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Problems when I write np array to binary file, new file is only half of the original one

I am trying to remove top 24 lines of a raw file, so I opened the original raw file(let's call it raw1.raw) and converted it to nparray, then I initialized a new array and remove the top24 lines, but after writing new array to the new binary file(raw2.raw), I found raw2 is 15.2mb only while the original file raw1.raw is like 30.6mb, my code:
import numpy as np
import imageio
import rawpy
import cv2
def ave():
fd = open('raw1.raw', 'rb')
rows = 3000 #around 3000, not the real rows
cols = 5100 #around 5100, not the real cols
f = np.fromfile(fd, dtype=np.uint8,count=rows*cols)
I_array = f.reshape((rows, cols)) #notice row, column format
#print(I_array)
fd.close()
im = np.zeros((rows - 24 , cols))
for i in range (len(I_array) - 24):
for j in range(len(I_array[i])):
im[i][j] = I_array[i + 24][j]
#print(im)
newFile = open("raw2.raw", "wb")
im.astype('uint8').tofile(newFile)
newFile.close()
if __name__ == "__main__":
ave()
I tried to use im.astype('uint16') when write in the binary file, but the value would be wrong if I use uint16.

There must clearly be more data in your 'raw1.raw' file that you are not using. Are you sure that file wasn't created using 'uint16' data and you are just pulling out the first half as 'uint8' data? I just checked the writing of random data.
import os, numpy as np
x = np.random.randint(0,256,size=(3000,5100),dtype='uint8')
x.tofile(open('testfile.raw','w'))
print(os.stat('testfile.raw').st_size) #I get 15.3MB.
So, 'uint8' for a 3000 by 5100 clearly takes up 15.3MB. I don't know how you got 30+.
############################ EDIT #########
Just to add more clarification. Do you realize that dtype does nothing more than change the "view" of your data? It doesn't effect the actual data that is saved in memory. This also goes for data that you read from a file. Take for example:
import numpy as np
#The way to understand x, is that x is taking 12 bytes in memory and using
#that information to hold 3 values. The first 4 bytes are the first value,
#the second 4 bytes are the second, etc.
x = np.array([1,2,3],dtype='uint32')
#Change x to display those 12 bytes at 6 different values. Doing this does
#NOT change the data that the array is holding. You are only changing the
#'view' of the data.
x.dtype = 'uint16'
print(x)
In general (there are few special cases), changing the dtype doesn't change the underlying data. However, the conversion function .astype() does change the underlying data. If you have any array of 12 bytes viewed as 'int32' then running .astype('uint8') will take each entry (4 bytes) and covert it (known as casting) to a uint8 entry (1 byte). The new array will only have 3 bytes for the 3 entries. You can see this litterally:
x = np.array([1,2,3],dtype='uint32')
print(x.tobytes())
y = x.astype('uint8')
print(y.tobytes())
So, when we say that a file is 30mb, we mean that the file has (minus some header information) is 30,000,000 bytes which are exactly uint8s. 1 uint8 is 1 byte. If any array has 6000by5100 uint8s (bytes), then the array has 30,600,000 bytes of information in memory.
Likewise, if you read a file (DOES NOT MATTER THE FILE) and write np.fromfile(,dtype=np.uint8,count=15_300_000) then you told python to read EXACTLY 15_300_000 bytes (again 1 byte is 1 uint8) of information (15mb). If your file is 100mb, 40mb, or even 30mb, it would be completely irrelevant because you told python to only read the first 15mb of data.

How to skip bytes after reading data using numpy fromfile

I'm trying to read noncontiguous fields from a binary file in Python using numpy fromfile function. It's based on this Matlab code using fread:
fseek(file, 0, 'bof');
q = fread(file, inf, 'float32', 8);
8 indicates the number of bytes I want to skip after reading each value. I was wondering if there was a similar option in fromfile, or if there is another way of reading specific values from a binary file in Python. Thanks for your help.
Henrik

Something like this should work, untested:
import struct
floats = []
with open(filename, 'rb') as f:
while True:
buff = f.read(4) # 'f' is 4-bytes wide
if len(buff) < 4: break
x = struct.unpack('f', buff)[0] # Convert buffer to float (get from returned tuple)
floats.append(x) # Add float to list (for example)
f.seek(8, 1) # The second arg 1 specifies relative offset
Using struct.unpack()

Extract required bytes from a file in Python

I have a binary file here:
ftp://n5eil01u.ecs.nsidc.org/SAN/GLAS/GLA06.034/2003.02.21/GLA06_634_1102_001_0079_3_01_0001.DAT
I have to extract the following data from that file:
Byte Offset: 176
Data type: 4-byte (long) integer
Total bytes: 160
I tried as follows:
import numpy as np
fname = 'GLA06_634_1102_001_0079_3_01_0001.DAT'
with open(fname,'rb') as fi:
fi.seek (176,0)
data= np.fromfile(fi,dtype='long',count=160)
print data
No success, what's wrong with my idea?

Using a hard coded offset is a rather fragile solution. But assuming you know what you are doing:
Byte Offset: 176
Data type: 4-byte (long) integer
Total bytes: 160
AKAICT, that leads to 160/4 = 40 values to read (could you confirm that?)
In addition, the type should be one of the numpy defined type. Here np.int32 might be the right one:
data= np.fromfile(fi,dtype=np.int32,count=40)
On my computer, this produces the following result:
[1919251297 997485633 1634494218 1936678771 1634885475 825124212
808333629 808464432 942813232 1818692155 1868526433 1918854003
1600484449 1702125924 842871086 758329392 841822768 1728723760
1601397100 1600353135 1702125938 1835627615 1026633317 809119792
808466992 1668483643 1668509535 1952543327 1026633317 960048688
960051513 909654073 926037812 1668483643 1668509535 1952543327
1633967973 825124212 808464957 842018099]
If this is not what expected, maybe you have a problem of endianness.
Numpy as support for custom defined types to solve that problem:
For example:
np.dtype('<i4') is 4 bytes (signed) integer little endian
np.dtype('>i4') is 4 bytes (signed) integer big endian
In you case, to force reading data as little endian, you might write:
dt = np.dtype('<i4')
with open(fname,'rb') as fi:
fi.seek (176,0)
data= np.fromfile(fi,dtype=dt,count=40)
print data

how to convert wav file to float amplitude

so I asked everything in the title:
I have a wav file (written by PyAudio from an input audio) and I want to convert it in float data corresponding of the sound level (amplitude) to do some fourier transformation etc...
Anyone have an idea to convert WAV data to float?

I have identified two decent ways of doing this.
Method 1: using the wavefile module
Use this method if you don't mind installing some extra libraries which involved a bit of messing around on my Mac but which was easy on my Ubuntu server.
https://github.com/vokimon/python-wavefile
import wavefile
# returns the contents of the wav file as a double precision float array
def wav_to_floats(filename = 'file1.wav'):
w = wavefile.load(filename)
return w[1][0]
signal = wav_to_floats(sys.argv[1])
print "read "+str(len(signal))+" frames"
print "in the range "+str(min(signal))+" to "+str(max(signal))
Method 2: using the wave module
Use this method if you want less module install hassles.
Reads a wav file from the filesystem and converts it into floats in the range -1 to 1. It works with 16 bit files and if they are > 1 channel, will interleave the samples in the same way they are found in the file. For other bit depths, change the 'h' in the argument to struct.unpack according to the table at the bottom of this page:
https://docs.python.org/2/library/struct.html
It will not work for 24 bit files as there is no data type that is 24 bit, so there is no way to tell struct.unpack what to do.
import wave
import struct
import sys
def wav_to_floats(wave_file):
w = wave.open(wave_file)
astr = w.readframes(w.getnframes())
# convert binary chunks to short
a = struct.unpack("%ih" % (w.getnframes()* w.getnchannels()), astr)
a = [float(val) / pow(2, 15) for val in a]
return a
# read the wav file specified as first command line arg
signal = wav_to_floats(sys.argv[1])
print "read "+str(len(signal))+" frames"
print "in the range "+str(min(signal))+" to "+str(max(signal))

I spent hours trying to find the answer to this. The solution turns out to be really simple: struct.unpack is what you're looking for. The final code will look something like this:
rawdata=stream.read() # The raw PCM data in need of conversion
from struct import unpack # Import unpack -- this is what does the conversion
npts=len(rawdata) # Number of data points to be converted
formatstr='%ih' % npts # The format to convert the data; use '%iB' for unsigned PCM
int_data=unpack(formatstr,rawdata) # Convert from raw PCM to integer tuple
Most of the credit goes to Interpreting WAV Data. The only trick is getting the format right for unpack: it has to be the right number of bytes and the right format (signed or unsigned).

Most wave files are in PCM 16-bit integer format.
What you will want to:
Parse the header to known which format it is (check the link from Xophmeister)
Read the data, take the integer values and convert them to float
Integer values range from -32768 to 32767, and you need to convert to values from -1.0 to 1.0 in floating points.
I don't have the code in python, however in C++, here is a code excerpt if the PCM data is 16-bit integer, and convert it to float (32-bit):
short* pBuffer = (short*)pReadBuffer;
const float ONEOVERSHORTMAX = 3.0517578125e-5f; // 1/32768
unsigned int uFrameRead = dwRead / m_fmt.Format.nBlockAlign;
for ( unsigned int i = 0; i < uFrameCount * m_fmt.Format.nChannels; ++i )
{
short i16In = pBuffer[i];
out_pBuffer[i] = (float)i16In * ONEOVERSHORTMAX;
}
Be careful with stereo files, as the stereo PCM data in wave files is interleaved, meaning the data looks like LRLRLRLRLRLRLRLR (instead of LLLLLLLLRRRRRRRR). You may or may not need to de-interleave depending what you do with the data.

This version reads a wav file from the filesystem and converts it into floats in the range -1 to 1. It works with files of all sample widths and it will interleave the samples in the same way they are found in the file.
import wave
def read_wav_file(filename):
def get_int(bytes_obj):
an_int = int.from_bytes(bytes_obj, 'little', signed=sampwidth!=1)
return an_int - 128 * (sampwidth == 1)
with wave.open(filename, 'rb') as file:
sampwidth = file.getsampwidth()
frames = file.readframes(-1)
bytes_samples = (frames[i : i+sampwidth] for i in range(0, len(frames), sampwidth))
return [get_int(b) / pow(2, sampwidth * 8 - 1) for b in bytes_samples]
Also here is a link to the function that converts floats back to ints and writes them to desired wav file:
https://gto76.github.io/python-cheatsheet/#writefloatsamplestowavfile

The Microsoft WAVE format is fairly well documented. See https://ccrma.stanford.edu/courses/422/projects/WaveFormat/ for example. It wouldn't take much to write a file parser to open and interpret the data to get the information you require... That said, it's almost certainly been done before, so I'm sure someone will give an "easier" answer ;)

Interpreting WAV Data

I'm trying to write a program to display PCM data. I've been very frustrated trying to find a library with the right level of abstraction, but I've found the python wave library and have been using that. However, I'm not sure how to interpret the data.
The wave.getparams function returns (2 channels, 2 bytes, 44100 Hz, 96333 frames, No compression, No compression). This all seems cheery, but then I tried printing a single frame:'\xc0\xff\xd0\xff' which is 4 bytes. I suppose it's possible that a frame is 2 samples, but the ambiguities do not end there.
96333 frames * 2 samples/frame * (1/44.1k sec/sample) = 4.3688 seconds
However, iTunes reports the time as closer to 2 seconds and calculations based on file size and bitrate are in the ballpark of 2.7 seconds. What's going on here?
Additionally, how am I to know if the bytes are signed or unsigned?
Many thanks!

Thank you for your help! I got it working and I'll post the solution here for everyone to use in case some other poor soul needs it:
import wave
import struct
def pcm_channels(wave_file):
"""Given a file-like object or file path representing a wave file,
decompose it into its constituent PCM data streams.
Input: A file like object or file path
Output: A list of lists of integers representing the PCM coded data stream channels
and the sample rate of the channels (mixed rate channels not supported)
"""
stream = wave.open(wave_file,"rb")
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.")
integer_data = struct.unpack(fmt, raw_data)
del raw_data # Keep memory tidy (who knows how big it might be)
channels = [ [] for time in range(num_channels) ]
for index, value in enumerate(integer_data):
bucket = index % num_channels
channels[bucket].append(value)
return channels, sample_rate

"Two channels" means stereo, so it makes no sense to sum each channel's duration -- so you're off by a factor of two (2.18 seconds, not 4.37). As for signedness, as explained for example here, and I quote:
8-bit samples are stored as unsigned
bytes, ranging from 0 to 255. 16-bit
samples are stored as 2's-complement
signed integers, ranging from -32768
to 32767.
This is part of the specs of the WAV format (actually of its superset RIFF) and thus not dependent on what library you're using to deal with a WAV file.

I know that an answer has already been accepted, but I did some things with audio a while ago and you have to unpack the wave doing something like this.
pcmdata = wave.struct.unpack("%dh"%(wavedatalength),wavedata)
Also, one package that I used was called PyAudio, though I still had to use the wave package with it.

Each sample is 16 bits and there 2 channels, so the frame takes 4 bytes

The duration is simply the number of frames divided by the number of frames per second. From your data this is: 96333 / 44100 = 2.18 seconds.

Building upon this answer, you can get a good performance boost by using numpy.fromstring or numpy.fromfile. Also see this answer.
Here is what I did:
def interpret_wav(raw_bytes, n_frames, n_channels, sample_width, interleaved = True):
if sample_width == 1:
dtype = np.uint8 # unsigned char
elif sample_width == 2:
dtype = np.int16 # signed 2-byte short
else:
raise ValueError("Only supports 8 and 16 bit audio formats.")
channels = np.fromstring(raw_bytes, dtype=dtype)
if interleaved:
# channels are interleaved, i.e. sample N of channel M follows sample N of channel M-1 in raw data
channels.shape = (n_frames, n_channels)
channels = channels.T
else:
# channels are not interleaved. All samples from channel M occur before all samples from channel M-1
channels.shape = (n_channels, n_frames)
return channels
Assigning a new value to shape will throw an error if it requires data to be copied in memory. This is a good thing, since you want to use the data in place (using less time and memory overall). The ndarray.T function also does not copy (i.e. returns a view) if possible, but I'm not sure how you ensure that it does not copy.
Reading directly from the file with np.fromfile will be even better, but you would have to skip the header using a custom dtype. I haven't tried this yet.