def __init__(self, wf=None): """ wf = None : mic """ # Visualizations of result self.vis = Visualizations() # network parameter self.numCols = 2**9 # 2**9 = 512 sparsity = 0.10 self.numInput = int(self.numCols * sparsity) # encoder of audiostream self.e = BitmapArrayEncoder(self.numCols, 1) # setting audio p = pyaudio.PyAudio() if wf == None: self.wf = None channels = 1 rate = 44100 # sampling周波数: 1秒間に44100回 secToRecord = .1 # self.buffersize = 2**12 self.buffersToRecord=int(rate*secToRecord/self.buffersize) if not self.buffersToRecord: self.buffersToRecord = 1 audio_format = pyaudio.paInt32 else: self.printWaveInfo(wf) channels = wf.getnchannels() self.wf = wf rate = wf.getframerate() secToRecord = wf.getsampwidth() self.buffersize = 1024 self.buffersToRecord=int(rate*secToRecord/self.buffersize) if not self.buffersToRecord: self.buffersToRecord = 1 audio_format = p.get_format_from_width(secToRecord) self.inStream = p.open( format=audio_format, channels=channels, rate=rate, input=True, output=True, frames_per_buffer=self.buffersize) self.audio = numpy.empty((self.buffersToRecord*self.buffersize), dtype="uint32") # filters in Hertz # max lowHertz = (buffersize / 2-1) * rate / buffersize highHertz = 500 lowHertz = 10000 # Convert filters from Hertz to bins self.highpass = max(int(highHertz * self.buffersize /rate), 1) self.lowpass = min(int(lowHertz * self.buffersize / rate), self.buffersize/2 -1) # Temporal Pooler self.tp = TP( numberOfCols = self.numCols, cellsPerColumn = 4, initialPerm = 0.5, connectedPerm = 0.5, minThreshold = 10, newSynapseCount = 10, permanenceInc = 0.1, permanenceDec = 0.07, activationThreshold = 8, globalDecay = 0.02, burnIn = 2, checkSynapseConsistency = False, pamLength = 100 ) print("Number of columns: ", str(self.numCols)) print("Max size of input: ", str(self.numInput)) print("Sampling rate(Hz): ", str(rate)) print("Passband filter(Hz): ", str(highHertz), " - ", str(lowHertz)) print("Passband filter(bin):", str(self.highpass), " - ", str(self.lowpass)) print("Bin difference: ", str(self.lowpass - self.highpass)) print("Buffersize: ", str(self.buffersize))
class AudioStream: def printWaveInfo(self, wf): """ WAVEファイルの情報を取得 """ print() print("チャンネル数:", wf.getnchannels() ) print("サンプル幅:", wf.getsampwidth() ) print("サンプリング周波数:", wf.getframerate() ) print("フレーム数:", wf.getnframes() ) print("パラメータ:", wf.getparams() ) print("長さ(秒):", float(wf.getnframes()) / wf.getframerate() ) print() def __init__(self, wf=None): """ wf = None : mic """ # Visualizations of result self.vis = Visualizations() # network parameter self.numCols = 2**9 # 2**9 = 512 sparsity = 0.10 self.numInput = int(self.numCols * sparsity) # encoder of audiostream self.e = BitmapArrayEncoder(self.numCols, 1) # setting audio p = pyaudio.PyAudio() if wf == None: self.wf = None channels = 1 rate = 44100 # sampling周波数: 1秒間に44100回 secToRecord = .1 # self.buffersize = 2**12 self.buffersToRecord=int(rate*secToRecord/self.buffersize) if not self.buffersToRecord: self.buffersToRecord = 1 audio_format = pyaudio.paInt32 else: self.printWaveInfo(wf) channels = wf.getnchannels() self.wf = wf rate = wf.getframerate() secToRecord = wf.getsampwidth() self.buffersize = 1024 self.buffersToRecord=int(rate*secToRecord/self.buffersize) if not self.buffersToRecord: self.buffersToRecord = 1 audio_format = p.get_format_from_width(secToRecord) self.inStream = p.open( format=audio_format, channels=channels, rate=rate, input=True, output=True, frames_per_buffer=self.buffersize) self.audio = numpy.empty((self.buffersToRecord*self.buffersize), dtype="uint32") # filters in Hertz # max lowHertz = (buffersize / 2-1) * rate / buffersize highHertz = 500 lowHertz = 10000 # Convert filters from Hertz to bins self.highpass = max(int(highHertz * self.buffersize /rate), 1) self.lowpass = min(int(lowHertz * self.buffersize / rate), self.buffersize/2 -1) # Temporal Pooler self.tp = TP( numberOfCols = self.numCols, cellsPerColumn = 4, initialPerm = 0.5, connectedPerm = 0.5, minThreshold = 10, newSynapseCount = 10, permanenceInc = 0.1, permanenceDec = 0.07, activationThreshold = 8, globalDecay = 0.02, burnIn = 2, checkSynapseConsistency = False, pamLength = 100 ) print("Number of columns: ", str(self.numCols)) print("Max size of input: ", str(self.numInput)) print("Sampling rate(Hz): ", str(rate)) print("Passband filter(Hz): ", str(highHertz), " - ", str(lowHertz)) print("Passband filter(bin):", str(self.highpass), " - ", str(self.lowpass)) print("Bin difference: ", str(self.lowpass - self.highpass)) print("Buffersize: ", str(self.buffersize)) # # setup plot # plt.ion() # bin = range(self.highpass, self.lowpass) # xs = numpy.arange(len(bin)*rate/self.buffersize + highHertz) # self.freqPlot = plt.plot(xs, xs)[0] # plt.ylim(0, 10**12) def plotPerformance(self, values, window=1000): plt.clf() plt.plot(values[-window:]) plt.gcf().canvas.draw() # Without the next line, the pyplot plot won't actually show up. plt.pause(0.001) def playAudio(self): """ 指定されているwaveを再生 同時に波形をplot """ chunk = 22050 # 音源が0.5秒毎に切り替わっていたため. data = self.wf.readframes(chunk) #plt.ion() #data_list = [] predictedInt = None plt.figure(figsize=(15, 5)) while data != '': dat = numpy.fromstring(data, dtype = "uint32") #print(dat.shape, dat) # plot data_list = dat.tolist() self.plotPerformance(data_list, window=500) # plt.plot(dat) # plt.show(block = False) # plt.draw() # 音ならす. self.inStream.write(data) # sampling値 -> SDR actualInt, actual = self.encoder(data_list) # actualInt, predictedInt 比較 if not predictedInt == None: compare = self.vis.compareArray(actualInt, predictedInt) print("." . join(compare) ) anomaly = self.vis.calcAnomaly(actualInt, predictedInt) print(self.vis.hashtagAnomaly(anomaly) ) # TP predict predictedInt = self.tp_learn_and_predict(actual) # 次のデータ data = self.wf.readframes(chunk) self.inStream.close() p.terminate() def tp_learn_and_predict(self, data): self.tp.compute(data, enableLearn = True, computeInfOutput = True) predictedInt = self.tp.getPredictedState().max(axis=1) return predictedInt def encoder(self, data): # sampling 値 -> 周波数成分 ys = self.fft(data, self.highpass, self.lowpass) # 1. 強い周波数成分の上位numInputのindexを取得する. # 2. 数字のレンジをnumColsに合わせる. # 3. uniqにする. (いるの?) fs = numpy.sort(ys.argsort()[-self.numInput:]) rfs = fs.astype(numpy.float32) / (self.lowpass - self.highpass) * self.numCols ufs = numpy.unique(rfs) # encode actualInt = self.e.encode(ufs) actual = actualInt.astype(numpy.float32) return actualInt, actual def fft(self, audio, highpass, lowpass): left, right = numpy.split(numpy.abs(numpy.fft.fft(audio)), 2) output = left[highpass:lowpass] return output def formatRow(self, x): s = '' for c in range(len(x)): if c > 0 and c % 10 == 0: s += ' ' s += str(x[c]) s += ' ' return s def getAudioString(self): if self.wf == None: print(self.buffersToRecord) for i in range(self.buffersToRecord): try: audioString = self.inStream.read(self.buffersize) except IOError: print("Overflow error from 'audiostring = inStream.read(buffersize)'. Try decreasing buffersize.") quit() self.audio[i*self.buffersize:(i+1)*self.buffersize] = numpy.fromstring(audioString, dtype = "uint32") else: for i in range(self.buffersToRecord): audioString = self.wf.readframes(self.buffersize) self.audio[i*self.buffersize:(i+1)*self.buffersize] = numpy.fromstring(audioString, dtype = "uint32") def plotWave(self): self.getAudioString() print(self.audio) plt.plot(audiostream.audio[0:1000]) plt.show()
def __init__(self): """ Instantiate temporal pooler, encoder, audio sampler, filter, & freq plot """ self.vis = Visualizations() """ The number of columns in the input and therefore the TP 2**9 = 512 Trial and error pulled that out numCols should be tested during benchmarking """ self.numCols = 2 ** 9 sparsity = 0.10 self.numInput = int(self.numCols * sparsity) """ Create a bit map encoder From the encoder's __init__ method: 1st arg: the total bits in input 2nd arg: the number of bits used to encode each input bit """ self.e = BitmapArrayEncoder(self.numCols, 1) """ Sampling details rate: The sampling rate in Hz of my soundcard buffersize: The size of the array to which we will save audio segments (2^12 = 4096 is very good) secToRecord: The length of each sampling buffersToRecord: how many multiples of buffers are we recording? """ rate = 44100 secToRecord = 0.1 self.buffersize = 2 ** 12 self.buffersToRecord = int(rate * secToRecord / self.buffersize) if not self.buffersToRecord: self.buffersToRecord = 1 """ Filters in Hertz highHertz: lower limit of the bandpass filter, in Hertz lowHertz: upper limit of the bandpass filter, in Hertz max lowHertz = (buffersize / 2 - 1) * rate / buffersize """ highHertz = 500 lowHertz = 10000 """ Convert filters from Hertz to bins highpass: convert the highHertz into a bin for the FFT lowpass: convert the lowHertz into a bin for the FFt NOTES: highpass is at least the 1st bin since most mics only pick up >=20Hz lowpass is no higher than buffersize/2 - 1 (highest array index) passband needs to be wider than size of numInput - not checking for that """ self.highpass = max(int(highHertz * self.buffersize / rate), 1) self.lowpass = min(int(lowHertz * self.buffersize / rate), self.buffersize / 2 - 1) """ The call to create the temporal pooler region """ self.tp = TP( numberOfCols=self.numCols, cellsPerColumn=4, initialPerm=0.5, connectedPerm=0.5, minThreshold=10, newSynapseCount=10, permanenceInc=0.1, permanenceDec=0.07, activationThreshold=8, globalDecay=0.02, burnIn=2, checkSynapseConsistency=False, pamLength=100, ) """ Creating the audio stream from our mic """ p = pyaudio.PyAudio() self.inStream = p.open( format=pyaudio.paInt32, channels=1, rate=rate, input=True, frames_per_buffer=self.buffersize ) """ Setting up the array that will handle the timeseries of audio data from our input """ self.audio = numpy.empty((self.buffersToRecord * self.buffersize), dtype="uint32") """ Print out the inputs """ print "Number of columns:\t" + str(self.numCols) print "Max size of input:\t" + str(self.numInput) print "Sampling rate (Hz):\t" + str(rate) print "Passband filter (Hz):\t" + str(highHertz) + " - " + str(lowHertz) print "Passband filter (bin):\t" + str(self.highpass) + " - " + str(self.lowpass) print "Bin difference:\t\t" + str(self.lowpass - self.highpass) print "Buffersize:\t\t" + str(self.buffersize) """ Setup the plot Use the bandpass filter frequency range as the x-axis Rescale the y-axis """ plt.ion() bin = range(self.highpass, self.lowpass) xs = numpy.arange(len(bin)) * rate / self.buffersize + highHertz self.freqPlot = plt.plot(xs, xs)[0] plt.ylim(0, 10 ** 12) while True: self.processAudio()
class AudioStream: def __init__(self): """ Instantiate temporal pooler, encoder, audio sampler, filter, & freq plot """ self.vis = Visualizations() """ The number of columns in the input and therefore the TP 2**9 = 512 Trial and error pulled that out numCols should be tested during benchmarking """ self.numCols = 2 ** 9 sparsity = 0.10 self.numInput = int(self.numCols * sparsity) """ Create a bit map encoder From the encoder's __init__ method: 1st arg: the total bits in input 2nd arg: the number of bits used to encode each input bit """ self.e = BitmapArrayEncoder(self.numCols, 1) """ Sampling details rate: The sampling rate in Hz of my soundcard buffersize: The size of the array to which we will save audio segments (2^12 = 4096 is very good) secToRecord: The length of each sampling buffersToRecord: how many multiples of buffers are we recording? """ rate = 44100 secToRecord = 0.1 self.buffersize = 2 ** 12 self.buffersToRecord = int(rate * secToRecord / self.buffersize) if not self.buffersToRecord: self.buffersToRecord = 1 """ Filters in Hertz highHertz: lower limit of the bandpass filter, in Hertz lowHertz: upper limit of the bandpass filter, in Hertz max lowHertz = (buffersize / 2 - 1) * rate / buffersize """ highHertz = 500 lowHertz = 10000 """ Convert filters from Hertz to bins highpass: convert the highHertz into a bin for the FFT lowpass: convert the lowHertz into a bin for the FFt NOTES: highpass is at least the 1st bin since most mics only pick up >=20Hz lowpass is no higher than buffersize/2 - 1 (highest array index) passband needs to be wider than size of numInput - not checking for that """ self.highpass = max(int(highHertz * self.buffersize / rate), 1) self.lowpass = min(int(lowHertz * self.buffersize / rate), self.buffersize / 2 - 1) """ The call to create the temporal pooler region """ self.tp = TP( numberOfCols=self.numCols, cellsPerColumn=4, initialPerm=0.5, connectedPerm=0.5, minThreshold=10, newSynapseCount=10, permanenceInc=0.1, permanenceDec=0.07, activationThreshold=8, globalDecay=0.02, burnIn=2, checkSynapseConsistency=False, pamLength=100, ) """ Creating the audio stream from our mic """ p = pyaudio.PyAudio() self.inStream = p.open( format=pyaudio.paInt32, channels=1, rate=rate, input=True, frames_per_buffer=self.buffersize ) """ Setting up the array that will handle the timeseries of audio data from our input """ self.audio = numpy.empty((self.buffersToRecord * self.buffersize), dtype="uint32") """ Print out the inputs """ print "Number of columns:\t" + str(self.numCols) print "Max size of input:\t" + str(self.numInput) print "Sampling rate (Hz):\t" + str(rate) print "Passband filter (Hz):\t" + str(highHertz) + " - " + str(lowHertz) print "Passband filter (bin):\t" + str(self.highpass) + " - " + str(self.lowpass) print "Bin difference:\t\t" + str(self.lowpass - self.highpass) print "Buffersize:\t\t" + str(self.buffersize) """ Setup the plot Use the bandpass filter frequency range as the x-axis Rescale the y-axis """ plt.ion() bin = range(self.highpass, self.lowpass) xs = numpy.arange(len(bin)) * rate / self.buffersize + highHertz self.freqPlot = plt.plot(xs, xs)[0] plt.ylim(0, 10 ** 12) while True: self.processAudio() def processAudio(self): """ Sample audio, encode, send it to the TP Pulls the audio from the mic Conditions that audio as an SDR Computes a prediction via the TP Update the visualizations """ """ Cycle through the multiples of the buffers we're sampling Sample audio to store for each frame in buffersize Mic voltage-level timeseries is saved as 32-bit binary Convert that 32-bit binary into integers, and save to array for the FFT """ for i in range(self.buffersToRecord): try: audioString = self.inStream.read(self.buffersize) except IOError: print "Overflow error from 'audiostring = inStream.read(buffersize)'. Try decreasing buffersize." quit() self.audio[i * self.buffersize : (i + 1) * self.buffersize] = numpy.fromstring(audioString, dtype="uint32") """ Get int array of strength for each bin of frequencies via fast fourier transform Get the indices of the strongest frequencies (the top 'numInput') Scale the indices so that the frequencies fit to within numCols Pick out the unique indices (we've reduced the mapping, so we likely have multiples) Encode those indices into an SDR via the BitmapArrayEncoder Cast the SDR as a float for the TP """ ys = self.fft(self.audio, self.highpass, self.lowpass) fs = numpy.sort(ys.argsort()[-self.numInput :]) rfs = fs.astype(numpy.float32) / (self.lowpass - self.highpass) * self.numCols ufs = numpy.unique(rfs) actualInt = self.e.encode(ufs) actual = actualInt.astype(numpy.float32) """ Pass the SDR to the TP Collect the prediction SDR from the TP Pass the prediction & actual SDRS to the anomaly calculator & array comparer Update the frequency plot """ self.tp.compute(actual, enableLearn=True, computeInfOutput=True) predictedInt = self.tp.getPredictedState().max(axis=1) compare = self.vis.compareArray(actualInt, predictedInt) anomaly = self.vis.calcAnomaly(actualInt, predictedInt) print ".".join(compare) print self.vis.hashtagAnomaly(anomaly) self.freqPlot.set_ydata(ys) plt.show(block=False) plt.draw() def fft(self, audio, highpass, lowpass): """ Fast fourier transform conditioning Output: 'output' contains the strength of each frequency in the audio signal frequencies are marked by its position in 'output': frequency = index * rate / buffesize output.size = buffersize/2 Method: Use numpy's FFT (numpy.fft.fft) Find the magnitude of the complex numbers returned (abs value) Split the FFT array in half, because we have mirror frequencies (they're the complex conjugates) Use just the first half to apply the bandpass filter Great info here: http://stackoverflow.com/questions/4364823/how-to-get-frequency-from-fft-result """ left, right = numpy.split(numpy.abs(numpy.fft.fft(audio)), 2) output = left[highpass:lowpass] return output
class AudioStream: def __init__(self): """ Instantiate temporal pooler, encoder, audio sampler, filter, & freq plot """ self.vis = Visualizations() """ The number of columns in the input and therefore the TP 2**9 = 512 Trial and error pulled that out numCols should be tested during benchmarking """ self.numCols = 2**9 sparsity = 0.10 self.numInput = int(self.numCols * sparsity) """ Create a bit map encoder From the encoder's __init__ method: 1st arg: the total bits in input 2nd arg: the number of bits used to encode each input bit """ self.e = BitmapArrayEncoder(self.numCols, 1) """ Sampling details rate: The sampling rate in Hz of my soundcard buffersize: The size of the array to which we will save audio segments (2^12 = 4096 is very good) secToRecord: The length of each sampling buffersToRecord: how many multiples of buffers are we recording? """ rate = 44100 secToRecord = .1 self.buffersize = 2**12 self.buffersToRecord = int(rate * secToRecord / self.buffersize) if not self.buffersToRecord: self.buffersToRecord = 1 """ Filters in Hertz highHertz: lower limit of the bandpass filter, in Hertz lowHertz: upper limit of the bandpass filter, in Hertz max lowHertz = (buffersize / 2 - 1) * rate / buffersize """ highHertz = 500 lowHertz = 10000 """ Convert filters from Hertz to bins highpass: convert the highHertz into a bin for the FFT lowpass: convert the lowHertz into a bin for the FFt NOTES: highpass is at least the 1st bin since most mics only pick up >=20Hz lowpass is no higher than buffersize/2 - 1 (highest array index) passband needs to be wider than size of numInput - not checking for that """ self.highpass = max(int(highHertz * self.buffersize / rate), 1) self.lowpass = min(int(lowHertz * self.buffersize / rate), self.buffersize / 2 - 1) """ The call to create the temporal pooler region """ self.tp = TP(numberOfCols=self.numCols, cellsPerColumn=4, initialPerm=0.5, connectedPerm=0.5, minThreshold=10, newSynapseCount=10, permanenceInc=0.1, permanenceDec=0.07, activationThreshold=8, globalDecay=0.02, burnIn=2, checkSynapseConsistency=False, pamLength=100) """ Creating the audio stream from our mic """ p = pyaudio.PyAudio() self.inStream = p.open(format=pyaudio.paInt32, channels=1, rate=rate, input=True, frames_per_buffer=self.buffersize) """ Setting up the array that will handle the timeseries of audio data from our input """ self.audio = numpy.empty((self.buffersToRecord * self.buffersize), dtype="uint32") """ Print out the inputs """ print "Number of columns:\t" + str(self.numCols) print "Max size of input:\t" + str(self.numInput) print "Sampling rate (Hz):\t" + str(rate) print "Passband filter (Hz):\t" + str(highHertz) + " - " + str( lowHertz) print "Passband filter (bin):\t" + str(self.highpass) + " - " + str( self.lowpass) print "Bin difference:\t\t" + str(self.lowpass - self.highpass) print "Buffersize:\t\t" + str(self.buffersize) """ Setup the plot Use the bandpass filter frequency range as the x-axis Rescale the y-axis """ plt.ion() bin = range(self.highpass, self.lowpass) xs = numpy.arange(len(bin)) * rate / self.buffersize + highHertz self.freqPlot = plt.plot(xs, xs)[0] plt.ylim(0, 10**12) while True: self.processAudio() def processAudio(self): """ Sample audio, encode, send it to the TP Pulls the audio from the mic Conditions that audio as an SDR Computes a prediction via the TP Update the visualizations """ """ Cycle through the multiples of the buffers we're sampling Sample audio to store for each frame in buffersize Mic voltage-level timeseries is saved as 32-bit binary Convert that 32-bit binary into integers, and save to array for the FFT """ for i in range(self.buffersToRecord): try: audioString = self.inStream.read(self.buffersize) except IOError: print "Overflow error from 'audiostring = inStream.read(buffersize)'. Try decreasing buffersize." quit() self.audio[i * self.buffersize:(i + 1) * self.buffersize] = numpy.fromstring(audioString, dtype="uint32") """ Get int array of strength for each bin of frequencies via fast fourier transform Get the indices of the strongest frequencies (the top 'numInput') Scale the indices so that the frequencies fit to within numCols Pick out the unique indices (we've reduced the mapping, so we likely have multiples) Encode those indices into an SDR via the BitmapArrayEncoder Cast the SDR as a float for the TP """ ys = self.fft(self.audio, self.highpass, self.lowpass) fs = numpy.sort(ys.argsort()[-self.numInput:]) rfs = fs.astype( numpy.float32) / (self.lowpass - self.highpass) * self.numCols ufs = numpy.unique(rfs) actualInt = self.e.encode(ufs) actual = actualInt.astype(numpy.float32) """ Pass the SDR to the TP Collect the prediction SDR from the TP Pass the prediction & actual SDRS to the anomaly calculator & array comparer Update the frequency plot """ self.tp.compute(actual, enableLearn=True, computeInfOutput=True) predictedInt = self.tp.getPredictedState().max(axis=1) compare = self.vis.compareArray(actualInt, predictedInt) anomaly = self.vis.calcAnomaly(actualInt, predictedInt) print ".".join(compare) print self.vis.hashtagAnomaly(anomaly) self.freqPlot.set_ydata(ys) plt.show(block=False) plt.draw() def fft(self, audio, highpass, lowpass): """ Fast fourier transform conditioning Output: 'output' contains the strength of each frequency in the audio signal frequencies are marked by its position in 'output': frequency = index * rate / buffesize output.size = buffersize/2 Method: Use numpy's FFT (numpy.fft.fft) Find the magnitude of the complex numbers returned (abs value) Split the FFT array in half, because we have mirror frequencies (they're the complex conjugates) Use just the first half to apply the bandpass filter Great info here: http://stackoverflow.com/questions/4364823/how-to-get-frequency-from-fft-result """ left, right = numpy.split(numpy.abs(numpy.fft.fft(audio)), 2) output = left[highpass:lowpass] return output
def __init__(self): """ Instantiate temporal pooler, encoder, audio sampler, filter, & freq plot """ self.vis = Visualizations() """ The number of columns in the input and therefore the TP 2**9 = 512 Trial and error pulled that out numCols should be tested during benchmarking """ self.numCols = 2**9 sparsity = 0.10 self.numInput = int(self.numCols * sparsity) """ Create a bit map encoder From the encoder's __init__ method: 1st arg: the total bits in input 2nd arg: the number of bits used to encode each input bit """ self.e = BitmapArrayEncoder(self.numCols, 1) """ Sampling details rate: The sampling rate in Hz of my soundcard buffersize: The size of the array to which we will save audio segments (2^12 = 4096 is very good) secToRecord: The length of each sampling buffersToRecord: how many multiples of buffers are we recording? """ rate = 44100 secToRecord = .1 self.buffersize = 2**12 self.buffersToRecord = int(rate * secToRecord / self.buffersize) if not self.buffersToRecord: self.buffersToRecord = 1 """ Filters in Hertz highHertz: lower limit of the bandpass filter, in Hertz lowHertz: upper limit of the bandpass filter, in Hertz max lowHertz = (buffersize / 2 - 1) * rate / buffersize """ highHertz = 500 lowHertz = 10000 """ Convert filters from Hertz to bins highpass: convert the highHertz into a bin for the FFT lowpass: convert the lowHertz into a bin for the FFt NOTES: highpass is at least the 1st bin since most mics only pick up >=20Hz lowpass is no higher than buffersize/2 - 1 (highest array index) passband needs to be wider than size of numInput - not checking for that """ self.highpass = max(int(highHertz * self.buffersize / rate), 1) self.lowpass = min(int(lowHertz * self.buffersize / rate), self.buffersize / 2 - 1) """ The call to create the temporal pooler region """ self.tp = TP(numberOfCols=self.numCols, cellsPerColumn=4, initialPerm=0.5, connectedPerm=0.5, minThreshold=10, newSynapseCount=10, permanenceInc=0.1, permanenceDec=0.07, activationThreshold=8, globalDecay=0.02, burnIn=2, checkSynapseConsistency=False, pamLength=100) """ Creating the audio stream from our mic """ p = pyaudio.PyAudio() self.inStream = p.open(format=pyaudio.paInt32, channels=1, rate=rate, input=True, frames_per_buffer=self.buffersize) """ Setting up the array that will handle the timeseries of audio data from our input """ self.audio = numpy.empty((self.buffersToRecord * self.buffersize), dtype="uint32") """ Print out the inputs """ print "Number of columns:\t" + str(self.numCols) print "Max size of input:\t" + str(self.numInput) print "Sampling rate (Hz):\t" + str(rate) print "Passband filter (Hz):\t" + str(highHertz) + " - " + str( lowHertz) print "Passband filter (bin):\t" + str(self.highpass) + " - " + str( self.lowpass) print "Bin difference:\t\t" + str(self.lowpass - self.highpass) print "Buffersize:\t\t" + str(self.buffersize) """ Setup the plot Use the bandpass filter frequency range as the x-axis Rescale the y-axis """ plt.ion() bin = range(self.highpass, self.lowpass) xs = numpy.arange(len(bin)) * rate / self.buffersize + highHertz self.freqPlot = plt.plot(xs, xs)[0] plt.ylim(0, 10**12) while True: self.processAudio()