219 lines
6.5 KiB
Python
Executable File
219 lines
6.5 KiB
Python
Executable File
"""
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Sends live audio analysis to the terminal.
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Based on musicinformationretrieval.com/realtime_spectrogram.py
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For more examples using PyAudio:
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https://github.com/mwickert/scikit-dsp-comm/blob/master/sk_dsp_comm/pyaudio_helper.py
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"""
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from __future__ import print_function
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import argparse
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import json
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import librosa
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import math
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import numpy
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import os
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import pyaudio
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import redis
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import statistics
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import sys
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import time
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def debug(*args, **kwargs):
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if( verbose == False ):
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return
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print(*args, file=sys.stderr, **kwargs)
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# Define default variables.
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BAND_OCTAVES = 10 # 12 * 9 octaves
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_BAND_TONES = BAND_OCTAVES * 12 # octaves * notes per octave
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_CHANNELS = 1
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_FRAMES_PER_BUFFER = 4410
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_N_FFT = 4096
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_RATE = 44100
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_SAMPLING_FREQUENCY = 0.1
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# Argument parsing
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# Audio Args
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parser = argparse.ArgumentParser(prog='realtime_redis')
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parser.add_argument('--list-devices','-L', action='store_true', help='Which devices are detected by pyaudio')
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parser.add_argument('--mode','-m', required=False, default='spectrum', choices=['spectrum', 'bpm'], type=str, help='Which mode to use. Default=spectrum')
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parser.add_argument('--device','-d', required=False, type=int, help='Which pyaudio device to use')
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#parser.add_argument('--frames','-f', required=False, default=4410, type=int, help='How many frames per buffer. Default={}'.format(_FRAMES_PER_BUFFER))
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parser.add_argument('--sampling-frequency','-s', required=False, default=0.1, type=float, help='Which frequency, in seconds. Default={}f '.format(_SAMPLING_FREQUENCY))
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parser.add_argument('--channels','-c', required=False, default=_CHANNELS, type=int, help='How many channels. Default={} '.format(_CHANNELS))
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parser.add_argument('--rate','-r', required=False, default=44100, type=int, help='Which rate. Default={} '.format(_RATE))
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# Redis Args
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parser.add_argument("-i","--ip",help="IP address of the Redis server ",default="127.0.0.1",type=str)
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parser.add_argument("-p","--port",help="Port of the Redis server ",default="6379",type=str)
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# Stardard Args
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parser.add_argument("-v","--verbose",action="store_true",help="Verbose")
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args = parser.parse_args()
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# global
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bpm = 120.0
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# Set real variables
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F_LO = librosa.note_to_hz('C0')
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F_HI = librosa.note_to_hz('C10')
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BAND_TONES = _BAND_TONES
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CHANNELS = args.channels
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DEVICE = args.device
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FRAMES_PER_BUFFER = int(args.rate * args.sampling_frequency )
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LIST_DEVICES = args.list_devices
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MODE = args.mode
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N_FFT = _N_FFT
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RATE = args.rate
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SAMPLING_FREQUENCY = args.sampling_frequency
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ip = args.ip
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port = args.port
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verbose = args.verbose
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if( MODE == "bpm" and SAMPLING_FREQUENCY < 0.5 ):
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debug( "You should use a --sampling_frequency superior to 0.5 in BPM mode...")
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melFilter = librosa.filters.mel(RATE, N_FFT, BAND_TONES, fmin=F_LO, fmax=F_HI)
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r = redis.Redis(
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host=ip,
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port=port)
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# Early exit to list devices
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# As it may crash later if not properly configured
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#
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def list_devices():
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# List all audio input devices
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p = pyaudio.PyAudio()
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i = 0
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n = p.get_device_count()
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print("\nFound {} devices\n".format(n))
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print(" {} {}".format('ID', 'Device name'))
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while i < n:
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dev = p.get_device_info_by_index(i)
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if dev['maxInputChannels'] > 0:
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print(" {} {}".format(i, dev['name']))
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i += 1
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if( LIST_DEVICES ):
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list_devices()
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os._exit(1)
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p = pyaudio.PyAudio()
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def m_bpm(audio_data):
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"""
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This function saves slow analysis to redis
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* onset
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* bpm
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* beat
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"""
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global bpm
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if( bpm <= 10):
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bpm = 10
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onset = librosa.onset.onset_detect(
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y = audio_data,
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sr = RATE
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)
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new_bpm, beats = librosa.beat.beat_track(
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y = audio_data,
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sr = RATE,
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trim = False,
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start_bpm = bpm,
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units = "time"
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)
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# Save to Redis
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r.set( 'onset', json.dumps( onset.tolist() ) )
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r.set( 'bpm', json.dumps( new_bpm ) )
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r.set( 'beats', json.dumps( beats.tolist() ) )
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bpm = new_bpm
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debug( "bpm:{} onset:{} beats:{}".format(bpm,onset,beats) )
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return True
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def m_spectrum(audio_data):
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"""
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This function saves fast analysis to redis
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"""
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# Compute real FFT.
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fft = numpy.fft.rfft(audio_data, n=N_FFT)
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# Compute mel spectrum.
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melspectrum = melFilter.dot(abs(fft))
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# Initialize output characters to display.
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spectrum_120 = [0]*BAND_TONES
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spectrum_10 = [0]*BAND_OCTAVES
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spectrum_oct = [[] for i in range(10)]
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# Assign values
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for i in range(BAND_TONES):
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val = round(melspectrum[i],2)
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spectrum_120[i] = val
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key = int(math.floor( i / 12 ))
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spectrum_oct[key].append(val)
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for i in range(BAND_OCTAVES):
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spectrum_10[i] = round(sum( spectrum_oct[i] ) / len( spectrum_oct[i]),2)
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# Get RMS
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#rms = librosa.feature.rms( S=melspectrum )
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rms = librosa.feature.rms( y=audio_data ).tolist()[0]
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rms_avg = round(sum(rms) / len(rms),2)
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# Save to redis
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#debug( 'spectrum_120:{} '.format(spectrum_120))
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#debug( 'spectrum_10:{}'.format(spectrum_10))
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#debug( 'rms:{}'.format(rms_avg))
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r.set( 'spectrum_120', json.dumps( spectrum_120 ) )
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r.set( 'spectrum_10', json.dumps( spectrum_10 ) )
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r.set( 'rms', "{}".format(rms_avg) )
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return True
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def callback(in_data, frame_count, time_info, status):
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audio_data = numpy.fromstring(in_data, dtype=numpy.float32)
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start = time.time()
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if MODE == 'spectrum':
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m_spectrum(audio_data)
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elif MODE == 'bpm':
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m_bpm( audio_data)
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else:
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debug( "Unknown mode. Exiting")
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os._exit(2)
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end = time.time()
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debug ("\rLoop took {:.2}s on {}s ".format(end - start, SAMPLING_FREQUENCY))
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return (in_data, pyaudio.paContinue)
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debug( "\n\nRunning! Using mode {}.\n\n".format(MODE))
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if MODE == 'spectrum':
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debug("In this mode, we will set keys: rms, spectrum, tuning")
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elif MODE == 'bpm':
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debug("In this mode, we will set keys: onset, bpm, beats")
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stream = p.open(format=pyaudio.paFloat32,
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channels=CHANNELS,
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rate=RATE,
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input=True, # Do record input.
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output=False, # Do not play back output.
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frames_per_buffer=FRAMES_PER_BUFFER,
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input_device_index = DEVICE,
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stream_callback=callback)
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stream.start_stream()
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while stream.is_active():
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time.sleep(SAMPLING_FREQUENCY)
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stream.stop_stream()
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stream.close()
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p.terminate()
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