Real-time Speech Recognition
This Python code implements real-time speech recognition using Google Cloud Speech-to-Text API and the PyAudio library. It captures audio from the microphone, streams it to the Speech-to-Text API, and prints the transcribed text. The MicrophoneStream class handles audio input, and the main function sets up the speech recognition client and processes the audio stream.
import os
import argparse
import io
import sys
import time
from google.cloud import speech
import pyaudio
from six.moves import queue
# 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 audio interface using PyAudio
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/z726ff
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
self._buff = queue.Queue()
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, record_seconds):
start_time = time.time()
while not self.closed and time.time() - start_time < record_seconds:
# 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)
def close(self):
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_stream.close()
self._audio_interface.terminate()
def __enter__(self):
return self
def __exit__(self, type, value, traceback):
self.close()
def main(record_seconds=10, language_code='en-US'):
# 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 = speech.RecognitionConfig(
encoding=speech.RecognitionConfig.AudioEncoding.LINEAR16,
sample_rate_hertz=RATE,
language_code=language_code,
model="latest_long",
)
streaming_config = speech.StreamingRecognitionConfig(
config=config,
interim_results=True)
with MicrophoneStream(RATE, CHUNK) as stream:
audio_generator = stream.generator(record_seconds)
requests = (speech.StreamingRecognizeRequest(audio_content=content)
for content in audio_generator)
responses = client.streaming_recognize(streaming_config, requests)
# Now, put the transcription responses to use.
transcript = ""
for response in responses:
print(response)
# Once the transcription is done, print the result.
for result in response.results:
if result.is_final:
alternative = result.alternatives[0]
transcript += alternative.transcript + " "
print(u'Transcript: {}'.format(transcript))
if __name__ == '__main__':
parser = argparse.ArgumentParser(description="Real-time speech recognition with adjustable duration.")
parser.add_argument('--duration', type=int, default=10, help="Duration of recording in seconds.")
parser.add_argument('--language_code', type=str, default='en-US', help="Language code for transcription.")
args = parser.parse_args()
print("Please speak...")
main(record_seconds=args.duration, language_code=args.language_code)