在Python中检测并loggingaudio
我需要捕获audio剪辑作为WAV文件，然后我可以传递到另一个Python的处理。 问题是，我需要确定什么时候有audio存在，然后logging下来，当它静音时停止，然后将该文件传递给处理模块。
我认为应该可以用波模块来检测何时有纯粹的静音，并且一旦检测到静音以外的东西就开始录音，然后当线路再次静音时停止录音。
只是不能让我的头，周围的任何人都可以让我开始一个基本的例子。
作为Nick Fortescue的回答，下面是一个更完整的例子，说明如何从麦克风录音并处理结果数据：
from sys import byteorder from array import array from struct import pack import pyaudio import wave THRESHOLD = 500 CHUNK_SIZE = 1024 FORMAT = pyaudio.paInt16 RATE = 44100 def is_silent(snd_data): "Returns 'True' if below the 'silent' threshold" return max(snd_data) < THRESHOLD def normalize(snd_data): "Average the volume out" MAXIMUM = 16384 times = float(MAXIMUM)/max(abs(i) for i in snd_data) r = array('h') for i in snd_data: r.append(int(i*times)) return r def trim(snd_data): "Trim the blank spots at the start and end" def _trim(snd_data): snd_started = False r = array('h') for i in snd_data: if not snd_started and abs(i)>THRESHOLD: snd_started = True r.append(i) elif snd_started: r.append(i) return r # Trim to the left snd_data = _trim(snd_data) # Trim to the right snd_data.reverse() snd_data = _trim(snd_data) snd_data.reverse() return snd_data def add_silence(snd_data, seconds): "Add silence to the start and end of 'snd_data' of length 'seconds' (float)" r = array('h', [0 for i in xrange(int(seconds*RATE))]) r.extend(snd_data) r.extend([0 for i in xrange(int(seconds*RATE))]) return r def record(): """ Record a word or words from the microphone and return the data as an array of signed shorts. Normalizes the audio, trims silence from the start and end, and pads with 0.5 seconds of blank sound to make sure VLC et al can play it without getting chopped off. """ p = pyaudio.PyAudio() stream = p.open(format=FORMAT, channels=1, rate=RATE, input=True, output=True, frames_per_buffer=CHUNK_SIZE) num_silent = 0 snd_started = False r = array('h') while 1: # little endian, signed short snd_data = array('h', stream.read(CHUNK_SIZE)) if byteorder == 'big': snd_data.byteswap() r.extend(snd_data) silent = is_silent(snd_data) if silent and snd_started: num_silent += 1 elif not silent and not snd_started: snd_started = True if snd_started and num_silent > 30: break sample_width = p.get_sample_size(FORMAT) stream.stop_stream() stream.close() p.terminate() r = normalize(r) r = trim(r) r = add_silence(r, 0.5) return sample_width, r def record_to_file(path): "Records from the microphone and outputs the resulting data to 'path'" sample_width, data = record() data = pack('<' + ('h'*len(data)), *data) wf = wave.open(path, 'wb') wf.setnchannels(1) wf.setsampwidth(sample_width) wf.setframerate(RATE) wf.writeframes(data) wf.close() if __name__ == '__main__': print("please speak a word into the microphone") record_to_file('demo.wav') print("done - result written to demo.wav") 我相信WAVE模块不支持录制，只是处理现有的文件。 你可能想看看PyAudio的实际logging。 WAV是世界上最简单的文件格式。 在pa16中，你只需要一个有符号的整数来代表一个关卡，而接近于0就更安静了。 我不记得如果WAV文件是高字节的第一个或低字节，但是这样的事情应该工作（对不起，我不是一个真正的Python程序员：
from array import array # you'll probably want to experiment on threshold # depends how noisy the signal threshold = 10 max_value = 0 as_ints = array('h', data) max_value = max(as_ints) if max_value > threshold: # not silence
用于logging的PyAudio代码仅供参考：
import pyaudio import sys chunk = 1024 FORMAT = pyaudio.paInt16 CHANNELS = 1 RATE = 44100 RECORD_SECONDS = 5 p = pyaudio.PyAudio() stream = p.open(format=FORMAT, channels=CHANNELS, rate=RATE, input=True, output=True, frames_per_buffer=chunk) print "* recording" for i in range(0, 44100 / chunk * RECORD_SECONDS): data = stream.read(chunk) # check for silence here by comparing the level with 0 (or some threshold) for # the contents of data. # then write data or not to a file print "* done" stream.stop_stream() stream.close() p.terminate()
感谢cryo改进版本，我基于我的testing代码如下：
#Instead of adding silence at start and end of recording (values=0) I add the original audio . This makes audio sound more natural as volume is >0. See trim() #I also fixed issue with the previous code - accumulated silence counter needs to be cleared once recording is resumed. from array import array from struct import pack from sys import byteorder import copy import pyaudio import wave THRESHOLD = 500 # audio levels not normalised. CHUNK_SIZE = 1024 SILENT_CHUNKS = 3 * 44100 / 1024 # about 3sec FORMAT = pyaudio.paInt16 FRAME_MAX_VALUE = 2 ** 15 - 1 NORMALIZE_MINUS_ONE_dB = 10 ** (-1.0 / 20) RATE = 44100 CHANNELS = 1 TRIM_APPEND = RATE / 4 def is_silent(data_chunk): """Returns 'True' if below the 'silent' threshold""" return max(data_chunk) < THRESHOLD def normalize(data_all): """Amplify the volume out to max -1dB""" # MAXIMUM = 16384 normalize_factor = (float(NORMALIZE_MINUS_ONE_dB * FRAME_MAX_VALUE) / max(abs(i) for i in data_all)) r = array('h') for i in data_all: r.append(int(i * normalize_factor)) return r def trim(data_all): _from = 0 _to = len(data_all) - 1 for i, b in enumerate(data_all): if abs(b) > THRESHOLD: _from = max(0, i - TRIM_APPEND) break for i, b in enumerate(reversed(data_all)): if abs(b) > THRESHOLD: _to = min(len(data_all) - 1, len(data_all) - 1 - i + TRIM_APPEND) break return copy.deepcopy(data_all[_from:(_to + 1)]) def record(): """Record a word or words from the microphone and return the data as an array of signed shorts.""" p = pyaudio.PyAudio() stream = p.open(format=FORMAT, channels=CHANNELS, rate=RATE, input=True, output=True, frames_per_buffer=CHUNK_SIZE) silent_chunks = 0 audio_started = False data_all = array('h') while True: # little endian, signed short data_chunk = array('h', stream.read(CHUNK_SIZE)) if byteorder == 'big': data_chunk.byteswap() data_all.extend(data_chunk) silent = is_silent(data_chunk) if audio_started: if silent: silent_chunks += 1 if silent_chunks > SILENT_CHUNKS: break else: silent_chunks = 0 elif not silent: audio_started = True sample_width = p.get_sample_size(FORMAT) stream.stop_stream() stream.close() p.terminate() data_all = trim(data_all) # we trim before normalize as threshhold applies to un-normalized wave (as well as is_silent() function) data_all = normalize(data_all) return sample_width, data_all def record_to_file(path): "Records from the microphone and outputs the resulting data to 'path'" sample_width, data = record() data = pack('<' + ('h' * len(data)), *data) wave_file = wave.open(path, 'wb') wave_file.setnchannels(CHANNELS) wave_file.setsampwidth(sample_width) wave_file.setframerate(RATE) wave_file.writeframes(data) wave_file.close() if __name__ == '__main__': print("Wait in silence to begin recording; wait in silence to terminate") record_to_file('demo.wav') print("done - result written to demo.wav")
pyaudio网站有很多很简短的例子： http ://people.csail.mit.edu/hubert/pyaudio/
import pyaudio import wave from array import array FORMAT=pyaudio.paInt16 CHANNELS=2 RATE=44100 CHUNK=1024 RECORD_SECONDS=15 FILE_NAME="RECORDING.wav" audio=pyaudio.PyAudio() #instantiate the pyaudio #recording prerequisites stream=audio.open(format=FORMAT,channels=CHANNELS, rate=RATE, input=True, frames_per_buffer=CHUNK) #starting recording frames=[] for i in range(0,int(RATE/CHUNK*RECORD_SECONDS)): data=stream.read(CHUNK) data_chunk=array('h',data) vol=max(data_chunk) if(vol>=500): print("something said") frames.append(data) else: print("nothing") print("\n") #end of recording stream.stop_stream() stream.close() audio.terminate() #writing to file wavfile=wave.open(FILE_NAME,'wb') wavfile.setnchannels(CHANNELS) wavfile.setsampwidth(audio.get_sample_size(FORMAT)) wavfile.setframerate(RATE) wavfile.writeframes(b''.join(frames))#append frames recorded to file wavfile.close()
我认为这样会有帮助。它是一个简单的脚本，用来检查是否存在静音。如果检测到静音，则不会录制，否则会录制。
你也可以看看csounds 。 它有几个API，包括Python。 它可能能够与AD接口交互并收集声音样本。
