gajim-plural/src/common/jingle_rtp.py

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##
## Copyright (C) 2006 Gajim Team
##
## This program 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 2 only.
##
## This program 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.
##
"""
Handles Jingle RTP sessions (XEP 0167)
"""
from collections import deque
import gobject
import socket
import xmpp
import farstream, gst
from glib import GError
import gajim
from jingle_transport import JingleTransportICEUDP
from jingle_content import contents, JingleContent, JingleContentSetupException
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from connection_handlers_events import InformationEvent
import logging
log = logging.getLogger('gajim.c.jingle_rtp')
class JingleRTPContent(JingleContent):
def __init__(self, session, media, transport=None):
if transport is None:
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transport = JingleTransportICEUDP(None)
JingleContent.__init__(self, session, transport)
self.media = media
self._dtmf_running = False
self.farstream_media = {'audio': farstream.MEDIA_TYPE_AUDIO,
'video': farstream.MEDIA_TYPE_VIDEO}[media]
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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.element_factory_make('fsrtpconference')
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, (errnum, errstr):
log.warn('Lookup of stun ip failed: %s' % errstr)
else:
params['stun-ip'] = ip
self.p2pstream = self.p2psession.new_stream(participant,
farstream.DIRECTION_RECV)
self.p2pstream.connect('src-pad-added', on_src_pad_added)
self.p2pstream.set_transmitter('nice', params)
def is_ready(self):
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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 GError, error_str:
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,
error_str)))
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:
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raise Exception("There is a DTMF batch already running")
events = deque(events)
self._dtmf_running = True
self._start_dtmf(events.popleft())
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gobject.timeout_add(500, self._next_dtmf, events)
def _next_dtmf(self, events):
self._stop_dtmf()
if events:
self._start_dtmf(events.popleft())
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gobject.timeout_add(500, self._next_dtmf, events)
else:
self._dtmf_running = False
def _start_dtmf(self, event):
if event in ('*', '#'):
event = {'*': farstream.DTMF_EVENT_STAR,
'#': farstream.DTMF_EVENT_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(xmpp.NS_JINGLE_RTP + ' description',
attrs={'media': self.media}, payload=self.iter_codecs())
def _setup_funnel(self):
self.funnel = gst.element_factory_make('fsfunnel')
self.pipeline.add(self.funnel)
self.funnel.set_state(gst.STATE_PLAYING)
self.sink.set_state(gst.STATE_PLAYING)
self.funnel.link(self.sink)
def _on_src_pad_added(self, stream, pad, codec):
if not self.funnel:
self._setup_funnel()
pad.link(self.funnel.get_pad('sink%d'))
def _on_gst_message(self, bus, message):
if message.type == gst.MESSAGE_ELEMENT:
name = message.structure.get_name()
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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.get_property('codecs'):
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self.send_description_info()
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 = message.structure['candidate']
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.structure['state']
if state == farstream.STREAM_STATE_FAILED:
reason = xmpp.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' % (
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message.structure['error-no'],
message.structure['error-msg']))
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elif message.type == gst.MESSAGE_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.structure['gerror'],
message.structure['debug']))))
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sink_pad = self.p2psession.get_property('sink-pad')
# Remove old source
self.src_bin.get_pad('src').unlink(sink_pad)
self.src_bin.set_state(gst.STATE_NULL)
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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.get_pad('src').link(sink_pad)
self.stream_failed_once = True
else:
reason = xmpp.Node('reason')
reason.setTag('failed-application')
self.session.remove_content(self.creator, self.name, reason)
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# Start playing again
self.pipeline.set_state(gst.STATE_PLAYING)
def get_fallback_src(self):
return gst.element_factory_make('fakesrc')
def on_negotiated(self):
if self.accepted:
if self.transport.remote_candidates:
self.p2pstream.add_remote_candidates(self.transport.remote_candidates)
self.transport.remote_candidates = []
# TODO: farstream.DIRECTION_BOTH only if senders='both'
self.p2pstream.set_property('direction', farstream.DIRECTION_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(int(codec['id']), codec['name'],
self.farstream_media, int(codec['clockrate']))
if 'channels' in codec:
c.channels = int(codec['channels'])
else:
c.channels = 1
c.optional_params = [(str(p['name']), str(p['value'])) for p in \
codec.iterTags('parameter')]
codecs.append(c)
if codecs:
# FIXME: Handle this case:
# glib.GError: There was no intersection between the remote codecs and
# the local ones
self.p2pstream.set_remote_codecs(codecs)
def iter_codecs(self):
codecs = self.p2psession.get_property('codecs')
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 = (xmpp.Node('parameter', {'name': name, 'value': value})
for name, value in codec.optional_params)
else:
payload = ()
yield xmpp.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(farstream.CODEC_ID_ANY, 'SPEEX',
farstream.MEDIA_TYPE_AUDIO, 16000),
farstream.Codec(farstream.CODEC_ID_ANY, 'SPEEX',
farstream.MEDIA_TYPE_AUDIO, 8000)]
self.p2psession.set_codec_preferences(codecs)
# the local parts
# TODO: Add queues?
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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"))
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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
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self.pipeline.add(self.sink, self.src_bin)
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self.src_bin.get_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):
JingleRTPContent.__init__(self, session, 'video', transport)
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)
# the local parts
if gajim.config.get('video_framerate'):
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framerate = 'videorate ! video/x-raw-yuv,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:
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video_size = 'video/x-raw-yuv,width=%s,height=%s ! ' % (w, h)
else:
video_size = ''
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self.src_bin = self.make_bin_from_config('video_input_device',
'%%s ! %svideoscale ! %sffmpegcolorspace' % (framerate, video_size),
_("video input"))
#caps = gst.element_factory_make('capsfilter')
#caps.set_property('caps', gst.caps_from_string('video/x-raw-yuv, width=320, height=240'))
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self.pipeline.add(self.src_bin)#, caps)
#src_bin.link(caps)
self.sink = self.make_bin_from_config('video_output_device',
'videoscale ! ffmpegcolorspace ! %s force-aspect-ratio=True',
_("video output"))
self.pipeline.add(self.sink)
self.src_bin.get_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)
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def get_fallback_src(self):
# TODO: Use avatar?
pipeline = 'videotestsrc is-live=true ! video/x-raw-yuv,framerate=10/1 ! ffmpegcolorspace'
return gst.parse_bin_from_description(pipeline, True)
def get_content(desc):
if desc['media'] == 'audio':
return JingleAudio
elif desc['media'] == 'video':
return JingleVideo
contents[xmpp.NS_JINGLE_RTP] = get_content