| | |
| | | # Voice Activity Detection |
| | | |
| | | ## Inference with pipeline |
| | | > **Note**: |
| | | > The modelscope pipeline supports all the models in [model zoo](https://alibaba-damo-academy.github.io/FunASR/en/modelscope_models.html#pretrained-models-on-modelscope) to inference and finetine. Here we take model of FSMN-VAD as example to demonstrate the usage. |
| | | |
| | | ## Inference |
| | | |
| | | ### Quick start |
| | | #### [FSMN-VAD model](https://modelscope.cn/models/damo/speech_fsmn_vad_zh-cn-16k-common-pytorch/summary) |
| | | ```python |
| | | from modelscope.pipelines import pipeline |
| | | from modelscope.utils.constant import Tasks |
| | | |
| | | ### Inference with you data |
| | | inference_pipeline = pipeline( |
| | | task=Tasks.voice_activity_detection, |
| | | model='damo/speech_fsmn_vad_zh-cn-16k-common-pytorch', |
| | | ) |
| | | |
| | | ### Inference with multi-threads on CPU |
| | | segments_result = inference_pipeline(audio_in='https://isv-data.oss-cn-hangzhou.aliyuncs.com/ics/MaaS/ASR/test_audio/vad_example.wav') |
| | | print(segments_result) |
| | | ``` |
| | | #### [FSMN-VAD-online model](https://modelscope.cn/models/damo/speech_fsmn_vad_zh-cn-16k-common-pytorch/summary) |
| | | ```python |
| | | inference_pipeline = pipeline( |
| | | task=Tasks.auto_speech_recognition, |
| | | model='damo/speech_fsmn_vad_zh-cn-16k-common-pytorch', |
| | | ) |
| | | import soundfile |
| | | speech, sample_rate = soundfile.read("example/asr_example.wav") |
| | | |
| | | ### Inference with multi GPU |
| | | param_dict = {"in_cache": dict(), "is_final": False} |
| | | chunk_stride = 1600# 100ms |
| | | # first chunk, 100ms |
| | | speech_chunk = speech[0:chunk_stride] |
| | | rec_result = inference_pipeline(audio_in=speech_chunk, param_dict=param_dict) |
| | | print(rec_result) |
| | | # next chunk, 480ms |
| | | speech_chunk = speech[chunk_stride:chunk_stride+chunk_stride] |
| | | rec_result = inference_pipeline(audio_in=speech_chunk, param_dict=param_dict) |
| | | print(rec_result) |
| | | ``` |
| | | Full code of demo, please ref to [demo](https://github.com/alibaba-damo-academy/FunASR/discussions/236) |
| | | |
| | | |
| | | #### API-reference |
| | | ##### define pipeline |
| | | - `task`: `Tasks.auto_speech_recognition` |
| | | - `model`: model name in [model zoo](https://alibaba-damo-academy.github.io/FunASR/en/modelscope_models.html#pretrained-models-on-modelscope), or model path in local disk |
| | | - `ngpu`: 1 (Defalut), decoding on GPU. If ngpu=0, decoding on CPU |
| | | - `ncpu`: 1 (Defalut), sets the number of threads used for intraop parallelism on CPU |
| | | - `output_dir`: None (Defalut), the output path of results if set |
| | | - `batch_size`: 1 (Defalut), 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.`: |
| | | ```cat wav.scp |
| | | 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 (Defalut), the output path of results if set |
| | | |
| | | ### Inference with multi-thread CPUs or multi GPUs |
| | | FunASR also offer recipes [run.sh](https://github.com/alibaba-damo-academy/FunASR/blob/main/egs_modelscope/asr/paraformer/speech_paraformer-large_asr_nat-zh-cn-16k-common-vocab8404-pytorch/infer.sh) to decode with multi-thread CPUs, or multi GPUs. |
| | | |
| | | - Setting parameters in `infer.sh` |
| | | - <strong>model:</strong> # model name on ModelScope |
| | | - <strong>data_dir:</strong> # the dataset dir needs to include `${data_dir}/wav.scp`. If `${data_dir}/text` is also exists, CER will be computed |
| | | - <strong>output_dir:</strong> # result dir |
| | | - <strong>batch_size:</strong> # batchsize of inference |
| | | - <strong>gpu_inference:</strong> # whether to perform gpu decoding, set false for cpu decoding |
| | | - <strong>gpuid_list:</strong> # set gpus, e.g., gpuid_list="0,1" |
| | | - <strong>njob:</strong> # the number of jobs for CPU decoding, if `gpu_inference`=false, use CPU decoding, please set `njob` |
| | | |
| | | - Decode with multi GPUs: |
| | | ```shell |
| | | bash infer.sh \ |
| | | --model "damo/speech_fsmn_vad_zh-cn-16k-common-pytorch" \ |
| | | --data_dir "./data/test" \ |
| | | --output_dir "./results" \ |
| | | --gpu_inference true \ |
| | | --gpuid_list "0,1" |
| | | ``` |
| | | - Decode with multi-thread CPUs: |
| | | ```shell |
| | | bash infer.sh \ |
| | | --model "damo/speech_fsmn_vad_zh-cn-16k-common-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 |
| | | |