pce / scripts /ctcalign.py
catiR
queries only sentences at least 10 speakers
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from transformers import Wav2Vec2ForCTC, Wav2Vec2Processor
import torch
import numpy as np
import soundfile as sf
from dataclasses import dataclass
# read wav audio, make mono and 16khz if necessary
def wav16m(sound_path):
aud, sr = sf.read(sound_path, dtype=np.float32)
if len(aud.shape) == 2:
aud = aud.mean(1)
if sr != 16000:
alen = int(aud.shape[0] / sr * 16000)
aud = signal.resample(aud, alen)
return aud
def aligner(model_path,model_word_separator = '|', model_blank_token = '[PAD]'):
# build labels dict from a processor where it is not directly accessible
def get_processor_labels(processor,word_sep,max_labels=100):
ixs = sorted(list(range(max_labels)),reverse=True)
return {processor.tokenizer.decode(n) or word_sep:n for n in ixs}
#------------------------------------------
# setup wav2vec2
#------------------------------------------
device = torch.device("cuda" if torch.cuda.is_available() else "cpu")
torch.random.manual_seed(0)
max_labels = 100 # any reasonable number higher than vocab + extra + special tokens in any language used
model = Wav2Vec2ForCTC.from_pretrained(model_path).to(device)
processor = Wav2Vec2Processor.from_pretrained(model_path)
labels_dict = get_processor_labels(processor,model_word_separator)
blank_id = labels_dict[model_blank_token]
#convert frame-numbers to timestamps in seconds
# w2v2 step size is about 20ms, or 50 frames per second
def f2s(fr):
return fr/50
#------------------------------------------
# forced alignment with ctc decoder
# based on implementation of
# https://pytorch.org/audio/main/tutorials/forced_alignment_tutorial.html
#------------------------------------------
# return the label class probability of each audio frame
# wav is the wav data already read in, NOT the file path.
def get_frame_probs(wav):
with torch.inference_mode(): # similar to with torch.no_grad():
input_values = processor(wav,sampling_rate=16000).input_values[0]
input_values = torch.tensor(input_values, device=device).unsqueeze(0)
emits = model(input_values).logits
emits = torch.log_softmax(emits, dim=-1)
return emits[0].cpu().detach()
def get_trellis(emission, tokens, blank_id):
num_frame = emission.size(0)
num_tokens = len(tokens)
trellis = torch.empty((num_frame + 1, num_tokens + 1))
trellis[0, 0] = 0
trellis[1:, 0] = torch.cumsum(emission[:, 0], 0) # len of this slice of trellis is len of audio frames)
trellis[0, -num_tokens:] = -float("inf") # len of this slice of trellis is len of transcript tokens
trellis[-num_tokens:, 0] = float("inf")
for t in range(num_frame):
trellis[t + 1, 1:] = torch.maximum(
# Score for staying at the same token
trellis[t, 1:] + emission[t, blank_id],
# Score for changing to the next token
trellis[t, :-1] + emission[t, tokens],
)
return trellis
@dataclass
class Point:
token_index: int
time_index: int
score: float
@dataclass
class Segment:
label: str
start: int
end: int
score: float
@property
def mfaform(self):
return f"{f2s(self.start)},{f2s(self.end)},{self.label}"
@property
def length(self):
return self.end - self.start
def backtrack(trellis, emission, tokens, blank_id):
# Note:
# j and t are indices for trellis, which has extra dimensions
# for time and tokens at the beginning.
# When referring to time frame index `T` in trellis,
# the corresponding index in emission is `T-1`.
# Similarly, when referring to token index `J` in trellis,
# the corresponding index in transcript is `J-1`.
j = trellis.size(1) - 1
t_start = torch.argmax(trellis[:, j]).item()
path = []
for t in range(t_start, 0, -1):
# 1. Figure out if the current position was stay or change
# `emission[J-1]` is the emission at time frame `J` of trellis dimension.
# Score for token staying the same from time frame J-1 to T.
stayed = trellis[t - 1, j] + emission[t - 1, blank_id]
# Score for token changing from C-1 at T-1 to J at T.
changed = trellis[t - 1, j - 1] + emission[t - 1, tokens[j - 1]]
# 2. Store the path with frame-wise probability.
prob = emission[t - 1, tokens[j - 1] if changed > stayed else 0].exp().item()
# Return token index and time index in non-trellis coordinate.
path.append(Point(j - 1, t - 1, prob))
# 3. Update the token
if changed > stayed:
j -= 1
if j == 0:
break
else:
raise ValueError("Failed to align")
return path[::-1]
def merge_repeats(path,transcript):
i1, i2 = 0, 0
segments = []
while i1 < len(path):
while i2 < len(path) and path[i1].token_index == path[i2].token_index: # while both path steps point to the same token index
i2 += 1
score = sum(path[k].score for k in range(i1, i2)) / (i2 - i1)
segments.append( # when i2 finally switches to a different token,
Segment(
transcript[path[i1].token_index],# to the list of segments, append the token from i1
path[i1].time_index, # time of the first path-point of that token
path[i2 - 1].time_index + 1, # time of the final path-point for that token.
score,
)
)
i1 = i2
return segments
def merge_words(segments, separator):
words = []
i1, i2 = 0, 0
while i1 < len(segments):
if i2 >= len(segments) or segments[i2].label == separator:
if i1 != i2:
segs = segments[i1:i2]
word = "".join([seg.label for seg in segs])
score = sum(seg.score * seg.length for seg in segs) / sum(seg.length for seg in segs)
words.append(Segment(word, segments[i1].start, segments[i2 - 1].end, score))
i1 = i2 + 1
i2 = i1
else:
i2 += 1
return words
#------------------------------------------
# handle, i/o, etc.
#------------------------------------------
# generate mfa format for character (phone) and word alignments
# skip the word separator as it is not a phone
def mfalike(chars,wds,wsep):
hed = ['Begin,End,Label,Type,Speaker\n']
wlines = [f'{w.mfaform},words,000\n' for w in wds]
slines = [f'{ch.mfaform},phones,000\n' for ch in chars if ch.label != wsep]
return (''.join(hed+wlines+slines))
# generate basic exportable list format for character OR word alignments
# skip the word separator as it is not a phone
def basic(segs,wsep="|"):
return [[s.label,f2s(s.start),f2s(s.end)] for s in segs if s.label != wsep]
# generate numbered dicts to use in dtw
# alignment is given in numbered frames, not converted to timestamps
def fordtw(words,segments):
# index i, and word/seg, startframe, endframe
# preppend the index i to the word or seg
def _ix(i,elem):
return [f'{i:03d}__{elem.label}', elem.start, elem.end]
w_al = [_ix(i,wse) for i,wse in enumerate(words)] # from tuple to list
wsegdict = {}
for w,s,e in w_al:
nlett = len(w.split('__')[1])
wsegs = segments[:nlett]
wstart = s
wsegs = [_ix(i,cse) for i,cse in enumerate(wsegs)]
wsegs = [[seg, ss-s, se-s] for seg,ss,se in wsegs]
wsegdict[w] = wsegs
segments = segments[nlett:]
return w_al, wsegdict
# basic cleaning
# skip with is_normed=True
# if transcript was already normalised externally
def normalise_transcript(xcp):
xcp = xcp.lower()
xcp = xcp.replace('-','')
while ' ' in xcp:
xcp = xcp.replace(' ', ' ')
return xcp
# needs pad labels added to correctly time first segment
# and therefore add word sep character as placeholder in transcript
def prep_transcript(xcp,is_normed):
if not is_normed:
xcp = normalise_transcript(xcp)
xcp = xcp.replace(' ',model_word_separator)
label_ids = [labels_dict[c] for c in xcp]
label_ids = [blank_id] + label_ids + [blank_id]
xcp = f'{model_word_separator}{xcp}{model_word_separator}'
return xcp,label_ids
def _align(wav_data,transcript,is_normed=False):
norm_transcript,rec_label_ids = prep_transcript(transcript,is_normed)
emit = get_frame_probs(wav_data)
trellis = get_trellis(emit, rec_label_ids, blank_id)
path = backtrack(trellis, emit, rec_label_ids, blank_id)
segments = merge_repeats(path,norm_transcript)
words = merge_words(segments, model_word_separator)
#return fordtw(words,model_word_separator), basic(segments,model_word_separator)
return basic(words,model_word_separator)
return _align
# usage:
# from ctcalign import aligner, wav16m
# model_path ="/home/caitlinr/work/models/LVL/wav2vec2-large-xlsr-53-icelandic-ep10-1000h"
# model_word_sep = '|'
# model_blank_tk = '[PAD]'
# caligner = aligner(model_path,model_word_sep,model_blank_tk)
# word_aln, seg_aln = caligner(wav16m(wav_path),transcript_string)