From aa54e65287ff99ab364bc9893bff79900c8c4cc8 Mon Sep 17 00:00:00 2001
From: Lizerui9926 <110582652+Lizerui9926@users.noreply.github.com>
Date: 星期四, 20 四月 2023 11:38:45 +0800
Subject: [PATCH] Merge pull request #386 from alibaba-damo-academy/dev_sx

---
 docs/modescope_pipeline/asr_pipeline.md |   33 +++++++++++++++++++++++++++++----
 1 files changed, 29 insertions(+), 4 deletions(-)

diff --git a/docs/modescope_pipeline/asr_pipeline.md b/docs/modescope_pipeline/asr_pipeline.md
index de015b0..ee4b3ff 100644
--- a/docs/modescope_pipeline/asr_pipeline.md
+++ b/docs/modescope_pipeline/asr_pipeline.md
@@ -1,9 +1,12 @@
 # Speech Recognition
 
+> **Note**: 
+> The modelscope pipeline supports all the models in [model zoo] to inference and finetine. Here we take model of Paraformer and Paraformer-online as example to demonstrate the usage.
+
 ## Inference
 
 ### Quick start
-#### Paraformer model
+#### [Paraformer model](https://www.modelscope.cn/models/damo/speech_paraformer-large_asr_nat-zh-cn-16k-common-vocab8404-pytorch/summary)
 ```python
 from modelscope.pipelines import pipeline
 from modelscope.utils.constant import Tasks
@@ -16,6 +19,27 @@
 rec_result = inference_pipeline(audio_in='https://isv-data.oss-cn-hangzhou.aliyuncs.com/ics/MaaS/ASR/test_audio/asr_example_zh.wav')
 print(rec_result)
 ```
+#### [Paraformer-online model](https://www.modelscope.cn/models/damo/speech_paraformer_asr_nat-zh-cn-16k-common-vocab8404-online/summary)
+```python
+inference_pipeline = pipeline(
+    task=Tasks.auto_speech_recognition,
+    model='damo/speech_paraformer_asr_nat-zh-cn-16k-common-vocab8404-online',
+    )
+import soundfile
+speech, sample_rate = soundfile.read("example/asr_example.wav")
+
+param_dict = {"cache": dict(), "is_final": False}
+chunk_stride = 7680# 480ms
+# first chunk, 480ms
+speech_chunk = speech[0:chunk_stride] 
+rec_result = inference_pipeline(audio_in=speech_chunk, param_dict=param_dict)
+# next chunk, 480ms
+speech_chunk = speech[chunk_stride:chunk_stride+chunk_stride]
+rec_result = inference_pipeline(audio_in=speech_chunk, param_dict=param_dict)
+
+print(rec_result)
+```
+Full code of demo, please ref to [demo](https://github.com/alibaba-damo-academy/FunASR/discussions/241)
 
 #### API-reference
 ##### define pipeline
@@ -39,11 +63,12 @@
   In this case of `wav.scp` input, `output_dir` must be set to save the output results
 - `audio_fs`: audio sampling rate, only set when audio_in is pcm audio
 
-#### Inference with you data
 
-#### Inference with multi-threads on CPU
+### Inference with you data
 
-#### Inference with multi GPU
+### Inference with multi-threads on CPU
+
+### Inference with multi GPU
 
 ## Finetune with pipeline
 

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