def spectral_features(filelist):
"""
Given a list of files, retrieve them, analyse the first 100mS of each file and return
a feature table.
"""
number_of_files = len(filelist)
number_of_features = 5
features = np.zeros([number_of_files, number_of_features])
sample_rate = 44100
for file_index, url in enumerate(filelist):
print url
urllib.urlretrieve(url, filename='/tmp/localfile.wav')
audio = MonoLoader(filename = '/tmp/localfile.wav', sampleRate = sample_rate)()
zcr = ZeroCrossingRate()
hamming_window = Windowing(type = 'hamming') # we need to window the frame to avoid FFT artifacts.
spectrum = Spectrum()
central_moments = CentralMoments()
distributionshape = DistributionShape()
spectral_centroid = Centroid()
frame_size = int(round(0.100 * sample_rate)) # 100ms
# Only do the first frame for now.
# TODO we should generate values for the entire file, probably by averaging the features.
current_frame = audio[0 : frame_size]
features[file_index, 0] = zcr(current_frame)
spectral_magnitude = spectrum(hamming_window(current_frame))
centroid = spectral_centroid(spectral_magnitude)
spectral_moments = distributionshape(central_moments(spectral_magnitude))
features[file_index, 1] = centroid
features[file_index, 2:5] = spectral_moments
return features
def file_to_hpcp(filename):
audio = MonoLoader(filename=filename)()
windowing = Windowing(type='blackmanharris62')
spectrum = Spectrum()
spectral_peaks = SpectralPeaks(orderBy='magnitude',
magnitudeThreshold=0.001,
maxPeaks=20,
minFrequency=20,
maxFrequency=8000)
hpcp = HPCP(maxFrequency=8000) # ,
# normalized='unitSum') #VERIFICAR QUE ISTO E O Q FAZ SENTIDO FAZER
spec_group = []
hpcp_group = []
for frame in FrameGenerator(audio, frameSize=1024, hopSize=512):
windowed = windowing(frame)
fft = spectrum(windowed)
frequencies, magnitudes = spectral_peaks(fft)
final_hpcp = hpcp(frequencies, magnitudes)
spec_group.append(fft)
hpcp_group.append(final_hpcp)
mean_hpcp = np.mean(np.array(hpcp_group).T, axis=1)
return mean_hpcp
def hfc(filename):
audio = MonoLoader(filename=filename, sampleRate=44100)()
features = []
for frame in FrameGenerator(audio, frameSize = 1024, hopSize = 512):
mag, phase =CartesianToPolar()(FFT()(Windowing(type='hann')(frame)))
features.append(OnsetDetection(method='hfc')(mag, phase))
return Onsets()(array([features]),[1])
def noveltycurve(filename):
audio = MonoLoader(filename=filename, sampleRate=44100)()
band_energy = []
for frame in FrameGenerator(audio, frameSize = 1024, hopSize = 512):
mag, phase, = CartesianToPolar()(FFT()(Windowing(type='hann')(frame)))
band_energy.append(FrequencyBands()(mag))
novelty = NoveltyCurve()(band_energy)
return Onsets()(np.array([novelty]),[1])
def feature_allframes(input_features, frame_indexer = None): audio = input_features['audio'] beats = input_features['beats'] # Initialise the algorithms w = Windowing(type = 'hann') spectrum = Spectrum() # FFT would return complex FFT, we only want magnitude melbands = MelBands(numberBands = NUMBER_BANDS) #~ mfcc = MFCC(numberBands = NUMBER_BANDS, numberCoefficients = NUMBER_COEFF) pool = Pool() if frame_indexer is None: frame_indexer = range(4,len(beats) - 1) # Exclude first frame, because it has no predecessor to calculate difference with # 13 MFCC coefficients # 40 Mel band energies #~ mfcc_coeffs = np.zeros((len(beats), NUMBER_COEFF)) mfcc_bands = np.zeros((len(beats), NUMBER_BANDS)) # 1 cosine distance value between every mfcc feature vector # 13 differences between MFCC coefficient of this frame and previous frame # 13 differences between MFCC coefficient of this frame and frame - 4 # 13 differences between the differences above # Idem for mel band energies #~ mfcc_coeff_diff = np.zeros((len(beats), NUMBER_COEFF)) mfcc_bands_diff = np.zeros((len(beats), NUMBER_BANDS * 4)) # Step 1: Calculate framewise for all output frames # Calculate this for all frames where this frame, or its successor, is in the frame_indexer for i in [i for i in range(len(beats)) if (i in frame_indexer) or (i+1 in frame_indexer) or (i-1 in frame_indexer) or (i-2 in frame_indexer) or (i-3 in frame_indexer)]: SAMPLE_RATE = 44100 start_sample = int(beats[i] * SAMPLE_RATE) end_sample = int(beats[i+1] * SAMPLE_RATE) #print start_sample, end_sample frame = audio[start_sample : end_sample if (start_sample - end_sample) % 2 == 0 else end_sample - 1] bands = melbands(spectrum(w(frame))) #~ bands, coeffs = mfcc(spectrum(w(frame))) #~ mfcc_coeffs[i] = coeffs mfcc_bands[i] = bands # Step 2: Calculate the cosine distance between the MFCC values for i in frame_indexer: # The norm of difference is usually very high around downbeat, because of melodic changes there! #~ mfcc_coeff_diff[i] = mfcc_coeffs[i+1] - mfcc_coeffs[i] mfcc_bands_diff[i][0*NUMBER_BANDS : 1*NUMBER_BANDS] = mfcc_bands[i+1] - mfcc_bands[i] mfcc_bands_diff[i][1*NUMBER_BANDS : 2*NUMBER_BANDS] = mfcc_bands[i+2] - mfcc_bands[i] mfcc_bands_diff[i][2*NUMBER_BANDS : 3*NUMBER_BANDS] = mfcc_bands[i+3] - mfcc_bands[i] mfcc_bands_diff[i][3*NUMBER_BANDS : 4*NUMBER_BANDS] = mfcc_bands[i] - mfcc_bands[i-1] # Include the MFCC coefficients as features result = mfcc_bands_diff[frame_indexer] #~ result = np.append(mfcc_coeff_diff[frame_indexer], mfcc_bands_diff[frame_indexer], axis=1) #~ print np.shape(result), np.shape(mfcc_coeff_diff), np.shape(mfcc_bands_diff) return preprocessing.scale(result)
def spectralCentroid(audio,params):
""" hop size, frame size, window type """
hopSize, frameSize, wtype = params
w = Windowing(type=wtype)
spec = Spectrum()
result = []
centroid = ess.Centroid(range=int(44100/2))
for frame in ess.FrameGenerator(audio, frameSize = frameSize, hopSize = hopSize):
sf = spec(w(frame))
result.append(centroid(sf))
return np.asarray(result),hopSize
def rms(audio,params):
""" hop size, frame size, window type """
hopSize, frameSize, wtype = params
w = Windowing(type=wtype)
spec = Spectrum()
result = []
RMS = ess.RMS()
for frame in ess.FrameGenerator(audio, frameSize = frameSize, hopSize = hopSize):
sf = spec(w(frame))
result.append(RMS(sf))
return np.asarray(result),hopSize
def __init__(self, input_blocksize=1024, input_stepsize=512):
super(Essentia_Dissonance, self).__init__()
self.input_blocksize = input_blocksize
self.input_stepsize = min(input_stepsize, self.input_blocksize)
self.windower = Windowing(type='blackmanharris62')
self.spec_alg = None
self.spec_peaks_alg = None
self.dissonance_alg = Dissonance()
self.dissonance = []
def calculateDownbeats(self, audio, bpm, phase):
# Step 0: calculate the CSD (Complex Spectral Difference) features
# and the associated onset detection function ON LOWPASSED SIGNAL
spec = Spectrum(size=self.FRAME_SIZE)
w = Windowing(type='hann')
fft = FFT()
c2p = CartesianToPolar()
od_csd = OnsetDetection(method='complex')
lowpass = LowPass(cutoffFrequency=1500)
pool = Pool()
# TODO test faster (numpy) way
#audio = lowpass(audio)
for frame in FrameGenerator(audio,
frameSize=self.FRAME_SIZE,
hopSize=self.HOP_SIZE):
mag, ph = c2p(fft(w(frame)))
pool.add('onsets.complex', od_csd(mag, ph))
# Step 1: normalise the data using an adaptive mean threshold
novelty_mean = self.adaptive_mean(pool['onsets.complex'], 16.0)
# Step 2: half-wave rectify the result
novelty_hwr = (pool['onsets.complex'] - novelty_mean).clip(min=0)
# Step 7 (experimental): Determine downbeat locations as subsequence with highest complex spectral difference
for i in range(4):
phase_frames = (phase * 44100.0) / (512.0)
frames = (
np.round(
np.arange(phase_frames + i * self.numFramesPerBeat(bpm),
np.size(novelty_hwr),
4 * self.numFramesPerBeat(bpm))).astype('int')
)[:
-1] # Discard last value to prevent reading beyond array (last value rounded up for example)
pool.add('output.downbeat',
np.sum(novelty_hwr[frames]) / np.size(frames))
plt.subplot(4, 1, i + 1)
plt.plot(novelty_hwr)
for f in frames:
plt.axvline(x=f)
print pool['output.downbeat']
downbeatIndex = np.argmax(pool['output.downbeat'])
plt.show()
# experimental
return 1.0 * self.beats[downbeatIndex::4]
def feature_allframes(input_features, frame_indexer = None): audio = input_features['audio'] beats = input_features['beats'] # Initialise the algorithms w = Windowing(type = 'hann') loudness = Loudness() if frame_indexer is None: frame_indexer = range(1,len(beats) - 1) # Exclude first frame, because it has no predecessor to calculate difference with # 1 loudness value by default loudness_values = np.zeros((len(beats), 1)) # 1 difference value between loudness value cur and cur-1 # 1 difference value between loudness value cur and cur-4 # 1 difference value between differences above loudness_differences = np.zeros((len(beats), 9)) # Step 1: Calculate framewise for all output frames # Calculate this for all frames where this frame, or its successor, is in the frame_indexer for i in [i for i in range(len(beats)) if (i in frame_indexer) or (i+1 in frame_indexer) or (i-1 in frame_indexer) or (i-2 in frame_indexer) or (i-3 in frame_indexer) or (i-4 in frame_indexer) or (i-5 in frame_indexer) or (i-6 in frame_indexer) or (i-7 in frame_indexer) or (i-8 in frame_indexer)]: SAMPLE_RATE = 44100 start_sample = int(beats[i] * SAMPLE_RATE) end_sample = int(beats[i+1] * SAMPLE_RATE) #print start_sample, end_sample frame = audio[start_sample : end_sample if (start_sample - end_sample) % 2 == 0 else end_sample - 1] loudness_values[i] = loudness(w(frame)) # Step 2: Calculate the cosine distance between the MFCC values for i in frame_indexer: loudness_differences[i][0] = (loudness_values[i] - loudness_values[i-1]) loudness_differences[i][1] = (loudness_values[i+1] - loudness_values[i]) loudness_differences[i][2] = (loudness_values[i+2] - loudness_values[i]) loudness_differences[i][3] = (loudness_values[i+3] - loudness_values[i]) loudness_differences[i][4] = (loudness_values[i+4] - loudness_values[i]) loudness_differences[i][5] = (loudness_values[i+5] - loudness_values[i]) loudness_differences[i][6] = (loudness_values[i+6] - loudness_values[i]) loudness_differences[i][7] = (loudness_values[i+7] - loudness_values[i]) loudness_differences[i][8] = (loudness_values[i-1] - loudness_values[i+1]) # Include the raw values as absolute features result = loudness_differences[frame_indexer] #~ print np.shape(result), np.shape(loudness_values), np.shape(loudness_differences) return preprocessing.scale(result)
def __call__(self, audio, SR, sumThreshold=1e-5):
self.__reset__()
if audio.ndim > 1:
audio = np.sum(audio, axis=1) / audio.ndim
fcIndexArr = []
self.hist = np.zeros(int(self.frameSize / 2 + 1))
fft = FFT(size=self.frameSize) # declare FFT function
window = Windowing(size=self.frameSize,
type="hann") # declare windowing function
self.avgFrames = np.zeros(int(self.frameSize / 2) + 1)
maxNrg = max([
sum(abs(fft(window(frame)))**2)
for frame in FrameGenerator(audio,
frameSize=self.frameSize,
hopSize=self.hopSize,
startFromZero=True)
])
for i, frame in enumerate(
FrameGenerator(audio,
frameSize=self.frameSize,
hopSize=self.hopSize,
startFromZero=True)):
frame = window(frame) # apply window to the frame
frameFft = abs(fft(frame))
nrg = sum(frameFft**2)
if nrg >= 0.1 * maxNrg:
for j in reversed(range(len(frameFft))):
if sum(frameFft[j:] / j) >= sumThreshold:
fcIndexArr.append(j)
self.hist[j] += nrg
break
self.avgFrames = self.avgFrames + frameFft
if len(fcIndexArr) == 0:
fcIndexArr.append(int(self.frameSize / 2) + 1)
self.hist[int(self.frameSize / 2)] += 1
self.avgFrames /= (i + 1)
self.mostLikelyBin, conf, binary = self.__computeMeanFc(
fcIndexArr, np.arange(int(self.frameSize / 2) + 2), hist=self.hist)
return self.mostLikelyBin * SR / self.frameSize, conf, binary
def mel40_analyzer():
window = Windowing(size=256, type='blackmanharris62')
spectrum = Spectrum(size=256)
mel = MelBands(
inputSize=129,
numberBands=40,
lowFrequencyBound=27.5,
highFrequencyBound=8000.0,
sampleRate=16000.0)
def analyzer(samples):
feats = []
for frame in FrameGenerator(samples, 256, 160):
frame_feats = mel(spectrum(window(frame)))
frame_feats = np.log(frame_feats + 1e-16)
feats.append(frame_feats)
return np.array(feats)
return analyzer
def rms_centroids(filename, frameSize=1024, hopSize=512, sampleRate=44100):
# load our audio into an array
audio = MonoLoader(filename=filename, sampleRate=44100)()
# create the pool and the necessary algorithms
w = Windowing()
spec = Spectrum()
rms = RMS()
centroid = Centroid(range=int(sampleRate / 2))
cs = []
rmss = []
# compute the centroid for all frames in our audio and add it to the pool
for frame in FrameGenerator(audio, frameSize=frameSize, hopSize=hopSize):
sf = spec(w(frame))
cs.append(centroid(sf))
rmss.append(rms(sf))
return np.array(rmss), np.array(cs)
def create_analyzers(fs=44100.0,
nhop=512,
nffts=[1024, 2048, 4096],
mel_nband=80,
mel_freqlo=27.5,
mel_freqhi=16000.0):
analyzers = []
for nfft in nffts:
window = Windowing(size=nfft, type='blackmanharris62')
spectrum = Spectrum(size=nfft)
mel = MelBands(inputSize=(nfft // 2) + 1,
numberBands=mel_nband,
lowFrequencyBound=mel_freqlo,
highFrequencyBound=mel_freqhi,
sampleRate=fs)
analyzers.append((window, spectrum, mel))
return analyzers
def hcdf(filename):
audio = MonoLoader(filename=filename)()
windowing = Windowing(type='hann')
for frame in FrameGenerator(audio, frameSize=32768, hopSize=4096):
windowed = windowing(frame)
print('window', windowed)
# ConstantQ transform
# constant_q = ConstantQ(binsPerOctave=36, minFrequency=110, maxFrequency=3520, sampleRate=11025)
# kk = constant_q(windowed)
# 12 bin tunned Chromagram
# pedirle al ruso que lo ponga
chroma = Chromagram(numberBins=12,
binsPerOctave=36,
minFrequency=110,
windowType='hann') # maxFrequency=3520
pitch_class_vectors = chroma(frame)
print('pitch_class_vectors', pitch_class_vectors)
def ninos(filename,gamma=0.94):
"""
reference: Mounir, M., Karsmakers, P., & Van Waterschoot, T. (2016). Guitar note onset detection based on a spectral sparsity measure.
European Signal Processing Conference. https://doi.org/10.1109/EUSIPCO.2016.7760394
"""
N = 2048
hopSize = int(N/10)
J = int(N*gamma/2)
audio = MonoLoader(filename=filename, sampleRate=44100)()
mag = []
for frame in FrameGenerator(audio, frameSize = N, hopSize = hopSize):
m = CartesianToPolar()(FFT()(Windowing(type='hann')(frame)))[0]
m = np.asarray(m)
idx = np.argsort(m)[::-1][:J]
mag.append(m[idx])
mag = np.asarray(mag)
x2 = mag*mag
inos=np.sum(x2,axis=1)/(np.sum(x2*x2,axis=1)**(0.25))
ninos = inos/(J**(0.25))
return OnsetPeakPickingProcessor(threshold=0.03,fps=44100/hopSize)(ninos)
def shared_main(source, dest, display_result):
source_audio = _loader(source)
destination_audio = _loader(dest)
source_frame = FrameGenerator(source_audio, frameSize=2048, hopSize=512)
destination_frame = FrameGenerator(destination_audio,
frameSize=2048,
hopSize=512)
window = Windowing(type='hann') # window function
spectrum = Spectrum() # spectrum function
pitch_yin_fft = PitchYinFFT() # pitch extractor
pitch_saliennce = PitchSalience()
loudness = Loudness()
# draw_plot(source_frame, window, spectrum, pitch_yin_fft)
min_cost, match_result = compare(source_frame, destination_frame, window, \
spectrum, pitch_yin_fft, 5, 1, 1, display_result, loudness)
return min_cost, match_result
def run(self, audio):
# Calculate the melflux onset detection function
pool = Pool()
w = Windowing(type='hann')
fft = np.fft.fft
od_flux = OnsetDetection(method='melflux')
for frame in FrameGenerator(audio,
frameSize=self.FRAME_SIZE,
hopSize=self.HOP_SIZE):
pool.add('audio.windowed_frames', w(frame))
fft_result = fft(pool['audio.windowed_frames']).astype('complex64')
fft_result_mag = np.absolute(fft_result)
fft_result_ang = np.angle(fft_result)
self.fft_mag_1024_512 = fft_result_mag
self.fft_phase_1024_512 = fft_result_ang
for mag, phase in zip(fft_result_mag, fft_result_ang):
pool.add('onsets.complex', od_flux(mag, phase))
odf = pool['onsets.complex']
# Given the ODF, calculate the tempo and the phase
tempo, tempo_curve, phase, phase_curve = BeatTracker.get_tempo_and_phase_from_odf(
odf, self.HOP_SIZE)
# Calculate the beat annotations
spb = 60. / tempo #seconds per beat
beats = (np.arange(phase,
(np.size(audio) / self.SAMPLE_RATE) - spb + phase,
spb).astype('single'))
# Store all the results
self.bpm = tempo
self.phase = phase
self.beats = beats
self.onset_curve = BeatTracker.hwr(pool['onsets.complex'])
def feature_allframes(audio, beats, frame_indexer = None): # Initialise the algorithms w = Windowing(type = 'blackmanharris92') spectrum = Spectrum() specPeaks = SpectralPeaks() hpcp = HPCP() if frame_indexer is None: frame_indexer = range(1,len(beats) - 1) # Exclude first frame, because it has no predecessor to calculate difference with # 12 chromagram values by default chroma_values = np.zeros((len(beats), 12)) # Difference between chroma vectors chroma_differences = np.zeros((len(beats), 3)) # Step 1: Calculate framewise for all output frames # Calculate this for all frames where this frame, or its successor, is in the frame_indexer for i in [i for i in range(len(beats)) if (i in frame_indexer) or (i+1 in frame_indexer) or (i+1 in frame_indexer)]: SAMPLE_RATE = 44100 start_sample = int(beats[i] * SAMPLE_RATE) end_sample = int(beats[i+1] * SAMPLE_RATE) #print start_sample, end_sample frame = audio[start_sample : (end_sample if (start_sample - end_sample) % 2 == 0 else end_sample - 1)] freq, mag = specPeaks(spectrum(w(frame))) chroma_values[i] = hpcp(freq, mag) # Step 2: Calculate the cosine distance between the MFCC values for i in frame_indexer: chroma_differences[i][0] = np.linalg.norm(chroma_values[i] - chroma_values[i-1]) chroma_differences[i][1] = np.linalg.norm(chroma_values[i] - chroma_values[i+1]) chroma_differences[i][2] = np.linalg.norm(chroma_values[i-1] - chroma_values[i+1]) # Include the raw values as absolute features result = np.append(chroma_values[frame_indexer], chroma_differences[frame_indexer], axis=1) #~ print np.shape(result), np.shape(chroma_values), np.shape(chroma_differences) return preprocessing.scale(result)
def f_essentia_extract(Audio):
## METODOS DE LIBRERIA QUE DETECTAN DONDE OCURRE CADA NOTA RESPECTO AL TIEMPO
od2 = OnsetDetection(method='complex')
# Let's also get the other algorithms we will need, and a pool to store the results
w = Windowing(type='hann')
fft = FFT() # this gives us a complex FFT
c2p = CartesianToPolar(
) # and this turns it into a pair (magnitude, phase)
pool = essentia.Pool()
# Computing onset detection functions.
for frame in FrameGenerator(Audio, frameSize=1024, hopSize=512):
mag, phase, = c2p(fft(w(frame)))
pool.add('features.complex', od2(mag, phase))
## inicio de cada "nota"
onsets = Onsets()
tiempos_detectados_essentia = onsets(
essentia.array([pool['features.complex']]), [1])
#print(tiempos_detectados_essentia)
return tiempos_detectados_essentia
def detectBW(audio: list,
SR: float,
frame_size=256,
hop_size=128,
floor_db=-90,
oversample_f=1):
frame_size *= oversample_f # if an oversample factor is desired, apply it
fc_index_arr = []
fft = FFT(size=frame_size) # declare FFT function
window = Windowing(size=frame_size,
type="hann") # declare windowing function
for frame in FrameGenerator(audio,
frameSize=frame_size,
hopSize=hop_size,
startFromZero=True):
frame_fft = abs(fft(window(frame)))
frame_fft_db = 20 * np.log10(
frame_fft + eps) # calculate frame fft values in db
# compute the linear interpolation between the values of the maxima of the spectrum
interp_frame = compute_spectral_envelope(frame_fft_db, "linear")
interp_frame = modify_floor(interp_frame, floor_db, log=True)
fc_index = compute_fc(interp_frame)
if energy_verification(frame_fft, fc_index):
fc_index_arr.append(fc_index)
if len(fc_index_arr) == 0:
fc_index_arr = [frame_size]
fc_bin, conf, binary = compute_mean_fc(fc_index_arr,
np.arange(len(frame_fft)), SR)
# print("mean_fc: ", fc_bin*SR/frame_size ," conf: ", conf ," binary_result: ", binary)
return fc_bin * SR / frame_size, conf, binary
def feature_allframes(audio, beats, frame_indexer = None): # Initialise the algorithms w = Windowing(type = 'hann') loudness = Loudness() if frame_indexer is None: frame_indexer = range(1,len(beats) - 1) # Exclude first frame, because it has no predecessor to calculate difference with # 1 loudness value by default loudness_values = np.zeros((len(beats), 1)) # 1 difference value between loudness value cur and cur-1 # 1 difference value between loudness value cur and cur-4 # 1 difference value between differences above loudness_feature_vector = np.zeros((len(beats), 4)) # Step 1: Calculate framewise for all output frames # Calculate this for all frames where this frame, or its successor, is in the frame_indexer for i in [i for i in range(len(beats)) if (i in frame_indexer) or (i-1 in frame_indexer) or (i-2 in frame_indexer) or (i-3 in frame_indexer)]: SAMPLE_RATE = 44100 start_sample = int(beats[i] * SAMPLE_RATE) end_sample = int(beats[i+1] * SAMPLE_RATE) #print start_sample, end_sample frame = audio[start_sample : end_sample if (start_sample - end_sample) % 2 == 0 else end_sample - 1] loudness_values[i] = loudness(w(frame)) loudness_values = preprocessing.scale(loudness_values) # Step 2: construct feature vector for i in frame_indexer: loudness_feature_vector[i] = np.reshape(loudness_values[i:i+4], (4,)) # Include the raw values as absolute features result = loudness_feature_vector[frame_indexer] #~ print np.shape(result), np.shape(loudness_values), np.shape(loudness_differences) return result
def extract_features_from_frame(self, frame):
""" Return dictionary of features for the given frame."""
centroid = Centroid(range=22050)
hamming_window = Windowing(type='hamming')
zcr = ess.ZeroCrossingRate()
spectrum = ess.Spectrum()
central_moments = ess.CentralMoments()
# Spectrum can only compute FFT of array of even size (don't know why)
if len(frame) % 2 == 1:
frame = frame[:-1]
spectral_magnitude = spectrum(hamming_window(frame))
feat_dic = {'zrc':zcr(frame), 'centroid':centroid(spectral_magnitude)}
# Central moments
central_moments = ess.CentralMoments()
central_moms = central_moments(hamming_window(frame))
for idx, icm in enumerate(central_moms):
feat_dic['cm{}'.format(idx)] = icm
# Distribution shape
distributionshape = ess.DistributionShape()
for idx, ism in enumerate(distributionshape(central_moms)):
feat_dic['sm{}'.format(idx)] = ism
return feat_dic
beats = beatTracker.getBeats()
bpm = beatTracker.getBpm()
phase = beatTracker.getPhase()
beats = beats - phase
print 'Bpm: ', bpm
print 'Frame size in samples: ', 44100 * (60.0 / bpm)
# Followed approach from Foote
# Adjust the frame size to the length of a beat, to extract beat-aligned information (zelf-uitgevonden)
FRAME_SIZE = int(44100 * (60.0 / bpm))
HOP_SIZE = FRAME_SIZE / 2
frames_per_second = (44100.0 / FRAME_SIZE) * (FRAME_SIZE / HOP_SIZE)
beats = beats * frames_per_second
spec = Spectrum(size=FRAME_SIZE - FRAME_SIZE % 2)
w = Windowing(type='hann')
spectrum = Spectrum() # FFT would return complex FFT, we only want magnitude
mfcc = MFCC()
pool = Pool()
# Step 0: align audio with phase
beats = beats - 0.5
start_sample = int((phase) * (44100.0 * 60 / bpm))
# Step 1: Calculate framewise MFCC
for frame in FrameGenerator(audio[start_sample:],
frameSize=FRAME_SIZE,
hopSize=HOP_SIZE):
mfcc_bands, mfcc_coeffs = mfcc(
def run(self, audio):
# TODO put this in some util class
# Step 0: calculate the CSD (Complex Spectral Difference) features
# and the associated onset detection function
spec = Spectrum(size=self.FRAME_SIZE)
w = Windowing(type='hann')
fft = FFT()
c2p = CartesianToPolar()
od_csd = OnsetDetection(method='complex')
pool = Pool()
# TODO test faster (numpy) way
for frame in FrameGenerator(audio,
frameSize=self.FRAME_SIZE,
hopSize=self.HOP_SIZE):
mag, phase = c2p(fft(w(frame)))
pool.add('onsets.complex', od_csd(mag, phase))
# Step 1: normalise the data using an adaptive mean threshold
novelty_mean = self.adaptive_mean(pool['onsets.complex'], 16.0)
# Step 2: half-wave rectify the result
novelty_hwr = (pool['onsets.complex'] - novelty_mean).clip(min=0)
# Step 3: then calculate the autocorrelation of this signal
novelty_autocorr = self.autocorr(novelty_hwr)
# Step 4: Sum over constant intervals to detect most likely BPM
valid_bpms = np.arange(self.minBpm, self.maxBpm, self.stepBpm)
for bpm in valid_bpms:
frames = (
np.round(
np.arange(0, np.size(novelty_autocorr),
self.numFramesPerBeat(bpm))).astype('int')
)[:
-1] # Discard last value to prevent reading beyond array (last value rounded up for example)
pool.add('output.bpm',
np.sum(novelty_autocorr[frames]) / np.size(frames))
bpm = valid_bpms[np.argmax(pool['output.bpm'])]
# Step 5: Calculate phase information
valid_phases = np.arange(0.0, 60.0 / bpm,
0.001) # Valid phases in SECONDS
for phase in valid_phases:
# Convert phase from seconds to frames
phase_frames = (phase * 44100.0) / (512.0)
frames = (
np.round(
np.arange(phase_frames, np.size(novelty_hwr),
self.numFramesPerBeat(bpm))).astype('int')
)[:
-1] # Discard last value to prevent reading beyond array (last value rounded up for example)
pool.add('output.phase',
np.sum(novelty_hwr[frames]) / np.size(frames))
phase = valid_phases[np.argmax(pool['output.phase'])]
print 'PHASE', phase
# Step 6: Determine the beat locations
spb = 60. / bpm #seconds per beat
beats = (np.arange(phase, (np.size(audio) / 44100) - spb + phase,
spb).astype('single'))
# Store all the results
self.bpm = bpm
self.phase = phase
self.beats = beats
self.downbeats = self.calculateDownbeats(audio, bpm, phase)
def run(self, audio):
def numFramesPerBeat(bpm):
return (60.0 * self.SAMPLE_RATE) / (self.HOP_SIZE * bpm)
def autocorr(x):
result = np.correlate(x, x, mode='full')
return result[result.size / 2:]
def adaptive_mean(x, N):
return np.convolve(x, [1.0] * int(N), mode='same') / N
# Step 0: calculate the CSD (Complex Spectral Difference) features
# and the associated onset detection function
spec = Spectrum(size=self.FRAME_SIZE)
w = Windowing(type='hann')
fft = np.fft.fft
c2p = CartesianToPolar()
od_csd = OnsetDetection(method='melflux')
pool = Pool()
for frame in FrameGenerator(audio,
frameSize=self.FRAME_SIZE,
hopSize=self.HOP_SIZE):
pool.add('audio.windowed_frames', w(frame))
fft_result = fft(pool['audio.windowed_frames']).astype('complex64')
fft_result_mag = np.absolute(fft_result)
fft_result_ang = np.angle(fft_result)
for mag, phase in zip(fft_result_mag, fft_result_ang):
pool.add('onsets.complex', od_csd(mag, phase))
# Step 1: normalise the data using an adaptive mean threshold
novelty_mean = adaptive_mean(pool['onsets.complex'], 16.0)
# Step 2: half-wave rectify the result
novelty_hwr = (pool['onsets.complex'] - novelty_mean).clip(min=0)
# Step 3: then calculate the autocorrelation of this signal
novelty_autocorr = autocorr(novelty_hwr)
# Step 4: Sum over constant intervals to detect most likely BPM
valid_bpms = np.arange(self.minBpm, self.maxBpm, self.stepBpm)
for bpm in valid_bpms:
frames = (
np.round(
np.arange(0, np.size(novelty_autocorr),
numFramesPerBeat(bpm))).astype('int')
)[:
-1] # Discard last value to prevent reading beyond array (last value rounded up for example)
pool.add('output.bpm',
np.sum(novelty_autocorr[frames]) / np.size(frames))
bpm = valid_bpms[np.argmax(pool['output.bpm'])]
# Step 5: Calculate phase information
valid_phases = np.arange(0.0, 60.0 / bpm,
0.001) # Valid phases in SECONDS
for phase in valid_phases:
# Convert phase from seconds to frames
phase_frames = (phase * 44100.0) / (512.0)
frames = (
np.round(
np.arange(phase_frames, np.size(novelty_hwr),
numFramesPerBeat(bpm))).astype('int')
)[:
-1] # Discard last value to prevent reading beyond array (last value rounded up for example)
pool.add('output.phase',
np.sum(novelty_hwr[frames]) / np.size(frames))
phase = valid_phases[np.argmax(pool['output.phase'])]
# Step 6: Determine the beat locations
spb = 60. / bpm #seconds per beat
beats = (np.arange(phase, (np.size(audio) / 44100) - spb + phase,
spb).astype('single'))
# Store all the results
self.bpm = bpm
self.phase = phase
self.beats = beats
import time import essentia from essentia.standard import Extractor, MonoLoader, Trimmer, Mean, FrameGenerator, Spectrum, SpectralPeaks, Dissonance, BarkBands, Windowing, \ ZeroCrossingRate, OddToEvenHarmonicEnergyRatio, EnergyBand, MetadataReader, OnsetDetection, Onsets, CartesianToPolar, FFT, MFCC, SingleGaussian from build_map import build_map sampleRate = 44100 frameSize = 2048 hopSize = 1024 windowType = "hann" mean = Mean() keyDetector = essentia.standard.Key(pcpSize=12) spectrum = Spectrum() window = Windowing(size=frameSize, zeroPadding=0, type=windowType) mfcc = MFCC() gaussian = SingleGaussian() od = OnsetDetection(method='hfc') fft = FFT() # this gives us a complex FFT c2p = CartesianToPolar() # and this turns it into a pair (magnitude, phase) onsets = Onsets(alpha=1) # dissonance spectralPeaks = SpectralPeaks(sampleRate=sampleRate, orderBy='frequency') dissonance = Dissonance() # barkbands barkbands = BarkBands(sampleRate=sampleRate) # zero crossing rate
def test_windowing(self):
Windowing(type='hann')
def feature_allframes(audio, beats, frame_indexer = None):
# Initialise the algorithms
FRAME_SIZE = 1024
HOP_SIZE = 512
spec = Spectrum(size = FRAME_SIZE)
w = Windowing(type = 'hann')
fft = np.fft.fft
od_csd = OnsetDetection(method = 'complex')
od_hfc = OnsetDetection(method = 'flux')
pool = Pool()
# Calculate onset detection curve on audio
for frame in FrameGenerator(audio, frameSize = FRAME_SIZE, hopSize = HOP_SIZE):
pool.add('windowed_frames', w(frame))
fft_result = fft(pool['windowed_frames']).astype('complex64')
fft_result_mag = np.absolute(fft_result)
fft_result_ang = np.angle(fft_result)
for mag,phase in zip(fft_result_mag, fft_result_ang):
pool.add('onsets.flux', od_hfc(mag, phase))
# Normalize and half-rectify onset detection curve
def adaptive_mean(x, N):
return np.convolve(x, [1.0]*int(N), mode='same')/N
novelty_mean = adaptive_mean(pool['onsets.flux'], 16.0)
novelty_hwr = (pool['onsets.flux'] - novelty_mean).clip(min=0)
novelty_hwr = novelty_hwr / np.average(novelty_hwr)
# For every frame in frame_indexer,
if frame_indexer is None:
frame_indexer = list(range(4,len(beats) - 1)) # Exclude first frame, because it has no predecessor to calculate difference with
# Feature: correlation between current frame onset detection f and of previous frame
# Feature: correlation between current frame onset detection f and of next frame
# Feature: diff between correlation between current frame onset detection f and corr cur and next
onset_integrals = np.zeros((2 * len(beats), 1))
frame_i = (np.array(beats) * 44100.0/ HOP_SIZE).astype('int')
onset_correlations = np.zeros((len(beats), 21))
for i in [i for i in range(len(beats)) if (i in frame_indexer) or (i+1 in frame_indexer)
or (i-1 in frame_indexer) or (i-2 in frame_indexer) or (i-3 in frame_indexer)
or (i-4 in frame_indexer) or (i-5 in frame_indexer) or (i-6 in frame_indexer) or (i-7 in frame_indexer)]:
half_i = int((frame_i[i] + frame_i[i+1]) / 2)
cur_frame_1st_half = novelty_hwr[frame_i[i] : half_i]
cur_frame_2nd_half = novelty_hwr[half_i : frame_i[i+1]]
onset_integrals[2*i] = np.sum(cur_frame_1st_half)
onset_integrals[2*i + 1] = np.sum(cur_frame_2nd_half)
# Step 2: Calculate the cosine distance between the MFCC values
for i in frame_indexer:
onset_correlations[i][0] = max(np.correlate(novelty_hwr[frame_i[i-1] : frame_i[i]], novelty_hwr[frame_i[i] : frame_i[i+1]], mode='valid')) # Only 1 value
onset_correlations[i][1] = max(np.correlate(novelty_hwr[frame_i[i] : frame_i[i+1]], novelty_hwr[frame_i[i+1] : frame_i[i+2]], mode='valid')) # Only 1 value
onset_correlations[i][2] = max(np.correlate(novelty_hwr[frame_i[i] : frame_i[i+1]], novelty_hwr[frame_i[i+2] : frame_i[i+3]], mode='valid')) # Only 1 value
onset_correlations[i][3] = max(np.correlate(novelty_hwr[frame_i[i] : frame_i[i+1]], novelty_hwr[frame_i[i+3] : frame_i[i+4]], mode='valid')) # Only 1 value
# Difference in integrals of novelty curve between frames
# Quantifies the difference in number and prominence of onsets in this frame
onset_correlations[i][4] = onset_integrals[2*i] - onset_integrals[2*i-1]
onset_correlations[i][5] = onset_integrals[2*i+2] + onset_integrals[2*i+3] - onset_integrals[2*i-1] - onset_integrals[2*i-2]
for j in range(1,16):
onset_correlations[i][5 + j] = onset_integrals[2*i + j] - onset_integrals[2*i]
# Include the MFCC coefficients as features
result = onset_correlations[frame_indexer]
return preprocessing.scale(result)