我想在Python（2.x）中创建32位float WAV文件。虽然“标准” WAV文件通常使用int，但许多专业音频应用程序将音频数据处理（并保存）为浮点型。标准wave模块无法执行此操作：http://bugs.python.org/issue16525没有人在不使用补丁模块的情况下实现了这一目标吗？ tnx寻求帮助。
这听起来很有趣（请参见我的手柄），所以我锤了一些东西。也许您可以使用它。如果您的Python脚本生成的单音波形的数值介于[-1.0 .. 1.0]之间，请通过sample_array发送该波形并指定sample_rate（例如44100或48000）。这将返回一个数组，您可以将其作为.wav文件写入磁盘。
我在Windows Media Player，Apple QuickTime Player和VLC（全部在Windows 7中）中测试了所得的.wav输出。他们都玩过。
def float32_wav_file(sample_array, sample_rate):
byte_count = (len(sample_array)) * 4 # 32-bit floats
wav_file = ""
# write the header
wav_file += struct.pack('<ccccIccccccccIHHIIHH',
'R', 'I', 'F', 'F',
byte_count + 0x2c - 8, # header size
'W', 'A', 'V', 'E', 'f', 'm', 't', ' ',
0x10, # size of 'fmt ' header
3, # format 3 = floating-point PCM
1, # channels
sample_rate, # samples / second
sample_rate * 4, # bytes / second
4, # block alignment
32) # bits / sample
wav_file += struct.pack('<ccccI',
'd', 'a', 't', 'a', byte_count)
for sample in sample_array:
wav_file += struct.pack("<f", sample)
return wav_file
[这是我的贡献...包括任意字长和任意数量的频道。我已经自由更改了float32_wav_file，以包含一个保存用于测试的文件。请注意，文件结构的多通道数据部分是交错的。我敢肯定，该循环可能会被极大地Python化。
# see http://stackoverflow.com/questions/15576798/create-32bit-float-wav-file-in-python
# see... http://blog.theroyweb.com/extracting-wav-file-header-information-using-a-python-script
import struct
def float32_wav_file(file_name, sample_array, sample_rate):
(M,N)=sample_array.shape
#print "len sample_array=(%d,%d)" % (M,N)
byte_count = M * N * 4 # (len(sample_array)) * 4 # 32-bit floats
wav_file = ""
# write the header
wav_file += struct.pack('<ccccIccccccccIHHIIHH',
'R', 'I', 'F', 'F',
byte_count + 0x2c - 8, # header size
'W', 'A', 'V', 'E', 'f', 'm', 't', ' ',
0x10, # size of 'fmt ' header
3, # format 3 = floating-point PCM
M, # channels
sample_rate, # samples / second
sample_rate * 4, # bytes / second
4, # block alignment
32) # bits / sample
wav_file += struct.pack('<ccccI',
'd', 'a', 't', 'a', byte_count)
print "packing..."
for j in range(0,N):
for k in range(0,M):
wav_file += struct.pack("<f", sample_array[k,j])
print "saving..."
fi=open(file_name,'wb')
for value in wav_file:
fi.write(value)
fi.close()
return wav_file
import numpy as np
def wav_file_read(filename):
fi=open(filename,'rb')
data=fi.read()
fi.close()
A, B, C, D =struct.unpack('4c', data[0:4]) # 'RIFF'
ChunkSize =struct.unpack('<l', data[4:8])[0] #4+(8+SubChunk1Size)+8+SubChunk2Size)
A, B, C, D =struct.unpack('4c', data[8:12]) # 'WAVE'
A, B, C, D =struct.unpack('4c', data[12:16]) # 'fmt '
Subchunk1Size =struct.unpack('<l', data[16:20])[0] # LITTLE ENDIAN, long, 16
AudioFormat =struct.unpack('<h', data[20:22])[0] # LITTLE ENDIAN, short, 1
NumChannels =struct.unpack('<h', data[22:24])[0] # LITTLE ENDIAN, short, Mono = 1, Stereo = 2
SampleRate =struct.unpack('<l', data[24:28])[0] # LITTLE ENDIAN, long, sample rate in samples per second
ByteRate =struct.unpack('<l', data[28:32])[0] # self.SampleRate * self.NumChannels * self.BitsPerSample/8)) # (ByteRate) LITTLE ENDIAN, long
BlockAlign =struct.unpack('<h', data[32:34])[0] # self.NumChannels * self.BitsPerSample/8)) # (BlockAlign) LITTLE ENDIAN, short
BitsPerSample =struct.unpack('<h', data[34:36])[0] # LITTLE ENDIAN, short
A, B, C, D =struct.unpack('4c', data[36:40]) # BIG ENDIAN, char*4
SubChunk2Size =struct.unpack('<l', data[40:44])[0] # LITTLE ENDIAN, long
waveData=data[44:]
(M,N)=(len(waveData),len(waveData[0]))
print("ChunkSize =%d\nSubchunk1Size =%d\nAudioFormat =%d\nNumChannels =%d\nSampleRate =%d\nByteRate =%d\nBlockAlign =%d\nBitsPerSample =%d\nA:%c, B:%c, C:%c, D:%c\nSubChunk2Size =%d" %
(ChunkSize ,
Subchunk1Size,
AudioFormat ,
NumChannels ,
SampleRate ,
ByteRate ,
BlockAlign ,
BitsPerSample ,
A, B, C, D ,
SubChunk2Size ))
if BitsPerSample==8:
print "Unpacking 8 bits on len(waveData)=%d" % len(waveData)
d=np.fromstring(waveData,np.uint8)
floatdata=d.astype(np.float64)/np.float(127)
elif BitsPerSample==16:
print "Unpacking 16 bits on len(waveData)=%d" % len(waveData)
d=np.zeros(SubChunk2Size/2, dtype=np.int16)
j=0
for k in range(0, SubChunk2Size, 2):
d[j]=struct.unpack('<h',waveData[k:k+2])[0]
j=j+1
floatdata=d.astype(np.float64)/np.float(32767)
elif BitsPerSample==24:
print "Unpacking 24 bits on len(waveData)=%d" % len(waveData)
d=np.zeros(SubChunk2Size/3, dtype=np.int32)
j=0
for k in range(0, SubChunk2Size, 3):
d[j]=struct.unpack('<l',struct.pack('c',waveData[k])+waveData[k:k+3])[0]
j=j+1
floatdata=d.astype(np.float64)/np.float(2147483647)
else: # anything else will be considered 32 bits
print "Unpacking 32 bits on len(waveData)=%d" % len(waveData)
d=np.fromstring(waveData,np.int32)
floatdata=d.astype(np.float64)/np.float(2147483647)
v=floatdata[0::NumChannels]
for i in range(1,NumChannels):
v=np.vstack((v,floatdata[i::NumChannels]))
#return (np.vstack((floatdata[0::2],floatdata[1::2])), SampleRate, NumChannels, BitsPerSample)
return (v, SampleRate, NumChannels, BitsPerSample)
if __name__ == "__main__":
(array,SampleRate,NumChannels,BitsPerSample)=wav_file_read("my_input_file.wav")
wavefile=float32_wav_file("test_file.wav",array,SampleRate)
我尝试测试P Moran的代码，但绝对无法正常工作，例如'len（int（3））？'。抱歉，我认为它没有经过测试。但是不要担心，现代python有办法！
import numpy as np
songtime=np.arange(0,100,60/44100.,dtype=float)
coswav=np.cos(songtime)
sinewav=np.sin(songtime)
np.column_stack((coswav,sinwav))
songwav=np.column_stack((coswav,sinwav)).astype('float32')
import scipy.io.wavfile as scipy_io_wavfile
scipy_io_wavfile.write("5secof60Hz.wav",44100,songwav)
https://docs.python.org/3.7/library/wave.htmlhttps://docs.scipy.org/doc/scipy/reference/generated/scipy.io.wavfile.write.html