Merge branch 'main' into dev_zly

zhifu gao

2023-02-17 6baf10d5d15bed3948459b30567edd3c5898ff84

Merge branch 'main' into dev_zly

 README.md

@@ -17,21 +17,23 @@

## What's new: 

### 2023.2.16, funasr-0.2.0
- We support a new feature, export paraformer models into [onnx and torchscripts](https://github.com/alibaba-damo-academy/FunASR/tree/main/funasr/export) from modelscopes. The local finetuned models are also supported.
- We support a new feature, [onnxruntime](https://github.com/alibaba-damo-academy/FunASR/tree/main/funasr/runtime/python/onnxruntime/paraformer/rapid_paraformer), you could deploy the runtime without modelscope or funasr, for the [paraformer-large](https://www.modelscope.cn/models/damo/speech_paraformer-large_asr_nat-zh-cn-16k-common-vocab8404-pytorch/summary) model, the rtf of onnxruntime is 3x speedup(0.110->0.038) on cpu.
- We support e new feature, [grpc](https://github.com/alibaba-damo-academy/FunASR/tree/main/funasr/runtime/python/grpc), you could build the ASR service with grpc, by deploying the modelscope pipeline or onnxruntime.
### 2023.2.17, funasr-0.2.0, modelscope-1.3.0
- We support a new feature, export paraformer models into [onnx and torchscripts](https://github.com/alibaba-damo-academy/FunASR/tree/main/funasr/export) from modelscope. The local finetuned models are also supported.
- We support a new feature, [onnxruntime](https://github.com/alibaba-damo-academy/FunASR/tree/main/funasr/runtime/python/onnxruntime/paraformer/rapid_paraformer), you could deploy the runtime without modelscope or funasr, for the [paraformer-large](https://www.modelscope.cn/models/damo/speech_paraformer-large_asr_nat-zh-cn-16k-common-vocab8404-pytorch/summary) model, the rtf of onnxruntime is 3x speedup(0.110->0.038) on cpu, [details](https://github.com/alibaba-damo-academy/FunASR/tree/main/funasr/runtime/python/onnxruntime/paraformer/rapid_paraformer#speed).
- We support a new feature, [grpc](https://github.com/alibaba-damo-academy/FunASR/tree/main/funasr/runtime/python/grpc), you could build the ASR service with grpc, by deploying the modelscope pipeline or onnxruntime.
- We release a new model [paraformer-large-contextual](https://www.modelscope.cn/models/damo/speech_paraformer-large-contextual_asr_nat-zh-cn-16k-common-vocab8404/summary), which supports the hotword customization based on the incentive enhancement, and improves the recall and precision of hotwords.
- We optimize the timestamp alignment of [Paraformer-large-long](https://modelscope.cn/models/damo/speech_paraformer-large-vad-punc_asr_nat-zh-cn-16k-common-vocab8404-pytorch/summary), the prediction accuracy of timestamp is much improved, and achieving accumulated average shift (aas) of 74.7ms, [details](https://arxiv.org/abs/2301.12343).
- We release a new model, [8k VAD model](https://modelscope.cn/models/damo/speech_fsmn_vad_zh-cn-16k-common-pytorch/summary), which could predict the duration of none-silence speech. It could be freely integrated with any ASR models in [modelscope](https://www.modelscope.cn/models/damo/speech_paraformer-large-vad-punc_asr_nat-zh-cn-16k-common-vocab8404-pytorch/summary).
- We release a new model, [MFCCA](https://www.modelscope.cn/models/yufan6/speech_mfcca_asr-zh-cn-16k-alimeeting-vocab4950/summary), a multi-channel multi-speaker model which is independent of the number and geometry of microphones and supports Mandarin meeting transcription.
- We release a new model, [MFCCA](https://www.modelscope.cn/models/NPU-ASLP/speech_mfcca_asr-zh-cn-16k-alimeeting-vocab4950/summary), a multi-channel multi-speaker model which is independent of the number and geometry of microphones and supports Mandarin meeting transcription.
- We release several new UniASR model: 
[Southern Fujian Dialect model](https://modelscope.cn/models/damo/speech_UniASR_asr_2pass-minnan-16k-common-vocab3825/summary), 
[Southern Fujian Dialect model](https://modelscope.cn/models/damo/speech_UniASR_asr_2pass-minnan-16k-common-vocab3825/summary),
[French model](https://modelscope.cn/models/damo/speech_UniASR_asr_2pass-fr-16k-common-vocab3472-tensorflow1-online/summary), 
[German model](https://modelscope.cn/models/damo/speech_UniASR_asr_2pass-de-16k-common-vocab3690-tensorflow1-online/summary), 
[Vietnamese model](https://modelscope.cn/models/damo/speech_UniASR_asr_2pass-vi-16k-common-vocab1001-pytorch-online/summary), 
[Persian model](https://modelscope.cn/models/damo/speech_UniASR_asr_2pass-fa-16k-common-vocab1257-pytorch-online/summary).
- We release a new model, [paraformer-data2vec model](https://www.modelscope.cn/models/damo/speech_data2vec_pretrain-paraformer-zh-cn-aishell2-16k/summary), an unsupervised pretraining model on AISHELL-2, which is inited for paraformer model and then finetune on AISHEL-1.
### 2023.1.16, funasr-0.1.6
- Various new types of audio input types are now supported by modelscope inference pipeline, including: mp3、flac、ogg、opus...
### 2023.1.16, funasr-0.1.6， modelscope-1.2.0
- We release a new version model [Paraformer-large-long](https://modelscope.cn/models/damo/speech_paraformer-large-vad-punc_asr_nat-zh-cn-16k-common-vocab8404-pytorch/summary), which integrate the [VAD](https://modelscope.cn/models/damo/speech_fsmn_vad_zh-cn-16k-common-pytorch/summary) model, [ASR](https://www.modelscope.cn/models/damo/speech_paraformer-large_asr_nat-zh-cn-16k-common-vocab8404-pytorch/summary),
 [Punctuation](https://www.modelscope.cn/models/damo/punc_ct-transformer_zh-cn-common-vocab272727-pytorch/summary) model and timestamp together. The model could take in several hours long inputs.
- We release a new model, [16k VAD model](https://modelscope.cn/models/damo/speech_fsmn_vad_zh-cn-16k-common-pytorch/summary), which could predict the duration of none-silence speech. It could be freely integrated with any ASR models in [modelscope](https://www.modelscope.cn/models/damo/speech_paraformer-large-vad-punc_asr_nat-zh-cn-16k-common-vocab8404-pytorch/summary).
@@ -101,4 +103,10 @@
  booktitle={INTERSPEECH},
  year={2022}
}
@inproceedings{Shi2023AchievingTP,
  title={Achieving Timestamp Prediction While Recognizing with Non-Autoregressive End-to-End ASR Model},
  author={Xian Shi and Yanni Chen and Shiliang Zhang and Zhijie Yan},
  booktitle={arXiv preprint arXiv:2301.12343}
  year={2023}
}
```

 egs_modelscope/asr/mfcca/speech_mfcca_asr-zh-cn-16k-alimeeting-vocab4950/RESULTS.md

@@ -1,5 +1,5 @@
# Paraformer-Large
- Model link: <https://www.modelscope.cn/models/yufan6/speech_mfcca_asr-zh-cn-16k-alimeeting-vocab4950/summary>
- Model link: <https://www.modelscope.cn/models/NPU-ASLP/speech_mfcca_asr-zh-cn-16k-alimeeting-vocab4950/summary>
- Model size: 45M

# Environments

 egs_modelscope/asr/mfcca/speech_mfcca_asr-zh-cn-16k-alimeeting-vocab4950/finetune.py

@@ -24,12 +24,12 @@

if __name__ == '__main__':
    
    params = modelscope_args(model="yufan6/speech_mfcca_asr-zh-cn-16k-alimeeting-vocab4950")
    params = modelscope_args(model="NPU-ASLP/speech_mfcca_asr-zh-cn-16k-alimeeting-vocab4950")
    params.output_dir = "./checkpoint"              # m模型保存路径
    params.data_path = "./example_data/"            # 数据路径
    params.dataset_type = "small"                   # 小数据量设置small，若数据量大于1000小时，请使用large
    params.batch_bins = 1000                       # batch size，如果dataset_type="small"，batch_bins单位为fbank特征帧数，如果dataset_type="large"，batch_bins单位为毫秒，
    params.max_epoch = 10                           # 最大训练轮数
    params.lr = 0.0001                             # 设置学习率
    params.model_revision = 'v2.0.0'
    params.model_revision = 'v1.0.0'
    modelscope_finetune(params)

 egs_modelscope/asr/mfcca/speech_mfcca_asr-zh-cn-16k-alimeeting-vocab4950/infer.py

@@ -18,8 +18,8 @@
        os.environ['CUDA_VISIBLE_DEVICES'] = str(gpu_id)
    inference_pipline = pipeline(
        task=Tasks.auto_speech_recognition,
        model='yufan6/speech_mfcca_asr-zh-cn-16k-alimeeting-vocab4950',
        model_revision='v2.0.0',
        model='NPU-ASLP/speech_mfcca_asr-zh-cn-16k-alimeeting-vocab4950',
        model_revision='v1.0.0',
        output_dir=output_dir_job,
        batch_size=1,
    )

 egs_modelscope/asr/mfcca/speech_mfcca_asr-zh-cn-16k-alimeeting-vocab4950/infer_after_finetune.py

@@ -59,7 +59,7 @@

if __name__ == '__main__':
    params = {}
    params["modelscope_model_name"] = "yufan6/speech_mfcca_asr-zh-cn-16k-alimeeting-vocab4950"
    params["modelscope_model_name"] = "NPU-ASLP/speech_mfcca_asr-zh-cn-16k-alimeeting-vocab4950"
    params["required_files"] = ["feats_stats.npz", "decoding.yaml", "configuration.json"]
    params["output_dir"] = "./checkpoint"
    params["data_dir"] = "./example_data/validation"

 funasr/bin/asr_inference_paraformer.py

@@ -41,16 +41,7 @@
from funasr.utils import asr_utils, wav_utils, postprocess_utils
from funasr.models.frontend.wav_frontend import WavFrontend
from funasr.models.e2e_asr_paraformer import BiCifParaformer, ContextualParaformer


header_colors = '\033[95m'
end_colors = '\033[0m'

global_asr_language: str = 'zh-cn'
global_sample_rate: Union[int, Dict[Any, int]] = {
    'audio_fs': 16000,
    'model_fs': 16000
}
from funasr.export.models.e2e_asr_paraformer import Paraformer as Paraformer_export


class Speech2Text:
@@ -346,6 +337,160 @@
        # assert check_return_type(results)
        return results

class Speech2TextExport:
    """Speech2TextExport class

    """

    def __init__(
            self,
            asr_train_config: Union[Path, str] = None,
            asr_model_file: Union[Path, str] = None,
            cmvn_file: Union[Path, str] = None,
            lm_train_config: Union[Path, str] = None,
            lm_file: Union[Path, str] = None,
            token_type: str = None,
            bpemodel: str = None,
            device: str = "cpu",
            maxlenratio: float = 0.0,
            minlenratio: float = 0.0,
            dtype: str = "float32",
            beam_size: int = 20,
            ctc_weight: float = 0.5,
            lm_weight: float = 1.0,
            ngram_weight: float = 0.9,
            penalty: float = 0.0,
            nbest: int = 1,
            frontend_conf: dict = None,
            hotword_list_or_file: str = None,
            **kwargs,
    ):

        # 1. Build ASR model
        asr_model, asr_train_args = ASRTask.build_model_from_file(
            asr_train_config, asr_model_file, cmvn_file, device
        )
        frontend = None
        if asr_train_args.frontend is not None and asr_train_args.frontend_conf is not None:
            frontend = WavFrontend(cmvn_file=cmvn_file, **asr_train_args.frontend_conf)

        logging.info("asr_model: {}".format(asr_model))
        logging.info("asr_train_args: {}".format(asr_train_args))
        asr_model.to(dtype=getattr(torch, dtype)).eval()

        token_list = asr_model.token_list



        logging.info(f"Decoding device={device}, dtype={dtype}")

        # 5. [Optional] Build Text converter: e.g. bpe-sym -> Text
        if token_type is None:
            token_type = asr_train_args.token_type
        if bpemodel is None:
            bpemodel = asr_train_args.bpemodel

        if token_type is None:
            tokenizer = None
        elif token_type == "bpe":
            if bpemodel is not None:
                tokenizer = build_tokenizer(token_type=token_type, bpemodel=bpemodel)
            else:
                tokenizer = None
        else:
            tokenizer = build_tokenizer(token_type=token_type)
        converter = TokenIDConverter(token_list=token_list)
        logging.info(f"Text tokenizer: {tokenizer}")

        # self.asr_model = asr_model
        self.asr_train_args = asr_train_args
        self.converter = converter
        self.tokenizer = tokenizer

        self.device = device
        self.dtype = dtype
        self.nbest = nbest
        self.frontend = frontend

        model = Paraformer_export(asr_model, onnx=False)
        self.asr_model = model
        
    @torch.no_grad()
    def __call__(
            self, speech: Union[torch.Tensor, np.ndarray], speech_lengths: Union[torch.Tensor, np.ndarray] = None
    ):
        """Inference

        Args:
                speech: Input speech data
        Returns:
                text, token, token_int, hyp

        """
        assert check_argument_types()

        # Input as audio signal
        if isinstance(speech, np.ndarray):
            speech = torch.tensor(speech)

        if self.frontend is not None:
            feats, feats_len = self.frontend.forward(speech, speech_lengths)
            feats = to_device(feats, device=self.device)
            feats_len = feats_len.int()
            self.asr_model.frontend = None
        else:
            feats = speech
            feats_len = speech_lengths

        enc_len_batch_total = feats_len.sum()
        lfr_factor = max(1, (feats.size()[-1] // 80) - 1)
        batch = {"speech": feats, "speech_lengths": feats_len}

        # a. To device
        batch = to_device(batch, device=self.device)

        decoder_outs = self.asr_model(**batch)
        decoder_out, ys_pad_lens = decoder_outs[0], decoder_outs[1]
        
        results = []
        b, n, d = decoder_out.size()
        for i in range(b):
            am_scores = decoder_out[i, :ys_pad_lens[i], :]

            yseq = am_scores.argmax(dim=-1)
            score = am_scores.max(dim=-1)[0]
            score = torch.sum(score, dim=-1)
            # pad with mask tokens to ensure compatibility with sos/eos tokens
            yseq = torch.tensor(
                yseq.tolist(), device=yseq.device
            )
            nbest_hyps = [Hypothesis(yseq=yseq, score=score)]

            for hyp in nbest_hyps:
                assert isinstance(hyp, (Hypothesis)), type(hyp)

                # remove sos/eos and get results
                last_pos = -1
                if isinstance(hyp.yseq, list):
                    token_int = hyp.yseq[1:last_pos]
                else:
                    token_int = hyp.yseq[1:last_pos].tolist()

                # remove blank symbol id, which is assumed to be 0
                token_int = list(filter(lambda x: x != 0 and x != 2, token_int))

                # Change integer-ids to tokens
                token = self.converter.ids2tokens(token_int)

                if self.tokenizer is not None:
                    text = self.tokenizer.tokens2text(token)
                else:
                    text = None

                results.append((text, token, token_int, hyp, enc_len_batch_total, lfr_factor))

        return results


def inference(
        maxlenratio: float,
@@ -454,9 +599,11 @@
        level=log_level,
        format="%(asctime)s (%(module)s:%(lineno)d) %(levelname)s: %(message)s",
    )

    
    export_mode = False
    if param_dict is not None:
        hotword_list_or_file = param_dict.get('hotword')
        export_mode = param_dict.get("export_mode", False)
    else:
        hotword_list_or_file = None

@@ -490,7 +637,10 @@
        nbest=nbest,
        hotword_list_or_file=hotword_list_or_file,
    )
    speech2text = Speech2Text(**speech2text_kwargs)
    if export_mode:
        speech2text = Speech2TextExport(**speech2text_kwargs)
    else:
        speech2text = Speech2Text(**speech2text_kwargs)

    def _forward(
            data_path_and_name_and_type,

 funasr/bin/asr_inference_paraformer_vad.py

@@ -38,7 +38,6 @@
from funasr.utils import asr_utils, wav_utils, postprocess_utils
from funasr.models.frontend.wav_frontend import WavFrontend
from funasr.tasks.vad import VADTask
from funasr.utils.timestamp_tools import time_stamp_lfr6
from funasr.bin.punctuation_infer import Text2Punc
from funasr.bin.asr_inference_paraformer_vad_punc import Speech2Text
from funasr.bin.asr_inference_paraformer_vad_punc import Speech2VadSegment

 funasr/bin/asr_inference_paraformer_vad_punc.py

@@ -39,7 +39,7 @@
from funasr.utils import asr_utils, wav_utils, postprocess_utils
from funasr.models.frontend.wav_frontend import WavFrontend
from funasr.tasks.vad import VADTask
from funasr.utils.timestamp_tools import time_stamp_lfr6, time_stamp_lfr6_pl
from funasr.utils.timestamp_tools import time_stamp_lfr6_pl
from funasr.bin.punctuation_infer import Text2Punc
from funasr.models.e2e_asr_paraformer import BiCifParaformer

@@ -282,13 +282,10 @@
                else:
                    text = None

                if isinstance(self.asr_model, BiCifParaformer):
                    timestamp = time_stamp_lfr6_pl(us_alphas[i], us_cif_peak[i], copy.copy(token), begin_time, end_time)
                    results.append((text, token, token_int, timestamp, enc_len_batch_total, lfr_factor))
                else:
                    time_stamp = time_stamp_lfr6(alphas[i:i + 1, ], enc_len[i:i + 1, ], copy.copy(token), begin_time,
                                                 end_time)
                    results.append((text, token, token_int, time_stamp, enc_len_batch_total, lfr_factor))

                timestamp = time_stamp_lfr6_pl(us_alphas[i], us_cif_peak[i], copy.copy(token), begin_time, end_time)
                results.append((text, token, token_int, timestamp, enc_len_batch_total, lfr_factor))


        # assert check_return_type(results)
        return results
@@ -636,7 +633,8 @@
                text, token, token_int = result[0], result[1], result[2]
                time_stamp = None if len(result) < 4 else result[3]

                if use_timestamp and time_stamp is not None:

                if use_timestamp and time_stamp is not None: 
                    postprocessed_result = postprocess_utils.sentence_postprocess(token, time_stamp)
                else:
                    postprocessed_result = postprocess_utils.sentence_postprocess(token)

 funasr/export/export_model.py

@@ -44,6 +44,7 @@
            model,
            self.export_config,
        )
        model.eval()
        # self._export_onnx(model, verbose, export_dir)
        if self.onnx:
            self._export_onnx(model, verbose, export_dir)

 funasr/runtime/python/onnxruntime/paraformer/rapid_paraformer/paraformer_onnx.py

@@ -41,8 +41,8 @@
        self.ort_infer = OrtInferSession(model_file, device_id)
        self.batch_size = batch_size

    def __call__(self, wav_content: Union[str, np.ndarray, List[str]], fs: int = None) -> List:
        waveform_list = self.load_data(wav_content, fs)
    def __call__(self, wav_content: Union[str, np.ndarray, List[str]], **kwargs) -> List:
        waveform_list = self.load_data(wav_content, self.frontend.opts.samp_freq)
        waveform_nums = len(waveform_list)

        asr_res = []

 funasr/utils/timestamp_tools.py

@@ -4,88 +4,6 @@
import numpy as np
from typing import Any, List, Tuple, Union

def cut_interval(alphas: torch.Tensor, start: int, end: int, tail: bool):
    if not tail:
        if end == start + 1:
            cut = (end + start) / 2.0
        else:
            alpha = alphas[start+1: end].tolist()
            reverse_steps = 1
            for reverse_alpha in alpha[::-1]:
                if reverse_alpha > 0.35:
                    reverse_steps += 1
                else:
                    break
            cut = end - reverse_steps
    else:
        if end != len(alphas) - 1:
            cut = end + 1
        else:
            cut = start + 1
    return float(cut)

def time_stamp_lfr6(alphas: torch.Tensor, speech_lengths: torch.Tensor, raw_text: List[str], begin: int = 0, end: int = None):
    time_stamp_list = []
    alphas = alphas[0]
    text = copy.deepcopy(raw_text)
    if end is None:
        time = speech_lengths * 60 / 1000
        sacle_rate = (time / speech_lengths[0]).tolist()
    else:
        time = (end - begin) / 1000
        sacle_rate = (time / speech_lengths[0]).tolist()

    predictor = (alphas > 0.5).int()
    fire_places = torch.nonzero(predictor == 1).squeeze(1).tolist()
    
    cuts = []
    npeak = int(predictor.sum())
    nchar = len(raw_text)
    if npeak - 1 == nchar:
        fire_places = torch.where((alphas > 0.5) == 1)[0].tolist()
        for i in range(len(fire_places)):
            if fire_places[i] < len(alphas) - 1:
                if 0.05 < alphas[fire_places[i]+1] < 0.5:
                    fire_places[i] += 1
    elif npeak < nchar:
        lost_num = nchar - npeak
        lost_fire = speech_lengths[0].tolist() - fire_places[-1]
        interval_distance = lost_fire // (lost_num + 1)
        for i in range(1, lost_num + 1):
            fire_places.append(fire_places[-1] + interval_distance)
    elif npeak - 1 > nchar:
        redundance_num = npeak - 1 - nchar
        for i in range(redundance_num):
            fire_places.pop() 

    cuts.append(0)
    start_sil = True
    if start_sil:
        text.insert(0, '<sil>')

    for i in range(len(fire_places)-1):
        cuts.append(cut_interval(alphas, fire_places[i], fire_places[i+1], tail=(i==len(fire_places)-2)))

    for i in range(2, len(fire_places)-2):
        if fire_places[i-2] == fire_places[i-1] - 1 and fire_places[i-1] != fire_places[i] - 1:
            cuts[i-1] += 1

    if cuts[-1] != len(alphas) - 1:
        text.append('<sil>')
        cuts.append(speech_lengths[0].tolist())
    cuts.insert(-1, (cuts[-1] + cuts[-2]) * 0.5)
    sec_fire_places = np.array(cuts) * sacle_rate
    for i in range(1, len(sec_fire_places) - 1):
        start, end = sec_fire_places[i], sec_fire_places[i+1]
        if i == len(sec_fire_places) - 2:
            end = time
        time_stamp_list.append([int(round(start, 2) * 1000) + begin, int(round(end, 2) * 1000) + begin])
        text = text[1:]
    if npeak - 1 == nchar or npeak > nchar:
        return time_stamp_list[:-1]
    else:
        return time_stamp_list

def time_stamp_lfr6_pl(us_alphas, us_cif_peak, char_list, begin_time=0.0, end_time=None):
    START_END_THRESHOLD = 5
    TIME_RATE = 10.0 * 6 / 1000 / 3  #  3 times upsampled