def __init__(self, link_name, redis_host):
"""
Set up a new LinkConfig instance - needs to know the link name and
configuration host.
"""
self.int_properties = [
'port', 'jitter_buffer', 'opus_framesize', 'opus_complexity',
'bitrate', 'opus_loss_expectation'
]
self.bool_properties = ['opus_dtx', 'opus_fec', 'multicast']
self.link_name = link_name
self.redis_host = redis_host
self.logger_factory = LoggerFactory()
self.logger = self.logger_factory.getLogger('link.%s.config' %
self.link_name)
self.logger.info("Connecting to configuration host %s" %
self.redis_host)
self.redis = None
while True:
try:
self.redis = redis.StrictRedis(host=self.redis_host,
charset="utf-8",
decode_responses=True)
break
except Exception as e:
self.logger.error(
"Unable to connect to configuration host! Retrying. (%s)" %
e)
time.sleep(0.1)
def __init__(self, node_name, interface_name='default'):
self.interface_name = interface_name
self.node_name = node_name
self.logger_factory = LoggerFactory()
self.logger = self.logger_factory.getLogger(
'node.%s.audio_interface.%s' %
(self.node_name, self.interface_name))
self.config = dict()
def __init__(self, node_name, link_config, audio_interface):
"""Sets up a new RTP receiver"""
self.link_config = link_config
self.audio_interface = audio_interface
self.logger_factory = LoggerFactory()
self.logger = self.logger_factory.getLogger('node.%s.link.%s.%s' % (node_name, self.link_config.name, self.audio_interface.mode))
self.logger.info('Creating reception pipeline')
self.build_pipeline()
class LinkConfig(object):
"""
The LinkConfig class encapsulates link configuration. It's genderless;
a TX node should be able to set up a new link and an RX node should be
able (once the TX node has specified the port caps) to configure itself
to receive the stream using the data and methods in this config.
"""
def __init__(self, link_name, redis_host):
"""
Set up a new LinkConfig instance - needs to know the link name and
configuration host.
"""
self.link_name = link_name
self.redis_host = redis_host
self.logger_factory = LoggerFactory()
self.logger = self.logger_factory.getLogger('link.%s.config' %
self.link_name)
self.logger.info("Connecting to configuration host %s" %
self.redis_host)
self.redis = None
while True:
try:
self.redis = redis.StrictRedis(self.redis_host)
break
except Exception, e:
self.logger.error(
"Unable to connect to configuration host! Retrying. (%s)" %
e)
time.sleep(0.1)
def __init__(self, node_name, interface_name='default'):
self.interface_name = interface_name
self.node_name = node_name
self.logger_factory = LoggerFactory()
self.logger = self.logger_factory.getLogger('node.%s.audio_interface.%s'
% (self.node_name, self.interface_name))
self.config = dict()
class LinkConfig(object):
"""
The LinkConfig class encapsulates link configuration. It's genderless;
a TX node should be able to set up a new link and an RX node should be
able (once the TX node has specified the port caps) to configure itself
to receive the stream using the data and methods in this config.
"""
def __init__(self, link_name, redis_host):
"""
Set up a new LinkConfig instance - needs to know the link name and
configuration host.
"""
self.link_name = link_name
self.redis_host = redis_host
self.logger_factory = LoggerFactory()
self.logger = self.logger_factory.getLogger('link.%s.config' % self.link_name)
self.logger.info("Connecting to configuration host %s" % self.redis_host)
self.redis = None
while True:
try:
self.redis = redis.StrictRedis(self.redis_host)
break
except Exception, e:
self.logger.error(
"Unable to connect to configuration host! Retrying. (%s)"
% e
)
time.sleep(0.1)
def __init__(self, node_name, link_config, audio_interface):
"""Sets up a new RTP transmitter"""
self.link_config = link_config
self.audio_interface = audio_interface
self.logger_factory = LoggerFactory()
self.logger = self.logger_factory.getLogger('node.%s.link.%s.%s' % (node_name, self.link_config.name, self.audio_interface.mode))
self.logger.info('Creating transmission pipeline')
self.build_pipeline()
def __init__(self, link_name, redis_host):
"""
Set up a new LinkConfig instance - needs to know the link name and
configuration host.
"""
self.link_name = link_name
self.redis_host = redis_host
self.logger_factory = LoggerFactory()
self.logger = self.logger_factory.getLogger('link.%s.config' %
self.link_name)
self.logger.info("Connecting to configuration host %s" %
self.redis_host)
self.redis = None
while True:
try:
self.redis = redis.StrictRedis(self.redis_host)
break
except Exception, e:
self.logger.error(
"Unable to connect to configuration host! Retrying. (%s)" %
e)
time.sleep(0.1)
class AudioInterface(object):
"""
The AudioInterface class describes an audio interface on a Node.
The configuration is not shared across the network. The type property of
an AudioInterface should define the mode of link operation.
"""
def __init__(self, node_name, interface_name='default'):
self.interface_name = interface_name
self.node_name = node_name
self.logger_factory = LoggerFactory()
self.logger = self.logger_factory.getLogger('node.%s.audio_interface.%s'
% (self.node_name, self.interface_name))
self.config = dict()
def set(self, key, value):
"""Set a config value"""
self.logger.debug("Set %s to %s" % (key, value))
self.config[key] = value
def get(self, key):
"""Get a config value"""
value = self.config[key]
self.logger.debug("Fetched %s, got %s" % (key, value))
return value
def __getattr__(self, key):
"""Convenience method to access get"""
return self.get(key)
def set_from_argparse(self, opts):
"""Set up the audio interface from argparse options"""
self.set("mode", opts.mode)
if opts.mode == "tx":
self.set("type", opts.audio_input)
self.set("samplerate", opts.samplerate)
elif opts.mode == "rx":
self.set("type", opts.audio_output)
if self.get("type") == "alsa":
self.set("alsa_device", opts.alsa_device)
elif self.get("type") == "jack":
if opts.jack_auto is not False:
self.set("jack_auto", opts.jack_auto)
if opts.jack_name is not None:
self.set("jack_name", opts.jack_name)
else:
self.set("jack_name", "openob")
class AudioInterface(object):
"""
The AudioInterface class describes an audio interface on a Node.
The configuration is not shared across the network. The type property of
an AudioInterface should define the mode of link operation.
"""
def __init__(self, node_name, interface_name='default'):
self.interface_name = interface_name
self.node_name = node_name
self.logger_factory = LoggerFactory()
self.logger = self.logger_factory.getLogger(
'node.%s.audio_interface.%s' %
(self.node_name, self.interface_name))
self.config = dict()
def set(self, key, value):
"""Set a config value"""
self.logger.debug("Set %s to %s" % (key, value))
self.config[key] = value
def get(self, key):
"""Get a config value"""
value = self.config[key]
self.logger.debug("Fetched %s, got %s" % (key, value))
return value
def __getattr__(self, key):
"""Convenience method to access get"""
return self.get(key)
def set_from_argparse(self, opts):
"""Set up the audio interface from argparse options"""
self.set("mode", opts.mode)
if opts.mode == "tx":
self.set("type", opts.audio_input)
self.set("samplerate", opts.samplerate)
elif opts.mode == "rx":
self.set("type", opts.audio_output)
if self.get("type") == "alsa":
self.set("alsa_device", opts.alsa_device)
elif self.get("type") == "jack":
self.set("jack_auto", opts.jack_auto)
if opts.jack_name is not None:
self.set("jack_name", opts.jack_name)
else:
self.set("jack_name", "openob")
if opts.jack_port_pattern is not None:
self.set("jack_port_pattern", opts.jack_port_pattern)
def __init__(self, link_name, redis_host):
"""
Set up a new LinkConfig instance - needs to know the link name and
configuration host.
"""
self.link_name = link_name
self.redis_host = redis_host
self.logger_factory = LoggerFactory()
self.logger = self.logger_factory.getLogger('link.%s.config' % self.link_name)
self.logger.info("Connecting to configuration host %s" % self.redis_host)
self.redis = None
while True:
try:
self.redis = redis.StrictRedis(self.redis_host)
break
except Exception, e:
self.logger.error(
"Unable to connect to configuration host! Retrying. (%s)"
% e
)
time.sleep(0.1)
def __init__(self, link_name, redis_host):
"""
Set up a new LinkConfig instance - needs to know the link name and
configuration host.
"""
self.int_properties = ['port', 'jitter_buffer', 'opus_framesize', 'opus_complexity', 'bitrate', 'opus_loss_expectation']
self.bool_properties = ['opus_dtx', 'opus_fec', 'multicast']
self.link_name = link_name
self.redis_host = redis_host
self.logger_factory = LoggerFactory()
self.logger = self.logger_factory.getLogger('link.%s.config' % self.link_name)
self.logger.info("Connecting to configuration host %s" % self.redis_host)
self.redis = None
while True:
try:
self.redis = redis.StrictRedis(self.redis_host)
break
except Exception as e:
self.logger.error(
"Unable to connect to configuration host! Retrying. (%s)"
% e
)
time.sleep(0.1)
class RTPTransmitter(object):
def __init__(self, node_name, link_config, audio_interface):
"""Sets up a new RTP transmitter"""
self.started = False
self.caps = 'None'
self.pipeline = gst.Pipeline("tx")
self.bus = self.pipeline.get_bus()
self.bus.connect("message", self.on_message)
self.link_config = link_config
self.audio_interface = audio_interface
self.logger_factory = LoggerFactory()
self.logger = self.logger_factory.getLogger(
'node.%s.link.%s.%s' %
(node_name, self.link_config.name, self.audio_interface.mode))
self.logger.info("Creating RTP transmission pipeline")
# Audio input
if self.audio_interface.type == 'auto':
self.source = gst.element_factory_make('autoaudiosrc')
elif self.audio_interface.type == 'alsa':
self.source = gst.element_factory_make('alsasrc')
self.source.set_property('device',
self.audio_interface.alsa_device)
elif self.audio_interface.type == 'jack':
self.source = gst.element_factory_make("jackaudiosrc")
if self.audio_interface.jack_auto:
self.source.set_property('connect', 'auto')
else:
self.source.set_property('connect', 'none')
self.source.set_property('buffer-time', 50000)
self.source.set_property('name', self.audio_interface.jack_name)
self.source.set_property('client-name',
self.audio_interface.jack_name)
# Audio resampling and conversion
self.audioresample = gst.element_factory_make("audioresample")
self.audioconvert = gst.element_factory_make("audioconvert")
self.audioresample.set_property('quality', 9) # SRC
# Encoding and payloading
if self.link_config.encoding == 'opus':
self.encoder = gst.element_factory_make("opusenc", "encoder")
self.encoder.set_property('bitrate',
int(self.link_config.bitrate) * 1000)
self.encoder.set_property('tolerance', 80000000)
self.encoder.set_property('frame-size',
self.link_config.opus_framesize)
self.encoder.set_property('complexity',
int(self.link_config.opus_complexity))
self.encoder.set_property('inband-fec', self.link_config.opus_fec)
self.encoder.set_property(
'packet-loss-percentage',
int(self.link_config.opus_loss_expectation))
self.encoder.set_property('dtx', self.link_config.opus_dtx)
print(self.encoder.get_properties('bitrate', 'dtx', 'inband-fec'))
self.payloader = gst.element_factory_make("rtpopuspay",
"payloader")
elif self.link_config.encoding == 'pcm':
# we have no encoder for PCM operation
self.payloader = gst.element_factory_make("rtpL16pay", "payloader")
else:
self.logger.critical("Unknown encoding type %s" %
self.link_config.encoding)
# TODO: Add a tee here, and sort out creating multiple UDP sinks for multipath
# Now the RTP bits
# We'll send audio out on this
self.udpsink_rtpout = gst.element_factory_make("udpsink",
"udpsink_rtp")
self.udpsink_rtpout.set_property('host',
self.link_config.receiver_host)
self.udpsink_rtpout.set_property('port', self.link_config.port)
self.logger.info(
'Set receiver to %s:%i' %
(self.link_config.receiver_host, self.link_config.port))
if self.link_config.multicast:
self.udpsink_rtpout.set_property('auto_multicast', True)
self.logger.info('Multicast mode enabled')
# Our RTP manager
self.rtpbin = gst.element_factory_make("gstrtpbin", "gstrtpbin")
self.rtpbin.set_property('latency', 0)
# Our level monitor
self.level = gst.element_factory_make("level")
self.level.set_property('message', True)
self.level.set_property('interval', 1000000000)
# Add a capsfilter to allow specification of input sample rate
self.capsfilter = gst.element_factory_make("capsfilter")
# Add to the pipeline
self.pipeline.add(self.source, self.capsfilter, self.audioresample,
self.audioconvert, self.payloader,
self.udpsink_rtpout, self.rtpbin, self.level)
if self.link_config.encoding != 'pcm':
# Only add the encoder if we're not in PCM mode
self.pipeline.add(self.encoder)
# Decide which format to apply to the capsfilter (Jack uses float)
if self.audio_interface.type == 'jack':
data_type = 'audio/x-raw-float'
else:
data_type = 'audio/x-raw-int'
# if audio_rate has been specified, then add that to the capsfilter
if self.audio_interface.samplerate != 0:
self.capsfilter.set_property(
"caps",
gst.Caps('%s, channels=2, rate=%d' %
(data_type, self.audio_interface.samplerate)))
else:
self.capsfilter.set_property(
"caps", gst.Caps('%s, channels=2' % data_type))
# Then continue linking the pipeline together
gst.element_link_many(self.source, self.capsfilter, self.level,
self.audioresample, self.audioconvert)
# Now we get to link this up to our encoder/payloader
if self.link_config.encoding != 'pcm':
gst.element_link_many(self.audioconvert, self.encoder,
self.payloader)
else:
gst.element_link_many(self.audioconvert, self.payloader)
# And now the RTP bits
self.payloader.link_pads('src', self.rtpbin, 'send_rtp_sink_0')
self.rtpbin.link_pads('send_rtp_src_0', self.udpsink_rtpout, 'sink')
# self.udpsrc_rtcpin.link_pads('src', self.rtpbin, 'recv_rtcp_sink_0')
# # RTCP SRs
# self.rtpbin.link_pads('send_rtcp_src_0', self.udpsink_rtcpout, 'sink')
# Connect our bus up
self.bus.add_signal_watch()
self.bus.connect('message', self.on_message)
def run(self):
self.pipeline.set_state(gst.STATE_PLAYING)
while self.caps == 'None':
self.logger.debug(self.udpsink_rtpout.get_state())
self.caps = str(
self.udpsink_rtpout.get_pad('sink').get_property('caps'))
# Fix for gstreamer bug in rtpopuspay fixed in GST-plugins-bad
# 50140388d2b62d32dd9d0c071e3051ebc5b4083b, bug 686547
if self.link_config.encoding == 'opus':
self.caps = re.sub(r'(caps=.+ )', '', self.caps)
if self.caps == 'None':
self.logger.warn("Waiting for audio interface/caps")
time.sleep(0.1)
def loop(self):
try:
self.loop = gobject.MainLoop()
self.loop.run()
except Exception as e:
self.logger.exception(
"Encountered a problem in the MainLoop, tearing down the pipeline: %s"
% e)
self.pipeline.set_state(gst.STATE_NULL)
def on_message(self, bus, message):
if message.type == gst.MESSAGE_ELEMENT:
if message.structure.get_name() == 'level':
if self.started is False:
self.started = True
#gst.DEBUG_BIN_TO_DOT_FILE(self.pipeline, gst.DEBUG_GRAPH_SHOW_ALL, 'tx-graph')
#self.logger.debug(self.source.get_property('actual-buffer-time'))
if len(message.structure['peak']) == 1:
self.logger.info("Started mono audio transmission")
else:
self.logger.info("Started stereo audio transmission")
return True
def get_caps(self):
return self.caps
class LinkConfig(object):
"""
The LinkConfig class encapsulates link configuration. It's genderless;
a TX node should be able to set up a new link and an RX node should be
able (once the TX node has specified the port caps) to configure itself
to receive the stream using the data and methods in this config.
"""
def __init__(self, link_name, redis_host):
"""
Set up a new LinkConfig instance - needs to know the link name and
configuration host.
"""
self.int_properties = ['port', 'jitter_buffer', 'opus_framesize', 'opus_complexity', 'bitrate', 'opus_loss_expectation']
self.bool_properties = ['opus_dtx', 'opus_fec', 'multicast']
self.link_name = link_name
self.redis_host = redis_host
self.logger_factory = LoggerFactory()
self.logger = self.logger_factory.getLogger('link.%s.config' % self.link_name)
self.logger.info("Connecting to configuration host %s" % self.redis_host)
self.redis = None
while True:
try:
self.redis = redis.StrictRedis(self.redis_host)
break
except Exception as e:
self.logger.error(
"Unable to connect to configuration host! Retrying. (%s)"
% e
)
time.sleep(0.1)
def blocking_get(self, key):
"""Get a value, blocking until it's not None if needed"""
while True:
value = self.get(key)
if value is not None:
self.logger.debug("Fetched (blocking) %s, got %s" % (key, value))
return value
time.sleep(0.1)
def set(self, key, value):
"""Set a value in the config store"""
scoped_key = self.scoped_key(key)
self.redis.set(scoped_key, value)
self.logger.debug("Set %s to %s" % (scoped_key, value))
return value
def get(self, key):
"""Get a value from the config store"""
scoped_key = self.scoped_key(key)
value = self.redis.get(scoped_key)
# Do some typecasting
if key in self.int_properties:
value = int(value)
if key in self.bool_properties:
value = (value == 'True')
self.logger.debug("Fetched %s, got %s" % (scoped_key, value))
return value
def unset(self, key):
scoped_key = self.scoped_key(key)
self.redis.delete(scoped_key)
self.logger.debug("Unset %s" % scoped_key)
def __getattr__(self, key):
"""Convenience method to access get"""
return self.get(key)
def scoped_key(self, key):
"""Return an appropriate key name scoped to a link"""
return ("openob:%s:%s" % (self.link_name, key))
def set_from_argparse(self, opts):
"""Given an optparse object from bin/openob, configure this link"""
self.set("name", opts.link_name)
if opts.mode == "tx":
self.set("port", opts.port)
self.set("jitter_buffer", opts.jitter_buffer)
self.set("encoding", opts.encoding)
self.set("bitrate", opts.bitrate)
self.set("multicast", opts.multicast)
self.set("input_samplerate", opts.samplerate)
self.set("receiver_host", opts.receiver_host)
self.set("opus_framesize", opts.framesize)
self.set("opus_complexity", opts.complexity)
self.set("opus_fec", opts.fec)
self.set("opus_loss_expectation", opts.loss)
self.set("opus_dtx", opts.dtx)
def commit_changes(self, restart=False):
"""
To be called after calls to set() on a running link to signal
a reconfiguration event for that link. If restart is True, the link
should simply terminate itself so it can be restarted with the new
parameters. If restart is False, the link should set all parameters
it can which do not involve a restart.
"""
raise(NotImplementedError, "Link reconfiguration is not yet implemented")
def __init__(self, node_name, link_config, audio_interface):
"""Sets up a new RTP transmitter"""
self.started = False
self.caps = 'None'
self.pipeline = gst.Pipeline("tx")
self.bus = self.pipeline.get_bus()
self.bus.connect("message", self.on_message)
self.link_config = link_config
self.audio_interface = audio_interface
self.logger_factory = LoggerFactory()
self.logger = self.logger_factory.getLogger('node.%s.link.%s.%s' % (node_name, self.link_config.name, self.audio_interface.mode))
self.logger.info("Creating RTP transmission pipeline")
# Audio input
if self.audio_interface.type == 'auto':
self.source = gst.element_factory_make('autoaudiosrc')
elif self.audio_interface.type == 'alsa':
self.source = gst.element_factory_make('alsasrc')
self.source.set_property('device', self.audio_interface.alsa_device)
elif self.audio_interface.type == 'jack':
self.source = gst.element_factory_make("jackaudiosrc")
if self.audio_interface.jack_auto:
self.source.set_property('connect', 'auto')
else:
self.source.set_property('connect', 'none')
self.source.set_property('buffer-time', 50000)
self.source.set_property('name', self.audio_interface.jack_name)
self.source.set_property('client-name', self.audio_interface.jack_name)
# Audio resampling and conversion
self.audioresample = gst.element_factory_make("audioresample")
self.audioconvert = gst.element_factory_make("audioconvert")
self.audioresample.set_property('quality', 9) # SRC
# Encoding and payloading
if self.link_config.encoding == 'opus':
self.encoder = gst.element_factory_make("opusenc", "encoder")
self.encoder.set_property('bitrate', int(self.link_config.bitrate) * 1000)
self.encoder.set_property('tolerance', 80000000)
self.encoder.set_property('frame-size', self.link_config.opus_framesize)
self.encoder.set_property('complexity', int(self.link_config.opus_complexity))
self.encoder.set_property('inband-fec', self.link_config.opus_fec)
self.encoder.set_property('packet-loss-percentage', int(self.link_config.opus_loss_expectation))
self.encoder.set_property('dtx', self.link_config.opus_dtx)
print(self.encoder.get_properties('bitrate', 'dtx', 'inband-fec'))
self.payloader = gst.element_factory_make("rtpopuspay", "payloader")
elif self.link_config.encoding == 'pcm':
# we have no encoder for PCM operation
self.payloader = gst.element_factory_make("rtpL16pay", "payloader")
else:
self.logger.critical("Unknown encoding type %s" % self.link_config.encoding)
# TODO: Add a tee here, and sort out creating multiple UDP sinks for multipath
# Now the RTP bits
# We'll send audio out on this
self.udpsink_rtpout = gst.element_factory_make("udpsink", "udpsink_rtp")
self.udpsink_rtpout.set_property('host', self.link_config.receiver_host)
self.udpsink_rtpout.set_property('port', self.link_config.port)
self.logger.info('Set receiver to %s:%i' % (self.link_config.receiver_host, self.link_config.port))
if self.link_config.multicast:
self.udpsink_rtpout.set_property('auto_multicast', True)
self.logger.info('Multicast mode enabled')
# Our RTP manager
self.rtpbin = gst.element_factory_make("gstrtpbin", "gstrtpbin")
self.rtpbin.set_property('latency', 0)
# Our level monitor
self.level = gst.element_factory_make("level")
self.level.set_property('message', True)
self.level.set_property('interval', 1000000000)
# Add a capsfilter to allow specification of input sample rate
self.capsfilter = gst.element_factory_make("capsfilter")
# Add to the pipeline
self.pipeline.add(
self.source, self.capsfilter, self.audioresample, self.audioconvert,
self.payloader, self.udpsink_rtpout, self.rtpbin,
self.level)
if self.link_config.encoding != 'pcm':
# Only add the encoder if we're not in PCM mode
self.pipeline.add(self.encoder)
# Decide which format to apply to the capsfilter (Jack uses float)
if self.audio_interface.type == 'jack':
data_type = 'audio/x-raw-float'
else:
data_type = 'audio/x-raw-int'
# if audio_rate has been specified, then add that to the capsfilter
if self.audio_interface.samplerate != 0:
self.capsfilter.set_property(
"caps", gst.Caps('%s, channels=2, rate=%d' % (data_type, self.audio_interface.samplerate)))
else:
self.capsfilter.set_property(
"caps", gst.Caps('%s, channels=2' % data_type))
# Then continue linking the pipeline together
gst.element_link_many(
self.source, self.capsfilter, self.level, self.audioresample, self.audioconvert)
# Now we get to link this up to our encoder/payloader
if self.link_config.encoding != 'pcm':
gst.element_link_many(
self.audioconvert, self.encoder, self.payloader)
else:
gst.element_link_many(self.audioconvert, self.payloader)
# And now the RTP bits
self.payloader.link_pads('src', self.rtpbin, 'send_rtp_sink_0')
self.rtpbin.link_pads('send_rtp_src_0', self.udpsink_rtpout, 'sink')
# self.udpsrc_rtcpin.link_pads('src', self.rtpbin, 'recv_rtcp_sink_0')
# # RTCP SRs
# self.rtpbin.link_pads('send_rtcp_src_0', self.udpsink_rtcpout, 'sink')
# Connect our bus up
self.bus.add_signal_watch()
self.bus.connect('message', self.on_message)
class RTPTransmitter(object):
def __init__(self, node_name, link_config, audio_interface):
"""Sets up a new RTP transmitter"""
self.link_config = link_config
self.audio_interface = audio_interface
self.logger_factory = LoggerFactory()
self.logger = self.logger_factory.getLogger('node.%s.link.%s.%s' % (node_name, self.link_config.name, self.audio_interface.mode))
self.logger.info('Creating transmission pipeline')
self.build_pipeline()
def run(self):
self.pipeline.set_state(Gst.State.PLAYING)
# Gst.debug_bin_to_dot_file(self.pipeline, Gst.DebugGraphDetails.ALL, 'tx-graph')
while self.caps == None:
caps = self.transport.get_by_name('udpsink').get_static_pad('sink').get_property('caps')
if caps == None:
self.logger.warn('Waiting for audio interface/caps')
time.sleep(0.1)
else:
self.caps = caps.to_string()
def loop(self):
try:
loop = GLib.MainLoop()
loop.run()
except Exception as e:
self.logger.exception('Encountered a problem in the MainLoop, tearing down the pipeline: %s' % e)
self.pipeline.set_state(Gst.State.NULL)
def build_pipeline(self):
self.pipeline = Gst.Pipeline.new('tx')
self.started = False
self.caps = None
bus = self.pipeline.get_bus()
self.source = self.build_audio_interface()
self.encoder = self.build_encoder()
self.transport = self.build_transport()
self.pipeline.add(self.source)
self.pipeline.add(self.encoder)
self.pipeline.add(self.transport)
self.source.link(self.encoder)
self.encoder.link(self.transport)
# Connect our bus up
bus.add_signal_watch()
bus.connect('message', self.on_message)
def build_audio_interface(self):
self.logger.debug('Building audio input bin')
bin = Gst.Bin.new('audio')
# Audio input
if self.audio_interface.type == 'auto':
source = Gst.ElementFactory.make('autoaudiosrc')
elif self.audio_interface.type == 'alsa':
source = Gst.ElementFactory.make('alsasrc')
source.set_property('device', self.audio_interface.alsa_device)
elif self.audio_interface.type == 'jack':
source = Gst.ElementFactory.make('jackaudiosrc')
if self.audio_interface.jack_auto:
source.set_property('connect', 'auto')
else:
source.set_property('connect', 'none')
source.set_property('buffer-time', 50000)
source.set_property('name', self.audio_interface.jack_name)
source.set_property('client-name', self.audio_interface.jack_name)
if self.audio_interface.jack_port_pattern:
source.set_property('port-pattern', self.audio_interface.jack_port_pattern)
elif self.audio_interface.type == 'test':
source = Gst.ElementFactory.make('audiotestsrc')
bin.add(source)
# Our level monitor
level = Gst.ElementFactory.make('level')
level.set_property('message', True)
level.set_property('interval', 1000000000)
bin.add(level)
# Audio resampling and conversion
resample = Gst.ElementFactory.make('audioresample')
resample.set_property('quality', 9) # SRC
bin.add(resample)
convert = Gst.ElementFactory.make('audioconvert')
bin.add(convert)
# Add a capsfilter to allow specification of input sample rate
capsfilter = Gst.ElementFactory.make('capsfilter')
caps = Gst.Caps.new_empty_simple('audio/x-raw')
# if audio_rate has been specified, then add that to the capsfilter
if self.audio_interface.samplerate != 0:
caps.set_value('rate', self.audio_interface.samplerate)
self.logger.debug(caps.to_string())
capsfilter.set_property('caps', caps)
bin.add(capsfilter)
source.link(level)
level.link(resample)
resample.link(convert)
convert.link(capsfilter)
bin.add_pad(Gst.GhostPad.new('src', capsfilter.get_static_pad('src')))
return bin
def build_encoder(self):
self.logger.debug('Building encoder bin')
bin = Gst.Bin.new('encoder')
# Encoding and payloading
if self.link_config.encoding == 'opus':
encoder = Gst.ElementFactory.make('opusenc', 'encoder')
encoder.set_property('bitrate', int(self.link_config.bitrate) * 1000)
encoder.set_property('tolerance', 80000000)
encoder.set_property('frame-size', self.link_config.opus_framesize)
encoder.set_property('complexity', int(self.link_config.opus_complexity))
encoder.set_property('inband-fec', self.link_config.opus_fec)
encoder.set_property('packet-loss-percentage', int(self.link_config.opus_loss_expectation))
encoder.set_property('dtx', self.link_config.opus_dtx)
payloader = Gst.ElementFactory.make('rtpopuspay', 'payloader')
elif self.link_config.encoding == 'pcm':
# we have no encoder for PCM operation
payloader = Gst.ElementFactory.make('rtpL16pay', 'payloader')
else:
self.logger.critical('Unknown encoding type %s' % self.link_config.encoding)
bin.add(payloader)
if 'encoder' in locals():
bin.add(encoder)
encoder.link(payloader)
bin.add_pad(Gst.GhostPad.new('sink', encoder.get_static_pad('sink')))
else:
bin.add_pad(Gst.GhostPad.new('sink', payloader.get_static_pad('sink')))
bin.add_pad(Gst.GhostPad.new('src', payloader.get_static_pad('src')))
return bin
def build_transport(self):
self.logger.debug('Building RTP transport bin')
bin = Gst.Bin.new('transport')
# Our RTP manager
rtpbin = Gst.ElementFactory.make('rtpbin', 'rtpbin')
rtpbin.set_property('latency', 0)
bin.add(rtpbin)
# TODO: Add a tee here, and sort out creating multiple UDP sinks for multipath
udpsink = Gst.ElementFactory.make('udpsink', 'udpsink')
udpsink.set_property('host', self.link_config.receiver_host)
udpsink.set_property('port', self.link_config.port)
self.logger.info('Set receiver to %s:%i' % (self.link_config.receiver_host, self.link_config.port))
if self.link_config.multicast:
udpsink.set_property('auto_multicast', True)
self.logger.info('Multicast mode enabled')
bin.add(udpsink)
bin.add_pad(Gst.GhostPad.new('sink', rtpbin.get_request_pad('send_rtp_sink_0')))
rtpbin.link_pads('send_rtp_src_0', udpsink, 'sink')
return bin
def on_message(self, bus, message):
if message.type == Gst.MessageType.ELEMENT:
struct = message.get_structure()
if struct != None:
if struct.get_name() == 'level':
if self.started is False:
self.started = True
if len(struct.get_value('peak')) == 1:
self.logger.info('Started mono audio transmission')
else:
self.logger.info('Started stereo audio transmission')
else:
if len(struct.get_value('peak')) == 1:
self.logger.debug('Level: %.2f', struct.get_value('peak')[0])
else:
self.logger.debug('Levels: L %.2f R %.2f' % (struct.get_value('peak')[0], struct.get_value('peak')[1]))
return True
def get_caps(self):
return self.caps
def __init__(self, node_name):
"""Set up a new node."""
self.node_name = node_name
self.logger_factory = LoggerFactory()
self.logger = self.logger_factory.getLogger('node.%s' % self.node_name)
def __init__(self, node_name):
"""Set up a new node."""
self.node_name = node_name
self.logger_factory = LoggerFactory()
self.logger = self.logger_factory.getLogger('node.%s' % self.node_name)
class RTPReceiver(object):
def __init__(self, node_name, link_config, audio_interface):
"""Sets up a new RTP receiver"""
self.started = False
self.pipeline = gst.Pipeline("rx")
self.bus = self.pipeline.get_bus()
self.bus.connect("message", self.on_message)
self.link_config = link_config
self.audio_interface = audio_interface
self.logger_factory = LoggerFactory()
self.logger = self.logger_factory.getLogger(
'node.%s.link.%s.%s' %
(node_name, self.link_config.name, self.audio_interface.mode))
caps = self.link_config.get("caps")
# Audio output
if self.audio_interface.type == 'auto':
self.sink = gst.element_factory_make("autoaudiosink")
elif self.audio_interface.type == 'alsa':
self.sink = gst.element_factory_make("alsasink")
self.sink.set_property('device', self.audio_interface.alsa_device)
elif self.audio_interface.type == 'jack':
self.sink = gst.element_factory_make("jackaudiosink")
if self.audio_interface.jack_auto:
self.sink.set_property('connect', 'auto')
else:
self.sink.set_property('connect', 'none')
self.sink.set_property('name', self.audio_interface.jack_name)
# Audio conversion and resampling
self.audioconvert = gst.element_factory_make("audioconvert")
self.audioresample = gst.element_factory_make("audioresample")
self.audioresample.set_property('quality', 9)
self.audiorate = gst.element_factory_make("audiorate")
# Decoding and depayloading
if self.link_config.encoding == 'opus':
self.decoder = gst.element_factory_make("opusdec", "decoder")
self.decoder.set_property('use-inband-fec', True) # FEC
self.decoder.set_property('plc', True) # Packet loss concealment
self.depayloader = gst.element_factory_make(
"rtpopusdepay", "depayloader")
elif self.link_config.encoding == 'pcm':
self.depayloader = gst.element_factory_make(
"rtpL16depay", "depayloader")
# RTP stuff
self.rtpbin = gst.element_factory_make('gstrtpbin')
self.rtpbin.set_property('latency', self.link_config.jitter_buffer)
self.rtpbin.set_property('autoremove', True)
self.rtpbin.set_property('do-lost', True)
# Where audio comes in
self.udpsrc_rtpin = gst.element_factory_make('udpsrc')
self.udpsrc_rtpin.set_property('port', self.link_config.port)
if self.link_config.multicast:
self.udpsrc_rtpin.set_property('auto_multicast', True)
self.udpsrc_rtpin.set_property('multicast_group',
self.link_config.receiver_host)
caps = caps.replace('\\', '')
# Fix for gstreamer bug in rtpopuspay fixed in GST-plugins-bad
# 50140388d2b62d32dd9d0c071e3051ebc5b4083b, bug 686547
if self.link_config.encoding == 'opus':
caps = re.sub(r'(caps=.+ )', '', caps)
udpsrc_caps = gst.caps_from_string(caps)
self.udpsrc_rtpin.set_property('caps', udpsrc_caps)
self.udpsrc_rtpin.set_property('timeout', 3000000)
# Our level monitor, also used for continuous audio
self.level = gst.element_factory_make("level")
self.level.set_property('message', True)
self.level.set_property('interval', 1000000000)
# And now we've got it all set up we need to add the elements
self.pipeline.add(self.audiorate, self.audioresample,
self.audioconvert, self.sink, self.level,
self.depayloader, self.rtpbin, self.udpsrc_rtpin)
if self.link_config.encoding != 'pcm':
self.pipeline.add(self.decoder)
gst.element_link_many(self.depayloader, self.decoder,
self.audioconvert)
else:
gst.element_link_many(self.depayloader, self.audioconvert)
gst.element_link_many(self.audioconvert, self.audioresample,
self.audiorate, self.level, self.sink)
# Now the RTP pads
self.udpsrc_rtpin.link_pads('src', self.rtpbin, 'recv_rtp_sink_0')
# Attach callbacks for dynamic pads (RTP output) and busses
self.rtpbin.connect('pad-added', self.rtpbin_pad_added)
self.bus.add_signal_watch()
# Our RTPbin won't give us an audio pad till it receives, so we need to
# attach it here
def rtpbin_pad_added(self, obj, pad):
# Unlink first.
self.rtpbin.unlink(self.depayloader)
# Relink
self.rtpbin.link(self.depayloader)
def on_message(self, bus, message):
if message.type == gst.MESSAGE_ELEMENT:
if message.structure.get_name() == 'level':
if self.started is False:
self.started = True
if len(message.structure['peak']) == 1:
self.logger.info("Receiving mono audio transmission")
else:
self.logger.info("Receiving stereo audio transmission")
if message.structure.get_name() == 'GstUDPSrcTimeout':
# Only UDP source configured to emit timeouts is the audio
# input
self.logger.critical("No data received for 3 seconds!")
if self.started:
self.logger.critical("Shutting down receiver for restart")
self.pipeline.set_state(gst.STATE_NULL)
self.loop.quit()
return True
def run(self):
self.pipeline.set_state(gst.STATE_PLAYING)
def loop(self):
self.loop = gobject.MainLoop()
self.loop.run()
class LinkConfig(object):
"""
The LinkConfig class encapsulates link configuration. It's genderless;
a TX node should be able to set up a new link and an RX node should be
able (once the TX node has specified the port caps) to configure itself
to receive the stream using the data and methods in this config.
"""
def __init__(self, link_name, redis_host):
"""
Set up a new LinkConfig instance - needs to know the link name and
configuration host.
"""
self.int_properties = [
'port', 'jitter_buffer', 'opus_framesize', 'opus_complexity',
'bitrate', 'opus_loss_expectation'
]
self.bool_properties = ['opus_dtx', 'opus_fec', 'multicast']
self.link_name = link_name
self.redis_host = redis_host
self.logger_factory = LoggerFactory()
self.logger = self.logger_factory.getLogger('link.%s.config' %
self.link_name)
self.logger.info("Connecting to configuration host %s" %
self.redis_host)
self.redis = None
while True:
try:
self.redis = redis.StrictRedis(host=self.redis_host,
charset="utf-8",
decode_responses=True)
break
except Exception as e:
self.logger.error(
"Unable to connect to configuration host! Retrying. (%s)" %
e)
time.sleep(0.1)
def blocking_get(self, key):
"""Get a value, blocking until it's not None if needed"""
while True:
value = self.get(key)
if value is not None:
self.logger.debug("Fetched (blocking) %s, got %s" %
(key, value))
return value
time.sleep(0.1)
def set(self, key, value):
"""Set a value in the config store"""
scoped_key = self.scoped_key(key)
if key in self.bool_properties:
value = int(value)
self.redis.set(scoped_key, value)
self.logger.debug("Set %s to %s" % (scoped_key, value))
return value
def get(self, key):
"""Get a value from the config store"""
scoped_key = self.scoped_key(key)
value = self.redis.get(scoped_key)
# Do some typecasting
if key in self.int_properties:
value = int(value)
if key in self.bool_properties:
value = (value == 'True')
self.logger.debug("Fetched %s, got %s" % (scoped_key, value))
return value
def unset(self, key):
scoped_key = self.scoped_key(key)
self.redis.delete(scoped_key)
self.logger.debug("Unset %s" % scoped_key)
def __getattr__(self, key):
"""Convenience method to access get"""
return self.get(key)
def scoped_key(self, key):
"""Return an appropriate key name scoped to a link"""
return ("openob:%s:%s" % (self.link_name, key))
def set_from_argparse(self, opts):
"""Given an optparse object from bin/openob, configure this link"""
self.set("name", opts.link_name)
if opts.mode == "tx":
self.set("port", opts.port)
self.set("jitter_buffer", opts.jitter_buffer)
self.set("encoding", opts.encoding)
self.set("bitrate", opts.bitrate)
self.set("multicast", opts.multicast)
self.set("input_samplerate", opts.samplerate)
self.set("receiver_host", opts.receiver_host)
self.set("opus_framesize", opts.framesize)
self.set("opus_complexity", opts.complexity)
self.set("opus_fec", opts.fec)
self.set("opus_loss_expectation", opts.loss)
self.set("opus_dtx", opts.dtx)
def commit_changes(self, restart=False):
"""
To be called after calls to set() on a running link to signal
a reconfiguration event for that link. If restart is True, the link
should simply terminate itself so it can be restarted with the new
parameters. If restart is False, the link should set all parameters
it can which do not involve a restart.
"""
raise (NotImplementedError,
"Link reconfiguration is not yet implemented")
def __init__(self, node_name, link_config, audio_interface):
"""Sets up a new RTP receiver"""
self.started = False
self.pipeline = gst.Pipeline("rx")
self.bus = self.pipeline.get_bus()
self.bus.connect("message", self.on_message)
self.link_config = link_config
self.audio_interface = audio_interface
self.logger_factory = LoggerFactory()
self.logger = self.logger_factory.getLogger(
'node.%s.link.%s.%s' %
(node_name, self.link_config.name, self.audio_interface.mode))
caps = self.link_config.get("caps")
# Audio output
if self.audio_interface.type == 'auto':
self.sink = gst.element_factory_make("autoaudiosink")
elif self.audio_interface.type == 'alsa':
self.sink = gst.element_factory_make("alsasink")
self.sink.set_property('device', self.audio_interface.alsa_device)
elif self.audio_interface.type == 'jack':
self.sink = gst.element_factory_make("jackaudiosink")
if self.audio_interface.jack_auto:
self.sink.set_property('connect', 'auto')
else:
self.sink.set_property('connect', 'none')
self.sink.set_property('name', self.audio_interface.jack_name)
# Audio conversion and resampling
self.audioconvert = gst.element_factory_make("audioconvert")
self.audioresample = gst.element_factory_make("audioresample")
self.audioresample.set_property('quality', 9)
self.audiorate = gst.element_factory_make("audiorate")
# Decoding and depayloading
if self.link_config.encoding == 'opus':
self.decoder = gst.element_factory_make("opusdec", "decoder")
self.decoder.set_property('use-inband-fec', True) # FEC
self.decoder.set_property('plc', True) # Packet loss concealment
self.depayloader = gst.element_factory_make(
"rtpopusdepay", "depayloader")
elif self.link_config.encoding == 'pcm':
self.depayloader = gst.element_factory_make(
"rtpL16depay", "depayloader")
# RTP stuff
self.rtpbin = gst.element_factory_make('gstrtpbin')
self.rtpbin.set_property('latency', self.link_config.jitter_buffer)
self.rtpbin.set_property('autoremove', True)
self.rtpbin.set_property('do-lost', True)
# Where audio comes in
self.udpsrc_rtpin = gst.element_factory_make('udpsrc')
self.udpsrc_rtpin.set_property('port', self.link_config.port)
if self.link_config.multicast:
self.udpsrc_rtpin.set_property('auto_multicast', True)
self.udpsrc_rtpin.set_property('multicast_group',
self.link_config.receiver_host)
caps = caps.replace('\\', '')
# Fix for gstreamer bug in rtpopuspay fixed in GST-plugins-bad
# 50140388d2b62d32dd9d0c071e3051ebc5b4083b, bug 686547
if self.link_config.encoding == 'opus':
caps = re.sub(r'(caps=.+ )', '', caps)
udpsrc_caps = gst.caps_from_string(caps)
self.udpsrc_rtpin.set_property('caps', udpsrc_caps)
self.udpsrc_rtpin.set_property('timeout', 3000000)
# Our level monitor, also used for continuous audio
self.level = gst.element_factory_make("level")
self.level.set_property('message', True)
self.level.set_property('interval', 1000000000)
# And now we've got it all set up we need to add the elements
self.pipeline.add(self.audiorate, self.audioresample,
self.audioconvert, self.sink, self.level,
self.depayloader, self.rtpbin, self.udpsrc_rtpin)
if self.link_config.encoding != 'pcm':
self.pipeline.add(self.decoder)
gst.element_link_many(self.depayloader, self.decoder,
self.audioconvert)
else:
gst.element_link_many(self.depayloader, self.audioconvert)
gst.element_link_many(self.audioconvert, self.audioresample,
self.audiorate, self.level, self.sink)
# Now the RTP pads
self.udpsrc_rtpin.link_pads('src', self.rtpbin, 'recv_rtp_sink_0')
# Attach callbacks for dynamic pads (RTP output) and busses
self.rtpbin.connect('pad-added', self.rtpbin_pad_added)
self.bus.add_signal_watch()
def __init__(self, node_name, link_config, audio_interface):
"""Sets up a new RTP receiver"""
self.started = False
self.pipeline = gst.Pipeline("rx")
self.bus = self.pipeline.get_bus()
self.bus.connect("message", self.on_message)
self.link_config = link_config
self.audio_interface = audio_interface
self.logger_factory = LoggerFactory()
self.logger = self.logger_factory.getLogger('node.%s.link.%s.%s' % (node_name, self.link_config.name, self.audio_interface.mode))
caps = self.link_config.get("caps")
# Audio output
if self.audio_interface.type == 'auto':
self.sink = gst.element_factory_make("autoaudiosink")
elif self.audio_interface.type == 'alsa':
self.sink = gst.element_factory_make("alsasink")
self.sink.set_property('device', self.audio_interface.alsa_device)
elif self.audio_interface.type == 'jack':
self.sink = gst.element_factory_make("jackaudiosink")
if self.audio_interface.jack_auto:
self.sink.set_property('connect', 'auto')
else:
self.sink.set_property('connect', 'none')
self.sink.set_property('name', self.audio_interface.jack_name)
# Audio conversion and resampling
self.audioconvert = gst.element_factory_make("audioconvert")
self.audioresample = gst.element_factory_make("audioresample")
self.audioresample.set_property('quality', 9)
self.audiorate = gst.element_factory_make("audiorate")
# Decoding and depayloading
if self.link_config.encoding == 'opus':
self.decoder = gst.element_factory_make("opusdec", "decoder")
self.decoder.set_property('use-inband-fec', True) # FEC
self.decoder.set_property('plc', True) # Packet loss concealment
self.depayloader = gst.element_factory_make(
"rtpopusdepay", "depayloader")
elif self.link_config.encoding == 'pcm':
self.depayloader = gst.element_factory_make(
"rtpL16depay", "depayloader")
# RTP stuff
self.rtpbin = gst.element_factory_make('gstrtpbin')
self.rtpbin.set_property('latency', self.link_config.jitter_buffer)
self.rtpbin.set_property('autoremove', True)
self.rtpbin.set_property('do-lost', True)
# Where audio comes in
self.udpsrc_rtpin = gst.element_factory_make('udpsrc')
self.udpsrc_rtpin.set_property('port', self.link_config.port)
if self.link_config.multicast:
self.udpsrc_rtpin.set_property('auto_multicast', True)
self.udpsrc_rtpin.set_property('multicast_group', self.link_config.receiver_host)
caps = caps.replace('\\', '')
# Fix for gstreamer bug in rtpopuspay fixed in GST-plugins-bad
# 50140388d2b62d32dd9d0c071e3051ebc5b4083b, bug 686547
if self.link_config.encoding == 'opus':
caps = re.sub(r'(caps=.+ )', '', caps)
udpsrc_caps = gst.caps_from_string(caps)
self.udpsrc_rtpin.set_property('caps', udpsrc_caps)
self.udpsrc_rtpin.set_property('timeout', 3000000)
# Our level monitor, also used for continuous audio
self.level = gst.element_factory_make("level")
self.level.set_property('message', True)
self.level.set_property('interval', 1000000000)
# And now we've got it all set up we need to add the elements
self.pipeline.add(
self.audiorate, self.audioresample, self.audioconvert, self.sink,
self.level, self.depayloader, self.rtpbin, self.udpsrc_rtpin)
if self.link_config.encoding != 'pcm':
self.pipeline.add(self.decoder)
gst.element_link_many(
self.depayloader, self.decoder, self.audioconvert)
else:
gst.element_link_many(self.depayloader, self.audioconvert)
gst.element_link_many(
self.audioconvert, self.audioresample, self.audiorate, self.level,
self.sink)
# Now the RTP pads
self.udpsrc_rtpin.link_pads('src', self.rtpbin, 'recv_rtp_sink_0')
# Attach callbacks for dynamic pads (RTP output) and busses
self.rtpbin.connect('pad-added', self.rtpbin_pad_added)
self.bus.add_signal_watch()
def __init__(self, node_name, link_config, audio_interface):
"""Sets up a new RTP transmitter"""
self.started = False
self.caps = 'None'
self.pipeline = gst.Pipeline("tx")
self.bus = self.pipeline.get_bus()
self.bus.connect("message", self.on_message)
self.link_config = link_config
self.audio_interface = audio_interface
self.logger_factory = LoggerFactory()
self.logger = self.logger_factory.getLogger(
'node.%s.link.%s.%s' %
(node_name, self.link_config.name, self.audio_interface.mode))
self.logger.info("Starting up RTP transmitter")
# Audio input
if self.audio_interface.type == 'auto':
self.source = gst.element_factory_make('autoaudiosrc')
elif self.audio_interface.type == 'alsa':
self.source = gst.element_factory_make('alsasrc')
self.source.set_property('device',
self.audio_interface.alsa_device)
elif self.audio_interface.type == 'jack':
self.source = gst.element_factory_make("jackaudiosrc")
if self.audio_interface.jack_auto:
self.source.set_property('connect', 'auto')
else:
self.source.set_property('connect', 'none')
self.source.set_property('name', self.audio_interface.jack_name)
# Audio conversion and resampling
self.audioconvert = gst.element_factory_make("audioconvert")
self.audioresample = gst.element_factory_make("audioresample")
self.audioresample.set_property('quality', 9) # SRC
self.audiorate = gst.element_factory_make("audiorate")
# Encoding and payloading
if self.link_config.encoding == 'opus':
self.encoder = gst.element_factory_make("opusenc", "encoder")
self.encoder.set_property('bitrate',
int(self.link_config.bitrate) * 1000)
self.encoder.set_property('frame-size',
self.link_config.opus_framesize)
self.encoder.set_property('complexity',
int(self.link_config.opus_complexity))
self.encoder.set_property('inband-fec', self.link_config.opus_fec)
self.encoder.set_property(
'packet-loss-percentage',
int(self.link_config.opus_loss_expectation))
self.encoder.set_property('dtx', self.link_config.opus_dtx)
self.payloader = gst.element_factory_make("rtpopuspay",
"payloader")
elif self.link_config.encoding == 'pcm':
# we have no encoder for PCM operation
self.payloader = gst.element_factory_make("rtpL16pay", "payloader")
else:
self.logger.critical("Unknown encoding type %s" %
self.link_config.encoding)
# TODO: Add a tee here, and sort out creating multiple UDP sinks for multipath
# Now the RTP bits
# We'll send audio out on this
self.udpsink_rtpout = gst.element_factory_make("udpsink",
"udpsink_rtp")
self.udpsink_rtpout.set_property('host',
self.link_config.receiver_host)
self.udpsink_rtpout.set_property('port', self.link_config.port)
if self.link_config.multicast:
self.udpsink_rtpout.set_property('auto_multicast', True)
# Our RTP manager
self.rtpbin = gst.element_factory_make("gstrtpbin", "gstrtpbin")
# Our level monitor
self.level = gst.element_factory_make("level")
self.level.set_property('message', True)
self.level.set_property('interval', 1000000000)
# Add a capsfilter to allow specification of input sample rate
self.capsfilter = gst.element_factory_make("capsfilter")
# Add to the pipeline
self.pipeline.add(self.source, self.capsfilter, self.audioconvert,
self.audioresample, self.audiorate, self.payloader,
self.udpsink_rtpout, self.rtpbin, self.level)
if self.link_config.encoding != 'pcm':
# Only add the encoder if we're not in PCM mode
self.pipeline.add(self.encoder)
# Decide which format to apply to the capsfilter (Jack uses float)
if self.audio_interface.type == 'jack':
type = 'audio/x-raw-float'
else:
type = 'audio/x-raw-int'
# if audio_rate has been specified, then add that to the capsfilter
if self.audio_interface.samplerate != 0:
self.capsfilter.set_property(
"caps",
gst.Caps('%s, channels=2, rate=%d' %
(type, self.audio_interface.samplerate)))
else:
self.capsfilter.set_property("caps",
gst.Caps('%s, channels=2' % type))
# Then continue linking the pipeline together
gst.element_link_many(self.source, self.capsfilter, self.level,
self.audioresample, self.audiorate,
self.audioconvert)
# Now we get to link this up to our encoder/payloader
if self.link_config.encoding != 'pcm':
gst.element_link_many(self.audioconvert, self.encoder,
self.payloader)
else:
gst.element_link_many(self.audioconvert, self.payloader)
# And now the RTP bits
self.payloader.link_pads('src', self.rtpbin, 'send_rtp_sink_0')
self.rtpbin.link_pads('send_rtp_src_0', self.udpsink_rtpout, 'sink')
# self.udpsrc_rtcpin.link_pads('src', self.rtpbin, 'recv_rtcp_sink_0')
# # RTCP SRs
# self.rtpbin.link_pads('send_rtcp_src_0', self.udpsink_rtcpout, 'sink')
# Connect our bus up
self.bus.add_signal_watch()
self.bus.connect('message', self.on_message)
argv = sys.argv
sys.argv = []
from openob.logger import LoggerFactory
from openob.node import Node
from openob.link_config import LinkConfig
from openob.audio_interface import AudioInterface
sys.argv = argv
Config = ConfigParser.ConfigParser()
Config.read("/home/pi/openob-gui/outstreamer.ini")
if len(sys.argv) > 1 and sys.argv[1] == 'autostart':
if Config.get("outstreamer", "Boot") != '1':
print "Autostart off"
sys.exit()
Log_Level = logging.INFO #INFO / DEBUG
logger_factory = LoggerFactory(level=Log_Level)
link_config = LinkConfig("transmission", Config.get("outstreamer",
"Encoder_IP"))
link_config.set("jitter_buffer", Config.get("outstreamer", "Jitter_Buffer"))
audio_interface = AudioInterface("recepteur")
link_config.set("port", Config.get("outstreamer", "Listen_Port"))
audio_interface.set("mode", "rx")
audio_interface.set("type", "alsa")
audio_interface.set("alsa_device",
"hw:" + Config.get("outstreamer", "Soundcard_ID"))
node = Node("recepteur")
node.run_link(link_config, audio_interface)
class Node(object):
"""
OpenOB node instance.
Nodes run links. Each Node looks after its end of a link, ensuring
that it remains running and tries to recover from failures, as well as
responding to configuration changes.
Nodes have a name; everything else is link specific.
For instance, a node might be the 'studio' node, which would run a
'tx' end for the 'stl' link.
Nodes have a config host which is where they store their inter-Node
data and communicate with other Nodes.
"""
def __init__(self, node_name):
"""Set up a new node."""
self.node_name = node_name
self.logger_factory = LoggerFactory()
self.logger = self.logger_factory.getLogger('node.%s' % self.node_name)
def run_link(self, link_config, audio_interface):
"""
Run a new TX or RX node.
"""
# We're now entering the realm where we should desperately try and
# maintain a link under all circumstances forever.
self.logger.info("Link %s initial setup start on %s" %
(link_config.name, self.node_name))
link_logger = self.logger_factory.getLogger(
'node.%s.link.%s' % (self.node_name, link_config.name))
while True:
try:
if audio_interface.mode == 'tx':
try:
link_logger.info("Starting up transmitter")
transmitter = RTPTransmitter(self.node_name,
link_config,
audio_interface)
transmitter.run()
caps = transmitter.get_caps()
link_logger.debug(
"Got caps from transmitter, setting config")
link_config.set("caps", caps)
transmitter.loop()
except ElementNotFoundError as e:
link_logger.critical(
"GStreamer element missing: %s - will now exit" %
e)
sys.exit(1)
except Exception as e:
link_logger.exception(
"Transmitter crashed for some reason! Restarting..."
)
time.sleep(0.5)
elif audio_interface.mode == 'rx':
link_logger.info("Waiting for transmitter capabilities...")
caps = link_config.blocking_get("caps")
link_logger.info("Got caps from transmitter")
try:
link_logger.info("Starting up receiver")
receiver = RTPReceiver(self.node_name, link_config,
audio_interface)
receiver.run()
receiver.loop()
except ElementNotFoundError as e:
link_logger.critical(
"GStreamer element missing: %s - will now exit" %
e)
sys.exit(1)
except Exception as e:
link_logger.exception(
"Receiver crashed for some reason! Restarting...")
time.sleep(0.1)
else:
link_logger.critical("Unknown audio interface mode (%s)!" %
audio_interface.mode)
sys.exit(1)
except Exception as e:
link_logger.exception(
"Unknown exception thrown - please report this as a bug! %s"
% e)
raise
class RTPTransmitter(object):
def __init__(self, node_name, link_config, audio_interface):
"""Sets up a new RTP transmitter"""
self.link_config = link_config
self.audio_interface = audio_interface
self.logger_factory = LoggerFactory()
self.logger = self.logger_factory.getLogger(
'node.%s.link.%s.%s' %
(node_name, self.link_config.name, self.audio_interface.mode))
self.logger.info('Creating transmission pipeline')
self.build_pipeline()
def run(self):
self.pipeline.set_state(Gst.State.PLAYING)
# Gst.debug_bin_to_dot_file(self.pipeline, Gst.DebugGraphDetails.ALL, 'tx-graph')
while self.caps == None:
caps = self.transport.get_by_name('udpsink').get_static_pad(
'sink').get_property('caps')
if caps == None:
self.logger.warn('Waiting for audio interface/caps')
time.sleep(0.1)
else:
self.caps = caps.to_string()
def loop(self):
try:
loop = GLib.MainLoop()
loop.run()
except Exception as e:
self.logger.exception(
'Encountered a problem in the MainLoop, tearing down the pipeline: %s'
% e)
self.pipeline.set_state(Gst.State.NULL)
def build_pipeline(self):
self.pipeline = Gst.Pipeline.new('tx')
self.started = False
self.caps = None
bus = self.pipeline.get_bus()
self.source = self.build_audio_interface()
self.encoder = self.build_encoder()
self.transport = self.build_transport()
self.pipeline.add(self.source)
self.pipeline.add(self.encoder)
self.pipeline.add(self.transport)
self.source.link(self.encoder)
self.encoder.link(self.transport)
# Connect our bus up
bus.add_signal_watch()
bus.connect('message', self.on_message)
def build_audio_interface(self):
self.logger.debug('Building audio input bin')
bin = Gst.Bin.new('audio')
# Audio input
if self.audio_interface.type == 'auto':
source = Gst.ElementFactory.make('autoaudiosrc')
elif self.audio_interface.type == 'alsa':
source = Gst.ElementFactory.make('alsasrc')
source.set_property('device', self.audio_interface.alsa_device)
elif self.audio_interface.type == 'jack':
source = Gst.ElementFactory.make('jackaudiosrc')
if self.audio_interface.jack_auto:
source.set_property('connect', 'auto')
else:
source.set_property('connect', 'none')
source.set_property('buffer-time', 50000)
source.set_property('name', self.audio_interface.jack_name)
source.set_property('client-name', self.audio_interface.jack_name)
if self.audio_interface.jack_port_pattern:
source.set_property('port-pattern',
self.audio_interface.jack_port_pattern)
elif self.audio_interface.type == 'test':
source = Gst.ElementFactory.make('audiotestsrc')
bin.add(source)
# Our level monitor
level = Gst.ElementFactory.make('level')
level.set_property('message', True)
level.set_property('interval', 1000000000)
bin.add(level)
# Audio resampling and conversion
resample = Gst.ElementFactory.make('audioresample')
resample.set_property('quality', 9) # SRC
bin.add(resample)
convert = Gst.ElementFactory.make('audioconvert')
bin.add(convert)
# Add a capsfilter to allow specification of input sample rate
capsfilter = Gst.ElementFactory.make('capsfilter')
caps = Gst.Caps.new_empty_simple('audio/x-raw')
# if audio_rate has been specified, then add that to the capsfilter
if self.audio_interface.samplerate != 0:
caps.set_value('rate', self.audio_interface.samplerate)
self.logger.debug(caps.to_string())
capsfilter.set_property('caps', caps)
bin.add(capsfilter)
source.link(level)
level.link(resample)
resample.link(convert)
convert.link(capsfilter)
bin.add_pad(Gst.GhostPad.new('src', capsfilter.get_static_pad('src')))
return bin
def build_encoder(self):
self.logger.debug('Building encoder bin')
bin = Gst.Bin.new('encoder')
# Encoding and payloading
if self.link_config.encoding == 'opus':
encoder = Gst.ElementFactory.make('opusenc', 'encoder')
encoder.set_property('bitrate',
int(self.link_config.bitrate) * 1000)
encoder.set_property('tolerance', 80000000)
encoder.set_property('frame-size', self.link_config.opus_framesize)
encoder.set_property('complexity',
int(self.link_config.opus_complexity))
encoder.set_property('inband-fec', self.link_config.opus_fec)
encoder.set_property('packet-loss-percentage',
int(self.link_config.opus_loss_expectation))
encoder.set_property('dtx', self.link_config.opus_dtx)
payloader = Gst.ElementFactory.make('rtpopuspay', 'payloader')
elif self.link_config.encoding == 'pcm':
# we have no encoder for PCM operation
payloader = Gst.ElementFactory.make('rtpL16pay', 'payloader')
else:
self.logger.critical('Unknown encoding type %s' %
self.link_config.encoding)
bin.add(payloader)
if 'encoder' in locals():
bin.add(encoder)
encoder.link(payloader)
bin.add_pad(
Gst.GhostPad.new('sink', encoder.get_static_pad('sink')))
else:
bin.add_pad(
Gst.GhostPad.new('sink', payloader.get_static_pad('sink')))
bin.add_pad(Gst.GhostPad.new('src', payloader.get_static_pad('src')))
return bin
def build_transport(self):
self.logger.debug('Building RTP transport bin')
bin = Gst.Bin.new('transport')
# Our RTP manager
rtpbin = Gst.ElementFactory.make('rtpbin', 'rtpbin')
rtpbin.set_property('latency', 0)
bin.add(rtpbin)
# TODO: Add a tee here, and sort out creating multiple UDP sinks for multipath
udpsink = Gst.ElementFactory.make('udpsink', 'udpsink')
udpsink.set_property('host', self.link_config.receiver_host)
udpsink.set_property('port', self.link_config.port)
self.logger.info(
'Set receiver to %s:%i' %
(self.link_config.receiver_host, self.link_config.port))
if self.link_config.multicast:
udpsink.set_property('auto_multicast', True)
self.logger.info('Multicast mode enabled')
bin.add(udpsink)
bin.add_pad(
Gst.GhostPad.new('sink',
rtpbin.get_request_pad('send_rtp_sink_0')))
rtpbin.link_pads('send_rtp_src_0', udpsink, 'sink')
return bin
def on_message(self, bus, message):
if message.type == Gst.MessageType.ELEMENT:
struct = message.get_structure()
if struct != None:
if struct.get_name() == 'level':
if self.started is False:
self.started = True
if len(struct.get_value('peak')) == 1:
self.logger.info('Started mono audio transmission')
else:
self.logger.info(
'Started stereo audio transmission')
else:
if len(struct.get_value('peak')) == 1:
self.logger.debug('Level: %.2f',
struct.get_value('peak')[0])
else:
self.logger.debug('Levels: L %.2f R %.2f' %
(struct.get_value('peak')[0],
struct.get_value('peak')[1]))
return True
def get_caps(self):
return self.caps
def __init__(self, node_name, link_config, audio_interface):
"""Sets up a new RTP receiver"""
self.started = False
self.pipeline = gst.Pipeline("rx")
self.bus = self.pipeline.get_bus()
self.bus.connect("message", self.on_message)
self.link_config = link_config
self.audio_interface = audio_interface
self.logger_factory = LoggerFactory()
self.logger = self.logger_factory.getLogger(
"node.%s.link.%s.%s" % (node_name, self.link_config.name, self.audio_interface.mode)
)
self.logger.info("Creating RTP reception pipeline")
caps = self.link_config.get("caps")
# Audio output
if self.audio_interface.type == "auto":
self.sink = gst.element_factory_make("autoaudiosink")
elif self.audio_interface.type == "alsa":
self.sink = gst.element_factory_make("alsasink")
self.sink.set_property("device", self.audio_interface.alsa_device)
elif self.audio_interface.type == "jack":
self.sink = gst.element_factory_make("jackaudiosink")
if self.audio_interface.jack_auto:
self.sink.set_property("connect", "auto")
else:
self.sink.set_property("connect", "none")
self.sink.set_property("name", self.audio_interface.jack_name)
self.sink.set_property("client-name", self.audio_interface.jack_name)
# Audio resampling and conversion
self.audioresample = gst.element_factory_make("audioresample")
self.audioconvert = gst.element_factory_make("audioconvert")
self.audioresample.set_property("quality", 6)
# Decoding and depayloading
if self.link_config.encoding == "opus":
self.decoder = gst.element_factory_make("opusdec", "decoder")
self.decoder.set_property("use-inband-fec", True) # FEC
self.decoder.set_property("plc", True) # Packet loss concealment
self.depayloader = gst.element_factory_make("rtpopusdepay", "depayloader")
elif self.link_config.encoding == "pcm":
self.depayloader = gst.element_factory_make("rtpL16depay", "depayloader")
# RTP stuff
self.rtpbin = gst.element_factory_make("gstrtpbin")
self.rtpbin.set_property("latency", self.link_config.jitter_buffer)
self.rtpbin.set_property("autoremove", True)
self.rtpbin.set_property("do-lost", True)
# self.rtpbin.set_property('buffer-mode', 1)
# Where audio comes in
self.udpsrc_rtpin = gst.element_factory_make("udpsrc")
self.udpsrc_rtpin.set_property("port", self.link_config.port)
if self.link_config.multicast:
self.udpsrc_rtpin.set_property("auto_multicast", True)
self.udpsrc_rtpin.set_property("multicast_group", self.link_config.receiver_host)
self.logger.info("Multicast mode enabled")
caps = caps.replace("\\", "")
# Fix for gstreamer bug in rtpopuspay fixed in GST-plugins-bad
# 50140388d2b62d32dd9d0c071e3051ebc5b4083b, bug 686547
if self.link_config.encoding == "opus":
caps = re.sub(r"(caps=.+ )", "", caps)
udpsrc_caps = gst.caps_from_string(caps)
self.udpsrc_rtpin.set_property("caps", udpsrc_caps)
self.udpsrc_rtpin.set_property("timeout", 3000000)
# Our level monitor, also used for continuous audio
self.level = gst.element_factory_make("level")
self.level.set_property("message", True)
self.level.set_property("interval", 1000000000)
# And now we've got it all set up we need to add the elements
self.pipeline.add(
self.audioconvert,
self.audioresample,
self.sink,
self.level,
self.depayloader,
self.rtpbin,
self.udpsrc_rtpin,
)
if self.link_config.encoding != "pcm":
self.pipeline.add(self.decoder)
gst.element_link_many(self.depayloader, self.decoder, self.audioconvert)
else:
gst.element_link_many(self.depayloader, self.audioconvert)
gst.element_link_many(self.audioconvert, self.audioresample, self.level, self.sink)
self.logger.debug(self.sink)
# Now the RTP pads
self.udpsrc_rtpin.link_pads("src", self.rtpbin, "recv_rtp_sink_0")
# Attach callbacks for dynamic pads (RTP output) and busses
self.rtpbin.connect("pad-added", self.rtpbin_pad_added)
self.bus.add_signal_watch()
class RTPTransmitter(object):
def __init__(self, node_name, link_config, audio_interface):
"""Sets up a new RTP transmitter"""
self.started = False
self.caps = 'None'
self.pipeline = gst.Pipeline("tx")
self.bus = self.pipeline.get_bus()
self.bus.connect("message", self.on_message)
self.link_config = link_config
self.audio_interface = audio_interface
self.logger_factory = LoggerFactory()
self.logger = self.logger_factory.getLogger('node.%s.link.%s.%s' % (node_name, self.link_config.name, self.audio_interface.mode))
self.logger.info("Starting up RTP transmitter")
# Audio input
if self.audio_interface.type == 'auto':
self.source = gst.element_factory_make('autoaudiosrc')
elif self.audio_interface.type == 'alsa':
self.source = gst.element_factory_make('alsasrc')
self.source.set_property('device', self.audio_interface.alsa_device)
elif self.audio_interface.type == 'jack':
self.source = gst.element_factory_make("jackaudiosrc")
if self.audio_interface.jack_auto:
self.source.set_property('connect', 'auto')
else:
self.source.set_property('connect', 'none')
self.source.set_property('name', self.audio_interface.jack_name)
# Audio conversion and resampling
self.audioconvert = gst.element_factory_make("audioconvert")
self.audioresample = gst.element_factory_make("audioresample")
self.audioresample.set_property('quality', 9) # SRC
self.audiorate = gst.element_factory_make("audiorate")
# Encoding and payloading
if self.link_config.encoding == 'opus':
self.encoder = gst.element_factory_make("opusenc", "encoder")
self.encoder.set_property('bitrate', int(self.link_config.bitrate) * 1000)
self.encoder.set_property('frame-size', self.link_config.opus_framesize)
self.encoder.set_property('complexity', int(self.link_config.opus_complexity))
self.encoder.set_property('inband-fec', self.link_config.opus_fec)
self.encoder.set_property('packet-loss-percentage', int(self.link_config.opus_loss_expectation))
self.encoder.set_property('dtx', self.link_config.opus_dtx)
self.payloader = gst.element_factory_make("rtpopuspay", "payloader")
elif self.link_config.encoding == 'pcm':
# we have no encoder for PCM operation
self.payloader = gst.element_factory_make("rtpL16pay", "payloader")
else:
self.logger.critical("Unknown encoding type %s" % self.link_config.encoding)
# TODO: Add a tee here, and sort out creating multiple UDP sinks for multipath
# Now the RTP bits
# We'll send audio out on this
self.udpsink_rtpout = gst.element_factory_make(
"udpsink", "udpsink_rtp")
self.udpsink_rtpout.set_property('host', self.link_config.receiver_host)
self.udpsink_rtpout.set_property('port', self.link_config.port)
if self.link_config.multicast:
self.udpsink_rtpout.set_property('auto_multicast', True)
# Our RTP manager
self.rtpbin = gst.element_factory_make("gstrtpbin", "gstrtpbin")
# Our level monitor
self.level = gst.element_factory_make("level")
self.level.set_property('message', True)
self.level.set_property('interval', 1000000000)
# Add a capsfilter to allow specification of input sample rate
self.capsfilter = gst.element_factory_make("capsfilter")
# Add to the pipeline
self.pipeline.add(
self.source, self.capsfilter, self.audioconvert, self.audioresample,
self.audiorate, self.payloader, self.udpsink_rtpout, self.rtpbin,
self.level)
if self.link_config.encoding != 'pcm':
# Only add the encoder if we're not in PCM mode
self.pipeline.add(self.encoder)
# Decide which format to apply to the capsfilter (Jack uses float)
if self.audio_interface.type == 'jack':
type = 'audio/x-raw-float'
else:
type = 'audio/x-raw-int'
# if audio_rate has been specified, then add that to the capsfilter
if self.audio_interface.samplerate != 0:
self.capsfilter.set_property(
"caps", gst.Caps('%s, channels=2, rate=%d' % (type, self.audio_interface.samplerate)))
else:
self.capsfilter.set_property(
"caps", gst.Caps('%s, channels=2' % type))
# Then continue linking the pipeline together
gst.element_link_many(
self.source, self.capsfilter, self.level, self.audioresample,
self.audiorate, self.audioconvert)
# Now we get to link this up to our encoder/payloader
if self.link_config.encoding != 'pcm':
gst.element_link_many(
self.audioconvert, self.encoder, self.payloader)
else:
gst.element_link_many(self.audioconvert, self.payloader)
# And now the RTP bits
self.payloader.link_pads('src', self.rtpbin, 'send_rtp_sink_0')
self.rtpbin.link_pads('send_rtp_src_0', self.udpsink_rtpout, 'sink')
# self.udpsrc_rtcpin.link_pads('src', self.rtpbin, 'recv_rtcp_sink_0')
# # RTCP SRs
# self.rtpbin.link_pads('send_rtcp_src_0', self.udpsink_rtcpout, 'sink')
# Connect our bus up
self.bus.add_signal_watch()
self.bus.connect('message', self.on_message)
def run(self):
self.pipeline.set_state(gst.STATE_PLAYING)
while self.caps == 'None':
self.logger.warn("Waiting for audio interface/caps")
self.logger.debug(self.udpsink_rtpout.get_state())
self.caps = str(
self.udpsink_rtpout.get_pad('sink').get_property('caps'))
# Fix for gstreamer bug in rtpopuspay fixed in GST-plugins-bad
# 50140388d2b62d32dd9d0c071e3051ebc5b4083b, bug 686547
if self.link_config.encoding == 'opus':
self.caps = re.sub(r'(caps=.+ )', '', self.caps)
time.sleep(0.1)
def loop(self):
try:
self.loop = gobject.MainLoop()
self.loop.run()
except Exception, e:
self.logger.exception("Encountered a problem in the MainLoop, tearing down the pipeline")
self.pipeline.set_state(gst.STATE_NULL)
class RTPReceiver(object):
def __init__(self, node_name, link_config, audio_interface):
"""Sets up a new RTP receiver"""
self.started = False
self.pipeline = gst.Pipeline("rx")
self.bus = self.pipeline.get_bus()
self.bus.connect("message", self.on_message)
self.link_config = link_config
self.audio_interface = audio_interface
self.logger_factory = LoggerFactory()
self.logger = self.logger_factory.getLogger(
"node.%s.link.%s.%s" % (node_name, self.link_config.name, self.audio_interface.mode)
)
self.logger.info("Creating RTP reception pipeline")
caps = self.link_config.get("caps")
# Audio output
if self.audio_interface.type == "auto":
self.sink = gst.element_factory_make("autoaudiosink")
elif self.audio_interface.type == "alsa":
self.sink = gst.element_factory_make("alsasink")
self.sink.set_property("device", self.audio_interface.alsa_device)
elif self.audio_interface.type == "jack":
self.sink = gst.element_factory_make("jackaudiosink")
if self.audio_interface.jack_auto:
self.sink.set_property("connect", "auto")
else:
self.sink.set_property("connect", "none")
self.sink.set_property("name", self.audio_interface.jack_name)
self.sink.set_property("client-name", self.audio_interface.jack_name)
# Audio resampling and conversion
self.audioresample = gst.element_factory_make("audioresample")
self.audioconvert = gst.element_factory_make("audioconvert")
self.audioresample.set_property("quality", 6)
# Decoding and depayloading
if self.link_config.encoding == "opus":
self.decoder = gst.element_factory_make("opusdec", "decoder")
self.decoder.set_property("use-inband-fec", True) # FEC
self.decoder.set_property("plc", True) # Packet loss concealment
self.depayloader = gst.element_factory_make("rtpopusdepay", "depayloader")
elif self.link_config.encoding == "pcm":
self.depayloader = gst.element_factory_make("rtpL16depay", "depayloader")
# RTP stuff
self.rtpbin = gst.element_factory_make("gstrtpbin")
self.rtpbin.set_property("latency", self.link_config.jitter_buffer)
self.rtpbin.set_property("autoremove", True)
self.rtpbin.set_property("do-lost", True)
# self.rtpbin.set_property('buffer-mode', 1)
# Where audio comes in
self.udpsrc_rtpin = gst.element_factory_make("udpsrc")
self.udpsrc_rtpin.set_property("port", self.link_config.port)
if self.link_config.multicast:
self.udpsrc_rtpin.set_property("auto_multicast", True)
self.udpsrc_rtpin.set_property("multicast_group", self.link_config.receiver_host)
self.logger.info("Multicast mode enabled")
caps = caps.replace("\\", "")
# Fix for gstreamer bug in rtpopuspay fixed in GST-plugins-bad
# 50140388d2b62d32dd9d0c071e3051ebc5b4083b, bug 686547
if self.link_config.encoding == "opus":
caps = re.sub(r"(caps=.+ )", "", caps)
udpsrc_caps = gst.caps_from_string(caps)
self.udpsrc_rtpin.set_property("caps", udpsrc_caps)
self.udpsrc_rtpin.set_property("timeout", 3000000)
# Our level monitor, also used for continuous audio
self.level = gst.element_factory_make("level")
self.level.set_property("message", True)
self.level.set_property("interval", 1000000000)
# And now we've got it all set up we need to add the elements
self.pipeline.add(
self.audioconvert,
self.audioresample,
self.sink,
self.level,
self.depayloader,
self.rtpbin,
self.udpsrc_rtpin,
)
if self.link_config.encoding != "pcm":
self.pipeline.add(self.decoder)
gst.element_link_many(self.depayloader, self.decoder, self.audioconvert)
else:
gst.element_link_many(self.depayloader, self.audioconvert)
gst.element_link_many(self.audioconvert, self.audioresample, self.level, self.sink)
self.logger.debug(self.sink)
# Now the RTP pads
self.udpsrc_rtpin.link_pads("src", self.rtpbin, "recv_rtp_sink_0")
# Attach callbacks for dynamic pads (RTP output) and busses
self.rtpbin.connect("pad-added", self.rtpbin_pad_added)
self.bus.add_signal_watch()
# Our RTPbin won't give us an audio pad till it receives, so we need to
# attach it here
def rtpbin_pad_added(self, obj, pad):
# Unlink first.
self.rtpbin.unlink(self.depayloader)
# Relink
self.rtpbin.link(self.depayloader)
def on_message(self, bus, message):
if message.type == gst.MESSAGE_ELEMENT:
if message.structure.get_name() == "level":
if self.started is False:
self.started = True
# gst.DEBUG_BIN_TO_DOT_FILE(self.pipeline, gst.DEBUG_GRAPH_SHOW_ALL, 'rx-graph')
if len(message.structure["peak"]) == 1:
self.logger.info("Receiving mono audio transmission")
else:
self.logger.info("Receiving stereo audio transmission")
if message.structure.get_name() == "GstUDPSrcTimeout":
# Only UDP source configured to emit timeouts is the audio
# input
self.logger.critical("No data received for 3 seconds!")
if self.started:
self.logger.critical("Shutting down receiver for restart")
self.pipeline.set_state(gst.STATE_NULL)
self.loop.quit()
return True
def run(self):
self.pipeline.set_state(gst.STATE_PLAYING)
self.logger.info("Listening for stream on %s:%i" % (self.link_config.receiver_host, self.link_config.port))
def loop(self):
self.loop = gobject.MainLoop()
self.loop.run()
class Node(object):
"""
OpenOB node instance.
Nodes run links. Each Node looks after its end of a link, ensuring
that it remains running and tries to recover from failures, as well as
responding to configuration changes.
Nodes have a name; everything else is link specific.
For instance, a node might be the 'studio' node, which would run a
'tx' end for the 'stl' link.
Nodes have a config host which is where they store their inter-Node
data and communicate with other Nodes.
"""
def __init__(self, node_name):
"""Set up a new node."""
self.node_name = node_name
self.logger_factory = LoggerFactory()
self.logger = self.logger_factory.getLogger('node.%s' % self.node_name)
def run_link(self, link_config, audio_interface):
"""
Run a new TX or RX node.
"""
# We're now entering the realm where we should desperately try and
# maintain a link under all circumstances forever.
self.logger.info("Link %s initial setup start on %s" % (link_config.name, self.node_name))
link_logger = self.logger_factory.getLogger('node.%s.link.%s' % (self.node_name, link_config.name))
while True:
try:
if audio_interface.mode == 'tx':
try:
link_logger.info("Starting up transmitter")
transmitter = RTPTransmitter(self.node_name, link_config, audio_interface)
transmitter.run()
caps = transmitter.get_caps()
link_logger.debug("Got caps from transmitter, setting config")
link_config.set("caps", caps)
transmitter.loop()
except ElementNotFoundError as e:
link_logger.critical("GStreamer element missing: %s - will now exit" % e)
sys.exit(1)
except Exception as e:
link_logger.exception("Transmitter crashed for some reason! Restarting...")
time.sleep(0.5)
elif audio_interface.mode == 'rx':
link_logger.info("Waiting for transmitter capabilities...")
caps = link_config.blocking_get("caps")
link_logger.info("Got caps from transmitter")
try:
link_logger.info("Starting up receiver")
receiver = RTPReceiver(self.node_name, link_config, audio_interface)
receiver.run()
receiver.loop()
except ElementNotFoundError as e:
link_logger.critical("GStreamer element missing: %s - will now exit" % e)
sys.exit(1)
except Exception as e:
link_logger.exception("Receiver crashed for some reason! Restarting...")
time.sleep(0.1)
else:
link_logger.critical("Unknown audio interface mode (%s)!" % audio_interface.mode)
sys.exit(1)
except Exception as e:
link_logger.exception("Unknown exception thrown - please report this as a bug! %s" % e)
raise
sys.argv = []
from openob.logger import LoggerFactory
from openob.node import Node
from openob.link_config import LinkConfig
from openob.audio_interface import AudioInterface
sys.argv = argv
Config = configparser.ConfigParser()
Config.read("/home/pi/openob-gui/instreamer.ini")
if len(sys.argv) > 1 and sys.argv[1] == 'autostart':
if Config.get("instreamer", "Boot") != '1':
print("Autostart off")
sys.exit()
logger_factory = LoggerFactory(level=logging.INFO)
link_config = LinkConfig("transmission", Config.get("instreamer",
"Encoder_IP"))
audio_interface = AudioInterface("emetteur")
link_config.set("port", Config.get("instreamer", "Listen_Port"))
link_config.set("bitrate", int(Config.get("instreamer", "Bitrate")))
link_config.set("encoding", Config.get("instreamer", "Encoding"))
link_config.set("opus_framesize", "20")
link_config.set("opus_complexity", "9")
link_config.set("opus_loss_expectation", "0")
link_config.set("jitter_buffer", "40")
link_config.set("receiver_host", Config.get("instreamer", "Receiver_IP"))
audio_interface.set("mode", "tx")
audio_interface.set("samplerate", int(Config.get("instreamer", "Samplerate")))
audio_interface.set("type", "alsa")
class RTPReceiver(object):
def __init__(self, node_name, link_config, audio_interface):
"""Sets up a new RTP receiver"""
self.started = False
self.pipeline = gst.Pipeline("rx")
self.bus = self.pipeline.get_bus()
self.bus.connect("message", self.on_message)
self.link_config = link_config
self.audio_interface = audio_interface
self.logger_factory = LoggerFactory()
self.logger = self.logger_factory.getLogger('node.%s.link.%s.%s' % (node_name, self.link_config.name, self.audio_interface.mode))
caps = self.link_config.get("caps")
# Audio output
if self.audio_interface.type == 'auto':
self.sink = gst.element_factory_make("autoaudiosink")
elif self.audio_interface.type == 'alsa':
self.sink = gst.element_factory_make("alsasink")
self.sink.set_property('device', self.audio_interface.alsa_device)
elif self.audio_interface.type == 'jack':
self.sink = gst.element_factory_make("jackaudiosink")
if self.audio_interface.jack_auto:
self.sink.set_property('connect', 'auto')
else:
self.sink.set_property('connect', 'none')
self.sink.set_property('name', self.audio_interface.jack_name)
# Audio conversion and resampling
self.audioconvert = gst.element_factory_make("audioconvert")
self.audioresample = gst.element_factory_make("audioresample")
self.audioresample.set_property('quality', 9)
self.audiorate = gst.element_factory_make("audiorate")
# Decoding and depayloading
if self.link_config.encoding == 'opus':
self.decoder = gst.element_factory_make("opusdec", "decoder")
self.decoder.set_property('use-inband-fec', True) # FEC
self.decoder.set_property('plc', True) # Packet loss concealment
self.depayloader = gst.element_factory_make(
"rtpopusdepay", "depayloader")
elif self.link_config.encoding == 'pcm':
self.depayloader = gst.element_factory_make(
"rtpL16depay", "depayloader")
# RTP stuff
self.rtpbin = gst.element_factory_make('gstrtpbin')
self.rtpbin.set_property('latency', self.link_config.jitter_buffer)
self.rtpbin.set_property('autoremove', True)
self.rtpbin.set_property('do-lost', True)
# Where audio comes in
self.udpsrc_rtpin = gst.element_factory_make('udpsrc')
self.udpsrc_rtpin.set_property('port', self.link_config.port)
if self.link_config.multicast:
self.udpsrc_rtpin.set_property('auto_multicast', True)
self.udpsrc_rtpin.set_property('multicast_group', self.link_config.receiver_host)
caps = caps.replace('\\', '')
# Fix for gstreamer bug in rtpopuspay fixed in GST-plugins-bad
# 50140388d2b62d32dd9d0c071e3051ebc5b4083b, bug 686547
if self.link_config.encoding == 'opus':
caps = re.sub(r'(caps=.+ )', '', caps)
udpsrc_caps = gst.caps_from_string(caps)
self.udpsrc_rtpin.set_property('caps', udpsrc_caps)
self.udpsrc_rtpin.set_property('timeout', 3000000)
# Our level monitor, also used for continuous audio
self.level = gst.element_factory_make("level")
self.level.set_property('message', True)
self.level.set_property('interval', 1000000000)
# And now we've got it all set up we need to add the elements
self.pipeline.add(
self.audiorate, self.audioresample, self.audioconvert, self.sink,
self.level, self.depayloader, self.rtpbin, self.udpsrc_rtpin)
if self.link_config.encoding != 'pcm':
self.pipeline.add(self.decoder)
gst.element_link_many(
self.depayloader, self.decoder, self.audioconvert)
else:
gst.element_link_many(self.depayloader, self.audioconvert)
gst.element_link_many(
self.audioconvert, self.audioresample, self.audiorate, self.level,
self.sink)
# Now the RTP pads
self.udpsrc_rtpin.link_pads('src', self.rtpbin, 'recv_rtp_sink_0')
# Attach callbacks for dynamic pads (RTP output) and busses
self.rtpbin.connect('pad-added', self.rtpbin_pad_added)
self.bus.add_signal_watch()
# Our RTPbin won't give us an audio pad till it receives, so we need to
# attach it here
def rtpbin_pad_added(self, obj, pad):
# Unlink first.
self.rtpbin.unlink(self.depayloader)
# Relink
self.rtpbin.link(self.depayloader)
def on_message(self, bus, message):
if message.type == gst.MESSAGE_ELEMENT:
if message.structure.get_name() == 'level':
if self.started is False:
self.started = True
if len(message.structure['peak']) == 1:
self.logger.info("Receiving mono audio transmission")
else:
self.logger.info("Receiving stereo audio transmission")
if message.structure.get_name() == 'GstUDPSrcTimeout':
# Only UDP source configured to emit timeouts is the audio
# input
self.logger.critical("No data received for 3 seconds!")
if self.started:
self.logger.critical("Shutting down receiver for restart")
self.pipeline.set_state(gst.STATE_NULL)
self.loop.quit()
return True
def run(self):
self.pipeline.set_state(gst.STATE_PLAYING)
def loop(self):
self.loop = gobject.MainLoop()
self.loop.run()
class RTPReceiver(object):
def __init__(self, node_name, link_config, audio_interface):
"""Sets up a new RTP receiver"""
self.link_config = link_config
self.audio_interface = audio_interface
self.logger_factory = LoggerFactory()
self.logger = self.logger_factory.getLogger('node.%s.link.%s.%s' % (node_name, self.link_config.name, self.audio_interface.mode))
self.logger.info('Creating reception pipeline')
self.build_pipeline()
def run(self):
self.pipeline.set_state(Gst.State.PLAYING)
self.logger.info('Listening for stream on %s:%i' % (self.link_config.receiver_host, self.link_config.port))
def loop(self):
try:
self.main_loop = GLib.MainLoop()
self.main_loop.run()
except Exception as e:
self.logger.exception('Encountered a problem in the MainLoop, tearing down the pipeline: %s' % e)
self.pipeline.set_state(Gst.State.NULL)
def build_pipeline(self):
self.pipeline = Gst.Pipeline.new('rx')
self.started = False
bus = self.pipeline.get_bus()
self.transport = self.build_transport()
self.decoder = self.build_decoder()
self.output = self.build_audio_interface()
self.pipeline.add(self.transport)
self.pipeline.add(self.decoder)
self.pipeline.add(self.output)
self.transport.link(self.decoder)
self.decoder.link(self.output)
bus.add_signal_watch()
bus.connect('message', self.on_message)
def build_audio_interface(self):
self.logger.debug('Building audio output bin')
bin = Gst.Bin.new('audio')
# Audio output
if self.audio_interface.type == 'auto':
sink = Gst.ElementFactory.make('autoaudiosink')
elif self.audio_interface.type == 'alsa':
sink = Gst.ElementFactory.make('alsasink')
sink.set_property('device', self.audio_interface.alsa_device)
elif self.audio_interface.type == 'jack':
sink = Gst.ElementFactory.make('jackaudiosink')
if self.audio_interface.jack_auto:
sink.set_property('connect', 'auto')
else:
sink.set_property('connect', 'none')
sink.set_property('name', self.audio_interface.jack_name)
sink.set_property('client-name', self.audio_interface.jack_name)
if self.audio_interface.jack_port_pattern:
sink.set_property('port-pattern', self.audio_interface.jack_port_pattern)
elif self.audio_interface.type == 'test':
sink = Gst.ElementFactory.make('fakesink')
bin.add(sink)
# Audio resampling and conversion
resample = Gst.ElementFactory.make('audioresample')
resample.set_property('quality', 9)
bin.add(resample)
convert = Gst.ElementFactory.make('audioconvert')
bin.add(convert)
# Our level monitor, also used for continuous audio
level = Gst.ElementFactory.make('level')
level.set_property('message', True)
level.set_property('interval', 1000000000)
bin.add(level)
resample.link(convert)
convert.link(level)
level.link(sink)
bin.add_pad(Gst.GhostPad.new('sink', resample.get_static_pad('sink')))
return bin
def build_decoder(self):
self.logger.debug('Building decoder bin')
bin = Gst.Bin.new('decoder')
# Decoding and depayloading
if self.link_config.encoding == 'opus':
decoder = Gst.ElementFactory.make('opusdec', 'decoder')
decoder.set_property('use-inband-fec', True) # FEC
decoder.set_property('plc', True) # Packet loss concealment
depayloader = Gst.ElementFactory.make(
'rtpopusdepay', 'depayloader')
elif self.link_config.encoding == 'pcm':
depayloader = Gst.ElementFactory.make(
'rtpL16depay', 'depayloader')
else:
self.logger.critical('Unknown encoding type %s' % self.link_config.encoding)
bin.add(depayloader)
bin.add_pad(Gst.GhostPad.new('sink', depayloader.get_static_pad('sink')))
if 'decoder' in locals():
bin.add(decoder)
depayloader.link(decoder)
bin.add_pad(Gst.GhostPad.new('src', decoder.get_static_pad('src')))
else:
bin.add_pad(Gst.GhostPad.new('src', depayloader.get_static_pad('src')))
return bin
def build_transport(self):
self.logger.debug('Building RTP transport bin')
bin = Gst.Bin.new('transport')
caps = self.link_config.get('caps').replace('\\', '')
udpsrc_caps = Gst.Caps.from_string(caps)
# Where audio comes in
udpsrc = Gst.ElementFactory.make('udpsrc', 'udpsrc')
udpsrc.set_property('port', self.link_config.port)
udpsrc.set_property('caps', udpsrc_caps)
udpsrc.set_property('timeout', 3000000000)
if self.link_config.multicast:
udpsrc.set_property('auto_multicast', True)
udpsrc.set_property('multicast_group', self.link_config.receiver_host)
self.logger.info('Multicast mode enabled')
bin.add(udpsrc)
rtpbin = Gst.ElementFactory.make('rtpbin', 'rtpbin')
rtpbin.set_property('latency', self.link_config.jitter_buffer)
rtpbin.set_property('autoremove', True)
rtpbin.set_property('do-lost', True)
bin.add(rtpbin)
udpsrc.link_pads('src', rtpbin, 'recv_rtp_sink_0')
valve = Gst.ElementFactory.make('valve', 'valve')
bin.add(valve)
bin.add_pad(Gst.GhostPad.new('src', valve.get_static_pad('src')))
# Attach callbacks for dynamic pads (RTP output) and busses
rtpbin.connect('pad-added', self.rtpbin_pad_added)
return bin
# Our RTPbin won't give us an audio pad till it receives, so we need to
# attach it here
def rtpbin_pad_added(self, obj, pad):
valve = self.transport.get_by_name('valve')
rtpbin = self.transport.get_by_name('rtpbin')
# Unlink first.
rtpbin.unlink(valve)
# Relink
rtpbin.link(valve)
def on_message(self, bus, message):
if message.type == Gst.MessageType.ELEMENT:
struct = message.get_structure()
if struct != None:
if struct.get_name() == 'level':
if self.started is False:
self.started = True
if len(struct.get_value('peak')) == 1:
self.logger.info('Receiving mono audio transmission')
else:
self.logger.info('Receiving stereo audio transmission')
else:
if len(struct.get_value('peak')) == 1:
self.logger.debug('Level: %.2f', struct.get_value('peak')[0])
else:
self.logger.debug('Levels: L %.2f R %.2f' % (struct.get_value('peak')[0], struct.get_value('peak')[1]))
if struct.get_name() == 'GstUDPSrcTimeout':
# Gst.debug_bin_to_dot_file(self.pipeline, Gst.DebugGraphDetails.ALL, 'rx-graph')
# Only UDP source configured to emit timeouts is the audio input
self.logger.critical('No data received for 3 seconds!')
if self.started:
self.logger.critical('Shutting down receiver for restart')
self.pipeline.set_state(Gst.State.NULL)
self.main_loop.quit()
return True