Wie man BPM-tempo und audio-Funktionen in Python

Echo Nest API ist das, was du suchst:

http://echonest.github.io/remix/

Python-bindings sind Reich, obwohl die Installation von Echo Nest kann schmerzhaft sein, da das team scheinbar nicht in der Lage, eine solide Installateure.

Jedoch nicht in der lokalen Verarbeitung. Stattdessen berechnet er die audio-fingerprint-und uploads der song für die Echo-Nest-Server, die für die Extraktion von Informationen mithilfe von algorithmen, die Sie nicht aussetzen.

ich habe festgestellt, das der code von @scaperot hier, die Ihnen helfen könnte:

import wave, array, math, time, argparse, sys
import numpy, pywt
from scipy import signal
import pdb
import matplotlib.pyplot as plt

def read_wav(filename):

    #open file, get metadata for audio
    try:
        wf = wave.open(filename,'rb')
    except IOError, e:
        print e
        return

    # typ = choose_type( wf.getsampwidth() ) #TODO: implement choose_type
    nsamps = wf.getnframes();
    assert(nsamps > 0);

    fs = wf.getframerate()
    assert(fs > 0)

    # read entire file and make into an array
    samps = list(array.array('i',wf.readframes(nsamps)))
    #print 'Read', nsamps,'samples from', filename
    try:
        assert(nsamps == len(samps))
    except AssertionError, e:
        print  nsamps, "not equal to", len(samps)

    return samps, fs

# print an error when no data can be found
def no_audio_data():
    print "No audio data for sample, skipping..."
    return None, None

# simple peak detection
def peak_detect(data):
    max_val = numpy.amax(abs(data)) 
    peak_ndx = numpy.where(data==max_val)
    if len(peak_ndx[0]) == 0: #if nothing found then the max must be negative
        peak_ndx = numpy.where(data==-max_val)
    return peak_ndx

def bpm_detector(data,fs):
    cA = [] 
    cD = []
    correl = []
    cD_sum = []
    levels = 4
    max_decimation = 2**(levels-1);
    min_ndx = 60./ 220 * (fs/max_decimation)
    max_ndx = 60./ 40 * (fs/max_decimation)

    for loop in range(0,levels):
        cD = []
        # 1) DWT
        if loop == 0:
            [cA,cD] = pywt.dwt(data,'db4');
            cD_minlen = len(cD)/max_decimation+1;
            cD_sum = numpy.zeros(cD_minlen);
        else:
            [cA,cD] = pywt.dwt(cA,'db4');
        # 2) Filter
        cD = signal.lfilter([0.01],[1 -0.99],cD);

        # 4) Subtractargs.filename out the mean.

        # 5) Decimate for reconstruction later.
        cD = abs(cD[::(2**(levels-loop-1))]);
        cD = cD - numpy.mean(cD);
        # 6) Recombine the signal before ACF
        #    essentially, each level I concatenate 
        #    the detail coefs (i.e. the HPF values)
        #    to the beginning of the array
        cD_sum = cD[0:cD_minlen] + cD_sum;

    if [b for b in cA if b != 0.0] == []:
        return no_audio_data()
    # adding in the approximate data as well...    
    cA = signal.lfilter([0.01],[1 -0.99],cA);
    cA = abs(cA);
    cA = cA - numpy.mean(cA);
    cD_sum = cA[0:cD_minlen] + cD_sum;

    # ACF
    correl = numpy.correlate(cD_sum,cD_sum,'full') 

    midpoint = len(correl) / 2
    correl_midpoint_tmp = correl[midpoint:]
    peak_ndx = peak_detect(correl_midpoint_tmp[min_ndx:max_ndx]);
    if len(peak_ndx) > 1:
        return no_audio_data()

    peak_ndx_adjusted = peak_ndx[0]+min_ndx;
    bpm = 60./ peak_ndx_adjusted * (fs/max_decimation)
    print bpm
    return bpm,correl


if __name__ == '__main__':
    parser = argparse.ArgumentParser(description='Process .wav file to determine the Beats Per Minute.')
    parser.add_argument('--filename', required=True,
                   help='.wav file for processing')
    parser.add_argument('--window', type=float, default=3,
                   help='size of the the window (seconds) that will be scanned to determine the bpm.  Typically less than 10 seconds. [3]')

    args = parser.parse_args()
    samps,fs = read_wav(args.filename)

    data = []
    correl=[]
    bpm = 0
    n=0;
    nsamps = len(samps)
    window_samps = int(args.window*fs)         
    samps_ndx = 0;  #first sample in window_ndx 
    max_window_ndx = nsamps / window_samps;
    bpms = numpy.zeros(max_window_ndx)

    #iterate through all windows
    for window_ndx in xrange(0,max_window_ndx):

        #get a new set of samples
        #print n,":",len(bpms),":",max_window_ndx,":",fs,":",nsamps,":",samps_ndx
        data = samps[samps_ndx:samps_ndx+window_samps]
        if not ((len(data) % window_samps) == 0):
            raise AssertionError( str(len(data) ) ) 

        bpm, correl_temp = bpm_detector(data,fs)
        if bpm == None:
            continue
        bpms[window_ndx] = bpm
        correl = correl_temp

        #iterate at the end of the loop
        samps_ndx = samps_ndx+window_samps;
        n=n+1; #counter for debug...

    bpm = numpy.median(bpms)
    print 'Completed.  Estimated Beats Per Minute:', bpm

    n = range(0,len(correl))
    plt.plot(n,abs(correl)); 
    plt.show(False); #plot non-blocking
    time.sleep(10);
plt.close();

Nun, ich kam vor kurzem in Vampy die wrapper-plugin, das ermöglicht die Verwendung von Vamp-plugins in Python geschrieben in jedem Vamp-host. Vamp ist ein audio-Bearbeitungs-plugin-system für Erweiterungen, das extrahieren von beschreibenden Informationen von audio-Daten. Hoffe, es hilft.