diff --git a/LICENSE b/LICENSE
deleted file mode 100644
index e3b92c4ab..000000000
--- a/LICENSE
+++ /dev/null
@@ -1,73 +0,0 @@
-FunASR 模型开源协议
-
-版本号：1.0
-
-版权所有 (C) [2023-2028] [阿里巴巴集团]。保留所有权利。
-
-感谢您选择 FunASR 开源模型。FunASR 开源模型是一款免费且开源的语音模型，让大家可以使用、修改、分享和学习该模型。
-
-为了保证更好的社区合作，我们制定了以下协议，希望您仔细阅读并遵守本协议。
-
-1 定义
-本协议中，FunASR 指 FunASR 开源代码、模型及其衍生品，包括您finetune后的模型；您指使用、修改、分享和学习 FunASR 的个人或组织。
-
-2 许可和限制
-2.1 许可
-
-您可以在遵守本协议的前提下，自由地使用、复制、修改和分享 FunASR。
-
-2.2 限制
-
-您在使用、复制、修改和分享 FunASR 时，必须注明代码与模型出处以及作者信息。并且，将 FunASR 工业模型上传至任何第三方平台以供下载和使用，需要获得额外许可，可通过官方邮件（funasr@list.alibaba-inc.com）进行申请（免费）。
-
-3 责任和风险承担
-FunASR 仅作为参考和学习使用，不对您使用或修改 FunASR 造成的任何直接或间接损失承担任何责任。您对 FunASR 的使用和修改应该自行承担风险。
-
-4 终止
-如果您违反本协议的任何条款，您的许可将自动终止，您必须停止使用、复制、修改和分享 FunASR。
-
-5 修订
-本协议可能会不时更新和修订。修订后的协议将在 FunASR 官方仓库发布，并自动生效。如果您继续使用、复制、修改和分享 FunASR，即表示您同意修订后的协议。
-
-6 其他规定
-本协议受到 [国家/地区] 的法律管辖。如果任何条款被裁定为不合法、无效或无法执行，则该条款应被视为从本协议中删除，而其余条款应继续有效并具有约束力。
-
-如果您对本协议有任何问题或意见，请联系我们。
-
-版权所有© [2023-2028] [阿里巴巴集团]。保留所有权利。
-
-FunASR Model Open Source License
-Version 1.0
-
-Copyright (C) [2023-2028] Alibaba Group. All rights reserved.
-
-Thank you for choosing the FunASR open source model. The FunASR open source model is a free and open-source speech model that allows everyone to use, modify, share, and learn from it.
-
-To ensure better community collaboration, we have developed the following agreement and hope that you carefully read and abide by it.
-
-1 Definitions
-In this agreement, FunASR refers to the FunASR open source code, model, and its derivatives, including your fine-tuned models. You refer to individuals or organizations who use, modify, share, and learn from FunASR.
-
-2 License and Restrictions
-
-2.1 License
-You are free to use, copy, modify, and share FunASR under the conditions of this agreement.
-
-2.2 Restrictions
-You should indicate the code and model source and author information when using, copying, modifying and sharing FunASR. To upload the FunASR industrial model to any third-party platform for download and use, an additional license is required, which can be applied for free by sending an email to the official email address (funasr@list.alibaba-inc.com).
-
-3 Responsibility and Risk
-FunASR is for reference and learning purposes only and is not responsible for any direct or indirect losses caused by your use or modification of FunASR. You should take responsibility and risks for your use and modification of FunASR.
-
-4 Termination
-If you violate any terms of this agreement, your license will be automatically terminated, and you must stop using, copying, modifying, and sharing FunASR.
-
-5 Revision
-This agreement may be updated and revised from time to time. The revised agreement will be published in the FunASR official repository and automatically take effect. If you continue to use, copy, modify, and share FunASR, it means you agree to the revised agreement.
-
-6 Other Provisions
-This agreement is subject to the laws of [Country/Region]. If any provisions are found to be illegal, invalid, or unenforceable, they shall be deemed deleted from this agreement, and the remaining provisions shall remain valid and binding.
-
-If you have any questions or comments about this agreement, please contact us.
-
-Copyright (c) [2023-2028] Alibaba Group. All rights reserved.
\ No newline at end of file
diff --git a/MODEL_LICENSE b/MODEL_LICENSE
new file mode 100644
index 000000000..e50598012
--- /dev/null
+++ b/MODEL_LICENSE
@@ -0,0 +1,73 @@
+FunASR 模型开源协议
+
+版本号：1.0
+
+版权所有 (C) [2023-2028] [阿里巴巴集团]。保留所有权利。
+
+感谢您选择 FunASR 开源模型。FunASR 开源模型包含一系列免费且开源的工业模型，让大家可以使用、修改、分享和学习该模型。
+
+为了保证更好的社区合作，我们制定了以下协议，希望您仔细阅读并遵守本协议。
+
+1 定义
+本协议中，[FunASR 软件]指 FunASR 开源模型及其衍生品，包括 Finetune 后的模型；[您]指使用、修改、分享和学习[FunASR 软件]的个人或组织。
+
+2 许可和限制
+2.1 许可
+
+您可以在遵守本协议的前提下，自由地使用、复制、修改和分享[FunASR 软件]。
+
+2.2 限制
+
+您在使用、复制、修改和分享[FunASR 软件]时，必须注明出处以及作者信息。并且，将[FunASR 软件]上传至其他第三方平台以供下载，需要获得额外许可，可通过官方邮件（funasr@list.alibaba-inc.com）进行申请（免费）。
+
+3 责任和风险承担
+[FunASR 软件]仅作为参考和学习使用，不对您使用或修改[FunASR 软件]造成的任何直接或间接损失承担任何责任。您对[FunASR 软件]的使用和修改应该自行承担风险。
+
+4 终止
+如果您违反本协议的任何条款，您的许可将自动终止，您必须停止使用、复制、修改和分享[FunASR 软件]。
+
+5 修订
+本协议可能会不时更新和修订。修订后的协议将在[FunASR 软件]官方仓库发布，并自动生效。如果您继续使用、复制、修改和分享[FunASR 软件]，即表示您同意修订后的协议。
+
+6 其他规定
+本协议受到[国家/地区] 的法律管辖。如果任何条款被裁定为不合法、无效或无法执行，则该条款应被视为从本协议中删除，而其余条款应继续有效并具有约束力。
+
+如果您对本协议有任何问题或意见，请联系我们。
+
+版权所有© [2023-2028] [阿里巴巴集团]。保留所有权利。
+
+FunASR Model Open Source License
+Version 1.0
+
+Copyright (C) [2023-2028] Alibaba Group. All rights reserved.
+
+Thank you for choosing the FunASR open source models. The FunASR open source models contain a series of open-source models that allow everyone to use, modify, share, and learn from it.
+
+To ensure better community collaboration, we have developed the following agreement and hope that you carefully read and abide by it.
+
+1 Definitions
+In this agreement, [FunASR software] refers to the FunASR open source model, and its derivatives, including fine-tuned models. [You] refer to individuals or organizations who use, modify, share, and learn from [FunASR software].
+
+2 License and Restrictions
+
+2.1 License
+You are free to use, copy, modify, and share [FunASR software] under the conditions of this agreement.
+
+2.2 Restrictions
+You should indicate the code and model source and author information when using, copying, modifying and sharing [FunASR software]. To upload the [FunASR software] to other third-party platforms for download, an additional license is required, which can be applied for free by sending an email to the official email address (funasr@list.alibaba-inc.com).
+
+3 Responsibility and Risk
+[FunASR software] is for reference and learning purposes only and is not responsible for any direct or indirect losses caused by your use or modification of [FunASR software]. You should take responsibility and risks for your use and modification of [FunASR software].
+
+4 Termination
+If you violate any terms of this agreement, your license will be automatically terminated, and you must stop using, copying, modifying, and sharing [FunASR software].
+
+5 Revision
+This agreement may be updated and revised from time to time. The revised agreement will be published in the FunASR official repository and automatically take effect. If you continue to use, copy, modify, and share [FunASR software], it means you agree to the revised agreement.
+
+6 Other Provisions
+This agreement is subject to the laws of [Country/Region]. If any provisions are found to be illegal, invalid, or unenforceable, they shall be deemed deleted from this agreement, and the remaining provisions shall remain valid and binding.
+
+If you have any questions or comments about this agreement, please contact us.
+
+Copyright (c) [2023-2028] Alibaba Group. All rights reserved.
diff --git a/README.md b/README.md
index 1fcbcdd96..c31d6168c 100644
--- a/README.md
+++ b/README.md
@@ -88,6 +88,7 @@ If you have any questions about FunASR, please contact us by
## License
This project is licensed under the [The MIT License](https://opensource.org/licenses/MIT). FunASR also contains various third-party components and some code modified from other repos under other open source licenses.
+The use of pretraining model is subject to [model licencs](./MODEL_LICENSE)
## Citations
diff --git a/docs/academic_recipe/asr_recipe.md b/docs/academic_recipe/asr_recipe.md
index 611976f06..5a11dc5ee 100644
--- a/docs/academic_recipe/asr_recipe.md
+++ b/docs/academic_recipe/asr_recipe.md
@@ -37,7 +37,7 @@ tensorboard --logdir ${exp_dir}/exp/${model_dir}/tensorboard/train
```
Here is an example of loss:
-
+
The inference results are saved in `${exp_dir}/exp/${model_dir}/decode_asr_*/$dset`. The main two files are `text.cer` and `text.cer.txt`. `text.cer` saves the comparison between the recognized text and the reference text, like follows:
```text
@@ -106,18 +106,37 @@ This stage processes the dictionary, which is used as a mapping between label ch
龟
```
-* ``: indicates the blank token for CTC, must be in the first line
-* ``: indicates the start-of-sentence token, must be in the second line
-* ``: indicates the end-of-sentence token, must be in the third line
-* ``: indicates the out-of-vocabulary token, must be in the last line
+There are four tokens must be specified:
+* ``: (required), indicates the blank token for CTC, must be in the first line
+* ``: (required), indicates the start-of-sentence token, must be in the second line
+* ``: (required), indicates the end-of-sentence token, must be in the third line
+* ``: (required), indicates the out-of-vocabulary token, must be in the last line
### Stage 3: LM Training
### Stage 4: ASR Training
-This stage achieves the training of the specified model. To start training, users should manually set `exp_dir` to specify the path for saving experimental results. By default, the best `$keep_nbest_models` checkpoints on validation dataset will be averaged to generate a better model and adopted for decoding. FunASR implements `train.py` for training different models and users can configure the following parameters if necessary.
+This stage achieves the training of the specified model. To start training, users should manually set `exp_dir` to specify the path for saving experimental results. By default, the best `$keep_nbest_models` checkpoints on validation dataset will be averaged to generate a better model and adopted for decoding. FunASR implements `train.py` for training different models and users can configure the following parameters if necessary. The training command is as follows:
+
+```sh
+train.py \
+ --task_name asr \
+ --use_preprocessor true \
+ --token_list $token_list \
+ --data_dir ${feats_dir}/data \
+ --train_set ${train_set} \
+ --valid_set ${valid_set} \
+ --data_file_names "wav.scp,text" \
+ --cmvn_file ${feats_dir}/data/${train_set}/cmvn/am.mvn \
+ --speed_perturb ${speed_perturb} \
+ --resume true \
+ --output_dir ${exp_dir}/exp/${model_dir} \
+ --config $asr_config \
+ --ngpu $gpu_num \
+ ...
+```
* `task_name`: `asr` (Default), specify the task type of the current recipe
-* `gpu_num`: `2` (Default), specify the number of GPUs for training. When `gpu_num > 1`, DistributedDataParallel (DDP, the detail can be found [here](https://pytorch.org/tutorials/intermediate/ddp_tutorial.html)) training will be enabled. Correspondingly, `CUDA_VISIBLE_DEVICES` should be set to specify which ids of GPUs will be used.
+* `ngpu`: `2` (Default), specify the number of GPUs for training. When `ngpu > 1`, DistributedDataParallel (DDP, the detail can be found [here](https://pytorch.org/tutorials/intermediate/ddp_tutorial.html)) training will be enabled. Correspondingly, `CUDA_VISIBLE_DEVICES` should be set to specify which ids of GPUs will be used.
* `use_preprocessor`: `true` (Default), specify whether to use pre-processing on each sample
* `token_list`: the path of token list for training
* `dataset_type`: `small` (Default). FunASR supports `small` dataset type for training small datasets. Besides, an optional iterable-style DataLoader based on [Pytorch Iterable-style DataPipes](https://pytorch.org/data/beta/torchdata.datapipes.iter.html) for large datasets is supported and users can specify `dataset_type=large` to enable it.
@@ -125,6 +144,7 @@ This stage achieves the training of the specified model. To start training, user
* `data_file_names`: `"wav.scp,text"` specify the speech and text file names for ASR
* `cmvn_file`: the path of cmvn file
* `resume`: `true`, whether to enable "checkpoint training"
+* `output_dir`: the path for saving training results
* `config`: the path of configuration file, which is usually a YAML file in `conf` directory. In FunASR, the parameters of the training, including model, optimization, dataset, etc., can also be set in this file. Note that if the same parameters are specified in both recipe and config file, the parameters of recipe will be employed
### Stage 5: Decoding
@@ -156,25 +176,25 @@ res: 构 建 良 好 的 旅 游 市 场 环 境
## Change settings
Here we explain how to perform common custom settings, which can help users to modify scripts according to their own needs.
-* Training with specified GPUs
+### Training with specified GPUs
-For example, if users want to use 2 GPUs with id `2` and `3, users can run the following command:
+For example, if users want to use 2 GPUs with id `2` and `3`, users can run the following command:
```sh
. ./run.sh --CUDA_VISIBLE_DEVICES "2,3" --gpu_num 2
```
-* Start from/Stop at a specified stage
+### Start from/Stop at a specified stage
The recipe includes several stages. Users can start form or stop at any stage. For example, the following command achieves starting from the third stage and stopping at the fifth stage:
```sh
. ./run.sh --stage 3 --stop_stage 5
```
-* Training Steps
+### Specify total training steps
FunASR supports two parameters to specify the training steps, namely `max_epoch` and `max_update`. `max_epoch` indicates the total training epochs while `max_update` indicates the total training steps. If these two parameters are specified at the same time, once the training reaches any one of these two parameters, the training will be stopped.
-* Change the configuration of the model
+### Change the configuration of the model
The configuration of the model is set in the config file `conf/train_*.yaml`. Specifically, the default encoder configuration of paraformer is as follows:
```
@@ -199,28 +219,49 @@ encoder_conf:
```
Users can change the encoder configuration by modify these values. For example, if users want to use an encoder with 16 conformer blocks and each block has 8 attention heads, users just need to change `num_blocks` from 12 to 16 and change `attention_heads` from 4 to 8. Besides, the batch_size, learning rate and other training hyper-parameters are also set in this config file. To change these hyper-parameters, users just need to directly change the corresponding values in this file. For example, the default learning rate is `0.0005`. If users want to change the learning rate to 0.0002, set the value of lr as `lr: 0.0002`.
-* Use different input data type
+### Change different input data type
-FunASR supports different input data types, including `sound`, `kaldi_ark`, `npy`, `text` and `text_int`. Users can specify any number and any type of input, which is achieved by `data_file_names` (in `run.sh`), `data_names` and `data_types` (in config file). For example, ASR task usually requires speech and the corresponding transcripts as input. If speech is saved as raw audio (such as wav format) and transcripts are saved as text format, users need to set `data_file_names=wav.scp,text` (any name is allowed, denoting wav list and text list), set `data_names=speech,text` and set `data_types=sound,text`. When the input type changes to FBank, users just need to modify `data_types=kaldi_ark,text`.
+FunASR supports different input data types, including `sound`, `kaldi_ark`, `npy`, `text` and `text_int`. Users can specify any number and any type of input, which is achieved by `data_names` and `data_types` (in `config/train_*.yaml`). For example, ASR task usually requires speech and the transcripts as input. In FunASR, by default, speech is saved as raw audio (such as wav format) and transcripts are saved as text format. Correspondingly, `data_names` and `data_types` are set as follows (seen in `config/train_*.yaml`):
+```text
+dataset_conf:
+ data_names: speech,text
+ data_types: sound,text
+ ...
+```
+When the input type changes to FBank, users just need to modify as `data_types: kaldi_ark,text` in the config file. Note `data_file_names` used in `train.py` should also be changed to the new file name.
-* How to start from pre-trained models
-
-Users can start training from a pre-trained model by specifying the `init_param` parameter. Here `init_param` indicates the path of the pre-trained model. In addition to directly loading all the parameters from one pre-trained model, loading part of the parameters from different pre-trained models is supported. For example, to load encoder parameters from the pre-trained model A and decoder parameters from the pre-trained model B, users can set `init_param` twice as follows:
-```sh
-train.py ... --init_param ${model_A_path}:encoder --init_param ${model_B_path}:decoder ...
+### How to resume training process
+FunASR supports resuming training as follows:
+```shell
+train.py ... --resume true ...
```
-* How to freeze part model parameters
+### How to transfer / fine-tuning from pre-trained models
+
+FunASR supports transferring / fine-tuning from a pre-trained model by specifying the `init_param` parameter. The usage format is as follows:
+```shell
+train.py ... --init_param ::: ..
+```
+For example, the following command achieves loading all pretrained parameters starting from decoder except decoder.embed and set it to model.decoder2:
+```shell
+train.py ... --init_param model.pb:decoder:decoder2:decoder.embed ...
+```
+Besides, loading parameters from multiple pre-trained models is supported. For example, the following command achieves loading encoder parameters from the pre-trained model1 and decoder parameters from the pre-trained model2:
+```sh
+train.py ... --init_param model1.pb:encoder --init_param model2.pb:decoder ...
+```
+
+### How to freeze part of the model parameters
In certain situations, users may want to fix part of the model parameters update the rest model parameters. FunASR employs `freeze_param` to achieve this. For example, to fix all parameters like `encoder.*`, users need to set `freeze_param ` as follows:
```sh
train.py ... --freeze_param encoder ...
```
-* ModelScope Usage
+### ModelScope Usage
Users can use ModelScope for inference and fine-tuning based on a trained academic model. To achieve this, users need to run the stage 6 in the script. In this stage, relevant files required by ModelScope will be generated automatically. Users can then use the corresponding ModelScope interface by replacing the model name with the local trained model path. For the detailed usage of the ModelScope interface, please refer to [ModelScope Usage](https://alibaba-damo-academy.github.io/FunASR/en/modelscope_pipeline/quick_start.html).
-* Decoding by CPU or GPU
+### Decoding by CPU or GPU
We support CPU and GPU decoding. For CPU decoding, set `gpu_inference=false` and `njob` to specific the total number of CPU jobs. For GPU decoding, first set `gpu_inference=true`. Then set `gpuid_list` to specific which GPUs for decoding and `njob` to specific the number of decoding jobs on each GPU.
\ No newline at end of file
diff --git a/docs/model_zoo/modelscope_models.md b/docs/model_zoo/modelscope_models.md
index 8fe4e050b..cdbe3ffe6 100644
--- a/docs/model_zoo/modelscope_models.md
+++ b/docs/model_zoo/modelscope_models.md
@@ -1,7 +1,7 @@
# Pretrained Models on ModelScope
## Model License
-You are free to use, copy, modify, and share FunASR under the conditions of this agreement. You should indicate the code and model source and author information when using, copying, modifying and sharing FunASR. To upload the FunASR industrial model to any third-party platform for download and use, an additional license is required, which can be applied for free by sending an email to the official email address (funasr@list.alibaba-inc.com). Full license could see [license](https://github.com/alibaba-damo-academy/FunASR/blob/main/LICENSE)
+You are free to use, copy, modify, and share FunASR models under the conditions of this agreement. You should indicate the model source and author information when using, copying, modifying and sharing FunASR models. To upload the FunASR models to other third-party platforms for download, an additional license is required, which can be applied for free by sending an email to the official email address (funasr@list.alibaba-inc.com). Full model license could see [license](https://github.com/alibaba-damo-academy/FunASR/blob/main/MODEL_LICENSE)
## Model Zoo
diff --git a/egs_modelscope/asr/TEMPLATE/README.md b/egs_modelscope/asr/TEMPLATE/README.md
index 7ff04eb00..2c5433332 100644
--- a/egs_modelscope/asr/TEMPLATE/README.md
+++ b/egs_modelscope/asr/TEMPLATE/README.md
@@ -24,7 +24,8 @@ print(rec_result)
inference_pipeline = pipeline(
task=Tasks.auto_speech_recognition,
model='damo/speech_paraformer-large_asr_nat-zh-cn-16k-common-vocab8404-online',
- model_revision='v1.0.4'
+ model_revision='v1.0.6',
+ mode='paraformer_streaming'
)
import soundfile
speech, sample_rate = soundfile.read("example/asr_example.wav")
diff --git a/egs_modelscope/asr/paraformer/speech_paraformer-large_asr_nat-zh-cn-16k-common-vocab8404-online/README.md b/egs_modelscope/asr/paraformer/speech_paraformer-large_asr_nat-zh-cn-16k-common-vocab8404-online/README.md
new file mode 120000
index 000000000..bb55ab52e
--- /dev/null
+++ b/egs_modelscope/asr/paraformer/speech_paraformer-large_asr_nat-zh-cn-16k-common-vocab8404-online/README.md
@@ -0,0 +1 @@
+../../TEMPLATE/README.md
\ No newline at end of file
diff --git a/egs_modelscope/asr/paraformer/speech_paraformer-large_asr_nat-zh-cn-16k-common-vocab8404-online/demo.py b/egs_modelscope/asr/paraformer/speech_paraformer-large_asr_nat-zh-cn-16k-common-vocab8404-online/demo.py
index bc511bbd8..283f529a3 100644
--- a/egs_modelscope/asr/paraformer/speech_paraformer-large_asr_nat-zh-cn-16k-common-vocab8404-online/demo.py
+++ b/egs_modelscope/asr/paraformer/speech_paraformer-large_asr_nat-zh-cn-16k-common-vocab8404-online/demo.py
@@ -1,39 +1,12 @@
-import os
-import logging
-import torch
-import soundfile
-
from modelscope.pipelines import pipeline
from modelscope.utils.constant import Tasks
-from modelscope.utils.logger import get_logger
-logger = get_logger(log_level=logging.CRITICAL)
-logger.setLevel(logging.CRITICAL)
-
-os.environ["MODELSCOPE_CACHE"] = "./"
inference_pipeline = pipeline(
task=Tasks.auto_speech_recognition,
model='damo/speech_paraformer-large_asr_nat-zh-cn-16k-common-vocab8404-online',
- model_revision='v1.0.4'
+ model_revision='v1.0.6',
+ mode="paraformer_fake_streaming"
)
-
-model_dir = os.path.join(os.environ["MODELSCOPE_CACHE"], "damo/speech_paraformer-large_asr_nat-zh-cn-16k-common-vocab8404-online")
-speech, sample_rate = soundfile.read(os.path.join(model_dir, "example/asr_example.wav"))
-speech_length = speech.shape[0]
-
-sample_offset = 0
-chunk_size = [5, 10, 5] #[5, 10, 5] 600ms, [8, 8, 4] 480ms
-stride_size = chunk_size[1] * 960
-param_dict = {"cache": dict(), "is_final": False, "chunk_size": chunk_size}
-final_result = ""
-
-for sample_offset in range(0, speech_length, min(stride_size, speech_length - sample_offset)):
- if sample_offset + stride_size >= speech_length - 1:
- stride_size = speech_length - sample_offset
- param_dict["is_final"] = True
- rec_result = inference_pipeline(audio_in=speech[sample_offset: sample_offset + stride_size],
- param_dict=param_dict)
- if len(rec_result) != 0:
- final_result += rec_result['text']
- print(rec_result)
-print(final_result)
+audio_in='https://isv-data.oss-cn-hangzhou.aliyuncs.com/ics/MaaS/ASR/test_audio/asr_example_zh.wav'
+rec_result = inference_pipeline(audio_in=audio_in)
+print(rec_result)
diff --git a/egs_modelscope/asr/paraformer/speech_paraformer-large_asr_nat-zh-cn-16k-common-vocab8404-online/demo_online.py b/egs_modelscope/asr/paraformer/speech_paraformer-large_asr_nat-zh-cn-16k-common-vocab8404-online/demo_online.py
new file mode 100644
index 000000000..9d49d7dee
--- /dev/null
+++ b/egs_modelscope/asr/paraformer/speech_paraformer-large_asr_nat-zh-cn-16k-common-vocab8404-online/demo_online.py
@@ -0,0 +1,40 @@
+import os
+import logging
+import torch
+import soundfile
+
+from modelscope.pipelines import pipeline
+from modelscope.utils.constant import Tasks
+from modelscope.utils.logger import get_logger
+
+logger = get_logger(log_level=logging.CRITICAL)
+logger.setLevel(logging.CRITICAL)
+
+os.environ["MODELSCOPE_CACHE"] = "./"
+inference_pipeline = pipeline(
+ task=Tasks.auto_speech_recognition,
+ model='damo/speech_paraformer-large_asr_nat-zh-cn-16k-common-vocab8404-online',
+ model_revision='v1.0.6',
+ mode="paraformer_streaming"
+)
+
+model_dir = os.path.join(os.environ["MODELSCOPE_CACHE"], "damo/speech_paraformer-large_asr_nat-zh-cn-16k-common-vocab8404-online")
+speech, sample_rate = soundfile.read(os.path.join(model_dir, "example/asr_example.wav"))
+speech_length = speech.shape[0]
+
+sample_offset = 0
+chunk_size = [5, 10, 5] #[5, 10, 5] 600ms, [8, 8, 4] 480ms
+stride_size = chunk_size[1] * 960
+param_dict = {"cache": dict(), "is_final": False, "chunk_size": chunk_size}
+final_result = ""
+
+for sample_offset in range(0, speech_length, min(stride_size, speech_length - sample_offset)):
+ if sample_offset + stride_size >= speech_length - 1:
+ stride_size = speech_length - sample_offset
+ param_dict["is_final"] = True
+ rec_result = inference_pipeline(audio_in=speech[sample_offset: sample_offset + stride_size],
+ param_dict=param_dict)
+ if len(rec_result) != 0:
+ final_result += rec_result['text']
+ print(rec_result)
+print(final_result)
diff --git a/egs_modelscope/asr/paraformer/speech_paraformer-large_asr_nat-zh-cn-16k-common-vocab8404-online/finetune.py b/egs_modelscope/asr/paraformer/speech_paraformer-large_asr_nat-zh-cn-16k-common-vocab8404-online/finetune.py
new file mode 100644
index 000000000..a72321aa2
--- /dev/null
+++ b/egs_modelscope/asr/paraformer/speech_paraformer-large_asr_nat-zh-cn-16k-common-vocab8404-online/finetune.py
@@ -0,0 +1,37 @@
+import os
+
+from modelscope.metainfo import Trainers
+from modelscope.trainers import build_trainer
+
+from funasr.datasets.ms_dataset import MsDataset
+from funasr.utils.modelscope_param import modelscope_args
+
+
+def modelscope_finetune(params):
+ if not os.path.exists(params.output_dir):
+ os.makedirs(params.output_dir, exist_ok=True)
+ # dataset split ["train", "validation"]
+ ds_dict = MsDataset.load(params.data_path)
+ kwargs = dict(
+ model=params.model,
+ model_revision='v1.0.6',
+ data_dir=ds_dict,
+ dataset_type=params.dataset_type,
+ work_dir=params.output_dir,
+ batch_bins=params.batch_bins,
+ max_epoch=params.max_epoch,
+ lr=params.lr)
+ trainer = build_trainer(Trainers.speech_asr_trainer, default_args=kwargs)
+ trainer.train()
+
+
+if __name__ == '__main__':
+ params = modelscope_args(model="damo/speech_paraformer-large_asr_nat-zh-cn-16k-common-vocab8404-online", data_path="./data")
+ params.output_dir = "./checkpoint" # m模型保存路径
+ params.data_path = "./example_data/" # 数据路径
+ params.dataset_type = "small" # 小数据量设置small，若数据量大于1000小时，请使用large
+ params.batch_bins = 1000 # batch size，如果dataset_type="small"，batch_bins单位为fbank特征帧数，如果dataset_type="large"，batch_bins单位为毫秒，
+ params.max_epoch = 20 # 最大训练轮数
+ params.lr = 0.00005 # 设置学习率
+
+ modelscope_finetune(params)
diff --git a/egs_modelscope/asr/paraformer/speech_paraformer-large_asr_nat-zh-cn-16k-common-vocab8404-online/infer.py b/egs_modelscope/asr/paraformer/speech_paraformer-large_asr_nat-zh-cn-16k-common-vocab8404-online/infer.py
new file mode 100644
index 000000000..4a823aacd
--- /dev/null
+++ b/egs_modelscope/asr/paraformer/speech_paraformer-large_asr_nat-zh-cn-16k-common-vocab8404-online/infer.py
@@ -0,0 +1,32 @@
+import os
+import shutil
+import argparse
+from modelscope.pipelines import pipeline
+from modelscope.utils.constant import Tasks
+
+def modelscope_infer(args):
+ os.environ['CUDA_VISIBLE_DEVICES'] = str(args.gpuid)
+ inference_pipeline = pipeline(
+ task=Tasks.auto_speech_recognition,
+ model=args.model,
+ output_dir=args.output_dir,
+ batch_size=args.batch_size,
+ model_revision='v1.0.6',
+ mode="paraformer_fake_streaming",
+ param_dict={"decoding_model": args.decoding_mode, "hotword": args.hotword_txt}
+ )
+ inference_pipeline(audio_in=args.audio_in)
+
+if __name__ == "__main__":
+ parser = argparse.ArgumentParser()
+ parser.add_argument('--model', type=str, default="damo/speech_paraformer-large_asr_nat-zh-cn-16k-common-vocab8404-pytorch")
+ parser.add_argument('--audio_in', type=str, default="./data/test/wav.scp")
+ parser.add_argument('--output_dir', type=str, default="./results/")
+ parser.add_argument('--decoding_mode', type=str, default="normal")
+ parser.add_argument('--model_revision', type=str, default=None)
+ parser.add_argument('--mode', type=str, default=None)
+ parser.add_argument('--hotword_txt', type=str, default=None)
+ parser.add_argument('--batch_size', type=int, default=64)
+ parser.add_argument('--gpuid', type=str, default="0")
+ args = parser.parse_args()
+ modelscope_infer(args)
diff --git a/egs_modelscope/asr/paraformer/speech_paraformer-large_asr_nat-zh-cn-16k-common-vocab8404-online/infer.sh b/egs_modelscope/asr/paraformer/speech_paraformer-large_asr_nat-zh-cn-16k-common-vocab8404-online/infer.sh
new file mode 100644
index 000000000..6d4d625ef
--- /dev/null
+++ b/egs_modelscope/asr/paraformer/speech_paraformer-large_asr_nat-zh-cn-16k-common-vocab8404-online/infer.sh
@@ -0,0 +1,104 @@
+#!/usr/bin/env bash
+
+set -e
+set -u
+set -o pipefail
+
+stage=1
+stop_stage=2
+model="damo/speech_paraformer-large_asr_nat-zh-cn-16k-common-vocab8404-online"
+data_dir="./data/test"
+output_dir="./results"
+batch_size=32
+gpu_inference=true # whether to perform gpu decoding
+gpuid_list="0,1" # set gpus, e.g., gpuid_list="0,1"
+njob=32 # the number of jobs for CPU decoding, if gpu_inference=false, use CPU decoding, please set njob
+checkpoint_dir=
+checkpoint_name="valid.cer_ctc.ave.pb"
+
+. utils/parse_options.sh || exit 1;
+
+if ${gpu_inference} == "true"; then
+ nj=$(echo $gpuid_list | awk -F "," '{print NF}')
+else
+ nj=$njob
+ batch_size=1
+ gpuid_list=""
+ for JOB in $(seq ${nj}); do
+ gpuid_list=$gpuid_list"-1,"
+ done
+fi
+
+mkdir -p $output_dir/split
+split_scps=""
+for JOB in $(seq ${nj}); do
+ split_scps="$split_scps $output_dir/split/wav.$JOB.scp"
+done
+perl utils/split_scp.pl ${data_dir}/wav.scp ${split_scps}
+
+if [ -n "${checkpoint_dir}" ]; then
+ python utils/prepare_checkpoint.py ${model} ${checkpoint_dir} ${checkpoint_name}
+ model=${checkpoint_dir}/${model}
+fi
+
+if [ $stage -le 1 ] && [ $stop_stage -ge 1 ];then
+ echo "Decoding ..."
+ gpuid_list_array=(${gpuid_list//,/ })
+ for JOB in $(seq ${nj}); do
+ {
+ id=$((JOB-1))
+ gpuid=${gpuid_list_array[$id]}
+ mkdir -p ${output_dir}/output.$JOB
+ python infer.py \
+ --model ${model} \
+ --audio_in ${output_dir}/split/wav.$JOB.scp \
+ --output_dir ${output_dir}/output.$JOB \
+ --batch_size ${batch_size} \
+ --gpuid ${gpuid}
+ --mode "paraformer_fake_streaming"
+ }&
+ done
+ wait
+
+ mkdir -p ${output_dir}/1best_recog
+ for f in token score text; do
+ if [ -f "${output_dir}/output.1/1best_recog/${f}" ]; then
+ for i in $(seq "${nj}"); do
+ cat "${output_dir}/output.${i}/1best_recog/${f}"
+ done | sort -k1 >"${output_dir}/1best_recog/${f}"
+ fi
+ done
+fi
+
+if [ $stage -le 2 ] && [ $stop_stage -ge 2 ];then
+ echo "Computing WER ..."
+ cp ${output_dir}/1best_recog/text ${output_dir}/1best_recog/text.proc
+ cp ${data_dir}/text ${output_dir}/1best_recog/text.ref
+ python utils/compute_wer.py ${output_dir}/1best_recog/text.ref ${output_dir}/1best_recog/text.proc ${output_dir}/1best_recog/text.cer
+ tail -n 3 ${output_dir}/1best_recog/text.cer
+fi
+
+if [ $stage -le 3 ] && [ $stop_stage -ge 3 ];then
+ echo "SpeechIO TIOBE textnorm"
+ echo "$0 --> Normalizing REF text ..."
+ ./utils/textnorm_zh.py \
+ --has_key --to_upper \
+ ${data_dir}/text \
+ ${output_dir}/1best_recog/ref.txt
+
+ echo "$0 --> Normalizing HYP text ..."
+ ./utils/textnorm_zh.py \
+ --has_key --to_upper \
+ ${output_dir}/1best_recog/text.proc \
+ ${output_dir}/1best_recog/rec.txt
+ grep -v $'\t$' ${output_dir}/1best_recog/rec.txt > ${output_dir}/1best_recog/rec_non_empty.txt
+
+ echo "$0 --> computing WER/CER and alignment ..."
+ ./utils/error_rate_zh \
+ --tokenizer char \
+ --ref ${output_dir}/1best_recog/ref.txt \
+ --hyp ${output_dir}/1best_recog/rec_non_empty.txt \
+ ${output_dir}/1best_recog/DETAILS.txt | tee ${output_dir}/1best_recog/RESULTS.txt
+ rm -rf ${output_dir}/1best_recog/rec.txt ${output_dir}/1best_recog/rec_non_empty.txt
+fi
+
diff --git a/egs_modelscope/asr/paraformer/speech_paraformer-large_asr_nat-zh-cn-16k-common-vocab8404-online/utils b/egs_modelscope/asr/paraformer/speech_paraformer-large_asr_nat-zh-cn-16k-common-vocab8404-online/utils
new file mode 120000
index 000000000..a961ddc7a
--- /dev/null
+++ b/egs_modelscope/asr/paraformer/speech_paraformer-large_asr_nat-zh-cn-16k-common-vocab8404-online/utils
@@ -0,0 +1 @@
+../../TEMPLATE/utils/
\ No newline at end of file
diff --git a/egs_modelscope/asr/paraformer/speech_paraformer_asr_nat-zh-cn-16k-common-vocab8404-online/README.md b/egs_modelscope/asr/paraformer/speech_paraformer_asr_nat-zh-cn-16k-common-vocab8404-online/README.md
new file mode 120000
index 000000000..bb55ab52e
--- /dev/null
+++ b/egs_modelscope/asr/paraformer/speech_paraformer_asr_nat-zh-cn-16k-common-vocab8404-online/README.md
@@ -0,0 +1 @@
+../../TEMPLATE/README.md
\ No newline at end of file
diff --git a/egs_modelscope/asr/paraformer/speech_paraformer_asr_nat-zh-cn-16k-common-vocab8404-online/demo.py b/egs_modelscope/asr/paraformer/speech_paraformer_asr_nat-zh-cn-16k-common-vocab8404-online/demo.py
index abe66404d..5fa417b69 100644
--- a/egs_modelscope/asr/paraformer/speech_paraformer_asr_nat-zh-cn-16k-common-vocab8404-online/demo.py
+++ b/egs_modelscope/asr/paraformer/speech_paraformer_asr_nat-zh-cn-16k-common-vocab8404-online/demo.py
@@ -1,39 +1,12 @@
-import os
-import logging
-import torch
-import soundfile
-
from modelscope.pipelines import pipeline
from modelscope.utils.constant import Tasks
-from modelscope.utils.logger import get_logger
-logger = get_logger(log_level=logging.CRITICAL)
-logger.setLevel(logging.CRITICAL)
-
-os.environ["MODELSCOPE_CACHE"] = "./"
inference_pipeline = pipeline(
task=Tasks.auto_speech_recognition,
model='damo/speech_paraformer_asr_nat-zh-cn-16k-common-vocab8404-online',
- model_revision='v1.0.4'
+ model_revision='v1.0.6',
+ mode="paraformer_fake_streaming"
)
-
-model_dir = os.path.join(os.environ["MODELSCOPE_CACHE"], "damo/speech_paraformer_asr_nat-zh-cn-16k-common-vocab8404-online")
-speech, sample_rate = soundfile.read(os.path.join(model_dir, "example/asr_example.wav"))
-speech_length = speech.shape[0]
-
-sample_offset = 0
-chunk_size = [8, 8, 4] #[5, 10, 5] 600ms, [8, 8, 4] 480ms
-stride_size = chunk_size[1] * 960
-param_dict = {"cache": dict(), "is_final": False, "chunk_size": chunk_size}
-final_result = ""
-
-for sample_offset in range(0, speech_length, min(stride_size, speech_length - sample_offset)):
- if sample_offset + stride_size >= speech_length - 1:
- stride_size = speech_length - sample_offset
- param_dict["is_final"] = True
- rec_result = inference_pipeline(audio_in=speech[sample_offset: sample_offset + stride_size],
- param_dict=param_dict)
- if len(rec_result) != 0:
- final_result += rec_result['text']
- print(rec_result)
-print(final_result.strip())
+audio_in='https://isv-data.oss-cn-hangzhou.aliyuncs.com/ics/MaaS/ASR/test_audio/asr_example_zh.wav'
+rec_result = inference_pipeline(audio_in=audio_in)
+print(rec_result)
diff --git a/egs_modelscope/asr/paraformer/speech_paraformer_asr_nat-zh-cn-16k-common-vocab8404-online/demo_online.py b/egs_modelscope/asr/paraformer/speech_paraformer_asr_nat-zh-cn-16k-common-vocab8404-online/demo_online.py
new file mode 100644
index 000000000..d1dd441f0
--- /dev/null
+++ b/egs_modelscope/asr/paraformer/speech_paraformer_asr_nat-zh-cn-16k-common-vocab8404-online/demo_online.py
@@ -0,0 +1,40 @@
+import os
+import logging
+import torch
+import soundfile
+
+from modelscope.pipelines import pipeline
+from modelscope.utils.constant import Tasks
+from modelscope.utils.logger import get_logger
+
+logger = get_logger(log_level=logging.CRITICAL)
+logger.setLevel(logging.CRITICAL)
+
+os.environ["MODELSCOPE_CACHE"] = "./"
+inference_pipeline = pipeline(
+ task=Tasks.auto_speech_recognition,
+ model='damo/speech_paraformer_asr_nat-zh-cn-16k-common-vocab8404-online',
+ model_revision='v1.0.6',
+ mode="paraformer_streaming"
+)
+
+model_dir = os.path.join(os.environ["MODELSCOPE_CACHE"], "damo/speech_paraformer_asr_nat-zh-cn-16k-common-vocab8404-online")
+speech, sample_rate = soundfile.read(os.path.join(model_dir, "example/asr_example.wav"))
+speech_length = speech.shape[0]
+
+sample_offset = 0
+chunk_size = [8, 8, 4] #[5, 10, 5] 600ms, [8, 8, 4] 480ms
+stride_size = chunk_size[1] * 960
+param_dict = {"cache": dict(), "is_final": False, "chunk_size": chunk_size}
+final_result = ""
+
+for sample_offset in range(0, speech_length, min(stride_size, speech_length - sample_offset)):
+ if sample_offset + stride_size >= speech_length - 1:
+ stride_size = speech_length - sample_offset
+ param_dict["is_final"] = True
+ rec_result = inference_pipeline(audio_in=speech[sample_offset: sample_offset + stride_size],
+ param_dict=param_dict)
+ if len(rec_result) != 0:
+ final_result += rec_result['text']
+ print(rec_result)
+print(final_result)
diff --git a/egs_modelscope/asr/paraformer/speech_paraformer_asr_nat-zh-cn-16k-common-vocab8404-online/finetune.py b/egs_modelscope/asr/paraformer/speech_paraformer_asr_nat-zh-cn-16k-common-vocab8404-online/finetune.py
new file mode 100644
index 000000000..a9251eff9
--- /dev/null
+++ b/egs_modelscope/asr/paraformer/speech_paraformer_asr_nat-zh-cn-16k-common-vocab8404-online/finetune.py
@@ -0,0 +1,37 @@
+import os
+
+from modelscope.metainfo import Trainers
+from modelscope.trainers import build_trainer
+
+from funasr.datasets.ms_dataset import MsDataset
+from funasr.utils.modelscope_param import modelscope_args
+
+
+def modelscope_finetune(params):
+ if not os.path.exists(params.output_dir):
+ os.makedirs(params.output_dir, exist_ok=True)
+ # dataset split ["train", "validation"]
+ ds_dict = MsDataset.load(params.data_path)
+ kwargs = dict(
+ model=params.model,
+ model_revision='v1.0.6',
+ data_dir=ds_dict,
+ dataset_type=params.dataset_type,
+ work_dir=params.output_dir,
+ batch_bins=params.batch_bins,
+ max_epoch=params.max_epoch,
+ lr=params.lr)
+ trainer = build_trainer(Trainers.speech_asr_trainer, default_args=kwargs)
+ trainer.train()
+
+
+if __name__ == '__main__':
+ params = modelscope_args(model="damo/speech_paraformer_asr_nat-zh-cn-16k-common-vocab8404-online", data_path="./data")
+ params.output_dir = "./checkpoint" # m模型保存路径
+ params.data_path = "./example_data/" # 数据路径
+ params.dataset_type = "small" # 小数据量设置small，若数据量大于1000小时，请使用large
+ params.batch_bins = 1000 # batch size，如果dataset_type="small"，batch_bins单位为fbank特征帧数，如果dataset_type="large"，batch_bins单位为毫秒，
+ params.max_epoch = 20 # 最大训练轮数
+ params.lr = 0.00005 # 设置学习率
+
+ modelscope_finetune(params)
diff --git a/egs_modelscope/asr/paraformer/speech_paraformer_asr_nat-zh-cn-16k-common-vocab8404-online/infer.py b/egs_modelscope/asr/paraformer/speech_paraformer_asr_nat-zh-cn-16k-common-vocab8404-online/infer.py
new file mode 100644
index 000000000..4a823aacd
--- /dev/null
+++ b/egs_modelscope/asr/paraformer/speech_paraformer_asr_nat-zh-cn-16k-common-vocab8404-online/infer.py
@@ -0,0 +1,32 @@
+import os
+import shutil
+import argparse
+from modelscope.pipelines import pipeline
+from modelscope.utils.constant import Tasks
+
+def modelscope_infer(args):
+ os.environ['CUDA_VISIBLE_DEVICES'] = str(args.gpuid)
+ inference_pipeline = pipeline(
+ task=Tasks.auto_speech_recognition,
+ model=args.model,
+ output_dir=args.output_dir,
+ batch_size=args.batch_size,
+ model_revision='v1.0.6',
+ mode="paraformer_fake_streaming",
+ param_dict={"decoding_model": args.decoding_mode, "hotword": args.hotword_txt}
+ )
+ inference_pipeline(audio_in=args.audio_in)
+
+if __name__ == "__main__":
+ parser = argparse.ArgumentParser()
+ parser.add_argument('--model', type=str, default="damo/speech_paraformer-large_asr_nat-zh-cn-16k-common-vocab8404-pytorch")
+ parser.add_argument('--audio_in', type=str, default="./data/test/wav.scp")
+ parser.add_argument('--output_dir', type=str, default="./results/")
+ parser.add_argument('--decoding_mode', type=str, default="normal")
+ parser.add_argument('--model_revision', type=str, default=None)
+ parser.add_argument('--mode', type=str, default=None)
+ parser.add_argument('--hotword_txt', type=str, default=None)
+ parser.add_argument('--batch_size', type=int, default=64)
+ parser.add_argument('--gpuid', type=str, default="0")
+ args = parser.parse_args()
+ modelscope_infer(args)
diff --git a/egs_modelscope/asr/paraformer/speech_paraformer_asr_nat-zh-cn-16k-common-vocab8404-online/infer.sh b/egs_modelscope/asr/paraformer/speech_paraformer_asr_nat-zh-cn-16k-common-vocab8404-online/infer.sh
new file mode 100644
index 000000000..46c2bb3dc
--- /dev/null
+++ b/egs_modelscope/asr/paraformer/speech_paraformer_asr_nat-zh-cn-16k-common-vocab8404-online/infer.sh
@@ -0,0 +1,104 @@
+#!/usr/bin/env bash
+
+set -e
+set -u
+set -o pipefail
+
+stage=1
+stop_stage=2
+model="damo/speech_paraformer_asr_nat-zh-cn-16k-common-vocab8404-online"
+data_dir="./data/test"
+output_dir="./results"
+batch_size=32
+gpu_inference=true # whether to perform gpu decoding
+gpuid_list="0,1" # set gpus, e.g., gpuid_list="0,1"
+njob=32 # the number of jobs for CPU decoding, if gpu_inference=false, use CPU decoding, please set njob
+checkpoint_dir=
+checkpoint_name="valid.cer_ctc.ave.pb"
+
+. utils/parse_options.sh || exit 1;
+
+if ${gpu_inference} == "true"; then
+ nj=$(echo $gpuid_list | awk -F "," '{print NF}')
+else
+ nj=$njob
+ batch_size=1
+ gpuid_list=""
+ for JOB in $(seq ${nj}); do
+ gpuid_list=$gpuid_list"-1,"
+ done
+fi
+
+mkdir -p $output_dir/split
+split_scps=""
+for JOB in $(seq ${nj}); do
+ split_scps="$split_scps $output_dir/split/wav.$JOB.scp"
+done
+perl utils/split_scp.pl ${data_dir}/wav.scp ${split_scps}
+
+if [ -n "${checkpoint_dir}" ]; then
+ python utils/prepare_checkpoint.py ${model} ${checkpoint_dir} ${checkpoint_name}
+ model=${checkpoint_dir}/${model}
+fi
+
+if [ $stage -le 1 ] && [ $stop_stage -ge 1 ];then
+ echo "Decoding ..."
+ gpuid_list_array=(${gpuid_list//,/ })
+ for JOB in $(seq ${nj}); do
+ {
+ id=$((JOB-1))
+ gpuid=${gpuid_list_array[$id]}
+ mkdir -p ${output_dir}/output.$JOB
+ python infer.py \
+ --model ${model} \
+ --audio_in ${output_dir}/split/wav.$JOB.scp \
+ --output_dir ${output_dir}/output.$JOB \
+ --batch_size ${batch_size} \
+ --gpuid ${gpuid}
+ --mode "paraformer_fake_streaming"
+ }&
+ done
+ wait
+
+ mkdir -p ${output_dir}/1best_recog
+ for f in token score text; do
+ if [ -f "${output_dir}/output.1/1best_recog/${f}" ]; then
+ for i in $(seq "${nj}"); do
+ cat "${output_dir}/output.${i}/1best_recog/${f}"
+ done | sort -k1 >"${output_dir}/1best_recog/${f}"
+ fi
+ done
+fi
+
+if [ $stage -le 2 ] && [ $stop_stage -ge 2 ];then
+ echo "Computing WER ..."
+ cp ${output_dir}/1best_recog/text ${output_dir}/1best_recog/text.proc
+ cp ${data_dir}/text ${output_dir}/1best_recog/text.ref
+ python utils/compute_wer.py ${output_dir}/1best_recog/text.ref ${output_dir}/1best_recog/text.proc ${output_dir}/1best_recog/text.cer
+ tail -n 3 ${output_dir}/1best_recog/text.cer
+fi
+
+if [ $stage -le 3 ] && [ $stop_stage -ge 3 ];then
+ echo "SpeechIO TIOBE textnorm"
+ echo "$0 --> Normalizing REF text ..."
+ ./utils/textnorm_zh.py \
+ --has_key --to_upper \
+ ${data_dir}/text \
+ ${output_dir}/1best_recog/ref.txt
+
+ echo "$0 --> Normalizing HYP text ..."
+ ./utils/textnorm_zh.py \
+ --has_key --to_upper \
+ ${output_dir}/1best_recog/text.proc \
+ ${output_dir}/1best_recog/rec.txt
+ grep -v $'\t$' ${output_dir}/1best_recog/rec.txt > ${output_dir}/1best_recog/rec_non_empty.txt
+
+ echo "$0 --> computing WER/CER and alignment ..."
+ ./utils/error_rate_zh \
+ --tokenizer char \
+ --ref ${output_dir}/1best_recog/ref.txt \
+ --hyp ${output_dir}/1best_recog/rec_non_empty.txt \
+ ${output_dir}/1best_recog/DETAILS.txt | tee ${output_dir}/1best_recog/RESULTS.txt
+ rm -rf ${output_dir}/1best_recog/rec.txt ${output_dir}/1best_recog/rec_non_empty.txt
+fi
+
diff --git a/egs_modelscope/asr/paraformer/speech_paraformer_asr_nat-zh-cn-16k-common-vocab8404-online/utils b/egs_modelscope/asr/paraformer/speech_paraformer_asr_nat-zh-cn-16k-common-vocab8404-online/utils
new file mode 120000
index 000000000..a961ddc7a
--- /dev/null
+++ b/egs_modelscope/asr/paraformer/speech_paraformer_asr_nat-zh-cn-16k-common-vocab8404-online/utils
@@ -0,0 +1 @@
+../../TEMPLATE/utils/
\ No newline at end of file
diff --git a/funasr/bin/asr_infer.py b/funasr/bin/asr_infer.py
index 760fd0742..0e203c4db 100644
--- a/funasr/bin/asr_infer.py
+++ b/funasr/bin/asr_infer.py
@@ -305,6 +305,7 @@ class Speech2TextParaformer:
nbest: int = 1,
frontend_conf: dict = None,
hotword_list_or_file: str = None,
+ decoding_ind: int = 0,
**kwargs,
):
assert check_argument_types()
@@ -415,6 +416,7 @@ class Speech2TextParaformer:
self.nbest = nbest
self.frontend = frontend
self.encoder_downsampling_factor = 1
+ self.decoding_ind = decoding_ind
if asr_train_args.encoder == "data2vec_encoder" or asr_train_args.encoder_conf["input_layer"] == "conv2d":
self.encoder_downsampling_factor = 4
@@ -452,7 +454,7 @@ class Speech2TextParaformer:
batch = to_device(batch, device=self.device)
# b. Forward Encoder
- enc, enc_len = self.asr_model.encode(**batch)
+ enc, enc_len = self.asr_model.encode(**batch, ind=self.decoding_ind)
if isinstance(enc, tuple):
enc = enc[0]
# assert len(enc) == 1, len(enc)
diff --git a/funasr/bin/asr_inference_launch.py b/funasr/bin/asr_inference_launch.py
index ec5e17535..f84212d92 100644
--- a/funasr/bin/asr_inference_launch.py
+++ b/funasr/bin/asr_inference_launch.py
@@ -1638,6 +1638,8 @@ def inference_launch(**kwargs):
return inference_uniasr(**kwargs)
elif mode == "paraformer":
return inference_paraformer(**kwargs)
+ elif mode == "paraformer_fake_streaming":
+ return inference_paraformer(**kwargs)
elif mode == "paraformer_streaming":
return inference_paraformer_online(**kwargs)
elif mode.startswith("paraformer_vad"):
@@ -1920,4 +1922,4 @@ def main(cmd=None):
if __name__ == "__main__":
- main()
\ No newline at end of file
+ main()
diff --git a/funasr/bin/build_trainer.py b/funasr/bin/build_trainer.py
index 94f72627e..267e4054d 100644
--- a/funasr/bin/build_trainer.py
+++ b/funasr/bin/build_trainer.py
@@ -23,6 +23,8 @@ def parse_args(mode):
from funasr.tasks.asr import ASRTask as ASRTask
elif mode == "paraformer":
from funasr.tasks.asr import ASRTaskParaformer as ASRTask
+ elif mode == "paraformer_streaming":
+ from funasr.tasks.asr import ASRTaskParaformer as ASRTask
elif mode == "paraformer_vad_punc":
from funasr.tasks.asr import ASRTaskParaformer as ASRTask
elif mode == "uniasr":
diff --git a/funasr/build_utils/build_asr_model.py b/funasr/build_utils/build_asr_model.py
index ddc827fb3..46c11b0e6 100644
--- a/funasr/build_utils/build_asr_model.py
+++ b/funasr/build_utils/build_asr_model.py
@@ -23,7 +23,7 @@ from funasr.models.decoder.rnnt_decoder import RNNTDecoder
from funasr.models.joint_net.joint_network import JointNetwork
from funasr.models.e2e_asr import ASRModel
from funasr.models.e2e_asr_mfcca import MFCCA
-from funasr.models.e2e_asr_paraformer import Paraformer, ParaformerBert, BiCifParaformer, ContextualParaformer
+from funasr.models.e2e_asr_paraformer import Paraformer, ParaformerOnline, ParaformerBert, BiCifParaformer, ContextualParaformer
from funasr.models.e2e_tp import TimestampPredictor
from funasr.models.e2e_uni_asr import UniASR
from funasr.models.e2e_asr_transducer import TransducerModel, UnifiedTransducerModel
@@ -82,6 +82,7 @@ model_choices = ClassChoices(
asr=ASRModel,
uniasr=UniASR,
paraformer=Paraformer,
+ paraformer_online=ParaformerOnline,
paraformer_bert=ParaformerBert,
bicif_paraformer=BiCifParaformer,
contextual_paraformer=ContextualParaformer,
@@ -293,7 +294,7 @@ def build_asr_model(args):
token_list=token_list,
**args.model_conf,
)
- elif args.model in ["paraformer", "paraformer_bert", "bicif_paraformer", "contextual_paraformer"]:
+ elif args.model in ["paraformer", "paraformer_online", "paraformer_bert", "bicif_paraformer", "contextual_paraformer"]:
# predictor
predictor_class = predictor_choices.get_class(args.predictor)
predictor = predictor_class(**args.predictor_conf)
diff --git a/funasr/models/decoder/sanm_decoder.py b/funasr/models/decoder/sanm_decoder.py
index 18cd343e7..ed920bfbb 100644
--- a/funasr/models/decoder/sanm_decoder.py
+++ b/funasr/models/decoder/sanm_decoder.py
@@ -935,6 +935,7 @@ class ParaformerSANMDecoder(BaseTransformerDecoder):
hlens: torch.Tensor,
ys_in_pad: torch.Tensor,
ys_in_lens: torch.Tensor,
+ chunk_mask: torch.Tensor = None,
) -> Tuple[torch.Tensor, torch.Tensor]:
"""Forward decoder.
@@ -955,9 +956,13 @@ class ParaformerSANMDecoder(BaseTransformerDecoder):
"""
tgt = ys_in_pad
tgt_mask = myutils.sequence_mask(ys_in_lens, device=tgt.device)[:, :, None]
-
+
memory = hs_pad
memory_mask = myutils.sequence_mask(hlens, device=memory.device)[:, None, :]
+ if chunk_mask is not None:
+ memory_mask = memory_mask * chunk_mask
+ if tgt_mask.size(1) != memory_mask.size(1):
+ memory_mask = torch.cat((memory_mask, memory_mask[:, -2:-1, :]), dim=1)
x = tgt
x, tgt_mask, memory, memory_mask, _ = self.decoders(
diff --git a/funasr/models/e2e_asr_paraformer.py b/funasr/models/e2e_asr_paraformer.py
index 82acef2ae..686038e11 100644
--- a/funasr/models/e2e_asr_paraformer.py
+++ b/funasr/models/e2e_asr_paraformer.py
@@ -153,6 +153,7 @@ class Paraformer(FunASRModel):
speech_lengths: torch.Tensor,
text: torch.Tensor,
text_lengths: torch.Tensor,
+ decoding_ind: int = None,
) -> Tuple[torch.Tensor, Dict[str, torch.Tensor], torch.Tensor]:
"""Frontend + Encoder + Decoder + Calc loss
Args:
@@ -160,6 +161,7 @@ class Paraformer(FunASRModel):
speech_lengths: (Batch, )
text: (Batch, Length)
text_lengths: (Batch,)
+ decoding_ind: int
"""
assert text_lengths.dim() == 1, text_lengths.shape
# Check that batch_size is unified
@@ -176,7 +178,11 @@ class Paraformer(FunASRModel):
speech = speech[:, :speech_lengths.max()]
# 1. Encoder
- encoder_out, encoder_out_lens = self.encode(speech, speech_lengths)
+ if hasattr(self.encoder, "overlap_chunk_cls"):
+ ind = self.encoder.overlap_chunk_cls.random_choice(self.training, decoding_ind)
+ encoder_out, encoder_out_lens = self.encode(speech, speech_lengths, ind=ind)
+ else:
+ encoder_out, encoder_out_lens = self.encode(speech, speech_lengths)
intermediate_outs = None
if isinstance(encoder_out, tuple):
intermediate_outs = encoder_out[1]
@@ -272,12 +278,13 @@ class Paraformer(FunASRModel):
return {"feats": feats, "feats_lengths": feats_lengths}
def encode(
- self, speech: torch.Tensor, speech_lengths: torch.Tensor
+ self, speech: torch.Tensor, speech_lengths: torch.Tensor, ind: int = 0,
) -> Tuple[torch.Tensor, torch.Tensor]:
"""Frontend + Encoder. Note that this method is used by asr_inference.py
Args:
speech: (Batch, Length, ...)
speech_lengths: (Batch, )
+ ind: int
"""
with autocast(False):
# 1. Extract feats
@@ -299,11 +306,25 @@ class Paraformer(FunASRModel):
# feats: (Batch, Length, Dim)
# -> encoder_out: (Batch, Length2, Dim2)
if self.encoder.interctc_use_conditioning:
- encoder_out, encoder_out_lens, _ = self.encoder(
- feats, feats_lengths, ctc=self.ctc
- )
+ if hasattr(self.encoder, "overlap_chunk_cls"):
+ encoder_out, encoder_out_lens, _ = self.encoder(
+ feats, feats_lengths, ctc=self.ctc, ind=ind
+ )
+ encoder_out, encoder_out_lens = self.encoder.overlap_chunk_cls.remove_chunk(encoder_out,
+ encoder_out_lens,
+ chunk_outs=None)
+ else:
+ encoder_out, encoder_out_lens, _ = self.encoder(
+ feats, feats_lengths, ctc=self.ctc
+ )
else:
- encoder_out, encoder_out_lens, _ = self.encoder(feats, feats_lengths)
+ if hasattr(self.encoder, "overlap_chunk_cls"):
+ encoder_out, encoder_out_lens, _ = self.encoder(feats, feats_lengths, ind=ind)
+ encoder_out, encoder_out_lens = self.encoder.overlap_chunk_cls.remove_chunk(encoder_out,
+ encoder_out_lens,
+ chunk_outs=None)
+ else:
+ encoder_out, encoder_out_lens, _ = self.encoder(feats, feats_lengths)
intermediate_outs = None
if isinstance(encoder_out, tuple):
intermediate_outs = encoder_out[1]
@@ -592,9 +613,137 @@ class ParaformerOnline(Paraformer):
"""
def __init__(
- self, *args, **kwargs,
+ self,
+ vocab_size: int,
+ token_list: Union[Tuple[str, ...], List[str]],
+ frontend: Optional[AbsFrontend],
+ specaug: Optional[AbsSpecAug],
+ normalize: Optional[AbsNormalize],
+ encoder: AbsEncoder,
+ decoder: AbsDecoder,
+ ctc: CTC,
+ ctc_weight: float = 0.5,
+ interctc_weight: float = 0.0,
+ ignore_id: int = -1,
+ blank_id: int = 0,
+ sos: int = 1,
+ eos: int = 2,
+ lsm_weight: float = 0.0,
+ length_normalized_loss: bool = False,
+ report_cer: bool = True,
+ report_wer: bool = True,
+ sym_space: str = "",
+ sym_blank: str = "",
+ extract_feats_in_collect_stats: bool = True,
+ predictor=None,
+ predictor_weight: float = 0.0,
+ predictor_bias: int = 0,
+ sampling_ratio: float = 0.2,
+ decoder_attention_chunk_type: str = 'chunk',
+ share_embedding: bool = False,
+ preencoder: Optional[AbsPreEncoder] = None,
+ postencoder: Optional[AbsPostEncoder] = None,
+ use_1st_decoder_loss: bool = False,
):
- super().__init__(*args, **kwargs)
+ assert check_argument_types()
+ assert 0.0 <= ctc_weight <= 1.0, ctc_weight
+ assert 0.0 <= interctc_weight < 1.0, interctc_weight
+
+ super().__init__(
+ vocab_size=vocab_size,
+ token_list=token_list,
+ frontend=frontend,
+ specaug=specaug,
+ normalize=normalize,
+ preencoder=preencoder,
+ encoder=encoder,
+ postencoder=postencoder,
+ decoder=decoder,
+ ctc=ctc,
+ ctc_weight=ctc_weight,
+ interctc_weight=interctc_weight,
+ ignore_id=ignore_id,
+ blank_id=blank_id,
+ sos=sos,
+ eos=eos,
+ lsm_weight=lsm_weight,
+ length_normalized_loss=length_normalized_loss,
+ report_cer=report_cer,
+ report_wer=report_wer,
+ sym_space=sym_space,
+ sym_blank=sym_blank,
+ extract_feats_in_collect_stats=extract_feats_in_collect_stats,
+ predictor=predictor,
+ predictor_weight=predictor_weight,
+ predictor_bias=predictor_bias,
+ sampling_ratio=sampling_ratio,
+ )
+ # note that eos is the same as sos (equivalent ID)
+ self.blank_id = blank_id
+ self.sos = vocab_size - 1 if sos is None else sos
+ self.eos = vocab_size - 1 if eos is None else eos
+ self.vocab_size = vocab_size
+ self.ignore_id = ignore_id
+ self.ctc_weight = ctc_weight
+ self.interctc_weight = interctc_weight
+ self.token_list = token_list.copy()
+
+ self.frontend = frontend
+ self.specaug = specaug
+ self.normalize = normalize
+ self.preencoder = preencoder
+ self.postencoder = postencoder
+ self.encoder = encoder
+
+ if not hasattr(self.encoder, "interctc_use_conditioning"):
+ self.encoder.interctc_use_conditioning = False
+ if self.encoder.interctc_use_conditioning:
+ self.encoder.conditioning_layer = torch.nn.Linear(
+ vocab_size, self.encoder.output_size()
+ )
+
+ self.error_calculator = None
+
+ if ctc_weight == 1.0:
+ self.decoder = None
+ else:
+ self.decoder = decoder
+
+ self.criterion_att = LabelSmoothingLoss(
+ size=vocab_size,
+ padding_idx=ignore_id,
+ smoothing=lsm_weight,
+ normalize_length=length_normalized_loss,
+ )
+
+ if report_cer or report_wer:
+ self.error_calculator = ErrorCalculator(
+ token_list, sym_space, sym_blank, report_cer, report_wer
+ )
+
+ if ctc_weight == 0.0:
+ self.ctc = None
+ else:
+ self.ctc = ctc
+
+ self.extract_feats_in_collect_stats = extract_feats_in_collect_stats
+ self.predictor = predictor
+ self.predictor_weight = predictor_weight
+ self.predictor_bias = predictor_bias
+ self.sampling_ratio = sampling_ratio
+ self.criterion_pre = mae_loss(normalize_length=length_normalized_loss)
+ self.step_cur = 0
+ self.scama_mask = None
+ if hasattr(self.encoder, "overlap_chunk_cls") and self.encoder.overlap_chunk_cls is not None:
+ from funasr.modules.streaming_utils.chunk_utilis import build_scama_mask_for_cross_attention_decoder
+ self.build_scama_mask_for_cross_attention_decoder_fn = build_scama_mask_for_cross_attention_decoder
+ self.decoder_attention_chunk_type = decoder_attention_chunk_type
+
+ self.share_embedding = share_embedding
+ if self.share_embedding:
+ self.decoder.embed = None
+
+ self.use_1st_decoder_loss = use_1st_decoder_loss
def forward(
self,
@@ -602,6 +751,7 @@ class ParaformerOnline(Paraformer):
speech_lengths: torch.Tensor,
text: torch.Tensor,
text_lengths: torch.Tensor,
+ decoding_ind: int = None,
) -> Tuple[torch.Tensor, Dict[str, torch.Tensor], torch.Tensor]:
"""Frontend + Encoder + Decoder + Calc loss
Args:
@@ -609,6 +759,7 @@ class ParaformerOnline(Paraformer):
speech_lengths: (Batch, )
text: (Batch, Length)
text_lengths: (Batch,)
+ decoding_ind: int
"""
assert text_lengths.dim() == 1, text_lengths.shape
# Check that batch_size is unified
@@ -625,7 +776,11 @@ class ParaformerOnline(Paraformer):
speech = speech[:, :speech_lengths.max()]
# 1. Encoder
- encoder_out, encoder_out_lens = self.encode(speech, speech_lengths)
+ if hasattr(self.encoder, "overlap_chunk_cls"):
+ ind = self.encoder.overlap_chunk_cls.random_choice(self.training, decoding_ind)
+ encoder_out, encoder_out_lens = self.encode(speech, speech_lengths, ind=ind)
+ else:
+ encoder_out, encoder_out_lens = self.encode(speech, speech_lengths)
intermediate_outs = None
if isinstance(encoder_out, tuple):
intermediate_outs = encoder_out[1]
@@ -638,8 +793,12 @@ class ParaformerOnline(Paraformer):
# 1. CTC branch
if self.ctc_weight != 0.0:
+ if hasattr(self.encoder, "overlap_chunk_cls"):
+ encoder_out_ctc, encoder_out_lens_ctc = self.encoder.overlap_chunk_cls.remove_chunk(encoder_out,
+ encoder_out_lens,
+ chunk_outs=None)
loss_ctc, cer_ctc = self._calc_ctc_loss(
- encoder_out, encoder_out_lens, text, text_lengths
+ encoder_out_ctc, encoder_out_lens_ctc, text, text_lengths
)
# Collect CTC branch stats
@@ -652,8 +811,14 @@ class ParaformerOnline(Paraformer):
for layer_idx, intermediate_out in intermediate_outs:
# we assume intermediate_out has the same length & padding
# as those of encoder_out
+ if hasattr(self.encoder, "overlap_chunk_cls"):
+ encoder_out_ctc, encoder_out_lens_ctc = \
+ self.encoder.overlap_chunk_cls.remove_chunk(
+ intermediate_out,
+ encoder_out_lens,
+ chunk_outs=None)
loss_ic, cer_ic = self._calc_ctc_loss(
- intermediate_out, encoder_out_lens, text, text_lengths
+ encoder_out_ctc, encoder_out_lens_ctc, text, text_lengths
)
loss_interctc = loss_interctc + loss_ic
@@ -672,7 +837,7 @@ class ParaformerOnline(Paraformer):
# 2b. Attention decoder branch
if self.ctc_weight != 1.0:
- loss_att, acc_att, cer_att, wer_att, loss_pre = self._calc_att_loss(
+ loss_att, acc_att, cer_att, wer_att, loss_pre, pre_loss_att = self._calc_att_predictor_loss(
encoder_out, encoder_out_lens, text, text_lengths
)
@@ -684,8 +849,12 @@ class ParaformerOnline(Paraformer):
else:
loss = self.ctc_weight * loss_ctc + (1 - self.ctc_weight) * loss_att + loss_pre * self.predictor_weight
+ if self.use_1st_decoder_loss and pre_loss_att is not None:
+ loss = loss + pre_loss_att
+
# Collect Attn branch stats
stats["loss_att"] = loss_att.detach() if loss_att is not None else None
+ stats["pre_loss_att"] = pre_loss_att.detach() if pre_loss_att is not None else None
stats["acc"] = acc_att
stats["cer"] = cer_att
stats["wer"] = wer_att
@@ -697,14 +866,67 @@ class ParaformerOnline(Paraformer):
loss, stats, weight = force_gatherable((loss, stats, batch_size), loss.device)
return loss, stats, weight
+ def encode(
+ self, speech: torch.Tensor, speech_lengths: torch.Tensor, ind: int = 0,
+ ) -> Tuple[torch.Tensor, torch.Tensor]:
+ """Frontend + Encoder. Note that this method is used by asr_inference.py
+ Args:
+ speech: (Batch, Length, ...)
+ speech_lengths: (Batch, )
+ """
+ with autocast(False):
+ # 1. Extract feats
+ feats, feats_lengths = self._extract_feats(speech, speech_lengths)
+
+ # 2. Data augmentation
+ if self.specaug is not None and self.training:
+ feats, feats_lengths = self.specaug(feats, feats_lengths)
+
+ # 3. Normalization for feature: e.g. Global-CMVN, Utterance-CMVN
+ if self.normalize is not None:
+ feats, feats_lengths = self.normalize(feats, feats_lengths)
+ # Pre-encoder, e.g. used for raw input data
+ if self.preencoder is not None:
+ feats, feats_lengths = self.preencoder(feats, feats_lengths)
+
+ # 4. Forward encoder
+ # feats: (Batch, Length, Dim)
+ # -> encoder_out: (Batch, Length2, Dim2)
+ if self.encoder.interctc_use_conditioning:
+ encoder_out, encoder_out_lens, _ = self.encoder(
+ feats, feats_lengths, ctc=self.ctc, ind=ind
+ )
+ else:
+ encoder_out, encoder_out_lens, _ = self.encoder(feats, feats_lengths, ind=ind)
+ intermediate_outs = None
+ if isinstance(encoder_out, tuple):
+ intermediate_outs = encoder_out[1]
+ encoder_out = encoder_out[0]
+
+ # Post-encoder, e.g. NLU
+ if self.postencoder is not None:
+ encoder_out, encoder_out_lens = self.postencoder(
+ encoder_out, encoder_out_lens
+ )
+
+ assert encoder_out.size(0) == speech.size(0), (
+ encoder_out.size(),
+ speech.size(0),
+ )
+ assert encoder_out.size(1) <= encoder_out_lens.max(), (
+ encoder_out.size(),
+ encoder_out_lens.max(),
+ )
+
+ if intermediate_outs is not None:
+ return (encoder_out, intermediate_outs), encoder_out_lens
+
+ return encoder_out, encoder_out_lens
+
def encode_chunk(
self, speech: torch.Tensor, speech_lengths: torch.Tensor, cache: dict = None
) -> Tuple[torch.Tensor, torch.Tensor]:
"""Frontend + Encoder. Note that this method is used by asr_inference.py
-<<<<<<< HEAD
-=======
-
->>>>>>> 4cd79db451786548d8a100f25c3b03da0eb30f4b
Args:
speech: (Batch, Length, ...)
speech_lengths: (Batch, )
@@ -750,12 +972,241 @@ class ParaformerOnline(Paraformer):
return encoder_out, torch.tensor([encoder_out.size(1)])
+ def _calc_att_predictor_loss(
+ self,
+ encoder_out: torch.Tensor,
+ encoder_out_lens: torch.Tensor,
+ ys_pad: torch.Tensor,
+ ys_pad_lens: torch.Tensor,
+ ):
+ encoder_out_mask = (~make_pad_mask(encoder_out_lens, maxlen=encoder_out.size(1))[:, None, :]).to(
+ encoder_out.device)
+ if self.predictor_bias == 1:
+ _, ys_pad = add_sos_eos(ys_pad, self.sos, self.eos, self.ignore_id)
+ ys_pad_lens = ys_pad_lens + self.predictor_bias
+ mask_chunk_predictor = None
+ if self.encoder.overlap_chunk_cls is not None:
+ mask_chunk_predictor = self.encoder.overlap_chunk_cls.get_mask_chunk_predictor(None,
+ device=encoder_out.device,
+ batch_size=encoder_out.size(
+ 0))
+ mask_shfit_chunk = self.encoder.overlap_chunk_cls.get_mask_shfit_chunk(None, device=encoder_out.device,
+ batch_size=encoder_out.size(0))
+ encoder_out = encoder_out * mask_shfit_chunk
+ pre_acoustic_embeds, pre_token_length, pre_alphas, _ = self.predictor(encoder_out,
+ ys_pad,
+ encoder_out_mask,
+ ignore_id=self.ignore_id,
+ mask_chunk_predictor=mask_chunk_predictor,
+ target_label_length=ys_pad_lens,
+ )
+ predictor_alignments, predictor_alignments_len = self.predictor.gen_frame_alignments(pre_alphas,
+ encoder_out_lens)
+
+ scama_mask = None
+ if self.encoder.overlap_chunk_cls is not None and self.decoder_attention_chunk_type == 'chunk':
+ encoder_chunk_size = self.encoder.overlap_chunk_cls.chunk_size_pad_shift_cur
+ attention_chunk_center_bias = 0
+ attention_chunk_size = encoder_chunk_size
+ decoder_att_look_back_factor = self.encoder.overlap_chunk_cls.decoder_att_look_back_factor_cur
+ mask_shift_att_chunk_decoder = self.encoder.overlap_chunk_cls.\
+ get_mask_shift_att_chunk_decoder(None,
+ device=encoder_out.device,
+ batch_size=encoder_out.size(0)
+ )
+ scama_mask = self.build_scama_mask_for_cross_attention_decoder_fn(
+ predictor_alignments=predictor_alignments,
+ encoder_sequence_length=encoder_out_lens,
+ chunk_size=1,
+ encoder_chunk_size=encoder_chunk_size,
+ attention_chunk_center_bias=attention_chunk_center_bias,
+ attention_chunk_size=attention_chunk_size,
+ attention_chunk_type=self.decoder_attention_chunk_type,
+ step=None,
+ predictor_mask_chunk_hopping=mask_chunk_predictor,
+ decoder_att_look_back_factor=decoder_att_look_back_factor,
+ mask_shift_att_chunk_decoder=mask_shift_att_chunk_decoder,
+ target_length=ys_pad_lens,
+ is_training=self.training,
+ )
+ elif self.encoder.overlap_chunk_cls is not None:
+ encoder_out, encoder_out_lens = self.encoder.overlap_chunk_cls.remove_chunk(encoder_out,
+ encoder_out_lens,
+ chunk_outs=None)
+ # 0. sampler
+ decoder_out_1st = None
+ pre_loss_att = None
+ if self.sampling_ratio > 0.0:
+ if self.step_cur < 2:
+ logging.info("enable sampler in paraformer, sampling_ratio: {}".format(self.sampling_ratio))
+ if self.use_1st_decoder_loss:
+ sematic_embeds, decoder_out_1st, pre_loss_att = \
+ self.sampler_with_grad(encoder_out, encoder_out_lens, ys_pad,
+ ys_pad_lens, pre_acoustic_embeds, scama_mask)
+ else:
+ sematic_embeds, decoder_out_1st = \
+ self.sampler(encoder_out, encoder_out_lens, ys_pad,
+ ys_pad_lens, pre_acoustic_embeds, scama_mask)
+ else:
+ if self.step_cur < 2:
+ logging.info("disable sampler in paraformer, sampling_ratio: {}".format(self.sampling_ratio))
+ sematic_embeds = pre_acoustic_embeds
+
+ # 1. Forward decoder
+ decoder_outs = self.decoder(
+ encoder_out, encoder_out_lens, sematic_embeds, ys_pad_lens, scama_mask
+ )
+ decoder_out, _ = decoder_outs[0], decoder_outs[1]
+
+ if decoder_out_1st is None:
+ decoder_out_1st = decoder_out
+ # 2. Compute attention loss
+ loss_att = self.criterion_att(decoder_out, ys_pad)
+ acc_att = th_accuracy(
+ decoder_out_1st.view(-1, self.vocab_size),
+ ys_pad,
+ ignore_label=self.ignore_id,
+ )
+ loss_pre = self.criterion_pre(ys_pad_lens.type_as(pre_token_length), pre_token_length)
+
+ # Compute cer/wer using attention-decoder
+ if self.training or self.error_calculator is None:
+ cer_att, wer_att = None, None
+ else:
+ ys_hat = decoder_out_1st.argmax(dim=-1)
+ cer_att, wer_att = self.error_calculator(ys_hat.cpu(), ys_pad.cpu())
+
+ return loss_att, acc_att, cer_att, wer_att, loss_pre, pre_loss_att
+
+ def sampler(self, encoder_out, encoder_out_lens, ys_pad, ys_pad_lens, pre_acoustic_embeds, chunk_mask=None):
+
+ tgt_mask = (~make_pad_mask(ys_pad_lens, maxlen=ys_pad_lens.max())[:, :, None]).to(ys_pad.device)
+ ys_pad_masked = ys_pad * tgt_mask[:, :, 0]
+ if self.share_embedding:
+ ys_pad_embed = self.decoder.output_layer.weight[ys_pad_masked]
+ else:
+ ys_pad_embed = self.decoder.embed(ys_pad_masked)
+ with torch.no_grad():
+ decoder_outs = self.decoder(
+ encoder_out, encoder_out_lens, pre_acoustic_embeds, ys_pad_lens, chunk_mask
+ )
+ decoder_out, _ = decoder_outs[0], decoder_outs[1]
+ pred_tokens = decoder_out.argmax(-1)
+ nonpad_positions = ys_pad.ne(self.ignore_id)
+ seq_lens = (nonpad_positions).sum(1)
+ same_num = ((pred_tokens == ys_pad) & nonpad_positions).sum(1)
+ input_mask = torch.ones_like(nonpad_positions)
+ bsz, seq_len = ys_pad.size()
+ for li in range(bsz):
+ target_num = (((seq_lens[li] - same_num[li].sum()).float()) * self.sampling_ratio).long()
+ if target_num > 0:
+ input_mask[li].scatter_(dim=0, index=torch.randperm(seq_lens[li])[:target_num].cuda(), value=0)
+ input_mask = input_mask.eq(1)
+ input_mask = input_mask.masked_fill(~nonpad_positions, False)
+ input_mask_expand_dim = input_mask.unsqueeze(2).to(pre_acoustic_embeds.device)
+
+ sematic_embeds = pre_acoustic_embeds.masked_fill(~input_mask_expand_dim, 0) + ys_pad_embed.masked_fill(
+ input_mask_expand_dim, 0)
+ return sematic_embeds * tgt_mask, decoder_out * tgt_mask
+
+ def sampler_with_grad(self, encoder_out, encoder_out_lens, ys_pad, ys_pad_lens, pre_acoustic_embeds, chunk_mask=None):
+ tgt_mask = (~make_pad_mask(ys_pad_lens, maxlen=ys_pad_lens.max())[:, :, None]).to(ys_pad.device)
+ ys_pad_masked = ys_pad * tgt_mask[:, :, 0]
+ if self.share_embedding:
+ ys_pad_embed = self.decoder.output_layer.weight[ys_pad_masked]
+ else:
+ ys_pad_embed = self.decoder.embed(ys_pad_masked)
+ decoder_outs = self.decoder(
+ encoder_out, encoder_out_lens, pre_acoustic_embeds, ys_pad_lens, chunk_mask
+ )
+ pre_loss_att = self.criterion_att(decoder_outs[0], ys_pad)
+ decoder_out, _ = decoder_outs[0], decoder_outs[1]
+ pred_tokens = decoder_out.argmax(-1)
+ nonpad_positions = ys_pad.ne(self.ignore_id)
+ seq_lens = (nonpad_positions).sum(1)
+ same_num = ((pred_tokens == ys_pad) & nonpad_positions).sum(1)
+ input_mask = torch.ones_like(nonpad_positions)
+ bsz, seq_len = ys_pad.size()
+ for li in range(bsz):
+ target_num = (((seq_lens[li] - same_num[li].sum()).float()) * self.sampling_ratio).long()
+ if target_num > 0:
+ input_mask[li].scatter_(dim=0, index=torch.randperm(seq_lens[li])[:target_num].cuda(), value=0)
+ input_mask = input_mask.eq(1)
+ input_mask = input_mask.masked_fill(~nonpad_positions, False)
+ input_mask_expand_dim = input_mask.unsqueeze(2).to(pre_acoustic_embeds.device)
+
+ sematic_embeds = pre_acoustic_embeds.masked_fill(~input_mask_expand_dim, 0) + ys_pad_embed.masked_fill(
+ input_mask_expand_dim, 0)
+
+ return sematic_embeds * tgt_mask, decoder_out * tgt_mask, pre_loss_att
+
+ def calc_predictor(self, encoder_out, encoder_out_lens):
+
+ encoder_out_mask = (~make_pad_mask(encoder_out_lens, maxlen=encoder_out.size(1))[:, None, :]).to(
+ encoder_out.device)
+ mask_chunk_predictor = None
+ if self.encoder.overlap_chunk_cls is not None:
+ mask_chunk_predictor = self.encoder.overlap_chunk_cls.get_mask_chunk_predictor(None,
+ device=encoder_out.device,
+ batch_size=encoder_out.size(
+ 0))
+ mask_shfit_chunk = self.encoder.overlap_chunk_cls.get_mask_shfit_chunk(None, device=encoder_out.device,
+ batch_size=encoder_out.size(0))
+ encoder_out = encoder_out * mask_shfit_chunk
+ pre_acoustic_embeds, pre_token_length, pre_alphas, pre_peak_index = self.predictor(encoder_out,
+ None,
+ encoder_out_mask,
+ ignore_id=self.ignore_id,
+ mask_chunk_predictor=mask_chunk_predictor,
+ target_label_length=None,
+ )
+ predictor_alignments, predictor_alignments_len = self.predictor.gen_frame_alignments(pre_alphas,
+ encoder_out_lens+1 if self.predictor.tail_threshold > 0.0 else encoder_out_lens)
+
+ scama_mask = None
+ if self.encoder.overlap_chunk_cls is not None and self.decoder_attention_chunk_type == 'chunk':
+ encoder_chunk_size = self.encoder.overlap_chunk_cls.chunk_size_pad_shift_cur
+ attention_chunk_center_bias = 0
+ attention_chunk_size = encoder_chunk_size
+ decoder_att_look_back_factor = self.encoder.overlap_chunk_cls.decoder_att_look_back_factor_cur
+ mask_shift_att_chunk_decoder = self.encoder.overlap_chunk_cls.\
+ get_mask_shift_att_chunk_decoder(None,
+ device=encoder_out.device,
+ batch_size=encoder_out.size(0)
+ )
+ scama_mask = self.build_scama_mask_for_cross_attention_decoder_fn(
+ predictor_alignments=predictor_alignments,
+ encoder_sequence_length=encoder_out_lens,
+ chunk_size=1,
+ encoder_chunk_size=encoder_chunk_size,
+ attention_chunk_center_bias=attention_chunk_center_bias,
+ attention_chunk_size=attention_chunk_size,
+ attention_chunk_type=self.decoder_attention_chunk_type,
+ step=None,
+ predictor_mask_chunk_hopping=mask_chunk_predictor,
+ decoder_att_look_back_factor=decoder_att_look_back_factor,
+ mask_shift_att_chunk_decoder=mask_shift_att_chunk_decoder,
+ target_length=None,
+ is_training=self.training,
+ )
+ self.scama_mask = scama_mask
+
+ return pre_acoustic_embeds, pre_token_length, pre_alphas, pre_peak_index
+
def calc_predictor_chunk(self, encoder_out, cache=None):
pre_acoustic_embeds, pre_token_length = \
self.predictor.forward_chunk(encoder_out, cache["encoder"])
return pre_acoustic_embeds, pre_token_length
+ def cal_decoder_with_predictor(self, encoder_out, encoder_out_lens, sematic_embeds, ys_pad_lens):
+ decoder_outs = self.decoder(
+ encoder_out, encoder_out_lens, sematic_embeds, ys_pad_lens, self.scama_mask
+ )
+ decoder_out = decoder_outs[0]
+ decoder_out = torch.log_softmax(decoder_out, dim=-1)
+ return decoder_out, ys_pad_lens
+
def cal_decoder_with_predictor_chunk(self, encoder_out, sematic_embeds, cache=None):
decoder_outs = self.decoder.forward_chunk(
encoder_out, sematic_embeds, cache["decoder"]
@@ -1800,4 +2251,4 @@ class ContextualParaformer(Paraformer):
"torch tensor: {}, {}, loading from tf tensor: {}, {}".format(name, data_tf.size(), name_tf,
var_dict_tf[name_tf].shape))
- return var_dict_torch_update
\ No newline at end of file
+ return var_dict_torch_update
diff --git a/funasr/models/encoder/sanm_encoder.py b/funasr/models/encoder/sanm_encoder.py
index da675864c..46eabd1eb 100644
--- a/funasr/models/encoder/sanm_encoder.py
+++ b/funasr/models/encoder/sanm_encoder.py
@@ -633,6 +633,8 @@ class SANMEncoderChunkOpt(AbsEncoder):
self.embed = torch.nn.Linear(input_size, output_size)
elif input_layer == "pe":
self.embed = SinusoidalPositionEncoder()
+ elif input_layer == "pe_online":
+ self.embed = StreamSinusoidalPositionEncoder()
else:
raise ValueError("unknown input_layer: " + input_layer)
self.normalize_before = normalize_before
@@ -818,6 +820,59 @@ class SANMEncoderChunkOpt(AbsEncoder):
return (xs_pad, intermediate_outs), olens, None
return xs_pad, olens, None
+ def _add_overlap_chunk(self, feats: np.ndarray, cache: dict = {}):
+ if len(cache) == 0:
+ return feats
+ cache["feats"] = to_device(cache["feats"], device=feats.device)
+ overlap_feats = torch.cat((cache["feats"], feats), dim=1)
+ cache["feats"] = overlap_feats[:, -(cache["chunk_size"][0] + cache["chunk_size"][2]):, :]
+ return overlap_feats
+
+ def forward_chunk(self,
+ xs_pad: torch.Tensor,
+ ilens: torch.Tensor,
+ cache: dict = None,
+ ctc: CTC = None,
+ ):
+ xs_pad *= self.output_size() ** 0.5
+ if self.embed is None:
+ xs_pad = xs_pad
+ else:
+ xs_pad = self.embed(xs_pad, cache)
+ if cache["tail_chunk"]:
+ xs_pad = to_device(cache["feats"], device=xs_pad.device)
+ else:
+ xs_pad = self._add_overlap_chunk(xs_pad, cache)
+ encoder_outs = self.encoders0(xs_pad, None, None, None, None)
+ xs_pad, masks = encoder_outs[0], encoder_outs[1]
+ intermediate_outs = []
+ if len(self.interctc_layer_idx) == 0:
+ encoder_outs = self.encoders(xs_pad, None, None, None, None)
+ xs_pad, masks = encoder_outs[0], encoder_outs[1]
+ else:
+ for layer_idx, encoder_layer in enumerate(self.encoders):
+ encoder_outs = encoder_layer(xs_pad, None, None, None, None)
+ xs_pad, masks = encoder_outs[0], encoder_outs[1]
+ if layer_idx + 1 in self.interctc_layer_idx:
+ encoder_out = xs_pad
+
+ # intermediate outputs are also normalized
+ if self.normalize_before:
+ encoder_out = self.after_norm(encoder_out)
+
+ intermediate_outs.append((layer_idx + 1, encoder_out))
+
+ if self.interctc_use_conditioning:
+ ctc_out = ctc.softmax(encoder_out)
+ xs_pad = xs_pad + self.conditioning_layer(ctc_out)
+
+ if self.normalize_before:
+ xs_pad = self.after_norm(xs_pad)
+
+ if len(intermediate_outs) > 0:
+ return (xs_pad, intermediate_outs), None, None
+ return xs_pad, ilens, None
+
def gen_tf2torch_map_dict(self):
tensor_name_prefix_torch = self.tf2torch_tensor_name_prefix_torch
tensor_name_prefix_tf = self.tf2torch_tensor_name_prefix_tf
diff --git a/funasr/runtime/python/websocket/wss_srv_asr.py b/funasr/runtime/python/websocket/wss_srv_asr.py
index 71c97e63d..6460fbf68 100644
--- a/funasr/runtime/python/websocket/wss_srv_asr.py
+++ b/funasr/runtime/python/websocket/wss_srv_asr.py
@@ -58,7 +58,8 @@ inference_pipeline_asr_online = pipeline(
model=args.asr_model_online,
ngpu=args.ngpu,
ncpu=args.ncpu,
- model_revision='v1.0.4')
+ model_revision='v1.0.6',
+ mode='paraformer_streaming')
print("model loaded")
@@ -207,4 +208,4 @@ if len(args.certfile)>0:
else:
start_server = websockets.serve(ws_serve, args.host, args.port, subprotocols=["binary"], ping_interval=None)
asyncio.get_event_loop().run_until_complete(start_server)
-asyncio.get_event_loop().run_forever()
\ No newline at end of file
+asyncio.get_event_loop().run_forever()