chore(appveyor): Edit appveyor and remove old example

Author: ehdsouza

appveyor.yml

@@ -31,4 +31,4 @@ test_script:
 deploy: off
 
 matrix:
-  fast_finitsh: true
+  fast_finish: true

examples/microphone-speech-to-text.py

@@ -1,22 +1,24 @@
 # You need to install pyaudio to run this example
 # pip install pyaudio
 
-# Note that you need to record just once. You will not be able to send
-# more audio after the initial recording.
-
 from __future__ import print_function
 import pyaudio
-import tempfile
 from watson_developer_cloud import SpeechToTextV1
-from watson_developer_cloud.websocket import RecognizeCallback
+from watson_developer_cloud.websocket import RecognizeCallback, AudioSource
+from threading import Thread
+
+try:
+    from Queue import Queue, Full
+except ImportError:
+    from queue import Queue, Full
 
+# initialize speech to text service
 speech_to_text = SpeechToTextV1(
     username='YOUR SERVICE USERNAME',
     password='YOUR SERVICE PASSWORD',
     url='https://stream.watsonplatform.net/speech-to-text/api')
 
-
-# Example using websockets
+# define callback for the service
 class MyRecognizeCallback(RecognizeCallback):
     def __init__(self):
         RecognizeCallback.__init__(self)
@@ -39,35 +41,65 @@ def on_listening(self):
     def on_hypothesis(self, hypothesis):
         print(hypothesis)
 
+    def on_data(self, data):
+        print(data)
 
-mycallback = MyRecognizeCallback()
-tmp = tempfile.NamedTemporaryFile()
+    def on_close(self):
+        print("Connection closed")
 
+# Variables for recording the speech
 FORMAT = pyaudio.paInt16
 CHANNELS = 1
 RATE = 44100
 CHUNK = 1024
-RECORD_SECONDS = 5
-
+# Note: It will discard if the websocket client can't consumme fast enough
+BUF_MAX_SIZE = CHUNK * 10
+
+# Buffer to store audio
+q = Queue(maxsize=int(round(BUF_MAX_SIZE / CHUNK)))
+
+# define callback to store the recording in queue
+def callback(in_data, frame_count, time_info, status):
+    try:
+        q.put(in_data)
+    except Full:
+        pass # discard
+    return (None, pyaudio.paContinue)
+
+# get ready with service params
+audio_source = AudioSource(q, True, True)
+def recognize_using_weboscket(*args):
+    mycallback = MyRecognizeCallback()
+    speech_to_text.recognize_using_websocket(audio=audio_source,
+                                             content_type='audio/l16; rate=44100',
+                                             recognize_callback=mycallback)
+
+# instantiate pyaudio
 audio = pyaudio.PyAudio()
+
+# open stream using callback
 stream = audio.open(
     format=FORMAT,
     channels=CHANNELS,
     rate=RATE,
     input=True,
-    frames_per_buffer=CHUNK)
-
-print('recording....')
-with open(tmp.name, 'w') as f:
-    for i in range(0, int(RATE / CHUNK * RECORD_SECONDS)):
-        data = stream.read(CHUNK)
-        f.write(data)
-
-stream.stop_stream()
-stream.close()
-audio.terminate()
-print('Done recording...')
-
-with open(tmp.name) as f:
-    speech_to_text.recognize_with_websocket(
-        audio=f, recognize_callback=mycallback)
+    frames_per_buffer=CHUNK,
+    stream_callback=callback,
+    start=False
+)
+
+print("Enter CTRL+C to end recording...")
+stream.start_stream()
+
+try:
+    recognize_thread = Thread(target=recognize_using_weboscket, args=())
+    recognize_thread.start()
+
+    while True:
+        pass
+except KeyboardInterrupt:
+    # stop recording
+    audio_source.completed_recording()
+    stream.stop_stream()
+    stream.close()
+    audio.terminate()