| | |
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| | | We can send streaming audio data to server in real-time with grpc client every 300 ms e.g., and get transcribed text when stop speaking. |
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| | | The audio data is in streaming, the asr inference process is in offline. |
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| | | |
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| | | # Steps |
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| | | |
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| | | ## For the Server |
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| | | |
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| | |
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| | | |
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| | | Install the requirements for client |
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| | | ```shell |
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| | | git clone https://github.com/alibaba/FunASR.git && cd FunASR |
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| | | cd funasr/runtime/python/websocket |
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| | | pip install -r requirements_client.txt |
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| | | ``` |
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| | | |
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| | | Start client |
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| | | |
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| | | ```shell |
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| | | python ASR_client.py --host "localhost" --port 10095 --chunk_size 300 |
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| | | python ASR_client.py --host "127.0.0.1" --port 10095 --chunk_size 300 |
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| | | ``` |
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| | | |
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| | | ## Acknowledge |
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