当我尝试将wav文件转换为数据集（csv文件）时，一切似乎都工作正常，当我尝试对其进行FFT转换时，除了将其关闭6倍外，其他一切似乎都正常运行这是我应该担心的事情吗？
此外，我还有一个关于wav文件的问题，因此我在其中找到了数据并从中获取了整数。它们是2字节带符号的短裤，因此我知道它们的极限为-32767-32767，并且为振幅。但是，我对这些如何转换为实际体积感到困惑？
代码
import pyaudio
import wave
import time
import sys
import struct
import csv
# all files and stuffs need to contain their tags ('potatos.wav')
def wavToData(wavName):
stream = wave.open(wavName)
num_channels = stream.getnchannels()
sample_rate = stream.getframerate()
sample_width = stream.getsampwidth()
num_frames = stream.getnframes()
raw_data = stream.readframes( num_frames ) # Returns byte data
stream.close()
total_samples = num_frames * num_channels
if sample_width == 1:
fmt = "%iB" % total_samples # read unsigned chars
elif sample_width == 2:
fmt = "%ih" % total_samples # read signed 2 byte shorts
else:
raise ValueError("Only supports 8 and 16 bit audio formats.")
total_samples = num_frames * num_channels
integer_data = struct.unpack(fmt, raw_data) # this is amplitude
return integer_data, num_frames, sample_rate
def dataToWav(wavName,data):
# IMPORTANT hard coded standard values, refenced from the audiocheck.net files
stream = wave.open(wavName,"wb")
stream.setnchannels(1)
stream.setframerate(44100)
stream.setsampwidth(2)
numFrames = len(data)
stream.setnframes(numFrames)
fmt = "%ih" % numFrames
data = struct.pack(fmt,*data) #unpacks all
stream.writeframes(data)
def wavToCsv(wavName,csvName):
with open(csvName, 'wb') as csvfile: #contain all data in 1 row
# x = time (1 frame)
# y = freq
filewriter = csv.writer(csvfile, delimiter=',',
quotechar='|', quoting=csv.QUOTE_MINIMAL)
#get data
integer_data, num_frames, sample_rate = wavToData(wavName)
#adds headings and other needed data
frameCount = [x+1 for x in range(num_frames)] # x
fullData = [x for x in integer_data] # y
frameCount.insert(0, "Time (increments of %f)"%(1/sample_rate))
fullData.insert(0,"Frequency(dBfs)")
fullData = frameCount+fullData
filewriter.writerow(fullData)
def mergeWavData(wav1,wav2,mergeWav): #generates a brand new wav file
firstData = wavToData(wav1)[0]
secondData = wavToData(wav2)[0]
limitCheck = lambda x,y: ((x+y)> 32767 or (x+y) < -32767)
limitConstrain = lambda x,y: 32767 if (x+y) >32767 else -32767
newData = []
if len(firstData) < len(secondData): #adds the amplitudes of both wav file up to the shortest one
for i in range(len(firstData)):
data = 0
if(limitCheck(firstData[i],secondData[i])): #exceed limit
data = limitConstrain(firstData[i],secondData[i])
else:
data = firstData[i]+secondData[i]
newData.append(data)
else:
for i in range(len(secondData)):
data = 0
if(limitCheck(firstData[i],secondData[i])): #exceed limit
data = limitConstrain(firstData[i],secondData[i])
else:
data = firstData[i]+secondData[i]
if (data > 32767 or data <-32767):
print data
newData.append(data)
dataToWav(mergeWav,newData)
mergeWavData('audiocheck.net_sin_1000Hz_-3dBFS_3s.wav','audiocheck.net_sin_4000Hz_-3dBFS_3s.wav','1000Hz+4000Hz.wav')
wavToCsv('1000Hz+4000Hz.wav','1000Hz+4000Hz.csv') # maeks csv out of wav ayayayay
#include tags (.wav/.csv)
import pyaudio
import wave
import time
import sys
import struct
import csv
# all files and stuffs need to contain their tags ('potatos.wav')
def wavToData(wavName):
stream = wave.open(wavName)
num_channels = stream.getnchannels()
sample_rate = stream.getframerate()
sample_width = stream.getsampwidth()
num_frames = stream.getnframes()
raw_data = stream.readframes( num_frames ) # Returns byte data
stream.close()
total_samples = num_frames * num_channels
if sample_width == 1:
fmt = "%iB" % total_samples # read unsigned chars
elif sample_width == 2:
fmt = "%ih" % total_samples # read signed 2 byte shorts
else:
raise ValueError("Only supports 8 and 16 bit audio formats.")
total_samples = num_frames * num_channels
integer_data = struct.unpack(fmt, raw_data) # this is amplitude
return integer_data, num_frames, sample_rate
def dataToWav(wavName,data):
# IMPORTANT hard coded standard values, refenced from the audiocheck.net files
stream = wave.open(wavName,"wb")
stream.setnchannels(1)
stream.setframerate(44100)
stream.setsampwidth(2)
numFrames = len(data)
stream.setnframes(numFrames)
fmt = "%ih" % numFrames
data = struct.pack(fmt,*data) #unpacks all
stream.writeframes(data)
def wavToCsv(wavName,csvName):
with open(csvName, 'wb') as csvfile: #contain all data in 1 row
# x = time (1 frame)
# y = freq
filewriter = csv.writer(csvfile, delimiter=',',
quotechar='|', quoting=csv.QUOTE_MINIMAL)
#get data
integer_data, num_frames, sample_rate = wavToData(wavName)
#adds headings and other needed data
frameCount = [x+1 for x in range(num_frames)] # x
fullData = [x for x in integer_data] # y
frameCount.insert(0, "Time (increments of %f)"%(1/sample_rate))
fullData.insert(0,"Frequency(dBfs)")
fullData = frameCount+fullData
filewriter.writerow(fullData)
def mergeWavData(wav1,wav2,mergeWav): #generates a brand new wav file
firstData = wavToData(wav1)[0]
secondData = wavToData(wav2)[0]
limitCheck = lambda x,y: ((x+y)> 32767 or (x+y) < -32767)
limitConstrain = lambda x,y: 32767 if (x+y) >32767 else -32767
newData = []
if len(firstData) < len(secondData): #adds the amplitudes of both wav file up to the shortest one
for i in range(len(firstData)):
data = 0
if(limitCheck(firstData[i],secondData[i])): #exceed limit
data = limitConstrain(firstData[i],secondData[i])
else:
data = firstData[i]+secondData[i]
newData.append(data)
else:
for i in range(len(secondData)):
data = 0
if(limitCheck(firstData[i],secondData[i])): #exceed limit
data = limitConstrain(firstData[i],secondData[i])
else:
data = firstData[i]+secondData[i]
if (data > 32767 or data <-32767):
print data
newData.append(data)
dataToWav(mergeWav,newData)
mergeWavData('audiocheck.net_sin_1000Hz_-3dBFS_3s.wav','audiocheck.net_sin_4000Hz_-3dBFS_3s.wav','1000Hz+4000Hz.wav')
wavToCsv('1000Hz+4000Hz.wav','1000Hz+4000Hz.csv') # maeks csv out of wav ayayayay
#include tags (.wav/.csv)