FunASR/funasr/models/llm_asr_nar/model.py
zhifu gao c456abaf33
Dev gzf (#1405)
* init param
2024-02-29 16:23:52 +08:00

342 lines
13 KiB
Python

import logging
from typing import Union, Dict, List, Tuple, Optional
import time
import torch
import torch.nn as nn
import torch.nn.functional as F
from torch.cuda.amp import autocast
from funasr.models.scama.utils import sequence_mask
from funasr.losses.label_smoothing_loss import LabelSmoothingLoss
from funasr.models.ctc.ctc import CTC
from funasr.models.transformer.utils.add_sos_eos import add_sos_eos
from funasr.metrics.compute_acc import th_accuracy, compute_accuracy
# from funasr.models.e2e_asr_common import ErrorCalculator
from funasr.train_utils.device_funcs import force_gatherable
from funasr.utils.load_utils import load_audio_text_image_video, extract_fbank
from funasr.utils import postprocess_utils
from funasr.utils.datadir_writer import DatadirWriter
from funasr.register import tables
@tables.register("model_classes", "LLMASRNAR")
class LLMASRNAR(nn.Module):
""" """
def __init__(
self,
specaug: str = None,
specaug_conf: dict = None,
normalize: str = None,
normalize_conf: dict = None,
encoder: str = None,
encoder_conf: dict = None,
decoder: str = None,
decoder_conf: dict = None,
ctc: str = None,
ctc_conf: dict = None,
ctc_weight: float = 0.5,
llm: str = None,
llm_conf: dict = None,
adaptor: str = None,
adaptor_conf: dict = None,
input_size: int = 80,
vocab_size: int = -1,
ignore_id: int = -1,
blank_id: int = 0,
sos: int = 1,
eos: int = 2,
lsm_weight: float = 0.0,
length_normalized_loss: bool = False,
report_cer: bool = True,
report_wer: bool = True,
sym_space: str = "<space>",
sym_blank: str = "<blank>",
# extract_feats_in_collect_stats: bool = True,
share_embedding: bool = False,
# preencoder: Optional[AbsPreEncoder] = None,
# postencoder: Optional[AbsPostEncoder] = None,
**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)
# audio encoder
hub = encoder_conf.get("hub", None)
if hub == "funasr":
from funasr import AutoModel
init_param_path = encoder_conf.get("init_param_path", "iic/speech_paraformer-large_asr_nat-zh-cn-16k-common-vocab8404-pytorch")
model = AutoModel(model=init_param_path, model_revision="v2.0.4")
# frontend = model.kwargs.get("frontend")
model.model.decoder = None
self.audio_encoder = model.model
# self.frontend = frontend
elif hub == "hf":
pass
else:
encoder_class = tables.encoder_classes.get(encoder)
encoder = encoder_class(input_size=input_size, **encoder_conf)
encoder_output_size = encoder.output_size()
# llm
hub = llm_conf.get("hub", "hf")
self.llm = None
if hub == "hf":
from transformers import AutoModelForCausalLM, AutoTokenizer, AutoConfig
init_param_path = llm_conf.get("init_param_path", "vicuna-7b-v1.5")
model = AutoModelForCausalLM.from_pretrained(
init_param_path,
load_in_8bit=None,
device_map=None,
use_cache=None,
)
freeze = llm_conf.get("freeze", True)
if freeze:
for name, param in model.named_parameters():
param.requires_grad = False
model.eval()
self.llm = model
# adaptor
adaptor_class = tables.adaptor_classes.get(adaptor)
adaptor = adaptor_class(**adaptor_conf)
self.adaptor = adaptor
self.blank_id = blank_id
self.sos = sos if sos is not None else vocab_size - 1
self.eos = eos if eos is not None else vocab_size - 1
self.vocab_size = vocab_size
self.ignore_id = ignore_id
self.specaug = specaug
self.normalize = normalize
self.encoder = encoder
self.criterion_att = LabelSmoothingLoss(
size=vocab_size,
padding_idx=ignore_id,
smoothing=lsm_weight,
normalize_length=length_normalized_loss,
)
#
# if report_cer or report_wer:
# self.error_calculator = ErrorCalculator(
# token_list, sym_space, sym_blank, report_cer, report_wer
# )
#
self.error_calculator = None
self.length_normalized_loss = length_normalized_loss
self.beam_search = None
def forward(
self,
speech: torch.Tensor,
speech_lengths: torch.Tensor,
text: torch.Tensor,
text_lengths: torch.Tensor,
input_ids: torch.Tensor,
attention_mask:torch.Tensor,
labels_ids: torch.Tensor,
label_mask: torch.Tensor,
audio_mask: 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()
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]
# audio encoder
encoder_out, encoder_out_lens = self.encode(speech, speech_lengths, audio_mask=audio_mask)
# adaptor
encoder_out = self.adaptor(encoder_out)
if input_ids is not None:
input_ids[input_ids == -1] = 0
input_ids[input_ids == -100] = 0
if hasattr(self.llm.model, "embed_tokens"):
inputs_embeds = self.llm.model.embed_tokens(input_ids)
elif hasattr(self.llm.model.model, "embed_tokens"):
inputs_embeds = self.llm.model.model.embed_tokens(input_ids)
else:
inputs_embeds = self.llm.model.model.model.embed_tokens(input_ids)
if audio_mask is not None:
batch_size, token_num, dims = inputs_embeds.shape
_, l, _ = encoder_out.shape
encoder_outs_pad = F.pad(encoder_out, (0, 0, token_num-l-1, 1, 0, 0), value=0.0)
inputs_embeds = encoder_outs_pad * audio_mask[:, :, None] + inputs_embeds * (1.0-audio_mask[:, :, None])
inputs_embeds = F.pad(inputs_embeds[:, 1:, :], (0, 0, 0, 1, 0, 0), value=0.0)
model_outputs = self.llm(inputs_embeds=inputs_embeds, attention_mask=attention_mask, labels=labels_ids)
loss = model_outputs.loss
stats = {}
with torch.no_grad():
preds = torch.argmax(model_outputs.logits, -1)
acc_att = compute_accuracy(preds[:, :-1], labels_ids[:, 1:], ignore_label=-100)
stats["acc"] = acc_att
stats["loss"] = torch.clone(loss.detach())
# force_gatherable: to-device and to-tensor if scalar for DataParallel
if self.length_normalized_loss:
batch_size = int((text_lengths + 1).sum())
loss, stats, weight = force_gatherable((loss, stats, batch_size), loss.device)
return loss, stats, weight
def encode(
self, speech: torch.Tensor, speech_lengths: torch.Tensor, **kwargs,
) -> Tuple[torch.Tensor, torch.Tensor]:
audio_mask = kwargs.get("audio_mask", None)
audio_token_lengths = audio_mask.sum(-1) if audio_mask is not None else None
batch = {"speech": speech, "speech_lengths": speech_lengths}
enc, enc_lens = self.audio_encoder.encode(**batch)
with autocast(False):
enc_mask = sequence_mask(enc_lens, enc.size(1), device=enc.device)[:, None, :]
pre_acoustic_embeds, pre_token_length, _, _ = self.audio_encoder.predictor(enc,
mask=enc_mask,
target_label_length=audio_token_lengths,
)
return pre_acoustic_embeds, pre_token_length
def inference(self,
data_in,
data_lengths=None,
key: list = None,
tokenizer=None,
frontend=None,
**kwargs,
):
prompt = kwargs.get("prompt", "Transcribe speech to text.")
if kwargs.get("batch_size", 1) > 1:
raise NotImplementedError("batch decoding is not implemented")
meta_data = {}
if isinstance(data_in, torch.Tensor) and kwargs.get("data_type", "sound") == "fbank": # fbank
speech, speech_lengths = data_in, data_lengths
if len(speech.shape) < 3:
speech = speech[None, :, :]
if speech_lengths is None:
speech_lengths = speech.shape[1]
else:
# extract fbank feats
time1 = time.perf_counter()
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)
time2 = time.perf_counter()
meta_data["load_data"] = f"{time2 - time1:0.3f}"
speech, speech_lengths = extract_fbank(audio_sample_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)
# adaptor
encoder_out = self.adaptor(encoder_out)
prompt_pre = "USER: \nINSTRUCTION: {}\nINPUT: ".format(prompt)
prompt_ids = tokenizer.encode(prompt_pre)
prompt_length = len(prompt_ids)
prompt_ids = torch.tensor(prompt_ids, dtype=torch.int64).to(kwargs["device"])
if hasattr(self.llm.model, "embed_tokens"):
inputs_embeds = self.llm.model.embed_tokens(prompt_ids)
elif hasattr(self.llm.model.model, "embed_tokens"):
inputs_embeds = self.llm.model.model.embed_tokens(prompt_ids)
else:
inputs_embeds = self.llm.model.model.model.embed_tokens(prompt_ids)
inputs_embeds = torch.cat((inputs_embeds[None, :, :], encoder_out), dim=1) # [prompt, audio]
attention_mask = torch.ones(inputs_embeds.size()[:-1], dtype=torch.long).to(kwargs["device"])
# model_outputs = self.llm.generate(
# inputs_embeds=inputs_embeds,
# max_length=kwargs.get("max_length", 200),
# max_new_tokens=kwargs.get("max_new_tokens", 200),
# num_beams=kwargs.get("num_beams", 4),
# do_sample=kwargs.get("do_sample", False),
# min_length=kwargs.get("min_length", 1),
# top_p=kwargs.get("top_p", 1.0),
# repetition_penalty=kwargs.get("repetition_penalty", 1.0),
# length_penalty=kwargs.get("length_penalty", 1.0),
# temperature=kwargs.get("temperature", 1.0),
# attention_mask=attention_mask,
# bos_token_id=tokenizer.bos_token_id,
# eos_token_id=tokenizer.eos_token_id,
# pad_token_id=tokenizer.pad_token_id
# )
model_outputs = self.llm(inputs_embeds=inputs_embeds, attention_mask=attention_mask, labels=None)
preds = torch.argmax(model_outputs.logits, -1)
text = tokenizer.batch_decode(preds, add_special_tokens=False, skip_special_tokens=True)
text = text[0].split(': ')[-1]
text = text.strip()
# preds = torch.argmax(model_outputs.logits, -1)
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[f"{0 + 1}best_recog"]
results = []
result_i = {"key": key[0], "text": text}
results.append(result_i)
if ibest_writer is not None:
ibest_writer["text"][key[0]] = text
return results, meta_data