From 35b1c051f6db3649a818547902497d219c871b84 Mon Sep 17 00:00:00 2001
From: zhifu gao <zhifu.gzf@alibaba-inc.com>
Date: 星期四, 14 三月 2024 09:33:30 +0800
Subject: [PATCH] Dev gzf llm (#1493)
---
funasr/models/llm_asr_nar/model.py | 333 +++++++++++++++++++++++++++++++++++++++++++++++++++++++
1 files changed, 333 insertions(+), 0 deletions(-)
diff --git a/funasr/models/llm_asr_nar/model.py b/funasr/models/llm_asr_nar/model.py
index d83f571..a6096b2 100644
--- a/funasr/models/llm_asr_nar/model.py
+++ b/funasr/models/llm_asr_nar/model.py
@@ -16,6 +16,7 @@
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.models.paraformer.cif_predictor import mae_loss
from funasr.utils.datadir_writer import DatadirWriter
from funasr.register import tables
@@ -348,3 +349,335 @@
return results, meta_data
+
+@tables.register("model_classes", "LLMASRNARPrompt")
+class LLMASRNARPrompt(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,
+ predictor_weight: int = 1.0,
+ 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
+ self.predictor_weight = predictor_weight
+
+ 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,
+ )
+ self.criterion_pre = mae_loss(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, loss_pre = 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:
+ # inputs_embeds锛� [bos, prompt, input, pad, target]
+ prompt_bos_length = kwargs.get("prompt_bos_length", None)
+ assert prompt_bos_length is not None
+ prompt_bos_length = prompt_bos_length[0].item()
+ batch_size, token_num, dims = inputs_embeds.shape
+ _, l, _ = encoder_out.shape
+ encoder_outs_pad = F.pad(encoder_out, (0, 0, prompt_bos_length, token_num - prompt_bos_length - l, 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) # [prompt, input, pad, target, 0.0]
+
+ # labels_ids: [bos, prompt, input, target, eos] -> [-1, -1, input, target, eos]
+ # loss:
+ # inputs_embeds[:-1] -> [prompt, input, pad, target]
+ # labels_ids[1:] -> [prompt, input, target, eos] -> [-1, input, target, eos];
+ model_outputs = self.llm(inputs_embeds=inputs_embeds, attention_mask=attention_mask, labels=labels_ids)
+ loss_llm = model_outputs.loss
+ loss = loss_llm + loss_pre * self.predictor_weight
+ 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_pre"] = torch.clone(loss_pre.detach())
+ stats["loss_llm"] = torch.clone(loss_llm.detach())
+ 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,
+ ):
+
+ audio_mask = kwargs.get("audio_mask", None)
+ audio_token_lengths = audio_mask.sum(-1) if audio_mask is not None else None
+ text_token_int = kwargs.get("text_token_int", None)
+ if audio_token_lengths is None:
+ audio_token_lengths = torch.tensor([len(text_token_int)], dtype=torch.int64)
+
+ 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,
+ )
+ loss_pre = self.criterion_pre(audio_token_lengths.type_as(pre_token_length), pre_token_length)
+
+ return pre_acoustic_embeds, pre_token_length, loss_pre
+
+ 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=None)
+ if len(kwargs.get("data_type")) > 1:
+ audio_sample_list, text_token_int_list = audio_sample_list
+ text_token_int = text_token_int_list[0].replace(" ", "")
+ text_token_int = tokenizer.encode(text_token_int)
+ else:
+ text_token_int = None
+ 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, text_token_int=text_token_int)
+
+ # 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
--
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