3.8 KiB
Finetune
「简体中文」|「English」
Requirements
pip install funasr>=1.3.0
Data Prepare
Data examples
head -n1 data/train_example.jsonl | jq
{
"messages": [
{
"role": "system",
"content": "You are a helpful assistant."
},
{
"role": "user",
"content": "语音转写:<|startofspeech|>!https://modelscope.cn/datasets/FunAudioLLM/funasr-demo/resolve/master/audios/IT0011W0002.wav<|endofspeech|>"
},
{
"role": "assistant",
"content": "几点了?"
}
],
"speech_length": 145,
"text_length": 3
}
Full ref to data/train_example.jsonl
Description:
-
The content of systemis fixed as
You are a helpful assistant. -
The content of userincludes the prompt and the path to the audio file (enclosed between
<|startofspeech|>!and<|endofspeech|>).- The default prompts are
语音转写:andSpeech transcription:. - For corresponding languages, prompts can be combined, such as
语音转写成英文:andTranscribe speech into Chinese:. - When the text annotation corresponding to the audio file contains no Arabic numerals or punctuation marks, you can use
语音转写,不进行文本规整:andSpeech transcription without text normalization:.
- The default prompts are
-
The content of assistant corresponds to the text annotation of the audio file.
-
speech_length: The number of fbank frames of the audio file (10ms per frame).
-
text_length: The number of tokens in the annotation text of the audio file (encoded using
Qwen/Qwen3-0.6B). -
messages[2]["content"]: transcription -
speech_length: number of fbank frames of the audio file -
text_length: number of tokens of the transcription (tokenized byQwen3-0.6B)
We provide a data format conversion tool scp2jsonl.py, which can convert common speech recognition training data formats such as wav scp and transcription into the ChatML format.
train_wav.scp
BAC009S0764W0121 https://isv-data.oss-cn-hangzhou.aliyuncs.com/ics/MaaS/ASR/test_audio/BAC009S0764W0121.wav
BAC009S0916W0489 https://isv-data.oss-cn-hangzhou.aliyuncs.com/ics/MaaS/ASR/test_audio/BAC009S0916W0489.wav
train_text.txt
BAC009S0764W0121 甚至出现交易几乎停滞的情况
BAC009S0916W0489 湖北一公司以员工名义贷款数十员工负债千万
python tools/scp2jsonl.py \
++scp+file=data/train_wav.scp \
++transcript_file=data/train_text.txt \
++jsonl_file=data/train_example.jsonl
Finetune
Modify the audio_encoder_conf.freeze, audio_adaptor_conf.freeze, and llm_conf.freeze in finetune.sh.
Set the freeze parameter of the modules to be fine-tuned to false(by default, only the LLM is fine-tuned).
For more detailed parameters, refer to: SenseVoice Model Training and Testing
bash finetune.sh
Recommended Configuration
- For training data less than 1000 hours, it is recommended to fine-tune the audio_adaptor.
- For training data less than 5000 hours, it is recommended to fine-tune the audio_encoder and audio_adaptor.
- For training data greater than 10000 hours, it is recommended to perform full-parameter fine-tuning.
Model Evaluation
After model fine-tuning is completed, you can decode the model using the decode.py script:
python decode.py \
++model_dir=/path/to/finetuned \
++scp_file=data/val_wav.scp \
++output_file=output.txt
After decoding is completed, text inverse normalization needs to be applied to the annotations and recognition results, and then the WER should be calculated:
python tools/whisper_mix_normalize.py data/val_text.txt data/val_norm.txt
python tools/whisper_mix_normalize.py output.txt output_norm.txt
compute-wer data/val_norm.txt output_norm.txt cer.txt
tail -n8 cer.txt