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([简体中文](./README_zh.md)|English)
FunASR has open-sourced a large number of pre-trained models on industrial data. You are free to use, copy, modify, and share FunASR models under the [Model License Agreement](https://github.com/alibaba-damo-academy/FunASR/blob/main/MODEL_LICENSE). Below, we list some representative models. For a comprehensive list, please refer to our [Model Zoo](https://github.com/alibaba-damo-academy/FunASR/tree/main/model_zoo).
<div align="center">
<h4>
<a href="#Inference"> Model Inference </a>
|<a href="#Training"> Model Training and Testing </a>
|<a href="#Export"> Model Export and Testing </a>
</h4>
</div>
<a name="Inference"></a>
## Model Inference
### Quick Start
For command-line invocation:
```shell
funasr ++model=paraformer-zh ++vad_model="fsmn-vad" ++punc_model="ct-punc" ++input=asr_example_zh.wav
```
For python code invocation (recommended):
```python
from funasr import AutoModel
model = AutoModel(model="paraformer-zh")
res = model.generate(input="https://isv-data.oss-cn-hangzhou.aliyuncs.com/ics/MaaS/ASR/test_audio/vad_example.wav")
print(res)
```
### API Description
#### AutoModel Definition
```python
model = AutoModel(model=[str], device=[str], ncpu=[int], output_dir=[str], batch_size=[int], hub=[str], **kwargs)
```
- `model`(str): model name in the [Model Repository](https://github.com/alibaba-damo-academy/FunASR/tree/main/model_zoo), or a model path on local disk.
- `device`(str): `cuda:0` (default gpu0) for using GPU for inference, specify `cpu` for using CPU.
- `ncpu`(int): `4` (default), sets the number of threads for CPU internal operations.
- `output_dir`(str): `None` (default), set this to specify the output path for the results.
- `batch_size`(int): `1` (default), the number of samples per batch during decoding.
- `hub`(str):`ms` (default) to download models from ModelScope. Use `hf` to download models from Hugging Face.
- `**kwargs`(dict): Any parameters found in config.yaml can be directly specified here, for instance, the maximum segmentation length in the vad model max_single_segment_time=6000 (milliseconds).
#### AutoModel Inference
```python
res = model.generate(input=[str], output_dir=[str])
```
- `input`: The input to be decoded, which could be:
- A wav file path, e.g., asr_example.wav
- A pcm file path, e.g., asr_example.pcm, in this case, specify the audio sampling rate fs (default is 16000)
- An audio byte stream, e.g., byte data from a microphone
- A wav.scp, a Kaldi-style wav list (wav_id \t wav_path), for example:
```text
asr_example1 ./audios/asr_example1.wav
asr_example2 ./audios/asr_example2.wav
```
When using wav.scp as input, you must set output_dir to save the output results.
- Audio samples, `e.g.`: `audio, rate = soundfile.read("asr_example_zh.wav")`, data type is numpy.ndarray. Supports batch inputs, type is list:
```[audio_sample1, audio_sample2, ..., audio_sampleN]```
- fbank input, supports batch grouping. Shape is [batch, frames, dim], type is torch.Tensor.
- `output_dir`: None (default), if set, specifies the output path for the results.
- `**kwargs`(dict): Inference parameters related to the model, for example,`beam_size=10`,`decoding_ctc_weight=0.1`.
### More Usage Introduction
#### Speech Recognition (Non-streaming)
```python
from funasr import AutoModel
# paraformer-zh is a multi-functional asr model
# use vad, punc, spk or not as you need
model = AutoModel(model="paraformer-zh",
vad_model="fsmn-vad",
vad_kwargs={"max_single_segment_time": 60000},
punc_model="ct-punc",
# spk_model="cam++"
)
wav_file = f"{model.model_path}/example/asr_example.wav"
res = model.generate(input=wav_file, batch_size_s=300, batch_size_threshold_s=60, hotword='魔搭')
print(res)
```
Notes:
- Typically, the input duration for models is limited to under 30 seconds. However, when combined with `vad_model`, support for audio input of any length is enabled, not limited to the paraformer model—any audio input model can be used.
- Parameters related to model can be directly specified in the definition of AutoModel; parameters related to `vad_model` can be set through `vad_kwargs`, which is a dict; similar parameters include `punc_kwargs` and `spk_kwargs`.
- `max_single_segment_time`: Denotes the maximum audio segmentation length for `vad_model`, measured in milliseconds (ms).
- `batch_size_s` represents the use of dynamic batching, where the total audio duration within a batch is measured in seconds (s).
- `batch_size_threshold_s`: Indicates that when the duration of an audio segment post-VAD segmentation exceeds the batch_size_threshold_s threshold, the batch size is set to 1, measured in seconds (s).
Recommendations:
When you input long audio and encounter Out Of Memory (OOM) issues, since memory usage tends to increase quadratically with audio length, consider the following three scenarios:
a) At the beginning of inference, memory usage primarily depends on `batch_size_s`. Appropriately reducing this value can decrease memory usage.
b) During the middle of inference, when encountering long audio segments cut by VAD and the total token count is less than `batch_size_s`, yet still facing OOM, you can appropriately reduce `batch_size_threshold_s`. If the threshold is exceeded, the batch size is forced to 1.
c) Towards the end of inference, if long audio segments cut by VAD have a total token count less than `batch_size_s` and exceed the `threshold` batch_size_threshold_s, forcing the batch size to 1 and still facing OOM, you may reduce `max_single_segment_time` to shorten the VAD audio segment length.
#### Speech Recognition (Streaming)
```python
from funasr import AutoModel
chunk_size = [0, 10, 5] #[0, 10, 5] 600ms, [0, 8, 4] 480ms
encoder_chunk_look_back = 4 #number of chunks to lookback for encoder self-attention
decoder_chunk_look_back = 1 #number of encoder chunks to lookback for decoder cross-attention
model = AutoModel(model="paraformer-zh-streaming")
import soundfile
import os
wav_file = os.path.join(model.model_path, "example/asr_example.wav")
speech, sample_rate = soundfile.read(wav_file)
chunk_stride = chunk_size[1] * 960 # 600ms
cache = {}
total_chunk_num = int(len((speech)-1)/chunk_stride+1)
for i in range(total_chunk_num):
speech_chunk = speech[i*chunk_stride:(i+1)*chunk_stride]
is_final = i == total_chunk_num - 1
res = model.generate(input=speech_chunk, cache=cache, is_final=is_final, chunk_size=chunk_size, encoder_chunk_look_back=encoder_chunk_look_back, decoder_chunk_look_back=decoder_chunk_look_back)
print(res)
```
Note: `chunk_size` is the configuration for streaming latency.` [0,10,5]` indicates that the real-time display granularity is `10*60=600ms`, and the lookahead information is `5*60=300ms`. Each inference input is `600ms` (sample points are `16000*0.6=960`), and the output is the corresponding text. For the last speech segment input, `is_final=True` needs to be set to force the output of the last word.
#### Voice Activity Detection (Non-Streaming)
```python
from funasr import AutoModel
model = AutoModel(model="fsmn-vad")
wav_file = f"{model.model_path}/example/vad_example.wav"
res = model.generate(input=wav_file)
print(res)
```
Note: The output format of the VAD model is: `[[beg1, end1], [beg2, end2], ..., [begN, endN]]`, where `begN/endN` indicates the starting/ending point of the `N-th` valid audio segment, measured in milliseconds.
#### Voice Activity Detection (Streaming)
```python
from funasr import AutoModel
chunk_size = 200 # ms
model = AutoModel(model="fsmn-vad")
import soundfile
wav_file = f"{model.model_path}/example/vad_example.wav"
speech, sample_rate = soundfile.read(wav_file)
chunk_stride = int(chunk_size * sample_rate / 1000)
cache = {}
total_chunk_num = int(len((speech)-1)/chunk_stride+1)
for i in range(total_chunk_num):
speech_chunk = speech[i*chunk_stride:(i+1)*chunk_stride]
is_final = i == total_chunk_num - 1
res = model.generate(input=speech_chunk, cache=cache, is_final=is_final, chunk_size=chunk_size)
if len(res[0]["value"]):
print(res)
```
Note: The output format for the streaming VAD model can be one of four scenarios:
- `[[beg1, end1], [beg2, end2], .., [begN, endN]]`:The same as the offline VAD output result mentioned above.
- `[[beg, -1]]`:Indicates that only a starting point has been detected.
- `[[-1, end]]`:Indicates that only an ending point has been detected.
- `[]`:Indicates that neither a starting point nor an ending point has been detected.
The output is measured in milliseconds and represents the absolute time from the starting point.
#### Punctuation Restoration
```python
from funasr import AutoModel
model = AutoModel(model="ct-punc")
res = model.generate(input="那今天的会就到这里吧 happy new year 明年见")
print(res)
```
#### Timestamp Prediction
```python
from funasr import AutoModel
model = AutoModel(model="fa-zh")
wav_file = f"{model.model_path}/example/asr_example.wav"
text_file = f"{model.model_path}/example/text.txt"
res = model.generate(input=(wav_file, text_file), data_type=("sound", "text"))
print(res)
```
More examples ref to [docs](https://github.com/alibaba-damo-academy/FunASR/tree/main/examples/industrial_data_pretraining)
<a name="Training"></a>
## Model Training and Testing
### Quick Start
Execute via command line (for quick testing, not recommended):
```shell
funasr-train ++model=paraformer-zh ++train_data_set_list=data/list/train.jsonl ++valid_data_set_list=data/list/val.jsonl ++output_dir="./outputs" &> log.txt &
```
Execute with Python code (supports multi-node and multi-GPU, recommended):
```shell
cd examples/industrial_data_pretraining/paraformer
bash finetune.sh
# "log_file: ./outputs/log.txt"
```
Full code ref to [finetune.sh](https://github.com/alibaba-damo-academy/FunASR/blob/main/examples/industrial_data_pretraining/paraformer/finetune.sh)
### Detailed Parameter Description:
```shell
funasr/bin/train.py \
++model="${model_name_or_model_dir}" \
++train_data_set_list="${train_data}" \
++valid_data_set_list="${val_data}" \
++dataset_conf.batch_size=20000 \
++dataset_conf.batch_type="token" \
++dataset_conf.num_workers=4 \
++train_conf.max_epoch=50 \
++train_conf.log_interval=1 \
++train_conf.resume=false \
++train_conf.validate_interval=2000 \
++train_conf.save_checkpoint_interval=2000 \
++train_conf.keep_nbest_models=20 \
++train_conf.avg_nbest_model=10 \
++optim_conf.lr=0.0002 \
++output_dir="${output_dir}" &> ${log_file}
```
- `model`(str): The name of the model (the ID in the model repository), at which point the script will automatically download the model to local storage; alternatively, the path to a model already downloaded locally.
- `train_data_set_list`(str): The path to the training data, typically in jsonl format, for specific details refer to [examples](https://github.com/alibaba-damo-academy/FunASR/blob/main/data/list).
- `valid_data_set_list`(str):The path to the validation data, also generally in jsonl format, for specific details refer to examples](https://github.com/alibaba-damo-academy/FunASR/blob/main/data/list).
- `dataset_conf.batch_type`(str):example (default), the type of batch. example means batches are formed with a fixed number of batch_size samples; length or token means dynamic batching, with total length or number of tokens of the batch equalling batch_size.
- `dataset_conf.batch_size`(int):Used in conjunction with batch_type. When batch_type=example, it represents the number of samples; when batch_type=length, it represents the length of the samples, measured in fbank frames (1 frame = 10 ms) or the number of text tokens.
- `train_conf.max_epoch`(int):The total number of epochs for training.
- `train_conf.log_interval`(int):The number of steps between logging.
- `train_conf.resume`(int):Whether to enable checkpoint resuming for training.
- `train_conf.validate_interval`(int):The interval in steps to run validation tests during training.
- `train_conf.save_checkpoint_interval`(int):The interval in steps for saving the model during training.
- `train_conf.keep_nbest_models`(int):The maximum number of model parameters to retain, sorted by validation set accuracy, from highest to lowest.
- `train_conf.avg_nbest_model`(int):Average over the top n models with the highest accuracy.
- `optim_conf.lr`(float):The learning rate.
- `output_dir`(str):The path for saving the model.
- `**kwargs`(dict): Any parameters in config.yaml can be specified directly here, for example, to filter out audio longer than 20s: dataset_conf.max_token_length=2000, measured in fbank frames (1 frame = 10 ms) or the number of text tokens.
#### Multi-GPU Training
##### Single-Machine Multi-GPU Training
```shell
export CUDA_VISIBLE_DEVICES="0,1"
gpu_num=$(echo $CUDA_VISIBLE_DEVICES | awk -F "," '{print NF}')
torchrun --nnodes 1 --nproc_per_node ${gpu_num} --master_port 12345 \
../../../funasr/bin/train.py ${train_args}
```
--nnodes represents the total number of participating nodes, while --nproc_per_node indicates the number of processes running on each node. --master_port indicates the port is 12345
##### Multi-Machine Multi-GPU Training
On the master node, assuming the IP is 192.168.1.1 and the port is 12345, and you're using 2 GPUs, you would run the following command:
```shell
export CUDA_VISIBLE_DEVICES="0,1"
gpu_num=$(echo $CUDA_VISIBLE_DEVICES | awk -F "," '{print NF}')
torchrun --nnodes 2 --node_rank 0 --nproc_per_node ${gpu_num} --master_addr 192.168.1.1 --master_port 12345 \
../../../funasr/bin/train.py ${train_args}
```
On the worker node (assuming the IP is 192.168.1.2), you need to ensure that the MASTER_ADDR and MASTER_PORT environment variables are set to match those of the master node, and then run the same command:
```shell
export CUDA_VISIBLE_DEVICES="0,1"
gpu_num=$(echo $CUDA_VISIBLE_DEVICES | awk -F "," '{print NF}')
torchrun --nnodes 2 --node_rank 1 --nproc_per_node ${gpu_num} --master_addr 192.168.1.1 --master_port 12345 \
../../../funasr/bin/train.py ${train_args}
```
--nnodes indicates the total number of nodes participating in the training, --node_rank represents the ID of the current node, and --nproc_per_node specifies the number of processes running on each node (usually corresponds to the number of GPUs).
#### Data prepare
`jsonl` ref to([demo](https://github.com/alibaba-damo-academy/FunASR/blob/main/data/list)).
The instruction scp2jsonl can be used to generate from wav.scp and text.txt. The preparation process for wav.scp and text.txt is as follows:
`train_text.txt`
```bash
ID0012W0013 当客户风险承受能力评估依据发生变化时
ID0012W0014 所有只要处理 data 不管你是做 machine learning 做 deep learning
ID0012W0015 he tried to think how it could be
```
`train_wav.scp`
```bash
BAC009S0764W0121 https://isv-data.oss-cn-hangzhou.aliyuncs.com/ics/MaaS/ASR/test_audio/BAC009S0764W0121.wav
BAC009S0916W0489 https://isv-data.oss-cn-hangzhou.aliyuncs.com/ics/MaaS/ASR/test_audio/BAC009S0916W0489.wav
ID0012W0015 https://isv-data.oss-cn-hangzhou.aliyuncs.com/ics/MaaS/ASR/test_audio/asr_example_cn_en.wav
```
`Command`
```shell
# 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="../../../data/list/train.jsonl"
```
(Optional, not required) If you need to parse from jsonl back to wav.scp and text.txt, you can use the following command:
```shell
# generate wav.scp and text.txt from train.jsonl and val.jsonl
jsonl2scp \
++scp_file_list='["../../../data/list/train_wav.scp", "../../../data/list/train_text.txt"]' \
++data_type_list='["source", "target"]' \
++jsonl_file_in="../../../data/list/train.jsonl"
```
#### Training log
##### log.txt
```shell
tail log.txt
[2024-03-21 15:55:52,137][root][INFO] - train, rank: 3, epoch: 0/50, step: 6990/1, total step: 6990, (loss_avg_rank: 0.327), (loss_avg_epoch: 0.409), (ppl_avg_epoch: 1.506), (acc_avg_epoch: 0.795), (lr: 1.165e-04), [('loss_att', 0.259), ('acc', 0.825), ('loss_pre', 0.04), ('loss', 0.299), ('batch_size', 40)], {'data_load': '0.000', 'forward_time': '0.315', 'backward_time': '0.555', 'optim_time': '0.076', 'total_time': '0.947'}, GPU, memory: usage: 3.830 GB, peak: 18.357 GB, cache: 20.910 GB, cache_peak: 20.910 GB
[2024-03-21 15:55:52,139][root][INFO] - train, rank: 1, epoch: 0/50, step: 6990/1, total step: 6990, (loss_avg_rank: 0.334), (loss_avg_epoch: 0.409), (ppl_avg_epoch: 1.506), (acc_avg_epoch: 0.795), (lr: 1.165e-04), [('loss_att', 0.285), ('acc', 0.823), ('loss_pre', 0.046), ('loss', 0.331), ('batch_size', 36)], {'data_load': '0.000', 'forward_time': '0.334', 'backward_time': '0.536', 'optim_time': '0.077', 'total_time': '0.948'}, GPU, memory: usage: 3.943 GB, peak: 18.291 GB, cache: 19.619 GB, cache_peak: 19.619 GB
```
- `rank`:gpu id。
- `epoch`,`step`,`total step`:the current epoch, step, and total steps.
- `loss_avg_rank`:the average loss across all GPUs for the current step.
- `loss/ppl/acc_avg_epoch`:the overall average loss/perplexity/accuracy for the current epoch, up to the current step count. The last step of the epoch when it ends represents the total average loss/perplexity/accuracy for that epoch; it is recommended to use the accuracy metric.
- `lr`:the learning rate for the current step.
- `[('loss_att', 0.259), ('acc', 0.825), ('loss_pre', 0.04), ('loss', 0.299), ('batch_size', 40)]`:the specific data for the current GPU ID.
- `total_time`:the total time taken for a single step.
- `GPU, memory`:the model-used/peak memory and the model+cache-used/peak memory.
##### tensorboard
```bash
tensorboard --logdir /xxxx/FunASR/examples/industrial_data_pretraining/paraformer/outputs/log/tensorboard
```
http://localhost:6006/
### 训练后模型测试
#### With `configuration.json` file
Assuming the training model path is: ./model_dir, if a configuration.json file has been generated in this directory, you only need to change the model name to the model path in the above model inference method.
For example, for shell inference:
```shell
python -m funasr.bin.inference ++model="./model_dir" ++input=="${input}" ++output_dir="${output_dir}"
```
Python inference
```python
from funasr import AutoModel
model = AutoModel(model="./model_dir")
res = model.generate(input=wav_file)
print(res)
```
#### Without `configuration.json` file
If there is no configuration.json in the model path, you need to manually specify the exact configuration file path and the model path.
```shell
python -m funasr.bin.inference \
--config-path "${local_path}" \
--config-name "${config}" \
++init_param="${init_param}" \
++tokenizer_conf.token_list="${tokens}" \
++frontend_conf.cmvn_file="${cmvn_file}" \
++input="${input}" \
++output_dir="${output_dir}" \
++device="${device}"
```
Parameter Introduction
- `config-path`:This is the path to the config.yaml saved during the experiment, which can be found in the experiment's output directory.
- `config-name`:The name of the configuration file, usually config.yaml. It supports both YAML and JSON formats, for example config.json.
- `init_param`:The model parameters that need to be tested, usually model.pt. You can choose a specific model file as needed.
- `tokenizer_conf.token_list`:The path to the vocabulary file, which is normally specified in config.yaml. There is no need to manually specify it again unless the path in config.yaml is incorrect, in which case the correct path must be manually specified here.
- `frontend_conf.cmvn_file`:The CMVN (Cepstral Mean and Variance Normalization) file used when extracting fbank features from WAV files, which is usually specified in config.yaml. There is no need to manually specify it again unless the path in config.yaml is incorrect, in which case the correct path must be manually specified here.
Other parameters are the same as mentioned above. A complete [example](https://github.com/alibaba-damo-academy/FunASR/blob/main/examples/industrial_data_pretraining/paraformer/infer_from_local.sh) can be found here.
<a name="Export"></a>
## Export ONNX
### Command-line usage
```shell
funasr-export ++model=paraformer ++quantize=false ++device=cpu
```
### Python
```python
from funasr import AutoModel
model = AutoModel(model="paraformer", device="cpu")
res = model.export(quantize=False)
```
### Test ONNX
```python
# pip3 install -U funasr-onnx
from funasr_onnx import Paraformer
model_dir = "damo/speech_paraformer-large_asr_nat-zh-cn-16k-common-vocab8404-pytorch"
model = Paraformer(model_dir, batch_size=1, quantize=True)
wav_path = ['~/.cache/modelscope/hub/damo/speech_paraformer-large_asr_nat-zh-cn-16k-common-vocab8404-pytorch/example/asr_example.wav']
result = model(wav_path)
print(result)
```
More examples ref to [demo](https://github.com/alibaba-damo-academy/FunASR/tree/main/runtime/python/onnxruntime)
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(简体中文|[English](./README.md))
FunASR开源了大量在工业数据上预训练模型,您可以在 [模型许可协议](https://github.com/alibaba-damo-academy/FunASR/blob/main/MODEL_LICENSE)下自由使用、复制、修改和分享FunASR模型,下面列举代表性的模型,更多模型请参考 [模型仓库](https://github.com/alibaba-damo-academy/FunASR/tree/main/model_zoo)。
<div align="center">
<h4>
<a href="#模型推理"> 模型推理 </a>
|<a href="#模型训练与测试"> 模型训练与测试 </a>
|<a href="#模型导出与测试"> 模型导出与测试 </a>
</h4>
</div>
<a name="模型推理"></a>
## 模型推理
### 快速使用
命令行方式调用:
```shell
funasr ++model=paraformer-zh ++vad_model="fsmn-vad" ++punc_model="ct-punc" ++input=asr_example_zh.wav
```
python代码调用(推荐)
```python
from funasr import AutoModel
model = AutoModel(model="paraformer-zh")
res = model.generate(input="https://isv-data.oss-cn-hangzhou.aliyuncs.com/ics/MaaS/ASR/test_audio/vad_example.wav")
print(res)
```
### 接口说明
#### AutoModel 定义
```python
model = AutoModel(model=[str], device=[str], ncpu=[int], output_dir=[str], batch_size=[int], hub=[str], **kwargs)
```
- `model`(str): [模型仓库](https://github.com/alibaba-damo-academy/FunASR/tree/main/model_zoo) 中的模型名称,或本地磁盘中的模型路径
- `device`(str): `cuda:0`(默认gpu0),使用 GPU 进行推理,指定。如果为`cpu`,则使用 CPU 进行推理
- `ncpu`(int): `4` (默认),设置用于 CPU 内部操作并行性的线程数
- `output_dir`(str): `None` (默认),如果设置,输出结果的输出路径
- `batch_size`(int): `1` (默认),解码时的批处理,样本个数
- `hub`(str):`ms`(默认),从modelscope下载模型。如果为`hf`,从huggingface下载模型。
- `**kwargs`(dict): 所有在`config.yaml`中参数,均可以直接在此处指定,例如,vad模型中最大切割长度 `max_single_segment_time=6000` (毫秒)。
#### AutoModel 推理
```python
res = model.generate(input=[str], output_dir=[str])
```
- `input`: 要解码的输入,可以是:
- wav文件路径, 例如: asr_example.wav
- pcm文件路径, 例如: asr_example.pcm,此时需要指定音频采样率fs(默认为16000)
- 音频字节数流,例如:麦克风的字节数数据
- wav.scp,kaldi 样式的 wav 列表 (`wav_id \t wav_path`), 例如:
```text
asr_example1 ./audios/asr_example1.wav
asr_example2 ./audios/asr_example2.wav
```
在这种输入 `wav.scp` 的情况下,必须设置 `output_dir` 以保存输出结果
- 音频采样点,例如:`audio, rate = soundfile.read("asr_example_zh.wav")`, 数据类型为 numpy.ndarray。支持batch输入,类型为list:
```[audio_sample1, audio_sample2, ..., audio_sampleN]```
- fbank输入,支持组batch。shape为[batch, frames, dim],类型为torch.Tensor,例如
- `output_dir`: None (默认),如果设置,输出结果的输出路径
- `**kwargs`(dict): 与模型相关的推理参数,例如,`beam_size=10`,`decoding_ctc_weight=0.1`。
### 更多用法介绍
#### 非实时语音识别
```python
from funasr import AutoModel
# paraformer-zh is a multi-functional asr model
# use vad, punc, spk or not as you need
model = AutoModel(model="paraformer-zh",
vad_model="fsmn-vad",
vad_kwargs={"max_single_segment_time": 60000},
punc_model="ct-punc",
# spk_model="cam++"
)
wav_file = f"{model.model_path}/example/asr_example.wav"
res = model.generate(input=wav_file, batch_size_s=300, batch_size_threshold_s=60, hotword='魔搭')
print(res)
```
注意:
- 通常模型输入限制时长30s以下,组合`vad_model`后,支持任意时长音频输入,不局限于paraformer模型,所有音频输入模型均可以。
- `model`相关的参数可以直接在`AutoModel`定义中直接指定;与`vad_model`相关参数可以通过`vad_kwargs`来指定,类型为dict;类似的有`punc_kwargs`,`spk_kwargs`;
- `max_single_segment_time`: 表示`vad_model`最大切割音频时长, 单位是毫秒ms.
- `batch_size_s` 表示采用动态batch,batch中总音频时长,单位为秒s。
- `batch_size_threshold_s`: 表示`vad_model`切割后音频片段时长超过 `batch_size_threshold_s`阈值时,将batch_size数设置为1, 单位为秒s.
建议:当您输入为长音频,遇到OOM问题时,因为显存占用与音频时长呈平方关系增加,分为3种情况:
- a)推理起始阶段,显存主要取决于`batch_size_s`,适当减小该值,可以减少显存占用;
- b)推理中间阶段,遇到VAD切割的长音频片段,总token数小于`batch_size_s`,仍然出现OOM,可以适当减小`batch_size_threshold_s`,超过阈值,强制batch为1;
- c)推理快结束阶段,遇到VAD切割的长音频片段,总token数小于`batch_size_s`,且超过阈值`batch_size_threshold_s`,强制batch为1,仍然出现OOM,可以适当减小`max_single_segment_time`,使得VAD切割音频时长变短。
#### 实时语音识别
```python
from funasr import AutoModel
chunk_size = [0, 10, 5] #[0, 10, 5] 600ms, [0, 8, 4] 480ms
encoder_chunk_look_back = 4 #number of chunks to lookback for encoder self-attention
decoder_chunk_look_back = 1 #number of encoder chunks to lookback for decoder cross-attention
model = AutoModel(model="paraformer-zh-streaming")
import soundfile
import os
wav_file = os.path.join(model.model_path, "example/asr_example.wav")
speech, sample_rate = soundfile.read(wav_file)
chunk_stride = chunk_size[1] * 960 # 600ms
cache = {}
total_chunk_num = int(len((speech)-1)/chunk_stride+1)
for i in range(total_chunk_num):
speech_chunk = speech[i*chunk_stride:(i+1)*chunk_stride]
is_final = i == total_chunk_num - 1
res = model.generate(input=speech_chunk, cache=cache, is_final=is_final, chunk_size=chunk_size, encoder_chunk_look_back=encoder_chunk_look_back, decoder_chunk_look_back=decoder_chunk_look_back)
print(res)
```
注:`chunk_size`为流式延时配置,`[0,10,5]`表示上屏实时出字粒度为`10*60=600ms`,未来信息为`5*60=300ms`。每次推理输入为`600ms`(采样点数为`16000*0.6=960`),输出为对应文字,最后一个语音片段输入需要设置`is_final=True`来强制输出最后一个字。
#### 语音端点检测(非实时)
```python
from funasr import AutoModel
model = AutoModel(model="fsmn-vad")
wav_file = f"{model.model_path}/example/vad_example.wav"
res = model.generate(input=wav_file)
print(res)
```
注:VAD模型输出格式为:`[[beg1, end1], [beg2, end2], .., [begN, endN]]`,其中`begN/endN`表示第`N`个有效音频片段的起始点/结束点,
单位为毫秒。
#### 语音端点检测(实时)
```python
from funasr import AutoModel
chunk_size = 200 # ms
model = AutoModel(model="fsmn-vad")
import soundfile
wav_file = f"{model.model_path}/example/vad_example.wav"
speech, sample_rate = soundfile.read(wav_file)
chunk_stride = int(chunk_size * sample_rate / 1000)
cache = {}
total_chunk_num = int(len((speech)-1)/chunk_stride+1)
for i in range(total_chunk_num):
speech_chunk = speech[i*chunk_stride:(i+1)*chunk_stride]
is_final = i == total_chunk_num - 1
res = model.generate(input=speech_chunk, cache=cache, is_final=is_final, chunk_size=chunk_size)
if len(res[0]["value"]):
print(res)
```
注:流式VAD模型输出格式为4种情况:
- `[[beg1, end1], [beg2, end2], .., [begN, endN]]`:同上离线VAD输出结果。
- `[[beg, -1]]`:表示只检测到起始点。
- `[[-1, end]]`:表示只检测到结束点。
- `[]`:表示既没有检测到起始点,也没有检测到结束点
输出结果单位为毫秒,从起始点开始的绝对时间。
#### 标点恢复
```python
from funasr import AutoModel
model = AutoModel(model="ct-punc")
res = model.generate(input="那今天的会就到这里吧 happy new year 明年见")
print(res)
```
#### 时间戳预测
```python
from funasr import AutoModel
model = AutoModel(model="fa-zh")
wav_file = f"{model.model_path}/example/asr_example.wav"
text_file = f"{model.model_path}/example/text.txt"
res = model.generate(input=(wav_file, text_file), data_type=("sound", "text"))
print(res)
```
更多([示例](https://github.com/alibaba-damo-academy/FunASR/tree/main/examples/industrial_data_pretraining))
<a name="核心功能"></a>
## 模型训练与测试
### 快速开始
命令行执行(用于快速测试,不推荐):
```shell
funasr-train ++model=paraformer-zh ++train_data_set_list=data/list/train.jsonl ++valid_data_set_list=data/list/val.jsonl ++output_dir="./outputs" &> log.txt &
```
python代码执行(可以多机多卡,推荐)
```shell
cd examples/industrial_data_pretraining/paraformer
bash finetune.sh
# "log_file: ./outputs/log.txt"
```
详细完整的脚本参考 [finetune.sh](https://github.com/alibaba-damo-academy/FunASR/blob/main/examples/industrial_data_pretraining/paraformer/finetune.sh)
### 详细参数介绍
```shell
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=20000 \
++dataset_conf.batch_type="token" \
++dataset_conf.num_workers=4 \
++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 \
++output_dir="${output_dir}" &> ${log_file}
```
- `model`(str):模型名字(模型仓库中的ID),此时脚本会自动下载模型到本读;或者本地已经下载好的模型路径。
- `train_data_set_list`(str):训练数据路径,默认为jsonl格式,具体参考([例子](https://github.com/alibaba-damo-academy/FunASR/blob/main/data/list))。
- `valid_data_set_list`(str):验证数据路径,默认为jsonl格式,具体参考([例子](https://github.com/alibaba-damo-academy/FunASR/blob/main/data/list))。
- `dataset_conf.batch_type`(str):`example`(默认),batch的类型。`example`表示按照固定数目batch_size个样本组batch;`length` or `token` 表示动态组batch,batch总长度或者token数为batch_size。
- `dataset_conf.batch_size`(int):与 `batch_type` 搭配使用,当 `batch_type=example` 时,表示样本个数;当 `batch_type=length` 时,表示样本中长度,单位为fbank帧数(1帧10ms)或者文字token个数。
- `train_conf.max_epoch`(int):`100`(默认),训练总epoch数。
- `train_conf.log_interval`(int):`50`(默认),打印日志间隔step数。
- `train_conf.resume`(int):`True`(默认),是否开启断点重训。
- `train_conf.validate_interval`(int):`5000`(默认),训练中做验证测试的间隔step数。
- `train_conf.save_checkpoint_interval`(int):`5000`(默认),训练中模型保存间隔step数。
- `train_conf.avg_keep_nbest_models_type`(str):`acc`(默认),保留nbest的标准为acc(越大越好)。`loss`表示,保留nbest的标准为loss(越小越好)。
- `train_conf.keep_nbest_models`(int):`500`(默认),保留最大多少个模型参数,配合 `avg_keep_nbest_models_type` 按照验证集 acc/loss 保留最佳的n个模型,其他删除,节约存储空间。
- `train_conf.avg_nbest_model`(int):`10`(默认),保留最大多少个模型参数,配合 `avg_keep_nbest_models_type` 按照验证集 acc/loss 对最佳的n个模型平均。
- `train_conf.accum_grad`(int):`1`(默认),梯度累积功能。
- `train_conf.grad_clip`(float):`10.0`(默认),梯度截断功能。
- `train_conf.use_fp16`(bool):`False`(默认),开启fp16训练,加快训练速度。
- `optim_conf.lr`(float):学习率。
- `output_dir`(str):模型保存路径。
- `**kwargs`(dict): 所有在`config.yaml`中参数,均可以直接在此处指定,例如,过滤20s以上长音频:`dataset_conf.max_token_length=2000`,单位为音频fbank帧数(1帧10ms)或者文字token个数。
#### 多gpu训练
##### 单机多gpu训练
```shell
export CUDA_VISIBLE_DEVICES="0,1"
gpu_num=$(echo $CUDA_VISIBLE_DEVICES | awk -F "," '{print NF}')
torchrun --nnodes 1 --nproc_per_node ${gpu_num} --master_port 12345 \
../../../funasr/bin/train.py ${train_args}
```
--nnodes 表示参与的节点总数,--nproc_per_node 表示每个节点上运行的进程数,--master_port 表示端口号
##### 多机多gpu训练
在主节点上,假设IP为192.168.1.1,端口为12345,使用的是2个GPU,则运行如下命令:
```shell
export CUDA_VISIBLE_DEVICES="0,1"
gpu_num=$(echo $CUDA_VISIBLE_DEVICES | awk -F "," '{print NF}')
torchrun --nnodes 2 --node_rank 0 --nproc_per_node ${gpu_num} --master_addr 192.168.1.1 --master_port 12345 \
../../../funasr/bin/train.py ${train_args}
```
在从节点上(假设IP为192.168.1.2),你需要确保MASTER_ADDR和MASTER_PORT环境变量与主节点设置的一致,并运行同样的命令:
```shell
export CUDA_VISIBLE_DEVICES="0,1"
gpu_num=$(echo $CUDA_VISIBLE_DEVICES | awk -F "," '{print NF}')
torchrun --nnodes 2 --node_rank 1 --nproc_per_node ${gpu_num} --master_addr 192.168.1.1 --master_port 12345 \
../../../funasr/bin/train.py ${train_args}
```
--nnodes 表示参与的节点总数,--node_rank 表示当前节点id,--nproc_per_node 表示每个节点上运行的进程数(通常为gpu个数),--master_port 表示端口号
#### 准备数据
`jsonl`格式可以参考([例子](https://github.com/alibaba-damo-academy/FunASR/blob/main/data/list))。
可以用指令 `scp2jsonl` 从wav.scp与text.txt生成。wav.scp与text.txt准备过程如下:
`train_text.txt`
左边为数据唯一ID,需与`train_wav.scp`中的`ID`一一对应
右边为音频文件标注文本,格式如下:
```bash
ID0012W0013 当客户风险承受能力评估依据发生变化时
ID0012W0014 所有只要处理 data 不管你是做 machine learning 做 deep learning
ID0012W0015 he tried to think how it could be
```
`train_wav.scp`
左边为数据唯一ID,需与`train_text.txt`中的`ID`一一对应
右边为音频文件的路径,格式如下
```bash
BAC009S0764W0121 https://isv-data.oss-cn-hangzhou.aliyuncs.com/ics/MaaS/ASR/test_audio/BAC009S0764W0121.wav
BAC009S0916W0489 https://isv-data.oss-cn-hangzhou.aliyuncs.com/ics/MaaS/ASR/test_audio/BAC009S0916W0489.wav
ID0012W0015 https://isv-data.oss-cn-hangzhou.aliyuncs.com/ics/MaaS/ASR/test_audio/asr_example_cn_en.wav
```
`生成指令`
```shell
# 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="../../../data/list/train.jsonl"
```
(可选,非必需)如果需要从jsonl解析成wav.scp与text.txt,可以使用指令:
```shell
# generate wav.scp and text.txt from train.jsonl and val.jsonl
jsonl2scp \
++scp_file_list='["../../../data/list/train_wav.scp", "../../../data/list/train_text.txt"]' \
++data_type_list='["source", "target"]' \
++jsonl_file_in="../../../data/list/train.jsonl"
```
#### 查看训练日志
##### 查看实验log
```shell
tail log.txt
[2024-03-21 15:55:52,137][root][INFO] - train, rank: 3, epoch: 0/50, step: 6990/1, total step: 6990, (loss_avg_rank: 0.327), (loss_avg_epoch: 0.409), (ppl_avg_epoch: 1.506), (acc_avg_epoch: 0.795), (lr: 1.165e-04), [('loss_att', 0.259), ('acc', 0.825), ('loss_pre', 0.04), ('loss', 0.299), ('batch_size', 40)], {'data_load': '0.000', 'forward_time': '0.315', 'backward_time': '0.555', 'optim_time': '0.076', 'total_time': '0.947'}, GPU, memory: usage: 3.830 GB, peak: 18.357 GB, cache: 20.910 GB, cache_peak: 20.910 GB
[2024-03-21 15:55:52,139][root][INFO] - train, rank: 1, epoch: 0/50, step: 6990/1, total step: 6990, (loss_avg_rank: 0.334), (loss_avg_epoch: 0.409), (ppl_avg_epoch: 1.506), (acc_avg_epoch: 0.795), (lr: 1.165e-04), [('loss_att', 0.285), ('acc', 0.823), ('loss_pre', 0.046), ('loss', 0.331), ('batch_size', 36)], {'data_load': '0.000', 'forward_time': '0.334', 'backward_time': '0.536', 'optim_time': '0.077', 'total_time': '0.948'}, GPU, memory: usage: 3.943 GB, peak: 18.291 GB, cache: 19.619 GB, cache_peak: 19.619 GB
```
指标解释:
- `rank`:表示gpu id。
- `epoch`,`step`,`total step`:表示当前epoch,step,总step。
- `loss_avg_rank`:表示当前step,所有gpu平均loss。
- `loss/ppl/acc_avg_epoch`:表示当前epoch周期,截止当前step数时,总平均loss/ppl/acc。epoch结束时的最后一个step表示epoch总平均loss/ppl/acc,推荐使用acc指标。
- `lr`:当前step的学习率。
- `[('loss_att', 0.259), ('acc', 0.825), ('loss_pre', 0.04), ('loss', 0.299), ('batch_size', 40)]`:表示当前gpu id的具体数据。
- `total_time`:表示单个step总耗时。
- `GPU, memory`:分别表示,模型使用/峰值显存,模型+缓存使用/峰值显存。
##### tensorboard可视化
```bash
tensorboard --logdir /xxxx/FunASR/examples/industrial_data_pretraining/paraformer/outputs/log/tensorboard
```
浏览器中打开:http://localhost:6006/
### 训练后模型测试
#### 有configuration.json
假定,训练模型路径为:./model_dir,如果改目录下有生成configuration.json,只需要将 [上述模型推理方法](https://github.com/alibaba-damo-academy/FunASR/blob/main/examples/README_zh.md#%E6%A8%A1%E5%9E%8B%E6%8E%A8%E7%90%86) 中模型名字修改为模型路径即可
例如:
从shell推理
```shell
python -m funasr.bin.inference ++model="./model_dir" ++input=="${input}" ++output_dir="${output_dir}"
```
从python推理
```python
from funasr import AutoModel
model = AutoModel(model="./model_dir")
res = model.generate(input=wav_file)
print(res)
```
#### 无configuration.json时
如果模型路径中无configuration.json时,需要手动指定具体配置文件路径与模型路径
```shell
python -m funasr.bin.inference \
--config-path "${local_path}" \
--config-name "${config}" \
++init_param="${init_param}" \
++tokenizer_conf.token_list="${tokens}" \
++frontend_conf.cmvn_file="${cmvn_file}" \
++input="${input}" \
++output_dir="${output_dir}" \
++device="${device}"
```
参数介绍
- `config-path`:为实验中保存的 `config.yaml`,可以从实验输出目录中查找。
- `config-name`:配置文件名,一般为 `config.yaml`,支持yaml格式与json格式,例如 `config.json`
- `init_param`:需要测试的模型参数,一般为`model.pt`,可以自己选择具体的模型文件
- `tokenizer_conf.token_list`:词表文件路径,一般在 `config.yaml` 有指定,无需再手动指定,当 `config.yaml` 中路径不正确时,需要在此处手动指定。
- `frontend_conf.cmvn_file`:wav提取fbank中用到的cmvn文件,一般在 `config.yaml` 有指定,无需再手动指定,当 `config.yaml` 中路径不正确时,需要在此处手动指定。
其他参数同上,完整 [示例](https://github.com/alibaba-damo-academy/FunASR/blob/main/examples/industrial_data_pretraining/paraformer/infer_from_local.sh)
<a name="模型导出与测试"></a>
## 模型导出与测试
### 从命令行导出
```shell
funasr-export ++model=paraformer ++quantize=false
```
### 从Python导出
```python
from funasr import AutoModel
model = AutoModel(model="paraformer")
res = model.export(quantize=False)
```
### 测试ONNX
```python
# pip3 install -U funasr-onnx
from funasr_onnx import Paraformer
model_dir = "damo/speech_paraformer-large_asr_nat-zh-cn-16k-common-vocab8404-pytorch"
model = Paraformer(model_dir, batch_size=1, quantize=True)
wav_path = ['~/.cache/modelscope/hub/damo/speech_paraformer-large_asr_nat-zh-cn-16k-common-vocab8404-pytorch/example/asr_example.wav']
result = model(wav_path)
print(result)
```
更多例子请参考 [样例](https://github.com/alibaba-damo-academy/FunASR/tree/main/runtime/python/onnxruntime)
+14
View File
@@ -0,0 +1,14 @@
# Branchformer Result
## Training Config
- Feature info: using raw speech, extracting 80 dims fbank online, global cmvn, speed perturb(0.9, 1.0, 1.1), specaugment
- Train info: lr 0.001, batch_size 10000, 4 gpu(Tesla V100), acc_grad 1, 180 epochs
- Train config: conf/train_asr_branchformer.yaml
- LM config: LM was not used
## Results (CER)
| testset | CER(%) |
|:-----------:|:-------:|
| dev | 4.15 |
| test | 4.51 |
@@ -0,0 +1,122 @@
# This is an example that demonstrates how to configure a model file.
# You can modify the configuration according to your own requirements.
# to print the register_table:
# from funasr.register import tables
# tables.print()
# network architecture
model: Branchformer
model_conf:
ctc_weight: 0.3
lsm_weight: 0.1 # label smoothing option
length_normalized_loss: false
# encoder
encoder: BranchformerEncoder
encoder_conf:
output_size: 256
use_attn: true
attention_heads: 4
attention_layer_type: rel_selfattn
pos_enc_layer_type: rel_pos
rel_pos_type: latest
use_cgmlp: true
cgmlp_linear_units: 2048
cgmlp_conv_kernel: 31
use_linear_after_conv: false
gate_activation: identity
merge_method: concat
cgmlp_weight: 0.5 # used only if merge_method is "fixed_ave"
attn_branch_drop_rate: 0.0 # used only if merge_method is "learned_ave"
num_blocks: 24
dropout_rate: 0.1
positional_dropout_rate: 0.1
attention_dropout_rate: 0.1
input_layer: conv2d
stochastic_depth_rate: 0.0
# decoder
decoder: TransformerDecoder
decoder_conf:
attention_heads: 4
linear_units: 2048
num_blocks: 6
dropout_rate: 0.1
positional_dropout_rate: 0.1
self_attention_dropout_rate: 0.
src_attention_dropout_rate: 0.
# frontend related
frontend: WavFrontend
frontend_conf:
fs: 16000
window: hamming
n_mels: 80
frame_length: 25
frame_shift: 10
dither: 0.0
lfr_m: 1
lfr_n: 1
specaug: SpecAug
specaug_conf:
apply_time_warp: true
time_warp_window: 5
time_warp_mode: bicubic
apply_freq_mask: true
freq_mask_width_range:
- 0
- 30
num_freq_mask: 2
apply_time_mask: true
time_mask_width_range:
- 0
- 40
num_time_mask: 2
train_conf:
accum_grad: 1
grad_clip: 5
max_epoch: 180
keep_nbest_models: 10
avg_keep_nbest_models_type: acc
log_interval: 50
optim: adam
optim_conf:
lr: 0.001
weight_decay: 0.000001
scheduler: warmuplr
scheduler_conf:
warmup_steps: 35000
dataset: AudioDataset
dataset_conf:
index_ds: IndexDSJsonl
batch_sampler: EspnetStyleBatchSampler
batch_type: length # example or length
batch_size: 10000 # if batch_type is example, batch_size is the numbers of samples; if length, batch_size is source_token_len+target_token_len;
max_token_length: 2048 # filter samples if source_token_len+target_token_len > max_token_length,
buffer_size: 1024
shuffle: True
num_workers: 4
preprocessor_speech: SpeechPreprocessSpeedPerturb
preprocessor_speech_conf:
speed_perturb: [0.9, 1.0, 1.1]
tokenizer: CharTokenizer
tokenizer_conf:
unk_symbol: <unk>
ctc_conf:
dropout_rate: 0.0
ctc_type: builtin
reduce: true
ignore_nan_grad: true
normalize: null
beam_size: 10
decoding_ctc_weight: 0.4
@@ -0,0 +1 @@
../paraformer/demo_infer.sh
@@ -0,0 +1 @@
../paraformer/demo_train_or_finetune.sh
+12
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@@ -0,0 +1,12 @@
python -m funasr.bin.inference \
--config-path="/mnt/workspace/FunASR/examples/aishell/paraformer/exp/baseline_paraformer_conformer_12e_6d_2048_256_zh_char_exp3" \
--config-name="config.yaml" \
++init_param="/mnt/workspace/FunASR/examples/aishell/paraformer/exp/baseline_paraformer_conformer_12e_6d_2048_256_zh_char_exp3/model.pt.ep38" \
++tokenizer_conf.token_list="/mnt/nfs/zhifu.gzf/data/AISHELL-1-feats/DATA/data/zh_token_list/char/tokens.txt" \
++frontend_conf.cmvn_file="/mnt/nfs/zhifu.gzf/data/AISHELL-1-feats/DATA/data/train/am.mvn" \
++input="/mnt/nfs/zhifu.gzf/data/AISHELL-1/data_aishell/wav/train/S0002/BAC009S0002W0122.wav" \
++output_dir="./outputs/debug" \
++device="cuda:0" \
@@ -0,0 +1,66 @@
#!/bin/bash
# Copyright 2017 Xingyu Na
# Apache 2.0
#. ./path.sh || exit 1;
if [ $# != 3 ]; then
echo "Usage: $0 <audio-path> <text-path> <output-path>"
echo " $0 /export/a05/xna/data/data_aishell/wav /export/a05/xna/data/data_aishell/transcript data"
exit 1;
fi
aishell_audio_dir=$1
aishell_text=$2/aishell_transcript_v0.8.txt
output_dir=$3
train_dir=$output_dir/data/local/train
dev_dir=$output_dir/data/local/dev
test_dir=$output_dir/data/local/test
tmp_dir=$output_dir/data/local/tmp
mkdir -p $train_dir
mkdir -p $dev_dir
mkdir -p $test_dir
mkdir -p $tmp_dir
# data directory check
if [ ! -d $aishell_audio_dir ] || [ ! -f $aishell_text ]; then
echo "Error: $0 requires two directory arguments"
exit 1;
fi
# find wav audio file for train, dev and test resp.
find $aishell_audio_dir -iname "*.wav" > $tmp_dir/wav.flist
n=`cat $tmp_dir/wav.flist | wc -l`
[ $n -ne 141925 ] && \
echo Warning: expected 141925 data data files, found $n
grep -i "wav/train" $tmp_dir/wav.flist > $train_dir/wav.flist || exit 1;
grep -i "wav/dev" $tmp_dir/wav.flist > $dev_dir/wav.flist || exit 1;
grep -i "wav/test" $tmp_dir/wav.flist > $test_dir/wav.flist || exit 1;
rm -r $tmp_dir
# Transcriptions preparation
for dir in $train_dir $dev_dir $test_dir; do
echo Preparing $dir transcriptions
sed -e 's/\.wav//' $dir/wav.flist | awk -F '/' '{print $NF}' > $dir/utt.list
paste -d' ' $dir/utt.list $dir/wav.flist > $dir/wav.scp_all
utils/filter_scp.pl -f 1 $dir/utt.list $aishell_text > $dir/transcripts.txt
awk '{print $1}' $dir/transcripts.txt > $dir/utt.list
utils/filter_scp.pl -f 1 $dir/utt.list $dir/wav.scp_all | sort -u > $dir/wav.scp
sort -u $dir/transcripts.txt > $dir/text
done
mkdir -p $output_dir/data/train $output_dir/data/dev $output_dir/data/test
for f in wav.scp text; do
cp $train_dir/$f $output_dir/data/train/$f || exit 1;
cp $dev_dir/$f $output_dir/data/dev/$f || exit 1;
cp $test_dir/$f $output_dir/data/test/$f || exit 1;
done
echo "$0: AISHELL data preparation succeeded"
exit 0;
@@ -0,0 +1,105 @@
#!/usr/bin/env bash
# Copyright 2014 Johns Hopkins University (author: Daniel Povey)
# 2017 Xingyu Na
# Apache 2.0
remove_archive=false
if [ "$1" == --remove-archive ]; then
remove_archive=true
shift
fi
if [ $# -ne 3 ]; then
echo "Usage: $0 [--remove-archive] <data-base> <url-base> <corpus-part>"
echo "e.g.: $0 /export/a05/xna/data www.openslr.org/resources/33 data_aishell"
echo "With --remove-archive it will remove the archive after successfully un-tarring it."
echo "<corpus-part> can be one of: data_aishell, resource_aishell."
fi
data=$1
url=$2
part=$3
if [ ! -d "$data" ]; then
echo "$0: no such directory $data"
exit 1;
fi
part_ok=false
list="data_aishell resource_aishell"
for x in $list; do
if [ "$part" == $x ]; then part_ok=true; fi
done
if ! $part_ok; then
echo "$0: expected <corpus-part> to be one of $list, but got '$part'"
exit 1;
fi
if [ -z "$url" ]; then
echo "$0: empty URL base."
exit 1;
fi
if [ -f $data/$part/.complete ]; then
echo "$0: data part $part was already successfully extracted, nothing to do."
exit 0;
fi
# sizes of the archive files in bytes.
sizes="15582913665 1246920"
if [ -f $data/$part.tgz ]; then
size=$(/bin/ls -l $data/$part.tgz | awk '{print $5}')
size_ok=false
for s in $sizes; do if [ $s == $size ]; then size_ok=true; fi; done
if ! $size_ok; then
echo "$0: removing existing file $data/$part.tgz because its size in bytes $size"
echo "does not equal the size of one of the archives."
rm $data/$part.tgz
else
echo "$data/$part.tgz exists and appears to be complete."
fi
fi
if [ ! -f $data/$part.tgz ]; then
if ! command -v wget >/dev/null; then
echo "$0: wget is not installed."
exit 1;
fi
full_url=$url/$part.tgz
echo "$0: downloading data from $full_url. This may take some time, please be patient."
cd $data || exit 1
if ! wget --no-check-certificate $full_url; then
echo "$0: error executing wget $full_url"
exit 1;
fi
fi
cd $data || exit 1
if ! tar -xvzf $part.tgz; then
echo "$0: error un-tarring archive $data/$part.tgz"
exit 1;
fi
touch $data/$part/.complete
if [ $part == "data_aishell" ]; then
cd $data/$part/wav || exit 1
for wav in ./*.tar.gz; do
echo "Extracting wav from $wav"
tar -zxf $wav && rm $wav
done
fi
echo "$0: Successfully downloaded and un-tarred $data/$part.tgz"
if $remove_archive; then
echo "$0: removing $data/$part.tgz file since --remove-archive option was supplied."
rm $data/$part.tgz
fi
exit 0;
+203
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@@ -0,0 +1,203 @@
#!/usr/bin/env bash
CUDA_VISIBLE_DEVICES="0,1,2,3"
# general configuration
feats_dir="../DATA" #feature output dictionary
exp_dir=`pwd`
lang=zh
token_type=char
stage=0
stop_stage=5
# feature configuration
nj=32
inference_device="cuda" #"cpu"
inference_checkpoint="model.pt.avg10"
inference_scp="wav.scp"
inference_batch_size=1
# data
raw_data=../raw_data
data_url=www.openslr.org/resources/33
# exp tag
tag="exp1"
workspace=`pwd`
master_port=12345
. utils/parse_options.sh || exit 1;
# Set bash to 'debug' mode, it will exit on :
# -e 'error', -u 'undefined variable', -o ... 'error in pipeline', -x 'print commands',
set -e
set -u
set -o pipefail
train_set=train
valid_set=dev
test_sets="dev test"
config=branchformer_12e_6d_2048_256.yaml
model_dir="baseline_$(basename "${config}" .yaml)_${lang}_${token_type}_${tag}"
if [ ${stage} -le -1 ] && [ ${stop_stage} -ge -1 ]; then
echo "stage -1: Data Download"
mkdir -p ${raw_data}
local/download_and_untar.sh ${raw_data} ${data_url} data_aishell
local/download_and_untar.sh ${raw_data} ${data_url} resource_aishell
fi
if [ ${stage} -le 0 ] && [ ${stop_stage} -ge 0 ]; then
echo "stage 0: Data preparation"
# Data preparation
local/aishell_data_prep.sh ${raw_data}/data_aishell/wav ${raw_data}/data_aishell/transcript ${feats_dir}
for x in train dev test; do
cp ${feats_dir}/data/${x}/text ${feats_dir}/data/${x}/text.org
paste -d " " <(cut -f 1 -d" " ${feats_dir}/data/${x}/text.org) <(cut -f 2- -d" " ${feats_dir}/data/${x}/text.org | tr -d " ") \
> ${feats_dir}/data/${x}/text
utils/text2token.py -n 1 -s 1 ${feats_dir}/data/${x}/text > ${feats_dir}/data/${x}/text.org
mv ${feats_dir}/data/${x}/text.org ${feats_dir}/data/${x}/text
# convert wav.scp text to jsonl
scp_file_list_arg="++scp_file_list='[\"${feats_dir}/data/${x}/wav.scp\",\"${feats_dir}/data/${x}/text\"]'"
python ../../../funasr/datasets/audio_datasets/scp2jsonl.py \
++data_type_list='["source", "target"]' \
++jsonl_file_out=${feats_dir}/data/${x}/audio_datasets.jsonl \
${scp_file_list_arg}
done
fi
if [ ${stage} -le 1 ] && [ ${stop_stage} -ge 1 ]; then
echo "stage 1: Feature and CMVN Generation"
python ../../../funasr/bin/compute_audio_cmvn.py \
--config-path "${workspace}/conf" \
--config-name "${config}" \
++train_data_set_list="${feats_dir}/data/${train_set}/audio_datasets.jsonl" \
++cmvn_file="${feats_dir}/data/${train_set}/cmvn.json" \
fi
token_list=${feats_dir}/data/${lang}_token_list/$token_type/tokens.txt
echo "dictionary: ${token_list}"
if [ ${stage} -le 2 ] && [ ${stop_stage} -ge 2 ]; then
echo "stage 2: Dictionary Preparation"
mkdir -p ${feats_dir}/data/${lang}_token_list/$token_type/
echo "make a dictionary"
echo "<blank>" > ${token_list}
echo "<s>" >> ${token_list}
echo "</s>" >> ${token_list}
utils/text2token.py -s 1 -n 1 --space "" ${feats_dir}/data/$train_set/text | cut -f 2- -d" " | tr " " "\n" \
| sort | uniq | grep -a -v -e '^\s*$' | awk '{print $0}' >> ${token_list}
echo "<unk>" >> ${token_list}
fi
# LM Training Stage
if [ ${stage} -le 3 ] && [ ${stop_stage} -ge 3 ]; then
echo "stage 3: LM Training"
fi
# ASR Training Stage
if [ ${stage} -le 4 ] && [ ${stop_stage} -ge 4 ]; then
echo "stage 4: ASR Training"
mkdir -p ${exp_dir}/exp/${model_dir}
current_time=$(date "+%Y-%m-%d_%H-%M")
log_file="${exp_dir}/exp/${model_dir}/train.log.txt.${current_time}"
echo "log_file: ${log_file}"
export CUDA_VISIBLE_DEVICES=$CUDA_VISIBLE_DEVICES
gpu_num=$(echo $CUDA_VISIBLE_DEVICES | awk -F "," '{print NF}')
torchrun \
--nnodes 1 \
--nproc_per_node ${gpu_num} \
--master_port ${master_port} \
../../../funasr/bin/train.py \
--config-path "${workspace}/conf" \
--config-name "${config}" \
++train_data_set_list="${feats_dir}/data/${train_set}/audio_datasets.jsonl" \
++valid_data_set_list="${feats_dir}/data/${valid_set}/audio_datasets.jsonl" \
++tokenizer_conf.token_list="${token_list}" \
++frontend_conf.cmvn_file="${feats_dir}/data/${train_set}/am.mvn" \
++output_dir="${exp_dir}/exp/${model_dir}" &> ${log_file}
fi
# Testing Stage
if [ ${stage} -le 5 ] && [ ${stop_stage} -ge 5 ]; then
echo "stage 5: Inference"
if [ ${inference_device} == "cuda" ]; then
nj=$(echo $CUDA_VISIBLE_DEVICES | awk -F "," '{print NF}')
else
inference_batch_size=1
CUDA_VISIBLE_DEVICES=""
for JOB in $(seq ${nj}); do
CUDA_VISIBLE_DEVICES=$CUDA_VISIBLE_DEVICES"-1,"
done
fi
for dset in ${test_sets}; do
inference_dir="${exp_dir}/exp/${model_dir}/inference-${inference_checkpoint}/${dset}"
_logdir="${inference_dir}/logdir"
echo "inference_dir: ${inference_dir}"
mkdir -p "${_logdir}"
data_dir="${feats_dir}/data/${dset}"
key_file=${data_dir}/${inference_scp}
split_scps=
for JOB in $(seq "${nj}"); do
split_scps+=" ${_logdir}/keys.${JOB}.scp"
done
utils/split_scp.pl "${key_file}" ${split_scps}
gpuid_list_array=(${CUDA_VISIBLE_DEVICES//,/ })
for JOB in $(seq ${nj}); do
{
id=$((JOB-1))
gpuid=${gpuid_list_array[$id]}
export CUDA_VISIBLE_DEVICES=${gpuid}
python ../../../funasr/bin/inference.py \
--config-path="${exp_dir}/exp/${model_dir}" \
--config-name="config.yaml" \
++init_param="${exp_dir}/exp/${model_dir}/${inference_checkpoint}" \
++tokenizer_conf.token_list="${token_list}" \
++frontend_conf.cmvn_file="${feats_dir}/data/${train_set}/am.mvn" \
++input="${_logdir}/keys.${JOB}.scp" \
++output_dir="${inference_dir}/${JOB}" \
++device="${inference_device}" \
++ncpu=1 \
++disable_log=true \
++batch_size="${inference_batch_size}" &> ${_logdir}/log.${JOB}.txt
}&
done
wait
mkdir -p ${inference_dir}/1best_recog
for f in token score text; do
if [ -f "${inference_dir}/${JOB}/1best_recog/${f}" ]; then
for JOB in $(seq "${nj}"); do
cat "${inference_dir}/${JOB}/1best_recog/${f}"
done | sort -k1 >"${inference_dir}/1best_recog/${f}"
fi
done
echo "Computing WER ..."
python utils/postprocess_text_zh.py ${inference_dir}/1best_recog/text ${inference_dir}/1best_recog/text.proc
python utils/postprocess_text_zh.py ${data_dir}/text ${inference_dir}/1best_recog/text.ref
python utils/compute_wer.py ${inference_dir}/1best_recog/text.ref ${inference_dir}/1best_recog/text.proc ${inference_dir}/1best_recog/text.cer
tail -n 3 ${inference_dir}/1best_recog/text.cer
done
fi
+1
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@@ -0,0 +1 @@
../paraformer/utils
+16
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@@ -0,0 +1,16 @@
# Conformer Result
## Training Config
- Feature info: using 80 dims fbank, global cmvn, speed perturb(0.9, 1.0, 1.1), specaugment
- Train info: lr 5e-4, batch_size 25000, 2 gpu(Tesla V100), acc_grad 1, 50 epochs
- Train config: conf/train_asr_transformer.yaml
- LM config: LM was not used
- Model size: 46M
## Results (CER)
| testset | CER(%) |
|:-----------:|:-------:|
| dev | 4.42 |
| test | 4.87 |
@@ -0,0 +1,110 @@
# This is an example that demonstrates how to configure a model file.
# You can modify the configuration according to your own requirements.
# to print the register_table:
# from funasr.register import tables
# tables.print()
# network architecture
model: Conformer
model_conf:
ctc_weight: 0.3
lsm_weight: 0.1 # label smoothing option
length_normalized_loss: false
# encoder
encoder: ConformerEncoder
encoder_conf:
output_size: 256 # dimension of attention
attention_heads: 4
linear_units: 2048 # the number of units of position-wise feed forward
num_blocks: 12 # the number of encoder blocks
dropout_rate: 0.1
positional_dropout_rate: 0.1
attention_dropout_rate: 0.0
input_layer: conv2d # encoder architecture type
normalize_before: true
pos_enc_layer_type: rel_pos
selfattention_layer_type: rel_selfattn
activation_type: swish
macaron_style: true
use_cnn_module: true
cnn_module_kernel: 15
# decoder
decoder: TransformerDecoder
decoder_conf:
attention_heads: 4
linear_units: 2048
num_blocks: 6
dropout_rate: 0.1
positional_dropout_rate: 0.1
self_attention_dropout_rate: 0.0
src_attention_dropout_rate: 0.0
# frontend related
frontend: WavFrontend
frontend_conf:
fs: 16000
window: hamming
n_mels: 80
frame_length: 25
frame_shift: 10
lfr_m: 1
lfr_n: 1
specaug: SpecAug
specaug_conf:
apply_time_warp: true
time_warp_window: 5
time_warp_mode: bicubic
apply_freq_mask: true
freq_mask_width_range:
- 0
- 30
num_freq_mask: 2
apply_time_mask: true
time_mask_width_range:
- 0
- 40
num_time_mask: 2
train_conf:
accum_grad: 1
grad_clip: 5
max_epoch: 150
keep_nbest_models: 10
log_interval: 50
optim: adam
optim_conf:
lr: 0.0005
scheduler: warmuplr
scheduler_conf:
warmup_steps: 30000
dataset: AudioDataset
dataset_conf:
index_ds: IndexDSJsonl
batch_sampler: EspnetStyleBatchSampler
batch_type: length # example or length
batch_size: 25000 # if batch_type is example, batch_size is the numbers of samples; if length, batch_size is source_token_len+target_token_len;
max_token_length: 2048 # filter samples if source_token_len+target_token_len > max_token_length,
buffer_size: 1024
shuffle: True
num_workers: 4
preprocessor_speech: SpeechPreprocessSpeedPerturb
preprocessor_speech_conf:
speed_perturb: [0.9, 1.0, 1.1]
tokenizer: CharTokenizer
tokenizer_conf:
unk_symbol: <unk>
ctc_conf:
dropout_rate: 0.0
ctc_type: builtin
reduce: true
ignore_nan_grad: true
normalize: null
@@ -0,0 +1,124 @@
# This is an example that demonstrates how to configure a model file.
# You can modify the configuration according to your own requirements.
# to print the register_table:
# from funasr.register import tables
# tables.print()
# network architecture
model: Conformer
model_conf:
ctc_weight: 0.3
lsm_weight: 0.1 # label smoothing option
length_normalized_loss: false
# encoder
encoder: ConformerEncoder
encoder_conf:
output_size: 256 # dimension of attention
attention_heads: 4
linear_units: 2048 # the number of units of position-wise feed forward
num_blocks: 12 # the number of encoder blocks
dropout_rate: 0.1
positional_dropout_rate: 0.1
attention_dropout_rate: 0.0
input_layer: conv2d # encoder architecture type
normalize_before: true
pos_enc_layer_type: rel_pos
selfattention_layer_type: rel_selfattn
activation_type: swish
macaron_style: true
use_cnn_module: true
cnn_module_kernel: 15
# decoder
decoder: TransformerRWKVDecoder
decoder_conf:
attention_heads: 4
linear_units: 2048
num_blocks: 6
dropout_rate: 0.1
positional_dropout_rate: 0.1
self_attention_dropout_rate: 0.0
src_attention_dropout_rate: 0.0
input_layer: embed
rwkv_cfg:
n_embd: 256
dropout: 0
head_size_a: 64
ctx_len: 512
dim_att: 256 #${model_conf.rwkv_cfg.n_embd}
dim_ffn: null
head_size_divisor: 4
n_layer: 6
pre_ffn: 0
ln0: false
ln1: false
init_rwkv: true
# frontend related
frontend: WavFrontend
frontend_conf:
fs: 16000
window: hamming
n_mels: 80
frame_length: 25
frame_shift: 10
lfr_m: 1
lfr_n: 1
specaug: SpecAug
specaug_conf:
apply_time_warp: true
time_warp_window: 5
time_warp_mode: bicubic
apply_freq_mask: true
freq_mask_width_range:
- 0
- 30
num_freq_mask: 2
apply_time_mask: true
time_mask_width_range:
- 0
- 40
num_time_mask: 2
train_conf:
accum_grad: 1
grad_clip: 5
max_epoch: 150
keep_nbest_models: 10
log_interval: 50
optim: adam
optim_conf:
lr: 0.0005
scheduler: warmuplr
scheduler_conf:
warmup_steps: 30000
dataset: AudioDataset
dataset_conf:
index_ds: IndexDSJsonl
batch_sampler: EspnetStyleBatchSampler
batch_type: length # example or length
batch_size: 25000 # if batch_type is example, batch_size is the numbers of samples; if length, batch_size is source_token_len+target_token_len;
max_token_length: 2048 # filter samples if source_token_len+target_token_len > max_token_length,
buffer_size: 1024
shuffle: True
num_workers: 4
preprocessor_speech: SpeechPreprocessSpeedPerturb
preprocessor_speech_conf:
speed_perturb: [0.9, 1.0, 1.1]
tokenizer: CharTokenizer
tokenizer_conf:
unk_symbol: <unk>
ctc_conf:
dropout_rate: 0.0
ctc_type: builtin
reduce: true
ignore_nan_grad: true
normalize: null
+1
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@@ -0,0 +1 @@
../paraformer/demo_infer.sh
@@ -0,0 +1 @@
../paraformer/demo_train_or_finetune.sh
@@ -0,0 +1,66 @@
#!/bin/bash
# Copyright 2017 Xingyu Na
# Apache 2.0
#. ./path.sh || exit 1;
if [ $# != 3 ]; then
echo "Usage: $0 <audio-path> <text-path> <output-path>"
echo " $0 /export/a05/xna/data/data_aishell/wav /export/a05/xna/data/data_aishell/transcript data"
exit 1;
fi
aishell_audio_dir=$1
aishell_text=$2/aishell_transcript_v0.8.txt
output_dir=$3
train_dir=$output_dir/data/local/train
dev_dir=$output_dir/data/local/dev
test_dir=$output_dir/data/local/test
tmp_dir=$output_dir/data/local/tmp
mkdir -p $train_dir
mkdir -p $dev_dir
mkdir -p $test_dir
mkdir -p $tmp_dir
# data directory check
if [ ! -d $aishell_audio_dir ] || [ ! -f $aishell_text ]; then
echo "Error: $0 requires two directory arguments"
exit 1;
fi
# find wav audio file for train, dev and test resp.
find $aishell_audio_dir -iname "*.wav" > $tmp_dir/wav.flist
n=`cat $tmp_dir/wav.flist | wc -l`
[ $n -ne 141925 ] && \
echo Warning: expected 141925 data data files, found $n
grep -i "wav/train" $tmp_dir/wav.flist > $train_dir/wav.flist || exit 1;
grep -i "wav/dev" $tmp_dir/wav.flist > $dev_dir/wav.flist || exit 1;
grep -i "wav/test" $tmp_dir/wav.flist > $test_dir/wav.flist || exit 1;
rm -r $tmp_dir
# Transcriptions preparation
for dir in $train_dir $dev_dir $test_dir; do
echo Preparing $dir transcriptions
sed -e 's/\.wav//' $dir/wav.flist | awk -F '/' '{print $NF}' > $dir/utt.list
paste -d' ' $dir/utt.list $dir/wav.flist > $dir/wav.scp_all
utils/filter_scp.pl -f 1 $dir/utt.list $aishell_text > $dir/transcripts.txt
awk '{print $1}' $dir/transcripts.txt > $dir/utt.list
utils/filter_scp.pl -f 1 $dir/utt.list $dir/wav.scp_all | sort -u > $dir/wav.scp
sort -u $dir/transcripts.txt > $dir/text
done
mkdir -p $output_dir/data/train $output_dir/data/dev $output_dir/data/test
for f in wav.scp text; do
cp $train_dir/$f $output_dir/data/train/$f || exit 1;
cp $dev_dir/$f $output_dir/data/dev/$f || exit 1;
cp $test_dir/$f $output_dir/data/test/$f || exit 1;
done
echo "$0: AISHELL data preparation succeeded"
exit 0;
@@ -0,0 +1,105 @@
#!/usr/bin/env bash
# Copyright 2014 Johns Hopkins University (author: Daniel Povey)
# 2017 Xingyu Na
# Apache 2.0
remove_archive=false
if [ "$1" == --remove-archive ]; then
remove_archive=true
shift
fi
if [ $# -ne 3 ]; then
echo "Usage: $0 [--remove-archive] <data-base> <url-base> <corpus-part>"
echo "e.g.: $0 /export/a05/xna/data www.openslr.org/resources/33 data_aishell"
echo "With --remove-archive it will remove the archive after successfully un-tarring it."
echo "<corpus-part> can be one of: data_aishell, resource_aishell."
fi
data=$1
url=$2
part=$3
if [ ! -d "$data" ]; then
echo "$0: no such directory $data"
exit 1;
fi
part_ok=false
list="data_aishell resource_aishell"
for x in $list; do
if [ "$part" == $x ]; then part_ok=true; fi
done
if ! $part_ok; then
echo "$0: expected <corpus-part> to be one of $list, but got '$part'"
exit 1;
fi
if [ -z "$url" ]; then
echo "$0: empty URL base."
exit 1;
fi
if [ -f $data/$part/.complete ]; then
echo "$0: data part $part was already successfully extracted, nothing to do."
exit 0;
fi
# sizes of the archive files in bytes.
sizes="15582913665 1246920"
if [ -f $data/$part.tgz ]; then
size=$(/bin/ls -l $data/$part.tgz | awk '{print $5}')
size_ok=false
for s in $sizes; do if [ $s == $size ]; then size_ok=true; fi; done
if ! $size_ok; then
echo "$0: removing existing file $data/$part.tgz because its size in bytes $size"
echo "does not equal the size of one of the archives."
rm $data/$part.tgz
else
echo "$data/$part.tgz exists and appears to be complete."
fi
fi
if [ ! -f $data/$part.tgz ]; then
if ! command -v wget >/dev/null; then
echo "$0: wget is not installed."
exit 1;
fi
full_url=$url/$part.tgz
echo "$0: downloading data from $full_url. This may take some time, please be patient."
cd $data || exit 1
if ! wget --no-check-certificate $full_url; then
echo "$0: error executing wget $full_url"
exit 1;
fi
fi
cd $data || exit 1
if ! tar -xvzf $part.tgz; then
echo "$0: error un-tarring archive $data/$part.tgz"
exit 1;
fi
touch $data/$part/.complete
if [ $part == "data_aishell" ]; then
cd $data/$part/wav || exit 1
for wav in ./*.tar.gz; do
echo "Extracting wav from $wav"
tar -zxf $wav && rm $wav
done
fi
echo "$0: Successfully downloaded and un-tarred $data/$part.tgz"
if $remove_archive; then
echo "$0: removing $data/$part.tgz file since --remove-archive option was supplied."
rm $data/$part.tgz
fi
exit 0;
+202
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@@ -0,0 +1,202 @@
#!/usr/bin/env bash
CUDA_VISIBLE_DEVICES="0,1"
# general configuration
feats_dir="../DATA" #feature output dictionary
exp_dir=`pwd`
lang=zh
token_type=char
stage=0
stop_stage=5
# feature configuration
nj=32
inference_device="cuda" #"cpu", "cuda:0", "cuda:1"
inference_checkpoint="model.pt.avg10"
inference_scp="wav.scp"
inference_batch_size=1
# data
raw_data=../raw_data
data_url=www.openslr.org/resources/33
# exp tag
tag="exp1"
workspace=`pwd`
master_port=12345
. utils/parse_options.sh || exit 1;
# Set bash to 'debug' mode, it will exit on :
# -e 'error', -u 'undefined variable', -o ... 'error in pipeline', -x 'print commands',
set -e
set -u
set -o pipefail
train_set=train
valid_set=dev
test_sets="dev test"
config=conformer_12e_6d_2048_256.yaml
model_dir="baseline_$(basename "${config}" .yaml)_${lang}_${token_type}_${tag}"
if [ ${stage} -le -1 ] && [ ${stop_stage} -ge -1 ]; then
echo "stage -1: Data Download"
mkdir -p ${raw_data}
local/download_and_untar.sh ${raw_data} ${data_url} data_aishell
local/download_and_untar.sh ${raw_data} ${data_url} resource_aishell
fi
if [ ${stage} -le 0 ] && [ ${stop_stage} -ge 0 ]; then
echo "stage 0: Data preparation"
# Data preparation
local/aishell_data_prep.sh ${raw_data}/data_aishell/wav ${raw_data}/data_aishell/transcript ${feats_dir}
for x in train dev test; do
cp ${feats_dir}/data/${x}/text ${feats_dir}/data/${x}/text.org
paste -d " " <(cut -f 1 -d" " ${feats_dir}/data/${x}/text.org) <(cut -f 2- -d" " ${feats_dir}/data/${x}/text.org | tr -d " ") \
> ${feats_dir}/data/${x}/text
utils/text2token.py -n 1 -s 1 ${feats_dir}/data/${x}/text > ${feats_dir}/data/${x}/text.org
mv ${feats_dir}/data/${x}/text.org ${feats_dir}/data/${x}/text
# convert wav.scp text to jsonl
scp_file_list_arg="++scp_file_list='[\"${feats_dir}/data/${x}/wav.scp\",\"${feats_dir}/data/${x}/text\"]'"
python ../../../funasr/datasets/audio_datasets/scp2jsonl.py \
++data_type_list='["source", "target"]' \
++jsonl_file_out=${feats_dir}/data/${x}/audio_datasets.jsonl \
${scp_file_list_arg}
done
fi
if [ ${stage} -le 1 ] && [ ${stop_stage} -ge 1 ]; then
echo "stage 1: Feature and CMVN Generation"
python ../../../funasr/bin/compute_audio_cmvn.py \
--config-path "${workspace}/conf" \
--config-name "${config}" \
++train_data_set_list="${feats_dir}/data/${train_set}/audio_datasets.jsonl" \
++cmvn_file="${feats_dir}/data/${train_set}/cmvn.json" \
fi
token_list=${feats_dir}/data/${lang}_token_list/$token_type/tokens.txt
echo "dictionary: ${token_list}"
if [ ${stage} -le 2 ] && [ ${stop_stage} -ge 2 ]; then
echo "stage 2: Dictionary Preparation"
mkdir -p ${feats_dir}/data/${lang}_token_list/$token_type/
echo "make a dictionary"
echo "<blank>" > ${token_list}
echo "<s>" >> ${token_list}
echo "</s>" >> ${token_list}
utils/text2token.py -s 1 -n 1 --space "" ${feats_dir}/data/$train_set/text | cut -f 2- -d" " | tr " " "\n" \
| sort | uniq | grep -a -v -e '^\s*$' | awk '{print $0}' >> ${token_list}
echo "<unk>" >> ${token_list}
fi
# LM Training Stage
if [ ${stage} -le 3 ] && [ ${stop_stage} -ge 3 ]; then
echo "stage 3: LM Training"
fi
# ASR Training Stage
if [ ${stage} -le 4 ] && [ ${stop_stage} -ge 4 ]; then
echo "stage 4: ASR Training"
mkdir -p ${exp_dir}/exp/${model_dir}
current_time=$(date "+%Y-%m-%d_%H-%M")
log_file="${exp_dir}/exp/${model_dir}/train.log.txt.${current_time}"
echo "log_file: ${log_file}"
export CUDA_VISIBLE_DEVICES=$CUDA_VISIBLE_DEVICES
gpu_num=$(echo $CUDA_VISIBLE_DEVICES | awk -F "," '{print NF}')
torchrun \
--nnodes 1 \
--nproc_per_node ${gpu_num} \
--master_port ${master_port} \
../../../funasr/bin/train.py \
--config-path "${workspace}/conf" \
--config-name "${config}" \
++train_data_set_list="${feats_dir}/data/${train_set}/audio_datasets.jsonl" \
++valid_data_set_list="${feats_dir}/data/${valid_set}/audio_datasets.jsonl" \
++tokenizer_conf.token_list="${token_list}" \
++frontend_conf.cmvn_file="${feats_dir}/data/${train_set}/am.mvn" \
++output_dir="${exp_dir}/exp/${model_dir}" &> ${log_file}
fi
# Testing Stage
if [ ${stage} -le 5 ] && [ ${stop_stage} -ge 5 ]; then
echo "stage 5: Inference"
if [ ${inference_device} == "cuda" ]; then
nj=$(echo $CUDA_VISIBLE_DEVICES | awk -F "," '{print NF}')
else
inference_batch_size=1
CUDA_VISIBLE_DEVICES=""
for JOB in $(seq ${nj}); do
CUDA_VISIBLE_DEVICES=$CUDA_VISIBLE_DEVICES"-1,"
done
fi
for dset in ${test_sets}; do
inference_dir="${exp_dir}/exp/${model_dir}/inference-${inference_checkpoint}/${dset}"
_logdir="${inference_dir}/logdir"
echo "inference_dir: ${inference_dir}"
mkdir -p "${_logdir}"
data_dir="${feats_dir}/data/${dset}"
key_file=${data_dir}/${inference_scp}
split_scps=
for JOB in $(seq "${nj}"); do
split_scps+=" ${_logdir}/keys.${JOB}.scp"
done
utils/split_scp.pl "${key_file}" ${split_scps}
gpuid_list_array=(${CUDA_VISIBLE_DEVICES//,/ })
for JOB in $(seq ${nj}); do
{
id=$((JOB-1))
gpuid=${gpuid_list_array[$id]}
export CUDA_VISIBLE_DEVICES=${gpuid}
python ../../../funasr/bin/inference.py \
--config-path="${exp_dir}/exp/${model_dir}" \
--config-name="config.yaml" \
++init_param="${exp_dir}/exp/${model_dir}/${inference_checkpoint}" \
++tokenizer_conf.token_list="${token_list}" \
++frontend_conf.cmvn_file="${feats_dir}/data/${train_set}/am.mvn" \
++input="${_logdir}/keys.${JOB}.scp" \
++output_dir="${inference_dir}/${JOB}" \
++device="${inference_device}" \
++ncpu=1 \
++disable_log=true \
++batch_size="${inference_batch_size}" &> ${_logdir}/log.${JOB}.txt
}&
done
wait
mkdir -p ${inference_dir}/1best_recog
for f in token score text; do
if [ -f "${inference_dir}/${JOB}/1best_recog/${f}" ]; then
for JOB in $(seq "${nj}"); do
cat "${inference_dir}/${JOB}/1best_recog/${f}"
done | sort -k1 >"${inference_dir}/1best_recog/${f}"
fi
done
echo "Computing WER ..."
python utils/postprocess_text_zh.py ${inference_dir}/1best_recog/text ${inference_dir}/1best_recog/text.proc
python utils/postprocess_text_zh.py ${data_dir}/text ${inference_dir}/1best_recog/text.ref
python utils/compute_wer.py ${inference_dir}/1best_recog/text.ref ${inference_dir}/1best_recog/text.proc ${inference_dir}/1best_recog/text.cer
tail -n 3 ${inference_dir}/1best_recog/text.cer
done
fi
+1
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@@ -0,0 +1 @@
../paraformer/utils
+14
View File
@@ -0,0 +1,14 @@
# E-Branchformer Result
## Training Config
- Feature info: using raw speech, extracting 80 dims fbank online, global cmvn, speed perturb(0.9, 1.0, 1.1), specaugment
- Train info: lr 0.001, batch_size 10000, 4 gpu(Tesla V100), acc_grad 1, 180 epochs
- Train config: conf/train_asr_e_branchformer.yaml
- LM config: LM was not used
## Results (CER)
| testset | CER(%) |
|:-----------:|:-------:|
| dev | 4.10 |
| test | 4.52 |
@@ -0,0 +1,118 @@
# This is an example that demonstrates how to configure a model file.
# You can modify the configuration according to your own requirements.
# to print the register_table:
# from funasr.register import tables
# tables.print()
# network architecture
model: Branchformer
model_conf:
ctc_weight: 0.3
lsm_weight: 0.1 # label smoothing option
length_normalized_loss: false
# encoder
encoder: EBranchformerEncoder
encoder_conf:
output_size: 256
attention_heads: 4
attention_layer_type: rel_selfattn
pos_enc_layer_type: rel_pos
rel_pos_type: latest
cgmlp_linear_units: 1024
cgmlp_conv_kernel: 31
use_linear_after_conv: false
gate_activation: identity
num_blocks: 12
dropout_rate: 0.1
positional_dropout_rate: 0.1
attention_dropout_rate: 0.1
input_layer: conv2d
layer_drop_rate: 0.0
linear_units: 1024
positionwise_layer_type: linear
use_ffn: true
macaron_ffn: true
merge_conv_kernel: 31
# decoder
decoder: TransformerDecoder
decoder_conf:
attention_heads: 4
linear_units: 2048
num_blocks: 6
dropout_rate: 0.1
positional_dropout_rate: 0.1
self_attention_dropout_rate: 0.
src_attention_dropout_rate: 0.
# frontend related
frontend: WavFrontend
frontend_conf:
fs: 16000
window: hamming
n_mels: 80
frame_length: 25
frame_shift: 10
dither: 0.0
lfr_m: 1
lfr_n: 1
specaug: SpecAug
specaug_conf:
apply_time_warp: true
time_warp_window: 5
time_warp_mode: bicubic
apply_freq_mask: true
freq_mask_width_range:
- 0
- 30
num_freq_mask: 2
apply_time_mask: true
time_mask_width_range:
- 0
- 40
num_time_mask: 2
train_conf:
accum_grad: 1
grad_clip: 5
max_epoch: 180
keep_nbest_models: 10
log_interval: 50
optim: adam
optim_conf:
lr: 0.001
weight_decay: 0.000001
scheduler: warmuplr
scheduler_conf:
warmup_steps: 35000
dataset: AudioDataset
dataset_conf:
index_ds: IndexDSJsonl
batch_sampler: EspnetStyleBatchSampler
batch_type: length # example or length
batch_size: 10000 # if batch_type is example, batch_size is the numbers of samples; if length, batch_size is source_token_len+target_token_len;
max_token_length: 2048 # filter samples if source_token_len+target_token_len > max_token_length,
buffer_size: 1024
shuffle: True
num_workers: 4
preprocessor_speech: SpeechPreprocessSpeedPerturb
preprocessor_speech_conf:
speed_perturb: [0.9, 1.0, 1.1]
tokenizer: CharTokenizer
tokenizer_conf:
unk_symbol: <unk>
ctc_conf:
dropout_rate: 0.0
ctc_type: builtin
reduce: true
ignore_nan_grad: true
normalize: null
@@ -0,0 +1 @@
../paraformer/demo_infer.sh
@@ -0,0 +1 @@
../paraformer/demo_train_or_finetune.sh
@@ -0,0 +1,66 @@
#!/bin/bash
# Copyright 2017 Xingyu Na
# Apache 2.0
#. ./path.sh || exit 1;
if [ $# != 3 ]; then
echo "Usage: $0 <audio-path> <text-path> <output-path>"
echo " $0 /export/a05/xna/data/data_aishell/wav /export/a05/xna/data/data_aishell/transcript data"
exit 1;
fi
aishell_audio_dir=$1
aishell_text=$2/aishell_transcript_v0.8.txt
output_dir=$3
train_dir=$output_dir/data/local/train
dev_dir=$output_dir/data/local/dev
test_dir=$output_dir/data/local/test
tmp_dir=$output_dir/data/local/tmp
mkdir -p $train_dir
mkdir -p $dev_dir
mkdir -p $test_dir
mkdir -p $tmp_dir
# data directory check
if [ ! -d $aishell_audio_dir ] || [ ! -f $aishell_text ]; then
echo "Error: $0 requires two directory arguments"
exit 1;
fi
# find wav audio file for train, dev and test resp.
find $aishell_audio_dir -iname "*.wav" > $tmp_dir/wav.flist
n=`cat $tmp_dir/wav.flist | wc -l`
[ $n -ne 141925 ] && \
echo Warning: expected 141925 data data files, found $n
grep -i "wav/train" $tmp_dir/wav.flist > $train_dir/wav.flist || exit 1;
grep -i "wav/dev" $tmp_dir/wav.flist > $dev_dir/wav.flist || exit 1;
grep -i "wav/test" $tmp_dir/wav.flist > $test_dir/wav.flist || exit 1;
rm -r $tmp_dir
# Transcriptions preparation
for dir in $train_dir $dev_dir $test_dir; do
echo Preparing $dir transcriptions
sed -e 's/\.wav//' $dir/wav.flist | awk -F '/' '{print $NF}' > $dir/utt.list
paste -d' ' $dir/utt.list $dir/wav.flist > $dir/wav.scp_all
utils/filter_scp.pl -f 1 $dir/utt.list $aishell_text > $dir/transcripts.txt
awk '{print $1}' $dir/transcripts.txt > $dir/utt.list
utils/filter_scp.pl -f 1 $dir/utt.list $dir/wav.scp_all | sort -u > $dir/wav.scp
sort -u $dir/transcripts.txt > $dir/text
done
mkdir -p $output_dir/data/train $output_dir/data/dev $output_dir/data/test
for f in wav.scp text; do
cp $train_dir/$f $output_dir/data/train/$f || exit 1;
cp $dev_dir/$f $output_dir/data/dev/$f || exit 1;
cp $test_dir/$f $output_dir/data/test/$f || exit 1;
done
echo "$0: AISHELL data preparation succeeded"
exit 0;
@@ -0,0 +1,105 @@
#!/usr/bin/env bash
# Copyright 2014 Johns Hopkins University (author: Daniel Povey)
# 2017 Xingyu Na
# Apache 2.0
remove_archive=false
if [ "$1" == --remove-archive ]; then
remove_archive=true
shift
fi
if [ $# -ne 3 ]; then
echo "Usage: $0 [--remove-archive] <data-base> <url-base> <corpus-part>"
echo "e.g.: $0 /export/a05/xna/data www.openslr.org/resources/33 data_aishell"
echo "With --remove-archive it will remove the archive after successfully un-tarring it."
echo "<corpus-part> can be one of: data_aishell, resource_aishell."
fi
data=$1
url=$2
part=$3
if [ ! -d "$data" ]; then
echo "$0: no such directory $data"
exit 1;
fi
part_ok=false
list="data_aishell resource_aishell"
for x in $list; do
if [ "$part" == $x ]; then part_ok=true; fi
done
if ! $part_ok; then
echo "$0: expected <corpus-part> to be one of $list, but got '$part'"
exit 1;
fi
if [ -z "$url" ]; then
echo "$0: empty URL base."
exit 1;
fi
if [ -f $data/$part/.complete ]; then
echo "$0: data part $part was already successfully extracted, nothing to do."
exit 0;
fi
# sizes of the archive files in bytes.
sizes="15582913665 1246920"
if [ -f $data/$part.tgz ]; then
size=$(/bin/ls -l $data/$part.tgz | awk '{print $5}')
size_ok=false
for s in $sizes; do if [ $s == $size ]; then size_ok=true; fi; done
if ! $size_ok; then
echo "$0: removing existing file $data/$part.tgz because its size in bytes $size"
echo "does not equal the size of one of the archives."
rm $data/$part.tgz
else
echo "$data/$part.tgz exists and appears to be complete."
fi
fi
if [ ! -f $data/$part.tgz ]; then
if ! command -v wget >/dev/null; then
echo "$0: wget is not installed."
exit 1;
fi
full_url=$url/$part.tgz
echo "$0: downloading data from $full_url. This may take some time, please be patient."
cd $data || exit 1
if ! wget --no-check-certificate $full_url; then
echo "$0: error executing wget $full_url"
exit 1;
fi
fi
cd $data || exit 1
if ! tar -xvzf $part.tgz; then
echo "$0: error un-tarring archive $data/$part.tgz"
exit 1;
fi
touch $data/$part/.complete
if [ $part == "data_aishell" ]; then
cd $data/$part/wav || exit 1
for wav in ./*.tar.gz; do
echo "Extracting wav from $wav"
tar -zxf $wav && rm $wav
done
fi
echo "$0: Successfully downloaded and un-tarred $data/$part.tgz"
if $remove_archive; then
echo "$0: removing $data/$part.tgz file since --remove-archive option was supplied."
rm $data/$part.tgz
fi
exit 0;
+203
View File
@@ -0,0 +1,203 @@
#!/usr/bin/env bash
CUDA_VISIBLE_DEVICES="0,1,2,3"
# general configuration
feats_dir="../DATA" #feature output dictionary
exp_dir=`pwd`
lang=zh
token_type=char
stage=0
stop_stage=5
# feature configuration
nj=32
inference_device="cuda" #"cpu"
inference_checkpoint="model.pt.avg10"
inference_scp="wav.scp"
inference_batch_size=1
# data
raw_data=../raw_data
data_url=www.openslr.org/resources/33
# exp tag
tag="exp1"
workspace=`pwd`
master_port=12345
. utils/parse_options.sh || exit 1;
# Set bash to 'debug' mode, it will exit on :
# -e 'error', -u 'undefined variable', -o ... 'error in pipeline', -x 'print commands',
set -e
set -u
set -o pipefail
train_set=train
valid_set=dev
test_sets="dev test"
config=e_branchformer_12e_6d_2048_256.yaml
model_dir="baseline_$(basename "${config}" .yaml)_${lang}_${token_type}_${tag}"
if [ ${stage} -le -1 ] && [ ${stop_stage} -ge -1 ]; then
echo "stage -1: Data Download"
mkdir -p ${raw_data}
local/download_and_untar.sh ${raw_data} ${data_url} data_aishell
local/download_and_untar.sh ${raw_data} ${data_url} resource_aishell
fi
if [ ${stage} -le 0 ] && [ ${stop_stage} -ge 0 ]; then
echo "stage 0: Data preparation"
# Data preparation
local/aishell_data_prep.sh ${raw_data}/data_aishell/wav ${raw_data}/data_aishell/transcript ${feats_dir}
for x in train dev test; do
cp ${feats_dir}/data/${x}/text ${feats_dir}/data/${x}/text.org
paste -d " " <(cut -f 1 -d" " ${feats_dir}/data/${x}/text.org) <(cut -f 2- -d" " ${feats_dir}/data/${x}/text.org | tr -d " ") \
> ${feats_dir}/data/${x}/text
utils/text2token.py -n 1 -s 1 ${feats_dir}/data/${x}/text > ${feats_dir}/data/${x}/text.org
mv ${feats_dir}/data/${x}/text.org ${feats_dir}/data/${x}/text
# convert wav.scp text to jsonl
scp_file_list_arg="++scp_file_list='[\"${feats_dir}/data/${x}/wav.scp\",\"${feats_dir}/data/${x}/text\"]'"
python ../../../funasr/datasets/audio_datasets/scp2jsonl.py \
++data_type_list='["source", "target"]' \
++jsonl_file_out=${feats_dir}/data/${x}/audio_datasets.jsonl \
${scp_file_list_arg}
done
fi
if [ ${stage} -le 1 ] && [ ${stop_stage} -ge 1 ]; then
echo "stage 1: Feature and CMVN Generation"
python ../../../funasr/bin/compute_audio_cmvn.py \
--config-path "${workspace}/conf" \
--config-name "${config}" \
++train_data_set_list="${feats_dir}/data/${train_set}/audio_datasets.jsonl" \
++cmvn_file="${feats_dir}/data/${train_set}/cmvn.json" \
fi
token_list=${feats_dir}/data/${lang}_token_list/$token_type/tokens.txt
echo "dictionary: ${token_list}"
if [ ${stage} -le 2 ] && [ ${stop_stage} -ge 2 ]; then
echo "stage 2: Dictionary Preparation"
mkdir -p ${feats_dir}/data/${lang}_token_list/$token_type/
echo "make a dictionary"
echo "<blank>" > ${token_list}
echo "<s>" >> ${token_list}
echo "</s>" >> ${token_list}
utils/text2token.py -s 1 -n 1 --space "" ${feats_dir}/data/$train_set/text | cut -f 2- -d" " | tr " " "\n" \
| sort | uniq | grep -a -v -e '^\s*$' | awk '{print $0}' >> ${token_list}
echo "<unk>" >> ${token_list}
fi
# LM Training Stage
if [ ${stage} -le 3 ] && [ ${stop_stage} -ge 3 ]; then
echo "stage 3: LM Training"
fi
# ASR Training Stage
if [ ${stage} -le 4 ] && [ ${stop_stage} -ge 4 ]; then
echo "stage 4: ASR Training"
mkdir -p ${exp_dir}/exp/${model_dir}
current_time=$(date "+%Y-%m-%d_%H-%M")
log_file="${exp_dir}/exp/${model_dir}/train.log.txt.${current_time}"
echo "log_file: ${log_file}"
export CUDA_VISIBLE_DEVICES=$CUDA_VISIBLE_DEVICES
gpu_num=$(echo $CUDA_VISIBLE_DEVICES | awk -F "," '{print NF}')
torchrun \
--nnodes 1 \
--nproc_per_node ${gpu_num} \
--master_port ${master_port} \
../../../funasr/bin/train.py \
--config-path "${workspace}/conf" \
--config-name "${config}" \
++train_data_set_list="${feats_dir}/data/${train_set}/audio_datasets.jsonl" \
++valid_data_set_list="${feats_dir}/data/${valid_set}/audio_datasets.jsonl" \
++tokenizer_conf.token_list="${token_list}" \
++frontend_conf.cmvn_file="${feats_dir}/data/${train_set}/am.mvn" \
++output_dir="${exp_dir}/exp/${model_dir}" &> ${log_file}
fi
# Testing Stage
if [ ${stage} -le 5 ] && [ ${stop_stage} -ge 5 ]; then
echo "stage 5: Inference"
if [ ${inference_device} == "cuda" ]; then
nj=$(echo $CUDA_VISIBLE_DEVICES | awk -F "," '{print NF}')
else
inference_batch_size=1
CUDA_VISIBLE_DEVICES=""
for JOB in $(seq ${nj}); do
CUDA_VISIBLE_DEVICES=$CUDA_VISIBLE_DEVICES"-1,"
done
fi
for dset in ${test_sets}; do
inference_dir="${exp_dir}/exp/${model_dir}/inference-${inference_checkpoint}/${dset}"
_logdir="${inference_dir}/logdir"
echo "inference_dir: ${inference_dir}"
mkdir -p "${_logdir}"
data_dir="${feats_dir}/data/${dset}"
key_file=${data_dir}/${inference_scp}
split_scps=
for JOB in $(seq "${nj}"); do
split_scps+=" ${_logdir}/keys.${JOB}.scp"
done
utils/split_scp.pl "${key_file}" ${split_scps}
gpuid_list_array=(${CUDA_VISIBLE_DEVICES//,/ })
for JOB in $(seq ${nj}); do
{
id=$((JOB-1))
gpuid=${gpuid_list_array[$id]}
export CUDA_VISIBLE_DEVICES=${gpuid}
python ../../../funasr/bin/inference.py \
--config-path="${exp_dir}/exp/${model_dir}" \
--config-name="config.yaml" \
++init_param="${exp_dir}/exp/${model_dir}/${inference_checkpoint}" \
++tokenizer_conf.token_list="${token_list}" \
++frontend_conf.cmvn_file="${feats_dir}/data/${train_set}/am.mvn" \
++input="${_logdir}/keys.${JOB}.scp" \
++output_dir="${inference_dir}/${JOB}" \
++device="${inference_device}" \
++ncpu=1 \
++disable_log=true \
++batch_size="${inference_batch_size}" &> ${_logdir}/log.${JOB}.txt
}&
done
wait
mkdir -p ${inference_dir}/1best_recog
for f in token score text; do
if [ -f "${inference_dir}/${JOB}/1best_recog/${f}" ]; then
for JOB in $(seq "${nj}"); do
cat "${inference_dir}/${JOB}/1best_recog/${f}"
done | sort -k1 >"${inference_dir}/1best_recog/${f}"
fi
done
echo "Computing WER ..."
python utils/postprocess_text_zh.py ${inference_dir}/1best_recog/text ${inference_dir}/1best_recog/text.proc
python utils/postprocess_text_zh.py ${data_dir}/text ${inference_dir}/1best_recog/text.ref
python utils/compute_wer.py ${inference_dir}/1best_recog/text.ref ${inference_dir}/1best_recog/text.proc ${inference_dir}/1best_recog/text.cer
tail -n 3 ${inference_dir}/1best_recog/text.cer
done
fi
+1
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@@ -0,0 +1 @@
../paraformer/utils
+24
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@@ -0,0 +1,24 @@
# Paraformer
pretrained model in [ModelScope](https://www.modelscope.cn/home):[speech_paraformer_asr_nat-aishell1-pytorch](https://www.modelscope.cn/models/damo/speech_paraformer_asr_nat-aishell1-pytorch/summary)
## Training Config
- Feature info: using 80 dims fbank, global cmvn, speed perturb(0.9, 1.0, 1.1), specaugment
- Train info: lr 5e-4, batch_size 25000, 2 gpu(Tesla V100), acc_grad 1, 50 epochs
- Train config: conf/train_asr_paraformer_conformer_12e_6d_2048_256.yaml
- LM config: LM was not used
## Results (CER)
- Decode config: conf/decode_asr_transformer_noctc_1best.yaml (ctc weight:0.0)
| testset | CER(%) |
|:-----------:|:-------:|
| dev | 4.66 |
| test | 5.11 |
- Decode config: conf/decode_asr_transformer.yaml (ctc weight:0.5)
| testset | CER(%) |
|:-----------:|:-------:|
| dev | 4.52 |
| test | 4.94 |
@@ -0,0 +1,118 @@
# network architecture
model: Paraformer
model_conf:
ctc_weight: 0.3
lsm_weight: 0.1
length_normalized_loss: false
predictor_weight: 1.0
sampling_ratio: 0.4
use_1st_decoder_loss: true
# encoder
encoder: ConformerEncoder
encoder_conf:
output_size: 256 # dimension of attention
attention_heads: 4
linear_units: 2048 # the number of units of position-wise feed forward
num_blocks: 12 # the number of encoder blocks
dropout_rate: 0.1
positional_dropout_rate: 0.1
attention_dropout_rate: 0.0
input_layer: conv2d # encoder architecture type
normalize_before: true
pos_enc_layer_type: rel_pos
selfattention_layer_type: rel_selfattn
activation_type: swish
macaron_style: true
use_cnn_module: true
cnn_module_kernel: 15
# decoder
decoder: ParaformerSANDecoder
decoder_conf:
attention_heads: 4
linear_units: 2048
num_blocks: 6
dropout_rate: 0.1
positional_dropout_rate: 0.1
self_attention_dropout_rate: 0.0
src_attention_dropout_rate: 0.0
# predictor
predictor: CifPredictor
predictor_conf:
idim: 256
threshold: 1.0
l_order: 1
r_order: 1
tail_threshold: 0.45
# frontend related
frontend: WavFrontend
frontend_conf:
fs: 16000
window: hamming
n_mels: 80
frame_length: 25
frame_shift: 10
lfr_m: 1
lfr_n: 1
specaug: SpecAug
specaug_conf:
apply_time_warp: true
time_warp_window: 5
time_warp_mode: bicubic
apply_freq_mask: true
freq_mask_width_range:
- 0
- 30
num_freq_mask: 2
apply_time_mask: true
time_mask_width_range:
- 0
- 40
num_time_mask: 2
train_conf:
accum_grad: 1
grad_clip: 5
max_epoch: 150
keep_nbest_models: 10
avg_nbest_model: 10
log_interval: 50
optim: adam
optim_conf:
lr: 0.0005
scheduler: warmuplr
scheduler_conf:
warmup_steps: 30000
dataset: AudioDataset
dataset_conf:
index_ds: IndexDSJsonl
batch_sampler: EspnetStyleBatchSampler
batch_type: length # example or length
batch_size: 25000 # if batch_type is example, batch_size is the numbers of samples; if length, batch_size is source_token_len+target_token_len;
max_token_length: 2048 # filter samples if source_token_len+target_token_len > max_token_length,
buffer_size: 1024
shuffle: True
num_workers: 4
preprocessor_speech: SpeechPreprocessSpeedPerturb
preprocessor_speech_conf:
speed_perturb: [0.9, 1.0, 1.1]
tokenizer: CharTokenizer
tokenizer_conf:
unk_symbol: <unk>
ctc_conf:
dropout_rate: 0.0
ctc_type: builtin
reduce: true
ignore_nan_grad: true
normalize: null
+15
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@@ -0,0 +1,15 @@
# Copyright FunASR (https://github.com/alibaba-damo-academy/FunASR). All Rights Reserved.
# MIT License (https://opensource.org/licenses/MIT)
python -m funasr.bin.inference \
--config-path="/mnt/workspace/FunASR/examples/aishell/paraformer/exp/baseline_paraformer_conformer_12e_6d_2048_256_zh_char_exp3" \
--config-name="config.yaml" \
++init_param="/mnt/workspace/FunASR/examples/aishell/paraformer/exp/baseline_paraformer_conformer_12e_6d_2048_256_zh_char_exp3/model.pt.ep38" \
++tokenizer_conf.token_list="/mnt/nfs/zhifu.gzf/data/AISHELL-1-feats/DATA/data/zh_token_list/char/tokens.txt" \
++frontend_conf.cmvn_file="/mnt/nfs/zhifu.gzf/data/AISHELL-1-feats/DATA/data/train/am.mvn" \
++input="/mnt/nfs/zhifu.gzf/data/AISHELL-1/data_aishell/wav/train/S0002/BAC009S0002W0122.wav" \
++output_dir="./outputs/debug" \
++device="cuda:0" \
@@ -0,0 +1,51 @@
# 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="0,1"
gpu_num=$(echo $CUDA_VISIBLE_DEVICES | awk -F "," '{print NF}')
# data dir, which contains: train.json, val.json, tokens.jsonl/tokens.txt, am.mvn
data_dir="/Users/zhifu/funasr1.0/data/list"
## generate jsonl from wav.scp and text.txt
#python -m funasr.datasets.audio_datasets.scp2jsonl \
#++scp_file_list='["/Users/zhifu/funasr1.0/test_local/wav.scp", "/Users/zhifu/funasr1.0/test_local/text.txt"]' \
#++data_type_list='["source", "target"]' \
#++jsonl_file_out=/Users/zhifu/funasr1.0/test_local/audio_datasets.jsonl
train_data="${data_dir}/train.jsonl"
val_data="${data_dir}/val.jsonl"
tokens="${data_dir}/tokens.json"
cmvn_file="${data_dir}/am.mvn"
# exp output dir
output_dir="/Users/zhifu/exp"
log_file="${output_dir}/log.txt"
workspace=`pwd`
config="paraformer_conformer_12e_6d_2048_256.yaml"
init_param="${output_dir}/model.pt"
mkdir -p ${output_dir}
echo "log_file: ${log_file}"
torchrun \
--nnodes 1 \
--nproc_per_node ${gpu_num} \
../../../funasr/bin/train.py \
--config-path "${workspace}/conf" \
--config-name "${config}" \
++train_data_set_list="${train_data}" \
++valid_data_set_list="${val_data}" \
++tokenizer_conf.token_list="${tokens}" \
++frontend_conf.cmvn_file="${cmvn_file}" \
++dataset_conf.batch_size=32 \
++dataset_conf.batch_type="example" \
++dataset_conf.num_workers=4 \
++train_conf.max_epoch=150 \
++optim_conf.lr=0.0002 \
++init_param="${init_param}" \
++output_dir="${output_dir}" &> ${log_file}
@@ -0,0 +1,66 @@
#!/bin/bash
# Copyright 2017 Xingyu Na
# Apache 2.0
#. ./path.sh || exit 1;
if [ $# != 3 ]; then
echo "Usage: $0 <audio-path> <text-path> <output-path>"
echo " $0 /export/a05/xna/data/data_aishell/wav /export/a05/xna/data/data_aishell/transcript data"
exit 1;
fi
aishell_audio_dir=$1
aishell_text=$2/aishell_transcript_v0.8.txt
output_dir=$3
train_dir=$output_dir/data/local/train
dev_dir=$output_dir/data/local/dev
test_dir=$output_dir/data/local/test
tmp_dir=$output_dir/data/local/tmp
mkdir -p $train_dir
mkdir -p $dev_dir
mkdir -p $test_dir
mkdir -p $tmp_dir
# data directory check
if [ ! -d $aishell_audio_dir ] || [ ! -f $aishell_text ]; then
echo "Error: $0 requires two directory arguments"
exit 1;
fi
# find wav audio file for train, dev and test resp.
find $aishell_audio_dir -iname "*.wav" > $tmp_dir/wav.flist
n=`cat $tmp_dir/wav.flist | wc -l`
[ $n -ne 141925 ] && \
echo Warning: expected 141925 data data files, found $n
grep -i "wav/train" $tmp_dir/wav.flist > $train_dir/wav.flist || exit 1;
grep -i "wav/dev" $tmp_dir/wav.flist > $dev_dir/wav.flist || exit 1;
grep -i "wav/test" $tmp_dir/wav.flist > $test_dir/wav.flist || exit 1;
rm -r $tmp_dir
# Transcriptions preparation
for dir in $train_dir $dev_dir $test_dir; do
echo Preparing $dir transcriptions
sed -e 's/\.wav//' $dir/wav.flist | awk -F '/' '{print $NF}' > $dir/utt.list
paste -d' ' $dir/utt.list $dir/wav.flist > $dir/wav.scp_all
utils/filter_scp.pl -f 1 $dir/utt.list $aishell_text > $dir/transcripts.txt
awk '{print $1}' $dir/transcripts.txt > $dir/utt.list
utils/filter_scp.pl -f 1 $dir/utt.list $dir/wav.scp_all | sort -u > $dir/wav.scp
sort -u $dir/transcripts.txt > $dir/text
done
mkdir -p $output_dir/data/train $output_dir/data/dev $output_dir/data/test
for f in wav.scp text; do
cp $train_dir/$f $output_dir/data/train/$f || exit 1;
cp $dev_dir/$f $output_dir/data/dev/$f || exit 1;
cp $test_dir/$f $output_dir/data/test/$f || exit 1;
done
echo "$0: AISHELL data preparation succeeded"
exit 0;
@@ -0,0 +1,105 @@
#!/usr/bin/env bash
# Copyright 2014 Johns Hopkins University (author: Daniel Povey)
# 2017 Xingyu Na
# Apache 2.0
remove_archive=false
if [ "$1" == --remove-archive ]; then
remove_archive=true
shift
fi
if [ $# -ne 3 ]; then
echo "Usage: $0 [--remove-archive] <data-base> <url-base> <corpus-part>"
echo "e.g.: $0 /export/a05/xna/data www.openslr.org/resources/33 data_aishell"
echo "With --remove-archive it will remove the archive after successfully un-tarring it."
echo "<corpus-part> can be one of: data_aishell, resource_aishell."
fi
data=$1
url=$2
part=$3
if [ ! -d "$data" ]; then
echo "$0: no such directory $data"
exit 1;
fi
part_ok=false
list="data_aishell resource_aishell"
for x in $list; do
if [ "$part" == $x ]; then part_ok=true; fi
done
if ! $part_ok; then
echo "$0: expected <corpus-part> to be one of $list, but got '$part'"
exit 1;
fi
if [ -z "$url" ]; then
echo "$0: empty URL base."
exit 1;
fi
if [ -f $data/$part/.complete ]; then
echo "$0: data part $part was already successfully extracted, nothing to do."
exit 0;
fi
# sizes of the archive files in bytes.
sizes="15582913665 1246920"
if [ -f $data/$part.tgz ]; then
size=$(/bin/ls -l $data/$part.tgz | awk '{print $5}')
size_ok=false
for s in $sizes; do if [ $s == $size ]; then size_ok=true; fi; done
if ! $size_ok; then
echo "$0: removing existing file $data/$part.tgz because its size in bytes $size"
echo "does not equal the size of one of the archives."
rm $data/$part.tgz
else
echo "$data/$part.tgz exists and appears to be complete."
fi
fi
if [ ! -f $data/$part.tgz ]; then
if ! command -v wget >/dev/null; then
echo "$0: wget is not installed."
exit 1;
fi
full_url=$url/$part.tgz
echo "$0: downloading data from $full_url. This may take some time, please be patient."
cd $data || exit 1
if ! wget --no-check-certificate $full_url; then
echo "$0: error executing wget $full_url"
exit 1;
fi
fi
cd $data || exit 1
if ! tar -xvzf $part.tgz; then
echo "$0: error un-tarring archive $data/$part.tgz"
exit 1;
fi
touch $data/$part/.complete
if [ $part == "data_aishell" ]; then
cd $data/$part/wav || exit 1
for wav in ./*.tar.gz; do
echo "Extracting wav from $wav"
tar -zxf $wav && rm $wav
done
fi
echo "$0: Successfully downloaded and un-tarred $data/$part.tgz"
if $remove_archive; then
echo "$0: removing $data/$part.tgz file since --remove-archive option was supplied."
rm $data/$part.tgz
fi
exit 0;
+201
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@@ -0,0 +1,201 @@
#!/usr/bin/env bash
CUDA_VISIBLE_DEVICES="0,1"
# general configuration
feats_dir="../DATA" #feature output dictionary
exp_dir=`pwd`
lang=zh
token_type=char
stage=0
stop_stage=5
# feature configuration
nj=32
inference_device="cuda" #"cpu"
inference_checkpoint="model.pt.avg10"
inference_scp="wav.scp"
inference_batch_size=32
# data
raw_data=../raw_data
data_url=www.openslr.org/resources/33
# exp tag
tag="exp1"
workspace=`pwd`
master_port=12345
. utils/parse_options.sh || exit 1;
# Set bash to 'debug' mode, it will exit on :
# -e 'error', -u 'undefined variable', -o ... 'error in pipeline', -x 'print commands',
set -e
set -u
set -o pipefail
train_set=train
valid_set=dev
test_sets="dev test"
config=paraformer_conformer_12e_6d_2048_256.yaml
model_dir="baseline_$(basename "${config}" .yaml)_${lang}_${token_type}_${tag}"
if [ ${stage} -le -1 ] && [ ${stop_stage} -ge -1 ]; then
echo "stage -1: Data Download"
mkdir -p ${raw_data}
local/download_and_untar.sh ${raw_data} ${data_url} data_aishell
local/download_and_untar.sh ${raw_data} ${data_url} resource_aishell
fi
if [ ${stage} -le 0 ] && [ ${stop_stage} -ge 0 ]; then
echo "stage 0: Data preparation"
# Data preparation
local/aishell_data_prep.sh ${raw_data}/data_aishell/wav ${raw_data}/data_aishell/transcript ${feats_dir}
for x in train dev test; do
cp ${feats_dir}/data/${x}/text ${feats_dir}/data/${x}/text.org
paste -d " " <(cut -f 1 -d" " ${feats_dir}/data/${x}/text.org) <(cut -f 2- -d" " ${feats_dir}/data/${x}/text.org | tr -d " ") \
> ${feats_dir}/data/${x}/text
utils/text2token.py -n 1 -s 1 ${feats_dir}/data/${x}/text > ${feats_dir}/data/${x}/text.org
mv ${feats_dir}/data/${x}/text.org ${feats_dir}/data/${x}/text
# convert wav.scp text to jsonl
scp_file_list_arg="++scp_file_list='[\"${feats_dir}/data/${x}/wav.scp\",\"${feats_dir}/data/${x}/text\"]'"
python ../../../funasr/datasets/audio_datasets/scp2jsonl.py \
++data_type_list='["source", "target"]' \
++jsonl_file_out=${feats_dir}/data/${x}/audio_datasets.jsonl \
${scp_file_list_arg}
done
fi
if [ ${stage} -le 1 ] && [ ${stop_stage} -ge 1 ]; then
echo "stage 1: Feature and CMVN Generation"
python ../../../funasr/bin/compute_audio_cmvn.py \
--config-path "${workspace}/conf" \
--config-name "${config}" \
++train_data_set_list="${feats_dir}/data/${train_set}/audio_datasets.jsonl" \
++cmvn_file="${feats_dir}/data/${train_set}/cmvn.json"
fi
token_list=${feats_dir}/data/${lang}_token_list/$token_type/tokens.txt
echo "dictionary: ${token_list}"
if [ ${stage} -le 2 ] && [ ${stop_stage} -ge 2 ]; then
echo "stage 2: Dictionary Preparation"
mkdir -p ${feats_dir}/data/${lang}_token_list/$token_type/
echo "make a dictionary"
echo "<blank>" > ${token_list}
echo "<s>" >> ${token_list}
echo "</s>" >> ${token_list}
utils/text2token.py -s 1 -n 1 --space "" ${feats_dir}/data/$train_set/text | cut -f 2- -d" " | tr " " "\n" \
| sort | uniq | grep -a -v -e '^\s*$' | awk '{print $0}' >> ${token_list}
echo "<unk>" >> ${token_list}
fi
# LM Training Stage
if [ ${stage} -le 3 ] && [ ${stop_stage} -ge 3 ]; then
echo "stage 3: LM Training"
fi
# ASR Training Stage
if [ ${stage} -le 4 ] && [ ${stop_stage} -ge 4 ]; then
echo "stage 4: ASR Training"
mkdir -p ${exp_dir}/exp/${model_dir}
current_time=$(date "+%Y-%m-%d_%H-%M")
log_file="${exp_dir}/exp/${model_dir}/train.log.txt.${current_time}"
echo "log_file: ${log_file}"
export CUDA_VISIBLE_DEVICES=$CUDA_VISIBLE_DEVICES
gpu_num=$(echo $CUDA_VISIBLE_DEVICES | awk -F "," '{print NF}')
torchrun \
--nnodes 1 \
--nproc_per_node ${gpu_num} \
--master_port ${master_port} \
../../../funasr/bin/train.py \
--config-path "${workspace}/conf" \
--config-name "${config}" \
++train_data_set_list="${feats_dir}/data/${train_set}/audio_datasets.jsonl" \
++valid_data_set_list="${feats_dir}/data/${valid_set}/audio_datasets.jsonl" \
++tokenizer_conf.token_list="${token_list}" \
++frontend_conf.cmvn_file="${feats_dir}/data/${train_set}/am.mvn" \
++output_dir="${exp_dir}/exp/${model_dir}" &> ${log_file}
fi
# Testing Stage
if [ ${stage} -le 5 ] && [ ${stop_stage} -ge 5 ]; then
echo "stage 5: Inference"
if [ ${inference_device} == "cuda" ]; then
nj=$(echo $CUDA_VISIBLE_DEVICES | awk -F "," '{print NF}')
else
inference_batch_size=1
CUDA_VISIBLE_DEVICES=""
for JOB in $(seq ${nj}); do
CUDA_VISIBLE_DEVICES=$CUDA_VISIBLE_DEVICES"-1,"
done
fi
for dset in ${test_sets}; do
inference_dir="${exp_dir}/exp/${model_dir}/inference-${inference_checkpoint}/${dset}"
_logdir="${inference_dir}/logdir"
echo "inference_dir: ${inference_dir}"
mkdir -p "${_logdir}"
data_dir="${feats_dir}/data/${dset}"
key_file=${data_dir}/${inference_scp}
split_scps=
for JOB in $(seq "${nj}"); do
split_scps+=" ${_logdir}/keys.${JOB}.scp"
done
utils/split_scp.pl "${key_file}" ${split_scps}
gpuid_list_array=(${CUDA_VISIBLE_DEVICES//,/ })
for JOB in $(seq ${nj}); do
{
id=$((JOB-1))
gpuid=${gpuid_list_array[$id]}
export CUDA_VISIBLE_DEVICES=${gpuid}
python ../../../funasr/bin/inference.py \
--config-path="${exp_dir}/exp/${model_dir}" \
--config-name="config.yaml" \
++init_param="${exp_dir}/exp/${model_dir}/${inference_checkpoint}" \
++tokenizer_conf.token_list="${token_list}" \
++frontend_conf.cmvn_file="${feats_dir}/data/${train_set}/am.mvn" \
++input="${_logdir}/keys.${JOB}.scp" \
++output_dir="${inference_dir}/${JOB}" \
++device="${inference_device}" \
++ncpu=1 \
++disable_log=true \
++batch_size="${inference_batch_size}" &> ${_logdir}/log.${JOB}.txt
}&
done
wait
mkdir -p ${inference_dir}/1best_recog
for f in token score text; do
if [ -f "${inference_dir}/${JOB}/1best_recog/${f}" ]; then
for JOB in $(seq "${nj}"); do
cat "${inference_dir}/${JOB}/1best_recog/${f}"
done | sort -k1 >"${inference_dir}/1best_recog/${f}"
fi
done
echo "Computing WER ..."
python utils/postprocess_text_zh.py ${inference_dir}/1best_recog/text ${inference_dir}/1best_recog/text.proc
python utils/postprocess_text_zh.py ${data_dir}/text ${inference_dir}/1best_recog/text.ref
python utils/compute_wer.py ${inference_dir}/1best_recog/text.ref ${inference_dir}/1best_recog/text.proc ${inference_dir}/1best_recog/text.cer
tail -n 3 ${inference_dir}/1best_recog/text.cer
done
fi
@@ -0,0 +1,197 @@
import os
import numpy as np
import sys
def compute_wer(ref_file, hyp_file, cer_detail_file):
rst = {
"Wrd": 0,
"Corr": 0,
"Ins": 0,
"Del": 0,
"Sub": 0,
"Snt": 0,
"Err": 0.0,
"S.Err": 0.0,
"wrong_words": 0,
"wrong_sentences": 0,
}
hyp_dict = {}
ref_dict = {}
with open(hyp_file, "r") as hyp_reader:
for line in hyp_reader:
key = line.strip().split()[0]
value = line.strip().split()[1:]
hyp_dict[key] = value
with open(ref_file, "r") as ref_reader:
for line in ref_reader:
key = line.strip().split()[0]
value = line.strip().split()[1:]
ref_dict[key] = value
cer_detail_writer = open(cer_detail_file, "w")
for hyp_key in hyp_dict:
if hyp_key in ref_dict:
out_item = compute_wer_by_line(hyp_dict[hyp_key], ref_dict[hyp_key])
rst["Wrd"] += out_item["nwords"]
rst["Corr"] += out_item["cor"]
rst["wrong_words"] += out_item["wrong"]
rst["Ins"] += out_item["ins"]
rst["Del"] += out_item["del"]
rst["Sub"] += out_item["sub"]
rst["Snt"] += 1
if out_item["wrong"] > 0:
rst["wrong_sentences"] += 1
cer_detail_writer.write(hyp_key + print_cer_detail(out_item) + "\n")
cer_detail_writer.write(
"ref:" + "\t" + " ".join(list(map(lambda x: x.lower(), ref_dict[hyp_key]))) + "\n"
)
cer_detail_writer.write(
"hyp:" + "\t" + " ".join(list(map(lambda x: x.lower(), hyp_dict[hyp_key]))) + "\n"
)
if rst["Wrd"] > 0:
rst["Err"] = round(rst["wrong_words"] * 100 / rst["Wrd"], 2)
if rst["Snt"] > 0:
rst["S.Err"] = round(rst["wrong_sentences"] * 100 / rst["Snt"], 2)
cer_detail_writer.write("\n")
cer_detail_writer.write(
"%WER "
+ str(rst["Err"])
+ " [ "
+ str(rst["wrong_words"])
+ " / "
+ str(rst["Wrd"])
+ ", "
+ str(rst["Ins"])
+ " ins, "
+ str(rst["Del"])
+ " del, "
+ str(rst["Sub"])
+ " sub ]"
+ "\n"
)
cer_detail_writer.write(
"%SER "
+ str(rst["S.Err"])
+ " [ "
+ str(rst["wrong_sentences"])
+ " / "
+ str(rst["Snt"])
+ " ]"
+ "\n"
)
cer_detail_writer.write(
"Scored "
+ str(len(hyp_dict))
+ " sentences, "
+ str(len(hyp_dict) - rst["Snt"])
+ " not present in hyp."
+ "\n"
)
def compute_wer_by_line(hyp, ref):
hyp = list(map(lambda x: x.lower(), hyp))
ref = list(map(lambda x: x.lower(), ref))
len_hyp = len(hyp)
len_ref = len(ref)
cost_matrix = np.zeros((len_hyp + 1, len_ref + 1), dtype=np.int16)
ops_matrix = np.zeros((len_hyp + 1, len_ref + 1), dtype=np.int8)
for i in range(len_hyp + 1):
cost_matrix[i][0] = i
for j in range(len_ref + 1):
cost_matrix[0][j] = j
for i in range(1, len_hyp + 1):
for j in range(1, len_ref + 1):
if hyp[i - 1] == ref[j - 1]:
cost_matrix[i][j] = cost_matrix[i - 1][j - 1]
else:
substitution = cost_matrix[i - 1][j - 1] + 1
insertion = cost_matrix[i - 1][j] + 1
deletion = cost_matrix[i][j - 1] + 1
compare_val = [substitution, insertion, deletion]
min_val = min(compare_val)
operation_idx = compare_val.index(min_val) + 1
cost_matrix[i][j] = min_val
ops_matrix[i][j] = operation_idx
match_idx = []
i = len_hyp
j = len_ref
rst = {"nwords": len_ref, "cor": 0, "wrong": 0, "ins": 0, "del": 0, "sub": 0}
while i >= 0 or j >= 0:
i_idx = max(0, i)
j_idx = max(0, j)
if ops_matrix[i_idx][j_idx] == 0: # correct
if i - 1 >= 0 and j - 1 >= 0:
match_idx.append((j - 1, i - 1))
rst["cor"] += 1
i -= 1
j -= 1
elif ops_matrix[i_idx][j_idx] == 2: # insert
i -= 1
rst["ins"] += 1
elif ops_matrix[i_idx][j_idx] == 3: # delete
j -= 1
rst["del"] += 1
elif ops_matrix[i_idx][j_idx] == 1: # substitute
i -= 1
j -= 1
rst["sub"] += 1
if i < 0 and j >= 0:
rst["del"] += 1
elif j < 0 and i >= 0:
rst["ins"] += 1
match_idx.reverse()
wrong_cnt = cost_matrix[len_hyp][len_ref]
rst["wrong"] = wrong_cnt
return rst
def print_cer_detail(rst):
return (
"("
+ "nwords="
+ str(rst["nwords"])
+ ",cor="
+ str(rst["cor"])
+ ",ins="
+ str(rst["ins"])
+ ",del="
+ str(rst["del"])
+ ",sub="
+ str(rst["sub"])
+ ") corr:"
+ "{:.2%}".format(rst["cor"] / rst["nwords"])
+ ",cer:"
+ "{:.2%}".format(rst["wrong"] / rst["nwords"])
)
if __name__ == "__main__":
if len(sys.argv) != 4:
print("usage : python compute-wer.py test.ref test.hyp test.wer")
sys.exit(0)
ref_file = sys.argv[1]
hyp_file = sys.argv[2]
cer_detail_file = sys.argv[3]
compute_wer(ref_file, hyp_file, cer_detail_file)
@@ -0,0 +1,49 @@
from transformers import AutoTokenizer, AutoModel, pipeline
import numpy as np
import sys
import os
import torch
from kaldiio import WriteHelper
import re
text_file_json = sys.argv[1]
out_ark = sys.argv[2]
out_scp = sys.argv[3]
out_shape = sys.argv[4]
device = int(sys.argv[5])
model_path = sys.argv[6]
model = AutoModel.from_pretrained(model_path)
tokenizer = AutoTokenizer.from_pretrained(model_path)
extractor = pipeline(task="feature-extraction", model=model, tokenizer=tokenizer, device=device)
with open(text_file_json, "r") as f:
js = f.readlines()
f_shape = open(out_shape, "w")
with WriteHelper("ark,scp:{},{}".format(out_ark, out_scp)) as writer:
with torch.no_grad():
for idx, line in enumerate(js):
id, tokens = line.strip().split(" ", 1)
tokens = re.sub(" ", "", tokens.strip())
tokens = " ".join([j for j in tokens])
token_num = len(tokens.split(" "))
outputs = extractor(tokens)
outputs = np.array(outputs)
embeds = outputs[0, 1:-1, :]
token_num_embeds, dim = embeds.shape
if token_num == token_num_embeds:
writer(id, embeds)
shape_line = "{} {},{}\n".format(id, token_num_embeds, dim)
f_shape.write(shape_line)
else:
print(
"{}, size has changed, {}, {}, {}".format(
id, token_num, token_num_embeds, tokens
)
)
f_shape.close()
@@ -0,0 +1,87 @@
#!/usr/bin/env perl
# Copyright 2010-2012 Microsoft Corporation
# Johns Hopkins University (author: Daniel Povey)
# Licensed under the Apache License, Version 2.0 (the "License");
# you may not use this file except in compliance with the License.
# You may obtain a copy of the License at
#
# http://www.apache.org/licenses/LICENSE-2.0
#
# THIS CODE IS PROVIDED *AS IS* BASIS, WITHOUT WARRANTIES OR CONDITIONS OF ANY
# KIND, EITHER EXPRESS OR IMPLIED, INCLUDING WITHOUT LIMITATION ANY IMPLIED
# WARRANTIES OR CONDITIONS OF TITLE, FITNESS FOR A PARTICULAR PURPOSE,
# MERCHANTABLITY OR NON-INFRINGEMENT.
# See the Apache 2 License for the specific language governing permissions and
# limitations under the License.
# This script takes a list of utterance-ids or any file whose first field
# of each line is an utterance-id, and filters an scp
# file (or any file whose "n-th" field is an utterance id), printing
# out only those lines whose "n-th" field is in id_list. The index of
# the "n-th" field is 1, by default, but can be changed by using
# the -f <n> switch
$exclude = 0;
$field = 1;
$shifted = 0;
do {
$shifted=0;
if ($ARGV[0] eq "--exclude") {
$exclude = 1;
shift @ARGV;
$shifted=1;
}
if ($ARGV[0] eq "-f") {
$field = $ARGV[1];
shift @ARGV; shift @ARGV;
$shifted=1
}
} while ($shifted);
if(@ARGV < 1 || @ARGV > 2) {
die "Usage: filter_scp.pl [--exclude] [-f <field-to-filter-on>] id_list [in.scp] > out.scp \n" .
"Prints only the input lines whose f'th field (default: first) is in 'id_list'.\n" .
"Note: only the first field of each line in id_list matters. With --exclude, prints\n" .
"only the lines that were *not* in id_list.\n" .
"Caution: previously, the -f option was interpreted as a zero-based field index.\n" .
"If your older scripts (written before Oct 2014) stopped working and you used the\n" .
"-f option, add 1 to the argument.\n" .
"See also: scripts/filter_scp.pl .\n";
}
$idlist = shift @ARGV;
open(F, "<$idlist") || die "Could not open id-list file $idlist";
while(<F>) {
@A = split;
@A>=1 || die "Invalid id-list file line $_";
$seen{$A[0]} = 1;
}
if ($field == 1) { # Treat this as special case, since it is common.
while(<>) {
$_ =~ m/\s*(\S+)\s*/ || die "Bad line $_, could not get first field.";
# $1 is what we filter on.
if ((!$exclude && $seen{$1}) || ($exclude && !defined $seen{$1})) {
print $_;
}
}
} else {
while(<>) {
@A = split;
@A > 0 || die "Invalid scp file line $_";
@A >= $field || die "Invalid scp file line $_";
if ((!$exclude && $seen{$A[$field-1]}) || ($exclude && !defined $seen{$A[$field-1]})) {
print $_;
}
}
}
# tests:
# the following should print "foo 1"
# ( echo foo 1; echo bar 2 ) | scripts/filter_scp.pl <(echo foo)
# the following should print "bar 2".
# ( echo foo 1; echo bar 2 ) | scripts/filter_scp.pl -f 2 <(echo 2)
@@ -0,0 +1,35 @@
#!/usr/bin/env bash
echo "$0 $@"
data_dir=$1
if [ ! -f ${data_dir}/wav.scp ]; then
echo "$0: wav.scp is not found"
exit 1;
fi
if [ ! -f ${data_dir}/text ]; then
echo "$0: text is not found"
exit 1;
fi
mkdir -p ${data_dir}/.backup
awk '{print $1}' ${data_dir}/wav.scp > ${data_dir}/.backup/wav_id
awk '{print $1}' ${data_dir}/text > ${data_dir}/.backup/text_id
sort ${data_dir}/.backup/wav_id ${data_dir}/.backup/text_id | uniq -d > ${data_dir}/.backup/id
cp ${data_dir}/wav.scp ${data_dir}/.backup/wav.scp
cp ${data_dir}/text ${data_dir}/.backup/text
mv ${data_dir}/wav.scp ${data_dir}/wav.scp.bak
mv ${data_dir}/text ${data_dir}/text.bak
utils/filter_scp.pl -f 1 ${data_dir}/.backup/id ${data_dir}/wav.scp.bak | sort -k1,1 -u > ${data_dir}/wav.scp
utils/filter_scp.pl -f 1 ${data_dir}/.backup/id ${data_dir}/text.bak | sort -k1,1 -u > ${data_dir}/text
rm ${data_dir}/wav.scp.bak
rm ${data_dir}/text.bak
@@ -0,0 +1,52 @@
#!/usr/bin/env bash
echo "$0 $@"
data_dir=$1
if [ ! -f ${data_dir}/feats.scp ]; then
echo "$0: feats.scp is not found"
exit 1;
fi
if [ ! -f ${data_dir}/text ]; then
echo "$0: text is not found"
exit 1;
fi
if [ ! -f ${data_dir}/speech_shape ]; then
echo "$0: feature lengths is not found"
exit 1;
fi
if [ ! -f ${data_dir}/text_shape ]; then
echo "$0: text lengths is not found"
exit 1;
fi
mkdir -p ${data_dir}/.backup
awk '{print $1}' ${data_dir}/feats.scp > ${data_dir}/.backup/wav_id
awk '{print $1}' ${data_dir}/text > ${data_dir}/.backup/text_id
sort ${data_dir}/.backup/wav_id ${data_dir}/.backup/text_id | uniq -d > ${data_dir}/.backup/id
cp ${data_dir}/feats.scp ${data_dir}/.backup/feats.scp
cp ${data_dir}/text ${data_dir}/.backup/text
cp ${data_dir}/speech_shape ${data_dir}/.backup/speech_shape
cp ${data_dir}/text_shape ${data_dir}/.backup/text_shape
mv ${data_dir}/feats.scp ${data_dir}/feats.scp.bak
mv ${data_dir}/text ${data_dir}/text.bak
mv ${data_dir}/speech_shape ${data_dir}/speech_shape.bak
mv ${data_dir}/text_shape ${data_dir}/text_shape.bak
utils/filter_scp.pl -f 1 ${data_dir}/.backup/id ${data_dir}/feats.scp.bak | sort -k1,1 -u > ${data_dir}/feats.scp
utils/filter_scp.pl -f 1 ${data_dir}/.backup/id ${data_dir}/text.bak | sort -k1,1 -u > ${data_dir}/text
utils/filter_scp.pl -f 1 ${data_dir}/.backup/id ${data_dir}/speech_shape.bak | sort -k1,1 -u > ${data_dir}/speech_shape
utils/filter_scp.pl -f 1 ${data_dir}/.backup/id ${data_dir}/text_shape.bak | sort -k1,1 -u > ${data_dir}/text_shape
rm ${data_dir}/feats.scp.bak
rm ${data_dir}/text.bak
rm ${data_dir}/speech_shape.bak
rm ${data_dir}/text_shape.bak
@@ -0,0 +1,97 @@
#!/usr/bin/env bash
# Copyright 2012 Johns Hopkins University (Author: Daniel Povey);
# Arnab Ghoshal, Karel Vesely
# Licensed under the Apache License, Version 2.0 (the "License");
# you may not use this file except in compliance with the License.
# You may obtain a copy of the License at
#
# http://www.apache.org/licenses/LICENSE-2.0
#
# THIS CODE IS PROVIDED *AS IS* BASIS, WITHOUT WARRANTIES OR CONDITIONS OF ANY
# KIND, EITHER EXPRESS OR IMPLIED, INCLUDING WITHOUT LIMITATION ANY IMPLIED
# WARRANTIES OR CONDITIONS OF TITLE, FITNESS FOR A PARTICULAR PURPOSE,
# MERCHANTABLITY OR NON-INFRINGEMENT.
# See the Apache 2 License for the specific language governing permissions and
# limitations under the License.
# Parse command-line options.
# To be sourced by another script (as in ". parse_options.sh").
# Option format is: --option-name arg
# and shell variable "option_name" gets set to value "arg."
# The exception is --help, which takes no arguments, but prints the
# $help_message variable (if defined).
###
### The --config file options have lower priority to command line
### options, so we need to import them first...
###
# Now import all the configs specified by command-line, in left-to-right order
for ((argpos=1; argpos<$#; argpos++)); do
if [ "${!argpos}" == "--config" ]; then
argpos_plus1=$((argpos+1))
config=${!argpos_plus1}
[ ! -r $config ] && echo "$0: missing config '$config'" && exit 1
. $config # source the config file.
fi
done
###
### Now we process the command line options
###
while true; do
[ -z "${1:-}" ] && break; # break if there are no arguments
case "$1" in
# If the enclosing script is called with --help option, print the help
# message and exit. Scripts should put help messages in $help_message
--help|-h) if [ -z "$help_message" ]; then echo "No help found." 1>&2;
else printf "$help_message\n" 1>&2 ; fi;
exit 0 ;;
--*=*) echo "$0: options to scripts must be of the form --name value, got '$1'"
exit 1 ;;
# If the first command-line argument begins with "--" (e.g. --foo-bar),
# then work out the variable name as $name, which will equal "foo_bar".
--*) name=`echo "$1" | sed s/^--// | sed s/-/_/g`;
# Next we test whether the variable in question is undefned-- if so it's
# an invalid option and we die. Note: $0 evaluates to the name of the
# enclosing script.
# The test [ -z ${foo_bar+xxx} ] will return true if the variable foo_bar
# is undefined. We then have to wrap this test inside "eval" because
# foo_bar is itself inside a variable ($name).
eval '[ -z "${'$name'+xxx}" ]' && echo "$0: invalid option $1" 1>&2 && exit 1;
oldval="`eval echo \\$$name`";
# Work out whether we seem to be expecting a Boolean argument.
if [ "$oldval" == "true" ] || [ "$oldval" == "false" ]; then
was_bool=true;
else
was_bool=false;
fi
# Set the variable to the right value-- the escaped quotes make it work if
# the option had spaces, like --cmd "queue.pl -sync y"
eval $name=\"$2\";
# Check that Boolean-valued arguments are really Boolean.
if $was_bool && [[ "$2" != "true" && "$2" != "false" ]]; then
echo "$0: expected \"true\" or \"false\": $1 $2" 1>&2
exit 1;
fi
shift 2;
;;
*) break;
esac
done
# Check for an empty argument to the --cmd option, which can easily occur as a
# result of scripting errors.
[ ! -z "${cmd+xxx}" ] && [ -z "$cmd" ] && echo "$0: empty argument to --cmd option" 1>&2 && exit 1;
true; # so this script returns exit code 0.
@@ -0,0 +1,30 @@
import sys
import re
in_f = sys.argv[1]
out_f = sys.argv[2]
with open(in_f, "r", encoding="utf-8") as f:
lines = f.readlines()
with open(out_f, "w", encoding="utf-8") as f:
for line in lines:
outs = line.strip().split(" ", 1)
if len(outs) == 2:
idx, text = outs
text = re.sub("</s>", "", text)
text = re.sub("<s>", "", text)
text = re.sub("@@", "", text)
text = re.sub("@", "", text)
text = re.sub("<unk>", "", text)
text = re.sub(" ", "", text)
text = text.lower()
else:
idx = outs[0]
text = " "
text = [x for x in text]
text = " ".join(text)
out = "{} {}\n".format(idx, text)
f.write(out)
@@ -0,0 +1,44 @@
#!/usr/bin/env perl
# Copyright 2013 Johns Hopkins University (author: Daniel Povey)
# Licensed under the Apache License, Version 2.0 (the "License");
# you may not use this file except in compliance with the License.
# You may obtain a copy of the License at
#
# http://www.apache.org/licenses/LICENSE-2.0
#
# THIS CODE IS PROVIDED *AS IS* BASIS, WITHOUT WARRANTIES OR CONDITIONS OF ANY
# KIND, EITHER EXPRESS OR IMPLIED, INCLUDING WITHOUT LIMITATION ANY IMPLIED
# WARRANTIES OR CONDITIONS OF TITLE, FITNESS FOR A PARTICULAR PURPOSE,
# MERCHANTABLITY OR NON-INFRINGEMENT.
# See the Apache 2 License for the specific language governing permissions and
# limitations under the License.
if ($ARGV[0] eq "--srand") {
$n = $ARGV[1];
$n =~ m/\d+/ || die "Bad argument to --srand option: \"$n\"";
srand($ARGV[1]);
shift;
shift;
} else {
srand(0); # Gives inconsistent behavior if we don't seed.
}
if (@ARGV > 1 || $ARGV[0] =~ m/^-.+/) { # >1 args, or an option we
# don't understand.
print "Usage: shuffle_list.pl [--srand N] [input file] > output\n";
print "randomizes the order of lines of input.\n";
exit(1);
}
@lines;
while (<>) {
push @lines, [ (rand(), $_)] ;
}
@lines = sort { $a->[0] cmp $b->[0] } @lines;
foreach $l (@lines) {
print $l->[1];
}
@@ -0,0 +1,246 @@
#!/usr/bin/env perl
# Copyright 2010-2011 Microsoft Corporation
# See ../../COPYING for clarification regarding multiple authors
#
# Licensed under the Apache License, Version 2.0 (the "License");
# you may not use this file except in compliance with the License.
# You may obtain a copy of the License at
#
# http://www.apache.org/licenses/LICENSE-2.0
#
# THIS CODE IS PROVIDED *AS IS* BASIS, WITHOUT WARRANTIES OR CONDITIONS OF ANY
# KIND, EITHER EXPRESS OR IMPLIED, INCLUDING WITHOUT LIMITATION ANY IMPLIED
# WARRANTIES OR CONDITIONS OF TITLE, FITNESS FOR A PARTICULAR PURPOSE,
# MERCHANTABLITY OR NON-INFRINGEMENT.
# See the Apache 2 License for the specific language governing permissions and
# limitations under the License.
# This program splits up any kind of .scp or archive-type file.
# If there is no utt2spk option it will work on any text file and
# will split it up with an approximately equal number of lines in
# each but.
# With the --utt2spk option it will work on anything that has the
# utterance-id as the first entry on each line; the utt2spk file is
# of the form "utterance speaker" (on each line).
# It splits it into equal size chunks as far as it can. If you use the utt2spk
# option it will make sure these chunks coincide with speaker boundaries. In
# this case, if there are more chunks than speakers (and in some other
# circumstances), some of the resulting chunks will be empty and it will print
# an error message and exit with nonzero status.
# You will normally call this like:
# split_scp.pl scp scp.1 scp.2 scp.3 ...
# or
# split_scp.pl --utt2spk=utt2spk scp scp.1 scp.2 scp.3 ...
# Note that you can use this script to split the utt2spk file itself,
# e.g. split_scp.pl --utt2spk=utt2spk utt2spk utt2spk.1 utt2spk.2 ...
# You can also call the scripts like:
# split_scp.pl -j 3 0 scp scp.0
# [note: with this option, it assumes zero-based indexing of the split parts,
# i.e. the second number must be 0 <= n < num-jobs.]
use warnings;
$num_jobs = 0;
$job_id = 0;
$utt2spk_file = "";
$one_based = 0;
for ($x = 1; $x <= 3 && @ARGV > 0; $x++) {
if ($ARGV[0] eq "-j") {
shift @ARGV;
$num_jobs = shift @ARGV;
$job_id = shift @ARGV;
}
if ($ARGV[0] =~ /--utt2spk=(.+)/) {
$utt2spk_file=$1;
shift;
}
if ($ARGV[0] eq '--one-based') {
$one_based = 1;
shift @ARGV;
}
}
if ($num_jobs != 0 && ($num_jobs < 0 || $job_id - $one_based < 0 ||
$job_id - $one_based >= $num_jobs)) {
die "$0: Invalid job number/index values for '-j $num_jobs $job_id" .
($one_based ? " --one-based" : "") . "'\n"
}
$one_based
and $job_id--;
if(($num_jobs == 0 && @ARGV < 2) || ($num_jobs > 0 && (@ARGV < 1 || @ARGV > 2))) {
die
"Usage: split_scp.pl [--utt2spk=<utt2spk_file>] in.scp out1.scp out2.scp ...
or: split_scp.pl -j num-jobs job-id [--one-based] [--utt2spk=<utt2spk_file>] in.scp [out.scp]
... where 0 <= job-id < num-jobs, or 1 <= job-id <- num-jobs if --one-based.\n";
}
$error = 0;
$inscp = shift @ARGV;
if ($num_jobs == 0) { # without -j option
@OUTPUTS = @ARGV;
} else {
for ($j = 0; $j < $num_jobs; $j++) {
if ($j == $job_id) {
if (@ARGV > 0) { push @OUTPUTS, $ARGV[0]; }
else { push @OUTPUTS, "-"; }
} else {
push @OUTPUTS, "/dev/null";
}
}
}
if ($utt2spk_file ne "") { # We have the --utt2spk option...
open($u_fh, '<', $utt2spk_file) || die "$0: Error opening utt2spk file $utt2spk_file: $!\n";
while(<$u_fh>) {
@A = split;
@A == 2 || die "$0: Bad line $_ in utt2spk file $utt2spk_file\n";
($u,$s) = @A;
$utt2spk{$u} = $s;
}
close $u_fh;
open($i_fh, '<', $inscp) || die "$0: Error opening input scp file $inscp: $!\n";
@spkrs = ();
while(<$i_fh>) {
@A = split;
if(@A == 0) { die "$0: Empty or space-only line in scp file $inscp\n"; }
$u = $A[0];
$s = $utt2spk{$u};
defined $s || die "$0: No utterance $u in utt2spk file $utt2spk_file\n";
if(!defined $spk_count{$s}) {
push @spkrs, $s;
$spk_count{$s} = 0;
$spk_data{$s} = []; # ref to new empty array.
}
$spk_count{$s}++;
push @{$spk_data{$s}}, $_;
}
# Now split as equally as possible ..
# First allocate spks to files by allocating an approximately
# equal number of speakers.
$numspks = @spkrs; # number of speakers.
$numscps = @OUTPUTS; # number of output files.
if ($numspks < $numscps) {
die "$0: Refusing to split data because number of speakers $numspks " .
"is less than the number of output .scp files $numscps\n";
}
for($scpidx = 0; $scpidx < $numscps; $scpidx++) {
$scparray[$scpidx] = []; # [] is array reference.
}
for ($spkidx = 0; $spkidx < $numspks; $spkidx++) {
$scpidx = int(($spkidx*$numscps) / $numspks);
$spk = $spkrs[$spkidx];
push @{$scparray[$scpidx]}, $spk;
$scpcount[$scpidx] += $spk_count{$spk};
}
# Now will try to reassign beginning + ending speakers
# to different scp's and see if it gets more balanced.
# Suppose objf we're minimizing is sum_i (num utts in scp[i] - average)^2.
# We can show that if considering changing just 2 scp's, we minimize
# this by minimizing the squared difference in sizes. This is
# equivalent to minimizing the absolute difference in sizes. This
# shows this method is bound to converge.
$changed = 1;
while($changed) {
$changed = 0;
for($scpidx = 0; $scpidx < $numscps; $scpidx++) {
# First try to reassign ending spk of this scp.
if($scpidx < $numscps-1) {
$sz = @{$scparray[$scpidx]};
if($sz > 0) {
$spk = $scparray[$scpidx]->[$sz-1];
$count = $spk_count{$spk};
$nutt1 = $scpcount[$scpidx];
$nutt2 = $scpcount[$scpidx+1];
if( abs( ($nutt2+$count) - ($nutt1-$count))
< abs($nutt2 - $nutt1)) { # Would decrease
# size-diff by reassigning spk...
$scpcount[$scpidx+1] += $count;
$scpcount[$scpidx] -= $count;
pop @{$scparray[$scpidx]};
unshift @{$scparray[$scpidx+1]}, $spk;
$changed = 1;
}
}
}
if($scpidx > 0 && @{$scparray[$scpidx]} > 0) {
$spk = $scparray[$scpidx]->[0];
$count = $spk_count{$spk};
$nutt1 = $scpcount[$scpidx-1];
$nutt2 = $scpcount[$scpidx];
if( abs( ($nutt2-$count) - ($nutt1+$count))
< abs($nutt2 - $nutt1)) { # Would decrease
# size-diff by reassigning spk...
$scpcount[$scpidx-1] += $count;
$scpcount[$scpidx] -= $count;
shift @{$scparray[$scpidx]};
push @{$scparray[$scpidx-1]}, $spk;
$changed = 1;
}
}
}
}
# Now print out the files...
for($scpidx = 0; $scpidx < $numscps; $scpidx++) {
$scpfile = $OUTPUTS[$scpidx];
($scpfile ne '-' ? open($f_fh, '>', $scpfile)
: open($f_fh, '>&', \*STDOUT)) ||
die "$0: Could not open scp file $scpfile for writing: $!\n";
$count = 0;
if(@{$scparray[$scpidx]} == 0) {
print STDERR "$0: eError: split_scp.pl producing empty .scp file " .
"$scpfile (too many splits and too few speakers?)\n";
$error = 1;
} else {
foreach $spk ( @{$scparray[$scpidx]} ) {
print $f_fh @{$spk_data{$spk}};
$count += $spk_count{$spk};
}
$count == $scpcount[$scpidx] || die "Count mismatch [code error]";
}
close($f_fh);
}
} else {
# This block is the "normal" case where there is no --utt2spk
# option and we just break into equal size chunks.
open($i_fh, '<', $inscp) || die "$0: Error opening input scp file $inscp: $!\n";
$numscps = @OUTPUTS; # size of array.
@F = ();
while(<$i_fh>) {
push @F, $_;
}
$numlines = @F;
if($numlines == 0) {
print STDERR "$0: error: empty input scp file $inscp\n";
$error = 1;
}
$linesperscp = int( $numlines / $numscps); # the "whole part"..
$linesperscp >= 1 || die "$0: You are splitting into too many pieces! [reduce \$nj ($numscps) to be smaller than the number of lines ($numlines) in $inscp]\n";
$remainder = $numlines - ($linesperscp * $numscps);
($remainder >= 0 && $remainder < $numlines) || die "bad remainder $remainder";
# [just doing int() rounds down].
$n = 0;
for($scpidx = 0; $scpidx < @OUTPUTS; $scpidx++) {
$scpfile = $OUTPUTS[$scpidx];
($scpfile ne '-' ? open($o_fh, '>', $scpfile)
: open($o_fh, '>&', \*STDOUT)) ||
die "$0: Could not open scp file $scpfile for writing: $!\n";
for($k = 0; $k < $linesperscp + ($scpidx < $remainder ? 1 : 0); $k++) {
print $o_fh $F[$n++];
}
close($o_fh) || die "$0: Eror closing scp file $scpfile: $!\n";
}
$n == $numlines || die "$n != $numlines [code error]";
}
exit ($error);
@@ -0,0 +1,131 @@
#!/usr/bin/env python3
# Copyright 2017 Johns Hopkins University (Shinji Watanabe)
# Apache 2.0 (http://www.apache.org/licenses/LICENSE-2.0)
import argparse
import codecs
import re
import sys
is_python2 = sys.version_info[0] == 2
def exist_or_not(i, match_pos):
start_pos = None
end_pos = None
for pos in match_pos:
if pos[0] <= i < pos[1]:
start_pos = pos[0]
end_pos = pos[1]
break
return start_pos, end_pos
def get_parser():
parser = argparse.ArgumentParser(
description="convert raw text to tokenized text",
formatter_class=argparse.ArgumentDefaultsHelpFormatter,
)
parser.add_argument(
"--nchar",
"-n",
default=1,
type=int,
help="number of characters to split, i.e., \
aabb -> a a b b with -n 1 and aa bb with -n 2",
)
parser.add_argument("--skip-ncols", "-s", default=0, type=int, help="skip first n columns")
parser.add_argument("--space", default="<space>", type=str, help="space symbol")
parser.add_argument(
"--non-lang-syms",
"-l",
default=None,
type=str,
help="list of non-linguistic symobles, e.g., <NOISE> etc.",
)
parser.add_argument("text", type=str, default=False, nargs="?", help="input text")
parser.add_argument(
"--trans_type",
"-t",
type=str,
default="char",
choices=["char", "phn"],
help="""Transcript type. char/phn. e.g., for TIMIT FADG0_SI1279 -
If trans_type is char,
read from SI1279.WRD file -> "bricks are an alternative"
Else if trans_type is phn,
read from SI1279.PHN file -> "sil b r ih sil k s aa r er n aa l
sil t er n ih sil t ih v sil" """,
)
return parser
def main():
parser = get_parser()
args = parser.parse_args()
rs = []
if args.non_lang_syms is not None:
with codecs.open(args.non_lang_syms, "r", encoding="utf-8") as f:
nls = [x.rstrip() for x in f.readlines()]
rs = [re.compile(re.escape(x)) for x in nls]
if args.text:
f = codecs.open(args.text, encoding="utf-8")
else:
f = codecs.getreader("utf-8")(sys.stdin if is_python2 else sys.stdin.buffer)
sys.stdout = codecs.getwriter("utf-8")(sys.stdout if is_python2 else sys.stdout.buffer)
line = f.readline()
n = args.nchar
while line:
x = line.split()
print(" ".join(x[: args.skip_ncols]), end=" ")
a = " ".join(x[args.skip_ncols :])
# get all matched positions
match_pos = []
for r in rs:
i = 0
while i >= 0:
m = r.search(a, i)
if m:
match_pos.append([m.start(), m.end()])
i = m.end()
else:
break
if args.trans_type == "phn":
a = a.split(" ")
else:
if len(match_pos) > 0:
chars = []
i = 0
while i < len(a):
start_pos, end_pos = exist_or_not(i, match_pos)
if start_pos is not None:
chars.append(a[start_pos:end_pos])
i = end_pos
else:
chars.append(a[i])
i += 1
a = chars
a = [a[j : j + n] for j in range(0, len(a), n)]
a_flat = []
for z in a:
a_flat.append("".join(z))
a_chars = [z.replace(" ", args.space) for z in a_flat]
if args.trans_type == "phn":
a_chars = [z.replace("sil", args.space) for z in a_chars]
print(" ".join(a_chars))
line = f.readline()
if __name__ == "__main__":
main()
@@ -0,0 +1,104 @@
import re
import argparse
def load_dict(seg_file):
seg_dict = {}
with open(seg_file, "r") as infile:
for line in infile:
s = line.strip().split()
key = s[0]
value = s[1:]
seg_dict[key] = " ".join(value)
return seg_dict
def forward_segment(text, dic):
word_list = []
i = 0
while i < len(text):
longest_word = text[i]
for j in range(i + 1, len(text) + 1):
word = text[i:j]
if word in dic:
if len(word) > len(longest_word):
longest_word = word
word_list.append(longest_word)
i += len(longest_word)
return word_list
def tokenize(txt, seg_dict):
out_txt = ""
pattern = re.compile(r"([\u4E00-\u9FA5A-Za-z0-9])")
for word in txt:
if pattern.match(word):
if word in seg_dict:
out_txt += seg_dict[word] + " "
else:
out_txt += "<unk>" + " "
else:
continue
return out_txt.strip()
def get_parser():
parser = argparse.ArgumentParser(
description="text tokenize",
formatter_class=argparse.ArgumentDefaultsHelpFormatter,
)
parser.add_argument(
"--text-file",
"-t",
default=False,
required=True,
type=str,
help="input text",
)
parser.add_argument(
"--seg-file",
"-s",
default=False,
required=True,
type=str,
help="seg file",
)
parser.add_argument(
"--txt-index",
"-i",
default=1,
required=True,
type=int,
help="txt index",
)
parser.add_argument(
"--output-dir",
"-o",
default=False,
required=True,
type=str,
help="output dir",
)
return parser
def main():
parser = get_parser()
args = parser.parse_args()
txt_writer = open("{}/text.{}.txt".format(args.output_dir, args.txt_index), "w")
shape_writer = open("{}/len.{}".format(args.output_dir, args.txt_index), "w")
seg_dict = load_dict(args.seg_file)
with open(args.text_file, "r") as infile:
for line in infile:
s = line.strip().split()
text_id = s[0]
text_list = forward_segment("".join(s[1:]).lower(), seg_dict)
text = tokenize(text_list, seg_dict)
lens = len(text.strip().split())
txt_writer.write(text_id + " " + text + "\n")
shape_writer.write(text_id + " " + str(lens) + "\n")
if __name__ == "__main__":
main()
@@ -0,0 +1,35 @@
#!/usr/bin/env bash
# Begin configuration section.
nj=32
cmd=utils/run.pl
echo "$0 $@"
. utils/parse_options.sh || exit 1;
# tokenize configuration
text_dir=$1
seg_file=$2
logdir=$3
output_dir=$4
txt_dir=${output_dir}/txt; mkdir -p ${output_dir}/txt
mkdir -p ${logdir}
$cmd JOB=1:$nj $logdir/text_tokenize.JOB.log \
python utils/text_tokenize.py -t ${text_dir}/txt/text.JOB.txt \
-s ${seg_file} -i JOB -o ${txt_dir} \
|| exit 1;
# concatenate the text files together.
for n in $(seq $nj); do
cat ${txt_dir}/text.$n.txt || exit 1
done > ${output_dir}/text || exit 1
for n in $(seq $nj); do
cat ${txt_dir}/len.$n || exit 1
done > ${output_dir}/text_shape || exit 1
echo "$0: Succeeded text tokenize"
@@ -0,0 +1,911 @@
#!/usr/bin/env python3
# coding=utf-8
# Authors:
# 2019.5 Zhiyang Zhou (https://github.com/Joee1995/chn_text_norm.git)
# 2019.9 Jiayu DU
#
# requirements:
# - python 3.X
# notes: python 2.X WILL fail or produce misleading results
import sys, os, argparse, codecs, string, re
# ================================================================================ #
# basic constant
# ================================================================================ #
CHINESE_DIGIS = "零一二三四五六七八九"
BIG_CHINESE_DIGIS_SIMPLIFIED = "零壹贰叁肆伍陆柒捌玖"
BIG_CHINESE_DIGIS_TRADITIONAL = "零壹貳參肆伍陸柒捌玖"
SMALLER_BIG_CHINESE_UNITS_SIMPLIFIED = "十百千万"
SMALLER_BIG_CHINESE_UNITS_TRADITIONAL = "拾佰仟萬"
LARGER_CHINESE_NUMERING_UNITS_SIMPLIFIED = "亿兆京垓秭穰沟涧正载"
LARGER_CHINESE_NUMERING_UNITS_TRADITIONAL = "億兆京垓秭穰溝澗正載"
SMALLER_CHINESE_NUMERING_UNITS_SIMPLIFIED = "十百千万"
SMALLER_CHINESE_NUMERING_UNITS_TRADITIONAL = "拾佰仟萬"
ZERO_ALT = "〇"
ONE_ALT = "幺"
TWO_ALTS = ["两", "兩"]
POSITIVE = ["正", "正"]
NEGATIVE = ["负", "負"]
POINT = ["点", "點"]
# PLUS = [u'加', u'加']
# SIL = [u'杠', u'槓']
FILLER_CHARS = ["呃", "啊"]
ER_WHITELIST = (
"(儿女|儿子|儿孙|女儿|儿媳|妻儿|"
"胎儿|婴儿|新生儿|婴幼儿|幼儿|少儿|小儿|儿歌|儿童|儿科|托儿所|孤儿|"
"儿戏|儿化|台儿庄|鹿儿岛|正儿八经|吊儿郎当|生儿育女|托儿带女|养儿防老|痴儿呆女|"
"佳儿佳妇|儿怜兽扰|儿无常父|儿不嫌母丑|儿行千里母担忧|儿大不由爷|苏乞儿)"
)
# 中文数字系统类型
NUMBERING_TYPES = ["low", "mid", "high"]
CURRENCY_NAMES = (
"(人民币|美元|日元|英镑|欧元|马克|法郎|加拿大元|澳元|港币|先令|芬兰马克|爱尔兰镑|"
"里拉|荷兰盾|埃斯库多|比塞塔|印尼盾|林吉特|新西兰元|比索|卢布|新加坡元|韩元|泰铢)"
)
CURRENCY_UNITS = (
"((亿|千万|百万|万|千|百)|(亿|千万|百万|万|千|百|)元|(亿|千万|百万|万|千|百|)块|角|毛|分)"
)
COM_QUANTIFIERS = (
"(匹|张|座|回|场|尾|条|个|首|阙|阵|网|炮|顶|丘|棵|只|支|袭|辆|挑|担|颗|壳|窠|曲|墙|群|腔|"
"砣|座|客|贯|扎|捆|刀|令|打|手|罗|坡|山|岭|江|溪|钟|队|单|双|对|出|口|头|脚|板|跳|枝|件|贴|"
"针|线|管|名|位|身|堂|课|本|页|家|户|层|丝|毫|厘|分|钱|两|斤|担|铢|石|钧|锱|忽|(千|毫|微)克|"
"毫|厘|分|寸|尺|丈|里|寻|常|铺|程|(千|分|厘|毫|微)米|撮|勺|合|升|斗|石|盘|碗|碟|叠|桶|笼|盆|"
"盒|杯|钟|斛|锅|簋|篮|盘|桶|罐|瓶|壶|卮|盏|箩|箱|煲|啖|袋|钵|年|月|日|季|刻|时|周|天|秒|分|旬|"
"纪|岁|世|更|夜|春|夏|秋|冬|代|伏|辈|丸|泡|粒|颗|幢|堆|条|根|支|道|面|片|张|颗|块)"
)
# punctuation information are based on Zhon project (https://github.com/tsroten/zhon.git)
CHINESE_PUNC_STOP = "!?。。"
CHINESE_PUNC_NON_STOP = ""#$%&'()*+,-/:;<=>@[\]^_`{|}~⦅⦆「」、、〃》「」『』【】〔〕〖〗〘〙〚〛〜〝〞〟〰〾〿–—‘’‛“”„‟…‧﹏"
CHINESE_PUNC_LIST = CHINESE_PUNC_STOP + CHINESE_PUNC_NON_STOP
# ================================================================================ #
# basic class
# ================================================================================ #
class ChineseChar(object):
"""
中文字符
每个字符对应简体和繁体,
e.g. 简体 = '负', 繁体 = '負'
转换时可转换为简体或繁体
"""
def __init__(self, simplified, traditional):
self.simplified = simplified
self.traditional = traditional
# self.__repr__ = self.__str__
def __str__(self):
return self.simplified or self.traditional or None
def __repr__(self):
return self.__str__()
class ChineseNumberUnit(ChineseChar):
"""
中文数字/数位字符
每个字符除繁简体外还有一个额外的大写字符
e.g. '陆' 和 '陸'
"""
def __init__(self, power, simplified, traditional, big_s, big_t):
super(ChineseNumberUnit, self).__init__(simplified, traditional)
self.power = power
self.big_s = big_s
self.big_t = big_t
def __str__(self):
return "10^{}".format(self.power)
@classmethod
def create(cls, index, value, numbering_type=NUMBERING_TYPES[1], small_unit=False):
if small_unit:
return ChineseNumberUnit(
power=index + 1,
simplified=value[0],
traditional=value[1],
big_s=value[1],
big_t=value[1],
)
elif numbering_type == NUMBERING_TYPES[0]:
return ChineseNumberUnit(
power=index + 8,
simplified=value[0],
traditional=value[1],
big_s=value[0],
big_t=value[1],
)
elif numbering_type == NUMBERING_TYPES[1]:
return ChineseNumberUnit(
power=(index + 2) * 4,
simplified=value[0],
traditional=value[1],
big_s=value[0],
big_t=value[1],
)
elif numbering_type == NUMBERING_TYPES[2]:
return ChineseNumberUnit(
power=pow(2, index + 3),
simplified=value[0],
traditional=value[1],
big_s=value[0],
big_t=value[1],
)
else:
raise ValueError(
"Counting type should be in {0} ({1} provided).".format(
NUMBERING_TYPES, numbering_type
)
)
class ChineseNumberDigit(ChineseChar):
"""
中文数字字符
"""
def __init__(self, value, simplified, traditional, big_s, big_t, alt_s=None, alt_t=None):
super(ChineseNumberDigit, self).__init__(simplified, traditional)
self.value = value
self.big_s = big_s
self.big_t = big_t
self.alt_s = alt_s
self.alt_t = alt_t
def __str__(self):
return str(self.value)
@classmethod
def create(cls, i, v):
return ChineseNumberDigit(i, v[0], v[1], v[2], v[3])
class ChineseMath(ChineseChar):
"""
中文数位字符
"""
def __init__(self, simplified, traditional, symbol, expression=None):
super(ChineseMath, self).__init__(simplified, traditional)
self.symbol = symbol
self.expression = expression
self.big_s = simplified
self.big_t = traditional
CC, CNU, CND, CM = ChineseChar, ChineseNumberUnit, ChineseNumberDigit, ChineseMath
class NumberSystem(object):
"""
中文数字系统
"""
pass
class MathSymbol(object):
"""
用于中文数字系统的数学符号 (繁/简体), e.g.
positive = ['正', '正']
negative = ['负', '負']
point = ['点', '點']
"""
def __init__(self, positive, negative, point):
self.positive = positive
self.negative = negative
self.point = point
def __iter__(self):
for v in self.__dict__.values():
yield v
# class OtherSymbol(object):
# """
# 其他符号
# """
#
# def __init__(self, sil):
# self.sil = sil
#
# def __iter__(self):
# for v in self.__dict__.values():
# yield v
# ================================================================================ #
# basic utils
# ================================================================================ #
def create_system(numbering_type=NUMBERING_TYPES[1]):
"""
根据数字系统类型返回创建相应的数字系统,默认为 mid
NUMBERING_TYPES = ['low', 'mid', 'high']: 中文数字系统类型
low: '兆' = '亿' * '十' = $10^{9}$, '京' = '兆' * '十', etc.
mid: '兆' = '亿' * '万' = $10^{12}$, '京' = '兆' * '万', etc.
high: '兆' = '亿' * '亿' = $10^{16}$, '京' = '兆' * '兆', etc.
返回对应的数字系统
"""
# chinese number units of '亿' and larger
all_larger_units = zip(
LARGER_CHINESE_NUMERING_UNITS_SIMPLIFIED, LARGER_CHINESE_NUMERING_UNITS_TRADITIONAL
)
larger_units = [CNU.create(i, v, numbering_type, False) for i, v in enumerate(all_larger_units)]
# chinese number units of '十, 百, 千, 万'
all_smaller_units = zip(
SMALLER_CHINESE_NUMERING_UNITS_SIMPLIFIED, SMALLER_CHINESE_NUMERING_UNITS_TRADITIONAL
)
smaller_units = [CNU.create(i, v, small_unit=True) for i, v in enumerate(all_smaller_units)]
# digis
chinese_digis = zip(
CHINESE_DIGIS, CHINESE_DIGIS, BIG_CHINESE_DIGIS_SIMPLIFIED, BIG_CHINESE_DIGIS_TRADITIONAL
)
digits = [CND.create(i, v) for i, v in enumerate(chinese_digis)]
digits[0].alt_s, digits[0].alt_t = ZERO_ALT, ZERO_ALT
digits[1].alt_s, digits[1].alt_t = ONE_ALT, ONE_ALT
digits[2].alt_s, digits[2].alt_t = TWO_ALTS[0], TWO_ALTS[1]
# symbols
positive_cn = CM(POSITIVE[0], POSITIVE[1], "+", lambda x: x)
negative_cn = CM(NEGATIVE[0], NEGATIVE[1], "-", lambda x: -x)
point_cn = CM(POINT[0], POINT[1], ".", lambda x, y: float(str(x) + "." + str(y)))
# sil_cn = CM(SIL[0], SIL[1], '-', lambda x, y: float(str(x) + '-' + str(y)))
system = NumberSystem()
system.units = smaller_units + larger_units
system.digits = digits
system.math = MathSymbol(positive_cn, negative_cn, point_cn)
# system.symbols = OtherSymbol(sil_cn)
return system
def chn2num(chinese_string, numbering_type=NUMBERING_TYPES[1]):
def get_symbol(char, system):
for u in system.units:
if char in [u.traditional, u.simplified, u.big_s, u.big_t]:
return u
for d in system.digits:
if char in [d.traditional, d.simplified, d.big_s, d.big_t, d.alt_s, d.alt_t]:
return d
for m in system.math:
if char in [m.traditional, m.simplified]:
return m
def string2symbols(chinese_string, system):
int_string, dec_string = chinese_string, ""
for p in [system.math.point.simplified, system.math.point.traditional]:
if p in chinese_string:
int_string, dec_string = chinese_string.split(p)
break
return [get_symbol(c, system) for c in int_string], [
get_symbol(c, system) for c in dec_string
]
def correct_symbols(integer_symbols, system):
"""
一百八 to 一百八十
一亿一千三百万 to 一亿 一千万 三百万
"""
if integer_symbols and isinstance(integer_symbols[0], CNU):
if integer_symbols[0].power == 1:
integer_symbols = [system.digits[1]] + integer_symbols
if len(integer_symbols) > 1:
if isinstance(integer_symbols[-1], CND) and isinstance(integer_symbols[-2], CNU):
integer_symbols.append(CNU(integer_symbols[-2].power - 1, None, None, None, None))
result = []
unit_count = 0
for s in integer_symbols:
if isinstance(s, CND):
result.append(s)
unit_count = 0
elif isinstance(s, CNU):
current_unit = CNU(s.power, None, None, None, None)
unit_count += 1
if unit_count == 1:
result.append(current_unit)
elif unit_count > 1:
for i in range(len(result)):
if (
isinstance(result[-i - 1], CNU)
and result[-i - 1].power < current_unit.power
):
result[-i - 1] = CNU(
result[-i - 1].power + current_unit.power, None, None, None, None
)
return result
def compute_value(integer_symbols):
"""
Compute the value.
When current unit is larger than previous unit, current unit * all previous units will be used as all previous units.
e.g. '两千万' = 2000 * 10000 not 2000 + 10000
"""
value = [0]
last_power = 0
for s in integer_symbols:
if isinstance(s, CND):
value[-1] = s.value
elif isinstance(s, CNU):
value[-1] *= pow(10, s.power)
if s.power > last_power:
value[:-1] = list(map(lambda v: v * pow(10, s.power), value[:-1]))
last_power = s.power
value.append(0)
return sum(value)
system = create_system(numbering_type)
int_part, dec_part = string2symbols(chinese_string, system)
int_part = correct_symbols(int_part, system)
int_str = str(compute_value(int_part))
dec_str = "".join([str(d.value) for d in dec_part])
if dec_part:
return "{0}.{1}".format(int_str, dec_str)
else:
return int_str
def num2chn(
number_string,
numbering_type=NUMBERING_TYPES[1],
big=False,
traditional=False,
alt_zero=False,
alt_one=False,
alt_two=True,
use_zeros=True,
use_units=True,
):
def get_value(value_string, use_zeros=True):
striped_string = value_string.lstrip("0")
# record nothing if all zeros
if not striped_string:
return []
# record one digits
elif len(striped_string) == 1:
if use_zeros and len(value_string) != len(striped_string):
return [system.digits[0], system.digits[int(striped_string)]]
else:
return [system.digits[int(striped_string)]]
# recursively record multiple digits
else:
result_unit = next(u for u in reversed(system.units) if u.power < len(striped_string))
result_string = value_string[: -result_unit.power]
return (
get_value(result_string)
+ [result_unit]
+ get_value(striped_string[-result_unit.power :])
)
system = create_system(numbering_type)
int_dec = number_string.split(".")
if len(int_dec) == 1:
int_string = int_dec[0]
dec_string = ""
elif len(int_dec) == 2:
int_string = int_dec[0]
dec_string = int_dec[1]
else:
raise ValueError(
"invalid input num string with more than one dot: {}".format(number_string)
)
if use_units and len(int_string) > 1:
result_symbols = get_value(int_string)
else:
result_symbols = [system.digits[int(c)] for c in int_string]
dec_symbols = [system.digits[int(c)] for c in dec_string]
if dec_string:
result_symbols += [system.math.point] + dec_symbols
if alt_two:
liang = CND(
2,
system.digits[2].alt_s,
system.digits[2].alt_t,
system.digits[2].big_s,
system.digits[2].big_t,
)
for i, v in enumerate(result_symbols):
if isinstance(v, CND) and v.value == 2:
next_symbol = result_symbols[i + 1] if i < len(result_symbols) - 1 else None
previous_symbol = result_symbols[i - 1] if i > 0 else None
if isinstance(next_symbol, CNU) and isinstance(previous_symbol, (CNU, type(None))):
if next_symbol.power != 1 and (
(previous_symbol is None) or (previous_symbol.power != 1)
):
result_symbols[i] = liang
# if big is True, '两' will not be used and `alt_two` has no impact on output
if big:
attr_name = "big_"
if traditional:
attr_name += "t"
else:
attr_name += "s"
else:
if traditional:
attr_name = "traditional"
else:
attr_name = "simplified"
result = "".join([getattr(s, attr_name) for s in result_symbols])
# if not use_zeros:
# result = result.strip(getattr(system.digits[0], attr_name))
if alt_zero:
result = result.replace(getattr(system.digits[0], attr_name), system.digits[0].alt_s)
if alt_one:
result = result.replace(getattr(system.digits[1], attr_name), system.digits[1].alt_s)
for i, p in enumerate(POINT):
if result.startswith(p):
return CHINESE_DIGIS[0] + result
# ^10, 11, .., 19
if (
len(result) >= 2
and result[1]
in [
SMALLER_CHINESE_NUMERING_UNITS_SIMPLIFIED[0],
SMALLER_CHINESE_NUMERING_UNITS_TRADITIONAL[0],
]
and result[0]
in [CHINESE_DIGIS[1], BIG_CHINESE_DIGIS_SIMPLIFIED[1], BIG_CHINESE_DIGIS_TRADITIONAL[1]]
):
result = result[1:]
return result
# ================================================================================ #
# different types of rewriters
# ================================================================================ #
class Cardinal:
"""
CARDINAL类
"""
def __init__(self, cardinal=None, chntext=None):
self.cardinal = cardinal
self.chntext = chntext
def chntext2cardinal(self):
return chn2num(self.chntext)
def cardinal2chntext(self):
return num2chn(self.cardinal)
class Digit:
"""
DIGIT类
"""
def __init__(self, digit=None, chntext=None):
self.digit = digit
self.chntext = chntext
# def chntext2digit(self):
# return chn2num(self.chntext)
def digit2chntext(self):
return num2chn(self.digit, alt_two=False, use_units=False)
class TelePhone:
"""
TELEPHONE类
"""
def __init__(self, telephone=None, raw_chntext=None, chntext=None):
self.telephone = telephone
self.raw_chntext = raw_chntext
self.chntext = chntext
# def chntext2telephone(self):
# sil_parts = self.raw_chntext.split('<SIL>')
# self.telephone = '-'.join([
# str(chn2num(p)) for p in sil_parts
# ])
# return self.telephone
def telephone2chntext(self, fixed=False):
if fixed:
sil_parts = self.telephone.split("-")
self.raw_chntext = "<SIL>".join(
[num2chn(part, alt_two=False, use_units=False) for part in sil_parts]
)
self.chntext = self.raw_chntext.replace("<SIL>", "")
else:
sp_parts = self.telephone.strip("+").split()
self.raw_chntext = "<SP>".join(
[num2chn(part, alt_two=False, use_units=False) for part in sp_parts]
)
self.chntext = self.raw_chntext.replace("<SP>", "")
return self.chntext
class Fraction:
"""
FRACTION类
"""
def __init__(self, fraction=None, chntext=None):
self.fraction = fraction
self.chntext = chntext
def chntext2fraction(self):
denominator, numerator = self.chntext.split("分之")
return chn2num(numerator) + "/" + chn2num(denominator)
def fraction2chntext(self):
numerator, denominator = self.fraction.split("/")
return num2chn(denominator) + "分之" + num2chn(numerator)
class Date:
"""
DATE类
"""
def __init__(self, date=None, chntext=None):
self.date = date
self.chntext = chntext
# def chntext2date(self):
# chntext = self.chntext
# try:
# year, other = chntext.strip().split('年', maxsplit=1)
# year = Digit(chntext=year).digit2chntext() + '年'
# except ValueError:
# other = chntext
# year = ''
# if other:
# try:
# month, day = other.strip().split('月', maxsplit=1)
# month = Cardinal(chntext=month).chntext2cardinal() + '月'
# except ValueError:
# day = chntext
# month = ''
# if day:
# day = Cardinal(chntext=day[:-1]).chntext2cardinal() + day[-1]
# else:
# month = ''
# day = ''
# date = year + month + day
# self.date = date
# return self.date
def date2chntext(self):
date = self.date
try:
year, other = date.strip().split("年", 1)
year = Digit(digit=year).digit2chntext() + "年"
except ValueError:
other = date
year = ""
if other:
try:
month, day = other.strip().split("月", 1)
month = Cardinal(cardinal=month).cardinal2chntext() + "月"
except ValueError:
day = date
month = ""
if day:
day = Cardinal(cardinal=day[:-1]).cardinal2chntext() + day[-1]
else:
month = ""
day = ""
chntext = year + month + day
self.chntext = chntext
return self.chntext
class Money:
"""
MONEY类
"""
def __init__(self, money=None, chntext=None):
self.money = money
self.chntext = chntext
# def chntext2money(self):
# return self.money
def money2chntext(self):
money = self.money
pattern = re.compile(r"(\d+(\.\d+)?)")
matchers = pattern.findall(money)
if matchers:
for matcher in matchers:
money = money.replace(matcher[0], Cardinal(cardinal=matcher[0]).cardinal2chntext())
self.chntext = money
return self.chntext
class Percentage:
"""
PERCENTAGE类
"""
def __init__(self, percentage=None, chntext=None):
self.percentage = percentage
self.chntext = chntext
def chntext2percentage(self):
return chn2num(self.chntext.strip().strip("百分之")) + "%"
def percentage2chntext(self):
return "百分之" + num2chn(self.percentage.strip().strip("%"))
def remove_erhua(text, er_whitelist):
"""
去除儿化音词中的儿:
他女儿在那边儿 -> 他女儿在那边
"""
er_pattern = re.compile(er_whitelist)
new_str = ""
while re.search("儿", text):
a = re.search("儿", text).span()
remove_er_flag = 0
if er_pattern.search(text):
b = er_pattern.search(text).span()
if b[0] <= a[0]:
remove_er_flag = 1
if remove_er_flag == 0:
new_str = new_str + text[0 : a[0]]
text = text[a[1] :]
else:
new_str = new_str + text[0 : b[1]]
text = text[b[1] :]
text = new_str + text
return text
# ================================================================================ #
# NSW Normalizer
# ================================================================================ #
class NSWNormalizer:
def __init__(self, raw_text):
self.raw_text = "^" + raw_text + "$"
self.norm_text = ""
def _particular(self):
text = self.norm_text
pattern = re.compile(r"(([a-zA-Z]+)二([a-zA-Z]+))")
matchers = pattern.findall(text)
if matchers:
# print('particular')
for matcher in matchers:
text = text.replace(matcher[0], matcher[1] + "2" + matcher[2], 1)
self.norm_text = text
return self.norm_text
def normalize(self):
text = self.raw_text
# 规范化日期
pattern = re.compile(r"\D+((([089]\d|(19|20)\d{2})年)?(\d{1,2}月(\d{1,2}[日号])?)?)")
matchers = pattern.findall(text)
if matchers:
# print('date')
for matcher in matchers:
text = text.replace(matcher[0], Date(date=matcher[0]).date2chntext(), 1)
# 规范化金钱
pattern = re.compile(
r"\D+((\d+(\.\d+)?)[多余几]?" + CURRENCY_UNITS + r"(\d" + CURRENCY_UNITS + r"?)?)"
)
matchers = pattern.findall(text)
if matchers:
# print('money')
for matcher in matchers:
text = text.replace(matcher[0], Money(money=matcher[0]).money2chntext(), 1)
# 规范化固话/手机号码
# 手机
# http://www.jihaoba.com/news/show/13680
# 移动:139、138、137、136、135、134、159、158、157、150、151、152、188、187、182、183、184、178、198
# 联通:130、131、132、156、155、186、185、176
# 电信:133、153、189、180、181、177
pattern = re.compile(r"\D((\+?86 ?)?1([38]\d|5[0-35-9]|7[678]|9[89])\d{8})\D")
matchers = pattern.findall(text)
if matchers:
# print('telephone')
for matcher in matchers:
text = text.replace(
matcher[0], TelePhone(telephone=matcher[0]).telephone2chntext(), 1
)
# 固话
pattern = re.compile(r"\D((0(10|2[1-3]|[3-9]\d{2})-?)?[1-9]\d{6,7})\D")
matchers = pattern.findall(text)
if matchers:
# print('fixed telephone')
for matcher in matchers:
text = text.replace(
matcher[0], TelePhone(telephone=matcher[0]).telephone2chntext(fixed=True), 1
)
# 规范化分数
pattern = re.compile(r"(\d+/\d+)")
matchers = pattern.findall(text)
if matchers:
# print('fraction')
for matcher in matchers:
text = text.replace(matcher, Fraction(fraction=matcher).fraction2chntext(), 1)
# 规范化百分数
text = text.replace("%", "%")
pattern = re.compile(r"(\d+(\.\d+)?%)")
matchers = pattern.findall(text)
if matchers:
# print('percentage')
for matcher in matchers:
text = text.replace(
matcher[0], Percentage(percentage=matcher[0]).percentage2chntext(), 1
)
# 规范化纯数+量词
pattern = re.compile(r"(\d+(\.\d+)?)[多余几]?" + COM_QUANTIFIERS)
matchers = pattern.findall(text)
if matchers:
# print('cardinal+quantifier')
for matcher in matchers:
text = text.replace(matcher[0], Cardinal(cardinal=matcher[0]).cardinal2chntext(), 1)
# 规范化数字编号
pattern = re.compile(r"(\d{4,32})")
matchers = pattern.findall(text)
if matchers:
# print('digit')
for matcher in matchers:
text = text.replace(matcher, Digit(digit=matcher).digit2chntext(), 1)
# 规范化纯数
pattern = re.compile(r"(\d+(\.\d+)?)")
matchers = pattern.findall(text)
if matchers:
# print('cardinal')
for matcher in matchers:
text = text.replace(matcher[0], Cardinal(cardinal=matcher[0]).cardinal2chntext(), 1)
self.norm_text = text
self._particular()
return self.norm_text.lstrip("^").rstrip("$")
def nsw_test_case(raw_text):
print("I:" + raw_text)
print("O:" + NSWNormalizer(raw_text).normalize())
print("")
def nsw_test():
nsw_test_case("固话:0595-23865596或23880880。")
nsw_test_case("固话:0595-23865596或23880880。")
nsw_test_case("手机:+86 19859213959或15659451527。")
nsw_test_case("分数:32477/76391。")
nsw_test_case("百分数:80.03%。")
nsw_test_case("编号:31520181154418。")
nsw_test_case("纯数:2983.07克或12345.60米。")
nsw_test_case("日期:1999年2月20日或09年3月15号。")
nsw_test_case("金钱:12块5,34.5元,20.1万")
nsw_test_case("特殊:O2O或B2C。")
nsw_test_case("3456万吨")
nsw_test_case("2938个")
nsw_test_case("938")
nsw_test_case("今天吃了115个小笼包231个馒头")
nsw_test_case("有62%的概率")
if __name__ == "__main__":
# nsw_test()
p = argparse.ArgumentParser()
p.add_argument("ifile", help="input filename, assume utf-8 encoding")
p.add_argument("ofile", help="output filename")
p.add_argument("--to_upper", action="store_true", help="convert to upper case")
p.add_argument("--to_lower", action="store_true", help="convert to lower case")
p.add_argument(
"--has_key", action="store_true", help="input text has Kaldi's key as first field."
)
p.add_argument(
"--remove_fillers", type=bool, default=True, help='remove filler chars such as "呃, 啊"'
)
p.add_argument(
"--remove_erhua", type=bool, default=True, help='remove erhua chars such as "这儿"'
)
p.add_argument(
"--log_interval", type=int, default=10000, help="log interval in number of processed lines"
)
args = p.parse_args()
ifile = codecs.open(args.ifile, "r", "utf8")
ofile = codecs.open(args.ofile, "w+", "utf8")
n = 0
for l in ifile:
key = ""
text = ""
if args.has_key:
cols = l.split(maxsplit=1)
key = cols[0]
if len(cols) == 2:
text = cols[1].strip()
else:
text = ""
else:
text = l.strip()
# cases
if args.to_upper and args.to_lower:
sys.stderr.write("text norm: to_upper OR to_lower?")
exit(1)
if args.to_upper:
text = text.upper()
if args.to_lower:
text = text.lower()
# Filler chars removal
if args.remove_fillers:
for ch in FILLER_CHARS:
text = text.replace(ch, "")
if args.remove_erhua:
text = remove_erhua(text, ER_WHITELIST)
# NSW(Non-Standard-Word) normalization
text = NSWNormalizer(text).normalize()
# Punctuations removal
old_chars = CHINESE_PUNC_LIST + string.punctuation # includes all CN and EN punctuations
new_chars = " " * len(old_chars)
del_chars = ""
text = text.translate(str.maketrans(old_chars, new_chars, del_chars))
#
if args.has_key:
ofile.write(key + "\t" + text + "\n")
else:
ofile.write(text + "\n")
n += 1
if n % args.log_interval == 0:
sys.stderr.write("text norm: {} lines done.\n".format(n))
sys.stderr.write("text norm: {} lines done in total.\n".format(n))
ifile.close()
ofile.close()
+16
View File
@@ -0,0 +1,16 @@
# Conformer Result
## Training Config
- Feature info: using 80 dims fbank, global cmvn, speed perturb(0.9, 1.0, 1.1), specaugment
- Train info: lr 5e-4, batch_size 25000, 2 gpu(Tesla V100), acc_grad 1, 50 epochs
- Train config: conf/train_asr_transformer.yaml
- LM config: LM was not used
- Model size: 46M
## Results (CER)
| testset | CER(%) |
|:-----------:|:------:|
| dev | 4.97 |
| test | 5.37 |
@@ -0,0 +1,104 @@
# This is an example that demonstrates how to configure a model file.
# You can modify the configuration according to your own requirements.
# to print the register_table:
# from funasr.register import tables
# tables.print()
# network architecture
model: Transformer
model_conf:
ctc_weight: 0.3
lsm_weight: 0.1 # label smoothing option
length_normalized_loss: false
# encoder
encoder: TransformerEncoder
encoder_conf:
output_size: 256 # dimension of attention
attention_heads: 4
linear_units: 2048 # the number of units of position-wise feed forward
num_blocks: 12 # the number of encoder blocks
dropout_rate: 0.1
positional_dropout_rate: 0.1
attention_dropout_rate: 0.0
input_layer: conv2d # encoder architecture type
normalize_before: true
# decoder
decoder: TransformerDecoder
decoder_conf:
attention_heads: 4
linear_units: 2048
num_blocks: 6
dropout_rate: 0.1
positional_dropout_rate: 0.1
self_attention_dropout_rate: 0.0
src_attention_dropout_rate: 0.0
# frontend related
frontend: WavFrontend
frontend_conf:
fs: 16000
window: hamming
n_mels: 80
frame_length: 25
frame_shift: 10
lfr_m: 1
lfr_n: 1
specaug: SpecAug
specaug_conf:
apply_time_warp: true
time_warp_window: 5
time_warp_mode: bicubic
apply_freq_mask: true
freq_mask_width_range:
- 0
- 30
num_freq_mask: 2
apply_time_mask: true
time_mask_width_range:
- 0
- 40
num_time_mask: 2
train_conf:
accum_grad: 1
grad_clip: 5
max_epoch: 150
keep_nbest_models: 10
log_interval: 50
optim: adam
optim_conf:
lr: 0.002
scheduler: warmuplr
scheduler_conf:
warmup_steps: 30000
dataset: AudioDataset
dataset_conf:
index_ds: IndexDSJsonl
batch_sampler: EspnetStyleBatchSampler
batch_type: length # example or length
batch_size: 25000 # if batch_type is example, batch_size is the numbers of samples; if length, batch_size is source_token_len+target_token_len;
max_token_length: 2048 # filter samples if source_token_len+target_token_len > max_token_length,
buffer_size: 1024
shuffle: True
num_workers: 4
preprocessor_speech: SpeechPreprocessSpeedPerturb
preprocessor_speech_conf:
speed_perturb: [0.9, 1.0, 1.1]
tokenizer: CharTokenizer
tokenizer_conf:
unk_symbol: <unk>
ctc_conf:
dropout_rate: 0.0
ctc_type: builtin
reduce: true
ignore_nan_grad: true
normalize: null
@@ -0,0 +1 @@
../paraformer/demo_infer.sh
@@ -0,0 +1 @@
../paraformer/demo_train_or_finetune.sh
@@ -0,0 +1,66 @@
#!/bin/bash
# Copyright 2017 Xingyu Na
# Apache 2.0
#. ./path.sh || exit 1;
if [ $# != 3 ]; then
echo "Usage: $0 <audio-path> <text-path> <output-path>"
echo " $0 /export/a05/xna/data/data_aishell/wav /export/a05/xna/data/data_aishell/transcript data"
exit 1;
fi
aishell_audio_dir=$1
aishell_text=$2/aishell_transcript_v0.8.txt
output_dir=$3
train_dir=$output_dir/data/local/train
dev_dir=$output_dir/data/local/dev
test_dir=$output_dir/data/local/test
tmp_dir=$output_dir/data/local/tmp
mkdir -p $train_dir
mkdir -p $dev_dir
mkdir -p $test_dir
mkdir -p $tmp_dir
# data directory check
if [ ! -d $aishell_audio_dir ] || [ ! -f $aishell_text ]; then
echo "Error: $0 requires two directory arguments"
exit 1;
fi
# find wav audio file for train, dev and test resp.
find $aishell_audio_dir -iname "*.wav" > $tmp_dir/wav.flist
n=`cat $tmp_dir/wav.flist | wc -l`
[ $n -ne 141925 ] && \
echo Warning: expected 141925 data data files, found $n
grep -i "wav/train" $tmp_dir/wav.flist > $train_dir/wav.flist || exit 1;
grep -i "wav/dev" $tmp_dir/wav.flist > $dev_dir/wav.flist || exit 1;
grep -i "wav/test" $tmp_dir/wav.flist > $test_dir/wav.flist || exit 1;
rm -r $tmp_dir
# Transcriptions preparation
for dir in $train_dir $dev_dir $test_dir; do
echo Preparing $dir transcriptions
sed -e 's/\.wav//' $dir/wav.flist | awk -F '/' '{print $NF}' > $dir/utt.list
paste -d' ' $dir/utt.list $dir/wav.flist > $dir/wav.scp_all
utils/filter_scp.pl -f 1 $dir/utt.list $aishell_text > $dir/transcripts.txt
awk '{print $1}' $dir/transcripts.txt > $dir/utt.list
utils/filter_scp.pl -f 1 $dir/utt.list $dir/wav.scp_all | sort -u > $dir/wav.scp
sort -u $dir/transcripts.txt > $dir/text
done
mkdir -p $output_dir/data/train $output_dir/data/dev $output_dir/data/test
for f in wav.scp text; do
cp $train_dir/$f $output_dir/data/train/$f || exit 1;
cp $dev_dir/$f $output_dir/data/dev/$f || exit 1;
cp $test_dir/$f $output_dir/data/test/$f || exit 1;
done
echo "$0: AISHELL data preparation succeeded"
exit 0;
@@ -0,0 +1,105 @@
#!/usr/bin/env bash
# Copyright 2014 Johns Hopkins University (author: Daniel Povey)
# 2017 Xingyu Na
# Apache 2.0
remove_archive=false
if [ "$1" == --remove-archive ]; then
remove_archive=true
shift
fi
if [ $# -ne 3 ]; then
echo "Usage: $0 [--remove-archive] <data-base> <url-base> <corpus-part>"
echo "e.g.: $0 /export/a05/xna/data www.openslr.org/resources/33 data_aishell"
echo "With --remove-archive it will remove the archive after successfully un-tarring it."
echo "<corpus-part> can be one of: data_aishell, resource_aishell."
fi
data=$1
url=$2
part=$3
if [ ! -d "$data" ]; then
echo "$0: no such directory $data"
exit 1;
fi
part_ok=false
list="data_aishell resource_aishell"
for x in $list; do
if [ "$part" == $x ]; then part_ok=true; fi
done
if ! $part_ok; then
echo "$0: expected <corpus-part> to be one of $list, but got '$part'"
exit 1;
fi
if [ -z "$url" ]; then
echo "$0: empty URL base."
exit 1;
fi
if [ -f $data/$part/.complete ]; then
echo "$0: data part $part was already successfully extracted, nothing to do."
exit 0;
fi
# sizes of the archive files in bytes.
sizes="15582913665 1246920"
if [ -f $data/$part.tgz ]; then
size=$(/bin/ls -l $data/$part.tgz | awk '{print $5}')
size_ok=false
for s in $sizes; do if [ $s == $size ]; then size_ok=true; fi; done
if ! $size_ok; then
echo "$0: removing existing file $data/$part.tgz because its size in bytes $size"
echo "does not equal the size of one of the archives."
rm $data/$part.tgz
else
echo "$data/$part.tgz exists and appears to be complete."
fi
fi
if [ ! -f $data/$part.tgz ]; then
if ! command -v wget >/dev/null; then
echo "$0: wget is not installed."
exit 1;
fi
full_url=$url/$part.tgz
echo "$0: downloading data from $full_url. This may take some time, please be patient."
cd $data || exit 1
if ! wget --no-check-certificate $full_url; then
echo "$0: error executing wget $full_url"
exit 1;
fi
fi
cd $data || exit 1
if ! tar -xvzf $part.tgz; then
echo "$0: error un-tarring archive $data/$part.tgz"
exit 1;
fi
touch $data/$part/.complete
if [ $part == "data_aishell" ]; then
cd $data/$part/wav || exit 1
for wav in ./*.tar.gz; do
echo "Extracting wav from $wav"
tar -zxf $wav && rm $wav
done
fi
echo "$0: Successfully downloaded and un-tarred $data/$part.tgz"
if $remove_archive; then
echo "$0: removing $data/$part.tgz file since --remove-archive option was supplied."
rm $data/$part.tgz
fi
exit 0;
+203
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@@ -0,0 +1,203 @@
#!/usr/bin/env bash
CUDA_VISIBLE_DEVICES="0,1"
# general configuration
feats_dir="../DATA" #feature output dictionary
exp_dir=`pwd`
lang=zh
token_type=char
stage=0
stop_stage=5
# feature configuration
nj=32
inference_device="cuda" #"cpu"
inference_checkpoint="model.pt.avg10"
inference_scp="wav.scp"
inference_batch_size=1
# data
raw_data=../raw_data
data_url=www.openslr.org/resources/33
# exp tag
tag="exp1"
workspace=`pwd`
master_port=12345
. utils/parse_options.sh || exit 1;
# Set bash to 'debug' mode, it will exit on :
# -e 'error', -u 'undefined variable', -o ... 'error in pipeline', -x 'print commands',
set -e
set -u
set -o pipefail
train_set=train
valid_set=dev
test_sets="dev test"
config=transformer_12e_6d_2048_256.yaml
model_dir="baseline_$(basename "${config}" .yaml)_${lang}_${token_type}_${tag}"
if [ ${stage} -le -1 ] && [ ${stop_stage} -ge -1 ]; then
echo "stage -1: Data Download"
mkdir -p ${raw_data}
local/download_and_untar.sh ${raw_data} ${data_url} data_aishell
local/download_and_untar.sh ${raw_data} ${data_url} resource_aishell
fi
if [ ${stage} -le 0 ] && [ ${stop_stage} -ge 0 ]; then
echo "stage 0: Data preparation"
# Data preparation
local/aishell_data_prep.sh ${raw_data}/data_aishell/wav ${raw_data}/data_aishell/transcript ${feats_dir}
for x in train dev test; do
cp ${feats_dir}/data/${x}/text ${feats_dir}/data/${x}/text.org
paste -d " " <(cut -f 1 -d" " ${feats_dir}/data/${x}/text.org) <(cut -f 2- -d" " ${feats_dir}/data/${x}/text.org | tr -d " ") \
> ${feats_dir}/data/${x}/text
utils/text2token.py -n 1 -s 1 ${feats_dir}/data/${x}/text > ${feats_dir}/data/${x}/text.org
mv ${feats_dir}/data/${x}/text.org ${feats_dir}/data/${x}/text
# convert wav.scp text to jsonl
scp_file_list_arg="++scp_file_list='[\"${feats_dir}/data/${x}/wav.scp\",\"${feats_dir}/data/${x}/text\"]'"
python ../../../funasr/datasets/audio_datasets/scp2jsonl.py \
++data_type_list='["source", "target"]' \
++jsonl_file_out=${feats_dir}/data/${x}/audio_datasets.jsonl \
${scp_file_list_arg}
done
fi
if [ ${stage} -le 1 ] && [ ${stop_stage} -ge 1 ]; then
echo "stage 1: Feature and CMVN Generation"
python ../../../funasr/bin/compute_audio_cmvn.py \
--config-path "${workspace}/conf" \
--config-name "${config}" \
++train_data_set_list="${feats_dir}/data/${train_set}/audio_datasets.jsonl" \
++cmvn_file="${feats_dir}/data/${train_set}/cmvn.json" \
fi
token_list=${feats_dir}/data/${lang}_token_list/$token_type/tokens.txt
echo "dictionary: ${token_list}"
if [ ${stage} -le 2 ] && [ ${stop_stage} -ge 2 ]; then
echo "stage 2: Dictionary Preparation"
mkdir -p ${feats_dir}/data/${lang}_token_list/$token_type/
echo "make a dictionary"
echo "<blank>" > ${token_list}
echo "<s>" >> ${token_list}
echo "</s>" >> ${token_list}
utils/text2token.py -s 1 -n 1 --space "" ${feats_dir}/data/$train_set/text | cut -f 2- -d" " | tr " " "\n" \
| sort | uniq | grep -a -v -e '^\s*$' | awk '{print $0}' >> ${token_list}
echo "<unk>" >> ${token_list}
fi
# LM Training Stage
if [ ${stage} -le 3 ] && [ ${stop_stage} -ge 3 ]; then
echo "stage 3: LM Training"
fi
# ASR Training Stage
if [ ${stage} -le 4 ] && [ ${stop_stage} -ge 4 ]; then
echo "stage 4: ASR Training"
mkdir -p ${exp_dir}/exp/${model_dir}
current_time=$(date "+%Y-%m-%d_%H-%M")
log_file="${exp_dir}/exp/${model_dir}/train.log.txt.${current_time}"
echo "log_file: ${log_file}"
export CUDA_VISIBLE_DEVICES=$CUDA_VISIBLE_DEVICES
gpu_num=$(echo $CUDA_VISIBLE_DEVICES | awk -F "," '{print NF}')
torchrun \
--nnodes 1 \
--nproc_per_node ${gpu_num} \
--master_port ${master_port} \
../../../funasr/bin/train.py \
--config-path "${workspace}/conf" \
--config-name "${config}" \
++train_data_set_list="${feats_dir}/data/${train_set}/audio_datasets.jsonl" \
++valid_data_set_list="${feats_dir}/data/${valid_set}/audio_datasets.jsonl" \
++tokenizer_conf.token_list="${token_list}" \
++frontend_conf.cmvn_file="${feats_dir}/data/${train_set}/am.mvn" \
++output_dir="${exp_dir}/exp/${model_dir}" &> ${log_file}
fi
# Testing Stage
if [ ${stage} -le 5 ] && [ ${stop_stage} -ge 5 ]; then
echo "stage 5: Inference"
if [ ${inference_device} == "cuda" ]; then
nj=$(echo $CUDA_VISIBLE_DEVICES | awk -F "," '{print NF}')
else
inference_batch_size=1
CUDA_VISIBLE_DEVICES=""
for JOB in $(seq ${nj}); do
CUDA_VISIBLE_DEVICES=$CUDA_VISIBLE_DEVICES"-1,"
done
fi
for dset in ${test_sets}; do
inference_dir="${exp_dir}/exp/${model_dir}/inference-${inference_checkpoint}/${dset}"
_logdir="${inference_dir}/logdir"
echo "inference_dir: ${inference_dir}"
mkdir -p "${_logdir}"
data_dir="${feats_dir}/data/${dset}"
key_file=${data_dir}/${inference_scp}
split_scps=
for JOB in $(seq "${nj}"); do
split_scps+=" ${_logdir}/keys.${JOB}.scp"
done
utils/split_scp.pl "${key_file}" ${split_scps}
gpuid_list_array=(${CUDA_VISIBLE_DEVICES//,/ })
for JOB in $(seq ${nj}); do
{
id=$((JOB-1))
gpuid=${gpuid_list_array[$id]}
export CUDA_VISIBLE_DEVICES=${gpuid}
python ../../../funasr/bin/inference.py \
--config-path="${exp_dir}/exp/${model_dir}" \
--config-name="config.yaml" \
++init_param="${exp_dir}/exp/${model_dir}/${inference_checkpoint}" \
++tokenizer_conf.token_list="${token_list}" \
++frontend_conf.cmvn_file="${feats_dir}/data/${train_set}/am.mvn" \
++input="${_logdir}/keys.${JOB}.scp" \
++output_dir="${inference_dir}/${JOB}" \
++device="${inference_device}" \
++ncpu=1 \
++disable_log=true \
++batch_size="${inference_batch_size}" &> ${_logdir}/log.${JOB}.txt
}&
done
wait
mkdir -p ${inference_dir}/1best_recog
for f in token score text; do
if [ -f "${inference_dir}/${JOB}/1best_recog/${f}" ]; then
for JOB in $(seq "${nj}"); do
cat "${inference_dir}/${JOB}/1best_recog/${f}"
done | sort -k1 >"${inference_dir}/1best_recog/${f}"
fi
done
echo "Computing WER ..."
python utils/postprocess_text_zh.py ${inference_dir}/1best_recog/text ${inference_dir}/1best_recog/text.proc
python utils/postprocess_text_zh.py ${data_dir}/text ${inference_dir}/1best_recog/text.ref
python utils/compute_wer.py ${inference_dir}/1best_recog/text.ref ${inference_dir}/1best_recog/text.proc ${inference_dir}/1best_recog/text.cer
tail -n 3 ${inference_dir}/1best_recog/text.cer
done
fi
+1
View File
@@ -0,0 +1 @@
../paraformer/utils
@@ -0,0 +1,19 @@
#!/usr/bin/env python3
# -*- encoding: utf-8 -*-
# Copyright FunASR (https://github.com/alibaba-damo-academy/FunASR). All Rights Reserved.
# MIT License (https://opensource.org/licenses/MIT)
from funasr import AutoModel
multilingual_wavs = [
"example_zh-CN.mp3",
"example_en.mp3",
"example_ja.mp3",
"example_ko.mp3",
]
model = AutoModel(model="iic/speech_whisper-large_lid_multilingual_pytorch")
for wav_id in multilingual_wavs:
wav_file = f"{model.model_path}/examples/{wav_id}"
res = model.generate(input=wav_file, data_type="sound", inference_clip_length=250)
print("detect sample {}: {}".format(wav_id, res))
@@ -0,0 +1,22 @@
#!/usr/bin/env python3
# -*- encoding: utf-8 -*-
# Copyright FunASR (https://github.com/alibaba-damo-academy/FunASR). All Rights Reserved.
# MIT License (https://opensource.org/licenses/MIT)
from modelscope.pipelines import pipeline
from modelscope.utils.constant import Tasks
multilingual_wavs = [
"https://www.modelscope.cn/api/v1/models/iic/speech_whisper-large_lid_multilingual_pytorch/repo?Revision=master&FilePath=examples/example_zh-CN.mp3",
"https://www.modelscope.cn/api/v1/models/iic/speech_whisper-large_lid_multilingual_pytorch/repo?Revision=master&FilePath=examples/example_en.mp3",
"https://www.modelscope.cn/api/v1/models/iic/speech_whisper-large_lid_multilingual_pytorch/repo?Revision=master&FilePath=examples/example_ja.mp3",
"https://www.modelscope.cn/api/v1/models/iic/speech_whisper-large_lid_multilingual_pytorch/repo?Revision=master&FilePath=examples/example_ko.mp3",
]
inference_pipeline = pipeline(
task=Tasks.auto_speech_recognition, model="iic/speech_whisper-large_lid_multilingual_pytorch"
)
for wav in multilingual_wavs:
rec_result = inference_pipeline(input=wav, inference_clip_length=250)
print(rec_result)
@@ -0,0 +1,20 @@
#!/usr/bin/env python3
# -*- encoding: utf-8 -*-
# Copyright FunASR (https://github.com/alibaba-damo-academy/FunASR). All Rights Reserved.
# MIT License (https://opensource.org/licenses/MIT)
from funasr import AutoModel
model = AutoModel(
model="iic/speech_paraformer-large-vad-punc_asr_nat-zh-cn-16k-common-vocab8404-pytorch",
vad_model="iic/speech_fsmn_vad_zh-cn-16k-common-pytorch",
punc_model="iic/punc_ct-transformer_cn-en-common-vocab471067-large",
# spk_model="iic/speech_campplus_sv_zh-cn_16k-common",
)
res = model.generate(
input="https://isv-data.oss-cn-hangzhou.aliyuncs.com/ics/MaaS/ASR/test_audio/asr_vad_punc_example.wav",
batch_size_s=300,
batch_size_threshold_s=60,
)
print(res)
@@ -0,0 +1,18 @@
model="iic/speech_paraformer-large-vad-punc_asr_nat-zh-cn-16k-common-vocab8404-pytorch"
vad_model="iic/speech_fsmn_vad_zh-cn-16k-common-pytorch"
#punc_model="iic/punc_ct-transformer_zh-cn-common-vocab272727-pytorch"
punc_model="iic/punc_ct-transformer_cn-en-common-vocab471067-large"
spk_model="iic/speech_campplus_sv_zh-cn_16k-common"
python funasr/bin/inference.py \
+model=${model} \
+vad_model=${vad_model} \
+punc_model=${punc_model} \
+spk_model=${spk_model} \
+input="https://isv-data.oss-cn-hangzhou.aliyuncs.com/ics/MaaS/ASR/test_audio/asr_vad_punc_example.wav" \
+output_dir="./outputs/debug" \
+device="cpu" \
+batch_size_s=300 \
+batch_size_threshold_s=60
@@ -0,0 +1,28 @@
#!/usr/bin/env python3
# -*- encoding: utf-8 -*-
# Copyright FunASR (https://github.com/alibaba-damo-academy/FunASR). All Rights Reserved.
# MIT License (https://opensource.org/licenses/MIT)
# method1, inference from model hub
from funasr import AutoModel
model = AutoModel(
model="iic/speech_paraformer-large-vad-punc_asr_nat-zh-cn-16k-common-vocab8404-pytorch",
device="cpu",
)
res = model.export(type="onnx", quantize=False)
print(res)
# method2, inference from local path
from funasr import AutoModel
model = AutoModel(
model="/Users/zhifu/.cache/modelscope/hub/iic/speech_paraformer-large_asr_nat-zh-cn-16k-common-vocab8404-pytorch",
device="cpu",
)
res = model.export(type="onnx", quantize=False)
print(res)
@@ -0,0 +1,23 @@
# Copyright FunASR (https://github.com/alibaba-damo-academy/FunASR). All Rights Reserved.
# MIT License (https://opensource.org/licenses/MIT)
# method1, inference from model hub
export HYDRA_FULL_ERROR=1
model="iic/speech_paraformer-large_asr_nat-zh-cn-16k-common-vocab8404-pytorch"
python -m funasr.bin.export \
++model=${model} \
++type="onnx" \
++quantize=false \
++device="cpu"
# method2, inference from local path
model="/Users/zhifu/.cache/modelscope/hub/iic/speech_paraformer-large_asr_nat-zh-cn-16k-common-vocab8404-pytorch"
python -m funasr.bin.export \
++model=${model} \
++type="onnx" \
++quantize=false \
++device="cpu"
@@ -0,0 +1,65 @@
# Copyright FunASR (https://github.com/alibaba-damo-academy/FunASR). All Rights Reserved.
# MIT License (https://opensource.org/licenses/MIT)
# method1, finetune from model hub
# which gpu to train or finetune
export CUDA_VISIBLE_DEVICES="0,1"
gpu_num=$(echo $CUDA_VISIBLE_DEVICES | awk -F "," '{print NF}')
# model_name from model_hub, or model_dir in local path
## option 1, download model automatically
model_name_or_model_dir="iic/speech_paraformer-large-vad-punc_asr_nat-zh-cn-16k-common-vocab8404-pytorch"
## option 2, download model by git
#local_path_root=${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 \
--nproc_per_node ${gpu_num} \
../../../funasr/bin/train.py \
++model="${model_name_or_model_dir}" \
++train_data_set_list="${train_data}" \
++valid_data_set_list="${val_data}" \
++dataset_conf.batch_size=20000 \
++dataset_conf.batch_type="token" \
++dataset_conf.num_workers=4 \
++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 \
++optim_conf.lr=0.0002 \
++output_dir="${output_dir}" &> ${log_file}
@@ -0,0 +1,13 @@
#!/usr/bin/env python3
# -*- encoding: utf-8 -*-
# Copyright FunASR (https://github.com/alibaba-damo-academy/FunASR). All Rights Reserved.
# MIT License (https://opensource.org/licenses/MIT)
from funasr import AutoModel
model = AutoModel(model="iic/speech_campplus_sv_zh-cn_16k-common")
res = model.generate(
input="https://isv-data.oss-cn-hangzhou.aliyuncs.com/ics/MaaS/ASR/test_audio/asr_example_zh.wav"
)
print(res)
@@ -0,0 +1,14 @@
#!/usr/bin/env python3
# -*- encoding: utf-8 -*-
# Copyright FunASR (https://github.com/alibaba-damo-academy/FunASR). All Rights Reserved.
# MIT License (https://opensource.org/licenses/MIT)
from funasr import AutoModel
model = AutoModel(model="iic/speech_conformer_asr_nat-zh-cn-16k-aishell2-vocab5212-pytorch")
res = model.generate(
input="https://modelscope.oss-cn-beijing.aliyuncs.com/test/audios/asr_example.wav",
decoding_ctc_weight=0.0,
)
print(res)
@@ -0,0 +1,9 @@
model="iic/speech_conformer_asr_nat-zh-cn-16k-aishell1-vocab4234-pytorch"
python funasr/bin/inference.py \
+model=${model} \
+input="https://isv-data.oss-cn-hangzhou.aliyuncs.com/ics/MaaS/ASR/test_audio/asr_example_zh.wav" \
+output_dir="./outputs/debug" \
+device="cpu" \
@@ -0,0 +1,14 @@
#!/usr/bin/env python3
# -*- encoding: utf-8 -*-
# Copyright FunASR (https://github.com/alibaba-damo-academy/FunASR). All Rights Reserved.
# MIT License (https://opensource.org/licenses/MIT)
from funasr import AutoModel
model = AutoModel(model="iic/speech_paraformer-large-contextual_asr_nat-zh-cn-16k-common-vocab8404")
res = model.generate(
input="https://isv-data.oss-cn-hangzhou.aliyuncs.com/ics/MaaS/ASR/test_audio/asr_example_zh.wav",
hotword="达摩院 魔搭",
)
print(res)
@@ -0,0 +1,11 @@
model="iic/speech_paraformer-large-contextual_asr_nat-zh-cn-16k-common-vocab8404"
python ../../../funasr/bin/inference.py \
+model=${model} \
+input="https://isv-data.oss-cn-hangzhou.aliyuncs.com/ics/MaaS/ASR/test_audio/asr_example_zh.wav" \
+output_dir="./outputs/debug" \
+device="cpu" \
+"hotword='达摩院 魔搭'"
@@ -0,0 +1,9 @@
python -m funasr.bin.inference \
--config-path="/nfs/yufan.yf/workspace/model_download/modelscope/hub/damo/speech_paraformer-large-contextual_asr_nat-zh-cn-16k-common-vocab8404" \
--config-name="config.yaml" \
++init_param="/nfs/yufan.yf/workspace/model_download/modelscope/hub/damo/speech_paraformer-large-contextual_asr_nat-zh-cn-16k-common-vocab8404/model.pb" \
++tokenizer_conf.token_list="/nfs/yufan.yf/workspace/model_download/modelscope/hub/damo/speech_paraformer-large-contextual_asr_nat-zh-cn-16k-common-vocab8404/tokens.txt" \
++frontend_conf.cmvn_file="/nfs/yufan.yf/workspace/model_download/modelscope/hub/damo/speech_paraformer-large-contextual_asr_nat-zh-cn-16k-common-vocab8404/am.mvn" \
++input="/nfs/yufan.yf/workspace/model_download/modelscope/hub/damo/speech_paraformer-large-contextual_asr_nat-zh-cn-16k-common-vocab8404/asr_example_zh.wav" \
++output_dir="./outputs/debug2" \
++device="" \
@@ -0,0 +1,67 @@
# Copyright FunASR (https://github.com/alibaba-damo-academy/FunASR). All Rights Reserved.
# MIT License (https://opensource.org/licenses/MIT)
# method1, finetune from model hub
# which gpu to train or finetune
export CUDA_VISIBLE_DEVICES="0,1"
gpu_num=$(echo $CUDA_VISIBLE_DEVICES | awk -F "," '{print NF}')
# model_name from model_hub, or model_dir in local path
## option 1, download model automatically
model_name_or_model_dir="iic/speech_paraformer-large-contextual_asr_nat-zh-cn-16k-common-vocab8404"
## option 2, download model by git
#local_path_root=${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 \
--nproc_per_node ${gpu_num} \
../../../funasr/bin/train.py \
++model="${model_name_or_model_dir}" \
++model_revision="${model_revision}" \
++train_data_set_list="${train_data}" \
++valid_data_set_list="${val_data}" \
++dataset_conf.batch_size=20000 \
++dataset_conf.batch_type="token" \
++dataset_conf.num_workers=4 \
++train_conf.max_epoch=50 \
++train_conf.log_interval=1 \
++train_conf.resume=false \
++train_conf.validate_interval=2000 \
++train_conf.save_checkpoint_interval=2000 \
++train_conf.keep_nbest_models=20 \
++optim_conf.lr=0.0002 \
++output_dir="${output_dir}" &> ${log_file}
@@ -0,0 +1,6 @@
export FUNASR_DIR=$PWD/../../../
# NOTE(kan-bayashi): Use UTF-8 in Python to avoid UnicodeDecodeError when LC_ALL=C
export PYTHONIOENCODING=UTF-8
export PATH=$FUNASR_DIR/funasr/bin:$PATH
export PYTHONPATH=$FUNASR_DIR/funasr/bin:$FUNASR_DIR/funasr:$FUNASR_DIR:$PYTHONPATH
@@ -0,0 +1,23 @@
#!/usr/bin/env python3
# -*- encoding: utf-8 -*-
# Copyright FunASR (https://github.com/alibaba-damo-academy/FunASR). All Rights Reserved.
# MIT License (https://opensource.org/licenses/MIT)
from funasr import AutoModel
model = AutoModel(model="iic/punc_ct-transformer_zh-cn-common-vocab272727-pytorch")
res = model.generate(
input="https://isv-data.oss-cn-hangzhou.aliyuncs.com/ics/MaaS/ASR/test_text/punc_example.txt"
)
print(res)
from funasr import AutoModel
model = AutoModel(model="iic/punc_ct-transformer_cn-en-common-vocab471067-large")
res = model.generate(
input="https://isv-data.oss-cn-hangzhou.aliyuncs.com/ics/MaaS/ASR/test_text/punc_example.txt"
)
print(res)
@@ -0,0 +1,12 @@
#model="iic/punc_ct-transformer_zh-cn-common-vocab272727-pytorch"
#
model="iic/punc_ct-transformer_cn-en-common-vocab471067-large"
python funasr/bin/inference.py \
+model=${model} \
+input="https://isv-data.oss-cn-hangzhou.aliyuncs.com/ics/MaaS/ASR/test_text/punc_example.txt" \
+output_dir="./outputs/debug" \
+device="cpu"
@@ -0,0 +1,26 @@
#!/usr/bin/env python3
# -*- encoding: utf-8 -*-
# Copyright FunASR (https://github.com/alibaba-damo-academy/FunASR). All Rights Reserved.
# MIT License (https://opensource.org/licenses/MIT)
# method1, inference from model hub
from funasr import AutoModel
model = AutoModel(
model="iic/punc_ct-transformer_zh-cn-common-vocab272727-pytorch",
)
res = model.export(type="onnx", quantize=False)
print(res)
# method2, inference from local path
from funasr import AutoModel
model = AutoModel(
model="/Users/zhifu/.cache/modelscope/hub/iic/punc_ct-transformer_zh-cn-common-vocab272727-pytorch"
)
res = model.export(type="onnx", quantize=False)
print(res)
@@ -0,0 +1,26 @@
# Copyright FunASR (https://github.com/alibaba-damo-academy/FunASR). All Rights Reserved.
# MIT License (https://opensource.org/licenses/MIT)
# method1, inference from model hub
export HYDRA_FULL_ERROR=1
model="iic/punc_ct-transformer_zh-cn-common-vocab272727-pytorch"
python -m funasr.bin.export \
++model=${model} \
++model_revision=${model_revision} \
++type="onnx" \
++quantize=false \
++device="cpu"
# method2, inference from local path
model="/Users/zhifu/.cache/modelscope/hub/iic/punc_ct-transformer_zh-cn-common-vocab272727-pytorch"
python -m funasr.bin.export \
++model=${model} \
++type="onnx" \
++quantize=false \
++device="cpu"
@@ -0,0 +1,18 @@
#!/usr/bin/env python3
# -*- encoding: utf-8 -*-
# Copyright FunASR (https://github.com/alibaba-damo-academy/FunASR). All Rights Reserved.
# MIT License (https://opensource.org/licenses/MIT)
from funasr import AutoModel
model = AutoModel(model="iic/punc_ct-transformer_zh-cn-common-vad_realtime-vocab272727")
inputs = "跨境河流是养育沿岸|人民的生命之源长期以来为帮助下游地区防灾减灾中方技术人员|在上游地区极为恶劣的自然条件下克服巨大困难甚至冒着生命危险|向印方提供汛期水文资料处理紧急事件中方重视印方在跨境河流问题上的关切|愿意进一步完善双方联合工作机制|凡是|中方能做的我们|都会去做而且会做得更好我请印度朋友们放心中国在上游的|任何开发利用都会经过科学|规划和论证兼顾上下游的利益"
vads = inputs.split("|")
rec_result_all = "outputs: "
cache = {}
for vad in vads:
rec_result = model.generate(input=vad, cache=cache)
rec_result_all += rec_result[0]["text"]
print(rec_result_all)
@@ -0,0 +1,9 @@
model="iic/punc_ct-transformer_zh-cn-common-vad_realtime-vocab272727"
python funasr/bin/inference.py \
+model=${model} \
+input="https://isv-data.oss-cn-hangzhou.aliyuncs.com/ics/MaaS/ASR/test_text/punc_example.txt" \
+output_dir="./outputs/debug" \
+device="cpu"
@@ -0,0 +1,26 @@
#!/usr/bin/env python3
# -*- encoding: utf-8 -*-
# Copyright FunASR (https://github.com/alibaba-damo-academy/FunASR). All Rights Reserved.
# MIT License (https://opensource.org/licenses/MIT)
# method1, inference from model hub
from funasr import AutoModel
model = AutoModel(
model="iic/punc_ct-transformer_zh-cn-common-vad_realtime-vocab272727",
)
res = model.export(type="onnx", quantize=False)
print(res)
# method2, inference from local path
from funasr import AutoModel
model = AutoModel(
model="/Users/zhifu/.cache/modelscope/hub/iic/punc_ct-transformer_zh-cn-common-vad_realtime-vocab272727"
)
res = model.export(type="onnx", quantize=False)
print(res)
@@ -0,0 +1,28 @@
# Copyright FunASR (https://github.com/alibaba-damo-academy/FunASR). All Rights Reserved.
# MIT License (https://opensource.org/licenses/MIT)
# method1, inference from model hub
export HYDRA_FULL_ERROR=1
model="iic/punc_ct-transformer_zh-cn-common-vad_realtime-vocab272727"
python -m funasr.bin.export \
++model=${model} \
++model_revision=${model_revision} \
++input="https://isv-data.oss-cn-hangzhou.aliyuncs.com/ics/MaaS/ASR/test_audio/vad_example.wav" \
++type="onnx" \
++quantize=false \
++device="cpu"
# method2, inference from local path
model="/Users/zhifu/.cache/modelscope/hub/iic/punc_ct-transformer_zh-cn-common-vad_realtime-vocab272727"
python -m funasr.bin.export \
++model=${model} \
++input="https://isv-data.oss-cn-hangzhou.aliyuncs.com/ics/MaaS/ASR/test_audio/vad_example.wav" \
++type="onnx" \
++quantize=false \
++device="cpu"
@@ -0,0 +1,49 @@
#!/usr/bin/env python3
# -*- encoding: utf-8 -*-
# Copyright FunASR (https://github.com/alibaba-damo-academy/FunASR). All Rights Reserved.
# MIT License (https://opensource.org/licenses/MIT)
from funasr import AutoModel
wav_file = "https://isv-data.oss-cn-hangzhou.aliyuncs.com/ics/MaaS/ASR/test_audio/vad_example.wav"
model = AutoModel(model="iic/speech_fsmn_vad_zh-cn-16k-common-pytorch")
res = model.generate(input=wav_file)
print(res)
# [[beg1, end1], [beg2, end2], .., [begN, endN]]
# beg/end: ms
import soundfile
import os
wav_file = os.path.join(model.model_path, "example/vad_example.wav")
speech, sample_rate = soundfile.read(wav_file)
chunk_size = 200 # ms
chunk_stride = int(chunk_size * sample_rate / 1000)
cache = {}
total_chunk_num = int(len((speech) - 1) / chunk_stride + 1)
for i in range(total_chunk_num):
speech_chunk = speech[i * chunk_stride : (i + 1) * chunk_stride]
is_final = i == total_chunk_num - 1
res = model.generate(
input=speech_chunk,
cache=cache,
is_final=is_final,
chunk_size=chunk_size,
disable_pbar=True,
)
# print(res)
if len(res[0]["value"]):
print(res)
# 1. [[beg1, end1], [beg2, end2], .., [begN, endN]]; [[beg, end]]; [[beg1, end1], [beg2, end2]]
# 2. [[beg, -1]]
# 3. [[-1, end]]
# beg/end: ms
@@ -0,0 +1,10 @@
model="iic/speech_fsmn_vad_zh-cn-16k-common-pytorch"
python funasr/bin/inference.py \
+model=${model} \
+input="https://isv-data.oss-cn-hangzhou.aliyuncs.com/ics/MaaS/ASR/test_audio/vad_example.wav" \
+output_dir="./outputs/debug" \
+device="cpu" \
@@ -0,0 +1,25 @@
#!/usr/bin/env python3
# -*- encoding: utf-8 -*-
# Copyright FunASR (https://github.com/alibaba-damo-academy/FunASR). All Rights Reserved.
# MIT License (https://opensource.org/licenses/MIT)
# method1, inference from model hub
from funasr import AutoModel
model = AutoModel(model="iic/speech_fsmn_vad_zh-cn-16k-common-pytorch")
res = model.export(type="onnx", quantize=False)
print(res)
# method2, inference from local path
from funasr import AutoModel
model = AutoModel(
model="/Users/zhifu/.cache/modelscope/hub/iic/speech_fsmn_vad_zh-cn-16k-common-pytorch"
)
res = model.export(type="onnx", quantize=False)
print(res)
@@ -0,0 +1,24 @@
# Copyright FunASR (https://github.com/alibaba-damo-academy/FunASR). All Rights Reserved.
# MIT License (https://opensource.org/licenses/MIT)
# method1, inference from model hub
export HYDRA_FULL_ERROR=1
model="iic/speech_fsmn_vad_zh-cn-16k-common-pytorch"
python -m funasr.bin.export \
++model=${model} \
++model_revision=${model_revision} \
++type="onnx" \
++quantize=false
# method2, inference from local path
model="/Users/zhifu/.cache/modelscope/hub/iic/speech_fsmn_vad_zh-cn-16k-common-pytorch"
python -m funasr.bin.export \
++model=${model} \
++type="onnx" \
++quantize=false
@@ -0,0 +1,735 @@
#!/usr/bin/env python3
# -*- coding: utf-8 -*-
from enum import Enum
import re, sys, unicodedata
import codecs
import argparse
from tqdm import tqdm
import os
import pdb
remove_tag = False
spacelist = [" ", "\t", "\r", "\n"]
puncts = [
"!",
",",
"?",
"、",
"。",
"!",
",",
";",
"?",
":",
"「",
"」",
"︰",
"『",
"』",
"《",
"》",
]
class Code(Enum):
match = 1
substitution = 2
insertion = 3
deletion = 4
class WordError(object):
def __init__(self):
self.errors = {
Code.substitution: 0,
Code.insertion: 0,
Code.deletion: 0,
}
self.ref_words = 0
def get_wer(self):
assert self.ref_words != 0
errors = (
self.errors[Code.substitution]
+ self.errors[Code.insertion]
+ self.errors[Code.deletion]
)
return 100.0 * errors / self.ref_words
def get_result_string(self):
return (
f"error_rate={self.get_wer():.4f}, "
f"ref_words={self.ref_words}, "
f"subs={self.errors[Code.substitution]}, "
f"ins={self.errors[Code.insertion]}, "
f"dels={self.errors[Code.deletion]}"
)
def characterize(string):
res = []
i = 0
while i < len(string):
char = string[i]
if char in puncts:
i += 1
continue
cat1 = unicodedata.category(char)
# https://unicodebook.readthedocs.io/unicode.html#unicode-categories
if cat1 == "Zs" or cat1 == "Cn" or char in spacelist: # space or not assigned
i += 1
continue
if cat1 == "Lo": # letter-other
res.append(char)
i += 1
else:
# some input looks like: <unk><noise>, we want to separate it to two words.
sep = " "
if char == "<":
sep = ">"
j = i + 1
while j < len(string):
c = string[j]
if ord(c) >= 128 or (c in spacelist) or (c == sep):
break
j += 1
if j < len(string) and string[j] == ">":
j += 1
res.append(string[i:j])
i = j
return res
def stripoff_tags(x):
if not x:
return ""
chars = []
i = 0
T = len(x)
while i < T:
if x[i] == "<":
while i < T and x[i] != ">":
i += 1
i += 1
else:
chars.append(x[i])
i += 1
return "".join(chars)
def normalize(sentence, ignore_words, cs, split=None):
"""sentence, ignore_words are both in unicode"""
new_sentence = []
for token in sentence:
x = token
if not cs:
x = x.upper()
if x in ignore_words:
continue
if remove_tag:
x = stripoff_tags(x)
if not x:
continue
if split and x in split:
new_sentence += split[x]
else:
new_sentence.append(x)
return new_sentence
class Calculator:
def __init__(self):
self.data = {}
self.space = []
self.cost = {}
self.cost["cor"] = 0
self.cost["sub"] = 1
self.cost["del"] = 1
self.cost["ins"] = 1
def calculate(self, lab, rec):
# Initialization
lab.insert(0, "")
rec.insert(0, "")
while len(self.space) < len(lab):
self.space.append([])
for row in self.space:
for element in row:
element["dist"] = 0
element["error"] = "non"
while len(row) < len(rec):
row.append({"dist": 0, "error": "non"})
for i in range(len(lab)):
self.space[i][0]["dist"] = i
self.space[i][0]["error"] = "del"
for j in range(len(rec)):
self.space[0][j]["dist"] = j
self.space[0][j]["error"] = "ins"
self.space[0][0]["error"] = "non"
for token in lab:
if token not in self.data and len(token) > 0:
self.data[token] = {"all": 0, "cor": 0, "sub": 0, "ins": 0, "del": 0}
for token in rec:
if token not in self.data and len(token) > 0:
self.data[token] = {"all": 0, "cor": 0, "sub": 0, "ins": 0, "del": 0}
# Computing edit distance
for i, lab_token in enumerate(lab):
for j, rec_token in enumerate(rec):
if i == 0 or j == 0:
continue
min_dist = sys.maxsize
min_error = "none"
dist = self.space[i - 1][j]["dist"] + self.cost["del"]
error = "del"
if dist < min_dist:
min_dist = dist
min_error = error
dist = self.space[i][j - 1]["dist"] + self.cost["ins"]
error = "ins"
if dist < min_dist:
min_dist = dist
min_error = error
if lab_token == rec_token.replace("<BIAS>", ""):
dist = self.space[i - 1][j - 1]["dist"] + self.cost["cor"]
error = "cor"
else:
dist = self.space[i - 1][j - 1]["dist"] + self.cost["sub"]
error = "sub"
if dist < min_dist:
min_dist = dist
min_error = error
self.space[i][j]["dist"] = min_dist
self.space[i][j]["error"] = min_error
# Tracing back
result = {
"lab": [],
"rec": [],
"code": [],
"all": 0,
"cor": 0,
"sub": 0,
"ins": 0,
"del": 0,
}
i = len(lab) - 1
j = len(rec) - 1
while True:
if self.space[i][j]["error"] == "cor": # correct
if len(lab[i]) > 0:
self.data[lab[i]]["all"] = self.data[lab[i]]["all"] + 1
self.data[lab[i]]["cor"] = self.data[lab[i]]["cor"] + 1
result["all"] = result["all"] + 1
result["cor"] = result["cor"] + 1
result["lab"].insert(0, lab[i])
result["rec"].insert(0, rec[j])
result["code"].insert(0, Code.match)
i = i - 1
j = j - 1
elif self.space[i][j]["error"] == "sub": # substitution
if len(lab[i]) > 0:
self.data[lab[i]]["all"] = self.data[lab[i]]["all"] + 1
self.data[lab[i]]["sub"] = self.data[lab[i]]["sub"] + 1
result["all"] = result["all"] + 1
result["sub"] = result["sub"] + 1
result["lab"].insert(0, lab[i])
result["rec"].insert(0, rec[j])
result["code"].insert(0, Code.substitution)
i = i - 1
j = j - 1
elif self.space[i][j]["error"] == "del": # deletion
if len(lab[i]) > 0:
self.data[lab[i]]["all"] = self.data[lab[i]]["all"] + 1
self.data[lab[i]]["del"] = self.data[lab[i]]["del"] + 1
result["all"] = result["all"] + 1
result["del"] = result["del"] + 1
result["lab"].insert(0, lab[i])
result["rec"].insert(0, "")
result["code"].insert(0, Code.deletion)
i = i - 1
elif self.space[i][j]["error"] == "ins": # insertion
if len(rec[j]) > 0:
self.data[rec[j]]["ins"] = self.data[rec[j]]["ins"] + 1
result["ins"] = result["ins"] + 1
result["lab"].insert(0, "")
result["rec"].insert(0, rec[j])
result["code"].insert(0, Code.insertion)
j = j - 1
elif self.space[i][j]["error"] == "non": # starting point
break
else: # shouldn't reach here
print(
"this should not happen , i = {i} , j = {j} , error = {error}".format(
i=i, j=j, error=self.space[i][j]["error"]
)
)
return result
def overall(self):
result = {"all": 0, "cor": 0, "sub": 0, "ins": 0, "del": 0}
for token in self.data:
result["all"] = result["all"] + self.data[token]["all"]
result["cor"] = result["cor"] + self.data[token]["cor"]
result["sub"] = result["sub"] + self.data[token]["sub"]
result["ins"] = result["ins"] + self.data[token]["ins"]
result["del"] = result["del"] + self.data[token]["del"]
return result
def cluster(self, data):
result = {"all": 0, "cor": 0, "sub": 0, "ins": 0, "del": 0}
for token in data:
if token in self.data:
result["all"] = result["all"] + self.data[token]["all"]
result["cor"] = result["cor"] + self.data[token]["cor"]
result["sub"] = result["sub"] + self.data[token]["sub"]
result["ins"] = result["ins"] + self.data[token]["ins"]
result["del"] = result["del"] + self.data[token]["del"]
return result
def keys(self):
return list(self.data.keys())
def width(string):
return sum(1 + (unicodedata.east_asian_width(c) in "AFW") for c in string)
def default_cluster(word):
unicode_names = [unicodedata.name(char) for char in word]
for i in reversed(range(len(unicode_names))):
if unicode_names[i].startswith("DIGIT"): # 1
unicode_names[i] = "Number" # 'DIGIT'
elif unicode_names[i].startswith("CJK UNIFIED IDEOGRAPH") or unicode_names[i].startswith(
"CJK COMPATIBILITY IDEOGRAPH"
):
# 明 / 郎
unicode_names[i] = "Mandarin" # 'CJK IDEOGRAPH'
elif unicode_names[i].startswith("LATIN CAPITAL LETTER") or unicode_names[i].startswith(
"LATIN SMALL LETTER"
):
# A / a
unicode_names[i] = "English" # 'LATIN LETTER'
elif unicode_names[i].startswith("HIRAGANA LETTER"): # は こ め
unicode_names[i] = "Japanese" # 'GANA LETTER'
elif (
unicode_names[i].startswith("AMPERSAND")
or unicode_names[i].startswith("APOSTROPHE")
or unicode_names[i].startswith("COMMERCIAL AT")
or unicode_names[i].startswith("DEGREE CELSIUS")
or unicode_names[i].startswith("EQUALS SIGN")
or unicode_names[i].startswith("FULL STOP")
or unicode_names[i].startswith("HYPHEN-MINUS")
or unicode_names[i].startswith("LOW LINE")
or unicode_names[i].startswith("NUMBER SIGN")
or unicode_names[i].startswith("PLUS SIGN")
or unicode_names[i].startswith("SEMICOLON")
):
# & / ' / @ / ℃ / = / . / - / _ / # / + / ;
del unicode_names[i]
else:
return "Other"
if len(unicode_names) == 0:
return "Other"
if len(unicode_names) == 1:
return unicode_names[0]
for i in range(len(unicode_names) - 1):
if unicode_names[i] != unicode_names[i + 1]:
return "Other"
return unicode_names[0]
def get_args():
parser = argparse.ArgumentParser(description="wer cal")
parser.add_argument("--ref", type=str, help="Text input path")
parser.add_argument("--ref_ocr", type=str, help="Text input path")
parser.add_argument("--rec_name", type=str, action="append", default=[])
parser.add_argument("--rec_file", type=str, action="append", default=[])
parser.add_argument("--verbose", type=int, default=1, help="show")
parser.add_argument("--char", type=bool, default=True, help="show")
args = parser.parse_args()
return args
def main(args):
cluster_file = ""
ignore_words = set()
tochar = args.char
verbose = args.verbose
padding_symbol = " "
case_sensitive = False
max_words_per_line = sys.maxsize
split = None
if not case_sensitive:
ig = set([w.upper() for w in ignore_words])
ignore_words = ig
default_clusters = {}
default_words = {}
ref_file = args.ref
ref_ocr = args.ref_ocr
rec_files = args.rec_file
rec_names = args.rec_name
assert len(rec_files) == len(rec_names)
# load ocr
ref_ocr_dict = {}
with codecs.open(ref_ocr, "r", "utf-8") as fh:
for line in fh:
if "$" in line:
line = line.replace("$", " ")
if tochar:
array = characterize(line)
else:
array = line.strip().split()
if len(array) == 0:
continue
fid = array[0]
ref_ocr_dict[fid] = normalize(array[1:], ignore_words, case_sensitive, split)
if split and not case_sensitive:
newsplit = dict()
for w in split:
words = split[w]
for i in range(len(words)):
words[i] = words[i].upper()
newsplit[w.upper()] = words
split = newsplit
rec_sets = {}
calculators_dict = dict()
ub_wer_dict = dict()
hotwords_related_dict = dict() # 记录recall相关的内容
for i, hyp_file in enumerate(rec_files):
rec_sets[rec_names[i]] = dict()
with codecs.open(hyp_file, "r", "utf-8") as fh:
for line in fh:
if tochar:
array = characterize(line)
else:
array = line.strip().split()
if len(array) == 0:
continue
fid = array[0]
rec_sets[rec_names[i]][fid] = normalize(
array[1:], ignore_words, case_sensitive, split
)
calculators_dict[rec_names[i]] = Calculator()
ub_wer_dict[rec_names[i]] = {"u_wer": WordError(), "b_wer": WordError(), "wer": WordError()}
hotwords_related_dict[rec_names[i]] = {"tp": 0, "tn": 0, "fp": 0, "fn": 0}
# tp: 热词在label里,同时在rec里
# tn: 热词不在label里,同时不在rec里
# fp: 热词不在label里,但是在rec里
# fn: 热词在label里,但是不在rec里
# record wrong label but in ocr
wrong_rec_but_in_ocr_dict = {}
for rec_name in rec_names:
wrong_rec_but_in_ocr_dict[rec_name] = 0
_file_total_len = 0
with os.popen("cat {} | wc -l".format(ref_file)) as pipe:
_file_total_len = int(pipe.read().strip())
# compute error rate on the interaction of reference file and hyp file
for line in tqdm(open(ref_file, "r", encoding="utf-8"), total=_file_total_len):
if tochar:
array = characterize(line)
else:
array = line.rstrip("\n").split()
if len(array) == 0:
continue
fid = array[0]
lab = normalize(array[1:], ignore_words, case_sensitive, split)
if verbose:
print("\nutt: %s" % fid)
ocr_text = ref_ocr_dict[fid]
ocr_set = set(ocr_text)
print("ocr: {}".format(" ".join(ocr_text)))
list_match = [] # 指label里面在ocr里面的内容
list_not_mathch = []
tmp_error = 0
tmp_match = 0
for index in range(len(lab)):
# text_list.append(uttlist[index+1])
if lab[index] not in ocr_set:
tmp_error += 1
list_not_mathch.append(lab[index])
else:
tmp_match += 1
list_match.append(lab[index])
print("label in ocr: {}".format(" ".join(list_match)))
# for each reco file
base_wrong_ocr_wer = None
ocr_wrong_ocr_wer = None
for rec_name in rec_names:
rec_set = rec_sets[rec_name]
if fid not in rec_set:
continue
rec = rec_set[fid]
# print(rec)
for word in rec + lab:
if word not in default_words:
default_cluster_name = default_cluster(word)
if default_cluster_name not in default_clusters:
default_clusters[default_cluster_name] = {}
if word not in default_clusters[default_cluster_name]:
default_clusters[default_cluster_name][word] = 1
default_words[word] = default_cluster_name
result = calculators_dict[rec_name].calculate(lab.copy(), rec.copy())
if verbose:
if result["all"] != 0:
wer = (
float(result["ins"] + result["sub"] + result["del"]) * 100.0 / result["all"]
)
else:
wer = 0.0
print("WER(%s): %4.2f %%" % (rec_name, wer), end=" ")
print(
"N=%d C=%d S=%d D=%d I=%d"
% (result["all"], result["cor"], result["sub"], result["del"], result["ins"])
)
# print(result['rec'])
wrong_rec_but_in_ocr = []
for idx in range(len(result["lab"])):
if result["lab"][idx] != "":
if result["lab"][idx] != result["rec"][idx].replace("<BIAS>", ""):
if result["lab"][idx] in list_match:
wrong_rec_but_in_ocr.append(result["lab"][idx])
wrong_rec_but_in_ocr_dict[rec_name] += 1
print("wrong_rec_but_in_ocr: {}".format(" ".join(wrong_rec_but_in_ocr)))
if rec_name == "base":
base_wrong_ocr_wer = len(wrong_rec_but_in_ocr)
if "ocr" in rec_name or "hot" in rec_name:
ocr_wrong_ocr_wer = len(wrong_rec_but_in_ocr)
if ocr_wrong_ocr_wer < base_wrong_ocr_wer:
print(
"{} {} helps, {} -> {}".format(
fid, rec_name, base_wrong_ocr_wer, ocr_wrong_ocr_wer
)
)
elif ocr_wrong_ocr_wer > base_wrong_ocr_wer:
print(
"{} {} hurts, {} -> {}".format(
fid, rec_name, base_wrong_ocr_wer, ocr_wrong_ocr_wer
)
)
# recall = 0
# false_alarm = 0
# for idx in range(len(result['lab'])):
# if "<BIAS>" in result['rec'][idx]:
# if result['rec'][idx].replace("<BIAS>", "") in list_match:
# recall += 1
# else:
# false_alarm += 1
# print("bias hotwords recall: {}, fa: {}, list_match {}, recall: {:.2f}, fa: {:.2f}".format(
# recall, false_alarm, len(list_match), recall / len(list_match) if len(list_match) != 0 else 0, false_alarm / len(list_match) if len(list_match) != 0 else 0
# ))
# tp: 热词在label里,同时在rec里
# tn: 热词不在label里,同时不在rec里
# fp: 热词不在label里,但是在rec里
# fn: 热词在label里,但是不在rec里
_rec_list = [word.replace("<BIAS>", "") for word in rec]
_label_list = [word for word in lab]
_tp = _tn = _fp = _fn = 0
hot_true_list = [hotword for hotword in ocr_text if hotword in _label_list]
hot_bad_list = [hotword for hotword in ocr_text if hotword not in _label_list]
for badhotword in hot_bad_list:
count = len([word for word in _rec_list if word == badhotword])
# print(f"bad {badhotword} count: {count}")
# for word in _rec_list:
# if badhotword == word:
# count += 1
if count == 0:
hotwords_related_dict[rec_name]["tn"] += 1
_tn += 1
# fp: 0
else:
hotwords_related_dict[rec_name]["fp"] += count
_fp += count
# tn: 0
# if badhotword in _rec_list:
# hotwords_related_dict[rec_name]['fp'] += 1
# else:
# hotwords_related_dict[rec_name]['tn'] += 1
for hotword in hot_true_list:
true_count = len([word for word in _label_list if hotword == word])
rec_count = len([word for word in _rec_list if hotword == word])
# print(f"good {hotword} true_count: {true_count}, rec_count: {rec_count}")
if rec_count == true_count:
hotwords_related_dict[rec_name]["tp"] += true_count
_tp += true_count
elif rec_count > true_count:
hotwords_related_dict[rec_name]["tp"] += true_count
# fp: 不在label里,但是在rec里
hotwords_related_dict[rec_name]["fp"] += rec_count - true_count
_tp += true_count
_fp += rec_count - true_count
else:
hotwords_related_dict[rec_name]["tp"] += rec_count
# fn: 热词在label里,但是不在rec里
hotwords_related_dict[rec_name]["fn"] += true_count - rec_count
_tp += rec_count
_fn += true_count - rec_count
print(
"hotword: tp: {}, tn: {}, fp: {}, fn: {}, all: {}, recall: {:.2f}%".format(
_tp,
_tn,
_fp,
_fn,
sum([_tp, _tn, _fp, _fn]),
_tp / (_tp + _fn) * 100 if (_tp + _fn) != 0 else 0,
)
)
# if hotword in _rec_list:
# hotwords_related_dict[rec_name]['tp'] += 1
# else:
# hotwords_related_dict[rec_name]['fn'] += 1
# 计算uwer, bwer, wer
for code, rec_word, lab_word in zip(result["code"], result["rec"], result["lab"]):
if code == Code.match:
ub_wer_dict[rec_name]["wer"].ref_words += 1
if lab_word in hot_true_list:
# tmp_ref.append(ref_tokens[ref_idx])
ub_wer_dict[rec_name]["b_wer"].ref_words += 1
else:
ub_wer_dict[rec_name]["u_wer"].ref_words += 1
elif code == Code.substitution:
ub_wer_dict[rec_name]["wer"].ref_words += 1
ub_wer_dict[rec_name]["wer"].errors[Code.substitution] += 1
if lab_word in hot_true_list:
# tmp_ref.append(ref_tokens[ref_idx])
ub_wer_dict[rec_name]["b_wer"].ref_words += 1
ub_wer_dict[rec_name]["b_wer"].errors[Code.substitution] += 1
else:
ub_wer_dict[rec_name]["u_wer"].ref_words += 1
ub_wer_dict[rec_name]["u_wer"].errors[Code.substitution] += 1
elif code == Code.deletion:
ub_wer_dict[rec_name]["wer"].ref_words += 1
ub_wer_dict[rec_name]["wer"].errors[Code.deletion] += 1
if lab_word in hot_true_list:
# tmp_ref.append(ref_tokens[ref_idx])
ub_wer_dict[rec_name]["b_wer"].ref_words += 1
ub_wer_dict[rec_name]["b_wer"].errors[Code.deletion] += 1
else:
ub_wer_dict[rec_name]["u_wer"].ref_words += 1
ub_wer_dict[rec_name]["u_wer"].errors[Code.deletion] += 1
elif code == Code.insertion:
ub_wer_dict[rec_name]["wer"].errors[Code.insertion] += 1
if rec_word in hot_true_list:
ub_wer_dict[rec_name]["b_wer"].errors[Code.insertion] += 1
else:
ub_wer_dict[rec_name]["u_wer"].errors[Code.insertion] += 1
space = {}
space["lab"] = []
space["rec"] = []
for idx in range(len(result["lab"])):
len_lab = width(result["lab"][idx])
len_rec = width(result["rec"][idx])
length = max(len_lab, len_rec)
space["lab"].append(length - len_lab)
space["rec"].append(length - len_rec)
upper_lab = len(result["lab"])
upper_rec = len(result["rec"])
lab1, rec1 = 0, 0
while lab1 < upper_lab or rec1 < upper_rec:
if verbose > 1:
print("lab(%s):" % fid.encode("utf-8"), end=" ")
else:
print("lab:", end=" ")
lab2 = min(upper_lab, lab1 + max_words_per_line)
for idx in range(lab1, lab2):
token = result["lab"][idx]
print("{token}".format(token=token), end="")
for n in range(space["lab"][idx]):
print(padding_symbol, end="")
print(" ", end="")
print()
if verbose > 1:
print("rec(%s):" % fid.encode("utf-8"), end=" ")
else:
print("rec:", end=" ")
rec2 = min(upper_rec, rec1 + max_words_per_line)
for idx in range(rec1, rec2):
token = result["rec"][idx]
print("{token}".format(token=token), end="")
for n in range(space["rec"][idx]):
print(padding_symbol, end="")
print(" ", end="")
print()
# print('\n', end='\n')
lab1 = lab2
rec1 = rec2
print("\n", end="\n")
# break
if verbose:
print("===========================================================================")
print()
print(wrong_rec_but_in_ocr_dict)
for rec_name in rec_names:
result = calculators_dict[rec_name].overall()
if result["all"] != 0:
wer = float(result["ins"] + result["sub"] + result["del"]) * 100.0 / result["all"]
else:
wer = 0.0
print("{} Overall -> {:4.2f} %".format(rec_name, wer), end=" ")
print(
"N=%d C=%d S=%d D=%d I=%d"
% (result["all"], result["cor"], result["sub"], result["del"], result["ins"])
)
print(f"WER: {ub_wer_dict[rec_name]['wer'].get_result_string()}")
print(f"U-WER: {ub_wer_dict[rec_name]['u_wer'].get_result_string()}")
print(f"B-WER: {ub_wer_dict[rec_name]['b_wer'].get_result_string()}")
print(
"hotword: tp: {}, tn: {}, fp: {}, fn: {}, all: {}, recall: {:.2f}%".format(
hotwords_related_dict[rec_name]["tp"],
hotwords_related_dict[rec_name]["tn"],
hotwords_related_dict[rec_name]["fp"],
hotwords_related_dict[rec_name]["fn"],
sum([v for k, v in hotwords_related_dict[rec_name].items()]),
(
hotwords_related_dict[rec_name]["tp"]
/ (
hotwords_related_dict[rec_name]["tp"]
+ hotwords_related_dict[rec_name]["fn"]
)
* 100
if hotwords_related_dict[rec_name]["tp"] + hotwords_related_dict[rec_name]["fn"]
!= 0
else 0
),
)
)
# tp: 热词在label里,同时在rec里
# tn: 热词不在label里,同时不在rec里
# fp: 热词不在label里,但是在rec里
# fn: 热词在label里,但是不在rec里
if not verbose:
print()
print()
if __name__ == "__main__":
args = get_args()
# print("")
print(args)
main(args)
@@ -0,0 +1,19 @@
#!/usr/bin/env python3
# -*- encoding: utf-8 -*-
# Copyright FunASR (https://github.com/alibaba-damo-academy/FunASR). All Rights Reserved.
# MIT License (https://opensource.org/licenses/MIT)
from funasr import AutoModel
model = AutoModel(model="iic/LCB-NET", model_revision="v1.0.0")
res = model.generate(
input=(
"https://www.modelscope.cn/api/v1/models/iic/LCB-NET/repo?Revision=master&FilePath=example/asr_example.wav",
"https://www.modelscope.cn/api/v1/models/iic/LCB-NET/repo?Revision=master&FilePath=example/ocr.txt",
),
data_type=("sound", "text"),
)
print(res)
@@ -0,0 +1,72 @@
file_dir="/home/yf352572/.cache/modelscope/hub/iic/LCB-NET/"
CUDA_VISIBLE_DEVICES="0,1"
inference_device="cuda"
if [ ${inference_device} == "cuda" ]; then
nj=$(echo $CUDA_VISIBLE_DEVICES | awk -F "," '{print NF}')
else
inference_batch_size=1
CUDA_VISIBLE_DEVICES=""
for JOB in $(seq ${nj}); do
CUDA_VISIBLE_DEVICES=$CUDA_VISIBLE_DEVICES"-1,"
done
fi
inference_dir="outputs/slidespeech_dev"
_logdir="${inference_dir}/logdir"
echo "inference_dir: ${inference_dir}"
mkdir -p "${_logdir}"
key_file1=${file_dir}/dev/wav.scp
key_file2=${file_dir}/dev/ocr.txt
split_scps1=
split_scps2=
for JOB in $(seq "${nj}"); do
split_scps1+=" ${_logdir}/wav.${JOB}.scp"
split_scps2+=" ${_logdir}/ocr.${JOB}.txt"
done
utils/split_scp.pl "${key_file1}" ${split_scps1}
utils/split_scp.pl "${key_file2}" ${split_scps2}
gpuid_list_array=(${CUDA_VISIBLE_DEVICES//,/ })
for JOB in $(seq ${nj}); do
{
id=$((JOB-1))
gpuid=${gpuid_list_array[$id]}
export CUDA_VISIBLE_DEVICES=${gpuid}
python -m funasr.bin.inference \
--config-path=${file_dir} \
--config-name="config.yaml" \
++init_param=${file_dir}/model.pt \
++tokenizer_conf.token_list=${file_dir}/tokens.txt \
++input=[${_logdir}/wav.${JOB}.scp,${_logdir}/ocr.${JOB}.txt] \
+data_type='["kaldi_ark", "text"]' \
++tokenizer_conf.bpemodel=${file_dir}/bpe.pt \
++normalize_conf.stats_file=${file_dir}/am.mvn \
++output_dir="${inference_dir}/${JOB}" \
++device="${inference_device}" \
++ncpu=1 \
++disable_log=true &> ${_logdir}/log.${JOB}.txt
}&
done
wait
mkdir -p ${inference_dir}/1best_recog
for JOB in $(seq "${nj}"); do
cat "${inference_dir}/${JOB}/1best_recog/token" >> "${inference_dir}/1best_recog/token"
done
echo "Computing WER ..."
sed -e 's/ /\t/' -e 's/ //g' -e 's/▁/ /g' -e 's/\t /\t/' ${inference_dir}/1best_recog/token > ${inference_dir}/1best_recog/token.proc
cp ${file_dir}/dev/text ${inference_dir}/1best_recog/token.ref
cp ${file_dir}/dev/ocr.list ${inference_dir}/1best_recog/ocr.list
python utils/compute_wer.py ${inference_dir}/1best_recog/token.ref ${inference_dir}/1best_recog/token.proc ${inference_dir}/1best_recog/token.cer
tail -n 3 ${inference_dir}/1best_recog/token.cer
./run_bwer_recall.sh ${inference_dir}/1best_recog/
tail -n 6 ${inference_dir}/1best_recog/BWER-UWER.results |head -n 5
@@ -0,0 +1,11 @@
#now_result_name=asr_conformer_acc1_lr002_warm20000/decode_asr_asr_model_valid.acc.ave
#hotword_type=ocr_1ngram_top10_hotwords_list
hot_exp_suf=$1
python compute_wer_details.py --v 1 \
--ref ${hot_exp_suf}/token.ref \
--ref_ocr ${hot_exp_suf}/ocr.list \
--rec_name base \
--rec_file ${hot_exp_suf}/token.proc \
> ${hot_exp_suf}/BWER-UWER.results
@@ -0,0 +1 @@
../../aishell/paraformer/utils
@@ -0,0 +1,89 @@
# This is an example that demonstrates how to configure a model file.
# You can modify the configuration according to your own requirements.
# to print the register_table:
# from funasr.register import tables
# tables.print()
# network architecture
model: LLMASR
model_conf:
lsm_weight: 0.1 # label smoothing option
length_normalized_loss: true
# encoder
encoder: WhisperWarp
encoder_conf:
hub: funasr
init_param_path: "/nfs/maziyang.mzy/models/Whisper-large-v2"
freeze: true
llm: Vicuna
llm_conf:
hub: hf
init_param_path: "/nfs/maziyang.mzy/models/vicuna-7b-v1.5"
freeze: true
adaptor: Linear
adaptor_conf:
downsample_rate: 5
llm_dim: 4096
encoder_dim: 512
# frontend related
frontend: WhisperFrontend
frontend_conf:
fs: 16000
whisper_model: large
do_pad_trim: true
specaug: SpecAugLFR
specaug_conf:
apply_time_warp: false
time_warp_window: 5
time_warp_mode: bicubic
apply_freq_mask: true
freq_mask_width_range:
- 0
- 30
lfr_rate: 6
num_freq_mask: 1
apply_time_mask: true
time_mask_width_range:
- 0
- 12
num_time_mask: 1
train_conf:
accum_grad: 1
grad_clip: 5
max_epoch: 150
keep_nbest_models: 10
log_interval: 10
optim: adamw
optim_conf:
lr: 0.0001
weight_decay: 0.000001
scheduler: warmuplr
scheduler_conf:
warmup_steps: 1500
dataset: AudioLLMDataset
dataset_conf:
index_ds: IndexDSJsonl
batch_sampler: BatchSampler
batch_type: example # example or length
batch_size: 8 # if batch_type is example, batch_size is the numbers of samples; if length, batch_size is source_token_len+target_token_len;
max_token_length: 2048 # filter samples if source_token_len+target_token_len > max_token_length,
buffer_size: 500
shuffle: True
num_workers: 4
preprocessor_text: TextPreprocessRemovePunctuation
tokenizer: HuggingfaceTokenizer
tokenizer_conf:
unk_symbol: <unk>
init_param_path: "/nfs/maziyang.mzy/models/vicuna-7b-v1.5"
@@ -0,0 +1,94 @@
# This is an example that demonstrates how to configure a model file.
# You can modify the configuration according to your own requirements.
# to print the register_table:
# from funasr.register import tables
# tables.print()
# network architecture
model: LLMASR
model_conf:
lsm_weight: 0.1 # label smoothing option
length_normalized_loss: true
# encoder
audio_encoder: "/nfs/zhifu.gzf/init_model/Whisper-large-v3" #iic/Whisper-large-v3
audio_encoder_conf:
hub: ms
freeze: true
llm: Qwen1.5-7b-chat
llm_conf:
hub: hf
freeze: true
init_param_path: "/nfs/zhifu.gzf/init_model/qwen/Qwen1___5-7B-Chat"
audio_adaptor: Linear
audio_adaptor_conf:
downsample_rate: 5
llm_dim: 4096
encoder_dim: 512
# frontend related
frontend: WhisperFrontend
frontend_conf:
fs: 16000
whisper_model: large-v3
do_pad_trim: true
permute: true # true: [bs, frames, dims]; false: [bs, dims, frames]
specaug: SpecAugLFR
specaug_conf:
apply_time_warp: false
time_warp_window: 5
time_warp_mode: bicubic
apply_freq_mask: true
freq_mask_width_range:
- 0
- 30
lfr_rate: 6
num_freq_mask: 1
apply_time_mask: true
time_mask_width_range:
- 0
- 12
num_time_mask: 1
train_conf:
accum_grad: 1
grad_clip: 5
max_epoch: 15
keep_nbest_models: 10
log_interval: 10
optim: adamw
optim_conf:
lr: 0.0001
weight_decay: 0.000000
scheduler: warmuplr
scheduler_conf:
warmup_steps: 1500
dataset: AudioLLMQwenAudioDataset
dataset_conf:
index_ds: IndexDSJsonl
batch_sampler: CustomDistributedBatchSampler
batch_type: example # example or length
batch_size: 4 # if batch_type is example, batch_size is the numbers of samples; if length, batch_size is source_token_len+target_token_len;
max_token_length: 3000 # filter samples if source_token_len+target_token_len > max_token_length,
shuffle: True
num_workers: 4
preprocessor_text: TextPreprocessRemovePunctuation
audio_adaptor_downsample_rate: ${audio_adaptor_conf.downsample_rate}
audio_encoder_downsample_rate: 2
# prompt: "<|startoftranscription|><|zh|><|transcribe|><|zh|><|notimestamps|><|wo_itn|>"
tokenizer: HuggingfaceTokenizer
tokenizer_conf:
unk_symbol: <unk>
init_param_path: "/nfs/zhifu.gzf/init_model/qwen/Qwen1___5-7B-Chat"
@@ -0,0 +1,93 @@
# This is an example that demonstrates how to configure a model file.
# You can modify the configuration according to your own requirements.
# to print the register_table:
# from funasr.register import tables
# tables.print()
# network architecture
model: LLMASR
model_conf:
lsm_weight: 0.1 # label smoothing option
length_normalized_loss: true
# encoder
audio_encoder: "/nfs/zhifu.gzf/init_model/Whisper-large-v3" #iic/Whisper-large-v3
audio_encoder_conf:
hub: ms
freeze: true
llm: Vicuna
llm_conf:
hub: hf
init_param_path: "/nfs/maziyang.mzy/models/vicuna-7b-v1.5"
freeze: true
audio_adaptor: Linear
audio_adaptor_conf:
downsample_rate: 5
llm_dim: 4096
encoder_dim: 512
# frontend related
frontend: WhisperFrontend
frontend_conf:
fs: 16000
whisper_model: large-v3
do_pad_trim: true
permute: true # true: [bs, frames, dims]; false: [bs, dims, frames]
specaug: SpecAugLFR
specaug_conf:
apply_time_warp: false
time_warp_window: 5
time_warp_mode: bicubic
apply_freq_mask: true
freq_mask_width_range:
- 0
- 30
lfr_rate: 6
num_freq_mask: 1
apply_time_mask: true
time_mask_width_range:
- 0
- 12
num_time_mask: 1
train_conf:
accum_grad: 1
grad_clip: 5
max_epoch: 15
keep_nbest_models: 10
log_interval: 10
optim: adamw
optim_conf:
lr: 0.0001
weight_decay: 0
scheduler: warmuplr
scheduler_conf:
warmup_steps: 1500
dataset: AudioLLMVicunaDataset
dataset_conf:
index_ds: IndexDSJsonl
batch_sampler: CustomDistributedBatchSampler
batch_type: example # example or length
batch_size: 4 # if batch_type is example, batch_size is the numbers of samples; if length, batch_size is source_token_len+target_token_len;
max_token_length: 3000 # filter samples if source_token_len+target_token_len > max_token_length,
shuffle: True
num_workers: 4
# preprocessor_text: TextPreprocessRemovePunctuation
audio_adaptor_downsample_rate: ${audio_adaptor_conf.downsample_rate}
audio_encoder_downsample_rate: 2
tokenizer: HuggingfaceTokenizer
tokenizer_conf:
unk_symbol: <unk>
init_param_path: "/nfs/maziyang.mzy/models/vicuna-7b-v1.5"
@@ -0,0 +1,14 @@
# Copyright FunASR (https://github.com/alibaba-damo-academy/FunASR). All Rights Reserved.
# MIT License (https://opensource.org/licenses/MIT)
python -m funasr.bin.inference \
--config-path="/root/FunASR/examples/aishell/llm_asr_nar/conf" \
--config-name="template.yaml" \
++init_param="/mnt/workspace/FunASR/examples/aishell/paraformer/exp/baseline_paraformer_conformer_12e_6d_2048_256_zh_char_exp3/model.pt.ep38" \
++input="/nfs/beinian.lzr/workspace/datasets/data/16k/opendata/aishell1/dev/wav/S0724/BAC009S0724W0121.wav" \
++scope_map="encoder.model,audio_encoder,encoder_projector,adaptor" \
++output_dir="./outputs/debug" \
++device="cpu" \
@@ -0,0 +1,46 @@
# Copyright FunASR (https://github.com/alibaba-damo-academy/FunASR). All Rights Reserved.
# MIT License (https://opensource.org/licenses/MIT)
# which gpu to train or finetune
export CUDA_VISIBLE_DEVICES="0"
gpu_num=$(echo $CUDA_VISIBLE_DEVICES | awk -F "," '{print NF}')
# data dir, which contains: train.json, val.json, tokens.jsonl/tokens.txt, am.mvn
#data_dir="/Users/zhifu/funasr1.0/data/list"
## generate jsonl from wav.scp and text.txt
#python -m funasr.datasets.audio_datasets.scp2jsonl \
#++scp_file_list='["/Users/zhifu/funasr1.0/test_local/wav.scp", "/Users/zhifu/funasr1.0/test_local/text.txt"]' \
#++data_type_list='["source", "target"]' \
#++jsonl_file_out=/Users/zhifu/funasr1.0/test_local/audio_datasets.jsonl
train_data="/nfs/maziyang.mzy/data/librispeech/librispeech_train_960h.jsonl"
val_data="/nfs/maziyang.mzy/data/librispeech/librispeech_dev_other_filtered.jsonl"
# exp output dir
output_dir="/nfs/zhifu.gzf/ckpt/exp/llm_asr_whisper_vicuna_exp1"
log_file="${output_dir}/log.txt"
workspace=`pwd`
config="whisper_vicuna_linear.yaml"
init_param="${output_dir}/model.pt"
mkdir -p ${output_dir}
echo "log_file: ${log_file}"
torchrun \
--nnodes 1 \
--nproc_per_node ${gpu_num} \
../../../funasr/bin/train.py \
--config-path "${workspace}/conf" \
--config-name "${config}" \
++train_data_set_list="${train_data}" \
++valid_data_set_list="${val_data}" \
++dataset_conf.batch_size=4 \
++dataset_conf.num_workers=4 \
++train_conf.max_epoch=15 \
++optim_conf.lr=0.0001 \
++init_param="${init_param}" \
++output_dir="${output_dir}" &> ${log_file} &
@@ -0,0 +1,42 @@
(简体中文|[English](./README.md))
# 语音识别
> **注意**:
> pipeline 支持 [modelscope模型仓库](https://alibaba-damo-academy.github.io/FunASR/en/model_zoo/modelscope_models.html#pretrained-models-on-modelscope) 中的所有模型进行推理和微调。这里我们以典型模型作为示例来演示使用方法。
## 推理
### 快速使用
#### [Paraformer 模型](https://www.modelscope.cn/models/damo/speech_paraformer-large_asr_nat-zh-cn-16k-common-vocab8404-pytorch/summary)
```python
from funasr import AutoModel
model = AutoModel(model="/Users/zhifu/Downloads/modelscope_models/speech_paraformer-large_asr_nat-zh-cn-16k-common-vocab8404-pytorch")
res = model(input="/Users/zhifu/Downloads/modelscope_models/speech_paraformer-large_asr_nat-zh-cn-16k-common-vocab8404-pytorch/example/asr_example.wav")
print(res)
```
### API接口说明
#### AutoModel 定义
- `model`: [模型仓库](https://alibaba-damo-academy.github.io/FunASR/en/model_zoo/modelscope_models.html#pretrained-models-on-modelscope) 中的模型名称,或本地磁盘中的模型路径
- `device`: `cuda`(默认),使用 GPU 进行推理。如果为`cpu`,则使用 CPU 进行推理
- `ncpu`: `None` (默认),设置用于 CPU 内部操作并行性的线程数
- `output_dir`: `None` (默认),如果设置,输出结果的输出路径
- `batch_size`: `1` (默认),解码时的批处理大小
#### AutoModel 推理
- `input`: 要解码的输入,可以是:
- wav文件路径, 例如: asr_example.wav
- pcm文件路径, 例如: asr_example.pcm,此时需要指定音频采样率fs(默认为16000)
- 音频字节数流,例如:麦克风的字节数数据
- wav.scp,kaldi 样式的 wav 列表 (`wav_id \t wav_path`), 例如:
```text
asr_example1 ./audios/asr_example1.wav
asr_example2 ./audios/asr_example2.wav
```
在这种输入 `wav.scp` 的情况下,必须设置 `output_dir` 以保存输出结果
- 音频采样点,例如:`audio, rate = soundfile.read("asr_example_zh.wav")`, 数据类型为 numpy.ndarray。支持batch输入,类型为list:
```[audio_sample1, audio_sample2, ..., audio_sampleN]```
- fbank输入,支持组batch。shape为[batch, frames, dim],类型为torch.Tensor,例如
- `output_dir`: None (默认),如果设置,输出结果的输出路径
@@ -0,0 +1,18 @@
#!/usr/bin/env python3
# -*- encoding: utf-8 -*-
# Copyright FunASR (https://github.com/alibaba-damo-academy/FunASR). All Rights Reserved.
# MIT License (https://opensource.org/licenses/MIT)
from funasr import AutoModel
model = AutoModel(model="iic/speech_timestamp_prediction-v1-16k-offline")
res = model.generate(
input=(
"https://isv-data.oss-cn-hangzhou.aliyuncs.com/ics/MaaS/ASR/test_audio/asr_example_zh.wav",
"欢迎大家来到魔搭社区进行体验",
),
data_type=("sound", "text"),
batch_size=2,
)
print(res)
@@ -0,0 +1,11 @@
model="iic/speech_timestamp_prediction-v1-16k-offline"
python funasr/bin/inference.py \
+model=${model} \
+input='["https://isv-data.oss-cn-hangzhou.aliyuncs.com/ics/MaaS/ASR/test_audio/asr_example_zh.wav", "欢迎大家来到魔搭社区进行体验"]' \
+data_type='["sound", "text"]' \
+output_dir="../outputs/debug" \
+device="cpu" \
+batch_size=2
@@ -0,0 +1,436 @@
(简体中文|[English](./README.md))
FunASR开源了大量在工业数据上预训练模型,您可以在 [模型许可协议](https://github.com/alibaba-damo-academy/FunASR/blob/main/MODEL_LICENSE)下自由使用、复制、修改和分享FunASR模型,下面列举代表性的模型,更多模型请参考 [模型仓库](https://github.com/alibaba-damo-academy/FunASR/tree/main/model_zoo)。
<div align="center">
<h4>
<a href="#模型推理"> 模型推理 </a>
|<a href="#模型训练与测试"> 模型训练与测试 </a>
|<a href="#模型导出与测试"> 模型导出与测试 </a>
</h4>
</div>
<a name="模型推理"></a>
## 模型推理
### 快速使用
命令行方式调用:
```shell
funasr ++model=paraformer-zh ++vad_model="fsmn-vad" ++punc_model="ct-punc" ++input=asr_example_zh.wav
```
python代码调用(推荐)
```python
from funasr import AutoModel
model = AutoModel(model="paraformer-zh")
res = model.generate(input="https://isv-data.oss-cn-hangzhou.aliyuncs.com/ics/MaaS/ASR/test_audio/vad_example.wav")
print(res)
```
### 接口说明
#### AutoModel 定义
```python
model = AutoModel(model=[str], device=[str], ncpu=[int], output_dir=[str], batch_size=[int], hub=[str], **kwargs)
```
- `model`(str): [模型仓库](https://github.com/alibaba-damo-academy/FunASR/tree/main/model_zoo) 中的模型名称,或本地磁盘中的模型路径
- `device`(str): `cuda:0`(默认gpu0),使用 GPU 进行推理,指定。如果为`cpu`,则使用 CPU 进行推理
- `ncpu`(int): `4` (默认),设置用于 CPU 内部操作并行性的线程数
- `output_dir`(str): `None` (默认),如果设置,输出结果的输出路径
- `batch_size`(int): `1` (默认),解码时的批处理,样本个数
- `hub`(str):`ms`(默认),从modelscope下载模型。如果为`hf`,从huggingface下载模型。
- `**kwargs`(dict): 所有在`config.yaml`中参数,均可以直接在此处指定,例如,vad模型中最大切割长度 `max_single_segment_time=6000` (毫秒)。
#### AutoModel 推理
```python
res = model.generate(input=[str], output_dir=[str])
```
- `input`: 要解码的输入,可以是:
- wav文件路径, 例如: asr_example.wav
- pcm文件路径, 例如: asr_example.pcm,此时需要指定音频采样率fs(默认为16000)
- 音频字节数流,例如:麦克风的字节数数据
- wav.scp,kaldi 样式的 wav 列表 (`wav_id \t wav_path`), 例如:
```text
asr_example1 ./audios/asr_example1.wav
asr_example2 ./audios/asr_example2.wav
```
在这种输入 `wav.scp` 的情况下,必须设置 `output_dir` 以保存输出结果
- 音频采样点,例如:`audio, rate = soundfile.read("asr_example_zh.wav")`, 数据类型为 numpy.ndarray。支持batch输入,类型为list:
```[audio_sample1, audio_sample2, ..., audio_sampleN]```
- fbank输入,支持组batch。shape为[batch, frames, dim],类型为torch.Tensor,例如
- `output_dir`: None (默认),如果设置,输出结果的输出路径
- `**kwargs`(dict): 与模型相关的推理参数,例如,`beam_size=10`,`decoding_ctc_weight=0.1`。
### 更多用法介绍
#### 非实时语音识别
```python
from funasr import AutoModel
# paraformer-zh is a multi-functional asr model
# use vad, punc, spk or not as you need
model = AutoModel(model="paraformer-zh",
vad_model="fsmn-vad",
vad_kwargs={"max_single_segment_time": 60000},
punc_model="ct-punc",
# spk_model="cam++"
)
wav_file = f"{model.model_path}/example/asr_example.wav"
res = model.generate(input=wav_file, batch_size_s=300, batch_size_threshold_s=60, hotword='魔搭')
print(res)
```
注意:
- 通常模型输入限制时长30s以下,组合`vad_model`后,支持任意时长音频输入,不局限于paraformer模型,所有音频输入模型均可以。
- `model`相关的参数可以直接在`AutoModel`定义中直接指定;与`vad_model`相关参数可以通过`vad_kwargs`来指定,类型为dict;类似的有`punc_kwargs`,`spk_kwargs`;
- `max_single_segment_time`: 表示`vad_model`最大切割音频时长, 单位是毫秒ms.
- `batch_size_s` 表示采用动态batch,batch中总音频时长,单位为秒s。
- `batch_size_threshold_s`: 表示`vad_model`切割后音频片段时长超过 `batch_size_threshold_s`阈值时,将batch_size数设置为1, 单位为秒s.
建议:当您输入为长音频,遇到OOM问题时,因为显存占用与音频时长呈平方关系增加,分为3种情况:
- a)推理起始阶段,显存主要取决于`batch_size_s`,适当减小该值,可以减少显存占用;
- b)推理中间阶段,遇到VAD切割的长音频片段,总token数小于`batch_size_s`,仍然出现OOM,可以适当减小`batch_size_threshold_s`,超过阈值,强制batch为1;
- c)推理快结束阶段,遇到VAD切割的长音频片段,总token数小于`batch_size_s`,且超过阈值`batch_size_threshold_s`,强制batch为1,仍然出现OOM,可以适当减小`max_single_segment_time`,使得VAD切割音频时长变短。
#### 实时语音识别
```python
from funasr import AutoModel
chunk_size = [0, 10, 5] #[0, 10, 5] 600ms, [0, 8, 4] 480ms
encoder_chunk_look_back = 4 #number of chunks to lookback for encoder self-attention
decoder_chunk_look_back = 1 #number of encoder chunks to lookback for decoder cross-attention
model = AutoModel(model="paraformer-zh-streaming")
import soundfile
import os
wav_file = os.path.join(model.model_path, "example/asr_example.wav")
speech, sample_rate = soundfile.read(wav_file)
chunk_stride = chunk_size[1] * 960 # 600ms
cache = {}
total_chunk_num = int(len((speech)-1)/chunk_stride+1)
for i in range(total_chunk_num):
speech_chunk = speech[i*chunk_stride:(i+1)*chunk_stride]
is_final = i == total_chunk_num - 1
res = model.generate(input=speech_chunk, cache=cache, is_final=is_final, chunk_size=chunk_size, encoder_chunk_look_back=encoder_chunk_look_back, decoder_chunk_look_back=decoder_chunk_look_back)
print(res)
```
注:`chunk_size`为流式延时配置,`[0,10,5]`表示上屏实时出字粒度为`10*60=600ms`,未来信息为`5*60=300ms`。每次推理输入为`600ms`(采样点数为`16000*0.6=960`),输出为对应文字,最后一个语音片段输入需要设置`is_final=True`来强制输出最后一个字。
#### 语音端点检测(非实时)
```python
from funasr import AutoModel
model = AutoModel(model="fsmn-vad")
wav_file = f"{model.model_path}/example/vad_example.wav"
res = model.generate(input=wav_file)
print(res)
```
注:VAD模型输出格式为:`[[beg1, end1], [beg2, end2], .., [begN, endN]]`,其中`begN/endN`表示第`N`个有效音频片段的起始点/结束点,
单位为毫秒。
#### 语音端点检测(实时)
```python
from funasr import AutoModel
chunk_size = 200 # ms
model = AutoModel(model="fsmn-vad")
import soundfile
wav_file = f"{model.model_path}/example/vad_example.wav"
speech, sample_rate = soundfile.read(wav_file)
chunk_stride = int(chunk_size * sample_rate / 1000)
cache = {}
total_chunk_num = int(len((speech)-1)/chunk_stride+1)
for i in range(total_chunk_num):
speech_chunk = speech[i*chunk_stride:(i+1)*chunk_stride]
is_final = i == total_chunk_num - 1
res = model.generate(input=speech_chunk, cache=cache, is_final=is_final, chunk_size=chunk_size)
if len(res[0]["value"]):
print(res)
```
注:流式VAD模型输出格式为4种情况:
- `[[beg1, end1], [beg2, end2], .., [begN, endN]]`:同上离线VAD输出结果。
- `[[beg, -1]]`:表示只检测到起始点。
- `[[-1, end]]`:表示只检测到结束点。
- `[]`:表示既没有检测到起始点,也没有检测到结束点
输出结果单位为毫秒,从起始点开始的绝对时间。
#### 标点恢复
```python
from funasr import AutoModel
model = AutoModel(model="ct-punc")
res = model.generate(input="那今天的会就到这里吧 happy new year 明年见")
print(res)
```
#### 时间戳预测
```python
from funasr import AutoModel
model = AutoModel(model="fa-zh")
wav_file = f"{model.model_path}/example/asr_example.wav"
text_file = f"{model.model_path}/example/text.txt"
res = model.generate(input=(wav_file, text_file), data_type=("sound", "text"))
print(res)
```
更多([示例](https://github.com/alibaba-damo-academy/FunASR/tree/main/examples/industrial_data_pretraining))
<a name="核心功能"></a>
## 模型训练与测试
### 快速开始
命令行执行(用于快速测试,不推荐):
```shell
funasr-train ++model=paraformer-zh ++train_data_set_list=data/list/train.jsonl ++valid_data_set_list=data/list/val.jsonl ++output_dir="./outputs" &> log.txt &
```
python代码执行(可以多机多卡,推荐)
```shell
cd examples/industrial_data_pretraining/paraformer
bash finetune.sh
# "log_file: ./outputs/log.txt"
```
详细完整的脚本参考 [finetune.sh](https://github.com/alibaba-damo-academy/FunASR/blob/main/examples/industrial_data_pretraining/paraformer/finetune.sh)
### 详细参数介绍
```shell
funasr/bin/train.py \
++model="${model_name_or_model_dir}" \
++train_data_set_list="${train_data}" \
++valid_data_set_list="${val_data}" \
++dataset_conf.batch_size=20000 \
++dataset_conf.batch_type="token" \
++dataset_conf.num_workers=4 \
++train_conf.max_epoch=50 \
++train_conf.log_interval=1 \
++train_conf.resume=false \
++train_conf.validate_interval=2000 \
++train_conf.save_checkpoint_interval=2000 \
++train_conf.keep_nbest_models=20 \
++train_conf.avg_nbest_model=10 \
++optim_conf.lr=0.0002 \
++output_dir="${output_dir}" &> ${log_file}
```
- `model`(str):模型名字(模型仓库中的ID),此时脚本会自动下载模型到本读;或者本地已经下载好的模型路径。
- `train_data_set_list`(str):训练数据路径,默认为jsonl格式,具体参考([例子](https://github.com/alibaba-damo-academy/FunASR/blob/main/data/list))。
- `valid_data_set_list`(str):验证数据路径,默认为jsonl格式,具体参考([例子](https://github.com/alibaba-damo-academy/FunASR/blob/main/data/list))。
- `dataset_conf.batch_type`(str):`example`(默认),batch的类型。`example`表示按照固定数目batch_size个样本组batch;`length` or `token` 表示动态组batch,batch总长度或者token数为batch_size。
- `dataset_conf.batch_size`(int):与 `batch_type` 搭配使用,当 `batch_type=example` 时,表示样本个数;当 `batch_type=length` 时,表示样本中长度,单位为fbank帧数(1帧10ms)或者文字token个数。
- `train_conf.max_epoch`(int):`100`(默认),训练总epoch数。
- `train_conf.log_interval`(int):`50`(默认),打印日志间隔step数。
- `train_conf.resume`(int):`True`(默认),是否开启断点重训。
- `train_conf.validate_interval`(int):`5000`(默认),训练中做验证测试的间隔step数。
- `train_conf.save_checkpoint_interval`(int):`5000`(默认),训练中模型保存间隔step数。
- `train_conf.avg_keep_nbest_models_type`(str):`acc`(默认),保留nbest的标准为acc(越大越好)。`loss`表示,保留nbest的标准为loss(越小越好)。
- `train_conf.keep_nbest_models`(int):`500`(默认),保留最大多少个模型参数,配合 `avg_keep_nbest_models_type` 按照验证集 acc/loss 保留最佳的n个模型,其他删除,节约存储空间。
- `train_conf.avg_nbest_model`(int):`10`(默认),保留最大多少个模型参数,配合 `avg_keep_nbest_models_type` 按照验证集 acc/loss 对最佳的n个模型平均。
- `train_conf.accum_grad`(int):`1`(默认),梯度累积功能。
- `train_conf.grad_clip`(float):`10.0`(默认),梯度截断功能。
- `train_conf.use_fp16`(bool):`False`(默认),开启fp16训练,加快训练速度。
- `optim_conf.lr`(float):学习率。
- `output_dir`(str):模型保存路径。
- `**kwargs`(dict): 所有在`config.yaml`中参数,均可以直接在此处指定,例如,过滤20s以上长音频:`dataset_conf.max_token_length=2000`,单位为音频fbank帧数(1帧10ms)或者文字token个数。
#### 多gpu训练
##### 单机多gpu训练
```shell
export CUDA_VISIBLE_DEVICES="0,1"
gpu_num=$(echo $CUDA_VISIBLE_DEVICES | awk -F "," '{print NF}')
torchrun --nnodes 1 --nproc_per_node ${gpu_num} \
../../../funasr/bin/train.py ${train_args}
```
--nnodes 表示参与的节点总数,--nproc_per_node 表示每个节点上运行的进程数
##### 多机多gpu训练
在主节点上,假设IP为192.168.1.1,端口为12345,使用的是2个GPU,则运行如下命令:
```shell
export CUDA_VISIBLE_DEVICES="0,1"
gpu_num=$(echo $CUDA_VISIBLE_DEVICES | awk -F "," '{print NF}')
torchrun --nnodes 2 --node_rank 0 --nproc_per_node ${gpu_num} --master_addr 192.168.1.1 --master_port 12345 \
../../../funasr/bin/train.py ${train_args}
```
在从节点上(假设IP为192.168.1.2),你需要确保MASTER_ADDR和MASTER_PORT环境变量与主节点设置的一致,并运行同样的命令:
```shell
export CUDA_VISIBLE_DEVICES="0,1"
gpu_num=$(echo $CUDA_VISIBLE_DEVICES | awk -F "," '{print NF}')
torchrun --nnodes 2 --node_rank 1 --nproc_per_node ${gpu_num} --master_addr 192.168.1.1 --master_port 12345 \
../../../funasr/bin/train.py ${train_args}
```
--nnodes 表示参与的节点总数,--node_rank 表示当前节点id,--nproc_per_node 表示每个节点上运行的进程数(通常为gpu个数)
#### 准备数据
`jsonl`格式可以参考([例子](https://github.com/alibaba-damo-academy/FunASR/blob/main/data/list))。
可以用指令 `scp2jsonl` 从wav.scp与text.txt生成。wav.scp与text.txt准备过程如下:
`train_text.txt`
左边为数据唯一ID,需与`train_wav.scp`中的`ID`一一对应
右边为音频文件标注文本,格式如下:
```bash
ID0012W0013 当客户风险承受能力评估依据发生变化时
ID0012W0014 所有只要处理 data 不管你是做 machine learning 做 deep learning
ID0012W0015 he tried to think how it could be
```
`train_wav.scp`
左边为数据唯一ID,需与`train_text.txt`中的`ID`一一对应
右边为音频文件的路径,格式如下
```bash
BAC009S0764W0121 https://isv-data.oss-cn-hangzhou.aliyuncs.com/ics/MaaS/ASR/test_audio/BAC009S0764W0121.wav
BAC009S0916W0489 https://isv-data.oss-cn-hangzhou.aliyuncs.com/ics/MaaS/ASR/test_audio/BAC009S0916W0489.wav
ID0012W0015 https://isv-data.oss-cn-hangzhou.aliyuncs.com/ics/MaaS/ASR/test_audio/asr_example_cn_en.wav
```
`生成指令`
```shell
# 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="../../../data/list/train.jsonl"
```
(可选,非必需)如果需要从jsonl解析成wav.scp与text.txt,可以使用指令:
```shell
# generate wav.scp and text.txt from train.jsonl and val.jsonl
jsonl2scp \
++scp_file_list='["../../../data/list/train_wav.scp", "../../../data/list/train_text.txt"]' \
++data_type_list='["source", "target"]' \
++jsonl_file_in="../../../data/list/train.jsonl"
```
#### 查看训练日志
##### 查看实验log
```shell
tail log.txt
[2024-03-21 15:55:52,137][root][INFO] - train, rank: 3, epoch: 0/50, step: 6990/1, total step: 6990, (loss_avg_rank: 0.327), (loss_avg_epoch: 0.409), (ppl_avg_epoch: 1.506), (acc_avg_epoch: 0.795), (lr: 1.165e-04), [('loss_att', 0.259), ('acc', 0.825), ('loss_pre', 0.04), ('loss', 0.299), ('batch_size', 40)], {'data_load': '0.000', 'forward_time': '0.315', 'backward_time': '0.555', 'optim_time': '0.076', 'total_time': '0.947'}, GPU, memory: usage: 3.830 GB, peak: 18.357 GB, cache: 20.910 GB, cache_peak: 20.910 GB
[2024-03-21 15:55:52,139][root][INFO] - train, rank: 1, epoch: 0/50, step: 6990/1, total step: 6990, (loss_avg_rank: 0.334), (loss_avg_epoch: 0.409), (ppl_avg_epoch: 1.506), (acc_avg_epoch: 0.795), (lr: 1.165e-04), [('loss_att', 0.285), ('acc', 0.823), ('loss_pre', 0.046), ('loss', 0.331), ('batch_size', 36)], {'data_load': '0.000', 'forward_time': '0.334', 'backward_time': '0.536', 'optim_time': '0.077', 'total_time': '0.948'}, GPU, memory: usage: 3.943 GB, peak: 18.291 GB, cache: 19.619 GB, cache_peak: 19.619 GB
```
指标解释:
- `rank`:表示gpu id。
- `epoch`,`step`,`total step`:表示当前epoch,step,总step。
- `loss_avg_rank`:表示当前step,所有gpu平均loss。
- `loss/ppl/acc_avg_epoch`:表示当前epoch周期,截止当前step数时,总平均loss/ppl/acc。epoch结束时的最后一个step表示epoch总平均loss/ppl/acc,推荐使用acc指标。
- `lr`:当前step的学习率。
- `[('loss_att', 0.259), ('acc', 0.825), ('loss_pre', 0.04), ('loss', 0.299), ('batch_size', 40)]`:表示当前gpu id的具体数据。
- `total_time`:表示单个step总耗时。
- `GPU, memory`:分别表示,模型使用/峰值显存,模型+缓存使用/峰值显存。
##### tensorboard可视化
```bash
tensorboard --logdir /xxxx/FunASR/examples/industrial_data_pretraining/paraformer/outputs/log/tensorboard
```
浏览器中打开:http://localhost:6006/
### 训练后模型测试
#### 有configuration.json
假定,训练模型路径为:./model_dir,如果改目录下有生成configuration.json,只需要将 [上述模型推理方法](https://github.com/alibaba-damo-academy/FunASR/blob/main/examples/README_zh.md#%E6%A8%A1%E5%9E%8B%E6%8E%A8%E7%90%86) 中模型名字修改为模型路径即可
例如:
从shell推理
```shell
python -m funasr.bin.inference ++model="./model_dir" ++input=="${input}" ++output_dir="${output_dir}"
```
从python推理
```python
from funasr import AutoModel
model = AutoModel(model="./model_dir")
res = model.generate(input=wav_file)
print(res)
```
#### 无configuration.json时
如果模型路径中无configuration.json时,需要手动指定具体配置文件路径与模型路径
```shell
python -m funasr.bin.inference \
--config-path "${local_path}" \
--config-name "${config}" \
++init_param="${init_param}" \
++tokenizer_conf.token_list="${tokens}" \
++frontend_conf.cmvn_file="${cmvn_file}" \
++input="${input}" \
++output_dir="${output_dir}" \
++device="${device}"
```
参数介绍
- `config-path`:为实验中保存的 `config.yaml`,可以从实验输出目录中查找。
- `config-name`:配置文件名,一般为 `config.yaml`,支持yaml格式与json格式,例如 `config.json`
- `init_param`:需要测试的模型参数,一般为`model.pt`,可以自己选择具体的模型文件
- `tokenizer_conf.token_list`:词表文件路径,一般在 `config.yaml` 有指定,无需再手动指定,当 `config.yaml` 中路径不正确时,需要在此处手动指定。
- `frontend_conf.cmvn_file`:wav提取fbank中用到的cmvn文件,一般在 `config.yaml` 有指定,无需再手动指定,当 `config.yaml` 中路径不正确时,需要在此处手动指定。
其他参数同上,完整 [示例](https://github.com/alibaba-damo-academy/FunASR/blob/main/examples/industrial_data_pretraining/paraformer/infer_from_local.sh)
<a name="模型导出与测试"></a>
## 模型导出与测试
### 从命令行导出
```shell
funasr-export ++model=paraformer ++quantize=false
```
### 从Python导出
```python
from funasr import AutoModel
model = AutoModel(model="paraformer")
res = model.export(quantize=False)
```
### 测试ONNX
```python
# pip3 install -U funasr-onnx
from funasr_onnx import Paraformer
model_dir = "damo/speech_paraformer-large_asr_nat-zh-cn-16k-common-vocab8404-pytorch"
model = Paraformer(model_dir, batch_size=1, quantize=True)
wav_path = ['~/.cache/modelscope/hub/damo/speech_paraformer-large_asr_nat-zh-cn-16k-common-vocab8404-pytorch/example/asr_example.wav']
result = model(wav_path)
print(result)
```
更多例子请参考 [样例](https://github.com/alibaba-damo-academy/FunASR/tree/main/runtime/python/onnxruntime)
Whitespace-only changes.
@@ -0,0 +1,19 @@
#!/usr/bin/env python3
# -*- encoding: utf-8 -*-
# Copyright FunASR (https://github.com/alibaba-damo-academy/FunASR). All Rights Reserved.
# MIT License (https://opensource.org/licenses/MIT)
from funasr import AutoModel
model = AutoModel(
model="iic/speech_seaco_paraformer_large_asr_nat-zh-cn-16k-common-vocab8404-pytorch",
vad_model="iic/speech_fsmn_vad_zh-cn-16k-common-pytorch",
punc_model="iic/punc_ct-transformer_zh-cn-common-vocab272727-pytorch",
spk_model="iic/speech_campplus_sv_zh-cn_16k-common",
)
res = model.generate(
input="https://isv-data.oss-cn-hangzhou.aliyuncs.com/ics/MaaS/ASR/test_audio/asr_example_zh.wav",
hotword="达摩院 磨搭",
)
print(res)
@@ -0,0 +1,16 @@
model="iic/speech_seaco_paraformer_large_asr_nat-zh-cn-16k-common-vocab8404-pytorch"
vad_model="iic/speech_fsmn_vad_zh-cn-16k-common-pytorch"
#punc_model="iic/punc_ct-transformer_zh-cn-common-vocab272727-pytorch"
punc_model="iic/punc_ct-transformer_cn-en-common-vocab471067-large"
spk_model="iic/speech_campplus_sv_zh-cn_16k-common"
python funasr/bin/inference.py \
++model=${model} \
++vad_model=${vad_model} \
++punc_model=${punc_model} \
++spk_model=${spk_model} \
++input="https://isv-data.oss-cn-hangzhou.aliyuncs.com/ics/MaaS/ASR/test_audio/asr_example_zh.wav" \
++output_dir="./outputs/debug" \
++device="cpu" \
++"hotword='达摩院 魔搭'"
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