পাইথনে * .wav ফাইলগুলি পড়া

90

আমার একটি .wav ফাইলে লেখা শব্দ বিশ্লেষণ করা দরকার। তার জন্য আমার এই ফাইলটি সংখ্যার সেটে রূপান্তর করা দরকার (উদাহরণস্বরূপ অ্যারে)। আমি মনে করি আমার তরঙ্গ প্যাকেজটি ব্যবহার করা দরকার। তবে এটি ঠিক কীভাবে কাজ করে তা আমি জানি না। উদাহরণস্বরূপ আমি নিম্নলিখিতটি করেছি:

import wave
w = wave.open('/usr/share/sounds/ekiga/voicemail.wav', 'r')
for i in range(w.getnframes()):
    frame = w.readframes(i)
    print frame

As a result of this code I expected to see sound pressure as function of time. In contrast I see a lot of strange, mysterious symbols (which are not hexadecimal numbers). Can anybody, pleas, help me with that?

python  audio  wav  wave 
— Roman
সূত্র

উত্তর:

110

Per the documentation, scipy.io.wavfile.read(somefile) returns a tuple of two items: the first is the sampling rate in samples per second, the second is a numpy array with all the data read from the file:

from scipy.io import wavfile
samplerate, data = wavfile.read('./output/audio.wav')
— Alex Martelli
সূত্র
You can combine this with command line conversion tools to open other formats.
— endolith
11
It seriously lacks the number of channels though. How are you supposed to work with audio without knowing the number of channels?
— bastibe
throwns some weird struct unpacking errors on my computer. I think it's using struct.unpack('<i',data) instead of the struct.unpack('<h',data) nak used below.
— Alex S
1
Does this library work? I run into a number of problems: scipy.io.wavfile.read('/usr/lib/python2.7/dist-packages/pygame/examples/data/house_lo.wav') -> No data. scipy.io.wavfile.read('/usr/lib/python2.7/dist-packages/pygame/examples/data/secosmic_lo.wav') -> ZeroDivisionError: integer division or modulo by zero
— Finn Årup Nielsen
6
@bastibe data is a 2-D numpy array so data.shape returns a tuple of (num_samples, num_channels)
— hobs
63

Using the struct module, you can take the wave frames (which are in 2's complementary binary between -32768 and 32767 (i.e. 0x8000 and 0x7FFF). This reads a MONO, 16-BIT, WAVE file. I found this webpage quite useful in formulating this:

import wave, struct

wavefile = wave.open('sine.wav', 'r')

length = wavefile.getnframes()
for i in range(0, length):
    wavedata = wavefile.readframes(1)
    data = struct.unpack("<h", wavedata)
    print(int(data[0]))

This snippet reads 1 frame. To read more than one frame (e.g., 13), use

wavedata = wavefile.readframes(13)
data = struct.unpack("<13h", wavedata)
— nak
সূত্র
2
how to handle 24bits stereo files ?
— Basj
14
this gives me the error: "struct.error: unpack requires a string argument of length 2"
— Coder404
1
If you run this piece of code with a very big audio file. Your computer will be die to due the memory need by this program. Need to process audio file by block for big audio file
— ArthurLambert
@Coder404 You probably have a stereo wave file, or a different bit depth.
— jmilloy
3
For those who, like me, are wondering what is 2s complementary binary, see here stackoverflow.com/questions/1049722/what-is-2s-complement
— Dennis Golomazov
34

Different Python modules to read wav:

There is at least these following libraries to read wave audio files:

The most simple example:

This is a simple example with SoundFile:

import soundfile as sf
data, samplerate = sf.read('existing_file.wav')

Format of the output:

Warning, the data are not always in the same format, that depends on the library. For instance:

from scikits import audiolab
from scipy.io import wavfile
from sys import argv
for filepath in argv[1:]:
    x, fs, nb_bits = audiolab.wavread(filepath)
    print('Reading with scikits.audiolab.wavread:', x)
    fs, x = wavfile.read(filepath)
    print('Reading with scipy.io.wavfile.read:', x)

Output:

Reading with scikits.audiolab.wavread: [ 0.          0.          0.         ..., -0.00097656 -0.00079346 -0.00097656]
Reading with scipy.io.wavfile.read: [  0   0   0 ..., -32 -26 -32]

SoundFile and Audiolab return floats between -1 and 1 (as matab does, that is the convention for audio signals). Scipy and wave return integers, which you can convert to floats according to the number of bits of encoding, for example:

from scipy.io.wavfile import read as wavread
samplerate, x = wavread(audiofilename)  # x is a numpy array of integers, representing the samples 
# scale to -1.0 -- 1.0
if x.dtype == 'int16':
    nb_bits = 16  # -> 16-bit wav files
elif x.dtype == 'int32':
    nb_bits = 32  # -> 32-bit wav files
max_nb_bit = float(2 ** (nb_bits - 1))
samples = x / (max_nb_bit + 1)  # samples is a numpy array of floats representing the samples
— PatriceG
সূত্র
14

IMHO, the easiest way to get audio data from a sound file into a NumPy array is SoundFile:

import soundfile as sf
data, fs = sf.read('/usr/share/sounds/ekiga/voicemail.wav')

This also supports 24-bit files out of the box.

There are many sound file libraries available, I've written an overview where you can see a few pros and cons. It also features a page explaining how to read a 24-bit wav file with the wave module.

— Matthias
সূত্র
Note: soundfile.read() normalizes by 2^(n_bits - 1) as in sandoval's scipy.io.wavfile example
— Quetzalcoatl
9

You can accomplish this using the scikits.audiolab module. It requires NumPy and SciPy to function, and also libsndfile.

Note, I was only able to get it to work on Ubunutu and not on OSX.

from scikits.audiolab import wavread

filename = "testfile.wav"

data, sample_frequency,encoding = wavread(filename)

Now you have the wav data

— ch3rryc0ke
সূত্র
scikits.audiolab has not been updated since 2010 and it's probably Python 2 only.
— Boris
4

If you want to procces an audio block by block, some of the given solutions are quite awful in the sense that they imply loading the whole audio into memory producing many cache misses and slowing down your program. python-wavefile provides some pythonic constructs to do NumPy block-by-block processing using efficient and transparent block management by means of generators. Other pythonic niceties are context manager for files, metadata as properties... and if you want the whole file interface, because you are developing a quick prototype and you don't care about efficency, the whole file interface is still there.

A simple example of processing would be:

import sys
from wavefile import WaveReader, WaveWriter

with WaveReader(sys.argv[1]) as r :
    with WaveWriter(
            'output.wav',
            channels=r.channels,
            samplerate=r.samplerate,
            ) as w :

        # Just to set the metadata
        w.metadata.title = r.metadata.title + " II"
        w.metadata.artist = r.metadata.artist

        # This is the prodessing loop
        for data in r.read_iter(size=512) :
            data[1] *= .8     # lower volume on the second channel
            w.write(data)

The example reuses the same block to read the whole file, even in the case of the last block that usually is less than the required size. In this case you get an slice of the block. So trust the returned block length instead of using a hardcoded 512 size for any further processing.

— vokimon
সূত্র
1

I needed to read a 1-channel 24-bit WAV file. The post above by Nak was very useful. However, as mentioned above by basj 24-bit is not straightforward. I finally got it working using the following snippet:

from scipy.io import wavfile
TheFile = 'example24bit1channelFile.wav'
[fs, x] = wavfile.read(TheFile)

# convert the loaded data into a 24bit signal

nx = len(x)
ny = nx/3*4    # four 3-byte samples are contained in three int32 words

y = np.zeros((ny,), dtype=np.int32)    # initialise array

# build the data left aligned in order to keep the sign bit operational.
# result will be factor 256 too high

y[0:ny:4] = ((x[0:nx:3] & 0x000000FF) << 8) | \
  ((x[0:nx:3] & 0x0000FF00) << 8) | ((x[0:nx:3] & 0x00FF0000) << 8)
y[1:ny:4] = ((x[0:nx:3] & 0xFF000000) >> 16) | \
  ((x[1:nx:3] & 0x000000FF) << 16) | ((x[1:nx:3] & 0x0000FF00) << 16)
y[2:ny:4] = ((x[1:nx:3] & 0x00FF0000) >> 8) | \
  ((x[1:nx:3] & 0xFF000000) >> 8) | ((x[2:nx:3] & 0x000000FF) << 24)
y[3:ny:4] = (x[2:nx:3] & 0x0000FF00) | \
  (x[2:nx:3] & 0x00FF0000) | (x[2:nx:3] & 0xFF000000)

y = y/256   # correct for building 24 bit data left aligned in 32bit words

Some additional scaling is required if you need results between -1 and +1. Maybe some of you out there might find this useful

— ProgJos
সূত্র
0

if its just two files and the sample rate is significantly high, you could just interleave them.

from scipy.io import wavfile
rate1,dat1 = wavfile.read(File1)
rate2,dat2 = wavfile.read(File2)

if len(dat2) > len(dat1):#swap shortest
    temp = dat2
    dat2 = dat1
    dat1 = temp

output = dat1
for i in range(len(dat2)/2): output[i*2]=dat2[i*2]

wavfile.write(OUTPUT,rate,dat)
— leec
সূত্র
0

u can also use simple import wavio library u also need have some basic knowledge of the sound.

— yunus
সূত্র
0

PyDub (http://pydub.com/) has not been mentioned and that should be fixed. IMO this is the most comprehensive library for reading audio files in Python right now, although not without its faults. Reading a wav file:

from pydub import AudioSegment

audio_file = AudioSegment.from_wav('path_to.wav')
# or
audio_file = AudioSegment.from_file('path_to.wav')

# do whatever you want with the audio, change bitrate, export, convert, read info, etc.
# Check out the API docs http://pydub.com/

PS. The example is about reading a wav file, but PyDub can handle a lot of various formats out of the box. The caveat is that it's based on both native Python wav support and ffmpeg, so you have to have ffmpeg installed and a lot of the pydub capabilities rely on the ffmpeg version. Usually if ffmpeg can do it, so can pydub (which is quite powerful).

Non-disclaimer: I'm not related to the project, but I am a heavy user.

— wanaryytel
সূত্র
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