NeMo / nemo /collections /speechlm2 /data /force_align.py
dlxj
update nemo==2.8.0.rc0
f5d2dd3
# Copyright (c) 2025, NVIDIA CORPORATION. All rights reserved.
#
# Licensed under the Apache License, Version 2.0 (the "License");
# you may not use this file except in compliance with the License.
# You may obtain a copy of the License at
#
# http://www.apache.org/licenses/LICENSE-2.0
#
# Unless required by applicable law or agreed to in writing, software
# distributed under the License is distributed on an "AS IS" BASIS,
# WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
# See the License for the specific language governing permissions and
# limitations under the License.
import logging
import multiprocessing as mp
import re
from typing import Any, Dict, List, Optional
import numpy as np
import torch
from lhotse import CutSet
# Use NeMo's force alignment utilities instead of torchaudio
from nemo.collections.asr.models.asr_model import ASRModel
from nemo.collections.asr.parts.utils.aligner_utils import (
add_t_start_end_to_utt_obj,
get_batch_variables,
viterbi_decoding,
)
class ForceAligner:
"""
Force alignment utility using NeMo CTC-based ASR models for speech-to-text alignment.
"""
def __init__(
self,
asr_model: Optional[ASRModel] = None,
device: str = None,
frame_length: float = 0.02,
asr_model_name: str = "stt_en_fastconformer_ctc_large",
):
"""
Initialize the ForceAligner.
Args:
asr_model: NeMo ASR model instance for alignment. If None, will load from asr_model_name
device: Device to run alignment on (default: auto-detect)
frame_length: Frame length in seconds for timestamp conversion
asr_model_name: Name of the NeMo ASR model to load if asr_model is None
"""
self.device = device or ("cuda" if torch.cuda.is_available() else "cpu")
self.frame_length = frame_length
self.asr_model_name = asr_model_name
self.asr_model = asr_model
self.output_timestep_duration = None
self._model_loaded = False
def _load_asr_model(self):
"""Load the NeMo ASR model."""
try:
if self.device == 'cuda' and mp.get_start_method(allow_none=True) == 'fork':
logging.warning(
"Detected 'fork' multiprocessing start method with CUDA device. "
"To avoid CUDA re-initialization errors in worker processes, "
"falling back to CPU for force alignment. "
"To use CUDA, set mp.set_start_method('spawn', force=True) in your main training script "
"before creating the DataLoader."
)
self.device = 'cpu'
device = torch.device(self.device)
logging.info(f"Loading NeMo ASR model '{self.asr_model_name}' for force alignment on device {device}")
if self.asr_model is None:
# Load ASR model from pretrained
self.asr_model = ASRModel.from_pretrained(self.asr_model_name, map_location=device)
else:
self.asr_model = self.asr_model.to(device)
self.asr_model.eval()
# Calculate output timestep duration
try:
self.output_timestep_duration = (
self.asr_model.cfg['preprocessor']['window_stride'] * self.asr_model.encoder.subsampling_factor
)
except Exception as e:
# Default fallback based on typical FastConformer settings
self.output_timestep_duration = 0.04
logging.warning(
f"Could not calculate output_timestep_duration from model config: {e}. "
f"Using default {self.output_timestep_duration}s"
)
logging.info(
f"NeMo ASR model loaded successfully for force alignment. "
f"Output timestep duration: {self.output_timestep_duration}s"
)
except Exception as e:
logging.error(f"Failed to load NeMo ASR model for force alignment: {e}")
self.asr_model = None
raise
def batch_force_align_user_audio(self, cuts: CutSet, source_sample_rate: int = 16000) -> CutSet:
"""
Perform batched force alignment on all user audio segments.
Collects all user segments, writes temp files, runs a single batched
get_batch_variables + viterbi_decoding call, then maps results back.
Args:
cuts: CutSet containing all cuts to process
source_sample_rate: Source sample rate of the audio
Returns:
CutSet with updated supervision texts (timestamped where alignment succeeded)
"""
if not self._model_loaded:
self._load_asr_model()
self._model_loaded = True
if self.asr_model is None:
logging.warning("ASR model not available for force alignment, returning empty cutset")
return CutSet.from_cuts([])
# Collect all user supervisions
user_supervisions = []
user_cuts = []
for cut in cuts:
for supervision in cut.supervisions:
if supervision.speaker.lower() == "user":
user_supervisions.append(supervision)
user_cuts.append(cut)
if not user_supervisions:
logging.info("No user supervisions found for force alignment")
return cuts
logging.info(f"Performing batched force alignment on {len(user_supervisions)} user audio segments")
# Prepare all audio arrays and texts for batched processing
audio_arrays = []
normalized_texts = []
valid_indices = [] # track which supervisions have valid audio/text
target_sample_rate = 16000
for i, (supervision, cut) in enumerate(zip(user_supervisions, user_cuts)):
try:
text = self._strip_timestamps(supervision.text)
normalized_text = self._normalize_transcript(text)
if not normalized_text.strip():
logging.warning(f"Text became empty after normalization: {supervision.text}")
continue
user_cut = cut.truncate(offset=supervision.start, duration=supervision.duration)
audio = user_cut.load_audio()
if audio.ndim > 1:
audio = audio.mean(axis=0)
if source_sample_rate != target_sample_rate:
from scipy import signal
num_samples = int(len(audio) * target_sample_rate / source_sample_rate)
audio = signal.resample(audio, num_samples)
# Add silence padding for better alignment at the end
silence_samples = int(0.64 * target_sample_rate)
audio = np.concatenate([audio, np.zeros(silence_samples)])
audio_arrays.append(audio)
normalized_texts.append(normalized_text)
valid_indices.append(i)
except Exception as e:
logging.error(f"Failed to prepare segment {i} for alignment: {e}")
if not audio_arrays:
logging.warning("No valid segments to align")
return cuts
# Batched ASR inference + Viterbi decoding
success_count = 0
failed_count = 0
try:
(
log_probs_batch,
y_batch,
T_batch,
U_batch,
utt_obj_batch,
output_timestep_duration,
) = get_batch_variables(
audio=audio_arrays,
model=self.asr_model,
gt_text_batch=normalized_texts,
align_using_pred_text=False,
output_timestep_duration=self.output_timestep_duration,
)
alignments_batch = viterbi_decoding(
log_probs_batch=log_probs_batch,
y_batch=y_batch,
T_batch=T_batch,
U_batch=U_batch,
viterbi_device=torch.device(self.device),
)
# Map results back to supervisions
for batch_idx, orig_idx in enumerate(valid_indices):
try:
if batch_idx >= len(alignments_batch) or batch_idx >= len(utt_obj_batch):
failed_count += 1
continue
utt_obj = utt_obj_batch[batch_idx]
if not utt_obj.token_ids_with_blanks:
failed_count += 1
continue
alignment = alignments_batch[batch_idx]
utt_obj = add_t_start_end_to_utt_obj(utt_obj, alignment, output_timestep_duration)
word_segments = self._extract_word_timestamps(utt_obj)
if word_segments:
timestamped_text = self._convert_alignment_to_timestamped_text(
word_segments, user_supervisions[orig_idx].text
)
user_supervisions[orig_idx].text = timestamped_text
success_count += 1
else:
failed_count += 1
except Exception as e:
logging.error(f"Failed to process alignment for segment {orig_idx}: {e}")
failed_count += 1
except Exception as e:
logging.error(f"Batched force alignment failed: {e}")
failed_count = len(valid_indices)
finally:
if self.device == 'cuda' and torch.cuda.is_available():
torch.cuda.empty_cache()
if failed_count > 0:
logging.warning(
f"Force alignment failed for {failed_count}/{len(user_supervisions)} user segments. "
f"Keeping original text for failed alignments."
)
else:
logging.info(f"Force alignment succeeded for all {success_count} user segments.")
return cuts
def _extract_word_timestamps(self, utt_obj) -> List[Dict[str, Any]]:
"""
Extract word-level timestamps from the utterance object returned by NeMo aligner.
Args:
utt_obj: Utterance object with timing information
Returns:
List of word segments with timing information
"""
word_segments = []
for segment_or_token in utt_obj.segments_and_tokens:
# Check if this is a Segment object (has words_and_tokens attribute)
if hasattr(segment_or_token, 'words_and_tokens'):
segment = segment_or_token
for word_or_token in segment.words_and_tokens:
# Check if this is a Word object (has 'text' and timing attributes)
if hasattr(word_or_token, 'text') and hasattr(word_or_token, 't_start'):
word = word_or_token
# Skip CTC blank tokens and include only words with valid timing
if (
word.text not in ('<b>', '')
and word.t_start is not None
and word.t_end is not None
and word.t_start >= 0
and word.t_end >= 0
):
word_segments.append(
{
'word': word.text,
'start': word.t_start,
'end': word.t_end,
'score': 1.0, # NeMo CTC alignment doesn't provide confidence scores
}
)
return word_segments
def _normalize_transcript(self, transcript: str) -> str:
"""
Normalize transcript for the ASR model's tokenizer.
Keeps it simple to match common ASR preprocessing.
"""
text = transcript.lower()
# Remove special characters except apostrophes and spaces
text = re.sub(r"[^a-z' ]", " ", text)
# Collapse multiple spaces
text = re.sub(r' +', ' ', text)
return text.strip()
def _convert_alignment_to_timestamped_text(
self, alignment_result: List[Dict[str, Any]], original_text: str
) -> str:
"""
Convert alignment results to timestamped text format.
Args:
alignment_result: List of word segments with timing information
original_text: Original text without timestamps
Returns:
Text with timestamp tokens in the format <|start_frame|>word<|end_frame|>
"""
timestamped_words = []
for word_seg in alignment_result:
word = word_seg["word"]
start_frame = int(word_seg["start"] / self.frame_length)
end_frame = int(word_seg["end"] / self.frame_length)
timestamped_words.append(f"<|{start_frame}|> {word} <|{end_frame}|>")
return " ".join(timestamped_words)
def _strip_timestamps(self, text: str) -> str:
"""
Strip timestamp tokens from text.
Args:
text: Text that may contain timestamp tokens
Returns:
Text with timestamp tokens removed
"""
text = re.sub(r'<\|[0-9]+\|>', '', text)
text = re.sub(r' +', ' ', text)
return text.strip()