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.metrics.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", "LLMASR") class LLMASR(nn.Module): """ """ def __init__( self, specaug: str = None, specaug_conf: dict = None, normalize: str = None, normalize_conf: dict = None, audio_encoder: str = None, audio_encoder_conf: dict = None, audio_adaptor: str = None, audio_adaptor_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, 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 = "", sym_blank: str = "", # 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 = audio_encoder_conf.get("hub", None) if hub == "ms": from funasr import AutoModel model = AutoModel(model=audio_encoder, model_revision="master") # frontend = model.kwargs.get("frontend") audio_encoder_output_size = model.model.encoder_output_size audio_encoder = model.model.model.encoder # self.frontend = frontend elif hub == "hf": pass else: encoder_class = tables.encoder_classes.get(audio_encoder) audio_encoder = encoder_class(input_size=input_size, **audio_encoder_conf) audio_encoder_output_size = audio_encoder.output_size() freeze = audio_encoder_conf.get("freeze", True) if freeze: for name, param in audio_encoder.named_parameters(): param.requires_grad = False audio_encoder.eval() self.audio_encoder = audio_encoder # 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(audio_adaptor) audio_adaptor_conf["encoder_dim"] = audio_encoder_output_size audio_adaptor = adaptor_class(**audio_adaptor_conf) self.audio_adaptor = audio_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.criterion_att = LabelSmoothingLoss( size=vocab_size, padding_idx=ignore_id, smoothing=lsm_weight, normalize_length=length_normalized_loss, ) 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_adaptor encoder_out = self.audio_adaptor(encoder_out) 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 # [audio, bos, prompt, input, pad] encoder_outs_pad = F.pad(encoder_out, (0, 0, 0, token_num - l, 0, 0), value=0.0) inputs_embeds = encoder_outs_pad * audio_mask[:, :, None] + inputs_embeds * ( 1.0 - audio_mask[:, :, None] ) 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, ): speech = speech.permute(0, 2, 1) res = self.audio_encoder(speech) if isinstance(res, (list, tuple)): encoder_out, encoder_out_lens = res[0], res[1] else: encoder_out, encoder_out_lens = res, speech_lengths return encoder_out, encoder_out_lens 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.audio_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"] ) preds = 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, ) 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