add camera and filedump
This commit is contained in:
parent
82cfa64527
commit
c002fc08eb
414
stream.py
414
stream.py
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@ -1,76 +1,360 @@
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#! /bin/python
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# this entire module is lovingly based on the gst-launch tool kindly provided
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# by the gstreamer wizards themselves...with unecessary crap cut out
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# we make the following assumptions here and optimize as such
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# - all streams are "live"
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# - will not need EOS (no mp4s)
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# - will not require SIGINT (this entire program won't understand them anyways)
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# - no tags or TOCs
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from auxilary import async
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from threading import Thread
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import gi, time
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gi.require_version('Gst', '1.0')
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gi.require_version('GObject', '2.0')
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from gi.repository import Gst
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from gi.repository import Gst, GObject
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Gst.init(None)
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pipe = Gst.Pipeline.new("streamer")
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bus = pipe.get_bus()
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vidSrc = Gst.ElementFactory.make("v4l2src", "vidSrc")
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vidConv = Gst.ElementFactory.make("videoconvert", "vidConv")
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vidScale = Gst.ElementFactory.make("videoscale", "vidScale")
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vidClock = Gst.ElementFactory.make("clockoverlay", "vidClock")
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vidEncode = Gst.ElementFactory.make("omxh264enc", "vidEncode")
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vidParse = Gst.ElementFactory.make("h264parse", "vidParse")
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mux = Gst.ElementFactory.make("mp4mux", "mux")
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#~ sink = Gst.ElementFactory.make("tcpserversink", "sink")
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sink = Gst.ElementFactory.make("filesink", "sink")
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vidSrc.set_property('device', '/dev/video0')
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#~ sink.set_property('host', '0.0.0.0')
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#~ sink.set_property('port', 8080)
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sink.set_property('location', '/home/alarm/testicle.mp4')
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vidRawCaps = Gst.Caps.from_string('video/x-raw,width=320,height=240,framerate=30/1')
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parseCaps = Gst.Caps.from_string('video/x-h264,stream-format=avc')
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pipe.add(vidSrc, vidConv, vidScale, vidClock, vidEncode, vidParse, mux, sink)
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print(vidSrc.link(vidConv))
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print(vidConv.link(vidScale))
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print(vidScale.link_filtered(vidClock, vidRawCaps))
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print(vidClock.link(vidEncode))
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print(vidEncode.link(vidParse))
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print(vidParse.link_filtered(mux, parseCaps))
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print(mux.link(sink))
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pipe.set_state(Gst.State.PLAYING)
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#~ signal.signal(signal.SIGTERM, exit())
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def terminate():
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pipe.set_state(Gst.State.NULL)
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exit()
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@async(daemon=True)
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def errorHandler():
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while 1:
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msg = bus.timed_pop_filtered(1e18, Gst.MessageType.ERROR)
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print('howdy')
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print(msg.parse_error())
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terminate()
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@async(daemon=True)
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def eosHandler():
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while 1:
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msg = bus.timed_pop_filtered(1e18, Gst.MessageType.EOS)
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print('EOS reached')
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terminate()
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try:
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errorHandler()
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eosHandler()
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while 1:
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time.sleep(3600)
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except KeyboardInterrupt:
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class GstException(Exception):
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pass
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def gstPrintMsg(pName, frmt, *args, sName=None):
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if sName:
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print('[{}] [{}] '.format(pName, sName) + frmt.format(*args))
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else:
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print('[{}] '.format(pName) + frmt.format(*args))
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def processErrorMessage(pName, sName, msg):
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error, debug = msg.parse_error()
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if debug:
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gstPrintMsg(pName, '{} - Additional debug info: {}', error.message, debug, sName=sName)
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else:
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gstPrintMsg(pName, error.message, sName=sName)
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raise GstException(error)
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def linkElements(e1, e2, caps=None):
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if caps:
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if not e1.link_filtered(e2, caps):
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raise GstException('cannot link \"{}\" to \"{}\" with caps {}'.format(e1.get_name(), e2.get_name(), caps.to_string()))
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else:
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if not e1.link(e2):
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raise GstException('cannot link \"{}\" to \"{}\"'.format(e1.get_name(), e2.get_name()))
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def linkTee(tee, *args):
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i = 0
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for e in args:
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teePad = tee.get_request_pad('src_{}'.format(i))
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ePad = e.get_static_pad('sink')
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teePad.link(ePad)
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i += 1
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# TODO: make it clearer which pipeline everything comes from
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def eventLoop(pipeline, block=True, doProgress=False, targetState=Gst.State.PLAYING):
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buffering = False
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inProgress = False
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prerolled = targetState != Gst.State.PAUSED
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pName = pipeline.get_name()
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bus = pipeline.get_bus()
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while 1:
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msg = bus.timed_pop(1e18 if block else 0)
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if not msg:
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return
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msgSrc = msg.src
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msgSrcName = msgSrc.get_name()
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msgType = msg.type
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msgTypeName = Gst.MessageType.get_name(msgType)
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# messages that involve manipulating the pipeline
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if msgType == Gst.MessageType.REQUEST_STATE:
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state = msg.parse_request_state()
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print('Setting state to {} as requested by {}'.format(
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Gst.Element.get_state_name(state),
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msgSrcName)
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)
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pipeline.set_state(state)
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elif msgType == Gst.MessageType.CLOCK_LOST:
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print('Clock lost. Getting new one.')
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pipeline.set_state(Gst.State.PAUSED)
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pipeline.set_state(Gst.State.PLAYING)
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elif msgType == Gst.MessageType.LATENCY:
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gstPrintMsg(pName, 'Redistributing latency', sName=msgSrcName)
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pipeline.recalculate_latency()
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# messages that do not require pipeline manipulation
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elif msgType == Gst.MessageType.BUFFERING:
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gstPrintMsg(pName, 'Buffering: {}', msg.parse_buffering(), sName=msgSrcName)
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elif msgType == Gst.MessageType.NEW_CLOCK:
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clock = msg.parse_new_clock()
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clock = clock.get_name() if clock else 'NULL'
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gstPrintMsg(pName, 'New clock: {}', clock)
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elif msgType == Gst.MessageType.INFO:
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error, debug = msg.parse_info()
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if debug:
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gstPrintMsg(pName, debug, sName=msgSrcName)
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elif msgType == Gst.MessageType.WARNING:
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error, debug = msg.parse_warning()
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if debug:
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gstPrintMsg(pName, '{} - Additional debug info: {}', error.message, debug, sName=msgSrcName)
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else:
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gstPrintMsg(pName, error.message, sName=msgSrcName)
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elif msgType == Gst.MessageType.ERROR:
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processErrorMessage(pName, msgSrcName, msg)
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elif msgType == Gst.MessageType.STATE_CHANGED:
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# we only care about pipeline level state changes
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if msgSrc == pipeline:
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old, new, pending = msg.parse_state_changed()
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# we only care if we reach the final target state
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if targetState == Gst.State.PAUSED and new == Gst.State.PAUSED:
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prerolled = True
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if buffering:
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gstPrintMsg(pName, 'Prerolled, waiting for buffering to finish')
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continue
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if inProgress:
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gstPrintMsg(pName, 'Prerolled, waiting for progress to finish')
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continue
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return
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elif msgType == Gst.MessageType.PROGRESS:
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progressType, code, text = msg.parse_progress()
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if (progressType == Gst.ProgressType.START or
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progressType == Gst.ProgressType.CONTINUE):
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if doProgress:
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inProgress = True
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block = True
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elif (progressType == Gst.ProgressType.COMPLETE or
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progressType == Gst.ProgressType.CANCELLED or
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progressType == Gst.ProgressType.ERROR):
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inProgress = False
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gstPrintMsg(pName, 'Progress: ({}) {}', code, text, sName=msgSrcName)
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if doProgress and not inProgress and not buffering and prerolled:
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return
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elif msgType == Gst.MessageType.HAVE_CONTEXT:
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context = msg.parse_have_context()
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gstPrintMsg(
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pName,
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'Got context: {}={}',
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context.get_context_type(),
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context.get_structure().to_string(),
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sName = msgSrcName
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)
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elif msgType == Gst.MessageType.PROPERTY_NOTIFY:
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obj, propName, val = msg.parse_property_notify()
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valStr = '(no value)'
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if val:
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if GObject.type_check_value_holds(val, GObject.TYPE_STRING):
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valStr = val.dup_string()
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elif val.g_type.is_a(Gst.Caps.__gtype__):
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valStr = val.get_boxed().to_string()
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else:
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valStr = Gst.value_serialize(val)
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gstPrintMsg(pName, '{}: {} = {}', obj.get_name(), propName, valStr, sName=msgSrcName)
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# these are things I might not need...
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elif msgType == Gst.MessageType.STREAM_START:
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if msgSrc == pipeline:
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gstPrintMsg(pName, 'Started stream')
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#~ elif msgType == Gst.MessageType.QOS:
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#~ frmt, processed, dropped = msg.parse_qos_stats()
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#~ jitter, proportion, quality = msg.parse_qos_values()
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#~ gstPrintMsg(
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#~ pName,
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#~ 'QOS stats: jitter={} dropped={}',
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#~ jitter,
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#~ '-' if frmt == Gst.Format.UNDEFINED else dropped,
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#~ sName = msgSrcName
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#~ )
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elif msgType == Gst.MessageType.ELEMENT:
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gstPrintMsg(pName, 'Unknown message ELEMENT', sName=msgSrcName)
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elif msgType == Gst.MessageType.UNKNOWN:
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gstPrintMsg(pName, 'Unknown message', sName=msgSrcname)
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@async(daemon=True)
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def startPipeline(pipeline, play=True):
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pName = pipeline.get_name()
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stateChange = pipeline.set_state(Gst.State.PAUSED)
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gstPrintMsg(pName, 'Setting to PAUSED')
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if stateChange == Gst.StateChangeReturn.FAILURE:
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gstPrintMsg(pName, 'Cannot set to PAUSE')
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eventLoop(pipeline, block=False, doProgress=False, targetState=Gst.State.VOID_PENDING)
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# we should always end up here because live
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elif stateChange == Gst.StateChangeReturn.NO_PREROLL:
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gstPrintMsg(pName, 'Live and does not need preroll')
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elif stateChange == Gst.StateChangeReturn.ASYNC:
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gstPrintMsg(pName, 'Prerolling')
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try:
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eventLoop(pipeline, block=True, doProgress=True, targetState=Gst.State.PAUSED)
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except GstException:
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gstPrintMsg(pName, 'Does not want to preroll')
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# some cleanup here?
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raise
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elif stateChange == Gst.StateChangeReturn.SUCCESS:
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gstPrintMsg(pName, 'Is prerolled')
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# this should always succeed...
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try:
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eventLoop(pipeline, block=False, doProgress=True, targetState=Gst.State.PLAYING)
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except GstException:
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gstPrintMsg(pName, 'Does not want to preroll')
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# some cleanup here?
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raise
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# ...and end up here
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else:
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if play:
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gstPrintMsg(pName, 'Setting to PLAYING')
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# and since this will ALWAYS be successful (maybe)...
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if pipeline.set_state(Gst.State.PLAYING) == Gst.StateChangeReturn.FAILURE:
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gstPrintMsg(pName, 'Cannot set to PLAYING')
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err = pipeline.get_bus().pop_filtered(Gst.MessageType.Error)
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processErrorMessage(pName, msgSrcName, err)
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# ...we and end up here and loop until Tool releases their next album
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try:
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eventLoop(pipeline, block=True, doProgress=False, targetState=Gst.State.PLAYING)
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except:
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# cleanup or recover
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raise
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def Camera(video=True, audio=True):
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pipeline = Gst.Pipeline.new("camera")
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if video:
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vSource = Gst.ElementFactory.make("v4l2src", "videoSource")
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vConvert = Gst.ElementFactory.make("videoconvert", "videoConvert")
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vScale = Gst.ElementFactory.make("videoscale", "videoScale")
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vClock = Gst.ElementFactory.make("clockoverlay", "videoClock")
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vEncode = Gst.ElementFactory.make("omxh264enc", "videoEncoder")
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vRTPPay = Gst.ElementFactory.make("rtph264pay", "videoRTPPayload")
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vRTPSink = Gst.ElementFactory.make("multiudpsink", "videoRTPSink")
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vSource.set_property('device', '/dev/video0')
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vRTPPay.set_property('config-interval', 1)
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vRTPPay.set_property('pt', 96)
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vRTPSink.set_property('clients', '127.0.0.1:9001,127.0.0.1:9002')
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vCaps = Gst.Caps.from_string('video/x-raw,width=640,height=480,framerate=30/1')
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pipeline.add(vSource, vConvert, vScale, vClock, vEncode, vRTPPay, vRTPSink)
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linkElements(vSource, vConvert)
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linkElements(vConvert, vScale)
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linkElements(vScale, vClock, vCaps)
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linkElements(vClock, vEncode)
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linkElements(vEncode, vRTPPay)
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linkElements(vRTPPay, vRTPSink)
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if audio:
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aSource = Gst.ElementFactory.make("alsasrc", "audioSource")
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aConvert = Gst.ElementFactory.make("audioconvert", "audioConvert")
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aScale = Gst.ElementFactory.make("audioresample", "audioResample")
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aEncode = Gst.ElementFactory.make("opusenc", "audioEncode")
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aRTPPay = Gst.ElementFactory.make("rtpopuspay", "audioRTPPayload")
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aRTPSink = Gst.ElementFactory.make("multiudpsink", "audioRTPSink")
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aSource.set_property('device', 'hw:1,0')
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aRTPSink.set_property('clients', '127.0.0.1:8001,127.0.0.1:8002')
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aCaps = Gst.Caps.from_string('audio/x-raw,rate=48000,channels=1')
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pipeline.add(aSource, aConvert, aScale, aEncode, aRTPPay, aRTPSink)
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linkElements(aSource, aConvert)
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linkElements(aConvert, aScale)
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linkElements(aScale, aEncode, aCaps)
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linkElements(aEncode, aRTPPay)
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linkElements(aRTPPay, aRTPSink)
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startPipeline(pipeline)
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class FileDump:
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def __init__(self):
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self.pipeline = Gst.Pipeline.new('filedump')
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aSource = Gst.ElementFactory.make('udpsrc', 'audioSource')
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aParse = Gst.ElementFactory.make('opusparse', 'audioParse')
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aQueue = Gst.ElementFactory.make('queue', 'audioQueue')
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vSource = Gst.ElementFactory.make('udpsrc', 'videoSource')
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vParse = Gst.ElementFactory.make('h264parse', 'videoParse')
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vQueue = Gst.ElementFactory.make('queue', 'videoQueue')
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mux = Gst.ElementFactory.make('matroskamux', 'mux')
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self.sink = Gst.ElementFactory.make('filesink', 'sink')
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self.sink.set_property('location', '/dev/null')
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aSource.set_property('port', 8000)
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vSource.set_property('port', 9000)
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self.pipeline.add(aSource, aParse, aQueue, vSource, vParse, vQueue, mux, self.sink)
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linkElements(vSource, vParse)
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linkElements(vParse, vQueue)
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linkElements(vQueue, mux)
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linkElements(aSource, aParse)
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linkElements(aParse, aQueue)
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linkElements(aQueue, mux)
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linkElements(mux, self.sink)
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startPipeline(self.pipeline, play=False)
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def setPath(self, path):
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self.sink.set_property('location', path)
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def play(self):
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self.pipeline.set_state(Gst.State.PLAYING)
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def pause(self):
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self.pipeline.set_state(Gst.State.PAUSED)
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Gst.init(None)
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Camera()
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while 1:
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time.sleep(3600)
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# this works for file dump
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# gst-launch-1.0 -v udpsrc port=8001 ! application/x-rtp,encoding-name=OPUS,payload=96 !
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# rtpjitterbuffer ! rtpopusdepay ! queue ! matroskamux name=mux ! filesink location=testicle.mkv \
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# udpsrc port=9001 ! application/x-rtp,encoding-name=H264,payload=96 ! rtpjitterbuffer ! \
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# rtph264depay ! h264parse ! queue ! mux.
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