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要開始使用Google Speech-to-Text API,首先需要在Python環境中安裝google-cloud-speech
包,并在Google Cloud項目中啟用Speech-to-Text API。
%pip install --upgrade --quiet google-cloud-speech
按照Google Cloud快速入門指南創建項目并啟用API。
使用Google Speech-to-Text API前,需要準備project_id
和file_path
。音頻文件可以是Google Cloud Storage的URI或本地文件路徑。
from google.cloud import speech
from google.cloud.speech import enums
from google.cloud.speech import types
client = speech.SpeechClient()
file_path = 'gs://cloud-samples-data/speech/brooklyn_bridge.raw'
audio = types.RecognitionAudio(uri=file_path)
config = types.RecognitionConfig(
encoding=enums.RecognitionConfig.AudioEncoding.LINEAR16,
sample_rate_hertz=16000,
language_code='en-US',
)
response = client.recognize(config=config, audio=audio)
for result in response.results:
print('Transcript: {}'.format(result.alternatives[0].transcript))
可以通過config
參數自定義識別配置,如選擇不同的語音識別模型和功能。
config = types.RecognitionConfig(
encoding=enums.RecognitionConfig.AudioEncoding.LINEAR16,
sample_rate_hertz=16000,
language_code='en-US',
enable_automatic_punctuation=True,
)
在某些地區訪問Google API可能會不穩定,推薦使用API代理服務提高訪問穩定性。例如,可以使用API代理服務。
Google Speech-to-Text API對單個音頻文件的長度有限制(60秒或10MB)。對于更長的音頻文件,可以將其分割成多個小文件進行處理。
確保config
中的language_code
與音頻文件中的語言一致,以獲得最佳的識別效果。
以下是一個使用Python將語音文件轉換為文本的完整示例。
from google.cloud import speech
client = speech.SpeechClient()
gcs_uri = 'gs://cloud-samples-data/speech/brooklyn_bridge.raw'
audio = speech.RecognitionAudio(uri=gcs_uri)
config = speech.RecognitionConfig(
encoding=speech.RecognitionConfig.AudioEncoding.LINEAR16,
sample_rate_hertz=16000,
language_code='en-US',
)
response = client.recognize(config=config, audio=audio)
for result in response.results:
print('Transcript: {}'.format(result.alternatives[0].transcript))
以下示例展示了如何使用麥克風實時捕捉語音并轉換為文本。
import os
from google.cloud import speech
from google.cloud.speech import enums
from google.cloud.speech import types
import pyaudio
from six.moves import queue
os.environ['GOOGLE_APPLICATION_CREDENTIALS'] = 'your-path-to-credentials.json'
RATE = 16000
CHUNK = int(RATE / 10)
class MicrophoneStream(object):
def __init__(self, rate, chunk):
self._rate = rate
self._chunk = chunk
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,
channels=1,
rate=self._rate,
input=True,
frames_per_buffer=self._chunk,
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
self._buff.put(None)
self._audio_interface.terminate()
def _fill_buffer(self, in_data, frame_count, time_info, status_flags):
self._buff.put(in_data)
return None, pyaudio.paContinue
def generator(self):
while not self.closed:
chunk = self._buff.get()
if chunk is None:
return
data = [chunk]
try:
while True:
chunk = self._buff.get(block=False)
if chunk is None:
return
data.append(chunk)
except queue.Empty:
break
yield b''.join(data)
def listen_print_loop(responses):
num_chars_printed = 0
for response in responses:
if not response.results:
continue
result = response.results[0]
if not result.alternatives:
continue
transcript = result.alternatives[0].transcript
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)
if re.search(r'b(exit|quit)b', transcript, re.I):
print('Exiting..')
break
num_chars_printed = 0
def main():
language_code = 'zh'
client = speech.SpeechClient()
config = speech.RecognitionConfig(
encoding=speech.RecognitionConfig.AudioEncoding.LINEAR16,
sample_rate_hertz=RATE,
language_code=language_code,
)
streaming_config = speech.StreamingRecognitionConfig(
config=config, interim_results=True
)
with MicrophoneStream(RATE, CHUNK) as stream:
audio_generator = stream.generator()
requests = (
speech.StreamingRecognizeRequest(audio_content=content)
for content in audio_generator
)
responses = client.streaming_recognize(streaming_config, requests)
listen_print_loop(responses)
if __name__ == '__main__':
main()
本文詳細介紹了Google語音識別技術的原理、應用場景、安裝設置、使用方法以及常見問題的解決方案。通過實際代碼示例,展示了如何使用Google Speech-to-Text API將音頻文件轉換為文本。希望本文能幫助您快速上手Google語音識別技術,并在實際項目中得到應用。
Google語音識別技術以其準確性和易用性,為開發者提供了強大的工具,推動了智能語音處理技術的發展。通過不斷學習和實踐,我們可以更好地利用這項技術,創造更多價值。
如果您對Google語音識別技術有更多的問題或想法,歡迎在評論區交流討論。