コード例 #1
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alpha = 0.9 # Koeffizient
N_sim = 200 # Anzahl Simulationsschritte

x = zeros(N_sim); x[0] = 1.0 # x ist dirac-Stoß mit Gew. 1.0

 # Koeffizientenquantisierung:
q_coeff = {'QI':0,'QF':3,'quant':'round','ovfl':'wrap'}
 # Quantisierung nach Summation im Filter:
#q_accu = {'QI':0,'QF':4,'quant':'fix','ovfl':'sat'} # keine Grenzzyklen
#q_accu = {'Q':1.4,'quant':'fix','ovfl':'wrap'}  # große Grenzzyklen bei QI = 0
q_accu = {'QI':0,'QF':4,'quant':'fix','ovfl':'wrap'} # kleine Grenzzyklen

# Keine Quantisierung ->  Werte für I, F beliebig
q_ideal = {'QI':0,'QF':0,'quant':'none','ovfl':'none'}

fx_coeff = fx.Fixed(q_coeff) # Fixpoint Object mit Parametern "q_coeff"
alpha_q  = fx_coeff.fix(alpha) # berechne  quantisierten Koeffizienten

fx_IIR_id = fx.Fixed(q_ideal) # Fixpoint-Objekt ohne Quantisierung und Overflow 
fx_IIR = fx.Fixed(q_accu) # Fixpoint-Objekt mit Parametern "q_accu"

n = arange(N_sim)
t1 = time.time()

y = IIR2(fx_IIR_id, x, alpha_q) # ohne Quantisierung
yq = IIR2(fx_IIR, x, alpha_q)
#
#yq = IIR1(fx_IIR, x, alpha_q)
t2 = time.time()
print('Time = ', t2 - t1)
print('Number of overflows = ', fx_IIR.N_over)
コード例 #2
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                channels=n_chan,
                rate=rate_in,
                output=True)

# Define quantization mode and create a quantization instance for each channel
# quantize with just a few bits:
q_obj = {
    'Q': -2.15,
    'quant': 'round',
    'ovfl': 'sat'
}  # try 'quant':'round', 'ovfl':'sat'

# Overflows QI = -1 means the MSB is 2^{-1} = 0.5
#q_obj = {'Q':-1.15,'quant':'fix','ovfl':'wrap'} # try  'ovfl':'sat'

fx_Q_l = fx.Fixed(q_obj)
fx_Q_r = fx.Fixed(q_obj)

# initialize arrays for audio samples
samples_in = zeros(CHUNK * 2, dtype=np_type)  # stereo int16
samples_out = zeros(CHUNK * 2, dtype=float)  # stereo float
samples_l = samples_r = zeros(CHUNK, dtype=np_type)  # separate channels int16

data_out = 'start'

while data_out:

    # read CHUNK stereo samples to string and convert to numpy array.
    # R / L samples are interleaved, each sample is 16 bit wide (dtype = np.int16)
    samples_in = np.fromstring(wf.readframes(CHUNK), dtype=np_type)
コード例 #3
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import matplotlib.pyplot as plt
from matplotlib.pyplot import (figure, plot, stem, grid, xlabel, ylabel,
                               subplot, title, clf, xlim, ylim)

import sys

sys.path.append('..')
import dsp_fpga_fix_lib as fx

b = [0.01623, 0, -0.06871, 0, 0.30399, 0.5, 0.30399, 0, -0.06871, 0, 0.01623]
#
q_obj7 = {'QI': 0, 'QF': 7, 'quant': 'floor', 'ovfl': 'none'}
q_obj17 = {'QI': 0, 'QF': 17, 'quant': 'floor', 'ovfl': 'none'}
#
fx_7 = fx.Fixed(q_obj7)
fx_17 = fx.Fixed(q_obj17)

bq7 = fx_7.fix(b)  # quantize a
bq17 = fx_17.fix(b)
title_str = "    b    |  bq(0.17)  | eps(0.17)|  bq(0.7)  | eps(0.7) "
print(title_str, "\n", "-" * len(title_str))
for i in range(len(b)):
    print("{0:8.5f} | {1:10.6g} | {2:9.2E}| {3:9.6f} | {4:9.2E}".format(
        b[i], bq17[i], b[i] - bq17[i], bq7[i], b[i] - bq7[i]))

w, H_id = sig.freqz(b)
w, H_q7 = sig.freqz(bq7)
F = w / (2 * pi)
fig = figure(1)
ax = fig.add_subplot(111)
コード例 #4
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x = a_sig * sin(2 * pi * t * fsig + 3)  # test signal with some starting phase
#x = a_sig * sin(2*pi*t*fsig+1) + 0.0001*sin(2*pi*t*2*fsig) #Two-Tone test signal
##################################################################
# Distort signal with second and third order non-linearities
# and calculate the resulting signal frequencies (with folding)
y = x + k2 * x * x - k3 * x**3
N_per = fold(N_per)
f_sig_fold = N_per / N_FFT * f_S
N_per2 = fold(N_per * 2)
f_sig2_fold = N_per2 / N_FFT * f_S
N_per3 = fold(N_per * 3)
f_sig3_fold = N_per3 / N_FFT * f_S
##################################################################
# Define and instantiate quantizer object
q_adc = {'QI': 0, 'QF': 13, 'quant': 'round', 'ovfl': 'sat'}
adc = fx.Fixed(q_adc)  # adc quantizer instance

# Quantize input signal (ADC) with or without dither
#y += adc.LSB/4. * np.random.rand(len(y)) # add dither, -1/4 LSB < eps_N < 1/4 LSB
yq = adc.fix(y)  # quantize with the parameters set by q_adc

if adc.N_over: print('%d Overflows in ADC:  ' % adc.N_over)

#################################################################
#
# Magnitude of DFT over f = 0 ... f_S/2, scaled mit 2 / N_FFT / sqrt(2)
#
# This yields correct scaling for:
# - SINGLE-SIDED spectra (0 ... f_S/2 ) : factor of 2
# - PERIODIC signalx (factor 1/N_FFT)
# - RMS values (factor 1 / sqrt(2) )
コード例 #5
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ファイル: NOI_intro_py.py プロジェクト: johndevitis/dsp_fpga
NOISE_LABEL = True  # add second axis for noise power density
NFFT = 2000
f_a = 2
N_per = 7
T_mess = N_per / f_a
T_S = T_mess / NFFT
t = linspace(0, T_mess, NFFT, endpoint=False)
a = 1.1 * sin(2 * pi * f_a * t)
#
q_obj = {
    'QI': 1,
    'QF': 3,
    'quant': 'round',
    'ovfl': 'wrap'
}  # versuchen Sie auch 'floor' / 'fix' und 'sat'
fx_a = fx.Fixed(q_obj)
aq = fx_a.fix(a)

print('Anzahl der Überläufe = ', fx_a.N_over)
#
figure(1)
title('Quantisiertes Sinussignal und Quantisierungsfehler')
plot(t, a, label=r'$a(t)$')
plt.step(t, aq, where='post', label=r'$a_Q(t)$')
plot(t, a - aq, label=r'$a(t) - a_Q(t)$')
plt.legend(fontsize=14)
xlabel(r'$t \; \mathrm{/ \; s} \; \rightarrow$')
ylabel(r'$a \rightarrow$')
A_max = 2**q_obj['QI'] - 2**-q_obj['QF']
plt.axhline(y=A_max, linestyle='--', color='k')
plt.axhline(y=-A_max, linestyle='--', color='k')
コード例 #6
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N = 10000
f_a = 1
t = linspace(0, 1, N, endpoint=False)
a = 1.1 * sin(2 * pi * f_a * t)
x = linspace(-2, 2, N, endpoint=False)
#
#q_obj = (0, 4, 'round', 'sat') # try 'round' ; 'sat'
q_obj = {
    'QI': 0,
    'QF': 4,
    'quant': 'floor',
    'ovfl': 'wrap'
}  # try 'round' ; 'sat'
#q_obj = {'OVFL':'sat'}
fx_a = fx.Fixed(q_obj)
fx_x = fx.Fixed(q_obj)

t_cpu = time.clock()

aq = fx_a.fix(a)  # quantize a
xq = fx_x.fix(x)  # quantize x
print('Anzahl der Überläufe = ', fx_x.N_over)
print('Total CPU time: %.5g ms' % ((time.clock() - t_cpu) * 1000))

#
figure(1)
title('Quantisiertes Sinussignal')
plot(t, a, label=r'$a(t)$')
plt.step(t, aq, where='post', label=r'$a_Q(t)$')
plot(t, a - aq, label=r'$a(t) - a_Q(t)$')
コード例 #7
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# Variante 3
# g = [0.16, 1/2.8, 1.4]
# a = [1, 0.96]
#
#b = ones(1,8)
#b = sig.remez(10, [0, 0.15, 0.35, 0.5],[1, 0],[1, 1]) # halfband-filter
b = [-0.123, 0, 0.379, 0.5, 0.379, 0, -0.123]  # halfband-filter
b = [0.01623, 0, -0.06871, 0, 0.30399, 0.5, 0.30399, 0, -0.06871, 0, 0.01623]
#b = [0.99, 0.]
#b=1
#a = array([1,])#,zeros(len(b)-1)]
a = 1.0
g = 1.0
##################################################################
# Instanziiere Quantizerobjekte und Fxp-Filter
adc = fx.Fixed(q_adc)  # adc quantizer instance
fil_ma_q = fx.FIX_filt_MA(q_mul, q_acc)  # fxpoint filter instance
dac = fx.Fixed(q_dac)  # dac quantizer instance
coeffq = fx.Fixed(q_coeff)

### quantize coefficients
aq = coeffq.fix(a)
bq = coeffq.fix(b)
gq = coeffq.fix(g)

if coeffq.N_over > 0:
    print("Coefficient overflows:", coeffq.N_over)

### Berechnung der Übertragungsfunktion mit idealen und quantisierten Koeff.
# idealer Betragsgang
[w, Hf] = sig.freqz(b, a, worN=int(N_FFT / 2))
コード例 #8
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ファイル: NOI_DFT.py プロジェクト: johndevitis/dsp_fpga
N_per = 53  # Anzahl der Perioden des Signals im DFT-Rechteckfenster
# N_per muss kleiner als N_FFT / 2 sein (sonst gibt's Aliasing!)
# und am besten eine Primzahl, z.B. 63, 131, 251, 509, 1021, ...
# N_per entspricht außerdem der Signalfrequenz!
a_sig = 1  # Signalamplitude
fsig = f_S * N_per / N_FFT
x = a_sig * sin(2 * pi * t * fsig + 1)  # + 2^-18 #Testsignal mit Startphase
#x = a_sig * sin(2*pi*t*fsig+1) + 0.001*sin(2*pi*t*251) #Two-Tone Testsignal

#########################################################################
# Definiere Quantizer-Objekte:
#

q_adc = {'QI': 1, 'QF': 15, 'quant': 'round', 'ovfl': 'sat'}

fx_adc = fx.Fixed(q_adc)  # instanstiiere Quantizer

##################################################################
### Input Quantization (ADC, q_adc)

if DITHER:
    ### add uniform dithering noise, -1/4 LSB < eps_N < 1/4 LSB
    x += 2**(-q_adc['QF'] - 2) * np.random.rand(len(x))

xq = fx_adc.fix(x)  # quantisiere

if fx_adc.N_over: print('Anzahl der Überläufe = ', fx_adc.N_over)

#################################################################
#
# FFT über N_FFT Datenpunkte skaliert mit 1/N_FFT über Freq. von 0 ... f_S