From 1eeffe9e24f4f8be117f7775eae0f9388b08dc2f Mon Sep 17 00:00:00 2001
From: Aleksey Lim <alsroot@member.fsf.org>
Date: Tue, 14 Jul 2009 03:05:09 +0000
Subject: Simplify piple by using identify plugin; handle warnings in gst bus

---
diff --git a/Speak.activity/espeak.py b/Speak.activity/espeak.py
index 4076e3b..36614f1 100644
--- a/Speak.activity/espeak.py
+++ b/Speak.activity/espeak.py
@@ -21,7 +21,7 @@ logger = logging.getLogger('speak')
 
 supported = True
 
-class AudioGrab(gobject.GObject):
+class BaseAudioGrab(gobject.GObject):
     __gsignals__ = {
         'new-buffer': (gobject.SIGNAL_RUN_FIRST, None, [gobject.TYPE_PYOBJECT])
     }
@@ -48,31 +48,47 @@ class AudioGrab(gobject.GObject):
         self.quiet = True
 
     def make_pipeline(self, cmd):
+        if self.pipeline is not None:
+            self.stop_sound_device()
+            del self.pipeline
+
         # 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
-        self.pipeline = gst.parse_launch(cmd)
-
-        def on_buffer(element, buffer, pad):
+        self.pipeline = gst.parse_launch(
+                cmd + ' ' \
+                '! decodebin ' \
+                '! queue ' \
+                '! identity name=valve ' \
+                '! autoaudiosink')
+
+        def on_buffer(element, buffer):
             # we got a new buffer of data, ask for another
             gobject.timeout_add(100, self._new_buffer, str(buffer))
             return True
 
-        # make a fakesink to capture audio
-        fakesink = gst.element_factory_make("fakesink", "fakesink")
-        fakesink.connect("handoff",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)
+        valve = self.pipeline.get_by_name('valve')
+        valve.props.signal_handoffs = True
+        valve.connect('handoff', on_buffer)
 
         def gstmessage_cb(bus, message):
-            if message.type in (gst.MESSAGE_EOS, gst.MESSAGE_ERROR):
+            self._was_message = True
+
+            if message.type == gst.MESSAGE_WARNING:
+                def check_after_warnings():
+                    if not self._was_message:
+                        self.stop_sound_device()
+                    return True
+
+                logger.debug(message.type)
+                self._was_message = False
+                gobject.timeout_add(500, self._new_buffer, str(buffer))
+
+            elif  message.type in (gst.MESSAGE_EOS, gst.MESSAGE_ERROR):
                 logger.debug(message.type)
                 self.stop_sound_device()
 
+        self._was_message = False
         bus = self.pipeline.get_bus()
         bus.add_signal_watch()
         bus.connect('message', gstmessage_cb)
diff --git a/Speak.activity/espeak_cmd.py b/Speak.activity/espeak_cmd.py
index f391e5b..682bfcd 100644
--- a/Speak.activity/espeak_cmd.py
+++ b/Speak.activity/espeak_cmd.py
@@ -25,16 +25,9 @@ RATE_MAX = 99
 PITCH_DEFAULT = PITCH_MAX/2
 RATE_DEFAULT = RATE_MAX/3
 
-class AudioGrabCmd(espeak.AudioGrab):
+class AudioGrabCmd(espeak.BaseAudioGrab):
     def speak(self, status, text):
-        self.make_pipeline(
-                'filesrc name=file-source ' \
-                '! decodebin ' \
-                '! tee name=tee ' \
-                'tee.! audioconvert' \
-                    '! alsasink ' \
-                'tee.! queue ' \
-                    '! audioconvert name=conv')
+        self.make_pipeline('filesrc name=file-source')
 
         # espeak uses 80 to 370
         rate = 80 + (370-80) * int(status.rate) / 100
diff --git a/Speak.activity/espeak_gst.py b/Speak.activity/espeak_gst.py
index f12453d..85cfa26 100644
--- a/Speak.activity/espeak_gst.py
+++ b/Speak.activity/espeak_gst.py
@@ -23,19 +23,9 @@ RATE_MAX = 200
 PITCH_DEFAULT = PITCH_MAX/2
 RATE_DEFAULT = RATE_MAX/2
 
-class AudioGrabGst(espeak.AudioGrab):
+class AudioGrabGst(espeak.BaseAudioGrab):
     def speak(self, status, text):
-        self.stop_sound_device()
-
-        self.make_pipeline(
-                'espeak name=espeak ' \
-                '! wavenc ! decodebin ' \
-                '! tee name=tee ' \
-                'tee.! audioconvert ' \
-                    '! alsasink ' \
-                'tee.! queue ' \
-                    '! audioconvert name=conv')
-
+        self.make_pipeline('espeak name=espeak ! wavenc')
         src = self.pipeline.get_by_name('espeak')
 
         pitch = int(status.pitch) - 100
diff --git a/Speak.activity/mouth.py b/Speak.activity/mouth.py
index 8b72f4b..b01155a 100644
--- a/Speak.activity/mouth.py
+++ b/Speak.activity/mouth.py
@@ -42,10 +42,15 @@ class Mouth(gtk.DrawingArea):
         audioSource.connect("new-buffer", self._new_buffer)
 
     def _new_buffer(self, obj, buf):
-        self.newest_buffer = list(unpack( str(int(len(buf))/2)+'h' , buf))
-        self.main_buffers += self.newest_buffer
-        if(len(self.main_buffers)>self.buffer_size):
-            del self.main_buffers[0:(len(self.main_buffers)-self.buffer_size)]
+        if len(buf) < 28:
+            self.newest_buffer = []
+        else:
+            self.newest_buffer = list(unpack( str(int(len(buf))/2)+'h' , buf))
+            self.main_buffers += self.newest_buffer
+            if(len(self.main_buffers)>self.buffer_size):
+                del self.main_buffers[0:(len(self.main_buffers)- \
+                        self.buffer_size)]
+
         self.queue_draw()
         return True
 
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