New class for video sessions, added an end_session method, and minor changes
This commit is contained in:
Thibaut GIRKA
parent
8b4cd1b368
commit
163b01e113
4 changed files with 71 additions and 25 deletions
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@ -198,7 +198,7 @@ gajim_common_features = [xmpp.NS_BYTESTREAM, xmpp.NS_SI, xmpp.NS_FILE,
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'sslc2s', 'stringprep', xmpp.NS_PING, xmpp.NS_TIME_REVISED, xmpp.NS_SSN,
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xmpp.NS_MOOD, xmpp.NS_ACTIVITY, xmpp.NS_NICK, xmpp.NS_ROSTERX,
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xmpp.NS_JINGLE, xmpp.NS_JINGLE_RTP, xmpp.NS_JINGLE_RTP_AUDIO,
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xmpp.NS_JINGLE_ICE_UDP]
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xmpp.NS_JINGLE_RTP_VIDEO, xmpp.NS_JINGLE_ICE_UDP]
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# Optional features gajim supports per account
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gajim_optional_features = {}
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@ -148,6 +148,14 @@ class JingleSession(object):
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reason.addChild('decline')
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self.__sessionTerminate(reason)
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def end_session(self):
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reason = xmpp.Node('reason')
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if self.state == JingleStates.active:
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reason.addChild('success')
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else:
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reason.addChild('abort')
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self.__sessionTerminate(reason)
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''' Middle-level functions to manage contents. Handle local content
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cache and send change notifications. '''
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def addContent(self, name, content, creator='we'):
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@ -337,12 +345,15 @@ class JingleSession(object):
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desc_ns = element.getTag('description').getNamespace()
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media = element.getTag('description')['media']
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tran_ns = element.getTag('transport').getNamespace()
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if desc_ns == xmpp.NS_JINGLE_RTP and media == 'audio':
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if desc_ns == xmpp.NS_JINGLE_RTP and media in ('audio', 'video'):
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contents_ok = True
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if tran_ns == xmpp.NS_JINGLE_ICE_UDP:
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# we've got voip content
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self.addContent(element['name'], JingleVoIP(self), 'peer')
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contents.append(('VOIP',))
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if media == 'audio':
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self.addContent(element['name'], JingleVoIP(self), 'peer')
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else:
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self.addContent(element['name'], JingleVideo(self), 'peer')
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contents.append((media,))
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transports_ok = True
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# If there's no content we understand...
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@ -600,6 +611,8 @@ class JingleRTPContent(JingleContent):
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def __init__(self, session, media, node=None):
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JingleContent.__init__(self, session, node)
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self.media = media
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self.farsight_media = {'audio': farsight.MEDIA_TYPE_AUDIO,
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'video': farsight.MEDIA_TYPE_VIDEO}[media]
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self.got_codecs = False
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self.callbacks['content-accept'] += [self.__getRemoteCodecsCB]
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@ -621,6 +634,15 @@ class JingleRTPContent(JingleContent):
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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.farsight_media)
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participant = self.conference.new_participant(self.session.peerjid)
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params = {'controlling-mode': self.session.weinitiate,# 'debug': False}
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'stun-ip': '69.0.208.27', 'debug': False}
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self.p2pstream = self.p2psession.new_stream(participant,
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farsight.DIRECTION_RECV, 'nice', params)
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def _fillContent(self, content):
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content.addChild(xmpp.NS_JINGLE_RTP + ' description',
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attrs={'media': self.media}, payload=self.iterCodecs())
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@ -646,7 +668,7 @@ class JingleRTPContent(JingleContent):
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elif name == 'farsight-component-state-changed':
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state = message.structure['state']
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print message.structure['component'], state
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if state==farsight.STREAM_STATE_CONNECTED:
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if state==farsight.STREAM_STATE_READY:
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self.negotiated = True
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#TODO: farsight.DIRECTION_BOTH only if senders='both'
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self.p2pstream.set_property('direction', farsight.DIRECTION_BOTH)
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@ -667,7 +689,7 @@ class JingleRTPContent(JingleContent):
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codecs = []
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for codec in content.getTag('description').iterTags('payload-type'):
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c = farsight.Codec(int(codec['id']), codec['name'],
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farsight.MEDIA_TYPE_AUDIO, int(codec['clockrate']))
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self.farsight_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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@ -709,8 +731,6 @@ class JingleVoIP(JingleRTPContent):
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def __init__(self, session, node=None):
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JingleRTPContent.__init__(self, session, 'audio', node)
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self.got_codecs = False
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self.setupStream()
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@ -718,13 +738,6 @@ class JingleVoIP(JingleRTPContent):
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def setupStream(self):
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JingleRTPContent.setupStream(self)
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# the network part
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participant = self.conference.new_participant(self.session.peerjid)
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params = {'controlling-mode': self.session.weinitiate,# 'debug': False}
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'stun-ip': '69.0.208.27', 'debug': False}
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self.p2psession = self.conference.new_session(farsight.MEDIA_TYPE_AUDIO)
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# Configure SPEEX
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#FIXME: codec ID is an important thing for psi (and pidgin?)
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# So, if it doesn't work with pidgin or psi, LOOK AT THIS
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@ -734,9 +747,6 @@ class JingleVoIP(JingleRTPContent):
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farsight.MEDIA_TYPE_AUDIO, 16000)]
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self.p2psession.set_codec_preferences(codecs)
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self.p2pstream = self.p2psession.new_stream(participant,
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farsight.DIRECTION_RECV, 'nice', params)
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# the local parts
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# TODO: use gconfaudiosink?
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# sink = get_first_gst_element(['alsasink', 'osssink', 'autoaudiosink'])
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@ -744,19 +754,17 @@ class JingleVoIP(JingleRTPContent):
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sink.set_property('sync', False)
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#sink.set_property('latency-time', 20000)
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#sink.set_property('buffer-time', 80000)
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self.pipeline.add(sink)
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# TODO: use gconfaudiosrc?
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self.src_mic = gst.element_factory_make('alsasrc')
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self.src_mic.set_property('blocksize', 320)
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self.pipeline.add(self.src_mic)
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src_mic = gst.element_factory_make('alsasrc')
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src_mic.set_property('blocksize', 320)
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self.mic_volume = gst.element_factory_make('volume')
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self.mic_volume.set_property('volume', 1)
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self.pipeline.add(self.mic_volume)
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# link gst elements
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self.src_mic.link(self.mic_volume)
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self.pipeline.add(sink, src_mic, self.mic_volume)
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src_mic.link(self.mic_volume)
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def src_pad_added (stream, pad, codec):
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if not self.funnel:
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@ -774,6 +782,43 @@ class JingleVoIP(JingleRTPContent):
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# The following is needed for farsight 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, node=None):
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JingleRTPContent.__init__(self, session, 'video', node)
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self.setupStream()
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''' Things to control the gstreamer's pipeline '''
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def setupStream(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.setupStream(self)
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# the local parts
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src_vid = gst.element_factory_make('videotestsrc')
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videoscale = gst.element_factory_make('videoscale')
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caps = gst.element_factory_make('capsfilter')
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caps.set_property('caps', gst.caps_from_string('video/x-raw-yuv, width=320, height=240'))
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colorspace = gst.element_factory_make('ffmpegcolorspace')
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self.pipeline.add(src_vid, videoscale, caps, colorspace)
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gst.element_link_many(src_vid, videoscale, caps, colorspace)
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def src_pad_added (stream, pad, codec):
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if not self.funnel:
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self.funnel = gst.element_factory_make('fsfunnel')
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self.pipeline.add(self.funnel)
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videosink = gst.element_factory_make('xvimagesink')
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self.pipeline.add(videosink)
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self.funnel.set_state (gst.STATE_PLAYING)
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videosink.set_state(gst.STATE_PLAYING)
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self.funnel.link(videosink)
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pad.link(self.funnel.get_pad('sink%d'))
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colorspace.get_pad('src').link(self.p2psession.get_property('sink-pad'))
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self.p2pstream.connect('src-pad-added', src_pad_added)
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# The following is needed for farsight to process ICE requests:
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self.pipeline.set_state(gst.STATE_PLAYING)
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class ConnectionJingle(object):
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''' This object depends on that it is a part of Connection class. '''
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@ -67,6 +67,7 @@ NS_JINGLE ='urn:xmpp:jingle:1' # XEP-01
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NS_JINGLE_ERRORS='urn:xmpp:jingle:errors:1' # XEP-0166
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NS_JINGLE_RTP ='urn:xmpp:jingle:apps:rtp:1' # XEP-0167
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NS_JINGLE_RTP_AUDIO='urn:xmpp:jingle:apps:rtp:audio' # XEP-0167
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NS_JINGLE_RTP_VIDEO='urn:xmpp:jingle:apps:rtp:video' # XEP-0167
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NS_JINGLE_RAW_UDP='urn:xmpp:jingle:transports:raw-udp:1' # XEP-0177
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NS_JINGLE_ICE_UDP='urn:xmpp:jingle:transports:ice-udp:1' # XEP-0176
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NS_LAST ='jabber:iq:last'
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@ -2087,7 +2087,7 @@ class Interface:
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content_types = set(c[0] for c in contents)
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# check type of jingle session
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if 'VOIP' in content_types:
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if 'audio' in content_types or 'video' in content_types:
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# a voip session...
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# we now handle only voip, so the only thing we will do here is
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# not to return from function
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