#!/usr/bin/env python # 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. """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_mic.py """ # [START import_libraries] from __future__ import division import re import sys from google.cloud import speech from google.cloud.speech import enums from google.cloud.speech import types import pyaudio from six.moves import queue # [END import_libraries] # Audio recording parameters RATE = 16000 CHUNK = int(RATE / 10) # 100ms class MicrophoneStream(object): """Opens a recording stream as a generator yielding the audio chunks.""" def __init__(self, rate, chunk): self._rate = rate self._chunk = chunk # Create a thread-safe buffer of audio data self._buff = queue.Queue() self.closed = True 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, ) self.closed = False return self 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() 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 def generator(self): while not self.closed: # 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] # 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 yield b''.join(data) # [END audio_stream] def listen_print_loop(responses): """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. """ num_chars_printed = 0 for response in responses: if not response.results: continue # 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 # Display the transcription of the top alternative. 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 = ' ' * (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(): # See http://g.co/cloud/speech/docs/languages # for a list of supported languages. language_code = 'en-US' # a BCP-47 language tag client = speech.SpeechClient() config = types.RecognitionConfig( encoding=enums.RecognitionConfig.AudioEncoding.LINEAR16, sample_rate_hertz=RATE, language_code=language_code) streaming_config = types.StreamingRecognitionConfig( config=config, interim_results=True) with MicrophoneStream(RATE, CHUNK) as stream: audio_generator = stream.generator() requests = (types.StreamingRecognizeRequest(audio_content=content) for content in audio_generator) responses = client.streaming_recognize(streaming_config, requests) # Now, put the transcription responses to use. listen_print_loop(responses) if __name__ == '__main__': main()