build the audiorecorder client
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clients/audiorecorder-pulse-udp/Util/PyAudioRecorder.py
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130
clients/audiorecorder-pulse-udp/Util/PyAudioRecorder.py
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#
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# The idea is to have only one thread accessing the audio input source instead
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# of every music enabled thread itself. also, different fuctions calculating
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# frequencys and so on shoud run in own threads to the leds get more updates
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#
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# in history one thread was calculating bpm, freqence average and max values in one thread
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# before updating the leds, what the pi was a bit slow for and you cloud count every update
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# of the leds
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# i think most of it is from https://github.com/shunfu/python-beat-detector/
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import numpy
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import pyaudio
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import threading
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import time
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class PyAudioRecorder:
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"""Simple, cross-platform class to record from the default input device."""
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def __init__(self):
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self.RATE = 44100
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self.BUFFERSIZE = 2**12
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self.secToRecord = .1
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self.kill_threads = False
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self.has_new_audio = False
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self.setup()
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# since the server only can handly one input (for now) this thread will calculate
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# some basic things for the musicEffectsThreads, so they can update the leds more often.
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# it is always posible to catch the fft() function and do your own thing in your musikEffect.
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# calculate bpm, this is the same for all clientThreads
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self.beats_idx = 0
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self.bpm_list = []
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self.prev_beat = time.perf_counter()
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self.low_freq_avg_list = []
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self.lastTime = time.time()
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self.recorderClients = []
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def setup(self):
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self.buffers_to_record = int(
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self.RATE * self.secToRecord / self.BUFFERSIZE)
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if self.buffers_to_record == 0:
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self.buffers_to_record = 1
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self.samples_to_record = int(self.BUFFERSIZE * self.buffers_to_record)
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self.chunks_to_record = int(self.samples_to_record / self.BUFFERSIZE)
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self.sec_per_point = 1. / self.RATE
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self.p = pyaudio.PyAudio()
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# start the PyAudio class
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# make sure the default input device is broadcasting the speaker output
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# there are a few ways to do this
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# e.g., stereo mix, VB audio cable for windows, soundflower for mac
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self.in_stream = self.p.open(format=pyaudio.paInt16,
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channels=1,
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rate=self.RATE,
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input=True,
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frames_per_buffer=self.BUFFERSIZE)
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print("Using default input device: {:s}".format(
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self.p.get_default_input_device_info()['name']))
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self.audio = numpy.empty(
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(self.chunks_to_record * self.BUFFERSIZE), dtype=numpy.int16)
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def close(self):
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print("pyAudioRecorder closed")
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self.kill_threads = True
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self.p.close(self.in_stream)
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### RECORDING AUDIO ###
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def get_audio(self):
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"""get a single buffer size worth of audio."""
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audio_string = self.in_stream.read(self.BUFFERSIZE)
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return numpy.fromstring(audio_string, dtype=numpy.int16)
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def recordo(self):
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while not self.kill_threads:
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for i in range(self.chunks_to_record):
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self.audio[i*self.BUFFERSIZE:(i+1)
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* self.BUFFERSIZE] = self.get_audio()
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self.has_new_audio = True
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def record(self):
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#while not self.kill_threads:
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for i in range(self.chunks_to_record):
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self.audio[i*self.BUFFERSIZE:(i+1)
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* self.BUFFERSIZE] = self.get_audio()
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#self.has_new_audio = True
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def start(self):
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print("pyAudioRecorder started")
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self.t = threading.Thread(target=self.record)
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self.t.start()
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### MATH ###
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def downsample(self, data, mult):
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"""Given 1D data, return the binned average."""
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overhang = len(data) % mult
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if overhang:
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data = data[:-overhang]
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data = numpy.reshape(data, (len(data) / mult, mult))
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data = numpy.average(data, 1)
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return data
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def fft(self, data=None, trim_by=20, log_scale=False, div_by=10000):
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self.record()
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if not data:
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data = self.audio.flatten()
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left, right = numpy.split(numpy.abs(numpy.fft.fft(data)), 2)
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ys = numpy.add(left, right[::-1])
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if log_scale:
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ys = numpy.multiply(20, numpy.log10(ys))
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xs = numpy.arange(self.BUFFERSIZE/2, dtype=float)
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if trim_by:
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i = int((self.BUFFERSIZE/2) / trim_by)
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ys = ys[:i]
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xs = xs[:i] * self.RATE / self.BUFFERSIZE
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if div_by:
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ys = ys / float(div_by)
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return [xs,ys]
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76
clients/audiorecorder-pulse-udp/audiorecorder.py
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76
clients/audiorecorder-pulse-udp/audiorecorder.py
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from Util.PyAudioRecorder import PyAudioRecorder
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from random import randint
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import time
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import socket
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import pickle
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remoteaddr = ("0.0.0.0",8002)
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# PyAudioRecorder uses the default
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pyAudioRecorder: PyAudioRecorder = PyAudioRecorder()
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#pyAudioRecorder.start()
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print(pyAudioRecorder.p.get_default_input_device_info())
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lastping = time.time()-1
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lastupdate = time.time()-1
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localaddr = ("",randint(6000,7000))
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udpsocket = socket.socket(socket.AF_INET, socket.SOCK_DGRAM)
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udpsocket.setsockopt(socket.SOL_SOCKET, socket.SO_BROADCAST, 1)
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udpsocket.settimeout(.002)
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registered:bool = False
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while True:
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#print(time.time() - self.lastping)
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if time.time() - lastping > .5 :
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udpsocket.sendto("s:ping".encode(), remoteaddr)
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lastping = time.time()
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ftt = pyAudioRecorder.fft()
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#print(ftt)
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if time.time() - lastupdate > 0.00833333333 and registered:
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#if pyAudioRecorder.has_new_audio:
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# there must be a better solution to do this.
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# this looks like my silly arduino code, but it works
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values: str = "d:xs"
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i = 0
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j = 0
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udpsocket.sendto(("dlen:"+str(len(ftt[0]))+":"+str(len(ftt[1]))+"").encode(), remoteaddr)
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for value in ftt[0]:
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values = values+":"+str(j)+"|"+str(value)
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i = i+1
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j = j+1
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if i == 20:
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udpsocket.sendto(values.encode(), remoteaddr)
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i = 0
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values= "d:xs"
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values = "d:ys"
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i = 0
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j = 0
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for value in ftt[1]:
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values = values+":"+str(j)+"|"+str(value)
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i = i+1
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j = j+1
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if i == 20:
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udpsocket.sendto(values.encode(), remoteaddr)
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i = 0
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values= "d:ys"
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udpsocket.sendto("d:c".encode(), remoteaddr)
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lastupdate = time.time()
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try:
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rbytes, address = udpsocket.recvfrom(4096)
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remoteaddr = address
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data = rbytes.decode('UTF-8')
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if data:
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if data[0] is "s" and data[1] is "r":
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registered = True
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udpsocket.sendto(("r:2:"+pyAudioRecorder.p.get_default_input_device_info()['name']+"").encode(), address)
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except socket.timeout:
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pass
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