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Super-squash branch 'main' using huggingface_hub
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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 torch
import logging
from typing import Dict, Tuple
from contextlib import contextmanager
from distutils.version import LooseVersion
from funasr_detach.register import tables
from funasr_detach.models.ctc.ctc import CTC
from funasr_detach.utils import postprocess_utils
from funasr_detach.metrics.compute_acc import th_accuracy
from funasr_detach.utils.datadir_writer import DatadirWriter
from funasr_detach.models.paraformer.model import Paraformer
from funasr_detach.models.paraformer.search import Hypothesis
from funasr_detach.models.paraformer.cif_predictor import mae_loss
from funasr_detach.train_utils.device_funcs import force_gatherable
from funasr_detach.losses.label_smoothing_loss import LabelSmoothingLoss
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, pad_list
from funasr_detach.utils.load_utils import load_audio_text_image_video, extract_fbank
if LooseVersion(torch.__version__) >= LooseVersion("1.6.0"):
from torch.cuda.amp import autocast
else:
# Nothing to do if torch<1.6.0
@contextmanager
def autocast(enabled=True):
yield
@tables.register("model_classes", "ParaformerStreaming")
class ParaformerStreaming(Paraformer):
"""
Author: Speech Lab of DAMO Academy, Alibaba Group
Paraformer: Fast and Accurate Parallel Transformer for Non-autoregressive End-to-End Speech Recognition
https://arxiv.org/abs/2206.08317
"""
def __init__(
self,
*args,
**kwargs,
):
super().__init__(*args, **kwargs)
# import pdb;
# pdb.set_trace()
self.sampling_ratio = kwargs.get("sampling_ratio", 0.2)
self.scama_mask = None
if (
hasattr(self.encoder, "overlap_chunk_cls")
and self.encoder.overlap_chunk_cls is not None
):
from funasr_detach.models.scama.chunk_utilis import (
build_scama_mask_for_cross_attention_decoder,
)
self.build_scama_mask_for_cross_attention_decoder_fn = (
build_scama_mask_for_cross_attention_decoder
)
self.decoder_attention_chunk_type = kwargs.get(
"decoder_attention_chunk_type", "chunk"
)
def forward(
self,
speech: torch.Tensor,
speech_lengths: torch.Tensor,
text: torch.Tensor,
text_lengths: torch.Tensor,
**kwargs,
) -> Tuple[torch.Tensor, Dict[str, torch.Tensor], torch.Tensor]:
"""Encoder + Decoder + Calc loss
Args:
speech: (Batch, Length, ...)
speech_lengths: (Batch, )
text: (Batch, Length)
text_lengths: (Batch,)
"""
# import pdb;
# pdb.set_trace()
decoding_ind = kwargs.get("decoding_ind")
if len(text_lengths.size()) > 1:
text_lengths = text_lengths[:, 0]
if len(speech_lengths.size()) > 1:
speech_lengths = speech_lengths[:, 0]
batch_size = speech.shape[0]
# Encoder
if hasattr(self.encoder, "overlap_chunk_cls"):
ind = self.encoder.overlap_chunk_cls.random_choice(
self.training, decoding_ind
)
encoder_out, encoder_out_lens = self.encode(speech, speech_lengths, ind=ind)
else:
encoder_out, encoder_out_lens = self.encode(speech, speech_lengths)
loss_ctc, cer_ctc = None, None
loss_pre = None
stats = dict()
# decoder: CTC branch
if self.ctc_weight > 0.0:
if hasattr(self.encoder, "overlap_chunk_cls"):
encoder_out_ctc, encoder_out_lens_ctc = (
self.encoder.overlap_chunk_cls.remove_chunk(
encoder_out, encoder_out_lens, chunk_outs=None
)
)
else:
encoder_out_ctc, encoder_out_lens_ctc = encoder_out, encoder_out_lens
loss_ctc, cer_ctc = self._calc_ctc_loss(
encoder_out_ctc, encoder_out_lens_ctc, text, text_lengths
)
# Collect CTC branch stats
stats["loss_ctc"] = loss_ctc.detach() if loss_ctc is not None else None
stats["cer_ctc"] = cer_ctc
# decoder: Attention decoder branch
loss_att, acc_att, cer_att, wer_att, loss_pre, pre_loss_att = (
self._calc_att_predictor_loss(
encoder_out, encoder_out_lens, text, text_lengths
)
)
# 3. CTC-Att loss definition
if self.ctc_weight == 0.0:
loss = loss_att + loss_pre * self.predictor_weight
else:
loss = (
self.ctc_weight * loss_ctc
+ (1 - self.ctc_weight) * loss_att
+ loss_pre * self.predictor_weight
)
# Collect Attn branch stats
stats["loss_att"] = loss_att.detach() if loss_att is not None else None
stats["pre_loss_att"] = (
pre_loss_att.detach() if pre_loss_att is not None else None
)
stats["acc"] = acc_att
stats["cer"] = cer_att
stats["wer"] = wer_att
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
if self.length_normalized_loss:
batch_size = (text_lengths + self.predictor_bias).sum()
loss, stats, weight = force_gatherable((loss, stats, batch_size), loss.device)
return loss, stats, weight
def encode_chunk(
self,
speech: torch.Tensor,
speech_lengths: torch.Tensor,
cache: dict = None,
**kwargs,
) -> Tuple[torch.Tensor, torch.Tensor]:
"""Frontend + 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.forward_chunk(
speech, speech_lengths, cache=cache["encoder"]
)
if isinstance(encoder_out, tuple):
encoder_out = encoder_out[0]
return encoder_out, torch.tensor([encoder_out.size(1)])
def _calc_att_predictor_loss(
self,
encoder_out: torch.Tensor,
encoder_out_lens: torch.Tensor,
ys_pad: torch.Tensor,
ys_pad_lens: torch.Tensor,
):
encoder_out_mask = (
~make_pad_mask(encoder_out_lens, maxlen=encoder_out.size(1))[:, None, :]
).to(encoder_out.device)
if self.predictor_bias == 1:
_, ys_pad = add_sos_eos(ys_pad, self.sos, self.eos, self.ignore_id)
ys_pad_lens = ys_pad_lens + self.predictor_bias
mask_chunk_predictor = None
if self.encoder.overlap_chunk_cls is not None:
mask_chunk_predictor = (
self.encoder.overlap_chunk_cls.get_mask_chunk_predictor(
None, device=encoder_out.device, batch_size=encoder_out.size(0)
)
)
mask_shfit_chunk = self.encoder.overlap_chunk_cls.get_mask_shfit_chunk(
None, device=encoder_out.device, batch_size=encoder_out.size(0)
)
encoder_out = encoder_out * mask_shfit_chunk
pre_acoustic_embeds, pre_token_length, pre_alphas, _ = self.predictor(
encoder_out,
ys_pad,
encoder_out_mask,
ignore_id=self.ignore_id,
mask_chunk_predictor=mask_chunk_predictor,
target_label_length=ys_pad_lens,
)
predictor_alignments, predictor_alignments_len = (
self.predictor.gen_frame_alignments(pre_alphas, encoder_out_lens)
)
scama_mask = None
if (
self.encoder.overlap_chunk_cls is not None
and self.decoder_attention_chunk_type == "chunk"
):
encoder_chunk_size = self.encoder.overlap_chunk_cls.chunk_size_pad_shift_cur
attention_chunk_center_bias = 0
attention_chunk_size = encoder_chunk_size
decoder_att_look_back_factor = (
self.encoder.overlap_chunk_cls.decoder_att_look_back_factor_cur
)
mask_shift_att_chunk_decoder = (
self.encoder.overlap_chunk_cls.get_mask_shift_att_chunk_decoder(
None, device=encoder_out.device, batch_size=encoder_out.size(0)
)
)
scama_mask = self.build_scama_mask_for_cross_attention_decoder_fn(
predictor_alignments=predictor_alignments,
encoder_sequence_length=encoder_out_lens,
chunk_size=1,
encoder_chunk_size=encoder_chunk_size,
attention_chunk_center_bias=attention_chunk_center_bias,
attention_chunk_size=attention_chunk_size,
attention_chunk_type=self.decoder_attention_chunk_type,
step=None,
predictor_mask_chunk_hopping=mask_chunk_predictor,
decoder_att_look_back_factor=decoder_att_look_back_factor,
mask_shift_att_chunk_decoder=mask_shift_att_chunk_decoder,
target_length=ys_pad_lens,
is_training=self.training,
)
elif self.encoder.overlap_chunk_cls is not None:
encoder_out, encoder_out_lens = self.encoder.overlap_chunk_cls.remove_chunk(
encoder_out, encoder_out_lens, chunk_outs=None
)
# 0. sampler
decoder_out_1st = None
pre_loss_att = None
if self.sampling_ratio > 0.0:
if self.step_cur < 2:
logging.info(
"enable sampler in paraformer, sampling_ratio: {}".format(
self.sampling_ratio
)
)
if self.use_1st_decoder_loss:
sematic_embeds, decoder_out_1st, pre_loss_att = self.sampler_with_grad(
encoder_out,
encoder_out_lens,
ys_pad,
ys_pad_lens,
pre_acoustic_embeds,
scama_mask,
)
else:
sematic_embeds, decoder_out_1st = self.sampler(
encoder_out,
encoder_out_lens,
ys_pad,
ys_pad_lens,
pre_acoustic_embeds,
scama_mask,
)
else:
if self.step_cur < 2:
logging.info(
"disable sampler in paraformer, sampling_ratio: {}".format(
self.sampling_ratio
)
)
sematic_embeds = pre_acoustic_embeds
# 1. Forward decoder
decoder_outs = self.decoder(
encoder_out, encoder_out_lens, sematic_embeds, ys_pad_lens, scama_mask
)
decoder_out, _ = decoder_outs[0], decoder_outs[1]
if decoder_out_1st is None:
decoder_out_1st = decoder_out
# 2. Compute attention loss
loss_att = self.criterion_att(decoder_out, ys_pad)
acc_att = th_accuracy(
decoder_out_1st.view(-1, self.vocab_size),
ys_pad,
ignore_label=self.ignore_id,
)
loss_pre = self.criterion_pre(
ys_pad_lens.type_as(pre_token_length), pre_token_length
)
# Compute cer/wer using attention-decoder
if self.training or self.error_calculator is None:
cer_att, wer_att = None, None
else:
ys_hat = decoder_out_1st.argmax(dim=-1)
cer_att, wer_att = self.error_calculator(ys_hat.cpu(), ys_pad.cpu())
return loss_att, acc_att, cer_att, wer_att, loss_pre, pre_loss_att
def sampler(
self,
encoder_out,
encoder_out_lens,
ys_pad,
ys_pad_lens,
pre_acoustic_embeds,
chunk_mask=None,
):
tgt_mask = (
~make_pad_mask(ys_pad_lens, maxlen=ys_pad_lens.max())[:, :, None]
).to(ys_pad.device)
ys_pad_masked = ys_pad * tgt_mask[:, :, 0]
if self.share_embedding:
ys_pad_embed = self.decoder.output_layer.weight[ys_pad_masked]
else:
ys_pad_embed = self.decoder.embed(ys_pad_masked)
with torch.no_grad():
decoder_outs = self.decoder(
encoder_out,
encoder_out_lens,
pre_acoustic_embeds,
ys_pad_lens,
chunk_mask,
)
decoder_out, _ = decoder_outs[0], decoder_outs[1]
pred_tokens = decoder_out.argmax(-1)
nonpad_positions = ys_pad.ne(self.ignore_id)
seq_lens = (nonpad_positions).sum(1)
same_num = ((pred_tokens == ys_pad) & nonpad_positions).sum(1)
input_mask = torch.ones_like(nonpad_positions)
bsz, seq_len = ys_pad.size()
for li in range(bsz):
target_num = (
((seq_lens[li] - same_num[li].sum()).float()) * self.sampling_ratio
).long()
if target_num > 0:
input_mask[li].scatter_(
dim=0,
index=torch.randperm(seq_lens[li])[:target_num].cuda(),
value=0,
)
input_mask = input_mask.eq(1)
input_mask = input_mask.masked_fill(~nonpad_positions, False)
input_mask_expand_dim = input_mask.unsqueeze(2).to(
pre_acoustic_embeds.device
)
sematic_embeds = pre_acoustic_embeds.masked_fill(
~input_mask_expand_dim, 0
) + ys_pad_embed.masked_fill(input_mask_expand_dim, 0)
return sematic_embeds * tgt_mask, decoder_out * tgt_mask
def calc_predictor(self, encoder_out, encoder_out_lens):
encoder_out_mask = (
~make_pad_mask(encoder_out_lens, maxlen=encoder_out.size(1))[:, None, :]
).to(encoder_out.device)
mask_chunk_predictor = None
if self.encoder.overlap_chunk_cls is not None:
mask_chunk_predictor = (
self.encoder.overlap_chunk_cls.get_mask_chunk_predictor(
None, device=encoder_out.device, batch_size=encoder_out.size(0)
)
)
mask_shfit_chunk = self.encoder.overlap_chunk_cls.get_mask_shfit_chunk(
None, device=encoder_out.device, batch_size=encoder_out.size(0)
)
encoder_out = encoder_out * mask_shfit_chunk
pre_acoustic_embeds, pre_token_length, pre_alphas, pre_peak_index = (
self.predictor(
encoder_out,
None,
encoder_out_mask,
ignore_id=self.ignore_id,
mask_chunk_predictor=mask_chunk_predictor,
target_label_length=None,
)
)
predictor_alignments, predictor_alignments_len = (
self.predictor.gen_frame_alignments(
pre_alphas,
(
encoder_out_lens + 1
if self.predictor.tail_threshold > 0.0
else encoder_out_lens
),
)
)
scama_mask = None
if (
self.encoder.overlap_chunk_cls is not None
and self.decoder_attention_chunk_type == "chunk"
):
encoder_chunk_size = self.encoder.overlap_chunk_cls.chunk_size_pad_shift_cur
attention_chunk_center_bias = 0
attention_chunk_size = encoder_chunk_size
decoder_att_look_back_factor = (
self.encoder.overlap_chunk_cls.decoder_att_look_back_factor_cur
)
mask_shift_att_chunk_decoder = (
self.encoder.overlap_chunk_cls.get_mask_shift_att_chunk_decoder(
None, device=encoder_out.device, batch_size=encoder_out.size(0)
)
)
scama_mask = self.build_scama_mask_for_cross_attention_decoder_fn(
predictor_alignments=predictor_alignments,
encoder_sequence_length=encoder_out_lens,
chunk_size=1,
encoder_chunk_size=encoder_chunk_size,
attention_chunk_center_bias=attention_chunk_center_bias,
attention_chunk_size=attention_chunk_size,
attention_chunk_type=self.decoder_attention_chunk_type,
step=None,
predictor_mask_chunk_hopping=mask_chunk_predictor,
decoder_att_look_back_factor=decoder_att_look_back_factor,
mask_shift_att_chunk_decoder=mask_shift_att_chunk_decoder,
target_length=None,
is_training=self.training,
)
self.scama_mask = scama_mask
return pre_acoustic_embeds, pre_token_length, pre_alphas, pre_peak_index
def calc_predictor_chunk(self, encoder_out, encoder_out_lens, cache=None, **kwargs):
is_final = kwargs.get("is_final", False)
return self.predictor.forward_chunk(
encoder_out, cache["encoder"], is_final=is_final
)
def cal_decoder_with_predictor(
self, encoder_out, encoder_out_lens, sematic_embeds, ys_pad_lens
):
decoder_outs = self.decoder(
encoder_out, encoder_out_lens, sematic_embeds, ys_pad_lens, self.scama_mask
)
decoder_out = decoder_outs[0]
decoder_out = torch.log_softmax(decoder_out, dim=-1)
return decoder_out, ys_pad_lens
def cal_decoder_with_predictor_chunk(
self, encoder_out, encoder_out_lens, sematic_embeds, ys_pad_lens, cache=None
):
decoder_outs = self.decoder.forward_chunk(
encoder_out, sematic_embeds, cache["decoder"]
)
decoder_out = decoder_outs
decoder_out = torch.log_softmax(decoder_out, dim=-1)
return decoder_out, ys_pad_lens
def init_cache(self, cache: dict = {}, **kwargs):
chunk_size = kwargs.get("chunk_size", [0, 10, 5])
encoder_chunk_look_back = kwargs.get("encoder_chunk_look_back", 0)
decoder_chunk_look_back = kwargs.get("decoder_chunk_look_back", 0)
batch_size = 1
enc_output_size = kwargs["encoder_conf"]["output_size"]
feats_dims = (
kwargs["frontend_conf"]["n_mels"] * kwargs["frontend_conf"]["lfr_m"]
)
cache_encoder = {
"start_idx": 0,
"cif_hidden": torch.zeros((batch_size, 1, enc_output_size)),
"cif_alphas": torch.zeros((batch_size, 1)),
"chunk_size": chunk_size,
"encoder_chunk_look_back": encoder_chunk_look_back,
"last_chunk": False,
"opt": None,
"feats": torch.zeros(
(batch_size, chunk_size[0] + chunk_size[2], feats_dims)
),
"tail_chunk": False,
}
cache["encoder"] = cache_encoder
cache_decoder = {
"decode_fsmn": None,
"decoder_chunk_look_back": decoder_chunk_look_back,
"opt": None,
"chunk_size": chunk_size,
}
cache["decoder"] = cache_decoder
cache["frontend"] = {}
cache["prev_samples"] = torch.empty(0)
return cache
def generate_chunk(
self,
speech,
speech_lengths=None,
key: list = None,
tokenizer=None,
frontend=None,
**kwargs,
):
cache = kwargs.get("cache", {})
speech = speech.to(device=kwargs["device"])
speech_lengths = speech_lengths.to(device=kwargs["device"])
# Encoder
#
encoder_out, encoder_out_lens = self.encode_chunk(
speech, speech_lengths, cache=cache, is_final=kwargs.get("is_final", False)
)
print(speech.shape, encoder_out.shape, encoder_out_lens)
if isinstance(encoder_out, tuple):
encoder_out = encoder_out[0]
# predictor
predictor_outs = self.calc_predictor_chunk(
encoder_out,
encoder_out_lens,
cache=cache,
is_final=kwargs.get("is_final", False),
)
pre_acoustic_embeds, pre_token_length, alphas, pre_peak_index = (
predictor_outs[0],
predictor_outs[1],
predictor_outs[2],
predictor_outs[3],
)
pre_token_length = pre_token_length.round().long()
if torch.max(pre_token_length) < 1:
return []
decoder_outs = self.cal_decoder_with_predictor_chunk(
encoder_out,
encoder_out_lens,
pre_acoustic_embeds,
pre_token_length,
cache=cache,
)
decoder_out, ys_pad_lens = decoder_outs[0], decoder_outs[1]
results = []
b, n, d = decoder_out.size()
if isinstance(key[0], (list, tuple)):
key = key[0]
for i in range(b):
x = encoder_out[i, : encoder_out_lens[i], :]
am_scores = decoder_out[i, : pre_token_length[i], :]
if self.beam_search is not None:
nbest_hyps = self.beam_search(
x=x,
am_scores=am_scores,
maxlenratio=kwargs.get("maxlenratio", 0.0),
minlenratio=kwargs.get("minlenratio", 0.0),
)
nbest_hyps = nbest_hyps[: self.nbest]
else:
yseq = am_scores.argmax(dim=-1)
score = am_scores.max(dim=-1)[0]
score = torch.sum(score, dim=-1)
# pad with mask tokens to ensure compatibility with sos/eos tokens
yseq = torch.tensor(
[self.sos] + yseq.tolist() + [self.eos], device=yseq.device
)
nbest_hyps = [Hypothesis(yseq=yseq, score=score)]
for nbest_idx, hyp in enumerate(nbest_hyps):
# remove sos/eos and get results
last_pos = -1
if isinstance(hyp.yseq, list):
token_int = hyp.yseq[1:last_pos]
else:
token_int = hyp.yseq[1:last_pos].tolist()
# remove blank symbol id, which is assumed to be 0
token_int = list(
filter(
lambda x: x != self.eos
and x != self.sos
and x != self.blank_id,
token_int,
)
)
# Change integer-ids to tokens
token = tokenizer.ids2tokens(token_int)
# text = tokenizer.tokens2text(token)
result_i = token
results.extend(result_i)
return results
def inference(
self,
data_in,
data_lengths=None,
key: list = None,
tokenizer=None,
frontend=None,
cache: dict = {},
**kwargs,
):
# init beamsearch
is_use_ctc = (
kwargs.get("decoding_ctc_weight", 0.0) > 0.00001 and self.ctc != None
)
is_use_lm = (
kwargs.get("lm_weight", 0.0) > 0.00001
and kwargs.get("lm_file", None) is not None
)
if self.beam_search is None and (is_use_lm or is_use_ctc):
logging.info("enable beam_search")
self.init_beam_search(**kwargs)
self.nbest = kwargs.get("nbest", 1)
if len(cache) == 0:
self.init_cache(cache, **kwargs)
meta_data = {}
chunk_size = kwargs.get("chunk_size", [0, 10, 5])
chunk_stride_samples = int(chunk_size[1] * 960) # 600ms
time1 = time.perf_counter()
cfg = {"is_final": kwargs.get("is_final", False)}
audio_sample_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,
cache=cfg,
)
# import pdb; pdb.set_trace()
_is_final = cfg["is_final"] # if data_in is a file or url, set is_final=True
time2 = time.perf_counter()
meta_data["load_data"] = f"{time2 - time1:0.3f}"
assert len(audio_sample_list) == 1, "batch_size must be set 1"
audio_sample = torch.cat((cache["prev_samples"], audio_sample_list[0]))
n = int(len(audio_sample) // chunk_stride_samples + int(_is_final))
m = int(len(audio_sample) % chunk_stride_samples * (1 - int(_is_final)))
tokens = []
for i in range(n):
kwargs["is_final"] = _is_final and i == n - 1
audio_sample_i = audio_sample[
i * chunk_stride_samples : (i + 1) * chunk_stride_samples
]
# extract fbank feats
speech, speech_lengths = extract_fbank(
[audio_sample_i],
data_type=kwargs.get("data_type", "sound"),
frontend=frontend,
cache=cache["frontend"],
is_final=kwargs["is_final"],
)
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
)
if len(speech) == 0:
break
tokens_i = self.generate_chunk(
speech,
speech_lengths,
key=key,
tokenizer=tokenizer,
cache=cache,
frontend=frontend,
**kwargs,
)
tokens.extend(tokens_i)
text_postprocessed, _ = postprocess_utils.sentence_postprocess(tokens)
result_i = {"key": key[0], "text": text_postprocessed}
result = [result_i]
cache["prev_samples"] = audio_sample[:-m]
if _is_final:
self.init_cache(cache, **kwargs)
if kwargs.get("output_dir"):
if not hasattr(self, "writer"):
self.writer = DatadirWriter(kwargs.get("output_dir"))
ibest_writer = self.writer[f"{1}best_recog"]
ibest_writer["token"][key[0]] = " ".join(tokens)
ibest_writer["text"][key[0]] = text_postprocessed
return result, meta_data
def infer_encoder(
self,
data_in,
data_lengths=None,
key: list = None,
tokenizer=None,
frontend=None,
cache: dict = {},
**kwargs,
):
if len(cache) == 0:
self.init_cache(cache, **kwargs)
meta_data = {}
chunk_size = kwargs.get("chunk_size", [0, 10, 5])
chunk_stride_samples = int(chunk_size[1] * 960) # 600ms
time1 = time.perf_counter()
cfg = {"is_final": kwargs.get("is_final", False)}
if isinstance(data_in[0], torch.Tensor):
audio_sample_list = data_in
else:
audio_sample_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,
cache=cfg,
)
_is_final = cfg["is_final"] # if data_in is a file or url, set is_final=True
time2 = time.perf_counter()
meta_data["load_data"] = f"{time2 - time1:0.3f}"
assert len(audio_sample_list) == 1, "batch_size must be set 1"
audio_sample = torch.cat((cache["prev_samples"], audio_sample_list[0]))
n = int(len(audio_sample) // chunk_stride_samples + int(_is_final))
m = int(len(audio_sample) % chunk_stride_samples * (1 - int(_is_final)))
encoder_outs = []
meta_data["batch_data_time"] = 0.0
meta_data["extract_feat"] = 0.0
for i in range(n):
kwargs["is_final"] = _is_final and i == n - 1
audio_sample_i = audio_sample[
i * chunk_stride_samples : (i + 1) * chunk_stride_samples
]
time2 = time.perf_counter()
# extract fbank feats
if kwargs["is_final"] and len(audio_sample_i) == 0:
break
try:
speech, speech_lengths = extract_fbank(
[audio_sample_i],
data_type=kwargs.get("data_type", "sound"),
frontend=frontend,
cache=cache["frontend"],
is_final=kwargs["is_final"],
)
except:
if i == n - 1 and audio_sample_i.shape[0] < 480:
print(f"Warning!!!, skip {audio_sample_i.shape[0]} samples")
break
else:
raise RuntimeError("infer failed")
time3 = time.perf_counter()
if len(speech) == 0 and kwargs["is_final"]:
break
meta_data["extract_feat"] = meta_data["extract_feat"] + time3 - time2
meta_data["batch_data_time"] = (
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_out, encoder_out_lens = self.encode_chunk(
speech,
speech_lengths,
cache=cache,
is_final=kwargs.get("is_final", False),
)
encoder_outs.append(encoder_out[:, (-speech_lengths[0]) :])
if i == n - 1:
break
speech_out = []
if len(encoder_outs) > 0:
speech_out = torch.cat(encoder_outs, dim=1)
result_i = {"key": key[0], "enc_out": speech_out}
result = [result_i]
if m > 0: # tail exists
cache["prev_samples"] = audio_sample[-m:]
else:
cache["prev_samples"] = torch.empty(0)
if _is_final:
self.init_cache(cache, **kwargs)
return result, meta_data, cache