def subtract(newdoc, newdoc2, current_user):
# newdoc_wav_path = newdoc.file
# print newdoc_wav_path
# recording_path = os.path.join(settings.MEDIA_ROOT, newdoc.file )
# print newdoc.file
# type newdoc.file
# print newdoc.file.url
# get paths for raw audio and music to be subtracted
raw_audio_filename = os.path.basename(newdoc.file.url)
original_audio_filename = os.path.basename(newdoc2.file.url)
# newdoc_url = newdoc.file.url
# print type(newdoc_url)
# Create paths to read in audio files
recording_path = os.path.join(settings.MEDIA_ROOT, raw_audio_filename)
original_audio_path = os.path.join(settings.MEDIA_ROOT, original_audio_filename)
# recording_path = os.path.join(settings.BASE_DIR, newdoc_url)
# print recording_path
# print settings.BASE_DIR
rate, data = scipy.io.wavfile.read(recording_path)
r = data.astype(np.float64)
rate, data = scipy.io.wavfile.read(original_audio_path)
o = data.astype(np.float64)
print len(o)
print len(r)
print 'applying adaptive filt'
# Apply adaptive filter
M = 100 #8000 (takelly long time) #100 # Number of filter taps in adaptive filter
step = 0.1 # Step size
y, e, w = adf.nlms(o, r, M, step, returnCoeffs=True)
scaled_e = np.int16(e/np.max(np.abs(e)) * 32767)
# d = room_simulate.room_sim(u)
# scaled_d = np.int16(d/np.max(np.abs(d)) * 32767)
# output_path = os.path.join(settings.MEDIA_ROOT, newdoc_filename+'TEST2.wav' )
output_filename = os.path.splitext(os.path.basename(newdoc.file.url))[0]
print output_filename
output_path = os.path.join(settings.MEDIA_ROOT,output_filename+'_SUBTRACTED.wav')
# output_path = os.path.join(settings.MEDIA_ROOT, ['/Ghosts_TEST2.wav'] )
write(output_path , 44100, scaled_e)
f = open(output_path)
output_file = File(f)
newdoc3 = Audio(user = current_user, file = output_file)
newdoc3.save()
return e
def comparison(u, d, filter_length, step_size, type):
if type is "NLMS":
y0, e0, w0 = LMSFilter(u, d, filter_length, step_size).normalized()
y1, e1, w1 = adaptfilt.nlms(u, d, filter_length, step_size)
elif type is "LMS":
y0, e0, w0 = LMSFilter(u, d, filter_length, step_size).regular()
y1, e1, w1 = adaptfilt.lms(u, d, filter_length, step_size)
else:
print("you must specify type of comparison: NLMS, LMS")
return False
fig1, axs = plt.subplots(3, 2, sharex='all')
plt.subplots_adjust(hspace=0.35)
fig1.suptitle(f"Comparison between 2 algorithms {type} filter")
x = [i for i in range(len(d))]
xt = [i for i in range(filter_length, len(d) + 1)]
axs[0, 0].plot(x, d, 'tab:blue')
axs[0, 0].set_title('desired signal')
axs[1, 0].plot(xt, y0, 'tab:purple')
axs[1, 0].set_title('my output')
axs[2, 0].plot(xt, e0, 'tab:red')
axs[2, 0].set_title('my error')
axs[0, 1].plot(x, d, 'tab:blue')
axs[0, 1].set_title('desired signal')
axs[1, 1].plot(xt, y1, 'tab:purple')
axs[1, 1].set_title('adaptfilt output')
axs[2, 1].plot(xt, e1, 'tab:red')
axs[2, 1].set_title('adaptfilt error')
for ax in axs.flat:
ax.label_outer()
ax.grid(True)
plt.savefig(f"{type}-comparison")
return True
def subtract(newdoc, newdoc2, current_user):
# newdoc_wav_path = newdoc.file
# print newdoc_wav_path
# recording_path = os.path.join(settings.MEDIA_ROOT, newdoc.file )
# print newdoc.file
# type newdoc.file
# print newdoc.file.url
# get paths for raw audio and music to be subtracted
raw_audio_filename = os.path.basename(newdoc.file.url)
original_audio_filename = os.path.basename(newdoc2.file.url)
# newdoc_url = newdoc.file.url
# print type(newdoc_url)
# Create paths to read in audio files
recording_path = os.path.join(settings.MEDIA_ROOT, raw_audio_filename)
original_audio_path = os.path.join(settings.MEDIA_ROOT,
original_audio_filename)
# recording_path = os.path.join(settings.BASE_DIR, newdoc_url)
# print recording_path
# print settings.BASE_DIR
rate, data = scipy.io.wavfile.read(recording_path)
r = data.astype(np.float64)
rate, data = scipy.io.wavfile.read(original_audio_path)
o = data.astype(np.float64)
print len(o)
print len(r)
print 'applying adaptive filt'
# Apply adaptive filter
M = 100 #8000 (takelly long time) #100 # Number of filter taps in adaptive filter
step = 0.1 # Step size
y, e, w = adf.nlms(o, r, M, step, returnCoeffs=True)
scaled_e = np.int16(e / np.max(np.abs(e)) * 32767)
# d = room_simulate.room_sim(u)
# scaled_d = np.int16(d/np.max(np.abs(d)) * 32767)
# output_path = os.path.join(settings.MEDIA_ROOT, newdoc_filename+'TEST2.wav' )
output_filename = os.path.splitext(os.path.basename(newdoc.file.url))[0]
print output_filename
output_path = os.path.join(settings.MEDIA_ROOT,
output_filename + '_SUBTRACTED.wav')
# output_path = os.path.join(settings.MEDIA_ROOT, ['/Ghosts_TEST2.wav'] )
write(output_path, 44100, scaled_e)
f = open(output_path)
output_file = File(f)
newdoc3 = Audio(user=current_user, file=output_file)
newdoc3.save()
return e
def nlms(u, d, M=100):
"""A classical adaptive filtering"""
# Apply adaptive filter
M = 100 # Number of filter taps in adaptive filter
step = 0.1 # Step size
y, e, w = adf.nlms(u, d, M, step, returnCoeffs=True)
return y, e, w
def adaptiveFiltering(ch0, ch1):
binSizeTim = np.size(ch0, 0)
biNum = np.size(ch0, 1)
M = 5
step = binSizeTim / 5.0
N = binSizeTim - M + 1
ch0out = np.zeros((N, biNum))
ch1out = np.zeros((N, biNum))
for bi in range(biNum):
ch0out[:, bi], ch1out[:,
bi], _ = adaptfilt.nlms(ch1[:, bi], ch0[:, bi],
M, step)
return ch0out, ch1out
d = np.convolve(snd, coeffs)
d = d/20.0
lis = lis/20.0
d = d[:len(lis)] # Trims sender's audio to the same length as that of the listener's in order to mix them
d = d + lis - (d*lis)/256.0 # Mix with listener's voice.
d = np.round(d,0)
# Hear how the mixed signal sounds before proceeding with the filtering.
dsound = d.astype('int16')
wavfile.write(waveout, lfs, dsound)
music = pyglet.resource.media('output.wav')
music.play()
time.sleep(len(dsound)/lfs)
# Apply adaptive filter
y, e, w = adf.nlms(snd[:len(d)], d, tap, step, returnCoeffs=True)
# The algorithm stores the processed result in the variable 'e', which is the mix of the error signal and the listener's voice.
# Hear how e sounds now. Ideally we on behalf of the sender, should hear only the listener's voice. Practically, some echo would still be present.
e = e.astype('int16')
wavfile.write('adapt.wav', lfs, e)
music = pyglet.resource.media(filtout)
music.play()
time.sleep(len(e)/lfs)
# Calculate and plot the mean square weight error
mswe = adf.mswe(w, coeffs)
plt.figure()
plt.title('Mean squared weight error')
plt.plot(mswe)
# Generate received signal d(n) using randomly chosen coefficients
coeffs = np.concatenate(([0.8], np.zeros(8), [-0.7], np.zeros(9),
[0.5], np.zeros(11), [-0.3], np.zeros(3),
[0.1], np.zeros(20), [-0.05]))
d = np.convolve(u, coeffs)
# Add background noise
v = np.random.randn(len(d)) * np.sqrt(5000)
d += v
# Apply adaptive filter
M = 100 # Number of filter taps in adaptive filter
step = 0.1 # Step size
y, e, w = adf.nlms(u, d, M, step, returnCoeffs=True)
# Calculate mean square weight error
mswe = adf.mswe(w, coeffs)
# Plot speech signals
plt.figure()
plt.title("Speech signals")
plt.plot(u, label="Emily's speech signal, u(n)")
plt.plot(d, label="Speech signal from John, d(n)")
plt.grid()
plt.legend()
plt.xlabel('Samples')
# Plot error signal - note how the measurement noise affects the error
plt.figure()
color='k',
label="music without noise")
plt.fill_between(times, s_data2[:, 0], s_data2[:, 1], label="music with noise")
plt.xlim(times[0], times[-1])
plt.grid()
plt.legend()
plt.xlabel('time')
d = ((data[:, 0]) / 2 + (data[:, 1]) / 2)
u = ((s_data2[:, 0]) / 2 + (s_data2[:, 1]) / 2)
# wavfile.write('rzeczywiscie.wav',samplerate,u.astype('int16'))
# filter
M = 1000
step = 0.1
y, e, w = adf.nlms(u, d, M, step)
dopasowanie2 = slice(0, len(y))
s_times = times[dopasowanie2]
plt.subplot(212)
plt.title("Music after filtration")
plt.fill_between(s_times, y, color='m', label="music after filtration")
plt.xlim(s_times[0], s_times[-1])
plt.grid()
plt.legend()
plt.xlabel('time')
plt.tight_layout(0, 0.1, 0)
plt.show()
dec = input('Do you want to see frequency spectum of signals? [Y/N]')
while i < 80:
music_frames = music.readframes(10000)
voice_frames = voice.readframes(10000)
u = np.fromstring(music_frames, np.int16)
u = np.float64(u)
print(i)
d = np.fromstring(voice_frames, np.int16)
d = np.float64(d)
# Apply adaptive filter
M = 20 # Number of filter taps in adaptive filter
step = 0.1 # Step size
y, e, w = adf.nlms(u, d, M, step, returnCoeffs=True)
full_e = np.concatenate([full_e, e])
full_d = np.concatenate([full_d, d])
full_u = np.concatenate([full_u, u])
i += 1
scaled = np.int16(full_e / np.max(np.abs(full_e)) * 32767)
write('splitaudio.wav', 32000, scaled)
# scaled_voice_file = wave.open("test.wav", "r")
# scaled_voice = scaled_voice_file.readframes(320000)
# print(len(scaled_voice))
def subtract(newdoc, newdoc2, current_user):
# newdoc_wav_path = newdoc.file
# print newdoc_wav_path
# recording_path = os.path.join(settings.MEDIA_ROOT, newdoc.file )
# print newdoc.file
# type newdoc.file
# print newdoc.file.url
# get paths for raw audio and music to be subtracted
raw_audio_filename = os.path.basename(newdoc.file.url)
original_audio_filename = os.path.basename(newdoc2.file.url)
# newdoc_url = newdoc.file.url
# print type(newdoc_url)
# Create paths to read in audio files
recording_path = os.path.join(settings.MEDIA_ROOT, raw_audio_filename)
original_audio_path = os.path.join(settings.MEDIA_ROOT, original_audio_filename)
# recording_path = os.path.join(settings.BASE_DIR, newdoc_url)
# print recording_path
# print settings.BASE_DIR
input_wav = wave.open (recording_path, "r")
(nchannels, sampwidth, framerate, nframes, comptype, compname) = input_wav.getparams ()
frames = input_wav.readframes (nframes * nchannels)
out = struct.unpack_from ("%dh" % nframes * nchannels, frames)
r = np.asarray(out, np.float64)
# rate, data = scipy.io.wavfile.read(recording_path)
# r = data.astype(np.float64)
input_wav = wave.open (original_audio_path, "r")
(nchannels, sampwidth, framerate, nframes, comptype, compname) = input_wav.getparams ()
frames = input_wav.readframes (nframes * nchannels)
out = struct.unpack_from ("%dh" % nframes * nchannels, frames)
o = np.asarray(out, np.float64)
# rate, data = scipy.io.wavfile.read(original_audio_path)
# o = data.astype(np.float64)
print len(o)
print len(r)
print 'applying adaptive filt'
# Apply adaptive filter
M = 100 #8000 (takelly long time) #100 # Number of filter taps in adaptive filter
step = 0.1 # Step size
y, e, w = adf.nlms(o, r, M, step, returnCoeffs=True)
scaled_e = np.int16(e/np.max(np.abs(e)) * 32767)
# d = room_simulate.room_sim(u)
# scaled_d = np.int16(d/np.max(np.abs(d)) * 32767)
# output_path = os.path.join(settings.MEDIA_ROOT, newdoc_filename+'TEST2.wav' )
output_filename = os.path.splitext(os.path.basename(newdoc.file.url))[0]
print output_filename
output_path = os.path.join(settings.MEDIA_ROOT,output_filename+'_SUBTRACTED.wav')
# output_path = os.path.join(settings.MEDIA_ROOT, ['/Ghosts_TEST2.wav'] )
output_wav = wave.open (output_path, "w")
output_wav.setparams((nchannels, sampwidth, framerate, nframes, comptype, compname))
output_wav.writeframes(scaled_e)
# write(output_path , 44100, scaled_e)
f = open(output_path)
output_file = File(f)
newdoc3 = Audio(user = current_user, file = output_file)
newdoc3.save()
return output_path
d = np.convolve(u, coeffs)
d = d / 20.0
v = v / 20.0
d = d[:len(
v
)] # Trims sender's audio to the same length as that of the listener's in order to mix them
d = d + v - (d * v) / 256.0 # Mix with listener's voice.
d = np.round(d, 0)
# Hear how the mixed signal sounds before proceeding with the filtering.
dsound = d.astype('int16')
wavfile.write(waveout, lfs, dsound)
winsound.PlaySound(waveout, winsound.SND_ALIAS)
# Apply adaptive filter
y, e, w = adf.nlms(u[:len(d)], d, M, step, returnCoeffs=True)
# The algorithm stores the processed result in the variable 'e', which is the mix of the error signal and the listener's voice.
# Hear how e sounds now. Ideally we on behalf of the sender, should hear only the listener's voice. Practically, some echo would still be present.
e = e.astype('int16')
wavfile.write(waveout, lfs, e)
winsound.PlaySound(waveout, winsound.SND_ALIAS)
# Calculate and plot the mean square weight error
mswe = adf.mswe(w, coeffs)
plt.figure()
plt.title('Mean squared weight error')
plt.plot(mswe)
plt.grid()
plt.xlabel('Samples')
def subtract(newdoc, newdoc2, current_user):
# newdoc_wav_path = newdoc.file
# print newdoc_wav_path
# recording_path = os.path.join(settings.MEDIA_ROOT, newdoc.file )
# print newdoc.file
# type newdoc.file
# print newdoc.file.url
# get paths for raw audio and music to be subtracted
raw_audio_filename = os.path.basename(newdoc.file.url)
original_audio_filename = os.path.basename(newdoc2.file.url)
# newdoc_url = newdoc.file.url
# print type(newdoc_url)
# Create paths to read in audio files
recording_path = os.path.join(settings.MEDIA_ROOT, raw_audio_filename)
original_audio_path = os.path.join(settings.MEDIA_ROOT,
original_audio_filename)
# recording_path = os.path.join(settings.BASE_DIR, newdoc_url)
# print recording_path
# print settings.BASE_DIR
input_wav = wave.open(recording_path, "r")
(nchannels, sampwidth, framerate, nframes, comptype,
compname) = input_wav.getparams()
frames = input_wav.readframes(nframes * nchannels)
out = struct.unpack_from("%dh" % nframes * nchannels, frames)
r = np.asarray(out, np.float64)
# rate, data = scipy.io.wavfile.read(recording_path)
# r = data.astype(np.float64)
input_wav = wave.open(original_audio_path, "r")
(nchannels, sampwidth, framerate, nframes, comptype,
compname) = input_wav.getparams()
frames = input_wav.readframes(nframes * nchannels)
out = struct.unpack_from("%dh" % nframes * nchannels, frames)
o = np.asarray(out, np.float64)
# rate, data = scipy.io.wavfile.read(original_audio_path)
# o = data.astype(np.float64)
print len(o)
print len(r)
print 'applying adaptive filt'
# Apply adaptive filter
M = 100 #8000 (takelly long time) #100 # Number of filter taps in adaptive filter
step = 0.1 # Step size
y, e, w = adf.nlms(o, r, M, step, returnCoeffs=True)
scaled_e = np.int16(e / np.max(np.abs(e)) * 32767)
# d = room_simulate.room_sim(u)
# scaled_d = np.int16(d/np.max(np.abs(d)) * 32767)
# output_path = os.path.join(settings.MEDIA_ROOT, newdoc_filename+'TEST2.wav' )
output_filename = os.path.splitext(os.path.basename(newdoc.file.url))[0]
print output_filename
output_path = os.path.join(settings.MEDIA_ROOT,
output_filename + '_SUBTRACTED.wav')
# output_path = os.path.join(settings.MEDIA_ROOT, ['/Ghosts_TEST2.wav'] )
output_wav = wave.open(output_path, "w")
output_wav.setparams(
(nchannels, sampwidth, framerate, nframes, comptype, compname))
output_wav.writeframes(scaled_e)
# write(output_path , 44100, scaled_e)
f = open(output_path)
output_file = File(f)
newdoc3 = Audio(user=current_user, file=output_file)
newdoc3.save()
return output_path
# Perform filtering
M = 60 # No. of taps to estimate
mu1 = 0.0008 # Step size 1 in LMS
mu2 = 0.0004 # Step size 1 in LMS
beta1 = 0.08 # Step size 2 in NLMS and AP
beta2 = 0.04 # Step size 2 in NLMS and AP
K = 3 # Projection order 1 in AP
# LMS
y_lms1, e_lms1, w_lms1 = adf.lms(u, d, M, mu1, returnCoeffs=True)
y_lms2, e_lms2, w_lms2 = adf.lms(u, d, M, mu2, returnCoeffs=True)
mswe_lms1 = adf.mswe(w_lms1, coeffs)
mswe_lms2 = adf.mswe(w_lms2, coeffs)
# NLMS
y_nlms1, e_nlms1, w_nlms1 = adf.nlms(u, d, M, beta1, returnCoeffs=True)
y_nlms2, e_nlms2, w_nlms2 = adf.nlms(u, d, M, beta2, returnCoeffs=True)
mswe_nlms1 = adf.mswe(w_nlms1, coeffs)
mswe_nlms2 = adf.mswe(w_nlms2, coeffs)
# AP
y_ap1, e_ap1, w_ap1 = adf.ap(u, d, M, beta1, K, returnCoeffs=True)
y_ap2, e_ap2, w_ap2 = adf.ap(u, d, M, beta2, K, returnCoeffs=True)
mswe_ap1 = adf.mswe(w_ap1, coeffs)
mswe_ap2 = adf.mswe(w_ap2, coeffs)
# Plot results
plt.figure()
plt.title('Convergence comparison of different adaptive filtering algorithms')
plt.plot(mswe_lms1, 'b', label='LMS with stepsize=%.4f' % mu1)
plt.plot(mswe_lms2, 'b--', label='LMS with stepsize=%.4f' % mu2)
def Removing_MotionArtifact(self, Int_FilterLength, Array_Signal):
_,_,Array_NoiseReference = self.ConvertInvertFourier(Array_Signal=Array_Signal)
Flt_StepSize = 0.1
Array_NoiseEst, Array_NoiseRemoved, Array_FilterCoefs = adaptfilt.nlms(u=Array_NoiseReference,d=Array_Signal, M=Int_FilterLength, step=Flt_StepSize)
return Array_NoiseRemoved
def estimate(y,y1,ref1):
data2=[]
sas=[]
sdelay=[]
##nT=5 ##samples in one time period
Fs=62.5 #500#sampling frequency in Hz, sampling interval=2ms
## window=int(Fs) ##samples per Beta
window =30
order=10
boundary=window-1 ##correction for dividion
length=len(y)
wave=thinkdsp.Wave(ys=y,framerate=Fs)
spectrum=wave.make_spectrum()
spectrum_heart=wave.make_spectrum()
spectrum_resp=wave.make_spectrum()
fft_mag=list(np.absolute(spectrum.hs))
fft_length= len(fft_mag)
spectrum_heart.high_pass(cutoff=0.5,factor=0.001)
spectrum_heart.low_pass(cutoff=4,factor=0.001)
fft_heart=list(np.absolute(spectrum_heart.hs))
spectrum_resp.high_pass(cutoff=0.15,factor=0)
spectrum_resp.low_pass(cutoff=0.4,factor=0)
fft_resp=list(np.absolute(spectrum_resp.hs))
max_fft_heart=max(fft_heart)
heart_sample=fft_heart.index(max_fft_heart)
fund_freq=heart_sample*Fs/length*60
## fund_freq-=0.2*fund_freq
nT=int(fund_freq/60*Fs)
max_fft_resp=max(fft_resp)
resp_sample=fft_resp.index(max_fft_resp)
rr=resp_sample*Fs/length*60
## FIRST ADAPTIVE FILTER
x1,e1,w1=adf.nlms(ref1,y,order,1)
x2,e2,w2=adf.nlms(ref1,y1,order,1)
## calculating fundamental time period of input signal x1
length=len(x1)
wave=thinkdsp.Wave(ys=x1,framerate=Fs)
spectrum=wave.make_spectrum()
spectrum_heart=wave.make_spectrum()
spectrum_resp=wave.make_spectrum()
fft_mag=list(np.absolute(spectrum.hs))
fft_length= len(fft_mag)
spectrum_heart.high_pass(cutoff=0.5,factor=0.001)
spectrum_heart.low_pass(cutoff=4,factor=0.001)
fft_heart=list(np.absolute(spectrum_heart.hs))
spectrum_resp.high_pass(cutoff=0.15,factor=0)
spectrum_resp.low_pass(cutoff=0.4,factor=0)
fft_resp=list(np.absolute(spectrum_resp.hs))
max_fft_heart=max(fft_heart)
heart_sample=fft_heart.index(max_fft_heart)
fund_freq=heart_sample*Fs/length*60
nT=int(fund_freq/60*Fs)
max_fft_resp=max(fft_resp)
resp_sample=fft_resp.index(max_fft_resp)
rr=resp_sample*Fs/length*60
##Second Filter algorithm
ex1s=.0
ex2s=.0
ex1x2=.0
esas=.0
esdx1=.0
esdx2=.0
ess=.0
esds=.0
essd=.0
rv=.8
alpha=.0
beta=[]
length=length-length%window
## calculating first/initial rv
for i in range(0,nT):
sas.append(x1[i]-rv*x2[i]) ##getting values of sas till nT for sdelay
ex1s=(ex1s*i+x1[i]**2)/(i+1)
ex2s=(ex2s*i+x2[i]**2)/(i+1)
ess=(ess*i+sas[i]**2)/(i+1)
esdx1=(esdx1*i+x1[i]*sas[i])/(i+1)
esdx2=(esdx2*i+x2[i]*sas[i])/(i+1)
ex1x2=(ex1x2*i+x1[i]*x2[i])/(i+1)
if((ex2s*esdx1 - ex1x2*esdx2)):
rv = (ex1x2*esdx1 - ex1s*esdx2)/(ex2s*esdx1 - ex1x2*esdx2)
## print esdx1,ex2s,esdx1,ex1x2,esdx2
## print (ex2s*esdx1 - ex1x2*esdx2),rv
##calculating rv,alpha and updating beta every 20 samples
for i in range(nT,length-window):
for j in range(0,window):
sas.append(x1[i]-rv*x2[i])
ex1s=(ex1s*j+x1[i]**2)/(j+1)
ex2s=(ex2s*j+x2[i]**2)/(j+1)
ess=(ess*j+sas[i]**2)/(j+1)
esdx1=(esdx1*j+x1[i]*sas[i-nT])/(j+1)
esdx2=(esdx2*j+x2[i]*sas[i-nT])/(j+1)
ex1x2=(ex1x2*j+x1[i]*x2[i])/(j+1)
## esds=(esds*j+sas[i-nT]**2)/(j+1)
## essd=(essd*j+sas[i]*sas[i-nT])/(j+1)
i=i+1
if((ex2s*esdx1 - ex1x2*esdx2)):
rv = (ex1x2*esdx1 - ex1s*esdx2)/(ex2s*esdx1 - ex1x2*esdx2)
## b = essd/esds
alpha = (ex1s-rv*ex1x2)/ess
for j in range(0,window):
beta.append((rv*alpha)/(1-alpha))
## print beta
i=i-1
venous_ref=[]
for i in range(0,length-nT):
venous_ref.append(x1[nT+i]- beta[i]*x2[nT+i])
xf1,ef1,wf1=adf.nlms(venous_ref,x2[nT:],2*order,1)
data2= data2 + list(xf1)
length=len(data2)
wave=thinkdsp.Wave(ys=data2,framerate=Fs)
spectrum=wave.make_spectrum()
spectrum_heart=wave.make_spectrum()
spectrum_resp=wave.make_spectrum()
fft_mag=list(np.absolute(spectrum.hs))
fft_length= len(fft_mag)
spectrum_heart.high_pass(cutoff=0.7,factor=0.001)
spectrum_heart.low_pass(cutoff=4,factor=0.001)
fft_heart=list(np.absolute(spectrum_heart.hs))
spectrum_resp.high_pass(cutoff=0.15,factor=0)
spectrum_resp.low_pass(cutoff=0.4,factor=0)
fft_resp=list(np.absolute(spectrum_resp.hs))
max_fft_heart=max(fft_heart)
heart_sample=fft_heart.index(max_fft_heart)
fund_freq=heart_sample*Fs/length*60
nT=int(fund_freq/60*Fs)
max_fft_resp=max(fft_resp)
resp_sample=fft_resp.index(max_fft_resp)
rr=resp_sample*Fs/length*60
## rr=10.2437645098
print "Heart rate [normal 60-100] : ", fund_freq,'BPM'
print "Respiration Rate [normal 10-20] : ", rr, "RPM"
return fund_freq
def adaptive_noise_cancellation(noise, signal, filter_size=64, learning_rate=0.1, normalized=True):
if normalized:
y, e, w = adaptfilt.nlms(noise, noise + signal, filter_size, learning_rate)
else:
y, e, w = adaptfilt.lms(noise, noise + signal, filter_size, learning_rate)
return numpy.concatenate((e[0:filter_size - 1], e))
for value in valuesfile:
#print value
if value is '':
break
a= value.split(' ')
if a[0] is '' or a[1] is '':break
sensor=float(a[0])
sensor1=float(a[1])
ref.append(sensor1-sensor)
data.append(sensor)
data1.append(sensor1)
length=len(data)
print "Length:", length
x1,e1,w1=adf.nlms(ref,data,5,1)
x2,e2,w2=adf.nlms(ref,data1,5,1)
ex1s=.0
ex2s=.0
ex1x2=.0
esas=.0
esdx1=.0
esdx2=.0
ess=.0
esds=.0
essd=.0
rv=0.5
alpha=.0
beta=[]
for i in range(0,20):
sas.append(data[i]-rv*data1[i])
def subtract(newdoc, newdoc2, current_user):
# Connect to Amazon S3
conn = S3Connection(settings.AWS_ACCESS_KEY_ID, settings.AWS_SECRET_ACCESS_KEY)
bucket = conn.get_bucket(settings.AWS_STORAGE_BUCKET_NAME)
k = Key(bucket)
# get filenames of raw audio and music to be subtracted
raw_audio_filename = os.path.basename(newdoc.file.url)
original_audio_filename = os.path.basename(newdoc2.file.url)
# Create upload paths to S3 for mixed audio
# raw_audio_s3_path = 'https://s3-us-west-2.amazonaws.com/audiofiles1234/'+raw_audio_filename
# original_audio_s3_path = 'https://s3-us-west-2.amazonaws.com/audiofiles1234/'+original_audio_filename
# Upload raw input to S3
k.key = raw_audio_filename#testes#output_filename # for now, key for bucket is filename (might want to change this in case of duplicates)
k.set_contents_from_file(newdoc.file, rewind=True)
k.make_public()
# Upload original audio to S3
k.key = original_audio_filename#testes#output_filename # for now, key for bucket is filename (might want to change this in case of duplicates)
k.set_contents_from_file(newdoc2.file, rewind=True)
k.make_public()
# Load file from AWS S3
url = "https://s3-us-west-2.amazonaws.com/audiofiles1234/Ghosts_echoed_RIR_noise_testfile.wav"
recorded_audio_file = urllib2.urlopen(url)
input_wav = wave.open (recorded_audio_file, "r")
# input_wav = wave.open (newdoc.file, "r") #attempt (failed?) at opening file directly from filefield object
# input_wav = wave.open (recording_path, "r") # READING FROM DISK METHOD (PUT BACK IN?)
# Read wav file into numpy array
(nchannels, sampwidth, framerate, nframes, comptype, compname) = input_wav.getparams ()
frames = input_wav.readframes (nframes * nchannels)
out = struct.unpack_from ("%dh" % nframes * nchannels, frames)
r = np.asarray(out, np.float64)
# Download file from AWS S3
url = "https://s3-us-west-2.amazonaws.com/audiofiles1234/GhostsNStuff_mono_4s.wav"
original_audio_file = urllib2.urlopen(url)
input_wav = wave.open (original_audio_file, "r")
# input_wav = wave.open (original_audio_path, "r") # READING FROM DISK METHOD (PUT BACK IN?)
# Read wav file into numpy array
(nchannels, sampwidth, framerate, nframes, comptype, compname) = input_wav.getparams ()
frames = input_wav.readframes (nframes * nchannels)
out = struct.unpack_from ("%dh" % nframes * nchannels, frames)
o = np.asarray(out, np.float64)
# print len(o)
# print len(r)
# print 'applying adaptive filt'
# Apply adaptive filter
M = 100 #8000 (takelly long time) #100 # Number of filter taps in adaptive filter
step = 0.1 # Step size
y, e, w = adf.nlms(o, r, M, step, returnCoeffs=True)
scaled_e = np.int16(e/np.max(np.abs(e)) * 32767)
# Create output filename
output_filename = os.path.splitext(os.path.basename(newdoc.file.url))[0]
print output_filename
output_filename_s3 = output_filename+'_SUBTRACTED_TEST.wav'
# output_path = os.path.join(settings.MEDIA_ROOT,output_filename+'_SUBTRACTED.wav')
output_path = 'https://s3-us-west-2.amazonaws.com/audiofiles1234/'+output_filename_s3
# Write output wav to S3
output_file = open(output_filename_s3, "w+") # create pointer to file object
output_wav = wave.open (output_file, "w")
output_wav.setparams((nchannels, sampwidth, framerate, nframes, comptype, compname))
output_wav.writeframes(scaled_e)
# output_wav.seek(0)
# output_filename_s3 = output_filename+'TEST'
k.key = output_filename_s3#testes#output_filename # for now, key for bucket is filename (might want to change this in case of duplicates)
k.set_contents_from_file(output_file, rewind=True)
k.make_public()
# write(output_path , 44100, scaled_e)
# f = open(output_path)
# output_file = File(f)
# newdoc3 = Audio(user = current_user, file = output_file)
# newdoc3.save()
return output_path
