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
67
68
69
70
71
72
73
74
75
76
77
78
79
80
81
82
83
84
85
86
87
88
89
90
91
92
93
94
95
96
97
98
99
100
101
102
103
104
105
106
107
108
109
110
111
112
113
114
115
116
117
118
119
120
121
122
123
124
125
126
127
128
129
130
131
132
133
134
135
136
137
138
139
140
141
142
143
144
145
146
147
148
149
150
151
152
153
154
155
156
157
158
159
160
161
162
163
164
165
166
167
168
169
170
171
172
173
174
175
176
177
178
179
180
181
182
183
184
185
186
187
188
189
190
191
192
193
194
195
196
197
198
199
200
201
202
203
204
205
206
207
208
209
210
211
212
213
214
215
216
217
218
219
220
221
222
223
224
225
226
227
228
229
|
#Copyright (c) 2008, Media Modifications Ltd.
#Copyright (c) 2011, Walter Bender
# This procedure is invoked when the user-definable block on the
# "extras" palette is selected.
# Usage: Import this code into a Python (user-definable) block; Pass
# it 'start' to start recording; 'stop' to stop recording; 'play'
# to play back your recording; or 'save' to save to the Sugar Journal.
def myblock(tw, arg):
''' Record and playback a sound (Sugar only) '''
import os
import time
import gtk
import gst
import gobject
gobject.threads_init()
from TurtleArt.tautils import get_path
from TurtleArt.tagplay import play_audio_from_file
from sugar.datastore import datastore
from sugar import profile
class Grecord:
def __init__(self, tw):
''' Set up the stream. '''
datapath = get_path(tw.parent, 'instance')
self.capture_file = os.path.join(datapath, 'output.wav')
self.save_file = os.path.join(datapath, 'output.ogg')
self._eos_cb = None
self._can_limit_framerate = False
self._playing = False
self._audio_transcode_handler = None
self._transcode_id = None
self._pipeline = gst.Pipeline("Record")
self._create_audiobin()
self._pipeline.add(self._audiobin)
bus = self._pipeline.get_bus()
bus.add_signal_watch()
bus.connect('message', self._bus_message_handler)
def _create_audiobin(self):
src = gst.element_factory_make("alsasrc", "absrc")
# attempt to use direct access to the 0,0 device, solving some A/V
# sync issues
src.set_property("device", "plughw:0,0")
hwdev_available = src.set_state(gst.STATE_PAUSED) != \
gst.STATE_CHANGE_FAILURE
src.set_state(gst.STATE_NULL)
if not hwdev_available:
src.set_property("device", "default")
srccaps = gst.Caps("audio/x-raw-int,rate=16000,channels=1,depth=16")
# guarantee perfect stream, important for A/V sync
rate = gst.element_factory_make("audiorate")
# without a buffer here, gstreamer struggles at the start of the
# recording and then the A/V sync is bad for the whole video
# (possibly a gstreamer/ALSA bug -- even if it gets caught up, it
# should be able to resync without problem)
queue = gst.element_factory_make("queue", "audioqueue")
queue.set_property("leaky", True) # prefer fresh data
queue.set_property("max-size-time", 5000000000) # 5 seconds
queue.set_property("max-size-buffers", 500)
queue.connect("overrun", self._log_queue_overrun)
enc = gst.element_factory_make("wavenc", "abenc")
sink = gst.element_factory_make("filesink", "absink")
sink.set_property("location", self.capture_file)
self._audiobin = gst.Bin("audiobin")
self._audiobin.add(src, rate, queue, enc, sink)
src.link(rate, srccaps)
gst.element_link_many(rate, queue, enc, sink)
def _log_queue_overrun(self, queue):
cbuffers = queue.get_property("current-level-buffers")
cbytes = queue.get_property("current-level-bytes")
ctime = queue.get_property("current-level-time")
def play(self):
if self._get_state() == gst.STATE_PLAYING:
return
self._pipeline.set_state(gst.STATE_PLAYING)
self._playing = True
def pause(self):
self._pipeline.set_state(gst.STATE_PAUSED)
self._playing = False
def stop(self):
self._pipeline.set_state(gst.STATE_NULL)
self._playing = False
def is_playing(self):
return self._playing
def _get_state(self):
return self._pipeline.get_state()[1]
def stop_recording_audio(self):
self.stop()
audio_path = self.capture_file
if not os.path.exists(audio_path) or \
os.path.getsize(audio_path) <= 0:
return
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)
vorbis_enc = audioline.get_by_name('audioVorbisenc')
audioFilesink = audioline.get_by_name('audioFilesink')
audioOggFilepath = self.save_file
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 blockedCb(self, x, y, z):
pass
def record_audio(self):
self.play()
def _transcodeUpdateCb(self, pipe):
position, duration = self._query_position(pipe)
if position != gst.CLOCK_TIME_NONE:
value = position * 100.0 / duration
value = value/100.0
return True
def _query_position(self, pipe):
try:
position, format = pipe.query_position(gst.FORMAT_TIME)
except:
position = gst.CLOCK_TIME_NONE
try:
duration, format = pipe.query_duration(gst.FORMAT_TIME)
except:
duration = gst.CLOCK_TIME_NONE
return (position, duration)
def _onMuxedAudioMessageCb(self, bus, message, pipe):
if message.type != gst.MESSAGE_EOS:
return True
gobject.source_remove(self._audio_transcode_handler)
self._audio_transcode_handler = None
gobject.source_remove(self._transcode_id)
self._transcode_id = None
pipe.set_state(gst.STATE_NULL)
pipe.get_bus().remove_signal_watch()
pipe.get_bus().disable_sync_message_emission()
wavFilepath = self.capture_file
oggFilepath = self.save_file
os.remove( wavFilepath )
return False
def _bus_message_handler(self, bus, message):
t = message.type
if t == gst.MESSAGE_EOS:
if self._eos_cb:
cb = self._eos_cb
self._eos_cb = None
cb()
elif t == gst.MESSAGE_ERROR:
# TODO: if we come out of suspend/resume with errors, then
# get us back up and running... TODO: handle "No space
# left on the resource.gstfilesink.c" err, debug =
# message.parse_error()
pass
# We store the stream as tw.grecord so that we can use it repeatedly.
if not hasattr(tw, 'grecord'):
tw.grecord = Grecord(tw)
# Sometime we need to pass multiple arguments, e.g., save, savename
savename = _('turtle audio recording')
if type(arg) == type([]):
cmd = arg[0]
if len(arg) > 1:
savename = str(arg[1])
else:
cmd = arg
if cmd == _('start'):
tw.grecord.record_audio()
elif cmd == _('stop'):
if os.path.exists(tw.grecord.save_file):
os.remove(tw.grecord.save_file)
tw.grecord.stop_recording_audio()
elif cmd == _('play'):
play_audio_from_file(tw.lc, tw.grecord.save_file)
elif cmd == _('save'):
if os.path.exists(tw.grecord.save_file):
if tw.running_sugar:
dsobject = datastore.create()
dsobject.metadata['title'] = savename
dsobject.metadata['icon-color'] = \
profile.get_color().to_string()
dsobject.metadata['mime_type'] = 'audio/ogg'
dsobject.set_file_path(tw.grecord.save_file)
datastore.write(dsobject)
dsobject.destroy()
|