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Speech sample using microphone #1013
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#!/usr/bin/env python | ||
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# Copyright 2017 Google Inc. All Rights Reserved. | ||
# | ||
# 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. | ||
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"""Google Cloud Speech API sample application using the streaming API. | ||
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NOTE: This module requires the additional dependency `pyaudio`. To install | ||
using pip: | ||
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pip install pyaudio | ||
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Example usage: | ||
python transcribe_streaming_mic.py | ||
""" | ||
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# [START import_libraries] | ||
from __future__ import division | ||
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import re | ||
import sys | ||
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from google.cloud import speech | ||
import pyaudio | ||
from six.moves import queue | ||
# [END import_libraries] | ||
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# Audio recording parameters | ||
RATE = 16000 | ||
CHUNK = int(RATE / 10) # 100ms | ||
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class MicAsFile(object): | ||
"""Opens a recording stream as a file-like object.""" | ||
def __init__(self, rate, chunk): | ||
self._rate = rate | ||
self._chunk = chunk | ||
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# Create a thread-safe buffer of audio data | ||
self._buff = queue.Queue() | ||
self.closed = True | ||
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def __enter__(self): | ||
self._audio_interface = pyaudio.PyAudio() | ||
self._audio_stream = self._audio_interface.open( | ||
format=pyaudio.paInt16, | ||
# The API currently only supports 1-channel (mono) audio | ||
# https://goo.gl/z757pE | ||
channels=1, rate=self._rate, | ||
input=True, frames_per_buffer=self._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=self._fill_buffer, | ||
) | ||
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self.closed = False | ||
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return self | ||
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def __exit__(self, type, value, traceback): | ||
self._audio_stream.stop_stream() | ||
self._audio_stream.close() | ||
self.closed = True | ||
# Flush out the read, just in case | ||
self._buff.put(None) | ||
self._audio_interface.terminate() | ||
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def _fill_buffer(self, in_data, frame_count, time_info, status_flags): | ||
"""Continuously collect data from the audio stream, into the buffer.""" | ||
self._buff.put(in_data) | ||
return None, pyaudio.paContinue | ||
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def read(self, chunk_size): | ||
if self.closed: | ||
return | ||
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# Use a blocking get() to ensure there's at least one chunk of data. | ||
data = [self._buff.get()] | ||
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# Now consume whatever other data's still buffered. | ||
while True: | ||
try: | ||
data.append(self._buff.get(block=False)) | ||
except queue.Empty: | ||
break | ||
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if self.closed: | ||
return | ||
return b''.join(data) | ||
# [END audio_stream] | ||
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def listen_print_loop(results_gen): | ||
"""Iterates through server responses and prints them. | ||
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The results_gen passed is a generator that will block until a response | ||
is provided by the server. When the transcription response comes, print it. | ||
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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 result in results_gen: | ||
if not result.alternatives: | ||
continue | ||
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# Display the top transcription | ||
transcript = result.transcript | ||
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# 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)) | ||
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if not result.is_final: | ||
sys.stdout.write(transcript + overwrite_chars + '\r') | ||
sys.stdout.flush() | ||
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num_chars_printed = len(transcript) | ||
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else: | ||
print(transcript + overwrite_chars) | ||
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# 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 | ||
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num_chars_printed = 0 | ||
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def main(): | ||
speech_client = speech.Client() | ||
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with MicAsFile(RATE, CHUNK) as stream: | ||
audio_sample = speech_client.sample( | ||
stream=stream, | ||
encoding=speech.encoding.Encoding.LINEAR16, | ||
sample_rate_hertz=RATE) | ||
# See http://g.co/cloud/speech/docs/languages | ||
# for a list of supported languages. | ||
language_code = 'en-US' # a BCP-47 language tag | ||
results_gen = audio_sample.streaming_recognize( | ||
language_code=language_code, interim_results=True) | ||
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# Now, put the transcription responses to use. | ||
listen_print_loop(results_gen) | ||
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if __name__ == '__main__': | ||
main() |
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# Copyright 2016, Google, Inc. | ||
# 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. | ||
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import os | ||
import re | ||
import threading | ||
import time | ||
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import mock | ||
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RESOURCES = os.path.join(os.path.dirname(__file__), 'resources') | ||
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class MockPyAudio(object): | ||
def __init__(self, audio_filename): | ||
self.audio_filename = audio_filename | ||
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def __call__(self, *args): | ||
return self | ||
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def open(self, stream_callback, rate, *args, **kwargs): | ||
self.rate = rate | ||
self.closed = threading.Event() | ||
self.stream_thread = threading.Thread( | ||
target=self.stream_audio, args=( | ||
self.audio_filename, stream_callback, self.closed)) | ||
self.stream_thread.start() | ||
return self | ||
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def close(self): | ||
self.closed.set() | ||
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def stop_stream(self): | ||
pass | ||
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def terminate(self): | ||
pass | ||
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def stream_audio(self, audio_filename, callback, closed, num_frames=512): | ||
with open(audio_filename, 'rb') as audio_file: | ||
while not closed.is_set(): | ||
# Approximate realtime by sleeping for the appropriate time for | ||
# the requested number of frames | ||
time.sleep(num_frames / float(self.rate)) | ||
# audio is 16-bit samples, whereas python byte is 8-bit | ||
num_bytes = 2 * num_frames | ||
chunk = audio_file.read(num_bytes) or b'\0' * num_bytes | ||
callback(chunk, None, None, None) | ||
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@mock.patch.dict('sys.modules', pyaudio=mock.MagicMock( | ||
PyAudio=MockPyAudio(os.path.join(RESOURCES, 'quit.raw')))) | ||
def test_main(capsys): | ||
import transcribe_streaming_mic | ||
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transcribe_streaming_mic.main() | ||
out, err = capsys.readouterr() | ||
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assert re.search(r'quit', out, re.DOTALL | re.I) |
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Copyright date on new file should be 2017