def __init__(self, width, height, device):
self.appsink = gst.parse_launch("appsink drop=true max-buffers=1")
cf_yuv = gst.parse_launch("capsfilter caps=\"video/x-raw-yuv,width="+str(width)+",height="+str(height)+"\"")
cf = gst.parse_launch("capsfilter caps=\"video/x-raw-rgb,width="+str(width)+",height="+str(height)+",bpp=24,red_mask=255, green_mask=65280, blue_mask=16711680, endianness=4321\"")
ff = gst.element_factory_make("ffmpegcolorspace", "converter")
src = gst.parse_launch("v4l2src device="+device)
print "creating pipe"
self.pipe = gst.Pipeline(name="ecvpipe")
self.pipe.add(src)
self.pipe.add(cf_yuv)
self.pipe.add(ff)
self.pipe.add(cf)
self.pipe.add(self.appsink)
print "done"
src.link(cf_yuv)
cf_yuv.link(ff)
ff.link(cf)
cf.link(self.appsink)
print "setting state \"playing\""
self.pipe.set_state(gst.STATE_PLAYING)
self.imagewidth = width
self.imageheight = height
def reference(self):
"""
Make the reference videos.
:returns: Paths to video files (see :attr:`files`) and video size (see :attr:`size`).
:rtype: tuple
"""
VTLOG.info("Making reference...")
self.pipeline = parse_launch('filesrc name=source ! decodebin ! videorate ! video/x-raw-yuv,framerate=' + self.conf['framerate'] + '/1 ! filesink name=sink1')
source = self.pipeline.get_by_name('source')
sink1 = self.pipeline.get_by_name('sink1')
location = self.video
self.files['original'].append(location)
source.props.location = location
location = self.conf['tempdir'] + self.conf['num'] + '_ref_original.yuv'
self.files['original'].append(location)
sink1.props.location = location
self.__play()
self.pipeline = parse_launch('filesrc name=source ! decodebin ! videorate ! video/x-raw-yuv,framerate=' + self.conf['framerate'] + '/1 ! ' + self.encoder + ' bitrate=' + self.bitrate \
+ ' ! tee name=t ! queue' + self.__add + ' ! filesink name=sink2 t. ! queue ! decodebin ! filesink name=sink3')
source = self.pipeline.get_by_name('source')
sink2 = self.pipeline.get_by_name('sink2')
sink3 = self.pipeline.get_by_name('sink3')
location = self.video
source.props.location = location
location = self.conf['tempdir'] + self.conf['num'] + '_ref.' + self.conf['codec']
self.files['coded'].append(location)
sink2.props.location = location
location = self.conf['tempdir'] + self.conf['num'] + '_ref.yuv'
self.files['coded'].append(location)
sink3.props.location = location
self.__play()
VTLOG.info("Reference made")
return self.files, self.size
def __init__(self):
gobject.GObject.__init__(self)
self.state = STATE_NULL
self.ready = False
self.eos = False
self.update_timeout = None
self.pipeline = gst.parse_launch('souphttpsrc name=src ! filesink name=sink sync=false')
self.src = self.pipeline.get_by_name('src')
self.sink = self.pipeline.get_by_name('sink')
self.foo = self.pipeline.get_by_name('foo')
self.bus = self.pipeline.get_bus()
self.bus.add_signal_watch()
self.bus.connect("message", self.bus_message_cb)
self.test_pipeline = gst.parse_launch('filesrc name=test_src ! decodebin2 name=decoder ! fakesink')
self.test_src = self.test_pipeline.get_by_name('test_src')
self.test_decoder = self.test_pipeline.get_by_name('decoder')
self.test_decoder.connect('autoplug-continue', self.autoplug_continue_cb)
self.test_bus = self.test_pipeline.get_bus()
self.test_bus.add_signal_watch()
self.test_bus.connect("message", self.test_bus_message_cb)
def videostream(self):
gtk.gdk.threads_enter()
nb = self.gm.get_object('dqmnotebook')
self.webcamarea = gtk.DrawingArea()
nb.prepend_page(self.webcamarea,gtk.Label('Webcam 2'))
nb.show_all()
self.player = gst.parse_launch('souphttpsrc location=http://axisminn02/mjpg/video.mjpg ! decodebin2 ! xvimagesink')
bus = self.player.get_bus()
bus.add_signal_watch()
bus.connect("message", self.deal_with_message)
bus.enable_sync_message_emission()
bus.connect("sync-message::element", self.sync_message)
self.player.set_state(gst.STATE_PLAYING)
self.webcamarea2 = gtk.DrawingArea()
nb.prepend_page(self.webcamarea2,gtk.Label('Webcam 1'))
nb.show_all()
self.player2 = gst.parse_launch('souphttpsrc location=http://axisminn01/mjpg/video.mjpg ! decodebin2 ! xvimagesink')
bus2 = self.player2.get_bus()
bus2.add_signal_watch()
bus2.connect("message", self.deal_with_message2)
bus2.enable_sync_message_emission()
bus2.connect("sync-message::element", self.sync_message2)
self.player2.set_state(gst.STATE_PLAYING)
gtk.gdk.threads_leave()
def create_pipeline(self, mimetype):
if platform.uname()[1].startswith('Nokia'):
self.bus = None
self.player = None
self.source = None
self.sink = None
if mimetype == 'application/ogg':
self.player = gst.parse_launch('gnomevfssrc name=source ! oggdemux ! ivorbisdec ! audioconvert ! dsppcmsink name=sink')
self.player.set_name('oggplayer')
self.set_volume = self.set_volume_dsp_pcm_sink
self.get_volume = self.get_volume_dsp_pcm_sink
else:
self.player = gst.parse_launch('gnomevfssrc name=source ! id3lib ! dspmp3sink name=sink')
self.player.set_name('mp3player')
self.set_volume = self.set_volume_dsp_mp3_sink
self.get_volume = self.get_volume_dsp_mp3_sink
self.source = self.player.get_by_name('source')
self.sink = self.player.get_by_name('sink')
self.player_uri = 'location'
else:
self.player = gst.element_factory_make('playbin', 'player')
self.player_uri = 'uri'
self.source = self.sink = self.player
self.set_volume = self.set_volume_playbin
self.get_volume = self.get_volume_playbin
self.bus = self.player.get_bus()
self.player_clean = True
def __init__(self):
window = gtk.Window(gtk.WINDOW_TOPLEVEL)
window.set_title("Webcam-Viewer")
window.set_default_size(500, 400)
window.connect("destroy", gtk.main_quit, "WM destroy")
vbox = gtk.VBox()
window.add(vbox)
self.movie_window = gtk.DrawingArea()
vbox.add(self.movie_window)
hbox = gtk.HBox()
vbox.pack_start(hbox, False)
hbox.set_border_width(10)
hbox.pack_start(gtk.Label())
self.button = gtk.Button("Start")
self.button.connect("clicked", self.start_stop)
hbox.pack_start(self.button, False)
self.button2 = gtk.Button("Quit")
self.button2.connect("clicked", self.exit)
hbox.pack_start(self.button2, False)
hbox.add(gtk.Label())
window.show_all()
#Skickar
options = "v4l2src ! video/x-raw-yuv,width=352,height=288,framerate=8/1 ! hantro4200enc ! rtph263pay ! udpsink host=130.236.218.162 port=5435"
self.player = gst.parse_launch ( options )
#visar
options2 = "udpsrc port=5434 caps=application/x-rtp,clock-rate=90000 ! rtph263depay ! hantro4100dec ! xvimagesink"
self.player1 = gst.parse_launch( options2 )
bus = self.player.get_bus()
bus.add_signal_watch()
bus.enable_sync_message_emission()
bus.connect("message", self.on_message)
bus.connect("sync-message::element", self.on_sync_message)
def __init__(self, source_pipeline_description, sink_pipeline_description):
gobject.threads_init()
imageprocessing = " ! ".join([
# Buffer the video stream, dropping frames if downstream
# processors aren't fast enough:
"queue name=q leaky=2",
# Convert to a colorspace that templatematch can handle:
"ffmpegcolorspace",
# Detect motion when requested:
"stbt-motiondetect name=motiondetect enabled=false",
# OpenCV image-processing library:
"stbt-templatematch name=templatematch method=1",
])
xvideo = " ! ".join([
# Convert to a colorspace that xvimagesink can handle:
"ffmpegcolorspace",
sink_pipeline_description,
])
screenshot = ("appsink name=screenshot max-buffers=1 drop=true "
"sync=false")
pipe = " ".join([
imageprocessing,
"! tee name=t",
"t. ! queue leaky=2 !", screenshot,
"t. ! queue leaky=2 !", xvideo
])
# Gstreamer loads plugin libraries on demand, when elements that need
# those libraries are first mentioned. There is a bug in gst-opencv
# where it erroneously claims to provide appsink, preventing the
# loading of the real appsink -- so we load it first.
# TODO: Fix gst-opencv so that it doesn't prevent appsink from being
# loaded.
gst.parse_launch("appsink")
self.source_pipeline_description = source_pipeline_description
self.source_bin = self.create_source_bin()
self.sink_bin = gst.parse_bin_from_description(pipe, True)
self.pipeline = gst.Pipeline("stb-tester")
self.pipeline.add(self.source_bin, self.sink_bin)
gst.element_link_many(self.source_bin, self.sink_bin)
self.templatematch = self.pipeline.get_by_name("templatematch")
self.motiondetect = self.pipeline.get_by_name("motiondetect")
self.screenshot = self.pipeline.get_by_name("screenshot")
self.bus = self.pipeline.get_bus()
self.bus.connect("message::error", self.on_error)
self.bus.connect("message::warning", self.on_warning)
self.bus.add_signal_watch()
self.pipeline.set_state(gst.STATE_PLAYING)
# Handle loss of video (but without end-of-stream event) from the
# Hauppauge HDPVR capture device.
self.queue = self.pipeline.get_by_name("q")
self.underrun_timeout_id = None
self.queue.connect("underrun", self.on_underrun)
self.queue.connect("running", self.on_running)
def __init__(self):
window = gtk.Window(gtk.WINDOW_TOPLEVEL)
window.set_title("Webcam-Viewer")
window.set_default_size(500, 400)
window.connect("destroy", gtk.main_quit, "WM destroy")
vbox = gtk.VBox()
window.add(vbox)
self.movie_window = gtk.DrawingArea()
vbox.add(self.movie_window)
hbox = gtk.HBox()
vbox.pack_start(hbox, False)
hbox.set_border_width(10)
hbox.pack_start(gtk.Label())
self.button = gtk.Button("Start")
self.button.connect("clicked", self.start_stop)
hbox.pack_start(self.button, False)
self.button2 = gtk.Button("Quit")
self.button2.connect("clicked", self.exit)
hbox.pack_start(self.button2, False)
hbox.add(gtk.Label())
window.show_all()
#Listening for Input:
# gst-launch udpsrc port=5434 caps=application/x-rtp,clock-rate=90000 ! rtph263depay ! hantro4100dec ! xvimagesink
#Sending Video Output:
# gst-launch v4l2src ! video/x-raw-yuv,width=352,height=288,framerate=8/1 ! hantro4200enc ! rtph263pay ! udpsink host=<other N800's ip> port=5434
#Stream to another device
self.sender = gst.parse_launch("v4l2src ! video/x-raw-yuv,width=320,height=240,framerate=8/1 ! hantro4200enc ! rtph263pay ! udpsink host=130.236.219.107 port=5434")
#Show the incoming video
self.player = gst.parse_launch("udpsrc port=5432 caps=application/x-rtp,clock-rate=90000 ! rtph263depay ! hantro4100dec ! xvimagesink")
#Stream both audio and video
# self.player = gst.parse_launch("v4l2src ! video/x-raw-yuv,width=320,height=240,framerate=15/1 ! hantro4200enc stream-type=1 profile-and-level=1001 !video/x-h263,framerate=15/1 ! rtph263ppay mtu=1438 ! udpsink host=130.236.219.107 port=5434 dsppcmsrc ! queue ! audio/x-raw-int,channels=1,rate=8000 ! mulawenc ! rtppcmupay mtu=1438 ! udpsink host=130.236.219.107 port=5432")
#Even try rate=48000
# Show my webcam
# self.player = gst.parse_launch ("v4l2src ! video/x-raw-yuv, width=320, height=240, framerate=8/1 ! autovideosink")
bus = self.player.get_bus()
bus.add_signal_watch()
bus.enable_sync_message_emission()
bus.connect("message", self.on_message)
bus.connect("sync-message::element", self.on_sync_message)
bus2 = self.sender.get_bus()
bus2.add_signal_watch()
bus2.enable_sync_message_emission()
bus2.connect("message", self.on_message)
bus2.connect("sync-message::element", self.on_sync_message)
def stop_stream(self):
self.button.set_label("Start Stream")
self.player.set_state(gst.STATE_NULL)
devname = self.combodevice.get_active_text()
if self.vdevicetype.get_active_text() == "v4l":
self.player = gst.parse_launch ("v4lsrc device="+devname+" ! video/x-raw-yuv,width=320,height=240 ! autovideosink")
else:
self.player = gst.parse_launch ("v4l2src device="+devname+" ! video/x-raw-yuv,width=320,height=240 ! autovideosink")
bus = self.player.get_bus()
bus.add_signal_watch()
bus.enable_sync_message_emission()
bus.connect("message", self.on_message)
bus.connect("sync-message::element", self.on_sync_message)
self.player.set_state(gst.STATE_PLAYING)
def create_pipeline(self, mimetype):
self.debug("creating pipeline")
if self._is_not_playbin2_friendly():
self.bus = None
self.player = None
self.source = None
self.sink = None
if mimetype == 'application/ogg':
self.player = gst.parse_launch('gnomevfssrc name=source ! oggdemux ! ivorbisdec ! audioconvert ! dsppcmsink name=sink')
self.player.set_name('oggplayer')
self.set_volume = self.set_volume_dsp_pcm_sink
self.get_volume = self.get_volume_dsp_pcm_sink
elif mimetype == 'application/flac':
self.player = gst.parse_launch('gnomevfssrc name=source ! flacdemux ! flacdec ! audioconvert ! dsppcmsink name=sink')
self.player.set_name('flacplayer')
self.set_volume = self.set_volume_dsp_pcm_sink
self.get_volume = self.get_volume_dsp_pcm_sink
else:
self.player = gst.parse_launch('gnomevfssrc name=source ! id3lib ! dspmp3sink name=sink')
self.player.set_name('mp3player')
self.set_volume = self.set_volume_dsp_mp3_sink
self.get_volume = self.get_volume_dsp_mp3_sink
self.source = self.player.get_by_name('source')
self.sink = self.player.get_by_name('sink')
self.player_uri = 'location'
self.mute = self.mute_hack
self.unmute = self.unmute_hack
self.get_mute = self.get_mute_hack
else:
self.player = gst.element_factory_make('playbin2', 'player')
self.player_uri = 'uri'
self.source = self.sink = self.player
self.set_volume = self.set_volume_playbin
self.get_volume = self.get_volume_playbin
self.mute = self.mute_playbin
self.unmute = self.unmute_playbin
self.get_mute = self.get_mute_playbin
audio_sink = gst.element_factory_make(self.audio_sink_name)
self._set_props(audio_sink, self.audio_sink_options)
self.player.set_property("audio-sink", audio_sink)
video_sink = gst.element_factory_make(self.video_sink_name)
self._set_props(video_sink, self.video_sink_options)
self.player.set_property("video-sink", video_sink)
self.bus = self.player.get_bus()
self.player_clean = True
self.bus.connect('message', self.on_message)
self.bus.add_signal_watch()
self.update_LC = LoopingCall(self.update)
def __init__(self, projectDir, posX, posY):
self.projectDir = projectDir
primaryLocation = "\""+projectDir+"/primary-dut.webm\""
secondaryLocation = "\""+projectDir+"/secondary-dut.webm\""
outLocation = "\""+projectDir+"/user-testing.webm\""
finalLocation = "\""+projectDir+"/final.webm\""
#posX, posY = self.get_primary_video_info (primaryLocation)
gstPipe = """filesrc
location="""+secondaryLocation+""" name=filein !
matroskademux name=demux1 ! queue !
vp8dec ! videorate force-fps=15/1 !
video/x-raw-yuv,width=320,height=240,framerate=15/1 !
queue ! videomixer name=mix
sink_0::xpos=0 sink_0::ypos=0
sink_1::xpos="""+str (posX)+"""
sink_1::ypos="""+str (posY)+""" ! vp8enc
quality=10 speed=2 threads=4 ! webmmux name=outmux !
filesink location="""+outLocation+"""
filesrc location="""+primaryLocation+"""
! matroskademux name=demux2 ! queue !
vp8dec ! videorate force-fps=15/1 ! videoscale add-borders=1 !
video/x-raw-yuv,framerate=15/1 ! mix."""
self.element = gst.parse_launch (gstPipe)
pipebus = self.element.get_bus ()
pipebus.add_signal_watch ()
pipebus.connect ("message", self.pipe1_changed_cb)
#second pass add audio - we could do this in the above pipeline but due to a bug it doesn't quite work..
gstPipe = """filesrc
location="""+secondaryLocation+""" ! queue ! matroskademux !
vorbisparse ! audio/x-vorbis ! queue ! outmux.audio_0
filesrc location="""+outLocation+""" ! queue !
matroskademux ! video/x-vp8 ! queue ! outmux.video_0 webmmux
name=outmux ! filesink location="""+finalLocation+""""""
self.element2 = gst.parse_launch (gstPipe)
pipebus2 = self.element2.get_bus ()
pipebus2.add_signal_watch ()
pipebus2.connect ("message", self.pipe2_changed_cb)
def __init__(self):
window = gtk.Window(gtk.WINDOW_TOPLEVEL)
window.set_title("Awesome AP")
window.set_default_size(500, 400)
window.connect("destroy", gtk.main_quit, "WM destroy")
vbox = gtk.VBox()
window.add(vbox)
self.movie_window = gtk.DrawingArea()
vbox.add(self.movie_window)
hbox = gtk.HBox()
vbox.pack_start(hbox, False)
hbox.set_border_width(10)
hbox.pack_start(gtk.Label())
self.button = gtk.Button("Start")
self.button.connect("clicked", self.start_stop)
hbox.pack_start(self.button, False)
self.button2 = gtk.Button("Quit")
self.button2.connect("clicked", self.exit)
hbox.pack_start(self.button2, False)
hbox.add(gtk.Label())
window.show_all()
#HOSTAR!
#filesrc location="$1" ! \
#audio/x-iLBC,rate=8000,channels=1,mode=20 ! \
#dspilbcsink
self.player = gst.parse_launch("udpsrc port=4999 ! audio/x-iLBC,rate=8000,channels=1,mode=20 ! dspilbcsink")
print "skickar ljud"
#CONNECTOR
self.player1 = gst.parse_launch("dspilbcsrc dtx=0 ! audio/x-iLBC,rate=8000,channels=1,mode=20 ! udpsink host=130.236.218.184 port=5000")
print "lyssnar video"
bus = self.player.get_bus()
bus.add_signal_watch()
bus.enable_sync_message_emission()
bus.connect("message", self.on_message)
bus.connect("sync-message::element", self.on_sync_message)
print "startar bus"
bus1 = self.player1.get_bus()
bus1.add_signal_watch()
bus1.enable_sync_message_emission()
bus1.connect("message", self.on_message)
bus1.connect("sync-message::element", self.on_sync_message)
print "startar bus1"
def init_gst(self):
"""Initialize the speech components"""
self.pipeline = gst.parse_launch('gconfaudiosrc ! audioconvert ! audioresample '
+
'! vader name=vad auto-threshold=true '
+ '! pocketsphinx name=asr ! fakesink')
asr = self.pipeline.get_by_name('asr')
asr.connect('partial_result', self.asr_partial_result)
asr.connect('result', self.asr_result)
# Turtle Grammar
# asr.set_property(
# 'lm', '/usr/share/pocketsphinx/model/lm/en/turtle.DMP')
# asr.set_property(
# 'dict', '/usr/share/pocketsphinx/model/lm/en/turtle.dic')
# asr.set_property(
# 'hmm', '/usr/share/pocketsphinx/model/hmm/en_US/hub4wsj_sc_8k')
#English Continuous
asr.set_property('lm', '/usr/share/pocketsphinx/model/lm/en_US/hub4.5000.DMP')
asr.set_property('dict', '/usr/share/pocketsphinx/model/lm/en_US/hub4.5000.dic')
asr.set_property('hmm', '/usr/share/pocketsphinx/model/hmm/en_US/hub4wsj_sc_8k')
asr.set_property('configured', True)
bus = self.pipeline.get_bus()
bus.add_signal_watch()
bus.connect('message::application', self.application_message)
self.pipeline.set_state(gst.STATE_PLAYING)
def __init__(self, options={}):
base.Base.__init__(self, options)
gst.Bin.__init__(self, self.options["name"])
aux = (
pipestr.replace("gc-v4l2-preview", "sink-" + self.options["name"])
.replace("gc-v4l2-enc", self.options["videoencoder"])
.replace("gc-v4l2-mux", self.options["muxer"])
)
if "image/jpeg" in self.options["caps"]:
aux = aux.replace("gc-v4l2-dec", "jpegdec max-errors=-1 ! queue !")
else:
aux = aux.replace("gc-v4l2-dec", "")
# bin = gst.parse_bin_from_description(aux, True)
bin = gst.parse_launch("( {} )".format(aux))
self.add(bin)
self.set_option_in_pipeline("location", "gc-v4l2-src", "device")
self.set_value_in_pipeline(path.join(self.options["path"], self.options["file"]), "gc-v4l2-sink", "location")
self.set_option_in_pipeline("caps", "gc-v4l2-filter", "caps", gst.Caps)
fr = re.findall("framerate *= *[0-9]+/[0-9]+", self.options["caps"])
if fr:
newcaps = "video/x-raw-yuv," + fr[0]
self.set_value_in_pipeline(newcaps, "gc-v4l2-vrate", "caps", gst.Caps)
for pos in ["right", "left", "top", "bottom"]:
self.set_option_in_pipeline("videocrop-" + pos, "gc-v4l2-crop", pos, int)
def __init__(self, uri):
self.pipeline = gst.parse_launch("""
filesrc location=/home/enzo/Videos/Part_Of_Me_15.mp4 !
typefind ! qtdemux name=mux mux.video_00 ! queue !
VideoSink location=%s
""" % uri)
freshness = 30 * 30
self.loop = gobject.MainLoop()
self.publisher = utils.RepoSocketPublisher()
self._handle = pyccn.CCN()
self._get_handle = pyccn.CCN()
self._basename = pyccn.Name('/test/VideoSite')
self._user_basename = self._basename.append('User')
self._register_info_name = self._user_basename.append('Register_Info')
# '/test/VideoSite/User/Register_Info'
self._key = pyccn.CCN.getDefaultKey()
# _name_key is the name of the content object for client to get the public key
self._name_key = self._basename.append('Key')
self._signed_info = pyccn.SignedInfo(self._key.publicKeyID, pyccn.KeyLocator(self._name_key), freshness = freshness)
self._singed_info_frames = pyccn.SignedInfo(self._key.publicKeyID, pyccn.KeyLocator(self._name_key), freshness = 1)
# client list
self.client = []
# key for encrypting content
self._dataKey_flag = False # if True, data key has been published
# self._data_key = os.urandom(16)
self._data_key = '123456789asdfghj'
self._name_of_dataKey = self._basename.append('Movie')
self._name_of_dataKey = self._name_of_dataKey.append('DataKey')
def main(args):
"Tagsetter test, test result with:"
"gst-launch -t playbin uri=file://$PWD/test.avi"
# create a new bin to hold the elements
bin = gst.parse_launch('audiotestsrc num-buffers=100 ! ' +
'lame ! ' +
'avimux name=mux ! ' +
'filesink location=test.avi')
mux = bin.get_by_name('mux')
bus = bin.get_bus()
bus.add_signal_watch()
bus.connect('message::eos', on_eos)
# prepare
bin.set_state(gst.STATE_READY)
# send tags
l = gst.TagList()
l[gst.TAG_ARTIST] = "Unknown Genius"
l[gst.TAG_TITLE] = "Unnamed Artwork"
mux.merge_tags(l, gst.TAG_MERGE_APPEND)
# start playing
bin.set_state(gst.STATE_PLAYING)
try:
mainloop.run()
except KeyboardInterrupt:
pass
# stop the bin
bin.set_state(gst.STATE_NULL)
def start_recognizer(self):
rospy.loginfo("Starting recognizer... ")
self.pipeline = gst.parse_launch(self.launch_config)
self.asr = self.pipeline.get_by_name('asr')
self.asr.connect('partial_result', self.asr_partial_result)
self.asr.connect('result', self.asr_result)
self.asr.set_property('configured', True)
self.asr.set_property('dsratio', 1)
# Configure language model
if rospy.has_param(self._lm_param):
lm = rospy.get_param(self._lm_param)
else:
rospy.logerr('Recognizer not started. Please specify a language model file.')
return
if rospy.has_param(self._dic_param):
dic = rospy.get_param(self._dic_param)
else:
rospy.logerr('Recognizer not started. Please specify a dictionary.')
return
self.asr.set_property('lm', lm)
self.asr.set_property('dict', dic)
self.bus = self.pipeline.get_bus()
self.bus.add_signal_watch()
self.bus_id = self.bus.connect('message::application', self.application_message)
self.pipeline.set_state(gst.STATE_PLAYING)
self.started = True
def __init__(self, stepper):
self.stepper = stepper
listener_desc = 'pulsesrc device="alsa_output.pci-0000_00_07.0.analog-stereo.monitor" ! spectrum bands=8 ! fakesink'
self.listener = gst.parse_launch(listener_desc)
bus = self.listener.get_bus()
bus.add_signal_watch()
bus.connect("message", self.on_message)
def setup(self):
self.srcs.reverse()
for src in self.srcs:
self.pipe.append(' '.join([src['src'].pipe, '! ' + self.name + '.' + src['sink']]))
print ' '.join(self.pipe)
self.process = gst.parse_launch(' '.join(self.pipe))
mixer = self.process.get_by_name("mixer")
for src in self.srcs:
src['pad'] = mixer.get_pad(src['sink'])
src['control'] = gst.Controller(src['pad'], "xpos", "ypos", "alpha")
src['control'].set_interpolation_mode("xpos", gst.INTERPOLATE_LINEAR)
src['control'].set("xpos", 5 * gst.SECOND, src['src'].xpos)
self.osc.add_method('/'+src['sink']+'/xpos', 'i', self.osc_callback)
src['control'].set_interpolation_mode("ypos", gst.INTERPOLATE_LINEAR)
src['control'].set("ypos", 5 * gst.SECOND, src['src'].ypos)
self.osc.add_method('/'+src['sink']+'/ypos', 'i', self.osc_callback)
src['control'].set_interpolation_mode("alpha", gst.INTERPOLATE_LINEAR)
src['control'].set("alpha", 5 * gst.SECOND, 1.0)
self.osc.add_method('/'+src['sink']+'/alpha', 'f', self.osc_callback)
def extractAudioTrack (device, track_number, sink, extra = None):
"""
Exctracts an audio track from a certain device. The 'extra' field is used
to send extra properties to the 'cdparanoia' element.
"""
bin = gst.parse_launch ("cdparanoia ! wavenc")
bin.set_state (gst.STATE_PAUSED)
TRACK_FORMAT = gst.format_get_by_nick ("track")
assert TRACK_FORMAT != 0
PLAY_TRACK = TRACK_FORMAT | gst.SEEK_METHOD_SET | gst.SEEK_FLAG_FLUSH
elements = bin.get_list ()
cdparanoia = elements[0]
wavenc = elements[-1]
cdparanoia.set_property ("device", device)
if extra is not None:
for key, value in extra.iteritems ():
cdparanoia.set_property (key, value)
src = cdparanoia.get_pad ("src")
evt = gst.event_new_segment_seek (PLAY_TRACK, track_number, track_number + 1)
src.send_event (evt)
bin.add (sink)
wavenc.link (sink)
return GstOperation(sink, bin)
def _stop_recording_audio(self):
self._pipeline.remove(self._audiobin)
audio_path = os.path.join(Instance.instancePath, "output.wav")
if not os.path.exists(audio_path) or os.path.getsize(audio_path) <= 0:
# FIXME: inform model of failure?
return
if self._audio_pixbuf:
self.model.still_ready(self._audio_pixbuf)
line = 'filesrc location=' + audio_path + ' name=audioFilesrc ! wavparse name=audioWavparse ! audioconvert name=audioAudioconvert ! vorbisenc name=audioVorbisenc ! oggmux name=audioOggmux ! filesink name=audioFilesink'
audioline = gst.parse_launch(line)
taglist = self._get_tags(constants.TYPE_AUDIO)
if self._audio_pixbuf:
pixbuf_b64 = utils.getStringEncodedFromPixbuf(self._audio_pixbuf)
taglist[gst.TAG_EXTENDED_COMMENT] = "coverart=" + pixbuf_b64
vorbis_enc = audioline.get_by_name('audioVorbisenc')
vorbis_enc.merge_tags(taglist, gst.TAG_MERGE_REPLACE_ALL)
audioFilesink = audioline.get_by_name('audioFilesink')
audioOggFilepath = os.path.join(Instance.instancePath, "output.ogg")
audioFilesink.set_property("location", audioOggFilepath)
audioBus = audioline.get_bus()
audioBus.add_signal_watch()
self._audio_transcode_handler = audioBus.connect('message', self._onMuxedAudioMessageCb, audioline)
self._transcode_id = gobject.timeout_add(200, self._transcodeUpdateCb, audioline)
audioline.set_state(gst.STATE_PLAYING)
def GenerateWaveform(self):
"""
Renders the level information for the GUI.
"""
pipe = """filesrc name=src ! decodebin ! audioconvert ! level message=true name=level_element ! fakesink"""
self.loadingPipeline = gst.parse_launch(pipe)
filesrc = self.loadingPipeline.get_by_name("src")
level = self.loadingPipeline.get_by_name("level_element")
filesrc.set_property("location", self.GetAbsFile())
level.set_property("interval", int(self.LEVEL_INTERVAL * gst.SECOND))
self.bus = self.loadingPipeline.get_bus()
self.bus.add_signal_watch()
self.bus.connect("message::element", self.bus_message)
self.bus.connect("message::tag", self.bus_message_tags)
self.bus.connect("message::state-changed", self.bus_message_statechange)
self.bus.connect("message::eos", self.bus_eos)
self.bus.connect("message::error", self.bus_error)
self.levels_list = LevelsList.LevelsList()
self.isLoading = True
self.emit("loading")
self.loadingPipeline.set_state(gst.STATE_PLAYING)
def play(self):
if not self.parsed:
command = " ! ".join(self.command)
debug("launching: '%s'" % command)
try:
self.pipeline = gst.parse_launch(command)
bus = self.pipeline.get_bus()
assert not self.connected_signals
self.connected_signals = []
for name, signal, callback in self.signals:
if name:
element = self.pipeline.get_by_name(name)
else:
element = bus
sid = element.connect(signal, callback)
self.connected_signals.append((element, sid))
self.parsed = True
except gobject.GError, e:
show_error("GStreamer error when creating pipeline", str(e))
self.error = str(e)
self.eos = True
self.done()
return
bus.add_signal_watch()
watch_id = bus.connect("message", self.on_message)
self.watch_id = watch_id
def __init__ (self, source):
super (IsWavPcm, self).__init__ ()
bin = gst.parse_launch (
"typefind name=iwp_typefind ! \
wavparse name=iwp_wavparse ! "
+ WavPcmParse +
" ! fakesink name=iwp_fakesink"
)
self.oper = GstOperation(pipeline = bin)
self.oper.listeners.append (self)
decoder = bin.get_by_name ("iwp_typefind")
sink = bin.get_by_name ("iwp_fakesink")
self.oper.query_element = sink
sink.set_property ("signal-handoffs", True)
sink.connect ("handoff", self.on_handoff)
waveparse = bin.get_by_name ("iwp_wavparse")
waveparse.connect (NEW_PAD_SIGNAL, self.on_new_pad)
self.oper.bin.add (source)
source.link (decoder)
self.isPcm = False
def copyThumbPic(self, fsink, buffer, pad, user_data=None):
if not self._thumb_exposure_open:
return
self._thumb_exposure_open = False
loader = gtk.gdk.pixbuf_loader_new_with_mime_type("image/jpeg")
loader.write(buffer)
loader.close()
self.thumbBuf = loader.get_pixbuf()
self.model.still_ready(self.thumbBuf)
self._thumb_element('thumb_tee').unlink(self._thumb_element('thumb_queue'))
oggFilepath = os.path.join(Instance.instancePath, "output.ogg") #ogv
wavFilepath = os.path.join(Instance.instancePath, "output.wav")
muxFilepath = os.path.join(Instance.instancePath, "mux.ogg") #ogv
muxline = gst.parse_launch('filesrc location=' + str(oggFilepath) + ' name=muxVideoFilesrc ! oggdemux name=muxOggdemux ! theoraparse ! oggmux name=muxOggmux ! filesink location=' + str(muxFilepath) + ' name=muxFilesink filesrc location=' + str(wavFilepath) + ' name=muxAudioFilesrc ! wavparse name=muxWavparse ! audioconvert name=muxAudioconvert ! vorbisenc name=muxVorbisenc ! muxOggmux.')
taglist = self._get_tags(constants.TYPE_VIDEO)
vorbis_enc = muxline.get_by_name('muxVorbisenc')
vorbis_enc.merge_tags(taglist, gst.TAG_MERGE_REPLACE_ALL)
muxBus = muxline.get_bus()
muxBus.add_signal_watch()
self._video_transcode_handler = muxBus.connect('message', self._onMuxedVideoMessageCb, muxline)
self._mux_pipes.append(muxline)
#add a listener here to monitor % of transcoding...
self._transcode_id = gobject.timeout_add(200, self._transcodeUpdateCb, muxline)
muxline.set_state(gst.STATE_PLAYING)
def __init__(self, options={}):
base.Base.__init__(self, options)
gst.Bin.__init__(self, self.options['name'])
aux = pipestr.replace("gc-hauppauge-preview", "sink-" + self.options['name'])
#bin = gst.parse_bin_from_description(aux, True)
bin = gst.parse_launch("( {} )".format(aux))
self.add(bin)
sink = self.get_by_name("gc-hauppauge-device-src")
sink.set_property("device", self.options["locprevideo"])
sink = self.get_by_name("gc-hauppauge-file-src")
sink.set_property("location", self.options["location"])
sink = self.get_by_name("gc-hauppauge-audio-src")
sink.set_property("location", self.options["locpreaudio"])
if self.options["player"] == False:
self.mute = True
element = self.get_by_name("gc-hauppauge-volume")
element.set_property("mute", True)
else:
self.mute = False
sink = self.get_by_name("gc-hauppauge-sink")
sink.set_property('location', path.join(self.options['path'], self.options['file']))
if self.options["vumeter"] == False:
level = self.get_by_name("gc-hauppauge-level")
level.set_property("message", False)
def __init__(self):
window = gtk.Window(gtk.WINDOW_TOPLEVEL)
window.set_title("Webcam-Viewer")
window.set_default_size(500, 400)
window.connect("destroy", gtk.main_quit, "WM destroy")
vbox = gtk.VBox()
window.add(vbox)
self.movie_window = gtk.DrawingArea()
vbox.add(self.movie_window)
hbox = gtk.HBox()
vbox.pack_start(hbox, False)
hbox.set_border_width(10)
hbox.pack_start(gtk.Label())
self.button = gtk.Button("Start")
self.button.connect("clicked", self.start_stop)
hbox.pack_start(self.button, False)
self.button2 = gtk.Button("Quit")
self.button2.connect("clicked", self.exit)
hbox.pack_start(self.button2, False)
hbox.add(gtk.Label())
window.show_all()
# Set up the gstreamer pipeline
self.player = gst.parse_launch ("v4l2src ! autovideosink")
bus = self.player.get_bus()
bus.add_signal_watch()
bus.enable_sync_message_emission()
bus.connect("message", self.on_message)
bus.connect("sync-message::element", self.on_sync_message)
def start(self):
self.pipeline = gst.parse_launch(
"audiotestsrc ! audioconvert ! vorbisenc ! "
"oggmux ! gdppay ! multifdsink name=sink")
self.sink = self.pipeline.get_by_name('sink')
self.sink.connect('client-removed', self.client_removed_handler)
self.pipeline.set_state(gst.STATE_PLAYING)
def setup(self, channels=None, samplerate=None, nframes=None):
super(WavEncoder, self).setup(channels, samplerate, nframes)
# TODO open file for writing
# the output data format we want
self.pipe = ''' appsrc name=src
! audioconvert
! wavenc
'''
if self.filename and self.streaming:
self.pipe += ''' ! tee name=t
! queue ! filesink location=%s
t. ! queue ! appsink name=app sync=False
''' % self.filename
elif self.filename :
self.pipe += '! filesink location=%s ' % self.filename
else:
self.pipe += '! appsink name=app sync=False'
self.pipeline = gst.parse_launch(self.pipe)
# store a pointer to appsrc in our encoder object
self.src = self.pipeline.get_by_name('src')
# store a pointer to appsink in our encoder object
self.app = self.pipeline.get_by_name('app')
srccaps = gst.Caps("""audio/x-raw-float,
endianness=(int)1234,
channels=(int)%s,
width=(int)32,
rate=(int)%d""" % (int(channels), int(samplerate)))
self.src.set_property("caps", srccaps)
# start pipeline
self.pipeline.set_state(gst.STATE_PLAYING)
def _video_eos(self):
self._pipeline.set_state(gst.STATE_NULL)
self._pipeline.get_by_name("tee").unlink(self._videobin)
self._pipeline.remove(self._videobin)
self._pipeline.remove(self._audiobin)
self.model.shutter_sound()
if len(self._thumb_pipes) > 0:
thumbline = self._thumb_pipes[-1]
thumbline.get_by_name('thumb_fakesink').disconnect(self._thumb_handoff_handler)
ogg_path = os.path.join(Instance.instancePath, "output.ogg") #ogv
if not os.path.exists(ogg_path) or os.path.getsize(ogg_path) <= 0:
# FIXME: inform model of failure?
return
line = 'filesrc location=' + ogg_path + ' name=thumbFilesrc ! oggdemux name=thumbOggdemux ! theoradec name=thumbTheoradec ! tee name=thumb_tee ! queue name=thumb_queue ! ffmpegcolorspace name=thumbFfmpegcolorspace ! jpegenc name=thumbJPegenc ! fakesink name=thumb_fakesink'
thumbline = gst.parse_launch(line)
thumb_queue = thumbline.get_by_name('thumb_queue')
thumb_queue.set_property("leaky", True)
thumb_queue.set_property("max-size-buffers", 1)
thumb_tee = thumbline.get_by_name('thumb_tee')
thumb_fakesink = thumbline.get_by_name('thumb_fakesink')
self._thumb_handoff_handler = thumb_fakesink.connect("handoff", self.copyThumbPic)
thumb_fakesink.set_property("signal-handoffs", True)
self._thumb_pipes.append(thumbline)
self._thumb_exposure_open = True
thumbline.set_state(gst.STATE_PLAYING)
def gst_play(self, fname, volume=100, balance=0):
"""Play the given file through gstreamer.
"""
if fname.startswith('file:') or fname.startswith('http:'):
uri = fname
elif config.data.os == 'win32':
uri = 'file:' + urllib.pathname2url(fname)
else:
uri = 'file://' + os.path.abspath(fname)
pipe = gst.parse_launch('uridecodebin name=decode uri=%s ! audioconvert ! audiopanorama panorama=%f ! audioamplify name=amplify amplification=%f ! autoaudiosink' % (uri, float(balance), int(volume) / 100.0 ))
bus = pipe.get_bus()
bus.add_signal_watch()
def eos_cb(b, m):
if m.src == pipe:
pipe.set_state(gst.STATE_NULL)
bus.connect('message::eos', eos_cb)
pipe.set_state(gst.STATE_PLAYING)
# FIXME: since we do not reuse the pipeline, we maybe should clean it up on state_change -> READY
return True
def stop_recording_audio(self):
# We should be able to simply pause and remove the audiobin, but
# this seems to cause a gstreamer segfault. So we stop the whole
# pipeline while manipulating it.
# http://dev.laptop.org/ticket/10183
self._pipeline.set_state(gst.STATE_NULL)
self.model.shutter_sound()
self._pipeline.remove(self._audiobin)
audio_path = os.path.join(Instance.instancePath, "output.wav")
if not os.path.exists(audio_path) or os.path.getsize(audio_path) <= 0:
# FIXME: inform model of failure?
return
if self._audio_pixbuf:
self.model.still_ready(self._audio_pixbuf)
line = 'filesrc location=' + audio_path + ' name=audioFilesrc ! wavparse name=audioWavparse ! audioconvert name=audioAudioconvert ! vorbisenc name=audioVorbisenc ! oggmux name=audioOggmux ! filesink name=audioFilesink'
audioline = gst.parse_launch(line)
taglist = self._get_tags(constants.TYPE_AUDIO)
if self._audio_pixbuf:
pixbuf_b64 = utils.getStringEncodedFromPixbuf(self._audio_pixbuf)
taglist[gst.TAG_EXTENDED_COMMENT] = "coverart=" + pixbuf_b64
vorbis_enc = audioline.get_by_name('audioVorbisenc')
vorbis_enc.merge_tags(taglist, gst.TAG_MERGE_REPLACE_ALL)
audioFilesink = audioline.get_by_name('audioFilesink')
audioOggFilepath = os.path.join(Instance.instancePath, "output.ogg")
audioFilesink.set_property("location", audioOggFilepath)
audioBus = audioline.get_bus()
audioBus.add_signal_watch()
self._audio_transcode_handler = audioBus.connect(
'message', self._onMuxedAudioMessageCb, audioline)
self._transcode_id = gobject.timeout_add(200, self._transcodeUpdateCb,
audioline)
audioline.set_state(gst.STATE_PLAYING)
def start_recognizer(self):
rospy.loginfo("开始语音识别... ")
# rospy.loginfo("Starting recognizer... ")
self.pipeline = gst.parse_launch(self.launch_config) # 解析 麦克风配置
self.asr = self.pipeline.get_by_name('asr') # 自动语音识别 模型
self.asr.connect('partial_result', self.asr_partial_result) # 后面的函数
self.asr.connect('result', self.asr_result)
#self.asr.set_property('configured', True) # 需要开启配置 hmm模型
self.asr.set_property('dsratio', 1)
# 配置语言模型
if rospy.has_param(self._lm_param):
lm = rospy.get_param(self._lm_param)
else:
rospy.logerr('请配置一个语言模型 lm.')
return
if rospy.has_param(self._dic_param):
dic = rospy.get_param(self._dic_param)
else:
rospy.logerr('请配置一个语言字典 dic.')
return
if rospy.has_param(self._hmm_param):
hmm = rospy.get_param(self._hmm_param)
else:
rospy.logerr('请配置一个语言识别模型 hmm.')
return
self.asr.set_property('lm', lm) # 设置asr的语言模型
self.asr.set_property('dict', dic) # 设置asr的字典
self.asr.set_property('hmm', hmm) # 设置asr的识别模型
self.bus = self.pipeline.get_bus()
self.bus.add_signal_watch()
self.bus_id = self.bus.connect('message::application',
self.application_message)
self.pipeline.set_state(gst.STATE_PLAYING)
self.started = True
def __init__(self, parent):
gobject.threads_init()
Frame.__init__(self, parent)
# Parent Object
self.parent = parent
self.parent.title("Lip Reader")
self.parent.geometry("840x640+0+0")
self.parent.resizable(width=FALSE, height=FALSE)
# Video Box
self.movie_window = Canvas(self, width=640, height=440, bg="black")
self.movie_window.pack(side=TOP, expand=YES, fill=BOTH)
# Buttons Box
self.ButtonBox = Frame(self, relief=RAISED, borderwidth=1)
self.ButtonBox.pack(side=BOTTOM, expand=YES, fill=BOTH)
self.closeButton = Button(self.ButtonBox, text="Close", command=self.quit)
self.closeButton.pack(side=RIGHT, padx=5, pady=5)
self.proButton = Button(self.ButtonBox, text="Process", command=self.process)
self.proButton.pack(side=RIGHT, padx=5, pady=5)
self.gotoButton = Button(self.ButtonBox, text="Start", command=self.start_stop)
self.gotoButton.pack(side=LEFT, padx=5, pady=5)
#self.capture =
#self.capture = Label(self, text = "Shoot a Video").grid(row = 0, column = 2, sticky = W+E+N+S)
# Set up the gstreamer pipeline
#self.player = gst.parse_launch ("v4l2src ! video/x-raw-yuv,width=640,height=480 ! ffmpegcolorspace ! xvimagesink ")
self.player = gst.parse_launch ("v4l2src ! tee name=t ! queue ! xvimagesink t. ! queue ! videorate ! video/x-raw-yuv, width=640, height=480, framerate=2/1 ! jpegenc ! multifilesink location=%05d.jpg ")
print "started"
bus = self.player.get_bus()
bus.add_signal_watch()
bus.enable_sync_message_emission()
bus.connect("message", self.on_message)
bus.connect("sync-message::element", self.on_sync_message)
def __init__ (self, parent=None):
global homeDir
QMainWindow.__init__(self, parent)
homeDir = os.environ['HOME'] + '/Documentos/audio/'
self.ui = Ui_MainWindow()
self.ui.setupUi(self)
self.centrar()
self.lista_dispositivos()
oparser = OptionParser()
oparser.add_option("-f", "--file", dest="path",help="save to FILE", metavar="FILE")
oparser.add_option("-d", "--device", dest="device",help="Use device DEVICE", metavar="DEVICE")
(options, args) = oparser.parse_args()
self.ctimer = QtCore.QTimer()
self.ctimer2 = QtCore.QTimer()
self.ui.lineEdit.setText(homeDir)
self.basedatos(0)
self.setWindowIcon(QtGui.QIcon('/usr/usr/share/GrabadorRadio/logo2.png'))
query = QSqlQuery(db)
if query.exec_("SELECT * FROM radios_nombre"):
while query.next():
self.ui.comboBox_2.addItem(query.value(1).toString())
db.close()
db.removeDatabase("rad0")
QtCore.QObject.connect(self.ui.pushButton,QtCore.SIGNAL("clicked()"), self.salir)
QtCore.QObject.connect(self.ui.pushButton_4,QtCore.SIGNAL("clicked()"), self.constant)
QtCore.QObject.connect(self.ui.pushButton_2,QtCore.SIGNAL("clicked()"), self.ruta)
QtCore.QObject.connect(self.ctimer, QtCore.SIGNAL("timeout()"), self.constantUpdate)
QtCore.QMetaObject.connectSlotsByName(self)
#self.basedatos()
self.player = gst.element_factory_make('playbin', 'player')
try:
# alsasink pulsesink osssink autoaudiosink
device = gst.parse_launch('alsasink')
except gobject.GError:
print 'Error: could not launch audio sink'
else:
self.player.set_property('audio-sink', device)
gobject.threads_init()
def __init__(self, loop, filename, frame_queue):
self.loop = loop
self.frame_queue = frame_queue
size = 'width=' + str(COLS) + ',height=' + str(ROWS)
if filename == "--video":
graph = 'v4l2src ! ffmpegcolorspace ! videoscale ! alphacolor ! video/x-raw-rgb,' + size + ',framerate=25/1 ! fakesink name=sink sync=1'
else:
graph = 'filesrc location=' + filename + ' ! decodebin name=decoder\n'
graph += 'decoder. ! ffmpegcolorspace ! videoscale ! alphacolor ! video/x-raw-rgb,' + size + ',framerate=25/1 ! fakesink name=sink sync=1\n'
graph += 'decoder. ! audioconvert ! audioresample ! alsasink'
self.pipeline = gst.parse_launch(graph)
bus = self.pipeline.get_bus()
bus.add_signal_watch()
bus.connect("message::eos", self.bus_watch)
fakesink = self.pipeline.get_by_name('sink')
fakesink.props.signal_handoffs = True
fakesink.connect("handoff", self.callback)
def get_peaks(filename):
global do_run
pipeline_txt = ('filesrc location="%s" ! decodebin ! audioconvert ! '
'audio/x-raw-int,channels=1,rate=44100,endianness=1234,'
'width=32,depth=32,signed=(bool)True !'
'level name=level interval=1000000000 !'
'fakesink' % filename)
pipeline = gst.parse_launch(pipeline_txt)
level = pipeline.get_by_name('level')
bus = pipeline.get_bus()
bus.add_signal_watch()
peaks = []
do_run = True
def show_peak(bus, message):
global do_run
if message.type == gst.MESSAGE_EOS:
pipeline.set_state(gst.STATE_NULL)
do_run = False
return
# filter only on level messages
if message.src is not level or \
not message.structure.has_key('peak'):
return
peaks.append(message.structure['peak'][0])
# connect the callback
bus.connect('message', show_peak)
# run the pipeline until we got eos
pipeline.set_state(gst.STATE_PLAYING)
ctx = gobject.gobject.main_context_default()
while ctx and do_run:
ctx.iteration()
return peaks
def start_recognizer(self):
rospy.loginfo("Starting recognizer... ")
self.pipeline = gst.parse_launch(self.launch_config)
self.asr = self.pipeline.get_by_name('asr')
self.asr.connect('partial_result', self.asr_partial_result)
self.asr.connect('result', self.asr_result)
#self.asr.set_property('configured', True)
self.asr.set_property('dsratio', 1)
# Configure language model
if rospy.has_param(self._lm_param):
lm = rospy.get_param(self._lm_param)
else:
rospy.logerr('Recognizer not started. Please specify a language model file.')
return
if rospy.has_param(self._dic_param):
dic = rospy.get_param(self._dic_param)
else:
rospy.logerr('Recognizer not started. Please specify a dictionary.')
return
if rospy.has_param(self._hmm_param):
hmm = rospy.get_param(self._hmm_param)
else:
rospy.logerr('what is param hmm?')
return
self.asr.set_property('lm', lm)
self.asr.set_property('dict', dic)
self.asr.set_property('hmm', hmm)
self.bus = self.pipeline.get_bus()
self.bus.add_signal_watch()
self.bus_id = self.bus.connect('message::application', self.application_message)
self.pipeline.set_state(gst.STATE_PLAYING)
self.started = True
def start(self, request=None):
self.info("PCMTranscoder start %r %r", request, self.uri)
self.pipeline = gst.parse_launch(
"%s ! decodebin ! audioconvert name=conv" % self.uri)
conv = self.pipeline.get_by_name('conv')
caps = gst.Caps(
"audio/x-raw-int,rate=44100,endianness=4321,channels=2,width=16,depth=16,signed=true"
)
#FIXME: UGLY. 'filter' is a python builtin!
filter = gst.element_factory_make("capsfilter", "filter")
filter.set_property("caps", caps)
self.pipeline.add(filter)
conv.link(filter)
sink = DataSink(destination=self.destination, request=request)
self.pipeline.add(sink)
filter.link(sink)
self.pipeline.set_state(gst.STATE_PLAYING)
d = request.notifyFinish()
d.addBoth(self.requestFinished)
def pipeline(self):
if self._pipeline is None:
gobject.threads_init()
command = " ".join(self.pipeline_command)
log.info('Pipeline: %s', command)
self._pipeline = gst.parse_launch(command)
sphinx = self._pipeline.get_by_name('sphinx')
sphinx.connect('partial_result', self.sphinx_partial_result)
sphinx.connect('result', self.sphinx_result)
sphinx.set_property('configured', True)
self.monitor = self._pipeline.get_by_name('monitor')
bus = self._pipeline.get_bus()
bus.add_signal_watch()
bus.connect('message', self.on_level)
self.pipeline.set_state(gst.STATE_PAUSED)
return self._pipeline
def testPadMonitorActivation(self):
pipeline = gst.parse_launch(
'fakesrc num-buffers=1 ! identity name=id ! fakesink')
identity = pipeline.get_by_name('id')
srcpad = identity.get_pad('src')
monitor = padmonitor.PadMonitor(srcpad, "identity-source",
lambda name: None, lambda name: None)
self.assertEquals(monitor.isActive(), False)
self._run_pipeline(pipeline)
# Now give the reactor a chance to process the callFromThread()
d = defer.Deferred()
def finishTest():
self.assertEquals(monitor.isActive(), True)
monitor.detach()
d.callback(True)
reactor.callLater(0.1, finishTest)
return d
def init_camera(self):
# TODO: This doesn't work when camera resolution is resized at runtime.
if self._pipeline:
self.release()
video_src = self._video_src
if video_src == 'v4l2src':
video_src += ' device=/dev/video%d' % self._index
elif video_src == 'dc1394src':
video_src += ' camera-number=%d' % self._index
GL_CAPS = 'video/x-raw-rgb,red_mask=(int)0xff0000,' + \
'green_mask=(int)0x00ff00,blue_mask=(int)0x0000ff'
pl = '%s ! decodebin name=decoder ! ffmpegcolorspace ! appsink ' + \
'name=camerasink emit-signals=True caps=%s'
self._pipeline = gst.parse_launch(pl % (video_src, GL_CAPS))
self._camerasink = self._pipeline.get_by_name('camerasink')
self._camerasink.connect('new-buffer', self._gst_new_buffer)
self._decodebin = self._pipeline.get_by_name('decoder')
if self._camerasink and not self.stopped:
self.start()
def testPadMonitorTimeout(self):
padmonitor.PadMonitor.PAD_MONITOR_PROBE_INTERVAL = 0.2
padmonitor.PadMonitor.PAD_MONITOR_CHECK_INTERVAL = 0.5
pipeline = gst.parse_launch(
'fakesrc num-buffers=1 ! identity name=id ! fakesink')
identity = pipeline.get_by_name('id')
srcpad = identity.get_pad('src')
# Now give the reactor a chance to process the callFromThread()
def finished():
monitor.detach()
d.callback(True)
def hasInactivated(name):
# We can't detach the monitor from this callback safely, so do
# it from a reactor.callLater()
reactor.callLater(0, finished)
def hasActivated():
self.assertEquals(monitor.isActive(), True)
# Now, we don't send any more data, and after our 0.5 second
# timeout we should go inactive. Pass our test if that happens.
# Otherwise trial will time out.
monitor = padmonitor.PadMonitor(srcpad, "identity-source",
lambda name: None,
hasInactivated)
self.assertEquals(monitor.isActive(), False)
self._run_pipeline(pipeline)
d = defer.Deferred()
reactor.callLater(0.2, hasActivated)
return d
def __init__(self, padre):
self.window = gtk.Window(gtk.WINDOW_TOPLEVEL)
self.window.set_title("Imagen de usuario")
self.window.set_decorated(False)
self.window.set_default_size(260, 232)
self.window.connect("destroy", gtk.main_quit, "WM destroy")
self.padre = padre
vbox = gtk.Fixed()
self.window.add(vbox)
self.movie_window = gtk.DrawingArea()
self.movie_window.set_size_request(260, 200)
vbox.put(self.movie_window, 0, 0)
hbox = gtk.HBox()
hbox.set_size_request(260, 32)
vbox.put(hbox, 0, 200)
self.button3 = gtk.Button("Capturar imagen")
self.button3.connect("clicked", self.take_snapshot)
hbox.pack_start(self.button3, True)
# Set up the gstreamer pipeline
self.player = gst.parse_launch("v4l2src ! autovideosink")
#self.player = gst.parse_launch ("v4l2src ! videorate ! video/x-raw-yuv,framerate=30/1,width=320,height=240 ! tee name=t_vid ! queue ! videoflip method=horizontal-flip ! xvimagesink sync=false t_vid. ! queue ! mux. alsasrc ! audio/x-raw-int,rate=48000,channels=2,depth=16 ! queue ! audioconvert ! queue ! mux. avimux name=mux")
bus = self.player.get_bus()
bus.add_signal_watch()
bus.enable_sync_message_emission()
bus.connect("message", self.on_message)
bus.connect("sync-message::element", self.on_sync_message)
self.window.set_border_width(3)
self.window.set_position(gtk.WIN_POS_CENTER_ALWAYS)
self.window.show_all()
self.start_stop(None)
gtk.gdk.threads_init()
time.sleep(0.1)
gtk.main()
def __init__(self):
""" Initialize the speech pipeline components. """
rospy.init_node('recognizer')
self.pub = rospy.Publisher('~output',String)
rospy.on_shutdown(self.shutdown)
# services to start/stop recognition
rospy.Service("~start", Empty, self.start)
rospy.Service("~stop", Empty, self.stop)
# configure pipeline
self.pipeline = gst.parse_launch('gconfaudiosrc ! audioconvert ! audioresample '
+ '! vader name=vad auto-threshold=true '
+ '! pocketsphinx name=asr ! fakesink')
asr = self.pipeline.get_by_name('asr')
asr.connect('partial_result', self.asr_partial_result)
asr.connect('result', self.asr_result)
asr.set_property('configured', True)
asr.set_property('dsratio', 1)
# parameters for lm and dic
try:
lm_ = rospy.get_param('~lm')
except:
rospy.logerr('Please specify a language model file')
return
try:
dict_ = rospy.get_param('~dict')
except:
rospy.logerr('Please specify a dictionary')
return
asr.set_property('lm',lm_)
asr.set_property('dict',dict_)
bus = self.pipeline.get_bus()
bus.add_signal_watch()
bus.connect('message::application', self.application_message)
self.start(None)
gtk.main()
def init_gst(self):
"""Initialize the speech components"""
self.pipeline = gst.parse_launch(
'gconfaudiosrc ! audioconvert ! audioresample ' +
'! vader name=vad auto-threshold=true ' +
'! pocketsphinx name=asr ! fakesink')
asr = self.pipeline.get_by_name('asr')
asr.connect('partial_result', self.asr_partial_result)
asr.connect('result', self.asr_result)
if self.ps_lm is not None:
asr.set_property('lm', self.ps_lm)
if self.ps_dict is not None:
asr.set_property('dict', self.ps_dict)
asr.set_property('configured', True)
bus = self.pipeline.get_bus()
bus.add_signal_watch()
bus.connect('message::application', self.application_message)
self.pipeline.set_state(gst.STATE_PAUSED)
def main(nom):
#pipeline pour le format ogg
#PIPELINE = "filesrc location=\"test.avi\" ! decodebin ! ffmpegcolorspace ! theoraenc quality=32 ! oggmux ! filesink name=sortie"
#pipeline pour le format mp4
PIPELINE = "filesrc location=\"/home/matthieu/nao/public/videoNao/record.avi\" ! decodebin ! ffmpegcolorspace ! x264enc ! ffmux_mp4 ! filesink name=sortie"
#pipeline à lancer dans le terminal pour la meme chose format ogg
#"gst-launch filesrc location=\"test.avi\" ! decodebin ! ffmpegcolorspace ! theoraenc quality=32 ! oggmux ! filesink location=\"test.ogg\""
FICHIER = "/home/matthieu/nao/public/videoNao/"+nom
pipeline = gst.parse_launch(PIPELINE)
filesink = pipeline.get_by_name("sortie")
filesink.set_property("location", FICHIER)
bus = pipeline.get_bus()
bus.add_signal_watch()
bus.connect( 'message', onSendMessage )
pipeline.set_state(gst.STATE_PLAYING)
time.sleep(5)
pipeline.set_state(gst.STATE_NULL)
print "c'est fini"
def source_to_wav(source, sink):
"""
Converts a given source element to wav format and sends it to sink element.
To convert a media file to a wav using gst-launch:
source ! decodebin ! audioconvert ! audioscale !$_WAV_PCM_PARSE ! wavenc
"""
bin = gst.parse_launch("decodebin name=stw_decodebin !"
"audioconvert ! " + _WAV_PCM_PARSE +
" ! wavenc name=stw_wavenc")
oper = GstOperation(sink, bin)
decoder = bin.get_by_name("stw_decodebin")
encoder = bin.get_by_name("stw_wavenc")
oper.bin.add(source)
oper.bin.add(sink)
source.link(decoder)
encoder.link(sink)
return oper
def start_visualization(self):
# TODO: overlay the data somehow, maybe with some cv+gstreamer thing?
# set up connections
peer = (self.ip, self.port)
iptype = socket.AF_INET6 if ':' in peer[0] else socket.AF_INET
vid_socket = socket.socket(iptype, socket.SOCK_DGRAM)
data_socket = socket.socket(iptype, socket.SOCK_DGRAM)
# set up keepalive
self._viz_running = True
self.vid_thread = threading.Thread(self.send_keepalive_msg, [vid_socket, KA_VIDEO_MSG, peer])
self.vid_thread.start()
self.data_thread = threading.Thread(self.send_keepalive_msg, [data_socket, KA_DATA_MSG, peer])
self.data_thread.start()
# set up gstreamer (blocked here so the rest of the class can run without it installed)
global pygst
global gst
if pygst is None:
import pygst
pygst.require('0.10')
import gst
PIPELINE_DEF = "udpsrc do-timestamp=true name=src blocksize=1316 closefd=false buffer-size=5600 !" \
"mpegtsdemux !" \
"queue !" \
"ffdec_h264 max-threads=0 !" \
"ffmpegcolorspace !" \
"xvimagesink name=video"
self._pipeline = None
try:
self._pipeline = gst.parse_launch(PIPELINE_DEF)
except Exception, e:
print e
self.stop_visualization()
return
def __init__(self, streamname='bbc'):
self.pipeline = None
self.streamname = streamname
streams = {
'bbc': 'http://bbcwssc.ic.llnwd.net/stream/bbcwssc_mp1_ws-eieuk',
'bbc4':
"http://wm-live.bbc.net.uk/wms/bbc_ami/radio4/radio4_bb_live_int_ep1_sl0?BBC-UID=b4adbcdf5ba7eea337440a77610973cee5bd1676f0b0524162a83074f35cc068",
'test': 'rtmp://216.246.37.52/fmr',
'capetalk': 'mms://46.4.25.237/capetalk-live',
'whatsplaying': 'http://commando.nitric.co.za/whatsplaying',
'npr': 'http://69.166.45.60/nwpr'
}
url = streams.get(
streamname,
"http://commando.nitric.co.za/twohundred/s" + streamname)
print "Starting stream pipeline: " + url
self.pipeline = gst.parse_launch('''
souphttpsrc location="%s" !
decodebin ! audio/x-raw-int !
appsink name=sink sync=False''' % url)
#http://mp32.bbc.streamuk.com:80/
#mms://a243.l3944038972.c39440.g.lm.akamaistream.net/D/243/39440/v0001/reflector:38972
self.sink = self.pipeline.get_by_name('sink')
self.pipeline.set_state(gst.STATE_PLAYING)
self.sink.connect('new-buffer', self.on_new_buffer)
self.sink.set_property('emit-signals', True)
self.streamrate = 22050.0 # Hz
if streamname == '220':
self.streamrate = 88200 # Hz
self.bufsize = 576
self.buffer = numpy.zeros(1024)
self.queue = Queue.Queue()
def __init__(self, infile, outfile):
self.player = gst.parse_launch(
'filesrc name=video_in' + ' ! decodebin name=decoder' +
' matroskamux name=mux' + ' ! filesink name=video_out' +
' queue name=q_audio_in' + ' ! audioconvert' + ' ! vorbisenc' +
' ! queue name=q_audio_out' + ' ! mux.' +
' queue name=q_orig_video_in' + ' ! ffmpegcolorspace' +
' ! pngenc compression-level=1 snapshot=false' +
' ! queue name=q_orig_video_out' + ' ! appsink name=png_sink' +
' appsrc name=png_src'
+ # TODO: capsset image/svg+xml ! rsvgdec; for now, output is png; URGENT: set framerate, res, etc.
' ! queue name=q_new_video_in' + ' ! pngdec' +
' ! ffmpegcolorspace' + ' ! theoraenc' +
' ! queue name=q_new_video_out' + ' ! mux.')
self.player.set_state(gst.STATE_NULL)
self.player.get_by_name('video_in').set_property('location', infile)
self.player.get_by_name('video_out').set_property('location', outfile)
self.decoder = self.player.get_by_name('decoder')
self.decoder.connect('new-decoded-pad', self.on_new_decoded_pad)
self.png_sink = self.player.get_by_name('png_sink')
self.png_sink.set_property('drop', False)
self.png_sink.set_property('max_buffers', 1)
self.png_src = self.player.get_by_name('png_src')
player_bus = self.player.get_bus()
player_bus.add_signal_watch()
player_bus.connect('message', self.on_message)
self.app_caps = False
self.ready_lock = threading.Lock()
self.ready_lock.acquire()
self.time_format = gst.Format(gst.FORMAT_TIME)
self.infile = infile
def __init__(self):
window = gtk.Window(gtk.WINDOW_TOPLEVEL)
window.set_title("Webcam-Viewer")
window.set_default_size(400, 350)
window.connect("destroy", gtk.main_quit, "WM destroy")
vbox = gtk.VBox()
window.add(vbox)
self.movie_window = gtk.DrawingArea()
vbox.add(self.movie_window)
hbox = gtk.HBox()
vbox.pack_start(hbox, False)
hbox.set_border_width(10)
hbox.pack_start(gtk.Label())
self.button = gtk.Button("Start")
self.button.connect("clicked", self.start_stop)
hbox.pack_start(self.button, False)
self.button2 = gtk.Button("Quit")
self.button2.connect("clicked", self.exit)
hbox.pack_start(self.button2, False)
hbox.add(gtk.Label())
window.show_all()
# Set up the gstreamer pipeline
#src.set_property("multicast-iface", "eth0")
# src.set_property("caps", Gst.caps_from_string( "application/x-rtp, media=(string)video, clock-rate=(int)90000, encoding-name=(string)MP2T-ES, payload=(int)33" ))
self.player = gst.parse_launch(
"udpsrc uri=udp://0.0.0.0:5000 caps = \"application/x-rtp, media=(string)video, clock-rate=(int)90000, encoding-name=(string)MP4V-ES, payload=(int)96\" ! rtpmp4vdepay ! ffdec_mpeg4 ! ffmpegcolorspace ! ximagesink"
)
#self.player = gst.parse_launch("udpsrc port="+self.MYPORT+" ! audio/x-iLBC,rate=8000,channels=1,mode=20 ! dspilbcsink")
#self.player = gst.parse_launch("dspilbcsrc dtx=0 ! audio/x-iLBC,rate=8000,channels=1,mode=20 ! udpsink host="+self.HOSTIP+" port= "+self.HOSTPORT)
#self.player = gst.parse_launch("v4l2src device=/dev/video1 ! ximagesink")
#self.player = gst.parse_launch("udpsrc port="+self.MYPORT+" caps=application/x-rtp,clock-rate=90000 ! rtph263depay ! hantro4100dec ! xvimagesink")
bus = self.player.get_bus()
bus.add_signal_watch()
bus.enable_sync_message_emission()
bus.connect("message", self.on_message)
bus.connect("sync-message::element", self.on_sync_message)
def copyThumbPic(self, fsink, buffer, pad, user_data=None):
if not self._thumb_exposure_open:
return
self._thumb_exposure_open = False
loader = gtk.gdk.pixbuf_loader_new_with_mime_type("image/jpeg")
loader.write(buffer)
loader.close()
self.thumbBuf = loader.get_pixbuf()
self.model.still_ready(self.thumbBuf)
self._thumb_element('thumb_tee').unlink(
self._thumb_element('thumb_queue'))
oggFilepath = os.path.join(Instance.instancePath, "output.ogg") #ogv
wavFilepath = os.path.join(Instance.instancePath, "output.wav")
muxFilepath = os.path.join(Instance.instancePath, "mux.ogg") #ogv
muxline = gst.parse_launch(
'filesrc location=' + str(oggFilepath) +
' name=muxVideoFilesrc ! oggdemux name=muxOggdemux ! theoraparse ! oggmux name=muxOggmux ! filesink location='
+ str(muxFilepath) + ' name=muxFilesink filesrc location=' +
str(wavFilepath) +
' name=muxAudioFilesrc ! wavparse name=muxWavparse ! audioconvert name=muxAudioconvert ! vorbisenc name=muxVorbisenc ! muxOggmux.'
)
taglist = self._get_tags(constants.TYPE_VIDEO)
vorbis_enc = muxline.get_by_name('muxVorbisenc')
vorbis_enc.merge_tags(taglist, gst.TAG_MERGE_REPLACE_ALL)
muxBus = muxline.get_bus()
muxBus.add_signal_watch()
self._video_transcode_handler = muxBus.connect(
'message', self._onMuxedVideoMessageCb, muxline)
self._mux_pipes.append(muxline)
#add a listener here to monitor % of transcoding...
self._transcode_id = gobject.timeout_add(200, self._transcodeUpdateCb,
muxline)
muxline.set_state(gst.STATE_PLAYING)
def build_pipeline(self):
if self.videofile is None:
return
videofile = self.videofile
if config.data.player['audio-record-device'] not in ('default', ''):
audiosrc = 'alsasrc device=' + config.data.player[
'audio-record-device']
else:
audiosrc = 'alsasrc'
videosrc = 'autovideosrc'
videosink = 'autovideosink'
if not config.data.player['record-video']:
# Generate black image
videosrc = 'videotestsrc pattern=2'
self.pipeline = gst.parse_launch(
'%(videosrc)s name=videosrc ! video/x-raw-yuv,width=352,pixel-aspect-ratio=(fraction)1/1 ! queue ! tee name=tee ! ffmpegcolorspace ! theoraenc drop-frames=1 ! queue ! oggmux name=mux ! filesink location=%(videofile)s %(audiosrc)s name=audiosrc ! audioconvert ! audiorate ! queue ! vorbisenc quality=0.5 ! mux. tee. ! queue ! %(videosink)s name=sink sync=false'
% locals())
self.imagesink = self.pipeline.get_by_name('sink')
self.videosrc = self.pipeline.get_by_name('videosrc')
self.audiosrc = self.pipeline.get_by_name('audiosrc')
self.player = self.pipeline
# Asynchronous XOverlay support.
bus = self.pipeline.get_bus()
bus.enable_sync_message_emission()
def on_sync_message(bus, message):
if message.structure is None:
return
if message.structure.get_name(
) == 'prepare-xwindow-id' and self.xid is not None:
message.src.set_xwindow_id(self.xid)
if hasattr(message.src.props, 'force-aspect-ratio'):
message.src.set_property("force-aspect-ratio", True)
bus.connect('sync-message::element', on_sync_message)
def sourceToWav (source, sink):
"""
Converts a given source element to wav format and sends it to sink element.
To convert a media file to a wav using gst-launch:
source ! decodebin ! audioconvert ! audioscale !
audio/x-raw-int, channels=2, rate=44100, width=16 ! wavenc
"""
bin = gst.parse_launch (
"decodebin ! audioconvert ! audioscale ! "
"audio/x-raw-int, channels=2,rate=44100, width=16 ! wavenc"
)
oper = GstOperation(sink, bin)
elements = bin.get_list ()
decoder = elements[0]
encoder = elements[-1]
oper.bin.add_many (source, sink)
source.link (decoder)
encoder.link (sink)
return oper
def __init__(self, callback=None):
self.is_playing = False
self.filename = None
self.bin = gst.parse_launch("gnomevfssrc name=source ! dspmp3sink")
self.source = self.bin.get_by_name("source")
self.bus = self.bin.get_bus()
self.bus.enable_sync_message_emission()
self.bus.add_signal_watch()
if callback:
def on_message(bus, message):
t = message.type
if t == gst.MESSAGE_EOS:
self.is_playing = False
self.filename = None
callback(self)
elif t == gst.MESSAGE_ERROR:
err, debug = message.parse_error()
print "Error: %s" % err, debug
self.bus.connect("message", on_message)
def __init__ (self, source):
operations.Operation.__init__ (self)
self.__can_start = True
bin = gst.parse_launch ("decodebin ! fakesink")
self.__oper = GstOperation(pipeline = bin)
self.__oper.listeners.append (self)
self.__metadata = {}
self.__error = None
bin.connect ("found-tag", self.__on_found_tag)
# Last element of the bin is the sinks
sink = bin.get_list ()[-1]
sink.set_property ("signal-handoffs", True)
# handoffs are called when it processes one chunk of data
sink.connect ("handoff", self.__on_handoff)
self.__oper.element = sink
# connect source to the first element on the pipeline
source.link (bin.get_list ()[0])
bin.add (source)
def videoSetup(self):
with_audio = ""
if not self.mute:
with_audio = "! queue ! audioconvert ! audiorate ! audioresample ! autoaudiosink"
s = "filesrc name=input ! decodebin2 name=dbin dbin. ! ffmpegcolorspace ! video/x-raw-rgb ! fakesink name=output signal-handoffs=true sync=true dbin. %s" % with_audio
self.player = gst.parse_launch(s)
self.input = self.player.get_by_name('input')
self.fakeSink = self.player.get_by_name('output')
self.input.set_property("location", self.videoSrc)
self.fakeSink.connect("handoff", self.newFrame)
# Catch the end of file as well as errors
# FIXME:
# Messages are sent if i use the following in run():
# gobject.MainLoop().get_context().iteration(True)
# BUT the main python thread then freezes after ~5 seconds...
# unless we use gobject.idle_add(self.player.elements)
bus = self.player.get_bus()
bus.add_signal_watch()
bus.enable_sync_message_emission()
bus.connect("message", self.onMessage)
# Required to prevent the main python thread from freezing, why?!
# Thanks to max26199 for finding this!
gobject.idle_add(self.player.elements)
def __init__(self):
window = gtk.Window(gtk.WINDOW_TOPLEVEL)
window.set_decorated(False)
window.set_title("SelfCam")
# You can change Width and Height below
window.set_default_size(400, 300)
window.connect("destroy", gtk.main_quit, "WM destroy")
vbox = gtk.VBox()
window.add(vbox)
self.cam_view = gtk.DrawingArea()
vbox.add(self.cam_view)
window.show_all()
self.player = gst.parse_launch("v4l2src ! autovideosink")
bus = self.player.get_bus()
bus.add_signal_watch()
bus.enable_sync_message_emission()
bus.connect('message', self.on_message)
bus.connect('sync-message::element', self.on_sync_message)
self.player.set_state(gst.STATE_PLAYING)
def start_pipeline(self, channels, samplerate):
self.pipeline = gst.parse_launch(self.pipe)
# store a pointer to appsrc in our encoder object
self.src = self.pipeline.get_by_name('src')
if self.streaming:
import Queue
self._streaming_queue = Queue.Queue(QUEUE_SIZE)
# store a pointer to appsink in our encoder object
self.app = self.pipeline.get_by_name('app')
self.app.set_property('max-buffers', GST_APPSINK_MAX_BUFFERS)
self.app.set_property("drop", False)
self.app.set_property('emit-signals', True)
self.app.connect("new-buffer", self._on_new_buffer_streaming)
#self.app.connect('new-preroll', self._on_new_preroll_streaming)
srccaps = gst.Caps("""audio/x-raw-float,
endianness=(int)1234,
channels=(int)%s,
width=(int)32,
rate=(int)%d""" % (int(channels), int(samplerate)))
self.src.set_property("caps", srccaps)
self.src.set_property('emit-signals', True)
self.src.set_property('num-buffers', -1)
self.src.set_property('block', False)
self.src.set_property('do-timestamp', True)
self.bus = self.pipeline.get_bus()
self.bus.add_signal_watch()
self.bus.connect("message", self._on_message_cb)
self.mainloop = gobject.MainLoop()
self.mainloopthread = MainloopThread(self.mainloop)
self.mainloopthread.start()
# start pipeline
self.pipeline.set_state(gst.STATE_PLAYING)
