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Move mic samples out of cloud-client so that cloud-client samples can…
… be run in cloud shell [(#2062)](GoogleCloudPlatform/python-docs-samples#2062)
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packages/google-cloud-speech/samples/microphone/README.rst
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.. This file is automatically generated. Do not edit this file directly. | ||
Google Cloud Speech API Python Samples | ||
=============================================================================== | ||
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.. image:: https://gstatic.com/cloudssh/images/open-btn.png | ||
:target: https://console.cloud.google.com/cloudshell/open?git_repo=https://github.com/GoogleCloudPlatform/python-docs-samples&page=editor&open_in_editor=speech/microphone/README.rst | ||
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This directory contains samples for Google Cloud Speech API. The `Google Cloud Speech API`_ enables easy integration of Google speech recognition technologies into developer applications. Send audio and receive a text transcription from the Cloud Speech API service. | ||
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- See the `migration guide`_ for information about migrating to Python client library v0.27. | ||
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.. _migration guide: https://cloud.google.com/speech/docs/python-client-migration | ||
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.. _Google Cloud Speech API: https://cloud.google.com/speech/docs/ | ||
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Setup | ||
------------------------------------------------------------------------------- | ||
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Authentication | ||
++++++++++++++ | ||
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This sample requires you to have authentication setup. Refer to the | ||
`Authentication Getting Started Guide`_ for instructions on setting up | ||
credentials for applications. | ||
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.. _Authentication Getting Started Guide: | ||
https://cloud.google.com/docs/authentication/getting-started | ||
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Install Dependencies | ||
++++++++++++++++++++ | ||
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#. Clone python-docs-samples and change directory to the sample directory you want to use. | ||
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.. code-block:: bash | ||
$ git clone https://github.com/GoogleCloudPlatform/python-docs-samples.git | ||
#. Install `pip`_ and `virtualenv`_ if you do not already have them. You may want to refer to the `Python Development Environment Setup Guide`_ for Google Cloud Platform for instructions. | ||
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.. _Python Development Environment Setup Guide: | ||
https://cloud.google.com/python/setup | ||
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#. Create a virtualenv. Samples are compatible with Python 2.7 and 3.4+. | ||
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.. code-block:: bash | ||
$ virtualenv env | ||
$ source env/bin/activate | ||
#. Install the dependencies needed to run the samples. | ||
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.. code-block:: bash | ||
$ pip install -r requirements.txt | ||
.. _pip: https://pip.pypa.io/ | ||
.. _virtualenv: https://virtualenv.pypa.io/ | ||
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The client library | ||
------------------------------------------------------------------------------- | ||
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This sample uses the `Google Cloud Client Library for Python`_. | ||
You can read the documentation for more details on API usage and use GitHub | ||
to `browse the source`_ and `report issues`_. | ||
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.. _Google Cloud Client Library for Python: | ||
https://googlecloudplatform.github.io/google-cloud-python/ | ||
.. _browse the source: | ||
https://github.com/GoogleCloudPlatform/google-cloud-python | ||
.. _report issues: | ||
https://github.com/GoogleCloudPlatform/google-cloud-python/issues | ||
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.. _Google Cloud SDK: https://cloud.google.com/sdk/ |
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packages/google-cloud-speech/samples/microphone/README.rst.in
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# This file is used to generate README.rst | ||
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product: | ||
name: Google Cloud Speech API | ||
short_name: Cloud Speech API | ||
url: https://cloud.google.com/speech/docs/ | ||
description: > | ||
The `Google Cloud Speech API`_ enables easy integration of Google speech | ||
recognition technologies into developer applications. Send audio and receive | ||
a text transcription from the Cloud Speech API service. | ||
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- See the `migration guide`_ for information about migrating to Python client library v0.27. | ||
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.. _migration guide: https://cloud.google.com/speech/docs/python-client-migration | ||
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setup: | ||
- auth | ||
- install_deps | ||
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cloud_client_library: true | ||
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folder: speech/microphone |
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packages/google-cloud-speech/samples/microphone/requirements.txt
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google-cloud-speech==0.36.3 | ||
pyaudio==0.2.11 | ||
six==1.12.0 |
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packages/google-cloud-speech/samples/microphone/transcribe_streaming_indefinite.py
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#!/usr/bin/env python | ||
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# Copyright 2018 Google LLC | ||
# | ||
# 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. | ||
NOTE: This module requires the additional dependency `pyaudio`. To install | ||
using pip: | ||
pip install pyaudio | ||
Example usage: | ||
python transcribe_streaming_indefinite.py | ||
""" | ||
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# [START speech_transcribe_infinite_streaming] | ||
from __future__ import division | ||
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import time | ||
import re | ||
import sys | ||
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from google.cloud import speech | ||
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import pyaudio | ||
from six.moves import queue | ||
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# Audio recording parameters | ||
STREAMING_LIMIT = 55000 | ||
SAMPLE_RATE = 16000 | ||
CHUNK_SIZE = int(SAMPLE_RATE / 10) # 100ms | ||
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def get_current_time(): | ||
return int(round(time.time() * 1000)) | ||
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def duration_to_secs(duration): | ||
return duration.seconds + (duration.nanos / float(1e9)) | ||
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class ResumableMicrophoneStream: | ||
"""Opens a recording stream as a generator yielding the audio chunks.""" | ||
def __init__(self, rate, chunk_size): | ||
self._rate = rate | ||
self._chunk_size = chunk_size | ||
self._num_channels = 1 | ||
self._max_replay_secs = 5 | ||
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# Create a thread-safe buffer of audio data | ||
self._buff = queue.Queue() | ||
self.closed = True | ||
self.start_time = get_current_time() | ||
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# 2 bytes in 16 bit samples | ||
self._bytes_per_sample = 2 * self._num_channels | ||
self._bytes_per_second = self._rate * self._bytes_per_sample | ||
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self._bytes_per_chunk = (self._chunk_size * self._bytes_per_sample) | ||
self._chunks_per_second = ( | ||
self._bytes_per_second // self._bytes_per_chunk) | ||
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def __enter__(self): | ||
self.closed = False | ||
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self._audio_interface = pyaudio.PyAudio() | ||
self._audio_stream = self._audio_interface.open( | ||
format=pyaudio.paInt16, | ||
channels=self._num_channels, | ||
rate=self._rate, | ||
input=True, | ||
frames_per_buffer=self._chunk_size, | ||
# 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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return self | ||
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def __exit__(self, type, value, traceback): | ||
self._audio_stream.stop_stream() | ||
self._audio_stream.close() | ||
self.closed = True | ||
# Signal the generator to terminate so that the client's | ||
# streaming_recognize method will not block the process termination. | ||
self._buff.put(None) | ||
self._audio_interface.terminate() | ||
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def _fill_buffer(self, in_data, *args, **kwargs): | ||
"""Continuously collect data from the audio stream, into the buffer.""" | ||
self._buff.put(in_data) | ||
return None, pyaudio.paContinue | ||
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def generator(self): | ||
while not self.closed: | ||
if get_current_time() - self.start_time > STREAMING_LIMIT: | ||
self.start_time = get_current_time() | ||
break | ||
# Use a blocking get() to ensure there's at least one chunk of | ||
# data, and stop iteration if the chunk is None, indicating the | ||
# end of the audio stream. | ||
chunk = self._buff.get() | ||
if chunk is None: | ||
return | ||
data = [chunk] | ||
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# Now consume whatever other data's still buffered. | ||
while True: | ||
try: | ||
chunk = self._buff.get(block=False) | ||
if chunk is None: | ||
return | ||
data.append(chunk) | ||
except queue.Empty: | ||
break | ||
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yield b''.join(data) | ||
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def listen_print_loop(responses, stream): | ||
"""Iterates through server responses and prints them. | ||
The responses passed is a generator that will block until a response | ||
is provided by the server. | ||
Each response may contain multiple results, and each result may contain | ||
multiple alternatives; for details, see https://goo.gl/tjCPAU. Here we | ||
print only the transcription for the top alternative of the top result. | ||
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. | ||
""" | ||
responses = (r for r in responses if ( | ||
r.results and r.results[0].alternatives)) | ||
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num_chars_printed = 0 | ||
for response in responses: | ||
if not response.results: | ||
continue | ||
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# The `results` list is consecutive. For streaming, we only care about | ||
# the first result being considered, since once it's `is_final`, it | ||
# moves on to considering the next utterance. | ||
result = response.results[0] | ||
if not result.alternatives: | ||
continue | ||
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# Display the transcription of the top alternative. | ||
top_alternative = result.alternatives[0] | ||
transcript = top_alternative.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 = ' ' * (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) | ||
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..') | ||
stream.closed = True | ||
break | ||
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num_chars_printed = 0 | ||
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def main(): | ||
client = speech.SpeechClient() | ||
config = speech.types.RecognitionConfig( | ||
encoding=speech.enums.RecognitionConfig.AudioEncoding.LINEAR16, | ||
sample_rate_hertz=SAMPLE_RATE, | ||
language_code='en-US', | ||
max_alternatives=1, | ||
enable_word_time_offsets=True) | ||
streaming_config = speech.types.StreamingRecognitionConfig( | ||
config=config, | ||
interim_results=True) | ||
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mic_manager = ResumableMicrophoneStream(SAMPLE_RATE, CHUNK_SIZE) | ||
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print('Say "Quit" or "Exit" to terminate the program.') | ||
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with mic_manager as stream: | ||
while not stream.closed: | ||
audio_generator = stream.generator() | ||
requests = (speech.types.StreamingRecognizeRequest( | ||
audio_content=content) | ||
for content in audio_generator) | ||
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responses = client.streaming_recognize(streaming_config, | ||
requests) | ||
# Now, put the transcription responses to use. | ||
listen_print_loop(responses, stream) | ||
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if __name__ == '__main__': | ||
main() | ||
# [END speech_transcribe_infinite_streaming] |
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