def test_002_square2_ff (self):
src_data0 = (
0.01+0.11j,
0.02+0.22j,
0.03+0.33j,
0.04+0.44j,
0.05+0.55j,
0.06+0.66j,
0.07+0.77j,
0.08+0.88j,
0.09+0.99j
)
src_coeff = (0.101, 0.102, 0.103, 0.104, 0.105)
scale = 15
expected_result = (245, 320, 395, 470, 445, 400, 334, 246, 135)
src0 = gr.vector_source_c (src_data0)
dsp.init()
# dsp.fir_ccf (
# coefficients,
# Q formatted can be between Q0 -> Q14
# scaling factor = [0, 15]
# interpolation factor
# currently not implemented "properly" a.k.a don't use just yet
# input signature
# 0 = normalized input
# 1 = scaled by a factor equal "scale"
# output signature
# 0 = normalized input
# 1 = scaled by a factor equal "scale"
# initialize DSP (set to 0 use dsp.init() instead
# 1 = yes
# 0 = no
# block ID
# 1 or 0 to uniquly identify block instance for now there's support for only two blocks
ccf_filt = dsp.fir_ccf (src_coeff, scale, 1, 0, 0, 0, 0)
# gccf = gr.interp_fir_filter_ccf (1, src_coeff)
dst0 = gr.vector_sink_c ()
dst1 = gr.vector_sink_c ()
self.tb.connect ((src0, 0), ccf_filt)
self.tb.connect (ccf_filt, dst0)
self.tb.run ()
result_data0 = dst0.data ()
result_data1 = dst1.data ()
print "Result DSP "
dsp.clean()
def __init__(self):
gr.top_block.__init__(self)
usage="%prog: [options] output_filename"
parser = OptionParser(option_class=eng_option, usage=usage)
parser.add_option("-I", "--audio-input", type="string", default="",
help="pcm input device name. E.g., hw:0,0 or /dev/dsp")
parser.add_option("-r", "--sample-rate", type="eng_float", default=48000,
help="set sample rate to RATE (48000)")
parser.add_option("-N", "--nsamples", type="eng_float", default=None,
help="number of samples to collect [default=+inf]")
(options, args) = parser.parse_args ()
if len(args) != 1:
parser.print_help()
raise SystemExit, 1
#filename = args[0]
sample_rate = int(options.sample_rate)
audio_coeffs = (
0.00058729130373770002,
0.0016584444738215582,
0.0015819269921330031,
0.0014607862142637573,
0.00020681278261230754,
-0.0013001097961560814,
-0.00249802658603143,
-0.0024276134129972843,
-0.00083069749014258953,
0.0017562878158492619,
0.003963761120687582,
0.0043075911442784871,
0.0020710872871114866,
-0.0020172640629268932,
-0.005882026963765212,
-0.0070692053073845166,
-0.0041954626649490937,
0.0019311082705710714,
0.0082980827342646387,
0.011045923787287403,
0.0076530405054369872,
-0.0012102332109476402,
-0.011372099802214802,
-0.016910189774436514,
-0.013347352799620162,
-0.00068013535845177706,
0.015578754320259895,
0.026379517186832846,
0.023618496101893545,
0.0051085800414948012,
-0.022608534445133374,
-0.045529642916534545,
-0.047580556152787695,
-0.018048092177406189,
0.042354392363985506,
0.11988807809069109,
0.19189052073753335,
0.2351677633079737,
0.2351677633079737,
0.19189052073753335,
0.11988807809069109,
0.042354392363985506,
-0.018048092177406189,
-0.047580556152787695,
-0.045529642916534545,
-0.022608534445133374,
0.0051085800414948012,
0.023618496101893545,
0.026379517186832846,
0.015578754320259895,
-0.00068013535845177706,
-0.013347352799620162,
-0.016910189774436514,
-0.011372099802214802,
-0.0012102332109476402,
0.0076530405054369872,
0.011045923787287403,
0.0082980827342646387,
0.0019311082705710714,
-0.0041954626649490937,
-0.0070692053073845166,
-0.005882026963765212,
-0.0020172640629268932,
0.0020710872871114866,
0.0043075911442784871,
0.003963761120687582,
0.0017562878158492619,
-0.00083069749014258953,
-0.0024276134129972843,
-0.00249802658603143,
-0.0013001097961560814,
0.00020681278261230754,
0.0014607862142637573,
0.0015819269921330031,
0.0016584444738215582,
0.00058729130373770002)
# initializes the DSP
dsp.init()
src = audio.source (sample_rate, options.audio_input)
#dst = gr.file_sink (gr.sizeof_float, filename)
dst = audio.sink (sample_rate, options.audio_output)
audio_filter = dsp.fir_fff(1, audio_coeffs)
# deletes DSP related buffers and resources
# dsp.clean()
if options.nsamples is None:
self.connect((src, 0), audio_filter, dst)
else:
head = gr.head(gr.sizeof_float, int(options.nsamples))
self.connect((src, 0), head, audio_filter, dst)