我是Python的新手，正在尝试如何在后台播放声音的情况下，从文件中实时转录音频语音。在
更新：@petezurich Sorry for the bad question. Currently, I can hear the
audio playing in the background. However, I am having trouble getting
Sphinx to transcribe the audio. Is there something wrong with the way
I am passing the audio to Sphinx?
It's constantly outputting "Sphinx error" message.
我正在使用PocketSpinx和Uberi/语音识别库。在
到目前为止，我总结了一下：
#!/usr/bin/env python
# recognitions.py : Transcribe Test from an Audio File
import os
import sys
import time
import wave
import pyaudio
import speech_recognition as sr
import threading
try:
import pocketsphinx
except:
print("PocketSphinx is not installed.")
# import audio file within script folder
from os import path
audio_file = path.join(os.path.abspath(os.path.dirname(sys.argv[0])), "samples/OSR_us_000_0061_8k.wav")
print("Transcribing... " + audio_file)
wf = wave.open(audio_file, 'rb')
# set PyAudio instance
pa = pyaudio.PyAudio()
# set recognizer instance (unmodified)
r = sr.Recognizer()
stream_buffer = bytes()
stream_counter = 0
audio_sampling_rate = 48000
def main_recognize(stream):
global audio_sampling_rate
# Create a new AudioData instance, which represents "mono" audio data
audio_data = sr.AudioData(stream, audio_sampling_rate, 2)
# recognize using CMU Sphinx (en-US only)
try:
print("Sphinx: " + r.recognize_sphinx(audio_data, language="en-US"))
except sr.UnknownValueError:
print("Sphinx error")
except sr.RequestError as e:
print("Sphinx error; {0}".format(e))
def stream_audio(data):
global stream_buffer
global stream_counter
buffer_set_size = 200
if stream_counter < buffer_set_size:
# force 'data' to BYTES to allow concat
data = bytes()
stream_buffer += data
stream_counter += 1
else:
threading.Thread(target=main_recognize, args=(stream_buffer,)).start()
# reset
stream_buffer = bytes()
stream_counter = 0
# define callback
def callback(in_data, frame_count, time_info, status):
data = wf.readframes(frame_count)
stream_audio(in_data)
return (data, pyaudio.paContinue)
# open audio stream
stream = pa.open(format=pa.get_format_from_width(wf.getsampwidth()),
channels=wf.getnchannels(),
rate=wf.getframerate(),
output=True,
stream_callback=callback)
# start the stream
stream.start_stream()
# wait for stream to finish
while stream.is_active():
time.sleep(0.1)
# stop stream
stream.stop_stream()
stream.close()
wf.close()
# close PyAudio
pa.terminate()
