def process(file):
# read in the file
f, sr, enc = wavread(file)
# compute the fourier transform & compute the window times:
D = librosa.stft(f)
times = librosa.frames_to_samples(np.arange(D.shape[1]))
# compute the onset strength envelope:
env = librosa.onset.onset_strength(y=f, sr=sr)
assert (len(times) == len(env))
# compute the onsets we are actually interested in, convert to samples:
onsets = librosa.onset.onset_detect(y=f, sr=sr)
onset_samps = librosa.frames_to_samples(onsets)
assert (onset_samps[-1] <= len(f))
# create a lookup table for retrieving onset strenghts:
lookup = []
prevval = 0
for v in onset_samps:
for i in xrange(prevval, len(times)):
if times[i] == v:
lookup.append(i)
prevval = i + 1
break
# create an empty audio buffer (result):
result = np.zeros(len(f))
# write envelope onset strength values at every onset point
# computed by the envelope:
for i in xrange(len(lookup)):
result[onset_samps[i]] = env[lookup[i]]
# write the result:
wavwrite(result, file[:-4] + '_proc.wav', sr, enc)
return
def gen_hihat(all_data, fs, fps, cand):
fps = librosa.samples_to_frames(fs, hop_length=hop_len, n_fft=win_len)
fps = 100
print(cand)
proc = BeatTrackingProcessor(look_aside=0.2, fps=fps)
act = RNNBeatProcessor()(all_data)
beat_times = proc(act)
song_len = librosa.samples_to_time(data.shape, sr=fs)[0]
hihat = np.zeros(all_data.shape)
idx = np.where(beat_times <= song_len)[0]
new_beat_times = np.zeros(idx.shape)
new_beat_times[idx] = beat_times[idx]
beat_samples = librosa.time_to_samples(new_beat_times, sr=fs)
start = librosa.frames_to_samples(cand[0], hop_len, n_fft=win_len)
end = librosa.frames_to_samples(cand[-1], hop_len, n_fft=win_len)
cand_len = end - start
i = 3
is_hihat = np.zeros(beat_samples.shape)
while i < len(beat_samples):
is_hihat[i] = 1
i = i + 4
for i, s in enumerate(beat_samples):
if is_hihat[i] == 1:
if s + cand_len > hihat.shape:
break
hihat[s:s + cand_len] = data[start:end]
return hihat, new_beat_times, beat_samples
def onsets_and_strength(all_onsets_strength, onsets_sorted, dly_onsets,
strongest_onset, strongest_onset_2, y_cut,
onset_strength):
print(all_onsets_strength)
print(onsets_sorted)
plt.subplot(211)
plt.vlines(librosa.frames_to_samples(dly_onsets), -1.0, 1.0, zorder=2)
plt.vlines(librosa.frames_to_samples(strongest_onset['onset']),
-1.0,
1.0,
colors='red',
zorder=3)
plt.vlines(librosa.frames_to_samples(strongest_onset_2['onset']),
-1.0,
1.0,
colors='green',
zorder=3)
plt.plot(y_cut, zorder=1)
plt.ylabel('Amplitude in Floating Point')
plt.xlabel('Samples')
plt.title('Onset Detection with Delay Effect')
plt.subplot(212)
plt.plot(onset_strength[0])
plt.ylabel('Onset Stength')
plt.xlabel('Frames')
plt.show()
def strip_audio(x, frame_length=1024, hop_length=256, rms_ths=0.2):
# compute energy
rmse = librosa.feature.rmse(x,
frame_length=frame_length,
hop_length=hop_length)[0]
rms_ratio = rmse / rmse.max()
active_frames = np.nonzero(rms_ratio > rms_ths)[0]
assert len(active_frames) > 0, "there is no voice part in the wav"
# strip continous active part
s_sample = librosa.frames_to_samples(active_frames[0],
hop_length=hop_length)[0]
e_sample = librosa.frames_to_samples(active_frames[-1],
hop_length=hop_length)[0]
# plot the rmse on the wavelet of x
# frames = range(len(energy))
# import matplot.pyplot as plt
# energy = np.array([
# sum(abs(x[i:i+frame_length]**2))
# for i in range(0, len(x), hop_length)
# ])
# t = librosa.frames_to_time(frames, sr=sr, hop_length=hop_length)
# librosa.display.waveplot(x, sr=sr, alpha=0.4)
# plt.plot(t, energy/energy.max(), 'r--') # normalized for visualization
# plt.plot(t[:len(rmse)], rmse/rmse.max(), color='g') # normalized for visualization
# plt.legend(('Energy', 'RMSE'))
return x[s_sample:e_sample]
def beat_match(song1, song2, sr):
"""
Creates two lists of length equal to the combined length of both songs. The first list is zero padded from the
end of the first song until the end of the second song. The second list is zero padded from the beginning of the
first song until the first beat of the last phrase of that same song. The second song is then appended to the second
list. The lists are then added together.
Input Parameters
------------------------
song1: 1-D array containing sample points of first song
song2: 1-D array containing sample points for second song
sr: integer representing the rate at which the song is being sampled
Returns
------------------------
a 1-D array containing a syncronized mixture of both songs
"""
print('begin beatmatch')
tempo1, beat1 = beat_track(song1)
tempo2, beat2 = beat_track(song2)
beat1 = librosa.frames_to_samples(beat1)
beat2 = librosa.frames_to_samples(beat2)
song2 = song2[beat2[0]:]
phrases1 = len(beat1)
fade_start = phrases1 - 32
fade_sample = beat1[fade_start]
fade_out_start = fade_sample
fade_out_end = len(song2)
phrases2 = len(beat2)
fade_in_start = len(song1[:fade_sample])
fade_in_end = fade_in_start + phrases2
song2 = fade(song2, type="in", end=beat2[32])
zeros2 = np.zeros(len(song1[:fade_sample]), dtype=np.float32)
list2 = np.append(zeros2, song2)
#list2 = fade(list2, type= "in", start = fade_in_start, end = fade_in_end)
song1 = fade(song1, type="out", start=fade_out_start)
zeros1 = np.zeros((len(song2) - len(song1[fade_sample:])),
dtype=np.float32)
list1 = np.append(song1, zeros1)
#list1 = fade(list1, type= "out", start = fade_out_start, end = fade_out_end)
mix = list1 + list2
print('end beatmatch')
return mix
def beat_match(song1, song2, sr):
"""
Creates two lists of length equal to the combined length of both songs. The first list is zero padded from the
end of the first song until the end of the second song. The second list is zero padded from the beginning of the
first song until the first beat of the last phrase of that same song. The second song is then appended to the second
list. The lists are then added together.
Input Parameters
------------------------
song1: 1-D array containing sample points of first song
song2: 1-D array containing sample points for second song
sr: integer representing the rate at which the song is being sampled
Returns
------------------------
a 1-D array containing a syncronized mixture of both songs
"""
print('begin beatmatch')
tempo1, beat1 = beat_track(song1)
tempo2, beat2 = beat_track(song2)
beat1 = librosa.frames_to_samples(beat1)
beat2 = librosa.frames_to_samples(beat2)
song2 = song2[beat2[0]:]
phrases1 = len(beat1)
fade_start = phrases1 - 32
fade_sample = beat1[fade_start]
fade_out_start = fade_sample
fade_out_end = len(song2)
phrases2 = len(beat2)
fade_in_start = len(song1[:fade_sample])
fade_in_end = fade_in_start + phrases2
song2 = fade(song2, type = "in", end = beat2[32])
zeros2 = np.zeros(len(song1[:fade_sample]), dtype = np.float32)
list2 = np.append(zeros2, song2)
#list2 = fade(list2, type= "in", start = fade_in_start, end = fade_in_end)
song1 = fade(song1, type = "out", start = fade_out_start)
zeros1 = np.zeros((len(song2)-len(song1[fade_sample:])), dtype = np.float32)
list1 = np.append(song1, zeros1)
#list1 = fade(list1, type= "out", start = fade_out_start, end = fade_out_end)
mix = list1 + list2
print('end beatmatch')
return mix
def restretch(self):
if self.final_offset >= 0:
offset_frame = librosa.frames_to_samples(self.final_offset)
noise = np.zeros(offset_frame)
self.restretch_data = np.concatenate((noise, self.y_2), axis=0)
else:
offset_frame = librosa.frames_to_samples(-self.final_offset)
# noise = np.zeros(offset_frame)
self.restretch_data = self.y_2[offset_frame:]
padding = np.zeros(len(self.y) - len(self.restretch_data))
self.final_vocal_audio = np.concatenate((self.restretch_data,padding), axis=0)
def feature_extract_blues(blues_track,
sr,
current_timesig,
onset_threshold=0.7):
# get rhythm overlay
hop_length = 512
blues_harm, blues_perc = librosa.effects.hpss(blues_track,
margin=(1.0, 5.0))
onset_env = librosa.onset.onset_strength(blues_perc,
sr=sr,
aggregate=np.median)
_, beats = librosa.beat.beat_track(onset_envelope=onset_env, sr=sr)
times = librosa.frames_to_time(np.arange(len(onset_env)),
sr=sr,
hop_length=hop_length)
prev_val = 0
for i, b in enumerate(beats[:-3]):
# get the corresponding onset env value
t_b = times[b]
on_f_b = librosa.time_to_frames([t_b], sr=sr, hop_length=hop_length)
if librosa.util.normalize(onset_env)[on_f_b] >= prev_val:
prev_val = librosa.util.normalize(onset_env)[on_f_b]
keep_beat_start = b
keep_beat_end = beats[i + 3]
alert_start = i
beat_start = librosa.frames_to_samples([keep_beat_start])[0]
beat_end = librosa.frames_to_samples([keep_beat_end])[0]
overlay_sample = blues_perc[beat_start:beat_end]
# get beat samples
beat_samples = librosa.frames_to_samples(beats, hop_length=hop_length)
# get extracted subsample - using VS pipeline
try:
rep_samples_audio, num_seg = extract.extract_sample(blues_harm, sr, 1)
signal_sample = rep_samples_audio[0][0]
except:
print "Could not extract sample from VS Pipeline, using default.."
mdpt = int(len(blues_harm) / 2)
signal_sample = blues_harm[mdpt:mdpt + sr]
return {
'overlay': overlay_sample,
'beats': beat_samples,
'alert': signal_sample
}
def slice_long_sample(y, sr, declick_samples=15, length_limit=None, fname=''):
if length_limit and len(y) / sr > length_limit:
y = y[0:length_limit * sr]
onsets = rosa.onset.onset_detect(y=y, sr=sr, backtrack=True)
onset_times = rosa.frames_to_samples(onsets)
onset_times = np.concatenate([onset_times, [len(y)]])
segmented = [
y[onset_times[n]:onset_times[n + 1]]
for n in range(len(onset_times) - 1)
]
segmented = [s for s in segmented if len(s) >= declick_samples]
if declick_samples > 1:
declick_envelope = np.linspace(1 / declick_samples,
1 - (1 / declick_samples),
declick_samples)
for i in range(len(segmented)):
segmented[i][0:declick_samples] *= declick_envelope
slices = []
for i, s in enumerate(segmented):
if not i % poll_every and i > 1:
print(
rf'calculating features for slice {i}/{len(segmented)} of {fname}...'
)
slices.append(ausl.AudioSlice(s, sr, fname))
return slices, onset_times
def onset(x, sr):
# Short-time Fourier transform (for EQ, must do inverse Fourier transform after)
X = librosa.stft(x)
# Find the frames when onsets occur
onset_frames = librosa.onset.onset_detect(x, sr=sr)
print("Onset Frames = " + str(onset_frames) + "\n ")
# Find the times, in seconds, when onsets occur in the audio signal
onset_times = librosa.frames_to_time(onset_frames, sr=sr)
print("Onset Times = " + str(onset_times) + "\n ")
# Convert the onset frames into sample indices to play "BEEB" sound on it
onset_samples = librosa.frames_to_samples(onset_frames)
print("Onset Samples = " + str(onset_samples) + "\n ")
# Use the "length" parameter so the click track is the same length as the original signal
clicks = librosa.clicks(times=onset_times, length=len(x))
# Play the click track "added to" the original signal
sd.play(x + clicks, sr)
# Display the waveform of the original signal
librosa.display.waveplot(x, sr)
plt.title("Original Signal")
plt.show() # Close window to resume
return onset_frames, onset_times, onset_samples
def __init__(self,
dataset,
sr=22050,
frameSize=2048,
hopSize=512,
transform=None,
cacheSize=4):
self.dataset = dataset
self.sr = sr
self.frameSize = frameSize
self.hopSize = hopSize
self.transform = transform
self.cacheSize = cacheSize
self.frameDt = float(frameSize) / sr
# count frames in dataset
nFramesList = []
for pathPair in dataset.pathPairs:
wavPath = pathPair.wav
duration = librosa.get_duration(filename=wavPath)
nSamples = librosa.time_to_samples(duration, sr=self.sr)
nFrames = 1 + int(
(nSamples - self.frameSize) / float(self.hopSize))
nFramesList.append(nFrames)
# check validation
sStart = librosa.frames_to_samples(nFrames - 1,
hop_length=self.hopSize)
sEnd = sStart + self.frameSize
assert (nSamples > 0) and (
sEnd <= nSamples), f'{nFrames}:{sStart}_{sEnd}, {nSamples}'
self.frameCumsum = np.cumsum(nFramesList)
# FIFO cache
self._sampleCache = deque(maxlen=cacheSize)
self._sampleIdxCache = deque(maxlen=cacheSize)
def generate_sine_midi_note(f0_info, sr, n_duration):
f0 = f0_info[0]
A = remap(f0_info[1], CdB.min(), CdB.max(), 0, 1)
duration = librosa.frames_to_time(n_duration, sr=fs, hop_length=hop_length)
# Generate music21 note
note_duration = 0.02 * np.around(
duration / 2 /
0.02) # Round to 2 decimal places for music21 compatibility
midi_velocity = int(round(remap(f0_info[1], CdB.min(), CdB.max(), 0, 127)))
if f0 == None:
try:
note_info = Rest(type=mm.secondsToDuration(note_duration).type)
except DurationException:
note_info = None
f0 = 0
else:
midi_note = round(librosa.hz_to_midi(f0))
try:
note = Note(midi_note,
type=mm.secondsToDuration(note_duration).type)
note.volume.velocity = midi_velocity
note_info = [note]
except DurationException:
note_info = None
if note_info is None:
return None
# Generate Sinewave
n = np.arange(librosa.frames_to_samples(n_duration, hop_length=hop_length))
sine_wave = A * np.sin(2 * np.pi * f0 * n / float(sr))
return [sine_wave, note_info]
def strip(y, frame_length, hop_length=512):
""" Removing leading silence from an audio track
:param y: (np.ndarray) audio signal
:param frame_length: (int)
:param hop_length: (int)
:return: Audio signal with leading silence removed
"""
# compute RMSE.
rms = librosa.feature.rms(y,
frame_length=frame_length,
hop_length=hop_length,
center=True)
# identify the first frame index where RMSE exceeds a threshold.
thresh = 0.01
frame_index = 0
while rms[0][frame_index] < thresh:
frame_index += 1
# convert units of frames to samples.
start_sample_index = librosa.frames_to_samples(frame_index,
hop_length=hop_length)
# return the trimmed signal.
return y[start_sample_index:]
def slicer(song, n_beats=16, duration=0):
'''
Takes in a song and its segments and computes the largest total segment in the dictionary.
To do this it sums up each of the dictionary entries using that disgusting(tm) comprehension below.
The segment has to be larger than the given duration in order to be considered in the sum.
It then takes the max dictionary entry and returns the segment with the bounds.
:param song: (Song) | song to slice
:param duration: (float) | min duration (in seconds)
:return: slice (Slice) | segmented slice
'''
largest_seg = max(
song.segments.items(),
key=lambda x: sum(
[z[1] - z[0] for z in x[1] if z[1] - z[0] >= duration]))[1]
max_pair = tuple(max(largest_seg, key=lambda pair: pair[1] - pair[0]))
slice = Slice(song.path, offset=max_pair[0], duration=max_pair[1])
perc_y = librosa.effects.percussive(slice.y)
beat_track = beatTrack(y=perc_y, sr=song.load.sr)
end_frame = librosa.frames_to_samples(beat_track.beats[n_beats])[0]
slice.y = slice.y[:end_frame]
return slice
def generate_note(self, f0_info, n_duration, round_to_sixteenth=True):
f0 = f0_info[0]
a = remap(f0_info[1], self.cqt.min(), self.cqt.max(), 0, 1)
duration = librosa.frames_to_time(n_duration, sr=self.sr, hop_length=self.hop_length)
note_duration = 0.02 * np.around(duration / 0.02) # Round to 2 decimal places for music21 compatibility
midi_duration = second_to_quarter(duration, self.tempo)
midi_velocity = int(round(remap(f0_info[1], self.cqt.min(), self.cqt.max(), 80, 120)))
if round_to_sixteenth:
midi_duration = round(midi_duration * 16) / 16
try:
if f0 is None:
midi_note = None
note_info = Rest(type=self.mm.secondsToDuration(note_duration).type)
f0 = 0
else:
midi_note = round(librosa.hz_to_midi(f0))
note = Note(librosa.midi_to_note(midi_note), type=self.mm.secondsToDuration(note_duration).type)
note.volume.velocity = midi_velocity
note_info = [note]
except DurationException:
if f0 is None:
midi_note = None
note_info = Rest(type='32nd')
f0 = 0
else:
midi_note = round(librosa.hz_to_midi(f0))
note = Note(librosa.midi_to_note(midi_note),
type='eighth')
note.volume.velocity = midi_velocity
note_info = [note]
midi_info = [midi_note, midi_duration, midi_velocity]
n = np.arange(librosa.frames_to_samples(n_duration, hop_length=self.hop_length))
sine_wave = a * np.sin(2 * np.pi * f0 * n / float(self.sr))
return [sine_wave, midi_info, note_info]
def find_localmax(
self,
signal,
noise_threshold=0.0, # Range: [0.0, 1.0].
jump=None,
frame_length=1024):
""" """
if not librosa_available:
print('ERROR: Error in find_localmax. Librosa not installed.')
index_list = []
return index_list
# Adjust for comparable results for low sampling rates.
if self.sampling_freq < 300000:
frame_length = int(frame_length / 2)
if jump is None:
jump = int(self.sampling_freq / 1000) # Default = 1 ms.
y = signal.copy()
if noise_threshold > 0.0:
y[(np.abs(y) < noise_threshold)] = 0.0
rmse = librosa.feature.rmse(y=y,
hop_length=jump,
frame_length=frame_length,
center=True)
locmax = librosa.util.localmax(rmse.T)
maxindexlist = [index for index, a in enumerate(locmax) if a == True]
# Original index list is related to jump length. Convert.
index_list = librosa.frames_to_samples(maxindexlist, hop_length=jump)
#
return index_list
def beats_to_sample(beats, y, sr):
"""
Aligning supposed beats to the peak of energy in the y signal
beats: np.ndarray
frames index where beats are supposed to be
y: np.ndarray
input signal
sr: int
samplerate
Returns
y_beat: np.ndarray
array with 1 when there is a beat
beats_indices: np.ndarray
array with indices of the beats
"""
y_beat = np.zeros(y.shape)
margin = int(0.1 * sr)
for beat in frames_to_samples(beats):
bs_index = beat - margin + np.argmax(
np.abs(y[beat - margin:beat + margin]))
y_beat[bs_index] = 1
return y_beat, np.where(y_beat == 1)[0]
def __test(x, y, hop_length, n_fft):
y_test = librosa.frames_to_samples(x,
hop_length=hop_length,
n_fft=n_fft)
assert np.allclose(y, y_test)
y = np.asanyarray(y)
assert y.shape == y_test.shape
assert y.ndim == y_test.ndim
def get_beats_samples(y, sr): # 起点强度（音符按键起始点） onset_env = librosa.onset.onset_strength(y=y, sr=sr) # 节拍点（帧索引） _, beats = librosa.beat.beat_track(onset_envelope=onset_env, sr=sr, hop_length=512) # 节拍点（采样引） beat_samples = librosa.frames_to_samples(beats, hop_length=512) return beats, beat_samples
def onset_detect(audio_vector, sr):
"""Returns the onsets detected of a given audio file"""
####### Onset and Tempo detection ##########
onset = librosa.onset.onset_detect(y=audio_vector, sr=sr, backtrack=True)
onset_env = librosa.onset.onset_strength(y=audio_vector, sr=sr)
onset_sec = librosa.frames_to_time(onset, sr=sr)
onset_frames = librosa.frames_to_samples(onset)
return onset_frames
def apply_trim_offset(self, frame):
return (
librosa.samples_to_frames(
librosa.frames_to_samples(frame) + self.trim_offset
)
if self.trim_offset
else frame
)
def __test(x, y, hop_length, n_fft):
y_test = librosa.frames_to_samples(x,
hop_length=hop_length,
n_fft=n_fft)
assert np.allclose(y, y_test)
y = np.asanyarray(y)
assert y.shape == y_test.shape
assert y.ndim == y_test.ndim
def overlap(siga, sigb, overlap_beats, sr, fade):
# Trim leading and trailing silence
siga = np.trim_zeros(siga)
sigb = np.trim_zeros(sigb)
# Get beat frames for each track and convert to track sample indices
atempo, abeatframes = librosa.beat.beat_track(y=siga, sr=sr)
abeats = librosa.frames_to_samples(abeatframes)
btempo, bbeatframes = librosa.beat.beat_track(y=sigb, sr=sr)
bbeats = librosa.frames_to_samples(bbeatframes)
# print atempo
# print btempo
# print siga.shape
# print sigb.shape
# print abeats.shape
# print bbeats.shape
# If fade is specified, cross-fade both tracks into each other
if (fade):
print "Fading tracks"
fadeindices = int(bbeats[overlap_beats])
fade = np.linspace(0, 1, num=fadeindices + 1)
for i in range(0, fade.shape[0]):
sigb[i] *= fade[i]
siga[siga.shape[0] - 1 - i] *= fade[i]
# print "Fade indicies: ",int(bbeats[overlap_beats])
# Create the output signal
mix = np.zeros(sigb.shape[0] + siga.shape[0])
# Prep it with the first track
mix[:siga.shape[0]] = siga
# The time frame of the beat where the second track should start
startframe = abeats[abeats.shape[0] - overlap_beats + 4]
# print "Start frames: ",startframe,startframe-bbeats[3],startframe-bbeats[3]+sigb.shape[0]
# print "Time beyond end of a ",(startframe-bbeats[3]+sigb.shape[0]-siga.shape[0])/44100.
# for i in range(overlap_beats):
# print abeats[abeats.shape[0]-overlap_beats+i]-abeats[abeats.shape[0]-overlap_beats+i-1], bbeats[i+1]-bbeats[i]
mix[startframe - bbeats[3]:startframe - bbeats[3] + sigb.shape[0]] += sigb
mix = np.trim_zeros(mix)
# print "Shape of mix ",mix.shape
return mix
def compute_segments_librosa(Y, sr, numparts):
myprint('Computing parts segmentation')
bounds = librosa.segment.agglomerative(Y, numparts)
bound_times = librosa.frames_to_time(bounds, sr=sr)
bound_samples = librosa.frames_to_samples(bounds,
hop_length=512,
n_fft=2048)
myprint('bound_samples = %s / %s' % (bound_samples.shape, bound_samples))
return bounds, bound_times, bound_samples
def analyze_signals(original_signal, test_signal, hop_size, start_time, end_time, sr, start_bpm, offbeat_factor):
# Offbeat_factor is how much off the beat the person is allowed to be
test, sr = librosa.load(test_signal, sr)
test_normalizer = np.max(test)
test = test/float(test_normalizer)
original, sr = librosa.load(original_signal, sr)
original_normalizer = np.max(original)
original = original/float(original_normalizer)
test_onset_env, test_beat_frames = estimated_beat(test_signal, hop_size, start_time, end_time, sr, start_bpm)
original_onset_env, original_beat_frames = estimated_beat(original_signal, hop_size, start_time, end_time, sr, start_bpm)
# plt.show()
test_beats = librosa.frames_to_samples(test_beat_frames, hop_length=hop_size)
original_beats = librosa.frames_to_samples(original_beat_frames, hop_length=hop_size)
beat_score = calculate_rank(original_beats, test_beats, sr, sr*offbeat_factor)
plt.figure(1)
ax1 = plt.subplot(2,1,1)
plt.plot(original, label='Signal')
plt.vlines(original_beats, -2, 2, alpha=.5, color='r',
linestyle='solid', linewidth=3, label='Beats')
plt.legend(frameon=True, framealpha=0.75)
# Limit the plot to a X-second window
plt.xlim([start_time * sr, end_time * sr])
plt.xticks(np.linspace(start_time, end_time, 5) * sr,
np.linspace(start_time, end_time, 5))
plt.xlabel('Time (s)')
plt.tight_layout()
ax2 = plt.subplot(2,1,2, sharex=ax1, sharey=ax1)
plt.plot(test, label='Signal')
plt.vlines(test_beats, -2, 2, alpha=.5, color='g',
linestyle='solid',linewidth=3, label='Beats')
plt.legend(frameon=True, framealpha=0.75)
# Limit the plot to a X-second window
plt.xlim([start_time * sr, end_time * sr])
plt.xticks(np.linspace(start_time, end_time, 5) * sr,
np.linspace(start_time, end_time, 5))
plt.xlabel('Time (s)')
plt.tight_layout()
plt.subplots_adjust(hspace=0)
plt.show()
return beat_score
def split_song(mix_in, mix_out, bpm, file_name):
file_path = './data/mp3/' + file_name
audio, _ = librosa.load(file_path, sr=SR)
_, beats = librosa.beat.beat_track(y=audio, sr=SR, bpm=bpm)
body = librosa.frames_to_samples(beats[mix_in + MIX_LEN:mix_out])
trans_in = librosa.frames_to_samples(beats[mix_in:mix_in + MIX_LEN])
trans_out = librosa.frames_to_samples(beats[mix_out:mix_out + MIX_LEN])
body_audio = audio[body[0]:body[-1]]
in_audio = audio[trans_in[0]:trans_in[-1]]
out_audio = audio[trans_out[0]:trans_out[-1]]
file_name = file_name.split('.')[0]
librosa.output.write_wav('./data/chopped/body/' + file_name + '.wav',
body_audio, SR)
librosa.output.write_wav('./data/chopped/trans_in/' + file_name + '.wav',
in_audio, SR)
librosa.output.write_wav('./data/chopped/trans_out/' + file_name + '.wav',
out_audio, SR)
def splitSamples(sourceDir, outputDir, mode=None):
print('- Splitting samples for dataset:', mode)
source_filelist = os.listdir(sourceDir)
for f in source_filelist:
print('processing', f)
outputSamples = os.path.join(outputDir, 'dataset_' + mode)
y, sr = librosa.load(os.path.join(sourceDir, f))
# trim silence at beginning and end
y, index = librosa.effects.trim(y)
# detect onsets
o_env = librosa.onset.onset_strength(y, sr=sr, feature=librosa.cqt)
onset_frames = librosa.onset.onset_detect(onset_envelope=o_env, sr=sr)
vectors = []
words = []
filenames = []
onset_samples = list(librosa.frames_to_samples(onset_frames))
onset_samples = np.concatenate(onset_samples, len(y))
starts = onset_samples[0:-1]
stops = onset_samples[1:]
analysis_folder = sourceDir
samples_folder = os.path.join(outputSamples, f)
try:
os.makedirs(samples_folder)
except:
pass
pbar = ProgressBar()
for i, (start, stop) in enumerate(pbar(zip(starts, stops))):
audio = y[start:stop]
filename = os.path.join(samples_folder, str(i) + '.wav')
librosa.output.write_wav(filename, audio, sr)
vector = get_fingerprint(audio, sr=sr)
word = basename(filename)
vectors.append(vector)
words.append(word)
filenames.append(filename)
np.savetxt(os.path.join(analysis_folder, 'vectors'),
vectors,
fmt='%.5f',
delimiter='\t')
np.savetxt(os.path.join(analysis_folder, 'words'), words, fmt='%s')
np.savetxt(os.path.join(analysis_folder, 'filenames.txt'),
filenames,
fmt='%s')
def feature_extract_jazz(jazz_track, sr, num_segments=8, seg_thresh=3):
# segment boundaries
mfcc = librosa.feature.mfcc(y=jazz_track, sr=sr)
bounds = librosa.segment.agglomerative(mfcc, num_segments)
sample_bounds = librosa.frames_to_samples(bounds)
sample_intervals = boundaries_to_intervals(sample_bounds)
# clean up short segments
del_list = []
for i, intr in enumerate(sample_intervals):
if intr[1] - intr[0] < seg_thresh * sr:
del_list.append(i)
sample_intervals = np.delete(sample_intervals, del_list, axis=0)
# corresponding intervals
# TODO: determine segment key/ progression
shift_by = []
# extracted subsample - using VS pipeline
jazz_harm, jazz_perc = librosa.effects.hpss(jazz_track)
try:
rep_samples_audio, num_seg = extract.extract_sample(jazz_harm, sr, 1)
signal_sample = rep_samples_audio[0][0]
except:
print "Could not extract sample from VS Pipeline, using default.."
mdpt = int(len(jazz_harm) / 2)
signal_sample = jazz_harm[mdpt:mdpt + sr]
# extract beats to overlay VS sample
onset_env = librosa.onset.onset_strength(jazz_perc,
sr=sr,
aggregate=np.median)
_, beats = librosa.beat.beat_track(onset_envelope=onset_env, sr=sr)
beat_samples = librosa.frames_to_samples(beats)
return {
'bounds': sample_intervals,
'shift': shift_by,
'alert': signal_sample,
'beats': beat_samples
}
def print_tf(tf, rate, file=sys.stdout):
fprint = lambda *args, **kwargs: print(*args, file=file, **kwargs)
fprint(tf.shape[1], end="")
for i in range(tf.shape[1]):
fprint(' {}'.format(librosa.frames_to_samples(i) / rate), end="")
fprint()
for i, freq in enumerate(librosa.fft_frequencies(sr=rate)):
row = tf[i]
fprint(freq, end="")
for x in tf[i]:
fprint(' {}'.format(np.abs(x)), end="")
fprint()
def test_frames_to_samples(frames, hop_length, n_fft):
samples = librosa.frames_to_samples(frames,
hop_length=hop_length,
n_fft=n_fft)
frames = np.asanyarray(frames)
assert frames.shape == samples.shape
assert frames.ndim == samples.ndim
if n_fft is None:
assert np.allclose(samples, frames * hop_length)
else:
assert np.allclose((samples - n_fft // 2) // hop_length, frames)
def get_downbeats(y, tempo, beat_frames, sr):
measures = len(beat_frames) // BEATS
beat_frames = librosa.samples_to_frames(beat_frames)
onset_env = librosa.onset.onset_strength(y, sr=sr, aggregate=np.median)
beat_strengths = onset_env[beat_frames]
measure_beat_strengths = beat_strengths[:measures * BEATS].reshape(
-1, BEATS)
beat_pos_strength = np.sum(measure_beat_strengths, axis=0)
downbeat_pos = np.argmax(beat_pos_strength)
full_measure_beats = beat_frames[:measures * BEATS].reshape(-1, BEATS)
downbeat_frames = full_measure_beats[:, downbeat_pos]
return librosa.frames_to_samples(downbeat_frames)
def strip(signal, frame_length=512, hop_length=256):
# Compute RMSE.
rmse = librosa.feature.rms(signal,
frame_length=frame_length,
hop_length=hop_length,
center=True)
# Identify the first frame index where RMSE exceeds a threshold.
thresh = 0.001
frame_index = 0
while rmse[0][frame_index] < thresh:
frame_index += 1
# Convert units of frames to samples.
start_sample_index = librosa.frames_to_samples(frame_index,
hop_length=hop_length)
signal = signal[start_sample_index:]
signal = np.array(list(signal)[::-1])
# Compute RMSE.
rmse = librosa.feature.rms(signal,
frame_length=frame_length,
hop_length=hop_length,
center=True)
# Identify the first frame index where RMSE exceeds a threshold.
thresh = 0.001
frame_index = 0
while rmse[0][frame_index] < thresh:
frame_index += 1
# Convert units of frames to samples.
start_sample_index = librosa.frames_to_samples(frame_index,
hop_length=hop_length)
signal = np.array(signal[start_sample_index:])
# Return the trimmed signal.
return np.array(list(signal)[::-1])
def reconstruct(features, n_fft=2048, sr=22050, hop_length=None):
if (hop_length == None):
hop_length = n_fft / 4
# will be a fraction shorter than the original
wave = np.zeros(lr.frames_to_samples(features.shape[0], hop_length,
n_fft)[0],
dtype=np.float32)
for frame, feature_slice in enumerate(features):
sample_start = lr.frames_to_samples(frame, hop_length, n_fft)[0]
wave_slice = reconstruct_slice(feature_slice, n_fft, sr)
sample_end = sample_start + len(wave_slice)
# not too sure about this
if len(wave[sample_start:sample_end]) < len(wave_slice):
wave_slice = wave_slice[:len(wave[sample_start:sample_end])]
wave[sample_start:sample_end] += wave_slice * np.hanning(
len(wave_slice)) # do I need to scale?
return wave
def process_audio(filename, frame_size, mel_bands, fmax, display):
""" act audio metadata and compute the dynamic spectrogram.
Prepare the audio file and process it to compute the dynamic spectrograms
block by block.
Args:
filename (str): Path to the audio file.
frame_size (int): frame size for the "per block" processing.
display_plot (boolean): display or save the plot.
Returns:
None
Todo:
- check if the samplerate of the file corresponds to the samplerate
in the configuration file
Note:
According to the 2016 base line code, the frame size is 40ms with
a hop size of 50%.
"""
samples = librosa.frames_to_samples(frame_size)
chan_nb, samplerate = extract_audio_data(filename)
counter = 0
for block in sfblocks(filename, blocksize=samples[0]):
counter += 1
# separate the channels to compute the spectrograms
for chan in np.arange(chan_nb):
if chan_nb < 2:
y = block
else:
y = block[:, chan]
# Compute the dynamic spectrogram
dynamic_spectrogram(
y,
filename,
block_nb=counter,
display=display)
static_spectrogram(
y,
filename,
counter,
mel_bands,
fmax, display=display)
def get_drum_wav(percussion, width=5, n=None):
# Compute volume shaper
percussion = librosa.util.normalize(percussion.ravel())
v = scipy.ndimage.median_filter(percussion,
width,
mode='mirror')
v = np.atleast_2d(v)
wav = synthesize(librosa.frames_to_samples(np.arange(v.shape[-1]),
hop_length=hop_length),
v,
fmin=librosa.midi_to_hz(0),
bins_per_octave=12,
wave=noise,
n=n)[0]
return wav
def __test(times, frames, sr, hop_length, click_freq, click_duration, click, length):
y = librosa.clicks(times=times,
frames=frames,
sr=sr,
hop_length=hop_length,
click_freq=click_freq,
click_duration=click_duration,
click=click,
length=length)
if times is not None:
nmax = librosa.time_to_samples(times, sr=sr).max()
else:
nmax = librosa.frames_to_samples(frames, hop_length=hop_length).max()
if length is not None:
assert len(y) == length
elif click is not None:
assert len(y) == nmax + len(click)
def get_wav(cq, nmin=60, nmax=120, width=5, max_peaks=1, wave=None, n=None):
# Slice down to the bass range
cq = cq[nmin:nmax]
# Pick peaks at each time
mask = peakgram(librosa.logamplitude(cq**2, top_db=60, ref_power=np.max),
max_peaks=max_peaks)
# Smooth in time
mask = scipy.ndimage.median_filter(mask,
size=(1, width),
mode='mirror')
# resynthesize with some magnitude compression
wav = synthesize(librosa.frames_to_samples(np.arange(cq.shape[-1]),
hop_length=hop_length),
mask * cq**(1./3),
fmin=librosa.midi_to_hz(nmin + MIDI_MIN),
bins_per_octave=12,
wave=wave,
n=n)[0]
return wav
#THE KEY FUNCTION of seperation *******
y_harmonic, y_percussive = librosa.effects.hpss(y)
x_harmonic, x_percussive = librosa.effects.hpss(x)
#beats
tempo_y, beats_y = librosa.beat.beat_track(y=y_percussive, sr=sr_y, trim=True)
tempo_x, beats_x = librosa.beat.beat_track(y=x_percussive, sr=sr_x, trim=True)
#adjust x to be the same tempo as y
ym = librosa.effects.time_stretch(y, tempo_x/tempo_y)
#remeasure tempo of y_matched
ym_harmonic, ym_percussive = librosa.effects.hpss(ym)
tempo_ym, beats_ym = librosa.beat.beat_track(y=ym_percussive, sr=sr_y, trim=True)
#PHASE the tracks
#get arrays of the sample indices of the beats
beats_i_x = librosa.frames_to_samples(beats_x)
beats_i_ym = librosa.frames_to_samples(beats_ym)
#cut off the tracks at the beats
x = x[beats_i_x[1]:]
ym = ym[beats_i_ym[1]:]
#mix the matched tracks
mix = np.array([(x + y)/2 for x, y in zip(ym, x)],dtype=np.float32)
#input array must be in numpy.float32!
librosa.output.write_wav('mixes/beat_matched.wav', mix, sr_y)
#LOAD OR CREATE S-MATRIX & NOVELTY VECTOR
s_matrix = init_smatrix(file_id,f,r, sample_duration)
novelty = init_novelty_vector(file_id, w, w_f, sample_duration, s_matrix)
#https://bmcfee.github.io/librosa/generated/librosa.util.peak_pick.html?
#TODO correlate to the beat, somehow
w_p = w_f/w_p_ratio
peaks = librosa.util.peak_pick(novelty, w_p, w_p, w_p, w_p, peak_window, w_p)
#cross reference beats and peaks
#peaks = cross_reference(beats, peaks, beat_threshold)
#assuming music is periodic...
peaks = filter_by_period(peaks, period_threshold,fpb)
#Sample a test segment
p_s = librosa.frames_to_samples(peaks)
if (len(p_s) > 2):
sample = y[p_s[1]:p_s[2]]
librosa.output.write_wav('mixes/sampled.wav', sample, sr)
loop = np.concatenate([sample,sample,sample])
librosa.output.write_wav('mixes/loop.wav', loop, sr)
#Shuffle a test segment
p_s = librosa.frames_to_samples(peaks)
if (len(p_s) >= 4):
s1 = y[p_s[0]:p_s[1]]
s2 = y[p_s[1]:p_s[2]]
s3 = y[p_s[2]:p_s[3]]
loop = np.concatenate([s3,s2,s1])
librosa.output.write_wav('mixes/shuffle.wav', loop, sr)
def __test(x, y, hop_length, n_fft):
y_test = librosa.frames_to_samples(x,
hop_length=hop_length,
n_fft=n_fft)
assert np.allclose(y, y_test)
from analysis.pitch import * from analysis.util import * file_id = 'all' audio_path = 'assets/'+file_id+'.wav' #sr = None disables resampling y, sr = (librosa.load(audio_path, sr=None,duration=40.0)) #THE KEY FUNCTION of seperation ******* y_harmonic, y_percussive = librosa.effects.hpss(y) tempo, beats = librosa.beat.beat_track(y=y_percussive, sr=sr, trim=False) #PHRASE DETECT phrases = get_phrase_intervals(file_id,y,sr, 1.0 , 1.0, 4.0, 15, 0.13, tempo) s_phrases = librosa.frames_to_samples(phrases) y1 = np.array(y[s_phrases[1]:s_phrases[2]]) y2 = np.array(y[s_phrases[3]:s_phrases[4]]) #THE KEY FUNCTION of seperation ******* y_harmonic, y_percussive = librosa.effects.hpss(y1) y_harmonic_2, y_percussive_2 = librosa.effects.hpss(y2) tempo, beats = librosa.beat.beat_track(y=y_percussive, sr=sr, trim=False) # We'll use a CQT-based chromagram here. An STFT-based implementation also exists in chroma_cqt() # We'll use the harmonic component to avoid pollution from transients C = librosa.feature.chroma_cqt(y=y_harmonic, sr=sr) C2 = librosa.feature.chroma_cqt(y=y_harmonic_2, sr=sr) pitch_sums_1 = get_pitch_sums(C)
