funasr1.0

README.md

@@ -76,57 +76,120 @@ FunASR has open-sourced a large number of pre-trained models on industrial data.
 
 <a name="quick-start"></a>
 ## Quick Start
-Quick start for new users（[tutorial](https://alibaba-damo-academy.github.io/FunASR/en/funasr/quick_start.html)）
-
-FunASR supports inference and fine-tuning of models trained on industrial data for tens of thousands of hours. For more details, please refer to [modelscope_egs](https://alibaba-damo-academy.github.io/FunASR/en/modelscope_pipeline/quick_start.html). It also supports training and fine-tuning of models on academic standard datasets. For more information, please refer to [egs](https://alibaba-damo-academy.github.io/FunASR/en/academic_recipe/asr_recipe.html).
 
 Below is a quick start tutorial. Test audio files ([Mandarin](https://isv-data.oss-cn-hangzhou.aliyuncs.com/ics/MaaS/ASR/test_audio/vad_example.wav), [English]()).
 
 ### Command-line usage
 
 ```shell
-funasr --model paraformer-zh asr_example_zh.wav
+funasr +model=paraformer-zh +vad_model="fsmn-vad" +punc_model="ct-punc" +input=asr_example_zh.wav
 ```
 
 Notes: Support recognition of single audio file, as well as file list in Kaldi-style wav.scp format: `wav_id wav_pat`
 
 ### Speech Recognition (Non-streaming)
 ```python
-from funasr import infer
+from funasr import AutoModel
 
-p = infer(model="paraformer-zh", vad_model="fsmn-vad", punc_model="ct-punc", model_hub="ms")
+model = AutoModel(model="paraformer-zh")
+# for the long duration wav, you could add vad model
+# model = AutoModel(model="paraformer-zh", vad_model="fsmn-vad", punc_model="ct-punc")
 
-res = p("asr_example_zh.wav", batch_size_token=5000)
+res = model(input="asr_example_zh.wav", batch_size=64)
 print(res)
 ```
 Note: `model_hub`: represents the model repository, `ms` stands for selecting ModelScope download, `hf` stands for selecting Huggingface download.
 
 ### Speech Recognition (Streaming)
 ```python
-from funasr import infer
-
-p = infer(model="paraformer-zh-streaming", model_hub="ms")
+from funasr import AutoModel
 
 chunk_size = [0, 10, 5] #[0, 10, 5] 600ms, [0, 8, 4] 480ms
-param_dict = {"cache": dict(), "is_final": False, "chunk_size": chunk_size, "encoder_chunk_look_back": 4, "decoder_chunk_look_back": 1}
-
-import torchaudio
-speech = torchaudio.load("asr_example_zh.wav")[0][0]
-speech_length = speech.shape[0]
-
-stride_size = chunk_size[1] * 960
-sample_offset = 0
-for sample_offset in range(0, speech_length, min(stride_size, speech_length - sample_offset)):
-    param_dict["is_final"] = True if sample_offset + stride_size >= speech_length - 1 else False
-    input = speech[sample_offset: sample_offset + stride_size]
-    rec_result = p(input=input, param_dict=param_dict)
-    print(rec_result)
+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", model_revision="v2.0.0")
+
+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(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.
 
-Quick start for new users can be found in [docs](https://alibaba-damo-academy.github.io/FunASR/en/funasr/quick_start_zh.html)
+### Voice Activity Detection (streaming)
+```python
+from funasr import AutoModel
 
+model = AutoModel(model="fsmn-vad", model_revision="v2.0.2")
+
+wav_file = f"{model.model_path}/example/asr_example.wav"
+res = model(input=wav_file)
+print(res)
+```
+### Voice Activity Detection (Non-streaming)
+```python
+from funasr import AutoModel
+
+chunk_size = 200 # ms
+model = AutoModel(model="fsmn-vad", model_revision="v2.0.2")
+
+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(input=speech_chunk,
+                cache=cache,
+                is_final=is_final,
+                chunk_size=chunk_size,
+                )
+    if len(res[0]["value"]):
+        print(res)
+```
+### Punctuation Restoration
+```python
+from funasr import AutoModel
 
+model = AutoModel(model="ct-punc", model_revision="v2.0.1")
+
+res = model(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", model_revision="v2.0.0")
+
+wav_file = f"{model.model_path}/example/asr_example.wav"
+text_file = f"{model.model_path}/example/asr_example.wav"
+res = model(input=(wav_file, text_file),
+            data_type=("sound", "text"))
+print(res)
+```
 [//]: # (FunASR supports inference and fine-tuning of models trained on industrial datasets of tens of thousands of hours. For more details, please refer to &#40;[modelscope_egs]&#40;https://alibaba-damo-academy.github.io/FunASR/en/modelscope_pipeline/quick_start.html&#41;&#41;. It also supports training and fine-tuning of models on academic standard datasets. For more details, please refer to&#40;[egs]&#40;https://alibaba-damo-academy.github.io/FunASR/en/academic_recipe/asr_recipe.html&#41;&#41;. The models include speech recognition &#40;ASR&#41;, speech activity detection &#40;VAD&#41;, punctuation recovery, language model, speaker verification, speaker separation, and multi-party conversation speech recognition. For a detailed list of models, please refer to the [Model Zoo]&#40;https://github.com/alibaba-damo-academy/FunASR/blob/main/docs/model_zoo/modelscope_models.md&#41;:)
 
 ## Deployment Service

README_zh.md

@@ -57,29 +57,28 @@ FunASR开源了大量在工业数据上预训练模型，您可以在[模型许
 （注：[🤗]()表示Huggingface模型仓库链接，[⭐]()表示ModelScope模型仓库链接）
 
 
-|                                                                              模型名字                                                                               |        任务详情        |     训练数据     | 参数量  |
-|:---------------------------------------------------------------------------------------------------------------------------------------------------------------:|:------------------:|:------------:|:----:|
-|     paraformer-zh <br> ([⭐](https://www.modelscope.cn/models/damo/speech_paraformer-large-vad-punc_asr_nat-zh-cn-16k-common-vocab8404-pytorch/summary)  [🤗]() )     |  语音识别，带时间戳输出，非实时   |  60000小时，中文  | 220M |
-|                 paraformer-zh-spk <br> ( [⭐](https://modelscope.cn/models/damo/speech_paraformer-large-vad-punc-spk_asr_nat-zh-cn/summary)  [🤗]() )                 | 分角色语音识别，带时间戳输出，非实时 |  60000小时，中文  | 220M |
-|        paraformer-zh-online <br> ( [⭐](https://modelscope.cn/models/damo/speech_paraformer-large_asr_nat-zh-cn-16k-common-vocab8404-online/summary) [🤗]() )         |      语音识别，实时       |  60000小时，中文  | 220M |
-|          paraformer-en <br> ( [⭐](https://www.modelscope.cn/models/damo/speech_paraformer-large-vad-punc_asr_nat-en-16k-common-vocab10020/summary) [🤗]() )          | 语音识别，非实时 |  50000小时，英文  | 220M |
-|                                                                paraformer-en-spk <br> ([⭐]() [🤗]() )                                                                |      语音识别，非实时      |  50000小时，英文  | 220M |
-|                      conformer-en <br> ( [⭐](https://modelscope.cn/models/damo/speech_conformer_asr-en-16k-vocab4199-pytorch/summary) [🤗]() )                       |      语音识别，非实时      |  50000小时，英文  | 220M |
-|                      ct-punc <br> ( [⭐](https://modelscope.cn/models/damo/punc_ct-transformer_cn-en-common-vocab471067-large/summary) [🤗]() )                       |      标点恢复      |  100M，中文与英文  | 1.1G | 
-|                           fsmn-vad <br> ( [⭐](https://modelscope.cn/models/damo/speech_fsmn_vad_zh-cn-16k-common-pytorch/summary) [🤗]() )                           |     语音端点检测，实时      | 5000小时，中文与英文 | 0.4M | 
-|                           fa-zh <br> ( [⭐](https://modelscope.cn/models/damo/speech_timestamp_prediction-v1-16k-offline/summary) [🤗]() )                            |   字级别时间戳预测         |  50000小时，中文  | 38M  |
+|                                                                             模型名字                                                                             |        任务详情        |     训练数据     | 参数量  |
+|:------------------------------------------------------------------------------------------------------------------------------------------------------------:|:------------------:|:------------:|:----:|
+| paraformer-zh <br> ([⭐](https://www.modelscope.cn/models/damo/speech_paraformer-large-vad-punc_asr_nat-zh-cn-16k-common-vocab8404-pytorch/summary)  [🤗]() ) |  语音识别，带时间戳输出，非实时   |  60000小时，中文  | 220M |
+|             paraformer-zh-spk <br> ( [⭐](https://modelscope.cn/models/damo/speech_paraformer-large-vad-punc-spk_asr_nat-zh-cn/summary)  [🤗]() )             | 分角色语音识别，带时间戳输出，非实时 |  60000小时，中文  | 220M |
+|   paraformer-zh-streaming <br> ( [⭐](https://modelscope.cn/models/damo/speech_paraformer-large_asr_nat-zh-cn-16k-common-vocab8404-online/summary) [🤗]() )   |      语音识别，实时       |  60000小时，中文  | 220M |
+|      paraformer-en <br> ( [⭐](https://www.modelscope.cn/models/damo/speech_paraformer-large-vad-punc_asr_nat-en-16k-common-vocab10020/summary) [🤗]() )      | 语音识别，非实时 |  50000小时，英文  | 220M |
+|                                                            paraformer-en-spk <br> ([⭐]() [🤗]() )                                                            |      语音识别，非实时      |  50000小时，英文  | 220M |
+|                  conformer-en <br> ( [⭐](https://modelscope.cn/models/damo/speech_conformer_asr-en-16k-vocab4199-pytorch/summary) [🤗]() )                   |      语音识别，非实时      |  50000小时，英文  | 220M |
+|                  ct-punc <br> ( [⭐](https://modelscope.cn/models/damo/punc_ct-transformer_cn-en-common-vocab471067-large/summary) [🤗]() )                   |      标点恢复      |  100M，中文与英文  | 1.1G | 
+|                       fsmn-vad <br> ( [⭐](https://modelscope.cn/models/damo/speech_fsmn_vad_zh-cn-16k-common-pytorch/summary) [🤗]() )                       |     语音端点检测，实时      | 5000小时，中文与英文 | 0.4M | 
+|                       fa-zh <br> ( [⭐](https://modelscope.cn/models/damo/speech_timestamp_prediction-v1-16k-offline/summary) [🤗]() )                        |   字级别时间戳预测         |  50000小时，中文  | 38M  |
 
 
 <a name="快速开始"></a>
 ## 快速开始
-FunASR支持数万小时工业数据训练的模型的推理和微调，详细信息可以参阅（[modelscope_egs](https://alibaba-damo-academy.github.io/FunASR/en/modelscope_pipeline/quick_start.html)）；也支持学术标准数据集模型的训练和微调，详细信息可以参阅（[egs](https://alibaba-damo-academy.github.io/FunASR/en/academic_recipe/asr_recipe.html)）。
 
 下面为快速上手教程，测试音频（[中文](https://isv-data.oss-cn-hangzhou.aliyuncs.com/ics/MaaS/ASR/test_audio/vad_example.wav)，[英文]()）
 
 ### 可执行命令行
 
 ```shell
-funasr --model paraformer-zh asr_example_zh.wav
+funasr +model=paraformer-zh +vad_model="fsmn-vad" +punc_model="ct-punc" +input=asr_example_zh.wav
 ```
 
 注：支持单条音频文件识别，也支持文件列表，列表为kaldi风格wav.scp：`wav_id   wav_path`
@@ -90,55 +89,109 @@ from funasr import AutoModel
 
 model = AutoModel(model="paraformer-zh")
 # for the long duration wav, you could add vad model
-# model = AutoModel(model="paraformer-zh", vad_model="fsmn-vad")
+# model = AutoModel(model="paraformer-zh", vad_model="fsmn-vad", punc_model="ct-punc")
 
 res = model(input="asr_example_zh.wav", batch_size=64)
 print(res)
 ```
 注：`model_hub`：表示模型仓库，`ms`为选择modelscope下载，`hf`为选择huggingface下载。
 
-[//]: # (### 实时语音识别)
+### 实时语音识别
 
-[//]: # (```python)
-
-[//]: # (from funasr import infer)
-
-[//]: # ()
-[//]: # (p = infer&#40;model="paraformer-zh-streaming", model_hub="ms"&#41;)
-
-[//]: # ()
-[//]: # (chunk_size = [0, 10, 5] #[0, 10, 5] 600ms, [0, 8, 4] 480ms)
-
-[//]: # (param_dict = {"cache": dict&#40;&#41;, "is_final": False, "chunk_size": chunk_size, "encoder_chunk_look_back": 4, "decoder_chunk_look_back": 1})
+```python
+from funasr import AutoModel
 
-[//]: # ()
-[//]: # (import torchaudio)
+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", model_revision="v2.0.0")
+
+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(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)
+```
 
-[//]: # (speech = torchaudio.load&#40;"asr_example_zh.wav"&#41;[0][0])
+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。
 
-[//]: # (speech_length = speech.shape[0])
+### 语音端点检测（非实时）
+```python
+from funasr import AutoModel
 
-[//]: # ()
-[//]: # (stride_size = chunk_size[1] * 960)
+model = AutoModel(model="fsmn-vad", model_revision="v2.0.2")
 
-[//]: # (sample_offset = 0)
+wav_file = f"{model.model_path}/example/asr_example.wav"
+res = model(input=wav_file)
+print(res)
+```
 
-[//]: # (for sample_offset in range&#40;0, speech_length, min&#40;stride_size, speech_length - sample_offset&#41;&#41;:)
+### 语音端点检测（实时）
+```python
+from funasr import AutoModel
 
-[//]: # (    param_dict["is_final"] = True if sample_offset + stride_size >= speech_length - 1 else False)
+chunk_size = 200 # ms
+model = AutoModel(model="fsmn-vad", model_revision="v2.0.2")
+
+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(input=speech_chunk,
+                cache=cache,
+                is_final=is_final,
+                chunk_size=chunk_size,
+                )
+    if len(res[0]["value"]):
+        print(res)
+```
 
-[//]: # (    input = speech[sample_offset: sample_offset + stride_size])
+### 标点恢复
+```python
+from funasr import AutoModel
 
-[//]: # (    rec_result = p&#40;input=input, param_dict=param_dict&#41;)
+model = AutoModel(model="ct-punc", model_revision="v2.0.1")
 
-[//]: # (    print&#40;rec_result&#41;)
+res = model(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
 
-[//]: # (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。)
+model = AutoModel(model="fa-zh", model_revision="v2.0.0")
 
-[//]: # ()
-[//]: # (更多详细用法（[新人文档]&#40;https://alibaba-damo-academy.github.io/FunASR/en/funasr/quick_start_zh.html&#41;）)
+wav_file = f"{model.model_path}/example/asr_example.wav"
+text_file = f"{model.model_path}/example/asr_example.wav"
+res = model(input=(wav_file, text_file),
+            data_type=("sound", "text"))
+print(res)
+```
+更多详细用法（[示例](examples/industrial_data_pretraining)）
 
 
 <a name="服务部署"></a>