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我想合併兩個代碼。
A.py是webpy代碼。
B.py是Google雲語音（STT）示例代碼。[python]我想合併兩個代碼，但發生webpy錯誤類型'exceptions.keyerror'
但是當我合併兩個代碼，它發生webpy錯誤
type 'exceptions.keyerror'
我插入A.py代碼B.py在main()第一線。
如何合併此代碼？
這是A.py
import web
urls = ("/.*", "hello")
app = web.application(urls, globals())
class hello:
def GET(self):
return 'Hello, world!'
if __name__ == "__main__":
app.run()
這是B.py（谷歌colud語音（STT）例如代碼）
#
# Licensed under the Apache License, Version 2.0 (the "License");
# you may not use this file except in compliance with the License.
# You may obtain a copy of the License at
#
# http://www.apache.org/licenses/LICENSE-2.0
#
# Unless required by applicable law or agreed to in writing, software
# distributed under the License is distributed on an "AS IS" BASIS,
# WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
# See the License for the specific language governing permissions and
# limitations under the License.
"""Sample that streams audio to the Google Cloud Speech API via GRPC."""
from __future__ import division
import contextlib
import functools
import re
import signal
import sys
import web
import google.auth
import google.auth.transport.grpc
import google.auth.transport.requests
from google.cloud.proto.speech.v1beta1 import cloud_speech_pb2
from google.rpc import code_pb2
import grpc
import pyaudio
from six.moves import queue
# Audio recording parameters
RATE = 48000
CHUNK = int(RATE/10) # 100ms
# The Speech API has a streaming limit of 60 seconds of audio*, so keep the
# connection alive for that long, plus some more to give the API time to figure
# out the transcription.
# * https://g.co/cloud/speech/limits#content
DEADLINE_SECS = 60 * 3 + 5
SPEECH_SCOPE = 'https://www.googleapis.com/auth/cloud-platform'
def make_channel(host, port):
"""Creates a secure channel with auth credentials from the environment."""
# Grab application default credentials from the environment
credentials, _ = google.auth.default(scopes=[SPEECH_SCOPE])
# Create a secure channel using the credentials.
http_request = google.auth.transport.requests.Request()
target = '{}:{}'.format(host, port)
return google.auth.transport.grpc.secure_authorized_channel(
credentials, http_request, target)
def _audio_data_generator(buff):
"""A generator that yields all available data in the given buffer.
Args:
buff - a Queue object, where each element is a chunk of data.
Yields:
A chunk of data that is the aggregate of all chunks of data in `buff`.
The function will block until at least one data chunk is available.
"""
stop = False
while not stop:
# Use a blocking get() to ensure there's at least one chunk of data.
data = [buff.get()]
# Now consume whatever other data's still buffered.
while True:
try:
data.append(buff.get(block=False))`enter code here`
except queue.Empty:
break
# `None` in the buffer signals that the audio stream is closed. Yield
# the final bit of the buffer and exit the loop.
if None in data:
stop = True
data.remove(None)
yield b''.join(data)
def _fill_buffer(buff, in_data, frame_count, time_info, status_flags):
"""Continuously collect data from the audio stream, into the buffer."""
buff.put(in_data)
return None, pyaudio.paContinue
# [START audio_stream]
@contextlib.contextmanager
def record_audio(rate, chunk):
"""Opens a recording stream in a context manager."""
# Create a thread-safe buffer of audio data
buff = queue.Queue()
audio_interface = pyaudio.PyAudio()
audio_stream = audio_interface.open(
format=pyaudio.paInt16,
# The API currently only supports 1-channel (mono) audio
channels=1, rate=rate,
input=True, frames_per_buffer=chunk,
# Run the audio stream asynchronously to fill the buffer object.
# This is necessary so that the input device's buffer doesn't overflow
# while the calling thread makes network requests, etc.
stream_callback=functools.partial(_fill_buffer, buff),
)
yield _audio_data_generator(buff)
audio_stream.stop_stream()
audio_stream.close()
# Signal the _audio_data_generator to finish
buff.put(None)
audio_interface.terminate()
# [END audio_stream]
def request_stream(data_stream, rate, interim_results=True):
"""Yields `StreamingRecognizeRequest`s constructed from a recording audio
stream.
Args:
data_stream: A generator that yields raw audio data to send.
rate: The sampling rate in hertz.
interim_results: Whether to return intermediate results, before the
transcription is finalized.
"""
# The initial request must contain metadata about the stream, so the
# server knows how to interpret it.
recognition_config = cloud_speech_pb2.RecognitionConfig(
# There are a bunch of config options you can specify. See
encoding='LINEAR16', # raw 16-bit signed LE samples
sample_rate=rate, # the rate in hertz
# See http://g.co/cloud/speech/docs/languages
# for a list of supported languages.
language_code='ko-KR', # a BCP-47 language tag
)
streaming_config = cloud_speech_pb2.StreamingRecognitionConfig(
interim_results=interim_results,
config=recognition_config,
)
yield cloud_speech_pb2.StreamingRecognizeRequest(
streaming_config=streaming_config)
for data in data_stream:
# Subsequent requests can all just have the content
yield cloud_speech_pb2.StreamingRecognizeRequest(audio_content=data)
def listen_print_loop(recognize_stream):
"""Iterates through server responses and prints them.
The recognize_stream passed is a generator that will block until a response
is provided by the server. When the transcription response comes, print it.
In this case, responses are provided for interim results as well. If the
response is an interim one, print a line feed at the end of it, to allow
the next result to overwrite it, until the response is a final one. For the
final one, print a newline to preserve the finalized transcription.
"""
num_chars_printed = 0
for resp in recognize_stream:
if resp.error.code != code_pb2.OK:
raise RuntimeError('Server error: ' + resp.error.message)
if not resp.results:
continue
# Display the top transcription
result = resp.results[0]
transcript = result.alternatives[0].transcript
# Display interim results, but with a carriage return at the end of the
# line, so subsequent lines will overwrite them.
#
# If the previous result was longer than this one, we need to print
# some extra spaces to overwrite the previous result
overwrite_chars = ' ' * max(0, num_chars_printed - len(transcript))
if not result.is_final:
sys.stdout.write(transcript + overwrite_chars + '\r')
sys.stdout.flush()
num_chars_printed = len(transcript)
else:
print(transcript + overwrite_chars)
# Exit recognition if any of the transcribed phrases could be
# one of our keywords.
if re.search(r'\b(exit|quit)\b', transcript, re.I):
print('Exiting..')
break
num_chars_printed = 0
def main():
urls = ("/.*", "hello")
app = web.application(urls, globals())
class hello:
def GET(self):
return 'Hello, world!'
app.run()
service = cloud_speech_pb2.SpeechStub(
make_channel('speech.googleapis.com', 443))
# For streaming audio from the microphone, there are three threads.
# First, a thread that collects audio data as it comes in
with record_audio(RATE, CHUNK) as buffered_audio_data:
# Second, a thread that sends requests with that data
requests = request_stream(buffered_audio_data, RATE)
# Third, a thread that listens for transcription responses
recognize_stream = service.StreamingRecognize(
requests, DEADLINE_SECS)
# Exit things cleanly on interrupt
signal.signal(signal.SIGINT, lambda *_: recognize_stream.cancel())
# Now, put the transcription responses to use.
try:
listen_print_loop(recognize_stream)
recognize_stream.cancel()
except grpc.RpcError as e:
code = e.code()
# CANCELLED is caused by the interrupt handler, which is expected.
if code is not code.CANCELLED:
raise
if __name__ == '__main__':
main()