update cmakelist

雾聪

2024-03-14 0cf5dfec2c8313fc2ed2aab8d10bf3dc4b9c283f

 funasr/models/llm_asr/model.py

@@ -12,7 +12,7 @@
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.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
@@ -20,8 +20,8 @@
from funasr.register import tables


@tables.register("model_classes", "LLMASRNAR")
class LLMASRNAR(nn.Module):
@tables.register("model_classes", "LLMASR")
class LLMASR(nn.Module):
    """ """
    
    def __init__(
@@ -30,8 +30,10 @@
        specaug_conf: dict = None,
        normalize: str = None,
        normalize_conf: dict = None,
        encoder: str = None,
        encoder_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,
@@ -39,8 +41,6 @@
        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,
@@ -70,23 +70,30 @@
            normalize = normalize_class(**normalize_conf)
        
        # audio encoder
        hub = encoder_conf.get("hub", None)
        if hub == "funasr":
        hub = audio_encoder_conf.get("hub", None)
        if hub == "ms":
            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")
            model = AutoModel(model=audio_encoder, model_revision="v2.0.4")
            # frontend = model.kwargs.get("frontend")
            model.model.decoder = None
            audio_encoder_output_size = model.model.encoder_output_size

            audio_encoder = model.model.model.encoder
            
            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()
            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")
@@ -95,6 +102,7 @@
            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,
@@ -109,10 +117,11 @@
            self.llm = model
        
        # adaptor
        adaptor_class = tables.adaptor_classes.get(adaptor)
        adaptor = adaptor_class(**adaptor_conf)
        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.adaptor = adaptor
        self.audio_adaptor = audio_adaptor
        
        
        self.blank_id = blank_id
@@ -122,8 +131,6 @@
        self.ignore_id = ignore_id
        self.specaug = specaug
        self.normalize = normalize
        self.encoder = encoder


        self.criterion_att = LabelSmoothingLoss(
            size=vocab_size,
@@ -131,12 +138,7 @@
            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
@@ -172,27 +174,26 @@
        batch_size = speech.shape[0]
        
        # audio encoder
        encoder_out, encoder_out_lens = self.encode(speech, speech_lengths, audio_mask=audio_mask)
        encoder_out, encoder_out_lens = self.encode(speech, speech_lengths)
        
        # adaptor
        encoder_out = self.adaptor(encoder_out)
        # 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 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)
        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
@@ -211,26 +212,18 @@
            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]:
    ):
        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
    
        audio_mask = kwargs.get("audio_mask")
        audio_token_lengths = audio_mask.sum(-1) if audio_mask 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,
@@ -275,7 +268,7 @@
        encoder_out, encoder_out_lens = self.encode(speech, speech_lengths)

        # adaptor
        encoder_out = self.adaptor(encoder_out)
        encoder_out = self.audio_adaptor(encoder_out)
        
    
        prompt_pre = "USER: \nINSTRUCTION: {}\nINPUT: ".format(prompt)
@@ -294,7 +287,7 @@
        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(
        preds = self.llm.generate(
            inputs_embeds=inputs_embeds,
            max_length=kwargs.get("max_length", 200),
            max_new_tokens=kwargs.get("max_new_tokens", 200),
@@ -311,7 +304,13 @@
            pad_token_id=tokenizer.pad_token_id
        )

        text = tokenizer.batch_decode(model_outputs, add_special_tokens=False, skip_special_tokens=True)

        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: