim.config.get('video_size').split('x')
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except:
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w = h = None
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if w and h:
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video_size = 'video/x-raw,width=%s,height=%s ! ' % (w, h)
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else:
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video_size = ''
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if gajim.config.get('video_see_self'):
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tee = '! tee name=t ! queue ! videoscale ! ' + \
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'video/x-raw,width=160,height=120 ! videoconvert ! ' + \
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'%s t. ! queue ' % gajim.config.get(
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'video_output_device')
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else:
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tee = ''
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self.src_bin = self.make_bin_from_config('video_input_device',
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'%%s %s! %svideoscale ! %svideoconvert' %
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(tee, framerate, video_size),
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_("video input"))
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self.pipeline.add(self.src_bin)
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self.pipeline.set_state(Gst.State.PLAYING)
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self.sink = self.make_bin_from_config('video_output_device',
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'videoscale ! videoconvert ! %s',
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_("video output"))
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self.pipeline.add(self.sink)
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self.src_bin.get_static_pad('src').link(self.p2psession.get_property(
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'sink-pad'))
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# The following is needed for farstream to process ICE requests:
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self.pipeline.set_state(Gst.State.PLAYING)
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def _on_sync_message(self, bus, message):
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if message.get_structure() is None:
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return False
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if message.get_structure().get_name() == 'prepare-window-handle':
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message.src.set_property('force-aspect-ratio', True)
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imagesink = message.src
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if gajim.config.get('video_see_self') and not self.out_xid_set:
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imagesink.set_window_handle(self.out_xid)
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self.out_xid_set = True
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else:
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imagesink.set_window_handle(self.in_xid)
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def get_fallback_src(self):
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# TODO: Use avatar?
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pipeline = 'videotestsrc is-live=true ! video/x-raw,framerate=10/1 ! videoconvert'
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return Gst.parse_bin_from_description(pipeline, True)
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def destroy(self):
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JingleRTPContent.destroy(self)
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self.pipeline.get_bus().disconnect_by_func(self._on_sync_message)
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def get_content(desc):
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if desc['media'] == 'audio':
|
||
return JingleAudio
|
||
elif desc['media'] == 'video':
|
||
return JingleVideo
|
||
|
||
contents[nbxmpp.NS_JINGLE_RTP] = get_content
|