Bootcamp Audio Day

import numpy as np

#################################################

############# NUMPY ARRAYS FOR AUDIO? ###########

#################################################

# I am a numpy array containing some audio

myAudio = np.array([-0.25,0.0,0.25,0.5])

# turn it up!

myAudio = myAudio*2.

# great!

print myAudio

# what can we find out about our numpy audio?

print "Dimensions: ",myAudio.ndim

print "Size of each dimension: ",myAudio.shape

print "Total number of elements: ", myAudio.size # this is product of shape dimensions

print "Data type: ",myAudio.dtype # ...or dtype.name

print "Bytes per element: ",myAudio.itemsize 

print "Second element: ",myAudio[1]

print "All elements up to (but not including) 2: ",myAudio[:2]

print "All elements from index 2 to end: ",myAudio[2:]

print "Abs!: ",np.abs(myAudio)

print "Min: ",np.min(myAudio)

print "Max: ",np.max(myAudio)

print "Sum: ",np.sum(myAudio)

print "Sqrt: ",np.sqrt(9)

print "FFT!: ",np.fft.fft(myAudio)

print "Random!: ",np.random.random(4)

print "Ones: ",np.ones(4)

print "Zeros: ",np.zeros(4)

import numpy as np

from scikits.audiolab import wavread

#################################################

############ EXTRACT AUDIO FROM FILE ############

#################################################

x, fs, enc = wavread("viola.wav")

#################################################

#### CALCULATE RMS OF EACH AUDIO BLOCK ####

#################################################

hop_size = 2048                             # set hop size

frame_size = 4096                           # set frame size

frame = np.zeros(frame_size)                # initialise frame with zeros

window = np.hanning(frame_size)             # create window of the same length as the hop size

# create empty numpy array to hold our 

rms = np.array([])

# run through signal frame by frame 

for n in range(0,x.size-hop_size,hop_size):

    # extract a segment of length hop_size

    buffer = x[n:n+hop_size]                               

    # add new segment to frame, shifting back samples of frame

    frame = np.append(frame[hop_size:frame_size],buffer)  

    # calculate RMS

    rms_val = np.sqrt(np.power(frame,2).mean())

    # add amplitude to our numpy array

    rms = np.append(rms,rms_val)

print rms

import numpy as np

from scikits.audiolab import Sndfile

from scikits.audiolab import Format

#################################################

######## CREATING A SOUND FILE INSTANCE #########

#################################################

# create Sndfile instance with our example audio file

f = Sndfile('viola.wav', 'r')

#################################################

######## EXTRACTING AUDIO FILE META-DATA ########

#################################################

# extract and print sample rate

fs = f.samplerate

print "sample rate: ",fs

# extract and print the number of channels

nc = f.channels

print "number of channels: ",nc

# extract the number of samples

nsamples = f.nframes

#################################################

######## READ AUDIO SAMPLES FROM THE FILE #######

#################################################

# we can read audio samples using the read_frame method

data = f.read_frames(nsamples)

#################################################

########## APPLY A DELAY TO THE SAMPLES #########

#################################################

# delay in samples

delay = fs/2

# volume of delayed sound

alpha = 0.75

# create an empty array for the output

out = np.zeros(data.size)

# for every sample in the array

for i in range(data.size):

    # if we are safe to apply the delay without negative indexing

    if (i >= delay):

        out[i] = data[i] + data[i-delay]*alpha

    else:

        out[i] = data[i] # hacky

#################################################

########## WRITING TO A NEW AUDIO FILE ##########

#################################################

# create a name for the new file

new_filename = 'delayed_audio.wav'

# Create a Sndfile instance for writing wav files @ 44100 Hz

format = Format('wav')

f = Sndfile(new_filename, 'w', format, 1, 44100)

# Write out the samples to an audio file

f.write_frames(out)

# close the audio file

f.close()

#################################################

#################################################

#################################################

import array as array

#################################################

############# PYTHON LISTS FOR AUDIO? ###########

#################################################

# I am a list of audio samples

myAudio = [-0.25,0.0,0.25,0.5]

# turn it up!

myAudio = myAudio*2

# oh dear!

print myAudio

# also, look what got into our array 

myAudio.append("hello!")

# oh dear again! - Python stores type information for every entry? Do we want

# to do that 44100 times a second?

print myAudio

#################################################

############# PYTHON ARRAYS FOR AUDIO? ##########

#################################################

# I am an array of audio samples

myAudio = array.array('d',[-0.25,0.0,0.25,0.5])

# this doesn't work, that's good

# myAudio.append("hello")

# but what about this!

myAudio = myAudio*2

# oh dear oh dear oh dear

print myAudio

import numpy as np

from scikits.audiolab import wavread

from matplotlib import pylab as plt

#################################################

############ EXTRACT AUDIO FROM FILE ############

#################################################

x, fs, enc = wavread("drums_mono.wav")

#################################################

#### CALCULATE RMS OF EACH AUDIO BLOCK ####

#################################################

hop_size = 1024                              # set hop size

frame_size = hop_size*2                     # set frame size

frame = np.zeros(frame_size)                # initialise frame with zeros

window = np.hanning(frame_size)             # create window of the same length as the hop size

# create empty numpy array to hold our 

rms = np.array([])

# run through signal frame by frame 

for n in range(0,x.size-hop_size,hop_size):

    # extract a segment of length hop_size

    buffer = x[n:n+hop_size]                               

    # add new segment to frame, shifting back samples of frame

    frame = np.append(frame[hop_size:frame_size],buffer)  

    # calculate RMS

    rms_val = np.sqrt(np.power(frame,2).mean())

    # add amplitude to our numpy array

    rms = np.append(rms,rms_val)

print rms

plt.plot(rms)

plt.title("RMS")

plt.xlabel("time")

plt.ylabel("value")

plt.show()

import os

from scikits.audiolab import Sndfile

from scikits.audiolab import Format

# get the current working directory

path = os.getcwd()

# get all file names in the current directory

files = os.listdir(path)

# for each file name

for filename in files:

    # if the file ends with a .wav extension

    if filename.endswith('.wav'):

        # create a Sndfile instance for the file

        f_in = Sndfile(filename, 'r')

        # extract the number of frames

        numframes = f_in.nframes

        # read all audio samples into 'data'

        data = f_in.read_frames(numframes)

        # extract the name (without extension from the file name)

        name,extension = os.path.splitext(filename)

        # create a new filename with a .aiff extension

        new_filename = name + '.aiff'

        # create the new format based on aiff

        format = Format('aiff')

        # create a new Sndfile instance for the output file

        f_out = Sndfile(new_filename, 'w', format, f_in.channels, f_in.samplerate)

        # write out audio samples to the new file

        f_out.write_frames(data[:numframes])

        # close the audio file

        f_out.close()

from scikits.audiolab import Sndfile

from scikits.audiolab import Format

from scikits.audiolab import play

#################################################

######## CREATING A SOUND FILE INSTANCE #########

#################################################

# create Sndfile instance with our example audio file

f = Sndfile('viola.wav', 'r')

#################################################

######## EXTRACTING AUDIO FILE META-DATA ########

#################################################

# extract and print sample rate

fs = f.samplerate

print "sample rate: ",fs

# extract and print the number of channels

nc = f.channels

print "number of channels: ",nc

# extract and print the encoding format

enc = f.encoding

print "encoding format: ",enc

# extract the number of frames - single samples for

# mono and pairs of samples for stereo

nsamples = f.nframes

#################################################

######## READ AUDIO SAMPLES FROM THE FILE #######

#################################################

# we can read audio samples using the read_frame method

data = f.read_frames(nsamples)

#################################################

############# PLAYING AN AUDIO FILE #############

#################################################

# play the audio file data in 'data' at 44100Hz

play(data,fs=44100)

#################################################

#################################################

#################################################

from scikits.audiolab import Sndfile

from scikits.audiolab import Format

from scikits.audiolab import play

#################################################

######## CREATING A SOUND FILE INSTANCE #########

#################################################

# create Sndfile instance with our example audio file

f = Sndfile('viola.wav', 'r')

#################################################

######## EXTRACTING AUDIO FILE META-DATA ########

#################################################

# extract and print sample rate

fs = f.samplerate

print "sample rate: ",fs

# extract and print the number of channels

nc = f.channels

print "number of channels: ",nc

# extract and print the encoding format

enc = f.encoding

print "encoding format: ",enc

# extract the number of frames - single samples for

# mono and pairs of samples for stereo

nsamples = f.nframes

#################################################

######## READ AUDIO SAMPLES FROM THE FILE #######

#################################################

# we can read audio samples using the read_frame method

data = f.read_frames(nsamples)

#################################################

########## WRITING TO A NEW AUDIO FILE ##########

#################################################

# create a name for the new file

new_filename = 'output_file.wav'

# create the output audio data, in this case a simple copy

output_data = data

# Create a Sndfile instance for writing wav files @ 44100 Hz

format = Format('wav')

f = Sndfile(new_filename, 'w', format, 1, 44100)

# Write out the first 3 seconds worth of samples (fs*3)

f.write_frames(output_data[:fs*3])

# close the audio file

f.close()

#################################################

#################################################

#################################################

import numpy as np

from scikits.audiolab import Sndfile

from scikits.audiolab import Format

#################################################

############ CREATE SOME RANDOM NOISE ###########

#################################################

# set our sampling frequency

fs = 44100

# create a numpy array that can hold 3 seconds of sound

noise = np.empty(3*fs)

# set each element to a random value (noise)

for i in range(noise.size):

    # generate random value and turn it down!

    noise[i] = np.random.random()*0.2

#################################################

########## WRITING NOISE TO AUDIO FILE ##########

#################################################

# create a name for the new file

new_filename = 'noise.wav'

# Create a Sndfile instance for writing wav files @ 44100 Hz

format = Format('wav')

f = Sndfile(new_filename, 'w', format, 1, fs)

# Write out the samples to the file

f.write_frames(noise)

# close the audio file

f.close()

import numpy as np

from scikits.audiolab import Sndfile

from scikits.audiolab import Format

#################################################

############## CREATE A SINE TONE ###############

#################################################

# set our sampling frequency

fs = 44100

# create a numpy array that can hold 3 seconds of sound

tone = np.zeros((3*fs,2))

# set frequency to 440Hz

freq = 440

# set volume to 0.3

amp = 0.3

# set values of each channel

for i in range(tone.shape[0]):

    # calculate phase value

    phaseVal = np.float(i)/np.float(fs)

    # generate tone and set volume for left and right

    tone[i][0] = np.sin(2*np.pi*freq*phaseVal)*amp

    tone[i][1] = np.sin(2*np.pi*freq*phaseVal)*amp

#################################################

########## WRITING TONES TO AUDIO FILE ##########

#################################################

# create a name for the new file

new_filename = 'tone.wav'

# Create a Sndfile instance for writing wav files @ 44100 Hz

format = Format('wav')

f = Sndfile(new_filename, 'w', format, 2, fs)

# Write out the samples to the file

f.write_frames(tone)

# close the audio file

f.close()

from scikits.audiolab import wavread

# specify a file name

filename = "viola3.wav"

# extract audio from file

x, fs, enc = wavread(filename)     

# print out the first 50 samples

print x[0:50]