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#!/usr/bin/env python3 | |
# -*- encoding: utf-8 -*- | |
# Copyright FunASR (https://github.com/alibaba-damo-academy/FunASR). All Rights Reserved. | |
# MIT License (https://opensource.org/licenses/MIT) | |
import time | |
import copy | |
import torch | |
from torch.cuda.amp import autocast | |
from typing import Union, Dict, List, Tuple, Optional | |
from funasr_detach.register import tables | |
from funasr_detach.models.ctc.ctc import CTC | |
from funasr_detach.utils import postprocess_utils | |
from funasr_detach.utils.datadir_writer import DatadirWriter | |
from funasr_detach.models.paraformer.cif_predictor import mae_loss | |
from funasr_detach.train_utils.device_funcs import force_gatherable | |
from funasr_detach.models.transformer.utils.add_sos_eos import add_sos_eos | |
from funasr_detach.models.transformer.utils.nets_utils import make_pad_mask | |
from funasr_detach.utils.timestamp_tools import ts_prediction_lfr6_standard | |
from funasr_detach.utils.load_utils import load_audio_text_image_video, extract_fbank | |
class MonotonicAligner(torch.nn.Module): | |
""" | |
Author: Speech Lab of DAMO Academy, Alibaba Group | |
Achieving timestamp prediction while recognizing with non-autoregressive end-to-end ASR model | |
https://arxiv.org/abs/2301.12343 | |
""" | |
def __init__( | |
self, | |
input_size: int = 80, | |
specaug: Optional[str] = None, | |
specaug_conf: Optional[Dict] = None, | |
normalize: str = None, | |
normalize_conf: Optional[Dict] = None, | |
encoder: str = None, | |
encoder_conf: Optional[Dict] = None, | |
predictor: str = None, | |
predictor_conf: Optional[Dict] = None, | |
predictor_bias: int = 0, | |
length_normalized_loss: bool = False, | |
**kwargs, | |
): | |
super().__init__() | |
if specaug is not None: | |
specaug_class = tables.specaug_classes.get(specaug) | |
specaug = specaug_class(**specaug_conf) | |
if normalize is not None: | |
normalize_class = tables.normalize_classes.get(normalize) | |
normalize = normalize_class(**normalize_conf) | |
encoder_class = tables.encoder_classes.get(encoder) | |
encoder = encoder_class(input_size=input_size, **encoder_conf) | |
encoder_output_size = encoder.output_size() | |
predictor_class = tables.predictor_classes.get(predictor) | |
predictor = predictor_class(**predictor_conf) | |
self.specaug = specaug | |
self.normalize = normalize | |
self.encoder = encoder | |
self.predictor = predictor | |
self.criterion_pre = mae_loss(normalize_length=length_normalized_loss) | |
self.predictor_bias = predictor_bias | |
def forward( | |
self, | |
speech: torch.Tensor, | |
speech_lengths: torch.Tensor, | |
text: torch.Tensor, | |
text_lengths: torch.Tensor, | |
) -> Tuple[torch.Tensor, Dict[str, torch.Tensor], torch.Tensor]: | |
"""Frontend + Encoder + Decoder + Calc loss | |
Args: | |
speech: (Batch, Length, ...) | |
speech_lengths: (Batch, ) | |
text: (Batch, Length) | |
text_lengths: (Batch,) | |
""" | |
assert text_lengths.dim() == 1, text_lengths.shape | |
# Check that batch_size is unified | |
assert ( | |
speech.shape[0] | |
== speech_lengths.shape[0] | |
== text.shape[0] | |
== text_lengths.shape[0] | |
), (speech.shape, speech_lengths.shape, text.shape, text_lengths.shape) | |
batch_size = speech.shape[0] | |
# for data-parallel | |
text = text[:, : text_lengths.max()] | |
speech = speech[:, : speech_lengths.max()] | |
# 1. Encoder | |
encoder_out, encoder_out_lens = self.encode(speech, speech_lengths) | |
encoder_out_mask = ( | |
~make_pad_mask(encoder_out_lens, maxlen=encoder_out.size(1))[:, None, :] | |
).to(encoder_out.device) | |
if self.predictor_bias == 1: | |
_, text = add_sos_eos(text, 1, 2, -1) | |
text_lengths = text_lengths + self.predictor_bias | |
_, _, _, _, pre_token_length2 = self.predictor( | |
encoder_out, text, encoder_out_mask, ignore_id=-1 | |
) | |
# loss_pre = self.criterion_pre(ys_pad_lens.type_as(pre_token_length), pre_token_length) | |
loss_pre = self.criterion_pre( | |
text_lengths.type_as(pre_token_length2), pre_token_length2 | |
) | |
loss = loss_pre | |
stats = dict() | |
# Collect Attn branch stats | |
stats["loss_pre"] = loss_pre.detach().cpu() if loss_pre is not None else None | |
stats["loss"] = torch.clone(loss.detach()) | |
# force_gatherable: to-device and to-tensor if scalar for DataParallel | |
loss, stats, weight = force_gatherable((loss, stats, batch_size), loss.device) | |
return loss, stats, weight | |
def calc_predictor_timestamp(self, encoder_out, encoder_out_lens, token_num): | |
encoder_out_mask = ( | |
~make_pad_mask(encoder_out_lens, maxlen=encoder_out.size(1))[:, None, :] | |
).to(encoder_out.device) | |
ds_alphas, ds_cif_peak, us_alphas, us_peaks = ( | |
self.predictor.get_upsample_timestamp( | |
encoder_out, encoder_out_mask, token_num | |
) | |
) | |
return ds_alphas, ds_cif_peak, us_alphas, us_peaks | |
def encode( | |
self, | |
speech: torch.Tensor, | |
speech_lengths: torch.Tensor, | |
**kwargs, | |
) -> Tuple[torch.Tensor, torch.Tensor]: | |
"""Encoder. Note that this method is used by asr_inference.py | |
Args: | |
speech: (Batch, Length, ...) | |
speech_lengths: (Batch, ) | |
ind: int | |
""" | |
with autocast(False): | |
# Data augmentation | |
if self.specaug is not None and self.training: | |
speech, speech_lengths = self.specaug(speech, speech_lengths) | |
# Normalization for feature: e.g. Global-CMVN, Utterance-CMVN | |
if self.normalize is not None: | |
speech, speech_lengths = self.normalize(speech, speech_lengths) | |
# Forward encoder | |
encoder_out, encoder_out_lens, _ = self.encoder(speech, speech_lengths) | |
if isinstance(encoder_out, tuple): | |
encoder_out = encoder_out[0] | |
return encoder_out, encoder_out_lens | |
def inference( | |
self, | |
data_in, | |
data_lengths=None, | |
key: list = None, | |
tokenizer=None, | |
frontend=None, | |
**kwargs, | |
): | |
meta_data = {} | |
# extract fbank feats | |
time1 = time.perf_counter() | |
audio_list, text_token_int_list = load_audio_text_image_video( | |
data_in, | |
fs=frontend.fs, | |
audio_fs=kwargs.get("fs", 16000), | |
data_type=kwargs.get("data_type", "sound"), | |
tokenizer=tokenizer, | |
) | |
time2 = time.perf_counter() | |
meta_data["load_data"] = f"{time2 - time1:0.3f}" | |
speech, speech_lengths = extract_fbank( | |
audio_list, data_type=kwargs.get("data_type", "sound"), frontend=frontend | |
) | |
time3 = time.perf_counter() | |
meta_data["extract_feat"] = f"{time3 - time2:0.3f}" | |
meta_data["batch_data_time"] = ( | |
speech_lengths.sum().item() * frontend.frame_shift * frontend.lfr_n / 1000 | |
) | |
speech = speech.to(device=kwargs["device"]) | |
speech_lengths = speech_lengths.to(device=kwargs["device"]) | |
# Encoder | |
encoder_out, encoder_out_lens = self.encode(speech, speech_lengths) | |
# predictor | |
text_lengths = torch.tensor([len(i) + 1 for i in text_token_int_list]).to( | |
encoder_out.device | |
) | |
_, _, us_alphas, us_peaks = self.calc_predictor_timestamp( | |
encoder_out, encoder_out_lens, token_num=text_lengths | |
) | |
results = [] | |
ibest_writer = None | |
if kwargs.get("output_dir") is not None: | |
if not hasattr(self, "writer"): | |
self.writer = DatadirWriter(kwargs.get("output_dir")) | |
ibest_writer = self.writer["tp_res"] | |
for i, (us_alpha, us_peak, token_int) in enumerate( | |
zip(us_alphas, us_peaks, text_token_int_list) | |
): | |
token = tokenizer.ids2tokens(token_int) | |
timestamp_str, timestamp = ts_prediction_lfr6_standard( | |
us_alpha[: encoder_out_lens[i] * 3], | |
us_peak[: encoder_out_lens[i] * 3], | |
copy.copy(token), | |
) | |
text_postprocessed, time_stamp_postprocessed, _ = ( | |
postprocess_utils.sentence_postprocess(token, timestamp) | |
) | |
result_i = { | |
"key": key[i], | |
"text": text_postprocessed, | |
"timestamp": time_stamp_postprocessed, | |
} | |
results.append(result_i) | |
if ibest_writer: | |
# ibest_writer["token"][key[i]] = " ".join(token) | |
ibest_writer["timestamp_list"][key[i]] = time_stamp_postprocessed | |
ibest_writer["timestamp_str"][key[i]] = timestamp_str | |
return results, meta_data | |