zhifu gao
2024-04-17 eaf9dda9e4d970af3d09db695e9e10c83ef94e25
Dev gzf exp (#1624)

* sensevoice finetune

* sensevoice finetune

* sensevoice finetune

* sensevoice finetune

* sensevoice finetune
8个文件已修改
5个文件已添加
535 ■■■■■ 已修改文件
examples/industrial_data_pretraining/sense_voice/demo.py 4 ●●●● 补丁 | 查看 | 原始文档 | blame | 历史
examples/industrial_data_pretraining/sense_voice/finetune.sh 69 ●●●●● 补丁 | 查看 | 原始文档 | blame | 历史
funasr/auto/auto_model.py 2 ●●●●● 补丁 | 查看 | 原始文档 | blame | 历史
funasr/bin/train.py 2 ●●● 补丁 | 查看 | 原始文档 | blame | 历史
funasr/datasets/audio_datasets/index_ds.py 23 ●●●●● 补丁 | 查看 | 原始文档 | blame | 历史
funasr/datasets/sense_voice_datasets/__init__.py 补丁 | 查看 | 原始文档 | blame | 历史
funasr/datasets/sense_voice_datasets/datasets.py 118 ●●●●● 补丁 | 查看 | 原始文档 | blame | 历史
funasr/losses/label_smoothing_loss.py 4 ●●●● 补丁 | 查看 | 原始文档 | blame | 历史
funasr/models/sense_voice/decoder.py 66 ●●●●● 补丁 | 查看 | 原始文档 | blame | 历史
funasr/models/sense_voice/encoder.py 67 ●●●●● 补丁 | 查看 | 原始文档 | blame | 历史
funasr/models/sense_voice/model.py 131 ●●●●● 补丁 | 查看 | 原始文档 | blame | 历史
funasr/models/sense_voice/whisper_lib/model.py 27 ●●●● 补丁 | 查看 | 原始文档 | blame | 历史
funasr/tokenizer/whisper_tokenizer.py 22 ●●●●● 补丁 | 查看 | 原始文档 | blame | 历史
examples/industrial_data_pretraining/sense_voice/demo.py
@@ -5,13 +5,13 @@
from funasr import AutoModel
model = AutoModel(model="/Users/zhifu/Downloads/modelscope_models/SenseVoice",
model = AutoModel(model="/Users/zhifu/Downloads/modelscope_models/SenseVoiceModelscope",
                  vad_model="iic/speech_fsmn_vad_zh-cn-16k-common-pytorch",
                  vad_kwargs={"max_single_segment_time": 30000},
                  )
input_wav = "https://isv-data.oss-cn-hangzhou.aliyuncs.com/ics/MaaS/SenseVoice/aed_ser/asr_bgm.wav"
input_wav = "https://isv-data.oss-cn-hangzhou.aliyuncs.com/ics/MaaS/ASR/test_audio/asr_example_zh.wav"
DecodingOptions = {
    "task": ("ASR", "AED", "SER"),
examples/industrial_data_pretraining/sense_voice/finetune.sh
New file
@@ -0,0 +1,69 @@
# Copyright FunASR (https://github.com/alibaba-damo-academy/FunASR). All Rights Reserved.
#  MIT License  (https://opensource.org/licenses/MIT)
# which gpu to train or finetune
export CUDA_VISIBLE_DEVICES="0"
gpu_num=$(echo $CUDA_VISIBLE_DEVICES | awk -F "," '{print NF}')
# model_name from model_hub, or model_dir in local path
## option 1, download model automatically
model_name_or_model_dir="iic/speech_paraformer-large_asr_nat-zh-cn-16k-common-vocab8404-pytorch"
model_name_or_model_dir="/Users/zhifu/Downloads/modelscope_models/SenseVoiceModelscope"
## option 2, download model by git
#local_path_root=${workspace}/modelscope_models
#mkdir -p ${local_path_root}/${model_name_or_model_dir}
#git clone https://www.modelscope.cn/${model_name_or_model_dir}.git ${local_path_root}/${model_name_or_model_dir}
#model_name_or_model_dir=${local_path_root}/${model_name_or_model_dir}
# data dir, which contains: train.json, val.json
data_dir="../../../data/list"
train_data="${data_dir}/train.jsonl"
val_data="${data_dir}/val.jsonl"
# generate train.jsonl and val.jsonl from wav.scp and text.txt
scp2jsonl \
++scp_file_list='["../../../data/list/train_wav.scp", "../../../data/list/train_text.txt"]' \
++data_type_list='["source", "target"]' \
++jsonl_file_out="${train_data}"
scp2jsonl \
++scp_file_list='["../../../data/list/val_wav.scp", "../../../data/list/val_text.txt"]' \
++data_type_list='["source", "target"]' \
++jsonl_file_out="${val_data}"
# exp output dir
output_dir="./outputs"
log_file="${output_dir}/log.txt"
mkdir -p ${output_dir}
echo "log_file: ${log_file}"
#torchrun \
#--nnodes 1 \
#--node_rank 0 \
#--nproc_per_node ${gpu_num} \
python \
../../../funasr/bin/train.py \
++model="${model_name_or_model_dir}" \
++train_data_set_list="${train_data}" \
++valid_data_set_list="${val_data}" \
++dataset_conf.batch_size=500 \
++dataset_conf.batch_type="token" \
++dataset_conf.num_workers=0 \
++train_conf.max_epoch=50 \
++train_conf.log_interval=1 \
++train_conf.resume=false \
++train_conf.validate_interval=2000 \
++train_conf.save_checkpoint_interval=2000 \
++train_conf.keep_nbest_models=20 \
++train_conf.avg_nbest_model=10 \
++optim_conf.lr=0.0002 \
++debug=true \
++device="cpu" \
++output_dir="${output_dir}" #&> ${log_file}
funasr/auto/auto_model.py
@@ -175,6 +175,8 @@
            kwargs["token_list"] = tokenizer.token_list if hasattr(tokenizer, "token_list") else None
            kwargs["token_list"] = tokenizer.get_vocab() if hasattr(tokenizer, "get_vocab") else kwargs["token_list"]
            vocab_size = len(kwargs["token_list"]) if kwargs["token_list"] is not None else -1
            if vocab_size == -1 and hasattr(tokenizer, "get_vocab_size"):
                vocab_size = tokenizer.get_vocab_size()
        else:
            vocab_size = -1
        kwargs["tokenizer"] = tokenizer
funasr/bin/train.py
@@ -102,7 +102,7 @@
    if use_ddp:
        model = model.cuda(local_rank)
        model = DDP(model, device_ids=[local_rank],
                    find_unused_parameters=kwargs.get("train_conf", {}).get("find_unused_parameters", False))
                    find_unused_parameters=kwargs.get("train_conf", {}).get("find_unused_parameters", True))
    elif use_fsdp:
        # model = FSDP(model).cuda(local_rank)
funasr/datasets/audio_datasets/index_ds.py
@@ -92,7 +92,7 @@
            for line in fin:
                data = json.loads(line.strip())
                if "text" in data:  # for sft
                    self.contents.append(data['text'])
                    contents.append(data['text'])
                if "source" in data:  # for speech lab pretrain
                    prompt = data.get("prompt", "<ASR>")
                    source = data["source"]
@@ -101,13 +101,20 @@
                    target_len = data.get("target_len", 0)
                    if "aishell" in source:
                        target = target.replace(" ", "")
                    contents.append({"source": source,
                                     "prompt": prompt,
                                     "target": target,
                                     "source_len": source_len,
                                     "target_len": target_len,
                                     }
                                    )
                    contents_i = {"source": source,
                                 "prompt": prompt,
                                 "target": target,
                                 "source_len": source_len,
                                 "target_len": target_len,
                                 }
                    text_language = data.get("text_language", None)
                    if text_language is not None:
                        contents_i["text_language"] = text_language
                    audio_language = data.get("audio_language", None)
                    if audio_language is not None:
                        contents_i["audio_language"] = audio_language
                    contents.append(contents_i)
        self.contents = contents
        
funasr/datasets/sense_voice_datasets/__init__.py
funasr/datasets/sense_voice_datasets/datasets.py
New file
@@ -0,0 +1,118 @@
import torch
import random
from funasr.register import tables
from funasr.utils.load_utils import extract_fbank, load_audio_text_image_video
@tables.register("dataset_classes", "SenseVoiceDataset")
class SenseVoiceDataset(torch.utils.data.Dataset):
    """
    SenseVoiceDataset
    """
    def __init__(self,
                 path,
                 index_ds: str = None,
                 frontend=None,
                 tokenizer=None,
                 int_pad_value: int = -1,
                 float_pad_value: float = 0.0,
                  **kwargs):
        super().__init__()
        index_ds_class = tables.index_ds_classes.get(index_ds)
        self.index_ds = index_ds_class(path, **kwargs)
        preprocessor_speech = kwargs.get("preprocessor_speech", None)
        if preprocessor_speech:
            preprocessor_speech_class = tables.preprocessor_classes.get(preprocessor_speech)
            preprocessor_speech = preprocessor_speech_class(**kwargs.get("preprocessor_speech_conf"))
        self.preprocessor_speech = preprocessor_speech
        preprocessor_text = kwargs.get("preprocessor_text", None)
        if preprocessor_text:
            preprocessor_text_class = tables.preprocessor_classes.get(preprocessor_text)
            preprocessor_text = preprocessor_text_class(**kwargs.get("preprocessor_text_conf"))
        self.preprocessor_text = preprocessor_text
        self.frontend = frontend
        self.fs = 16000 if frontend is None else frontend.fs
        self.data_type = "sound"
        self.tokenizer = tokenizer
        self.int_pad_value = int_pad_value
        self.float_pad_value = float_pad_value
        self.sos = kwargs.get("sos", "<|startoftranscript|>")
        self.eos = kwargs.get("eos", "<|endoftext|>")
    def get_source_len(self, index):
        item = self.index_ds[index]
        return self.index_ds.get_source_len(item)
    def get_target_len(self, index):
        item = self.index_ds[index]
        return self.index_ds.get_target_len(item)
    def __len__(self):
        return len(self.index_ds)
    def __getitem__(self, index):
        item = self.index_ds[index]
        # import pdb;
        # pdb.set_trace()
        source = item["source"]
        data_src = load_audio_text_image_video(source, fs=self.fs)
        if self.preprocessor_speech:
            data_src = self.preprocessor_speech(data_src, fs=self.fs)
        speech, speech_lengths = extract_fbank(data_src, data_type=self.data_type, frontend=self.frontend, is_final=True) # speech: [b, T, d]
        speech = speech.permute(0, 2, 1)
        target = item["target"]
        if self.preprocessor_text:
            target = self.preprocessor_text(target)
        task = item.get("prompt", "<|ASR|>")
        text_language = item.get("text_language", "<|zh|>")
        prompt = f"{self.sos}{task}{text_language}"
        prompt_ids = self.tokenizer.encode(prompt, allowed_special="all")
        prompt_ids_len = len(prompt_ids) - 1 # [sos, task]
        target_ids = self.tokenizer.encode(target, allowed_special="all")
        target_ids_len = len(target_ids) + 1 # [lid, text]
        eos = self.tokenizer.encode(self.eos, allowed_special="all") # [eos]
        ids = prompt_ids + target_ids + eos
        ids_lengths = len(ids)
        text = torch.tensor(ids, dtype=torch.int64)
        text_lengths = torch.tensor([ids_lengths], dtype=torch.int32)
        target_mask = [0] * (prompt_ids_len) + [1] * (target_ids_len) + [1]  # [sos, task, lid, text, eos]: [0, 0, 1, 1, 1]
        target_mask = torch.tensor(target_mask, dtype=torch.float32)
        return {"speech": speech[0, :, :],
                "speech_lengths": speech_lengths,
                "text": text,
                "text_lengths": text_lengths,
                "target_mask": target_mask,
                }
    def collator(self, samples: list=None):
        outputs = {}
        for sample in samples:
            for key in sample.keys():
                if key not in outputs:
                    outputs[key] = []
                outputs[key].append(sample[key])
        for key, data_list in outputs.items():
            if isinstance(data_list[0], torch.Tensor):
                if data_list[0].dtype == torch.int64 or data_list[0].dtype == torch.int32:
                    pad_value = self.int_pad_value
                else:
                    pad_value = self.float_pad_value
                outputs[key] = torch.nn.utils.rnn.pad_sequence(data_list, batch_first=True, padding_value=pad_value)
        return outputs
funasr/losses/label_smoothing_loss.py
@@ -50,8 +50,8 @@
        """
        assert x.size(2) == self.size
        batch_size = x.size(0)
        x = x.view(-1, self.size)
        target = target.view(-1)
        x = x.contiguous().view(-1, self.size)
        target = target.contiguous().view(-1)
        with torch.no_grad():
            true_dist = x.clone()
            true_dist.fill_(self.smoothing / (self.size - 1))
funasr/models/sense_voice/decoder.py
New file
@@ -0,0 +1,66 @@
import copy
from typing import Optional, Tuple, Union
import torch
import torch.nn as nn
import torch.nn.functional as F
from funasr.models.transformer.utils.nets_utils import make_pad_mask
def sense_voice_decode_forward(
    self,
    x: torch.Tensor,
    xa: torch.Tensor,
    kv_cache: Optional[dict] = None,
    **kwargs,
):
    """Forward decoder.
    Args:
        hs_pad: encoded memory, float32  (batch, maxlen_in, feat)
        hlens: (batch)
        ys_in_pad:
            input token ids, int64 (batch, maxlen_out)
            if input_layer == "embed"
            input tensor (batch, maxlen_out, #mels) in the other cases
        ys_in_lens: (batch)
    Returns:
        (tuple): tuple containing:
        x: decoded token score before softmax (batch, maxlen_out, token)
            if use_output_layer is True,
        olens: (batch, )
    """
    # import pdb;pdb.set_trace()
    use_padmask = self.use_padmask
    hlens = kwargs.get("hlens", None)
    ys_in_lens = kwargs.get("ys_in_lens", None)
    offset = next(iter(kv_cache.values())).shape[1] if kv_cache else 0
    tgt, memory = x, xa
    tgt[tgt==-1] = 0
    tgt = (
        self.token_embedding(tgt)
        + self.positional_embedding[offset : offset + tgt.size(1)]
    )
    # tgt = self.dropout(tgt)
    x = tgt.to(memory.dtype)
    if use_padmask and hlens is not None:
        memory_mask = (~make_pad_mask(hlens)[:, None, :]).to(memory.device)
    else:
        memory_mask = None
    for layer, block in enumerate(self.blocks):
        x = block(x, memory, mask=self.mask, memory_mask=memory_mask, is_pad_mask=False, is_pad_memory_mask=True)
    x = self.ln(x)
    x = (
        x @ torch.transpose(self.token_embedding.weight.to(x.dtype), 0, 1)
    ).float()
    return x
funasr/models/sense_voice/encoder.py
New file
@@ -0,0 +1,67 @@
import copy
from typing import Optional, Tuple, Union
import torch
import torch.nn as nn
import torch.nn.functional as F
from funasr.models.transformer.utils.nets_utils import make_pad_mask
def sense_voice_encode_forward(
    self,
    x: torch.Tensor,
    ilens: torch.Tensor = None,
    **kwargs,
):
    use_padmask = self.use_padmask
    x = F.gelu(self.conv1(x))
    x = F.gelu(self.conv2(x))
    x = x.permute(0, 2, 1)
    n_frames = x.size(1)
    max_pos = self.positional_embedding.size(0)
    max_pos = n_frames if n_frames < max_pos else max_pos
    x = (x[:, :max_pos, :] + self.positional_embedding[None, :max_pos, :]).to(x.dtype)
    if ilens is not None:
        if self.downsample_rate == 4:
            olens = (
                1
                + (
                    ilens
                    - self.conv1.kernel_size[0]
                    + 2 * self.conv1.padding[0]
                )
                // self.conv1.stride[0]
            )
        else:
            olens = ilens
        olens = (
            1
            + (
                olens
                - self.conv2.kernel_size[0]
                + 2 * self.conv2.padding[0]
            )
            // self.conv2.stride[0]
        )
        olens = torch.clamp(olens, max=max_pos)
    else:
        olens = None
    if use_padmask and olens is not None:
        padding_mask = (~make_pad_mask(olens)[:, None, :]).to(x.device)
    else:
        padding_mask = None
    for layer, block in enumerate(self.blocks):
        x = block(x, mask=padding_mask, is_pad_mask=True)
    x = self.ln_post(x)
    if ilens is None:
        return x
    else:
        return x, olens
funasr/models/sense_voice/model.py
@@ -1,35 +1,158 @@
from dataclasses import dataclass
from typing import Dict
from typing import Iterable, Optional
import types
import time
import numpy as np
import torch
import torch.nn.functional as F
from torch import Tensor
from torch import nn
from torch.cuda.amp import autocast
from funasr.metrics.compute_acc import compute_accuracy
from funasr.losses.label_smoothing_loss import LabelSmoothingLoss
from funasr.train_utils.device_funcs import force_gatherable
from . import whisper_lib as whisper
from funasr.utils.load_utils import load_audio_text_image_video, extract_fbank
from funasr.register import tables
@tables.register("model_classes", "SenseVoice")
class SenseVoice(nn.Module):
    def __init__(self, *args, **kwargs):
        super().__init__()
        hub = kwargs.get("hub", "funasr")
        dims = kwargs.get("dims", {})
        dims = whisper.model.ModelDimensions(**dims)
        model = whisper.model.Whisper(dims=dims)
        # encoder
        model.encoder.downsample_rate = kwargs.get("downsample_rate", 4)
        model.encoder.use_padmask = kwargs.get("use_padmask", True)
        from .encoder import sense_voice_encode_forward
        model.encoder.forward = types.MethodType(sense_voice_encode_forward, model.encoder)
        # decoder
        model.decoder.use_padmask = kwargs.get("use_padmask", True)
        from .decoder import sense_voice_decode_forward
        model.decoder.forward = types.MethodType(sense_voice_decode_forward, model.decoder)
        
        self.model = model
        
        self.encoder_output_size = self.model.dims.n_audio_state
        
    def forward(self, ):
        pass
        self.activation_checkpoint = kwargs.get("activation_checkpoint", False)
        self.ignore_id = kwargs.get("ignore_id", -1)
        self.vocab_size = kwargs.get("vocab_size", -1)
        self.length_normalized_loss = kwargs.get("length_normalized_loss", True)
        self.criterion_att = LabelSmoothingLoss(
            size=self.vocab_size,
            padding_idx=self.ignore_id,
            smoothing=kwargs.get("lsm_weight", 0.0),
            normalize_length=self.length_normalized_loss,
        )
        specaug = kwargs.get("specaug", None)
        if specaug is not None:
            specaug_class = tables.specaug_classes.get(specaug)
            specaug = specaug_class(**kwargs.get("specaug_conf", {}))
        self.specaug = specaug
    def forward(
        self,
        speech: torch.Tensor,
        speech_lengths: torch.Tensor,
        text: torch.Tensor,
        text_lengths: torch.Tensor,
        **kwargs,
    ):
        target_mask = kwargs.get("target_mask", None)
    
        # 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]
        if self.activation_checkpoint:
            from torch.utils.checkpoint import checkpoint
            encoder_out, encoder_out_lens = checkpoint(self.encode, speech, speech_lengths, use_reentrant=False)
        else:
            encoder_out, encoder_out_lens = self.encode(speech, speech_lengths)
        loss_att, acc_att, cer_att, wer_att = self._calc_att_loss(
            encoder_out, encoder_out_lens, text, text_lengths, target_mask=target_mask
        )
        loss = loss_att
        stats = {}
        stats["acc"] = acc_att
        stats["loss"] = torch.clone(loss.detach())
        stats["batch_size"] = batch_size
        # 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,
    ) :
        """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)
        # Forward encoder
        encoder_out, encoder_out_lens = self.model.encoder(speech.permute(0, 2, 1), speech_lengths)
        return encoder_out, encoder_out_lens
    def _calc_att_loss(
            self,
            encoder_out: torch.Tensor,
            encoder_out_lens: torch.Tensor,
            ys_pad: torch.Tensor,
            ys_pad_lens: torch.Tensor,
            **kwargs,
    ):
        target_mask = kwargs.get("target_mask", None)
        stats = {}
        # 1. Forward decoder
        decoder_out = self.model.decoder(
            x=ys_pad, xa=encoder_out, hlens=encoder_out_lens, ys_in_lens=ys_pad_lens
        )
        # 2. Compute attention loss
        mask = torch.ones_like(ys_pad) * (-1)
        ys_pad_mask = (ys_pad * target_mask + mask * (1-target_mask)).to(torch.int64)
        ys_pad_mask[ys_pad_mask == 0] = -1
        loss_att = self.criterion_att(decoder_out[:, :-1, :], ys_pad_mask[:, 1:])
        with torch.no_grad():
            preds = torch.argmax(decoder_out, -1)
            acc_att = compute_accuracy(preds[:, :-1], ys_pad_mask[:, 1:], ignore_label=self.ignore_id)
        return loss_att, acc_att, None, None
    def inference(self,
                  data_in,
                  data_lengths=None,
funasr/models/sense_voice/whisper_lib/model.py
@@ -74,7 +74,10 @@
        xa: Optional[Tensor] = None,
        mask: Optional[Tensor] = None,
        kv_cache: Optional[dict] = None,
        **kwargs,
    ):
        is_pad_mask = kwargs.get("is_pad_mask", False)
        q = self.query(x)
        if kv_cache is None or xa is None or self.key not in kv_cache:
@@ -87,12 +90,13 @@
            k = kv_cache[self.key]
            v = kv_cache[self.value]
        wv, qk = self.qkv_attention(q, k, v, mask)
        wv, qk = self.qkv_attention(q, k, v, mask, is_pad_mask=is_pad_mask)
        return self.out(wv), qk
    def qkv_attention(
        self, q: Tensor, k: Tensor, v: Tensor, mask: Optional[Tensor] = None
        self, q: Tensor, k: Tensor, v: Tensor, mask: Optional[Tensor] = None, **kwargs,
    ):
        is_pad_mask = kwargs.get("is_pad_mask", False)
        n_batch, n_ctx, n_state = q.shape
        scale = (n_state // self.n_head) ** -0.25
        q = q.view(*q.shape[:2], self.n_head, -1).permute(0, 2, 1, 3) * scale
@@ -101,10 +105,20 @@
        qk = q @ k
        if mask is not None:
            qk = qk + mask[:n_ctx, :n_ctx]
            if not is_pad_mask:
                qk = qk + mask[:n_ctx, :n_ctx]
            else:
                mask = mask.unsqueeze(1).eq(0)  # (batch, 1, *, time2)
                min_value = float(
                    np.finfo(torch.tensor(0, dtype=qk.dtype).numpy().dtype).min
                )
                qk = qk.masked_fill(mask, min_value)
        qk = qk.float()
        w = F.softmax(qk, dim=-1).to(q.dtype)
        if mask is not None and is_pad_mask:
            w = w.masked_fill(mask, 0.0)
        return (w @ v).permute(0, 2, 1, 3).flatten(start_dim=2), qk.detach()
@@ -132,10 +146,13 @@
        xa: Optional[Tensor] = None,
        mask: Optional[Tensor] = None,
        kv_cache: Optional[dict] = None,
        **kwargs,
    ):
        x = x + self.attn(self.attn_ln(x), mask=mask, kv_cache=kv_cache)[0]
        is_pad_mask = kwargs.get("is_pad_mask", False)
        is_pad_memory_mask = kwargs.get("is_pad_memory_mask", False)
        x = x + self.attn(self.attn_ln(x), mask=mask, kv_cache=kv_cache, is_pad_mask=is_pad_mask)[0]
        if self.cross_attn:
            x = x + self.cross_attn(self.cross_attn_ln(x), xa, kv_cache=kv_cache)[0]
            x = x + self.cross_attn(self.cross_attn_ln(x), xa, kv_cache=kv_cache, is_pad_mask=is_pad_memory_mask)[0]
        x = x + self.mlp(self.mlp_ln(x))
        return x
funasr/tokenizer/whisper_tokenizer.py
@@ -22,3 +22,25 @@
    
    return tokenizer
@tables.register("tokenizer_classes", "SenseVoiceTokenizer")
def SenseVoiceTokenizer(**kwargs):
    try:
        from funasr.models.sense_voice.whisper_lib.tokenizer import get_tokenizer
    except:
        print("Notice: If you want to use whisper, please `pip install -U openai-whisper`")
    language = kwargs.get("language", None)
    task = kwargs.get("task", None)
    is_multilingual = kwargs.get("is_multilingual", True)
    num_languages = kwargs.get("num_languages", 8749)
    vocab_path = kwargs.get("vocab_path", None)
    tokenizer = get_tokenizer(
        multilingual=is_multilingual,
        num_languages=num_languages,
        language=language,
        task=task,
        vocab_path=vocab_path,
    )
    return tokenizer