From d83c269464cf301fc71067ecf7702eefd6f5816d Mon Sep 17 00:00:00 2001
From: Aleksey Lim <alsroot@member.fsf.org>
Date: Thu, 05 Feb 2009 23:35:29 +0000
Subject: Switch Espeak to gst-plugins-espeak

---
(limited to 'Speak.activity/audio.py')

diff --git a/Speak.activity/audio.py b/Speak.activity/audio.py
index 1176fb5..646dac9 100644
--- a/Speak.activity/audio.py
+++ b/Speak.activity/audio.py
@@ -23,17 +23,17 @@
 
 # This code is a stripped down version of the audio grabber from Measure
 
-import pygst
-pygst.require("0.10")
 import gst
 import pygtk
 import gtk, gobject
 import signal, os
 import time
 import dbus
-import audioop
+import logging
 from struct import *
 
+logger = logging.getLogger('speak')
+
 class AudioGrab(gobject.GObject):
     __gsignals__ = {
         'new-buffer': (gobject.SIGNAL_RUN_FIRST, None, [gobject.TYPE_PYOBJECT])
@@ -43,13 +43,29 @@ class AudioGrab(gobject.GObject):
         gobject.GObject.__init__(self)
         self.pipeline = None
 
-    def playfile(self, filename):
+    def playfile(self, status, text):
+        pitch = int(status.pitch)
+        rate = int(status.rate)
+        # espeak uses 80 to 370
+        rate = 80 + (370-80) * 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 = 'filesrc name=file-source ! decodebin ! tee name=tee tee.! audioconvert ! alsasink tee.! queue ! audioconvert name=conv'
+        p = 'espeak text="%s" pitch=%d rate=%d voice=%s ' \
+            '! decodebin ' \
+            '! tee name=tee ' \
+            'tee.! audioconvert ' \
+                '! alsasink ' \
+            'tee.! queue ' \
+                '! audioconvert name=conv' \
+                % (text, pitch, rate, status.voice.name)
         self.pipeline = gst.parse_launch(p)
         
         # make a fakesink to capture audio
@@ -58,13 +74,14 @@ class AudioGrab(gobject.GObject):
         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()
         
@@ -72,42 +89,22 @@ class AudioGrab(gobject.GObject):
         # we should stop the sound device and stop emitting buffers
         # to save on CPU and battery usage when there is no audio playing
 
-    def playfd(self, fd):
-        self.stop_sound_device()
-
-        # 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
-        if self.pipeline is None:
-            p = 'fdsrc name=fd-source ! wavparse ! 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)
+    def _gstmessage_cb(self, bus, message):
+        type = message.type
 
-            # attach it to the pipeline
-            conv = self.pipeline.get_by_name("conv")
-            gst.element_link_many(conv, fakesink)
+        if type == gst.MESSAGE_EOS:
+            # END OF SOUND FILE
+            self.stop_sound_device()
+        elif type == gst.MESSAGE_ERROR:
+            self.stop_sound_device()
 
-        # set the source file
-        self.pipeline.get_by_name("fd-source").set_property('fd', fd)
-
-        # 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)
+        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):
@@ -116,9 +113,15 @@ class AudioGrab(gobject.GObject):
         return True
 
     def stop_sound_device(self):
-        if self.pipeline is not None:
-            self.pipeline.set_state(gst.STATE_NULL)
+        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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