FunASR/egs_modelscope/asr/TEMPLATE

Speech Recognition

Note

: The modelscope pipeline supports all the models in model zoo to inference and finetine. Here we take the typic models as examples to demonstrate the usage.

Inference

Quick start

Paraformer Model

from modelscope.pipelines import pipeline
from modelscope.utils.constant import Tasks

inference_pipeline = pipeline(
    task=Tasks.auto_speech_recognition,
    model='damo/speech_paraformer-large_asr_nat-zh-cn-16k-common-vocab8404-pytorch',
)

rec_result = inference_pipeline(audio_in='https://isv-data.oss-cn-hangzhou.aliyuncs.com/ics/MaaS/ASR/test_audio/asr_example_zh.wav')
print(rec_result)

Paraformer-online Model

Streaming Decoding

inference_pipeline = pipeline(
    task=Tasks.auto_speech_recognition,
    model='damo/speech_paraformer-large_asr_nat-zh-cn-16k-common-vocab8404-online',
    model_revision='v1.0.6',
    update_model=False,
    mode='paraformer_streaming'
    )
import soundfile
speech, sample_rate = soundfile.read("example/asr_example.wav")

chunk_size = [5, 10, 5] #[5, 10, 5] 600ms, [8, 8, 4] 480ms
param_dict = {"cache": dict(), "is_final": False, "chunk_size": chunk_size}
chunk_stride = chunk_size[1] * 960 # 600ms、480ms
# first chunk, 600ms
speech_chunk = speech[0:chunk_stride] 
rec_result = inference_pipeline(audio_in=speech_chunk, param_dict=param_dict)
print(rec_result)
# next chunk, 600ms
speech_chunk = speech[chunk_stride:chunk_stride+chunk_stride]
rec_result = inference_pipeline(audio_in=speech_chunk, param_dict=param_dict)
print(rec_result)

Fake Streaming Decoding

from modelscope.pipelines import pipeline
from modelscope.utils.constant import Tasks

inference_pipeline = pipeline(
    task=Tasks.auto_speech_recognition,
    model='damo/speech_paraformer-large_asr_nat-zh-cn-16k-common-vocab8404-online',
    model_revision='v1.0.6',
    update_model=False,
    mode="paraformer_fake_streaming"
)
audio_in='https://isv-data.oss-cn-hangzhou.aliyuncs.com/ics/MaaS/ASR/test_audio/asr_example_zh.wav'
rec_result = inference_pipeline(audio_in=audio_in)
print(rec_result)

Full code of demo, please ref to demo

UniASR Model

There are three decoding mode for UniASR model(fast、normal、offline), for more model details, please refer to docs

decoding_model = "fast" # "fast"、"normal"、"offline"
inference_pipeline = pipeline(
    task=Tasks.auto_speech_recognition,
    model='damo/speech_UniASR_asr_2pass-minnan-16k-common-vocab3825',
    param_dict={"decoding_model": decoding_model})

rec_result = inference_pipeline(audio_in='https://isv-data.oss-cn-hangzhou.aliyuncs.com/ics/MaaS/ASR/test_audio/asr_example_zh.wav')
print(rec_result)

The decoding mode of fast and normal is fake streaming, which could be used for evaluating of recognition accuracy. Full code of demo, please ref to demo

RNN-T-online model

Undo

MFCCA Model

For more model details, please refer to docs

from modelscope.pipelines import pipeline
from modelscope.utils.constant import Tasks

inference_pipeline = pipeline(
    task=Tasks.auto_speech_recognition,
    model='NPU-ASLP/speech_mfcca_asr-zh-cn-16k-alimeeting-vocab4950',
    model_revision='v3.0.0'
)

rec_result = inference_pipeline(audio_in='https://isv-data.oss-cn-hangzhou.aliyuncs.com/ics/MaaS/ASR/test_audio/asr_example_zh.wav')
print(rec_result)

API-reference

Define pipeline

• task: Tasks.auto_speech_recognition
• model: model name in model zoo, or model path in local disk
• ngpu: 1 (Default), decoding on GPU. If ngpu=0, decoding on CPU
• ncpu: 1 (Default), sets the number of threads used for intraop parallelism on CPU
• output_dir: None (Default), the output path of results if set
• batch_size: 1 (Default), batch size when decoding

Infer pipeline

• audio_in: the input to decode, which could be:
  • wav_path, e.g.: asr_example.wav,
  • pcm_path, e.g.: asr_example.pcm,
  • audio bytes stream, e.g.: bytes data from a microphone
  • audio sample point，e.g.: audio, rate = soundfile.read("asr_example_zh.wav"), the dtype is numpy.ndarray or torch.Tensor
  • wav.scp, kaldi style wav list (wav_id \t wav_path), e.g.:

  asr_example1  ./audios/asr_example1.wav
  asr_example2  ./audios/asr_example2.wav
  

  In this case of wav.scp input, output_dir must be set to save the output results
• audio_fs: audio sampling rate, only set when audio_in is pcm audio
• output_dir: None (Default), the output path of results if set

Inference with multi-thread CPUs or multi GPUs

FunASR also offer recipes egs_modelscope/asr/TEMPLATE/infer.sh to decode with multi-thread CPUs, or multi GPUs.

Settings of infer.sh

• model: model name in model zoo, or model path in local disk
• data_dir: the dataset dir needs to include wav.scp. If ${data_dir}/text is also exists, CER will be computed
• output_dir: output dir of the recognition results
• batch_size: 64 (Default), batch size of inference on gpu
• gpu_inference: true (Default), whether to perform gpu decoding, set false for CPU inference
• gpuid_list: 0,1 (Default), which gpu_ids are used to infer
• njob: only used for CPU inference (gpu_inference=false), 64 (Default), the number of jobs for CPU decoding
• checkpoint_dir: only used for infer finetuned models, the path dir of finetuned models
• checkpoint_name: only used for infer finetuned models, valid.cer_ctc.ave.pb (Default), which checkpoint is used to infer
• decoding_mode: normal (Default), decoding mode for UniASR model(fast、normal、offline)
• hotword_txt: None (Default), hotword file for contextual paraformer model(the hotword file name ends with .txt")

Decode with multi GPUs:

    bash infer.sh \
    --model "damo/speech_paraformer-large_asr_nat-zh-cn-16k-common-vocab8404-pytorch" \
    --data_dir "./data/test" \
    --output_dir "./results" \
    --batch_size 64 \
    --gpu_inference true \
    --gpuid_list "0,1"

Decode with multi-thread CPUs:

    bash infer.sh \
    --model "damo/speech_paraformer-large_asr_nat-zh-cn-16k-common-vocab8404-pytorch" \
    --data_dir "./data/test" \
    --output_dir "./results" \
    --gpu_inference false \
    --njob 64

Results

The decoding results can be found in $output_dir/1best_recog/text.cer, which includes recognition results of each sample and the CER metric of the whole test set.

If you decode the SpeechIO test sets, you can use textnorm with stage=3, and DETAILS.txt, RESULTS.txt record the results and CER after text normalization.

Finetune with pipeline

Quick start

finetune.py

import os
from modelscope.metainfo import Trainers
from modelscope.trainers import build_trainer
from modelscope.msdatasets.audio.asr_dataset import ASRDataset

def modelscope_finetune(params):
    if not os.path.exists(params.output_dir):
        os.makedirs(params.output_dir, exist_ok=True)
    # dataset split ["train", "validation"]
    ds_dict = ASRDataset.load(params.data_path, namespace='speech_asr')
    kwargs = dict(
        model=params.model,
        data_dir=ds_dict,
        dataset_type=params.dataset_type,
        work_dir=params.output_dir,
        batch_bins=params.batch_bins,
        max_epoch=params.max_epoch,
        lr=params.lr)
    trainer = build_trainer(Trainers.speech_asr_trainer, default_args=kwargs)
    trainer.train()


if __name__ == '__main__':
    from funasr.utils.modelscope_param import modelscope_args
    params = modelscope_args(model="damo/speech_paraformer-large_asr_nat-zh-cn-16k-common-vocab8404-pytorch")
    params.output_dir = "./checkpoint"                      # 模型保存路径
    params.data_path = "speech_asr_aishell1_trainsets"      # 数据路径，可以为modelscope中已上传数据，也可以是本地数据
    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 = 2000                                # 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 = 50                                   # 最大训练轮数
    params.lr = 0.00005                                     # 设置学习率
    
    modelscope_finetune(params)

python finetune.py &> log.txt &

Finetune with your data

• Modify finetune training related parameters in finetune.py

  • output_dir: result dir
  • data_dir: the dataset dir needs to include files: train/wav.scp, train/text; validation/wav.scp, validation/text
  • dataset_type: for dataset larger than 1000 hours, set as large, otherwise set as small
  • batch_bins: batch size. For dataset_type is small, batch_bins indicates the feature frames. For dataset_type is large, batch_bins indicates the duration in ms
  • max_epoch: number of training epoch
  • lr: learning rate

• Training data formats：

cat ./example_data/text
BAC009S0002W0122 而 对 楼 市 成 交 抑 制 作 用 最 大 的 限 购
BAC009S0002W0123 也 成 为 地 方 政 府 的 眼 中 钉
english_example_1 hello world
english_example_2 go swim 去 游 泳

cat ./example_data/wav.scp
BAC009S0002W0122 /mnt/data/wav/train/S0002/BAC009S0002W0122.wav
BAC009S0002W0123 /mnt/data/wav/train/S0002/BAC009S0002W0123.wav
english_example_1 /mnt/data/wav/train/S0002/english_example_1.wav
english_example_2 /mnt/data/wav/train/S0002/english_example_2.wav

• Then you can run the pipeline to finetune with:

python finetune.py

If you want finetune with multi-GPUs, you could:

CUDA_VISIBLE_DEVICES=1,2 python -m torch.distributed.launch --nproc_per_node 2 finetune.py > log.txt 2>&1

Inference with your finetuned model

• Setting parameters in egs_modelscope/asr/TEMPLATE/infer.sh is the same with docs, model is the model name from modelscope, which you finetuned.

• Decode with multi GPUs:

    bash infer.sh \
    --model "damo/speech_paraformer-large_asr_nat-zh-cn-16k-common-vocab8404-pytorch" \
    --data_dir "./data/test" \
    --output_dir "./results" \
    --batch_size 64 \
    --gpu_inference true \
    --gpuid_list "0,1" \
    --checkpoint_dir "./checkpoint" \
    --checkpoint_name "valid.cer_ctc.ave.pb"

• Decode with multi-thread CPUs:

    bash infer.sh \
    --model "damo/speech_paraformer-large_asr_nat-zh-cn-16k-common-vocab8404-pytorch" \
    --data_dir "./data/test" \
    --output_dir "./results" \
    --gpu_inference false \
    --njob 64 \
    --checkpoint_dir "./checkpoint" \
    --checkpoint_name "valid.cer_ctc.ave.pb"