1
2
3
4
5
6
7
8
9
10
11
12
13
14
15
16
17
18
19
20
21
22
23
24
25
26
27
28
29
30
31
32
33
34
35
36
37
38
39
40
41
42
43
44
45
46
47
48
49
50
51
52
53
54
55
56
57
58
59
60
61
62
63
64
65
66
|
import pygst
pygst.require("0.10")
import gst
import pygtk
import gtk, gobject
import signal, os
import time
import dbus
import audioop
from struct import *
class AudioGrab(gobject.GObject):
__gsignals__ = {
'new-buffer': (gobject.SIGNAL_RUN_FIRST, None, [gobject.TYPE_PYOBJECT])
}
def __init__(self, datastore, _jobject):
gobject.GObject.__init__(self)
self.pipeline = None
def playfile(self, filename):
# build a pipeline that reads the given file
# and sends it to both the real audio output
# and a fake one that we use to draw from
p = 'filesrc name=file-source ! decodebin ! tee name=tee tee.! audioconvert ! alsasink tee.! queue ! audioconvert name=conv'
self.pipeline = gst.parse_launch(p)
# make a fakesink to capture audio
fakesink = gst.element_factory_make("fakesink", "fakesink")
fakesink.connect("handoff",self.on_buffer)
fakesink.set_property("signal-handoffs",True)
self.pipeline.add(fakesink)
# attach it to the pipeline
conv = self.pipeline.get_by_name("conv")
gst.element_link_many(conv, fakesink)
# set the source file
self.pipeline.get_by_name("file-source").set_property('location', filename)
# play
self.restart_sound_device()
# how do we detect when the sample has finished playing?
# we should stop the sound device and stop emitting buffers
# to save on CPU and battery usage when there is no audio playing
def on_quit(self):
self.pipeline.set_state(gst.STATE_NULL)
def _new_buffer(self, buf):
# pass captured audio to anyone who is interested via the main thread
self.emit("new-buffer", buf)
return False
def on_buffer(self,element,buffer,pad):
# we got a new buffer of data, ask for another
gobject.timeout_add(100, self._new_buffer, str(buffer))
return True
def stop_sound_device(self):
self.pipeline.set_state(gst.STATE_NULL)
def restart_sound_device(self):
self.pipeline.set_state(gst.STATE_PLAYING)
|
