def myblock(tw, args):
''' Record and playback a sound (Sugar only) '''
import os
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
from gettext import gettext as _
class Grecord:
''' A class for creating a gstreamer session for recording audio. '''
def __init__(self, tw):
''' Set up the stream. We save to a raw .wav file and then
convert the sound to .ogg for saving. '''
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._recording = 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):
''' Assemble all the pieces we need. '''
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)
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 is_recording(self):
''' Are we recording? '''
return self._recording
def _get_state(self):
''' What is the state of our gstreamer pipeline? '''
return self._pipeline.get_state()[1]
def start_recording_audio(self):
''' Start the stream in order to start recording. '''
if self._get_state() == gst.STATE_PLAYING:
return
self._pipeline.set_state(gst.STATE_PLAYING)
self._recording = True
def stop_recording_audio(self):
''' Stop recording and then convert the results into a
.ogg file using a new stream. '''
self._pipeline.set_state(gst.STATE_NULL)
self._recording = False
if not os.path.exists(self.capture_file) or \
os.path.getsize(self.capture_file) <= 0:
return
# Remove previous transcoding results.
if os.path.exists(self.save_file):
os.remove(self.save_file)
line = 'filesrc location=' + self.capture_file + \
' 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')
audioFilesink.set_property("location", self.save_file)
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 _transcodeUpdateCb(self, pipe):
''' Where are we in the transcoding process? '''
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):
''' Where are we in the stream? '''
try:
position, format = pipe.query_position(gst.FORMAT_TIME)
except BaseException:
position = gst.CLOCK_TIME_NONE
try:
duration, format = pipe.query_duration(gst.FORMAT_TIME)
except BaseException:
duration = gst.CLOCK_TIME_NONE
return (position, duration)
def _onMuxedAudioMessageCb(self, bus, message, pipe):
''' Clean up at end of stream.'''
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()
os.remove(self.capture_file)
return False
def _bus_message_handler(self, bus, message):
''' Handle any messages associated with the stream. '''
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 audio-record stream instance as tw.grecord so that
# we can use it repeatedly.
if not hasattr(tw, 'grecord'):
tw.grecord = Grecord(tw)
# Sometime we need to parse multiple arguments, e.g., save, savename
save_name = '%s_%s' % (tw.activity.name, _('sound'))
cmd = args[0].lower()
if len(args) > 1:
save_name = str(args[1])
if cmd == 'start' or cmd == _('start').lower():
tw.grecord.start_recording_audio()
elif cmd == 'stop' or cmd == _('stop').lower():
tw.grecord.stop_recording_audio()
elif cmd == 'play' or cmd == _('play').lower():
play_audio_from_file(tw.lc, tw.grecord.save_file)
elif cmd == 'save' or cmd == _('save').lower():
if os.path.exists(tw.grecord.save_file) and tw.running_sugar:
dsobject = datastore.create()
dsobject.metadata['title'] = save_name
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()
def myblock(tw, args):
''' Record and playback a sound (Sugar only) '''
import os
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
from gettext import gettext as _
class Grecord:
''' A class for creating a gstreamer session for recording audio. '''
def __init__(self, tw):
''' Set up the stream. We save to a raw .wav file and then
convert the sound to .ogg for saving. '''
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._recording = 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):
''' Assemble all the pieces we need. '''
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):
''' We use a buffer, which may overflow. '''
cbuffers = queue.get_property("current-level-buffers")
cbytes = queue.get_property("current-level-bytes")
ctime = queue.get_property("current-level-time")
def is_recording(self):
''' Are we recording? '''
return self._recording
def _get_state(self):
''' What is the state of our gstreamer pipeline? '''
return self._pipeline.get_state()[1]
def start_recording_audio(self):
''' Start the stream in order to start recording. '''
if self._get_state() == gst.STATE_PLAYING:
return
self._pipeline.set_state(gst.STATE_PLAYING)
self._recording = True
def stop_recording_audio(self):
''' Stop recording and then convert the results into a
.ogg file using a new stream. '''
self._pipeline.set_state(gst.STATE_NULL)
self._recording = False
if not os.path.exists(self.capture_file) or \
os.path.getsize(self.capture_file) <= 0:
return
# Remove previous transcoding results.
if os.path.exists(self.save_file):
os.remove(self.save_file)
line = 'filesrc location=' + self.capture_file + \
' 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')
audioFilesink.set_property("location", self.save_file)
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 _transcodeUpdateCb(self, pipe):
''' Where are we in the transcoding process? '''
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):
''' Where are we in the stream? '''
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):
''' Clean up at end of stream.'''
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()
os.remove(self.capture_file)
return False
def _bus_message_handler(self, bus, message):
''' Handle any messages associated with the stream. '''
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 audio-record stream instance as tw.grecord so that
# we can use it repeatedly.
if not hasattr(tw, 'grecord'):
tw.grecord = Grecord(tw)
# Sometime we need to parse multiple arguments, e.g., save, savename
save_name = '%s_%s' % (tw.activity.name, _('sound'))
cmd = args[0].lower()
if len(args) > 1:
save_name = str(arg[1])
if cmd == 'start' or cmd == _('start').lower():
tw.grecord.start_recording_audio()
elif cmd == 'stop' or cmd == _('stop').lower():
tw.grecord.stop_recording_audio()
elif cmd == 'play' or cmd == _('play').lower():
play_audio_from_file(tw.lc, tw.grecord.save_file)
elif cmd == 'save' or cmd == _('save').lower():
if os.path.exists(tw.grecord.save_file) and tw.running_sugar:
dsobject = datastore.create()
dsobject.metadata['title'] = save_name
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()