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# Speak.activity
# A simple front end to the espeak text-to-speech engine on the XO laptop
# http://wiki.laptop.org/go/Speak
#
# Copyright (C) 2008 Joshua Minor
# This file is part of Speak.activity
#
# Parts of Speak.activity are based on code from Measure.activity
# Copyright (C) 2007 Arjun Sarwal - arjun@laptop.org
#
# Speak.activity is free software: you can redistribute it and/or modify
# it under the terms of the GNU General Public License as published by
# the Free Software Foundation, either version 3 of the License, or
# (at your option) any later version.
#
# Speak.activity is distributed in the hope that it will be useful,
# but WITHOUT ANY WARRANTY; without even the implied warranty of
# MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
# GNU General Public License for more details.
#
# You should have received a copy of the GNU General Public License
# along with Speak.activity. If not, see <http://www.gnu.org/licenses/>.
# This code is a stripped down version of the audio grabber from Measure
import subprocess
import pygst
pygst.require("0.10")
import gst
import pygtk
import gtk, gobject
import signal, os
import time
import dbus
import logging
from struct import *
logger = logging.getLogger('speak')
try:
import gst
gst.element_factory_make('espeak')
PITCH_MAX = 200
RATE_MAX = 200
PITCH_DEFAULT = PITCH_MAX/2
RATE_DEFAULT = RATE_MAX/2
except:
PITCH_MAX = 99
RATE_MAX = 99
PITCH_DEFAULT = PITCH_MAX/2
RATE_DEFAULT = RATE_MAX/3
class AudioGrab(gobject.GObject):
__gsignals__ = {
'new-buffer': (gobject.SIGNAL_RUN_FIRST, None, [gobject.TYPE_PYOBJECT])
}
def __init__(self):
gobject.GObject.__init__(self)
self.pipeline = None
def speak(self, status, text):
try:
self._speak(status, text)
except Exception, e:
logger.error('Cannot speak "%s": %s' % (text, e))
# espeak uses 80 to 370
rate = 80 + (370-80) * int(status.rate) / 100
wavpath = "/tmp/speak.wav"
subprocess.call(["espeak", "-w", wavpath, "-p", str(status.pitch), "-s", str(rate), "-v", status.voice.name, text], stdout=subprocess.PIPE)
self._playfile(wavpath)
def _speak(self, status, text):
pitch = int(status.pitch) - 100
rate = int(status.rate) - 100
logger.debug('pitch=%d rate=%d voice=%s text=%s' % (pitch, rate,
status.voice.name, text))
self.stop_sound_device()
self._quiet = False
# 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 = 'espeak name=espeak ' \
'! wavenc ! decodebin ' \
'! tee name=tee ' \
'tee.! audioconvert ' \
'! alsasink ' \
'tee.! queue ' \
'! audioconvert name=conv'
self.pipeline = gst.parse_launch(p)
espeak = self.pipeline.get_by_name('espeak')
espeak.props.text = text
espeak.props.pitch = pitch
espeak.props.rate = rate
espeak.props.voice = status.voice.name
# 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)
bus = self.pipeline.get_bus()
bus.add_signal_watch()
bus.connect('message', self._gstmessage_cb)
# attach it to the pipeline
conv = self.pipeline.get_by_name("conv")
gst.element_link_many(conv, fakesink)
# 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 _playfile(self, filename):
self.stop_sound_device()
self._quiet = False
# 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)
bus = self.pipeline.get_bus()
bus.add_signal_watch()
bus.connect('message', self._gstmessage_cb)
# 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 _gstmessage_cb(self, bus, message):
type = message.type
if type == gst.MESSAGE_EOS:
# END OF SOUND FILE
self.stop_sound_device()
elif type == gst.MESSAGE_ERROR:
self.stop_sound_device()
def on_quit(self):
self.pipeline.set_state(gst.STATE_NULL)
def _new_buffer(self, buf):
if not self._quiet:
# 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):
if self.pipeline is None:
return
self.pipeline.set_state(gst.STATE_NULL)
# Shut theirs mouths down
self._new_buffer('')
self._quiet = True
def restart_sound_device(self):
self.pipeline.set_state(gst.STATE_NULL)
self.pipeline.set_state(gst.STATE_PLAYING)
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