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generate.py
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generate.py
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import torch
import torchaudio
from datetime import timedelta
from dataclasses import dataclass
from srt import Subtitle, compose
import whisper
from pydub import AudioSegment
import re
import num2words
device = torch.device("cuda" if torch.cuda.is_available() else "cpu")
torch.cuda.empty_cache()
torch.random.manual_seed(0)
def force_align(SPEECH_FILE, transcript, start_index, start_time):
bundle = torchaudio.pipelines.WAV2VEC2_ASR_BASE_960H
model = bundle.get_model().to(device)
labels = bundle.get_labels()
with torch.inference_mode():
waveform, _ = torchaudio.load(SPEECH_FILE)
emissions, _ = model(waveform.to(device))
emissions = torch.log_softmax(emissions, dim=-1)
emission = emissions[0].cpu().detach()
dictionary = {c: i for i, c in enumerate(labels)}
tokens = [dictionary[c] for c in transcript]
def get_trellis(emission, tokens, blank_id=0):
num_frame = emission.size(0)
num_tokens = len(tokens)
# Trellis has extra diemsions for both time axis and tokens.
# The extra dim for tokens represents <SoS> (start-of-sentence)
# The extra dim for time axis is for simplification of the code.
trellis = torch.empty((num_frame + 1, num_tokens + 1))
trellis[0, 0] = 0
trellis[1:, 0] = torch.cumsum(emission[:, 0], 0)
trellis[0, -num_tokens:] = -float("inf")
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
trellis = get_trellis(emission, tokens)
@dataclass
class Point:
token_index: int
time_index: int
score: float
def backtrack(trellis, emission, tokens, blank_id=0):
# 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
# Note (again):
# `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]
path = backtrack(trellis, emission, tokens)
# Merge the labels
@dataclass
class Segment:
label: str
start: int
end: int
score: float
def __repr__(self):
return f"{self.label}\t({self.score:4.2f}): [{self.start:5d}, {self.end:5d})"
@property
def length(self):
return self.end - self.start
def merge_repeats(path):
i1, i2 = 0, 0
segments = []
while i1 < len(path):
while i2 < len(path) and path[i1].token_index == path[i2].token_index:
i2 += 1
score = sum(path[k].score for k in range(i1, i2)) / (i2 - i1)
segments.append(
Segment(
transcript[path[i1].token_index],
path[i1].time_index,
path[i2 - 1].time_index + 1,
score,
)
)
i1 = i2
return segments
segments = merge_repeats(path)
# Merge words
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
word_segments = merge_words(segments)
subs = []
for i,word in enumerate(word_segments):
ratio = waveform.size(1) / (trellis.size(0) - 1)
x0 = int(ratio * word.start)
x1 = int(ratio * word.end)
start = timedelta(seconds=start_time + x0 / bundle.sample_rate)
end = timedelta(seconds=start_time + x1 / bundle.sample_rate )
subtitle = Subtitle(start_index+i, start, end, word.label)
subs.append(subtitle)
print(compose(subs))
return subs
model = whisper.load_model("medium")
audio = whisper.load_audio("video.wav")
transcription = model.transcribe(audio)
# # print the recognized text
segments = transcription["segments"]
print("Transcription complete:")
print(transcription["text"])
print("Starting to force alignment...")
start_index = 0
total_subs = []
for i,segment in enumerate(segments):
text = segment["text"]
audioSegment = AudioSegment.from_wav("video.wav")[segment["start"]*1000:segment["end"]*1000]
audioSegment.export(str(i)+'.wav', format="wav") #Exports to a wav file in the current path.
transcript=text.strip().replace(" ", "|")
transcript = re.sub(r'[^\w|\s]', '', transcript)
transcript = re.sub(r"(\d+)", lambda x: num2words.num2words(int(x.group(0))), transcript)
print(segment["start"])
subs = force_align(str(i)+'.wav', transcript.upper(), start_index, segment["start"])
start_index += len(segment["text"])
total_subs.extend(subs)
CAPTION_FILE = open("caption.srt", "w")
CAPTION_FILE.write(compose(total_subs))
CAPTION_FILE.close()