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adding infinite streaming sample to samples folder, unlike indefinite…
…… [(#2161)](GoogleCloudPlatform/python-docs-samples#2161) * adding infinite streaming sample to samples folder, unlike indefinite streaming, this sample uses result_end_time to calculate unfinalized audio, and to resend it to maintain context across into the next streaming request * fixed some travis ci lint issues * fixed some travis ci lint issues * fixed some travis ci lint issues * fixed some travis ci lint issues
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packages/google-cloud-speech/samples/microphone/transcribe_streaming_infinite.py
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#!/usr/bin/env python | ||
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# Copyright 2019 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 dependencies `pyaudio` and `termcolor`. | ||
To install using pip: | ||
pip install pyaudio | ||
pip install termcolor | ||
Example usage: | ||
python transcribe_streaming_infinite.py | ||
""" | ||
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# [START speech_transcribe_infinite_streaming] | ||
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import time | ||
import re | ||
import sys | ||
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# uses result_end_time currently only avaialble in v1p1beta, will be in v1 soon | ||
from google.cloud import speech_v1p1beta1 as speech | ||
import pyaudio | ||
from six.moves import queue | ||
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# Audio recording parameters | ||
STREAMING_LIMIT = 10000 | ||
SAMPLE_RATE = 16000 | ||
CHUNK_SIZE = int(SAMPLE_RATE / 10) # 100ms | ||
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RED = '\033[0;31m' | ||
GREEN = '\033[0;32m' | ||
YELLOW = '\033[0;33m' | ||
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def get_current_time(): | ||
"""Return Current Time in MS.""" | ||
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return int(round(time.time() * 1000)) | ||
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class ResumableMicrophoneStream: | ||
"""Opens a recording stream as a generator yielding the audio chunks.""" | ||
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def __init__(self, rate, chunk_size): | ||
self._rate = rate | ||
self.chunk_size = chunk_size | ||
self._num_channels = 1 | ||
self._buff = queue.Queue() | ||
self.closed = True | ||
self.start_time = get_current_time() | ||
self.restart_counter = 0 | ||
self.audio_input = [] | ||
self.last_audio_input = [] | ||
self.result_end_time = 0 | ||
self.is_final_end_time = 0 | ||
self.final_request_end_time = 0 | ||
self.bridging_offset = 0 | ||
self.last_transcript_was_final = False | ||
self.new_stream = True | ||
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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def __enter__(self): | ||
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self.closed = False | ||
return self | ||
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def __exit__(self, type, value, traceback): | ||
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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.""" | ||
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self._buff.put(in_data) | ||
return None, pyaudio.paContinue | ||
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def generator(self): | ||
"""Stream Audio from microphone to API and to local buffer""" | ||
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while not self.closed: | ||
data = [] | ||
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if self.new_stream and self.last_audio_input: | ||
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chunk_time = STREAMING_LIMIT / len(self.last_audio_input) | ||
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if chunk_time != 0: | ||
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if self.bridging_offset < 0: | ||
self.bridging_offset = 0 | ||
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if self.bridging_offset > self.final_request_end_time: | ||
self.bridging_offset = self.final_request_end_time | ||
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chunks_from_ms = round((self.final_request_end_time - | ||
self.bridging_offset) / chunk_time) | ||
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self.bridging_offset = (round(( | ||
len(self.last_audio_input) - chunks_from_ms) | ||
* chunk_time)) | ||
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for i in range(chunks_from_ms, len(self.last_audio_input)): | ||
data.append(self.last_audio_input[i]) | ||
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self.new_stream = False | ||
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# 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() | ||
self.audio_input.append(chunk) | ||
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if chunk is None: | ||
return | ||
data.append(chunk) | ||
# Now consume whatever other data's still buffered. | ||
while True: | ||
try: | ||
chunk = self._buff.get(block=False) | ||
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if chunk is None: | ||
return | ||
data.append(chunk) | ||
self.audio_input.append(chunk) | ||
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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. | ||
""" | ||
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for response in responses: | ||
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if get_current_time() - stream.start_time > STREAMING_LIMIT: | ||
stream.start_time = get_current_time() | ||
break | ||
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if not response.results: | ||
continue | ||
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result = response.results[0] | ||
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if not result.alternatives: | ||
continue | ||
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transcript = result.alternatives[0].transcript | ||
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result_seconds = 0 | ||
result_nanos = 0 | ||
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if result.result_end_time.seconds: | ||
result_seconds = result.result_end_time.seconds | ||
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if result.result_end_time.nanos: | ||
result_nanos = result.result_end_time.nanos | ||
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stream.result_end_time = int((result_seconds * 1000) | ||
+ (result_nanos / 1000000)) | ||
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corrected_time = (stream.result_end_time - stream.bridging_offset | ||
+ (STREAMING_LIMIT * stream.restart_counter)) | ||
# Display interim results, but with a carriage return at the end of the | ||
# line, so subsequent lines will overwrite them. | ||
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if result.is_final: | ||
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sys.stdout.write(GREEN) | ||
sys.stdout.write('\033[K') | ||
sys.stdout.write(str(corrected_time) + ': ' + transcript + '\n') | ||
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stream.is_final_end_time = stream.result_end_time | ||
stream.last_transcript_was_final = True | ||
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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): | ||
sys.stdout.write(YELLOW) | ||
sys.stdout.write('Exiting...\n') | ||
stream.closed = True | ||
break | ||
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else: | ||
sys.stdout.write(RED) | ||
sys.stdout.write('\033[K') | ||
sys.stdout.write(str(corrected_time) + ': ' + transcript + '\r') | ||
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stream.last_transcript_was_final = False | ||
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def main(): | ||
"""start bidirectional streaming from microphone input to speech API""" | ||
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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) | ||
streaming_config = speech.types.StreamingRecognitionConfig( | ||
config=config, | ||
interim_results=True) | ||
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mic_manager = ResumableMicrophoneStream(SAMPLE_RATE, CHUNK_SIZE) | ||
print(mic_manager.chunk_size) | ||
sys.stdout.write(YELLOW) | ||
sys.stdout.write('\nListening, say "Quit" or "Exit" to stop.\n\n') | ||
sys.stdout.write('End (ms) Transcript Results/Status\n') | ||
sys.stdout.write('=====================================================\n') | ||
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with mic_manager as stream: | ||
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while not stream.closed: | ||
sys.stdout.write(YELLOW) | ||
sys.stdout.write('\n' + str( | ||
STREAMING_LIMIT * stream.restart_counter) + ': NEW REQUEST\n') | ||
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stream.audio_input = [] | ||
audio_generator = stream.generator() | ||
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requests = (speech.types.StreamingRecognizeRequest( | ||
audio_content=content)for content in audio_generator) | ||
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responses = client.streaming_recognize(streaming_config, | ||
requests) | ||
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# Now, put the transcription responses to use. | ||
listen_print_loop(responses, stream) | ||
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if stream.result_end_time > 0: | ||
stream.final_request_end_time = stream.is_final_end_time | ||
stream.result_end_time = 0 | ||
stream.last_audio_input = [] | ||
stream.last_audio_input = stream.audio_input | ||
stream.audio_input = [] | ||
stream.restart_counter = stream.restart_counter + 1 | ||
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if not stream.last_transcript_was_final: | ||
sys.stdout.write('\n') | ||
stream.new_stream = True | ||
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if __name__ == '__main__': | ||
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main() | ||
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# [END speech_transcribe_infinite_streaming] |