## ## Copyright (C) 2006 Gajim Team ## ## Gajim is free software; you can redistribute it and/or modify ## it under the terms of the GNU General Public License as published ## by the Free Software Foundation; version 3 only. ## ## Gajim is distributed in the hope that it will be useful, ## but WITHOUT ANY WARRANTY; without even the implied warranty of ## MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the ## GNU General Public License for more details. ## ## You should have received a copy of the GNU General Public License ## along with Gajim. If not, see . """ Handles Jingle RTP sessions (XEP 0167) """ from collections import deque from gi.repository import GLib import socket import nbxmpp from gi.repository import Farstream import gi gi.require_version('Gst', '1.0') from gi.repository import Gst gi.require_version('GdkX11', '3.0') from gi.repository import GdkX11 gi.require_version('GstVideo', '1.0') from gi.repository import GstVideo from gi.repository import GLib from common import gajim from common.jingle_transport import JingleTransportICEUDP from common.jingle_content import contents, JingleContent, JingleContentSetupException from common.connection_handlers_events import InformationEvent from common.jingle_session import FailedApplication import logging log = logging.getLogger('gajim.c.jingle_rtp') class JingleRTPContent(JingleContent): def __init__(self, session, media, transport=None): if transport is None: transport = JingleTransportICEUDP(None) JingleContent.__init__(self, session, transport) self.media = media self._dtmf_running = False self.farstream_media = {'audio': Farstream.MediaType.AUDIO, 'video': Farstream.MediaType.VIDEO}[media] self.pipeline = None self.src_bin = None self.stream_failed_once = False self.candidates_ready = False # True when local candidates are prepared self.callbacks['session-initiate'] += [self.__on_remote_codecs] self.callbacks['content-add'] += [self.__on_remote_codecs] self.callbacks['description-info'] += [self.__on_remote_codecs] self.callbacks['content-accept'] += [self.__on_remote_codecs] self.callbacks['session-accept'] += [self.__on_remote_codecs] self.callbacks['session-terminate'] += [self.__stop] self.callbacks['session-terminate-sent'] += [self.__stop] def setup_stream(self, on_src_pad_added): # pipeline and bus self.pipeline = Gst.Pipeline() bus = self.pipeline.get_bus() bus.add_signal_watch() bus.connect('message', self._on_gst_message) # conference self.conference = Gst.ElementFactory.make('fsrtpconference', None) self.pipeline.add(self.conference) self.funnel = None self.p2psession = self.conference.new_session(self.farstream_media) participant = self.conference.new_participant() # FIXME: Consider a workaround, here... # pidgin and telepathy-gabble don't follow the XEP, and it won't work # due to bad controlling-mode params = {'controlling-mode': self.session.weinitiate, 'debug': False} if gajim.config.get('use_stun_server'): stun_server = gajim.config.get('stun_server') if not stun_server and self.session.connection._stun_servers: stun_server = self.session.connection._stun_servers[0]['host'] if stun_server: try: ip = socket.getaddrinfo(stun_server, 0, socket.AF_UNSPEC, socket.SOCK_STREAM)[0][4][0] except socket.gaierror as e: log.warning('Lookup of stun ip failed: %s' % str(e)) else: params['stun-ip'] = ip self.p2pstream = self.p2psession.new_stream(participant, Farstream.StreamDirection.BOTH) self.p2pstream.connect('src-pad-added', on_src_pad_added) self.p2pstream.set_transmitter_ht('nice', params) def is_ready(self): return (JingleContent.is_ready(self) and self.candidates_ready) def make_bin_from_config(self, config_key, pipeline, text): pipeline = pipeline % gajim.config.get(config_key) try: bin = Gst.parse_bin_from_description(pipeline, True) return bin except GLib.GError as e: gajim.nec.push_incoming_event(InformationEvent(None, conn=self.session.connection, level='error', pri_txt=_('%s configuration error') % text.capitalize(), sec_txt=_("Couldn't setup %s. Check your configuration.\n\n" "Pipeline was:\n%s\n\nError was:\n%s") % (text, pipeline, str(e)))) raise JingleContentSetupException def add_remote_candidates(self, candidates): JingleContent.add_remote_candidates(self, candidates) # FIXME: connectivity should not be etablished yet # Instead, it should be etablished after session-accept! if self.sent: self.p2pstream.add_remote_candidates(candidates) def batch_dtmf(self, events): """ Send several DTMF tones """ if self._dtmf_running: raise Exception("There is a DTMF batch already running") events = deque(events) self._dtmf_running = True self._start_dtmf(events.popleft()) GLib.timeout_add(500, self._next_dtmf, events) def _next_dtmf(self, events): self._stop_dtmf() if events: self._start_dtmf(events.popleft()) GLib.timeout_add(500, self._next_dtmf, events) else: self._dtmf_running = False def _start_dtmf(self, event): if event in ('*', '#'): event = {'*': Farstream.DTMFEvent.STAR, '#': Farstream.DTMFEvent.POUND}[event] else: event = int(event) self.p2psession.start_telephony_event(event, 2) def _stop_dtmf(self): self.p2psession.stop_telephony_event() def _fill_content(self, content): content.addChild(nbxmpp.NS_JINGLE_RTP + ' description', attrs={'media': self.media}, payload=list(self.iter_codecs())) def _setup_funnel(self): self.funnel = Gst.ElementFactory.make('funnel', None) self.pipeline.add(self.funnel) self.funnel.link(self.sink) self.sink.set_state(Gst.State.PLAYING) self.funnel.set_state(Gst.State.PLAYING) def _on_src_pad_added(self, stream, pad, codec): if not self.funnel: self._setup_funnel() pad.link(self.funnel.get_request_pad('sink_%u')) def _on_gst_message(self, bus, message): if message.type == Gst.MessageType.ELEMENT: name = message.get_structure().get_name() log.debug('gst element message: %s: %s' % (name, message)) if name == 'farstream-new-active-candidate-pair': pass elif name == 'farstream-recv-codecs-changed': pass elif name == 'farstream-codecs-changed': if self.sent and self.p2psession.props.codecs_without_config: self.send_description_info() if self.transport.remote_candidates: # those lines MUST be done after we get info on our # codecs self.p2pstream.add_remote_candidates( self.transport.remote_candidates) self.transport.remote_candidates = [] self.p2pstream.set_property('direction', Farstream.StreamDirection.BOTH) elif name == 'farstream-local-candidates-prepared': self.candidates_ready = True if self.is_ready(): self.session.on_session_state_changed(self) elif name == 'farstream-new-local-candidate': candidate = self.p2pstream.parse_new_local_candidate(message)[1] self.transport.candidates.append(candidate) if self.sent: # FIXME: Is this case even possible? self.send_candidate(candidate) elif name == 'farstream-component-state-changed': state = message.get_structure().get_value('state') if state == Farstream.StreamState.FAILED: reason = nbxmpp.Node('reason') reason.setTag('failed-transport') self.session.remove_content(self.creator, self.name, reason) elif name == 'farstream-error': log.error('Farstream error #%d!\nMessage: %s' % ( message.get_structure().get_value('error-no'), message.get_structure().get_value('error-msg'))) elif message.type == Gst.MessageType.ERROR: # TODO: Fix it to fallback to videotestsrc anytime an error occur, # or raise an error, Jingle way # or maybe one-sided stream? if not self.stream_failed_once: gajim.nec.push_incoming_event(InformationEvent(None, conn=self.session.connection, level='error', pri_txt=_('GStreamer error'), sec_txt=_('Error: %s\nDebug: ' '%s' % (message.get_structure().get_value('gerror'), message.get_structure().get_value('debug'))))) sink_pad = self.p2psession.get_property('sink-pad') # Remove old source self.src_bin.get_static_pad('src').unlink(sink_pad) self.src_bin.set_state(Gst.State.NULL) self.pipeline.remove(self.src_bin) if not self.stream_failed_once: # Add fallback source self.src_bin = self.get_fallback_src() self.pipeline.add(self.src_bin) self.src_bin.link(sink_pad) self.stream_failed_once = True else: reason = nbxmpp.Node('reason') reason.setTag('failed-application') self.session.remove_content(self.creator, self.name, reason) # Start playing again self.pipeline.set_state(Gst.State.PLAYING) def get_fallback_src(self): return Gst.ElementFactory.make('fakesrc', None) def on_negotiated(self): if self.accepted: if self.p2psession.get_property('codecs'): # those lines MUST be done after we get info on our codecs if self.transport.remote_candidates: self.p2pstream.add_remote_candidates( self.transport.remote_candidates) self.transport.remote_candidates = [] # TODO: Farstream.StreamDirection.BOTH only if senders='both' # self.p2pstream.set_property('direction', # Farstream.StreamDirection.BOTH) JingleContent.on_negotiated(self) def __on_remote_codecs(self, stanza, content, error, action): """ Get peer codecs from what we get from peer """ codecs = [] for codec in content.getTag('description').iterTags('payload-type'): if not codec['id'] or not codec['name'] or not codec['clockrate']: # ignore invalid payload-types continue c = Farstream.Codec.new(int(codec['id']), codec['name'], self.farstream_media, int(codec['clockrate'])) if 'channels' in codec: c.channels = int(codec['channels']) else: c.channels = 1 for p in codec.iterTags('parameter'): c.add_optional_parameter(p['name'], str(p['value'])) codecs.append(c) if codecs: try: self.p2pstream.set_remote_codecs(codecs) except GLib.Error: raise FailedApplication def iter_codecs(self): codecs = self.p2psession.props.codecs_without_config for codec in codecs: attrs = {'name': codec.encoding_name, 'id': codec.id, 'channels': codec.channels} if codec.clock_rate: attrs['clockrate'] = codec.clock_rate if codec.optional_params: payload = list(nbxmpp.Node('parameter', {'name': p.name, 'value': p.value}) for p in codec.optional_params) else: payload = [] yield nbxmpp.Node('payload-type', attrs, payload) def __stop(self, *things): self.pipeline.set_state(Gst.State.NULL) def __del__(self): self.__stop() def destroy(self): JingleContent.destroy(self) self.p2pstream.disconnect_by_func(self._on_src_pad_added) self.pipeline.get_bus().disconnect_by_func(self._on_gst_message) class JingleAudio(JingleRTPContent): """ Jingle VoIP sessions consist of audio content transported over an ICE UDP protocol """ def __init__(self, session, transport=None): JingleRTPContent.__init__(self, session, 'audio', transport) self.setup_stream() def set_mic_volume(self, vol): """ vol must be between 0 ans 1 """ self.mic_volume.set_property('volume', vol) def set_out_volume(self, vol): """ vol must be between 0 ans 1 """ self.out_volume.set_property('volume', vol) def setup_stream(self): JingleRTPContent.setup_stream(self, self._on_src_pad_added) # Configure SPEEX # Workaround for psi (not needed since rev # 147aedcea39b43402fe64c533d1866a25449888a): # place 16kHz before 8kHz, as buggy psi versions will take in # account only the first codec codecs = [Farstream.Codec.new(Farstream.CODEC_ID_ANY, 'SPEEX', Farstream.MediaType.AUDIO, 16000), Farstream.Codec.new(Farstream.CODEC_ID_ANY, 'SPEEX', Farstream.MediaType.AUDIO, 8000)] self.p2psession.set_codec_preferences(codecs) # the local parts # TODO: Add queues? self.src_bin = self.make_bin_from_config('audio_input_device', '%s ! audioconvert', _("audio input")) self.sink = self.make_bin_from_config('audio_output_device', 'audioconvert ! volume name=gajim_out_vol ! %s', _("audio output")) self.mic_volume = self.src_bin.get_by_name('gajim_vol') self.out_volume = self.sink.get_by_name('gajim_out_vol') # link gst elements self.pipeline.add(self.sink) self.pipeline.add(self.src_bin) self.src_bin.get_static_pad('src').link(self.p2psession.get_property( 'sink-pad')) # The following is needed for farstream to process ICE requests: self.pipeline.set_state(Gst.State.PLAYING) class JingleVideo(JingleRTPContent): def __init__(self, session, transport=None, in_xid=0, out_xid=0): JingleRTPContent.__init__(self, session, 'video', transport) self.in_xid = in_xid self.out_xid = out_xid self.out_xid_set = False self.setup_stream() def setup_stream(self): # TODO: Everything is not working properly: # sometimes, one window won't show up, # sometimes it'll freeze... JingleRTPContent.setup_stream(self, self._on_src_pad_added) bus = self.pipeline.get_bus() bus.enable_sync_message_emission() bus.connect('sync-message::element', self._on_sync_message) # the local parts if gajim.config.get('video_framerate'): framerate = 'videorate ! video/x-raw,framerate=%s ! ' % \ gajim.config.get('video_framerate') else: framerate = '' try: w, h = gajim.config.get('video_size').split('x') except: w = h = None if w and h: video_size = 'video/x-raw,width=%s,height=%s ! ' % (w, h) else: video_size = '' if gajim.config.get('video_see_self'): tee = '! tee name=t ! queue ! videoscale ! ' + \ 'video/x-raw,width=160,height=120 ! videoconvert ! ' + \ '%s t. ! queue ' % gajim.config.get( 'video_output_device') else: tee = '' self.src_bin = self.make_bin_from_config('video_input_device', '%%s %s! %svideoscale ! %svideoconvert' % (tee, framerate, video_size), _("video input")) self.pipeline.add(self.src_bin) self.pipeline.set_state(Gst.State.PLAYING) self.sink = self.make_bin_from_config('video_output_device', 'videoscale ! videoconvert ! %s', _("video output")) self.pipeline.add(self.sink) self.src_bin.get_static_pad('src').link(self.p2psession.get_property( 'sink-pad')) # The following is needed for farstream to process ICE requests: self.pipeline.set_state(Gst.State.PLAYING) def _on_sync_message(self, bus, message): if message.get_structure() is None: return False if message.get_structure().get_name() == 'prepare-window-handle': message.src.set_property('force-aspect-ratio', True) imagesink = message.src if gajim.config.get('video_see_self') and not self.out_xid_set: imagesink.set_window_handle(self.out_xid) self.out_xid_set = True else: imagesink.set_window_handle(self.in_xid) def get_fallback_src(self): # TODO: Use avatar? pipeline = 'videotestsrc is-live=true ! video/x-raw,framerate=10/1 ! videoconvert' return Gst.parse_bin_from_description(pipeline, True) def destroy(self): JingleRTPContent.destroy(self) self.pipeline.get_bus().disconnect_by_func(self._on_sync_message) def get_content(desc): if desc['media'] == 'audio': return JingleAudio elif desc['media'] == 'video': return JingleVideo contents[nbxmpp.NS_JINGLE_RTP] = get_content