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| 1 | +#!/usr/bin/python |
| 2 | + |
| 3 | +# Copyright (C) 2016 Google Inc. |
| 4 | +# |
| 5 | +# Licensed under the Apache License, Version 2.0 (the "License"); |
| 6 | +# you may not use this file except in compliance with the License. |
| 7 | +# You may obtain a copy of the License at |
| 8 | +# |
| 9 | +# http://www.apache.org/licenses/LICENSE-2.0 |
| 10 | +# |
| 11 | +# Unless required by applicable law or agreed to in writing, software |
| 12 | +# distributed under the License is distributed on an "AS IS" BASIS, |
| 13 | +# WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied. |
| 14 | +# See the License for the specific language governing permissions and |
| 15 | +# limitations under the License. |
| 16 | + |
| 17 | +"""Sample that streams audio to the Google Cloud Speech API via GRPC. |
| 18 | +
|
| 19 | +This sample expands on transcribe_streaming.py to work around the 1-minute |
| 20 | +limit on streaming requests. It does this by transcribing normally until |
| 21 | +WRAP_IT_UP_SECS seconds before the 1-minute limit. At that point, it waits for |
| 22 | +the end of an utterance and once it hears it, it closes the current stream and |
| 23 | +opens a new one. It also keeps a buffer of audio around while this is |
| 24 | +happening, that it sends to the new stream in its initial request, to minimize |
| 25 | +losing any speech that occurs while this happens. |
| 26 | +
|
| 27 | +Note that you could do this a little more simply by simply re-starting the |
| 28 | +stream after every utterance, though this increases the possibility of audio |
| 29 | +being missed between streams. For learning purposes (and robustness), the more |
| 30 | +complex implementation is shown here. |
| 31 | +""" |
| 32 | + |
| 33 | +from __future__ import division |
| 34 | + |
| 35 | +import argparse |
| 36 | +import collections |
| 37 | +import contextlib |
| 38 | +import functools |
| 39 | +import logging |
| 40 | +import re |
| 41 | +import signal |
| 42 | +import sys |
| 43 | +import time |
| 44 | + |
| 45 | +import google.auth |
| 46 | +import google.auth.transport.grpc |
| 47 | +import google.auth.transport.requests |
| 48 | +from google.cloud.proto.speech.v1beta1 import cloud_speech_pb2 |
| 49 | +from google.rpc import code_pb2 |
| 50 | +import grpc |
| 51 | +import pyaudio |
| 52 | +from six.moves import queue |
| 53 | + |
| 54 | +# Seconds you have to wrap up your utterance |
| 55 | +WRAP_IT_UP_SECS = 15 |
| 56 | +SECS_OVERLAP = 1 |
| 57 | + |
| 58 | +# Audio recording parameters |
| 59 | +RATE = 16000 |
| 60 | +CHUNK = int(RATE / 10) # 100ms |
| 61 | + |
| 62 | +# The Speech API has a streaming limit of 60 seconds of audio*, so keep the |
| 63 | +# connection alive for that long, plus some more to give the API time to figure |
| 64 | +# out the transcription. |
| 65 | +# * https://g.co/cloud/speech/limits#content |
| 66 | +DEADLINE_SECS = 60 * 3 + 5 |
| 67 | +SPEECH_SCOPE = 'https://www.googleapis.com/auth/cloud-platform' |
| 68 | + |
| 69 | + |
| 70 | +def make_channel(host): |
| 71 | + """Creates a secure channel with auth credentials from the environment.""" |
| 72 | + # Grab application default credentials from the environment |
| 73 | + credentials, _ = google.auth.default(scopes=[SPEECH_SCOPE]) |
| 74 | + |
| 75 | + # Create a secure channel using the credentials. |
| 76 | + http_request = google.auth.transport.requests.Request() |
| 77 | + |
| 78 | + return google.auth.transport.grpc.secure_authorized_channel( |
| 79 | + credentials, http_request, host) |
| 80 | + |
| 81 | + |
| 82 | +def _audio_data_generator(buff, overlap_buffer): |
| 83 | + """A generator that yields all available data in the given buffer. |
| 84 | +
|
| 85 | + Args: |
| 86 | + buff (Queue): A Queue where each element is a chunk of data. |
| 87 | + overlap_buffer (deque): a ring buffer for storing trailing data chunks |
| 88 | + Yields: |
| 89 | + bytes: A chunk of data that is the aggregate of all chunks of data in |
| 90 | + `buff`. The function will block until at least one data chunk is |
| 91 | + available. |
| 92 | + """ |
| 93 | + if overlap_buffer: |
| 94 | + yield b''.join(overlap_buffer) |
| 95 | + overlap_buffer.clear() |
| 96 | + |
| 97 | + while True: |
| 98 | + # Use a blocking get() to ensure there's at least one chunk of data. |
| 99 | + data = [buff.get()] |
| 100 | + |
| 101 | + # Now consume whatever other data's still buffered. |
| 102 | + while True: |
| 103 | + try: |
| 104 | + data.append(buff.get(block=False)) |
| 105 | + except queue.Empty: |
| 106 | + break |
| 107 | + |
| 108 | + # `None` in the buffer signals that we should stop generating. Put the |
| 109 | + # data back into the buffer for the next generator. |
| 110 | + if None in data: |
| 111 | + data.remove(None) |
| 112 | + if data: |
| 113 | + buff.put(b''.join(data)) |
| 114 | + break |
| 115 | + else: |
| 116 | + overlap_buffer.extend(data) |
| 117 | + |
| 118 | + yield b''.join(data) |
| 119 | + |
| 120 | + |
| 121 | +def _fill_buffer(buff, in_data, frame_count, time_info, status_flags): |
| 122 | + """Continuously collect data from the audio stream, into the buffer.""" |
| 123 | + buff.put(in_data) |
| 124 | + return None, pyaudio.paContinue |
| 125 | + |
| 126 | + |
| 127 | +# [START audio_stream] |
| 128 | +@contextlib.contextmanager |
| 129 | +def record_audio(rate, chunk): |
| 130 | + """Opens a recording stream in a context manager.""" |
| 131 | + # Create a thread-safe buffer of audio data |
| 132 | + buff = queue.Queue() |
| 133 | + |
| 134 | + audio_interface = pyaudio.PyAudio() |
| 135 | + audio_stream = audio_interface.open( |
| 136 | + format=pyaudio.paInt16, |
| 137 | + # The API currently only supports 1-channel (mono) audio |
| 138 | + # https://goo.gl/z757pE |
| 139 | + channels=1, rate=rate, |
| 140 | + input=True, frames_per_buffer=chunk, |
| 141 | + # Run the audio stream asynchronously to fill the buffer object. |
| 142 | + # This is necessary so that the input device's buffer doesn't overflow |
| 143 | + # while the calling thread makes network requests, etc. |
| 144 | + stream_callback=functools.partial(_fill_buffer, buff), |
| 145 | + ) |
| 146 | + |
| 147 | + yield buff |
| 148 | + |
| 149 | + audio_stream.stop_stream() |
| 150 | + audio_stream.close() |
| 151 | + # Signal the _audio_data_generator to finish |
| 152 | + buff.put(None) |
| 153 | + audio_interface.terminate() |
| 154 | +# [END audio_stream] |
| 155 | + |
| 156 | + |
| 157 | +def request_stream(data_stream, rate, interim_results=True): |
| 158 | + """Yields `StreamingRecognizeRequest`s constructed from a recording audio |
| 159 | + stream. |
| 160 | +
|
| 161 | + Args: |
| 162 | + data_stream (generator): The raw audio data to send. |
| 163 | + rate (int): The sampling rate in hertz. |
| 164 | + interim_results (boolean): Whether to return intermediate results, |
| 165 | + before the transcription is finalized. |
| 166 | + """ |
| 167 | + # The initial request must contain metadata about the stream, so the |
| 168 | + # server knows how to interpret it. |
| 169 | + recognition_config = cloud_speech_pb2.RecognitionConfig( |
| 170 | + # There are a bunch of config options you can specify. See |
| 171 | + # https://goo.gl/KPZn97 for the full list. |
| 172 | + encoding='LINEAR16', # raw 16-bit signed LE samples |
| 173 | + sample_rate=rate, # the rate in hertz |
| 174 | + # See http://g.co/cloud/speech/docs/languages |
| 175 | + # for a list of supported languages. |
| 176 | + language_code='en-US', # a BCP-47 language tag |
| 177 | + ) |
| 178 | + streaming_config = cloud_speech_pb2.StreamingRecognitionConfig( |
| 179 | + interim_results=interim_results, |
| 180 | + config=recognition_config, |
| 181 | + ) |
| 182 | + |
| 183 | + yield cloud_speech_pb2.StreamingRecognizeRequest( |
| 184 | + streaming_config=streaming_config) |
| 185 | + |
| 186 | + for data in data_stream: |
| 187 | + # Subsequent requests can all just have the content |
| 188 | + yield cloud_speech_pb2.StreamingRecognizeRequest(audio_content=data) |
| 189 | + |
| 190 | + |
| 191 | +def listen_print_loop( |
| 192 | + recognize_stream, wrap_it_up_secs, buff, max_recog_secs=60): |
| 193 | + """Iterates through server responses and prints them. |
| 194 | +
|
| 195 | + The recognize_stream passed is a generator that will block until a response |
| 196 | + is provided by the server. When the transcription response comes, print it. |
| 197 | +
|
| 198 | + In this case, responses are provided for interim results as well. If the |
| 199 | + response is an interim one, print a line feed at the end of it, to allow |
| 200 | + the next result to overwrite it, until the response is a final one. For the |
| 201 | + final one, print a newline to preserve the finalized transcription. |
| 202 | + """ |
| 203 | + # What time should we switch to a new stream? |
| 204 | + time_to_switch = time.time() + max_recog_secs - wrap_it_up_secs |
| 205 | + graceful_exit = False |
| 206 | + num_chars_printed = 0 |
| 207 | + for resp in recognize_stream: |
| 208 | + if resp.error.code != code_pb2.OK: |
| 209 | + raise RuntimeError('Server error: ' + resp.error.message) |
| 210 | + |
| 211 | + if not resp.results: |
| 212 | + if resp.endpointer_type is resp.END_OF_SPEECH and ( |
| 213 | + time.time() > time_to_switch): |
| 214 | + graceful_exit = True |
| 215 | + buff.put(None) |
| 216 | + continue |
| 217 | + |
| 218 | + # Display the top transcription |
| 219 | + result = resp.results[0] |
| 220 | + transcript = result.alternatives[0].transcript |
| 221 | + |
| 222 | + # If the previous result was longer than this one, we need to print |
| 223 | + # some extra spaces to overwrite the previous result |
| 224 | + overwrite_chars = ' ' * max(0, num_chars_printed - len(transcript)) |
| 225 | + |
| 226 | + # Display interim results, but with a carriage return at the end of the |
| 227 | + # line, so subsequent lines will overwrite them. |
| 228 | + if not result.is_final: |
| 229 | + sys.stdout.write(transcript + overwrite_chars + '\r') |
| 230 | + sys.stdout.flush() |
| 231 | + |
| 232 | + num_chars_printed = len(transcript) |
| 233 | + |
| 234 | + else: |
| 235 | + print(transcript + overwrite_chars) |
| 236 | + |
| 237 | + # Exit recognition if any of the transcribed phrases could be |
| 238 | + # one of our keywords. |
| 239 | + if re.search(r'\b(exit|quit)\b', transcript, re.I): |
| 240 | + print('Exiting..') |
| 241 | + recognize_stream.cancel() |
| 242 | + |
| 243 | + elif graceful_exit: |
| 244 | + break |
| 245 | + |
| 246 | + num_chars_printed = 0 |
| 247 | + |
| 248 | + |
| 249 | +def main(): |
| 250 | + service = cloud_speech_pb2.SpeechStub( |
| 251 | + make_channel('speech.googleapis.com')) |
| 252 | + |
| 253 | + # For streaming audio from the microphone, there are three threads. |
| 254 | + # First, a thread that collects audio data as it comes in |
| 255 | + with record_audio(RATE, CHUNK) as buff: |
| 256 | + # Second, a thread that sends requests with that data |
| 257 | + overlap_buffer = collections.deque( |
| 258 | + maxlen=int(SECS_OVERLAP * RATE / CHUNK)) |
| 259 | + requests = request_stream( |
| 260 | + _audio_data_generator(buff, overlap_buffer), RATE) |
| 261 | + # Third, a thread that listens for transcription responses |
| 262 | + recognize_stream = service.StreamingRecognize( |
| 263 | + requests, DEADLINE_SECS) |
| 264 | + |
| 265 | + # Exit things cleanly on interrupt |
| 266 | + signal.signal(signal.SIGINT, lambda *_: recognize_stream.cancel()) |
| 267 | + |
| 268 | + # Now, put the transcription responses to use. |
| 269 | + try: |
| 270 | + while True: |
| 271 | + listen_print_loop(recognize_stream, WRAP_IT_UP_SECS, buff) |
| 272 | + |
| 273 | + # Discard this stream and create a new one. |
| 274 | + # Note: calling .cancel() doesn't immediately raise an RpcError |
| 275 | + # - it only raises when the iterator's next() is requested |
| 276 | + recognize_stream.cancel() |
| 277 | + |
| 278 | + logging.debug('Starting new stream') |
| 279 | + requests = request_stream(_audio_data_generator( |
| 280 | + buff, overlap_buffer), RATE) |
| 281 | + recognize_stream = service.StreamingRecognize( |
| 282 | + requests, DEADLINE_SECS) |
| 283 | + |
| 284 | + except grpc.RpcError: |
| 285 | + # This happens because of the interrupt handler |
| 286 | + pass |
| 287 | + |
| 288 | + |
| 289 | +if __name__ == '__main__': |
| 290 | + parser = argparse.ArgumentParser() |
| 291 | + parser.add_argument( |
| 292 | + '-v', '--verbose', help='increase output verbosity', |
| 293 | + action='store_true') |
| 294 | + args = parser.parse_args() |
| 295 | + if args.verbose: |
| 296 | + logging.basicConfig(level=logging.DEBUG) |
| 297 | + |
| 298 | + main() |
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