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IrvingGao committed 2024-05-18 15:50:56 +08:00
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examples/industrial_data_pretraining/bicif_paraformer/demo.py
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#!/usr/bin/env python3
# -*- encoding: utf-8 -*-
# Copyright FunASR (https://github.com/alibaba-damo-academy/FunASR). All Rights Reserved.
# MIT License (https://opensource.org/licenses/MIT)
from funasr import AutoModel
model = AutoModel(
model="iic/speech_paraformer-large-vad-punc_asr_nat-zh-cn-16k-common-vocab8404-pytorch",
vad_model="iic/speech_fsmn_vad_zh-cn-16k-common-pytorch",
punc_model="iic/punc_ct-transformer_cn-en-common-vocab471067-large",
# spk_model="iic/speech_campplus_sv_zh-cn_16k-common",
)
res = model.generate(
input="https://isv-data.oss-cn-hangzhou.aliyuncs.com/ics/MaaS/ASR/test_audio/asr_vad_punc_example.wav",
batch_size_s=300,
batch_size_threshold_s=60,
)
print(res)
examples/industrial_data_pretraining/bicif_paraformer/demo.sh
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model="iic/speech_paraformer-large-vad-punc_asr_nat-zh-cn-16k-common-vocab8404-pytorch"
vad_model="iic/speech_fsmn_vad_zh-cn-16k-common-pytorch"
#punc_model="iic/punc_ct-transformer_zh-cn-common-vocab272727-pytorch"
punc_model="iic/punc_ct-transformer_cn-en-common-vocab471067-large"
spk_model="iic/speech_campplus_sv_zh-cn_16k-common"
python funasr/bin/inference.py \
+model=${model} \
+vad_model=${vad_model} \
+punc_model=${punc_model} \
+spk_model=${spk_model} \
+input="https://isv-data.oss-cn-hangzhou.aliyuncs.com/ics/MaaS/ASR/test_audio/asr_vad_punc_example.wav" \
+output_dir="./outputs/debug" \
+device="cpu" \
+batch_size_s=300 \
+batch_size_threshold_s=60
examples/industrial_data_pretraining/bicif_paraformer/export.py
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#!/usr/bin/env python3
# -*- encoding: utf-8 -*-
# Copyright FunASR (https://github.com/alibaba-damo-academy/FunASR). All Rights Reserved.
# MIT License (https://opensource.org/licenses/MIT)
# method1, inference from model hub
from funasr import AutoModel
model = AutoModel(
model="iic/speech_paraformer-large-vad-punc_asr_nat-zh-cn-16k-common-vocab8404-pytorch",
device="cpu",
)
res = model.export(type="onnx", quantize=False)
print(res)
# method2, inference from local path
from funasr import AutoModel
model = AutoModel(
model="/Users/zhifu/.cache/modelscope/hub/iic/speech_paraformer-large_asr_nat-zh-cn-16k-common-vocab8404-pytorch",
device="cpu",
)
res = model.export(type="onnx", quantize=False)
print(res)
examples/industrial_data_pretraining/bicif_paraformer/export.sh
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# Copyright FunASR (https://github.com/alibaba-damo-academy/FunASR). All Rights Reserved.
# MIT License (https://opensource.org/licenses/MIT)
# method1, inference from model hub
export HYDRA_FULL_ERROR=1
model="iic/speech_paraformer-large_asr_nat-zh-cn-16k-common-vocab8404-pytorch"
python -m funasr.bin.export \
++model=${model} \
++type="onnx" \
++quantize=false \
++device="cpu"
# method2, inference from local path
model="/Users/zhifu/.cache/modelscope/hub/iic/speech_paraformer-large_asr_nat-zh-cn-16k-common-vocab8404-pytorch"
python -m funasr.bin.export \
++model=${model} \
++type="onnx" \
++quantize=false \
++device="cpu"
examples/industrial_data_pretraining/bicif_paraformer/finetune.sh
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# Copyright FunASR (https://github.com/alibaba-damo-academy/FunASR). All Rights Reserved.
# MIT License (https://opensource.org/licenses/MIT)
# method1, finetune from model hub
# which gpu to train or finetune
export CUDA_VISIBLE_DEVICES="0,1"
gpu_num=$(echo $CUDA_VISIBLE_DEVICES | awk -F "," '{print NF}')
# model_name from model_hub, or model_dir in local path
## option 1, download model automatically
model_name_or_model_dir="iic/speech_paraformer-large-vad-punc_asr_nat-zh-cn-16k-common-vocab8404-pytorch"
## option 2, download model by git
#local_path_root=${workspace}/modelscope_models
#mkdir -p ${local_path_root}/${model_name_or_model_dir}
#git clone https://www.modelscope.cn/${model_name_or_model_dir}.git ${local_path_root}/${model_name_or_model_dir}
#model_name_or_model_dir=${local_path_root}/${model_name_or_model_dir}
# data dir, which contains: train.json, val.json
data_dir="../../../data/list"
train_data="${data_dir}/train.jsonl"
val_data="${data_dir}/val.jsonl"
# generate train.jsonl and val.jsonl from wav.scp and text.txt
scp2jsonl \
++scp_file_list='["../../../data/list/train_wav.scp", "../../../data/list/train_text.txt"]' \
++data_type_list='["source", "target"]' \
++jsonl_file_out="${train_data}"
scp2jsonl \
++scp_file_list='["../../../data/list/val_wav.scp", "../../../data/list/val_text.txt"]' \
++data_type_list='["source", "target"]' \
++jsonl_file_out="${val_data}"
# exp output dir
output_dir="./outputs"
log_file="${output_dir}/log.txt"
mkdir -p ${output_dir}
echo "log_file: ${log_file}"
torchrun \
--nnodes 1 \
--nproc_per_node ${gpu_num} \
../../../funasr/bin/train.py \
++model="${model_name_or_model_dir}" \
++train_data_set_list="${train_data}" \
++valid_data_set_list="${val_data}" \
++dataset_conf.batch_size=20000 \
++dataset_conf.batch_type="token" \
++dataset_conf.num_workers=4 \
++train_conf.max_epoch=50 \
++train_conf.log_interval=1 \
++train_conf.resume=false \
++train_conf.validate_interval=2000 \
++train_conf.save_checkpoint_interval=2000 \
++train_conf.keep_nbest_models=20 \
++optim_conf.lr=0.0002 \
++output_dir="${output_dir}" &> ${log_file}
examples/industrial_data_pretraining/campplus_sv/demo.py
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#!/usr/bin/env python3
# -*- encoding: utf-8 -*-
# Copyright FunASR (https://github.com/alibaba-damo-academy/FunASR). All Rights Reserved.
# MIT License (https://opensource.org/licenses/MIT)
from funasr import AutoModel
model = AutoModel(model="iic/speech_campplus_sv_zh-cn_16k-common")
res = model.generate(
input="https://isv-data.oss-cn-hangzhou.aliyuncs.com/ics/MaaS/ASR/test_audio/asr_example_zh.wav"
)
print(res)
examples/industrial_data_pretraining/conformer/demo.py
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#!/usr/bin/env python3
# -*- encoding: utf-8 -*-
# Copyright FunASR (https://github.com/alibaba-damo-academy/FunASR). All Rights Reserved.
# MIT License (https://opensource.org/licenses/MIT)
from funasr import AutoModel
model = AutoModel(model="iic/speech_conformer_asr_nat-zh-cn-16k-aishell2-vocab5212-pytorch")
res = model.generate(
input="https://modelscope.oss-cn-beijing.aliyuncs.com/test/audios/asr_example.wav",
decoding_ctc_weight=0.0,
)
print(res)
examples/industrial_data_pretraining/conformer/demo.sh
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model="iic/speech_conformer_asr_nat-zh-cn-16k-aishell1-vocab4234-pytorch"
python funasr/bin/inference.py \
+model=${model} \
+input="https://isv-data.oss-cn-hangzhou.aliyuncs.com/ics/MaaS/ASR/test_audio/asr_example_zh.wav" \
+output_dir="./outputs/debug" \
+device="cpu" \
examples/industrial_data_pretraining/contextual_paraformer/demo.py
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#!/usr/bin/env python3
# -*- encoding: utf-8 -*-
# Copyright FunASR (https://github.com/alibaba-damo-academy/FunASR). All Rights Reserved.
# MIT License (https://opensource.org/licenses/MIT)
from funasr import AutoModel
model = AutoModel(model="iic/speech_paraformer-large-contextual_asr_nat-zh-cn-16k-common-vocab8404")
res = model.generate(
input="https://isv-data.oss-cn-hangzhou.aliyuncs.com/ics/MaaS/ASR/test_audio/asr_example_zh.wav",
hotword="达摩院 魔搭",
)
print(res)
examples/industrial_data_pretraining/contextual_paraformer/demo.sh
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model="iic/speech_paraformer-large-contextual_asr_nat-zh-cn-16k-common-vocab8404"
python ../../../funasr/bin/inference.py \
+model=${model} \
+input="https://isv-data.oss-cn-hangzhou.aliyuncs.com/ics/MaaS/ASR/test_audio/asr_example_zh.wav" \
+output_dir="./outputs/debug" \
+device="cpu" \
+"hotword='达摩院 魔搭'"
examples/industrial_data_pretraining/contextual_paraformer/demo2.sh
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python -m funasr.bin.inference \
--config-path="/nfs/yufan.yf/workspace/model_download/modelscope/hub/damo/speech_paraformer-large-contextual_asr_nat-zh-cn-16k-common-vocab8404" \
--config-name="config.yaml" \
++init_param="/nfs/yufan.yf/workspace/model_download/modelscope/hub/damo/speech_paraformer-large-contextual_asr_nat-zh-cn-16k-common-vocab8404/model.pb" \
++tokenizer_conf.token_list="/nfs/yufan.yf/workspace/model_download/modelscope/hub/damo/speech_paraformer-large-contextual_asr_nat-zh-cn-16k-common-vocab8404/tokens.txt" \
++frontend_conf.cmvn_file="/nfs/yufan.yf/workspace/model_download/modelscope/hub/damo/speech_paraformer-large-contextual_asr_nat-zh-cn-16k-common-vocab8404/am.mvn" \
++input="/nfs/yufan.yf/workspace/model_download/modelscope/hub/damo/speech_paraformer-large-contextual_asr_nat-zh-cn-16k-common-vocab8404/asr_example_zh.wav" \
++output_dir="./outputs/debug2" \
++device="" \
examples/industrial_data_pretraining/contextual_paraformer/finetune.sh
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# Copyright FunASR (https://github.com/alibaba-damo-academy/FunASR). All Rights Reserved.
# MIT License (https://opensource.org/licenses/MIT)
# method1, finetune from model hub
# which gpu to train or finetune
export CUDA_VISIBLE_DEVICES="0,1"
gpu_num=$(echo $CUDA_VISIBLE_DEVICES | awk -F "," '{print NF}')
# model_name from model_hub, or model_dir in local path
## option 1, download model automatically
model_name_or_model_dir="iic/speech_paraformer-large-contextual_asr_nat-zh-cn-16k-common-vocab8404"
## option 2, download model by git
#local_path_root=${workspace}/modelscope_models
#mkdir -p ${local_path_root}/${model_name_or_model_dir}
#git clone https://www.modelscope.cn/${model_name_or_model_dir}.git ${local_path_root}/${model_name_or_model_dir}
#model_name_or_model_dir=${local_path_root}/${model_name_or_model_dir}
# data dir, which contains: train.json, val.json
data_dir="../../../data/list"
train_data="${data_dir}/train.jsonl"
val_data="${data_dir}/val.jsonl"
# generate train.jsonl and val.jsonl from wav.scp and text.txt
scp2jsonl \
++scp_file_list='["../../../data/list/train_wav.scp", "../../../data/list/train_text.txt"]' \
++data_type_list='["source", "target"]' \
++jsonl_file_out="${train_data}"
scp2jsonl \
++scp_file_list='["../../../data/list/val_wav.scp", "../../../data/list/val_text.txt"]' \
++data_type_list='["source", "target"]' \
++jsonl_file_out="${val_data}"
# exp output dir
output_dir="./outputs"
log_file="${output_dir}/log.txt"
mkdir -p ${output_dir}
echo "log_file: ${log_file}"
torchrun \
--nnodes 1 \
--nproc_per_node ${gpu_num} \
../../../funasr/bin/train.py \
++model="${model_name_or_model_dir}" \
++model_revision="${model_revision}" \
++train_data_set_list="${train_data}" \
++valid_data_set_list="${val_data}" \
++dataset_conf.batch_size=20000 \
++dataset_conf.batch_type="token" \
++dataset_conf.num_workers=4 \
++train_conf.max_epoch=50 \
++train_conf.log_interval=1 \
++train_conf.resume=false \
++train_conf.validate_interval=2000 \
++train_conf.save_checkpoint_interval=2000 \
++train_conf.keep_nbest_models=20 \
++optim_conf.lr=0.0002 \
++output_dir="${output_dir}" &> ${log_file}
examples/industrial_data_pretraining/contextual_paraformer/path.sh
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export FUNASR_DIR=$PWD/../../../
# NOTE(kan-bayashi): Use UTF-8 in Python to avoid UnicodeDecodeError when LC_ALL=C
export PYTHONIOENCODING=UTF-8
export PATH=$FUNASR_DIR/funasr/bin:$PATH
export PYTHONPATH=$FUNASR_DIR/funasr/bin:$FUNASR_DIR/funasr:$FUNASR_DIR:$PYTHONPATH
examples/industrial_data_pretraining/ct_transformer/demo.py
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#!/usr/bin/env python3
# -*- encoding: utf-8 -*-
# Copyright FunASR (https://github.com/alibaba-damo-academy/FunASR). All Rights Reserved.
# MIT License (https://opensource.org/licenses/MIT)
from funasr import AutoModel
model = AutoModel(model="iic/punc_ct-transformer_zh-cn-common-vocab272727-pytorch")
res = model.generate(
input="https://isv-data.oss-cn-hangzhou.aliyuncs.com/ics/MaaS/ASR/test_text/punc_example.txt"
)
print(res)
from funasr import AutoModel
model = AutoModel(model="iic/punc_ct-transformer_cn-en-common-vocab471067-large")
res = model.generate(
input="https://isv-data.oss-cn-hangzhou.aliyuncs.com/ics/MaaS/ASR/test_text/punc_example.txt"
)
print(res)
examples/industrial_data_pretraining/ct_transformer/demo.sh
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#model="iic/punc_ct-transformer_zh-cn-common-vocab272727-pytorch"
#
model="iic/punc_ct-transformer_cn-en-common-vocab471067-large"
python funasr/bin/inference.py \
+model=${model} \
+input="https://isv-data.oss-cn-hangzhou.aliyuncs.com/ics/MaaS/ASR/test_text/punc_example.txt" \
+output_dir="./outputs/debug" \
+device="cpu"
examples/industrial_data_pretraining/ct_transformer/export.py
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#!/usr/bin/env python3
# -*- encoding: utf-8 -*-
# Copyright FunASR (https://github.com/alibaba-damo-academy/FunASR). All Rights Reserved.
# MIT License (https://opensource.org/licenses/MIT)
# method1, inference from model hub
from funasr import AutoModel
model = AutoModel(
model="iic/punc_ct-transformer_zh-cn-common-vocab272727-pytorch",
)
res = model.export(type="onnx", quantize=False)
print(res)
# method2, inference from local path
from funasr import AutoModel
model = AutoModel(
model="/Users/zhifu/.cache/modelscope/hub/iic/punc_ct-transformer_zh-cn-common-vocab272727-pytorch"
)
res = model.export(type="onnx", quantize=False)
print(res)
examples/industrial_data_pretraining/ct_transformer/export.sh
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# Copyright FunASR (https://github.com/alibaba-damo-academy/FunASR). All Rights Reserved.
# MIT License (https://opensource.org/licenses/MIT)
# method1, inference from model hub
export HYDRA_FULL_ERROR=1
model="iic/punc_ct-transformer_zh-cn-common-vocab272727-pytorch"
python -m funasr.bin.export \
++model=${model} \
++model_revision=${model_revision} \
++type="onnx" \
++quantize=false \
++device="cpu"
# method2, inference from local path
model="/Users/zhifu/.cache/modelscope/hub/iic/punc_ct-transformer_zh-cn-common-vocab272727-pytorch"
python -m funasr.bin.export \
++model=${model} \
++type="onnx" \
++quantize=false \
++device="cpu"
examples/industrial_data_pretraining/ct_transformer_streaming/demo.py
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#!/usr/bin/env python3
# -*- encoding: utf-8 -*-
# Copyright FunASR (https://github.com/alibaba-damo-academy/FunASR). All Rights Reserved.
# MIT License (https://opensource.org/licenses/MIT)
from funasr import AutoModel
model = AutoModel(model="iic/punc_ct-transformer_zh-cn-common-vad_realtime-vocab272727")
inputs = "跨境河流是养育沿岸|人民的生命之源长期以来为帮助下游地区防灾减灾中方技术人员|在上游地区极为恶劣的自然条件下克服巨大困难甚至冒着生命危险|向印方提供汛期水文资料处理紧急事件中方重视印方在跨境河流问题上的关切|愿意进一步完善双方联合工作机制|凡是|中方能做的我们|都会去做而且会做得更好我请印度朋友们放心中国在上游的|任何开发利用都会经过科学|规划和论证兼顾上下游的利益"
vads = inputs.split("|")
rec_result_all = "outputs: "
cache = {}
for vad in vads:
rec_result = model.generate(input=vad, cache=cache)
rec_result_all += rec_result[0]["text"]
print(rec_result_all)
examples/industrial_data_pretraining/ct_transformer_streaming/demo.sh
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model="iic/punc_ct-transformer_zh-cn-common-vad_realtime-vocab272727"
python funasr/bin/inference.py \
+model=${model} \
+input="https://isv-data.oss-cn-hangzhou.aliyuncs.com/ics/MaaS/ASR/test_text/punc_example.txt" \
+output_dir="./outputs/debug" \
+device="cpu"
examples/industrial_data_pretraining/ct_transformer_streaming/export.py
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#!/usr/bin/env python3
# -*- encoding: utf-8 -*-
# Copyright FunASR (https://github.com/alibaba-damo-academy/FunASR). All Rights Reserved.
# MIT License (https://opensource.org/licenses/MIT)
# method1, inference from model hub
from funasr import AutoModel
model = AutoModel(
model="iic/punc_ct-transformer_zh-cn-common-vad_realtime-vocab272727",
)
res = model.export(type="onnx", quantize=False)
print(res)
# method2, inference from local path
from funasr import AutoModel
model = AutoModel(
model="/Users/zhifu/.cache/modelscope/hub/iic/punc_ct-transformer_zh-cn-common-vad_realtime-vocab272727"
)
res = model.export(type="onnx", quantize=False)
print(res)
examples/industrial_data_pretraining/ct_transformer_streaming/export.sh
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# Copyright FunASR (https://github.com/alibaba-damo-academy/FunASR). All Rights Reserved.
# MIT License (https://opensource.org/licenses/MIT)
# method1, inference from model hub
export HYDRA_FULL_ERROR=1
model="iic/punc_ct-transformer_zh-cn-common-vad_realtime-vocab272727"
python -m funasr.bin.export \
++model=${model} \
++model_revision=${model_revision} \
++input="https://isv-data.oss-cn-hangzhou.aliyuncs.com/ics/MaaS/ASR/test_audio/vad_example.wav" \
++type="onnx" \
++quantize=false \
++device="cpu"
# method2, inference from local path
model="/Users/zhifu/.cache/modelscope/hub/iic/punc_ct-transformer_zh-cn-common-vad_realtime-vocab272727"
python -m funasr.bin.export \
++model=${model} \
++input="https://isv-data.oss-cn-hangzhou.aliyuncs.com/ics/MaaS/ASR/test_audio/vad_example.wav" \
++type="onnx" \
++quantize=false \
++device="cpu"
examples/industrial_data_pretraining/fsmn_vad_streaming/demo.py
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#!/usr/bin/env python3
# -*- encoding: utf-8 -*-
# Copyright FunASR (https://github.com/alibaba-damo-academy/FunASR). All Rights Reserved.
# MIT License (https://opensource.org/licenses/MIT)
from funasr import AutoModel
wav_file = "https://isv-data.oss-cn-hangzhou.aliyuncs.com/ics/MaaS/ASR/test_audio/vad_example.wav"
model = AutoModel(model="iic/speech_fsmn_vad_zh-cn-16k-common-pytorch")
res = model.generate(input=wav_file)
print(res)
# [[beg1, end1], [beg2, end2], .., [begN, endN]]
# beg/end: ms
import soundfile
import os
wav_file = os.path.join(model.model_path, "example/vad_example.wav")
speech, sample_rate = soundfile.read(wav_file)
chunk_size = 200 # ms
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.generate(
input=speech_chunk,
cache=cache,
is_final=is_final,
chunk_size=chunk_size,
disable_pbar=True,
)
# print(res)
if len(res[0]["value"]):
print(res)
# 1. [[beg1, end1], [beg2, end2], .., [begN, endN]]; [[beg, end]]; [[beg1, end1], [beg2, end2]]
# 2. [[beg, -1]]
# 3. [[-1, end]]
# beg/end: ms
examples/industrial_data_pretraining/fsmn_vad_streaming/demo.sh
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model="iic/speech_fsmn_vad_zh-cn-16k-common-pytorch"
python funasr/bin/inference.py \
+model=${model} \
+input="https://isv-data.oss-cn-hangzhou.aliyuncs.com/ics/MaaS/ASR/test_audio/vad_example.wav" \
+output_dir="./outputs/debug" \
+device="cpu" \
examples/industrial_data_pretraining/fsmn_vad_streaming/export.py
+25
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#!/usr/bin/env python3
# -*- encoding: utf-8 -*-
# Copyright FunASR (https://github.com/alibaba-damo-academy/FunASR). All Rights Reserved.
# MIT License (https://opensource.org/licenses/MIT)
# method1, inference from model hub
from funasr import AutoModel
model = AutoModel(model="iic/speech_fsmn_vad_zh-cn-16k-common-pytorch")
res = model.export(type="onnx", quantize=False)
print(res)
# method2, inference from local path
from funasr import AutoModel
model = AutoModel(
model="/Users/zhifu/.cache/modelscope/hub/iic/speech_fsmn_vad_zh-cn-16k-common-pytorch"
)
res = model.export(type="onnx", quantize=False)
print(res)
examples/industrial_data_pretraining/fsmn_vad_streaming/export.sh
+24
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@@ -0,0 +1,24 @@
# Copyright FunASR (https://github.com/alibaba-damo-academy/FunASR). All Rights Reserved.
# MIT License (https://opensource.org/licenses/MIT)
# method1, inference from model hub
export HYDRA_FULL_ERROR=1
model="iic/speech_fsmn_vad_zh-cn-16k-common-pytorch"
python -m funasr.bin.export \
++model=${model} \
++model_revision=${model_revision} \
++type="onnx" \
++quantize=false
# method2, inference from local path
model="/Users/zhifu/.cache/modelscope/hub/iic/speech_fsmn_vad_zh-cn-16k-common-pytorch"
python -m funasr.bin.export \
++model=${model} \
++type="onnx" \
++quantize=false
examples/industrial_data_pretraining/lcbnet/compute_wer_details.py
+735
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#!/usr/bin/env python3
# -*- coding: utf-8 -*-
from enum import Enum
import re, sys, unicodedata
import codecs
import argparse
from tqdm import tqdm
import os
import pdb
remove_tag = False
spacelist = [" ", "\t", "\r", "\n"]
puncts = [
"!",
",",
"?",
"、",
"。",
"！",
"，",
"；",
"？",
"：",
"「",
"」",
"︰",
"『",
"』",
"《",
"》",
]
class Code(Enum):
match = 1
substitution = 2
insertion = 3
deletion = 4
class WordError(object):
def __init__(self):
self.errors = {
Code.substitution: 0,
Code.insertion: 0,
Code.deletion: 0,
}
self.ref_words = 0
def get_wer(self):
assert self.ref_words != 0
errors = (
self.errors[Code.substitution]
+ self.errors[Code.insertion]
+ self.errors[Code.deletion]
)
return 100.0 * errors / self.ref_words
def get_result_string(self):
return (
f"error_rate={self.get_wer():.4f}, "
f"ref_words={self.ref_words}, "
f"subs={self.errors[Code.substitution]}, "
f"ins={self.errors[Code.insertion]}, "
f"dels={self.errors[Code.deletion]}"
)
def characterize(string):
res = []
i = 0
while i < len(string):
char = string[i]
if char in puncts:
i += 1
continue
cat1 = unicodedata.category(char)
# https://unicodebook.readthedocs.io/unicode.html#unicode-categories
if cat1 == "Zs" or cat1 == "Cn" or char in spacelist: # space or not assigned
i += 1
continue
if cat1 == "Lo": # letter-other
res.append(char)
i += 1
else:
# some input looks like: <unk><noise>, we want to separate it to two words.
sep = " "
if char == "<":
sep = ">"
j = i + 1
while j < len(string):
c = string[j]
if ord(c) >= 128 or (c in spacelist) or (c == sep):
break
j += 1
if j < len(string) and string[j] == ">":
j += 1
res.append(string[i:j])
i = j
return res
def stripoff_tags(x):
if not x:
return ""
chars = []
i = 0
T = len(x)
while i < T:
if x[i] == "<":
while i < T and x[i] != ">":
i += 1
i += 1
else:
chars.append(x[i])
i += 1
return "".join(chars)
def normalize(sentence, ignore_words, cs, split=None):
"""sentence, ignore_words are both in unicode"""
new_sentence = []
for token in sentence:
x = token
if not cs:
x = x.upper()
if x in ignore_words:
continue
if remove_tag:
x = stripoff_tags(x)
if not x:
continue
if split and x in split:
new_sentence += split[x]
else:
new_sentence.append(x)
return new_sentence
class Calculator:
def __init__(self):
self.data = {}
self.space = []
self.cost = {}
self.cost["cor"] = 0
self.cost["sub"] = 1
self.cost["del"] = 1
self.cost["ins"] = 1
def calculate(self, lab, rec):
# Initialization
lab.insert(0, "")
rec.insert(0, "")
while len(self.space) < len(lab):
self.space.append([])
for row in self.space:
for element in row:
element["dist"] = 0
element["error"] = "non"
while len(row) < len(rec):
row.append({"dist": 0, "error": "non"})
for i in range(len(lab)):
self.space[i][0]["dist"] = i
self.space[i][0]["error"] = "del"
for j in range(len(rec)):
self.space[0][j]["dist"] = j
self.space[0][j]["error"] = "ins"
self.space[0][0]["error"] = "non"
for token in lab:
if token not in self.data and len(token) > 0:
self.data[token] = {"all": 0, "cor": 0, "sub": 0, "ins": 0, "del": 0}
for token in rec:
if token not in self.data and len(token) > 0:
self.data[token] = {"all": 0, "cor": 0, "sub": 0, "ins": 0, "del": 0}
# Computing edit distance
for i, lab_token in enumerate(lab):
for j, rec_token in enumerate(rec):
if i == 0 or j == 0:
continue
min_dist = sys.maxsize
min_error = "none"
dist = self.space[i - 1][j]["dist"] + self.cost["del"]
error = "del"
if dist < min_dist:
min_dist = dist
min_error = error
dist = self.space[i][j - 1]["dist"] + self.cost["ins"]
error = "ins"
if dist < min_dist:
min_dist = dist
min_error = error
if lab_token == rec_token.replace("<BIAS>", ""):
dist = self.space[i - 1][j - 1]["dist"] + self.cost["cor"]
error = "cor"
else:
dist = self.space[i - 1][j - 1]["dist"] + self.cost["sub"]
error = "sub"
if dist < min_dist:
min_dist = dist
min_error = error
self.space[i][j]["dist"] = min_dist
self.space[i][j]["error"] = min_error
# Tracing back
result = {
"lab": [],
"rec": [],
"code": [],
"all": 0,
"cor": 0,
"sub": 0,
"ins": 0,
"del": 0,
}
i = len(lab) - 1
j = len(rec) - 1
while True:
if self.space[i][j]["error"] == "cor": # correct
if len(lab[i]) > 0:
self.data[lab[i]]["all"] = self.data[lab[i]]["all"] + 1
self.data[lab[i]]["cor"] = self.data[lab[i]]["cor"] + 1
result["all"] = result["all"] + 1
result["cor"] = result["cor"] + 1
result["lab"].insert(0, lab[i])
result["rec"].insert(0, rec[j])
result["code"].insert(0, Code.match)
i = i - 1
j = j - 1
elif self.space[i][j]["error"] == "sub": # substitution
if len(lab[i]) > 0:
self.data[lab[i]]["all"] = self.data[lab[i]]["all"] + 1
self.data[lab[i]]["sub"] = self.data[lab[i]]["sub"] + 1
result["all"] = result["all"] + 1
result["sub"] = result["sub"] + 1
result["lab"].insert(0, lab[i])
result["rec"].insert(0, rec[j])
result["code"].insert(0, Code.substitution)
i = i - 1
j = j - 1
elif self.space[i][j]["error"] == "del": # deletion
if len(lab[i]) > 0:
self.data[lab[i]]["all"] = self.data[lab[i]]["all"] + 1
self.data[lab[i]]["del"] = self.data[lab[i]]["del"] + 1
result["all"] = result["all"] + 1
result["del"] = result["del"] + 1
result["lab"].insert(0, lab[i])
result["rec"].insert(0, "")
result["code"].insert(0, Code.deletion)
i = i - 1
elif self.space[i][j]["error"] == "ins": # insertion
if len(rec[j]) > 0:
self.data[rec[j]]["ins"] = self.data[rec[j]]["ins"] + 1
result["ins"] = result["ins"] + 1
result["lab"].insert(0, "")
result["rec"].insert(0, rec[j])
result["code"].insert(0, Code.insertion)
j = j - 1
elif self.space[i][j]["error"] == "non": # starting point
break
else: # shouldn't reach here
print(
"this should not happen , i = {i} , j = {j} , error = {error}".format(
i=i, j=j, error=self.space[i][j]["error"]
)
)
return result
def overall(self):
result = {"all": 0, "cor": 0, "sub": 0, "ins": 0, "del": 0}
for token in self.data:
result["all"] = result["all"] + self.data[token]["all"]
result["cor"] = result["cor"] + self.data[token]["cor"]
result["sub"] = result["sub"] + self.data[token]["sub"]
result["ins"] = result["ins"] + self.data[token]["ins"]
result["del"] = result["del"] + self.data[token]["del"]
return result
def cluster(self, data):
result = {"all": 0, "cor": 0, "sub": 0, "ins": 0, "del": 0}
for token in data:
if token in self.data:
result["all"] = result["all"] + self.data[token]["all"]
result["cor"] = result["cor"] + self.data[token]["cor"]
result["sub"] = result["sub"] + self.data[token]["sub"]
result["ins"] = result["ins"] + self.data[token]["ins"]
result["del"] = result["del"] + self.data[token]["del"]
return result
def keys(self):
return list(self.data.keys())
def width(string):
return sum(1 + (unicodedata.east_asian_width(c) in "AFW") for c in string)
def default_cluster(word):
unicode_names = [unicodedata.name(char) for char in word]
for i in reversed(range(len(unicode_names))):
if unicode_names[i].startswith("DIGIT"): # 1
unicode_names[i] = "Number" # 'DIGIT'
elif unicode_names[i].startswith("CJK UNIFIED IDEOGRAPH") or unicode_names[i].startswith(
"CJK COMPATIBILITY IDEOGRAPH"
):
# 明 / 郎
unicode_names[i] = "Mandarin" # 'CJK IDEOGRAPH'
elif unicode_names[i].startswith("LATIN CAPITAL LETTER") or unicode_names[i].startswith(
"LATIN SMALL LETTER"
):
# A / a
unicode_names[i] = "English" # 'LATIN LETTER'
elif unicode_names[i].startswith("HIRAGANA LETTER"): # は こ め
unicode_names[i] = "Japanese" # 'GANA LETTER'
elif (
unicode_names[i].startswith("AMPERSAND")
or unicode_names[i].startswith("APOSTROPHE")
or unicode_names[i].startswith("COMMERCIAL AT")
or unicode_names[i].startswith("DEGREE CELSIUS")
or unicode_names[i].startswith("EQUALS SIGN")
or unicode_names[i].startswith("FULL STOP")
or unicode_names[i].startswith("HYPHEN-MINUS")
or unicode_names[i].startswith("LOW LINE")
or unicode_names[i].startswith("NUMBER SIGN")
or unicode_names[i].startswith("PLUS SIGN")
or unicode_names[i].startswith("SEMICOLON")
):
# & / ' / @ / ℃ / = / . / - / _ / # / + / ;
del unicode_names[i]
else:
return "Other"
if len(unicode_names) == 0:
return "Other"
if len(unicode_names) == 1:
return unicode_names[0]
for i in range(len(unicode_names) - 1):
if unicode_names[i] != unicode_names[i + 1]:
return "Other"
return unicode_names[0]
def get_args():
parser = argparse.ArgumentParser(description="wer cal")
parser.add_argument("--ref", type=str, help="Text input path")
parser.add_argument("--ref_ocr", type=str, help="Text input path")
parser.add_argument("--rec_name", type=str, action="append", default=[])
parser.add_argument("--rec_file", type=str, action="append", default=[])
parser.add_argument("--verbose", type=int, default=1, help="show")
parser.add_argument("--char", type=bool, default=True, help="show")
args = parser.parse_args()
return args
def main(args):
cluster_file = ""
ignore_words = set()
tochar = args.char
verbose = args.verbose
padding_symbol = " "
case_sensitive = False
max_words_per_line = sys.maxsize
split = None
if not case_sensitive:
ig = set([w.upper() for w in ignore_words])
ignore_words = ig
default_clusters = {}
default_words = {}
ref_file = args.ref
ref_ocr = args.ref_ocr
rec_files = args.rec_file
rec_names = args.rec_name
assert len(rec_files) == len(rec_names)
# load ocr
ref_ocr_dict = {}
with codecs.open(ref_ocr, "r", "utf-8") as fh:
for line in fh:
if "$" in line:
line = line.replace("$", " ")
if tochar:
array = characterize(line)
else:
array = line.strip().split()
if len(array) == 0:
continue
fid = array[0]
ref_ocr_dict[fid] = normalize(array[1:], ignore_words, case_sensitive, split)
if split and not case_sensitive:
newsplit = dict()
for w in split:
words = split[w]
for i in range(len(words)):
words[i] = words[i].upper()
newsplit[w.upper()] = words
split = newsplit
rec_sets = {}
calculators_dict = dict()
ub_wer_dict = dict()
hotwords_related_dict = dict() # 记录recall相关的内容
for i, hyp_file in enumerate(rec_files):
rec_sets[rec_names[i]] = dict()
with codecs.open(hyp_file, "r", "utf-8") as fh:
for line in fh:
if tochar:
array = characterize(line)
else:
array = line.strip().split()
if len(array) == 0:
continue
fid = array[0]
rec_sets[rec_names[i]][fid] = normalize(
array[1:], ignore_words, case_sensitive, split
)
calculators_dict[rec_names[i]] = Calculator()
ub_wer_dict[rec_names[i]] = {"u_wer": WordError(), "b_wer": WordError(), "wer": WordError()}
hotwords_related_dict[rec_names[i]] = {"tp": 0, "tn": 0, "fp": 0, "fn": 0}
# tp: 热词在label里，同时在rec里
# tn: 热词不在label里，同时不在rec里
# fp: 热词不在label里，但是在rec里
# fn: 热词在label里，但是不在rec里
# record wrong label but in ocr
wrong_rec_but_in_ocr_dict = {}
for rec_name in rec_names:
wrong_rec_but_in_ocr_dict[rec_name] = 0
_file_total_len = 0
with os.popen("cat {} | wc -l".format(ref_file)) as pipe:
_file_total_len = int(pipe.read().strip())
# compute error rate on the interaction of reference file and hyp file
for line in tqdm(open(ref_file, "r", encoding="utf-8"), total=_file_total_len):
if tochar:
array = characterize(line)
else:
array = line.rstrip("\n").split()
if len(array) == 0:
continue
fid = array[0]
lab = normalize(array[1:], ignore_words, case_sensitive, split)
if verbose:
print("\nutt: %s" % fid)
ocr_text = ref_ocr_dict[fid]
ocr_set = set(ocr_text)
print("ocr: {}".format(" ".join(ocr_text)))
list_match = [] # 指label里面在ocr里面的内容
list_not_mathch = []
tmp_error = 0
tmp_match = 0
for index in range(len(lab)):
# text_list.append(uttlist[index+1])
if lab[index] not in ocr_set:
tmp_error += 1
list_not_mathch.append(lab[index])
else:
tmp_match += 1
list_match.append(lab[index])
print("label in ocr: {}".format(" ".join(list_match)))
# for each reco file
base_wrong_ocr_wer = None
ocr_wrong_ocr_wer = None
for rec_name in rec_names:
rec_set = rec_sets[rec_name]
if fid not in rec_set:
continue
rec = rec_set[fid]
# print(rec)
for word in rec + lab:
if word not in default_words:
default_cluster_name = default_cluster(word)
if default_cluster_name not in default_clusters:
default_clusters[default_cluster_name] = {}
if word not in default_clusters[default_cluster_name]:
default_clusters[default_cluster_name][word] = 1
default_words[word] = default_cluster_name
result = calculators_dict[rec_name].calculate(lab.copy(), rec.copy())
if verbose:
if result["all"] != 0:
wer = (
float(result["ins"] + result["sub"] + result["del"]) * 100.0 / result["all"]
)
else:
wer = 0.0
print("WER(%s): %4.2f %%" % (rec_name, wer), end=" ")
print(
"N=%d C=%d S=%d D=%d I=%d"
% (result["all"], result["cor"], result["sub"], result["del"], result["ins"])
)
# print(result['rec'])
wrong_rec_but_in_ocr = []
for idx in range(len(result["lab"])):
if result["lab"][idx] != "":
if result["lab"][idx] != result["rec"][idx].replace("<BIAS>", ""):
if result["lab"][idx] in list_match:
wrong_rec_but_in_ocr.append(result["lab"][idx])
wrong_rec_but_in_ocr_dict[rec_name] += 1
print("wrong_rec_but_in_ocr: {}".format(" ".join(wrong_rec_but_in_ocr)))
if rec_name == "base":
base_wrong_ocr_wer = len(wrong_rec_but_in_ocr)
if "ocr" in rec_name or "hot" in rec_name:
ocr_wrong_ocr_wer = len(wrong_rec_but_in_ocr)
if ocr_wrong_ocr_wer < base_wrong_ocr_wer:
print(
"{} {} helps, {} -> {}".format(
fid, rec_name, base_wrong_ocr_wer, ocr_wrong_ocr_wer
)
)
elif ocr_wrong_ocr_wer > base_wrong_ocr_wer:
print(
"{} {} hurts, {} -> {}".format(
fid, rec_name, base_wrong_ocr_wer, ocr_wrong_ocr_wer
)
)
# recall = 0
# false_alarm = 0
# for idx in range(len(result['lab'])):
# if "<BIAS>" in result['rec'][idx]:
# if result['rec'][idx].replace("<BIAS>", "") in list_match:
# recall += 1
# else:
# false_alarm += 1
# print("bias hotwords recall: {}, fa: {}, list_match {}, recall: {:.2f}, fa: {:.2f}".format(
# recall, false_alarm, len(list_match), recall / len(list_match) if len(list_match) != 0 else 0, false_alarm / len(list_match) if len(list_match) != 0 else 0
# ))
# tp: 热词在label里，同时在rec里
# tn: 热词不在label里，同时不在rec里
# fp: 热词不在label里，但是在rec里
# fn: 热词在label里，但是不在rec里
_rec_list = [word.replace("<BIAS>", "") for word in rec]
_label_list = [word for word in lab]
_tp = _tn = _fp = _fn = 0
hot_true_list = [hotword for hotword in ocr_text if hotword in _label_list]
hot_bad_list = [hotword for hotword in ocr_text if hotword not in _label_list]
for badhotword in hot_bad_list:
count = len([word for word in _rec_list if word == badhotword])
# print(f"bad {badhotword} count: {count}")
# for word in _rec_list:
# if badhotword == word:
# count += 1
if count == 0:
hotwords_related_dict[rec_name]["tn"] += 1
_tn += 1
# fp: 0
else:
hotwords_related_dict[rec_name]["fp"] += count
_fp += count
# tn: 0
# if badhotword in _rec_list:
# hotwords_related_dict[rec_name]['fp'] += 1
# else:
# hotwords_related_dict[rec_name]['tn'] += 1
for hotword in hot_true_list:
true_count = len([word for word in _label_list if hotword == word])
rec_count = len([word for word in _rec_list if hotword == word])
# print(f"good {hotword} true_count: {true_count}, rec_count: {rec_count}")
if rec_count == true_count:
hotwords_related_dict[rec_name]["tp"] += true_count
_tp += true_count
elif rec_count > true_count:
hotwords_related_dict[rec_name]["tp"] += true_count
# fp: 不在label里，但是在rec里
hotwords_related_dict[rec_name]["fp"] += rec_count - true_count
_tp += true_count
_fp += rec_count - true_count
else:
hotwords_related_dict[rec_name]["tp"] += rec_count
# fn: 热词在label里，但是不在rec里
hotwords_related_dict[rec_name]["fn"] += true_count - rec_count
_tp += rec_count
_fn += true_count - rec_count
print(
"hotword: tp: {}, tn: {}, fp: {}, fn: {}, all: {}, recall: {:.2f}%".format(
_tp,
_tn,
_fp,
_fn,
sum([_tp, _tn, _fp, _fn]),
_tp / (_tp + _fn) * 100 if (_tp + _fn) != 0 else 0,
)
)
# if hotword in _rec_list:
# hotwords_related_dict[rec_name]['tp'] += 1
# else:
# hotwords_related_dict[rec_name]['fn'] += 1
# 计算uwer, bwer, wer
for code, rec_word, lab_word in zip(result["code"], result["rec"], result["lab"]):
if code == Code.match:
ub_wer_dict[rec_name]["wer"].ref_words += 1
if lab_word in hot_true_list:
# tmp_ref.append(ref_tokens[ref_idx])
ub_wer_dict[rec_name]["b_wer"].ref_words += 1
else:
ub_wer_dict[rec_name]["u_wer"].ref_words += 1
elif code == Code.substitution:
ub_wer_dict[rec_name]["wer"].ref_words += 1
ub_wer_dict[rec_name]["wer"].errors[Code.substitution] += 1
if lab_word in hot_true_list:
# tmp_ref.append(ref_tokens[ref_idx])
ub_wer_dict[rec_name]["b_wer"].ref_words += 1
ub_wer_dict[rec_name]["b_wer"].errors[Code.substitution] += 1
else:
ub_wer_dict[rec_name]["u_wer"].ref_words += 1
ub_wer_dict[rec_name]["u_wer"].errors[Code.substitution] += 1
elif code == Code.deletion:
ub_wer_dict[rec_name]["wer"].ref_words += 1
ub_wer_dict[rec_name]["wer"].errors[Code.deletion] += 1
if lab_word in hot_true_list:
# tmp_ref.append(ref_tokens[ref_idx])
ub_wer_dict[rec_name]["b_wer"].ref_words += 1
ub_wer_dict[rec_name]["b_wer"].errors[Code.deletion] += 1
else:
ub_wer_dict[rec_name]["u_wer"].ref_words += 1
ub_wer_dict[rec_name]["u_wer"].errors[Code.deletion] += 1
elif code == Code.insertion:
ub_wer_dict[rec_name]["wer"].errors[Code.insertion] += 1
if rec_word in hot_true_list:
ub_wer_dict[rec_name]["b_wer"].errors[Code.insertion] += 1
else:
ub_wer_dict[rec_name]["u_wer"].errors[Code.insertion] += 1
space = {}
space["lab"] = []
space["rec"] = []
for idx in range(len(result["lab"])):
len_lab = width(result["lab"][idx])
len_rec = width(result["rec"][idx])
length = max(len_lab, len_rec)
space["lab"].append(length - len_lab)
space["rec"].append(length - len_rec)
upper_lab = len(result["lab"])
upper_rec = len(result["rec"])
lab1, rec1 = 0, 0
while lab1 < upper_lab or rec1 < upper_rec:
if verbose > 1:
print("lab(%s):" % fid.encode("utf-8"), end=" ")
else:
print("lab:", end=" ")
lab2 = min(upper_lab, lab1 + max_words_per_line)
for idx in range(lab1, lab2):
token = result["lab"][idx]
print("{token}".format(token=token), end="")
for n in range(space["lab"][idx]):
print(padding_symbol, end="")
print(" ", end="")
print()
if verbose > 1:
print("rec(%s):" % fid.encode("utf-8"), end=" ")
else:
print("rec:", end=" ")
rec2 = min(upper_rec, rec1 + max_words_per_line)
for idx in range(rec1, rec2):
token = result["rec"][idx]
print("{token}".format(token=token), end="")
for n in range(space["rec"][idx]):
print(padding_symbol, end="")
print(" ", end="")
print()
# print('\n', end='\n')
lab1 = lab2
rec1 = rec2
print("\n", end="\n")
# break
if verbose:
print("===========================================================================")
print()
print(wrong_rec_but_in_ocr_dict)
for rec_name in rec_names:
result = calculators_dict[rec_name].overall()
if result["all"] != 0:
wer = float(result["ins"] + result["sub"] + result["del"]) * 100.0 / result["all"]
else:
wer = 0.0
print("{} Overall -> {:4.2f} %".format(rec_name, wer), end=" ")
print(
"N=%d C=%d S=%d D=%d I=%d"
% (result["all"], result["cor"], result["sub"], result["del"], result["ins"])
)
print(f"WER: {ub_wer_dict[rec_name]['wer'].get_result_string()}")
print(f"U-WER: {ub_wer_dict[rec_name]['u_wer'].get_result_string()}")
print(f"B-WER: {ub_wer_dict[rec_name]['b_wer'].get_result_string()}")
print(
"hotword: tp: {}, tn: {}, fp: {}, fn: {}, all: {}, recall: {:.2f}%".format(
hotwords_related_dict[rec_name]["tp"],
hotwords_related_dict[rec_name]["tn"],
hotwords_related_dict[rec_name]["fp"],
hotwords_related_dict[rec_name]["fn"],
sum([v for k, v in hotwords_related_dict[rec_name].items()]),
(
hotwords_related_dict[rec_name]["tp"]
/ (
hotwords_related_dict[rec_name]["tp"]
+ hotwords_related_dict[rec_name]["fn"]
)
* 100
if hotwords_related_dict[rec_name]["tp"] + hotwords_related_dict[rec_name]["fn"]
!= 0
else 0
),
)
)
# tp: 热词在label里，同时在rec里
# tn: 热词不在label里，同时不在rec里
# fp: 热词不在label里，但是在rec里
# fn: 热词在label里，但是不在rec里
if not verbose:
print()
print()
if __name__ == "__main__":
args = get_args()
# print("")
print(args)
main(args)
examples/industrial_data_pretraining/lcbnet/demo.py
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#!/usr/bin/env python3
# -*- encoding: utf-8 -*-
# Copyright FunASR (https://github.com/alibaba-damo-academy/FunASR). All Rights Reserved.
# MIT License (https://opensource.org/licenses/MIT)
from funasr import AutoModel
model = AutoModel(model="iic/LCB-NET", model_revision="v1.0.0")
res = model.generate(
input=(
"https://www.modelscope.cn/api/v1/models/iic/LCB-NET/repo?Revision=master&FilePath=example/asr_example.wav",
"https://www.modelscope.cn/api/v1/models/iic/LCB-NET/repo?Revision=master&FilePath=example/ocr.txt",
),
data_type=("sound", "text"),
)
print(res)
examples/industrial_data_pretraining/lcbnet/demo.sh
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file_dir="/home/yf352572/.cache/modelscope/hub/iic/LCB-NET/"
CUDA_VISIBLE_DEVICES="0,1"
inference_device="cuda"
if [ ${inference_device} == "cuda" ]; then
nj=$(echo $CUDA_VISIBLE_DEVICES | awk -F "," '{print NF}')
else
inference_batch_size=1
CUDA_VISIBLE_DEVICES=""
for JOB in $(seq ${nj}); do
CUDA_VISIBLE_DEVICES=$CUDA_VISIBLE_DEVICES"-1,"
done
fi
inference_dir="outputs/slidespeech_dev"
_logdir="${inference_dir}/logdir"
echo "inference_dir: ${inference_dir}"
mkdir -p "${_logdir}"
key_file1=${file_dir}/dev/wav.scp
key_file2=${file_dir}/dev/ocr.txt
split_scps1=
split_scps2=
for JOB in $(seq "${nj}"); do
split_scps1+=" ${_logdir}/wav.${JOB}.scp"
split_scps2+=" ${_logdir}/ocr.${JOB}.txt"
done
utils/split_scp.pl "${key_file1}" ${split_scps1}
utils/split_scp.pl "${key_file2}" ${split_scps2}
gpuid_list_array=(${CUDA_VISIBLE_DEVICES//,/ })
for JOB in $(seq ${nj}); do
{
id=$((JOB-1))
gpuid=${gpuid_list_array[$id]}
export CUDA_VISIBLE_DEVICES=${gpuid}
python -m funasr.bin.inference \
--config-path=${file_dir} \
--config-name="config.yaml" \
++init_param=${file_dir}/model.pt \
++tokenizer_conf.token_list=${file_dir}/tokens.txt \
++input=[${_logdir}/wav.${JOB}.scp,${_logdir}/ocr.${JOB}.txt] \
+data_type='["kaldi_ark", "text"]' \
++tokenizer_conf.bpemodel=${file_dir}/bpe.pt \
++normalize_conf.stats_file=${file_dir}/am.mvn \
++output_dir="${inference_dir}/${JOB}" \
++device="${inference_device}" \
++ncpu=1 \
++disable_log=true &> ${_logdir}/log.${JOB}.txt
}&
done
wait
mkdir -p ${inference_dir}/1best_recog
for JOB in $(seq "${nj}"); do
cat "${inference_dir}/${JOB}/1best_recog/token" >> "${inference_dir}/1best_recog/token"
done
echo "Computing WER ..."
sed -e 's/ /\t/' -e 's/ //g' -e 's/▁/ /g' -e 's/\t /\t/' ${inference_dir}/1best_recog/token > ${inference_dir}/1best_recog/token.proc
cp ${file_dir}/dev/text ${inference_dir}/1best_recog/token.ref
cp ${file_dir}/dev/ocr.list ${inference_dir}/1best_recog/ocr.list
python utils/compute_wer.py ${inference_dir}/1best_recog/token.ref ${inference_dir}/1best_recog/token.proc ${inference_dir}/1best_recog/token.cer
tail -n 3 ${inference_dir}/1best_recog/token.cer
./run_bwer_recall.sh ${inference_dir}/1best_recog/
tail -n 6 ${inference_dir}/1best_recog/BWER-UWER.results |head -n 5
examples/industrial_data_pretraining/lcbnet/run_bwer_recall.sh
+11
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#now_result_name=asr_conformer_acc1_lr002_warm20000/decode_asr_asr_model_valid.acc.ave
#hotword_type=ocr_1ngram_top10_hotwords_list
hot_exp_suf=$1
python compute_wer_details.py --v 1 \
--ref ${hot_exp_suf}/token.ref \
--ref_ocr ${hot_exp_suf}/ocr.list \
--rec_name base \
--rec_file ${hot_exp_suf}/token.proc \
> ${hot_exp_suf}/BWER-UWER.results
examples/industrial_data_pretraining/lcbnet/utils
+1
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@@ -0,0 +1 @@
../../aishell/paraformer/utils
examples/industrial_data_pretraining/llm_asr/conf/template.yaml
+89
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@@ -0,0 +1,89 @@
# This is an example that demonstrates how to configure a model file.
# You can modify the configuration according to your own requirements.
# to print the register_table:
# from funasr.register import tables
# tables.print()
# network architecture
model: LLMASR
model_conf:
lsm_weight: 0.1 # label smoothing option
length_normalized_loss: true
# encoder
encoder: WhisperWarp
encoder_conf:
hub: funasr
init_param_path: "/nfs/maziyang.mzy/models/Whisper-large-v2"
freeze: true
llm: Vicuna
llm_conf:
hub: hf
init_param_path: "/nfs/maziyang.mzy/models/vicuna-7b-v1.5"
freeze: true
adaptor: Linear
adaptor_conf:
downsample_rate: 5
llm_dim: 4096
encoder_dim: 512
# frontend related
frontend: WhisperFrontend
frontend_conf:
fs: 16000
whisper_model: large
do_pad_trim: true
specaug: SpecAugLFR
specaug_conf:
apply_time_warp: false
time_warp_window: 5
time_warp_mode: bicubic
apply_freq_mask: true
freq_mask_width_range:
- 0
- 30
lfr_rate: 6
num_freq_mask: 1
apply_time_mask: true
time_mask_width_range:
- 0
- 12
num_time_mask: 1
train_conf:
accum_grad: 1
grad_clip: 5
max_epoch: 150
keep_nbest_models: 10
log_interval: 10
optim: adamw
optim_conf:
lr: 0.0001
weight_decay: 0.000001
scheduler: warmuplr
scheduler_conf:
warmup_steps: 1500
dataset: AudioLLMDataset
dataset_conf:
index_ds: IndexDSJsonl
batch_sampler: BatchSampler
batch_type: example # example or length
batch_size: 8 # if batch_type is example, batch_size is the numbers of samples; if length, batch_size is source_token_len+target_token_len;
max_token_length: 2048 # filter samples if source_token_len+target_token_len > max_token_length,
buffer_size: 500
shuffle: True
num_workers: 4
preprocessor_text: TextPreprocessRemovePunctuation
tokenizer: HuggingfaceTokenizer
tokenizer_conf:
unk_symbol: <unk>
init_param_path: "/nfs/maziyang.mzy/models/vicuna-7b-v1.5"
examples/industrial_data_pretraining/llm_asr/conf/whisper_qwen_linear.yaml
+94
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# This is an example that demonstrates how to configure a model file.
# You can modify the configuration according to your own requirements.
# to print the register_table:
# from funasr.register import tables
# tables.print()
# network architecture
model: LLMASR
model_conf:
lsm_weight: 0.1 # label smoothing option
length_normalized_loss: true
# encoder
audio_encoder: "/nfs/zhifu.gzf/init_model/Whisper-large-v3" #iic/Whisper-large-v3
audio_encoder_conf:
hub: ms
freeze: true
llm: Qwen1.5-7b-chat
llm_conf:
hub: hf
freeze: true
init_param_path: "/nfs/zhifu.gzf/init_model/qwen/Qwen1___5-7B-Chat"
audio_adaptor: Linear
audio_adaptor_conf:
downsample_rate: 5
llm_dim: 4096
encoder_dim: 512
# frontend related
frontend: WhisperFrontend
frontend_conf:
fs: 16000
whisper_model: large-v3
do_pad_trim: true
permute: true # true: [bs, frames, dims]; false: [bs, dims, frames]
specaug: SpecAugLFR
specaug_conf:
apply_time_warp: false
time_warp_window: 5
time_warp_mode: bicubic
apply_freq_mask: true
freq_mask_width_range:
- 0
- 30
lfr_rate: 6
num_freq_mask: 1
apply_time_mask: true
time_mask_width_range:
- 0
- 12
num_time_mask: 1
train_conf:
accum_grad: 1
grad_clip: 5
max_epoch: 15
keep_nbest_models: 10
log_interval: 10
optim: adamw
optim_conf:
lr: 0.0001
weight_decay: 0.000000
scheduler: warmuplr
scheduler_conf:
warmup_steps: 1500
dataset: AudioLLMQwenAudioDataset
dataset_conf:
index_ds: IndexDSJsonl
batch_sampler: CustomDistributedBatchSampler
batch_type: example # example or length
batch_size: 4 # if batch_type is example, batch_size is the numbers of samples; if length, batch_size is source_token_len+target_token_len;
max_token_length: 3000 # filter samples if source_token_len+target_token_len > max_token_length,
shuffle: True
num_workers: 4
preprocessor_text: TextPreprocessRemovePunctuation
audio_adaptor_downsample_rate: ${audio_adaptor_conf.downsample_rate}
audio_encoder_downsample_rate: 2
# prompt: "<|startoftranscription|><|zh|><|transcribe|><|zh|><|notimestamps|><|wo_itn|>"
tokenizer: HuggingfaceTokenizer
tokenizer_conf:
unk_symbol: <unk>
init_param_path: "/nfs/zhifu.gzf/init_model/qwen/Qwen1___5-7B-Chat"
examples/industrial_data_pretraining/llm_asr/conf/whisper_vicuna_linear.yaml
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# This is an example that demonstrates how to configure a model file.
# You can modify the configuration according to your own requirements.
# to print the register_table:
# from funasr.register import tables
# tables.print()
# network architecture
model: LLMASR
model_conf:
lsm_weight: 0.1 # label smoothing option
length_normalized_loss: true
# encoder
audio_encoder: "/nfs/zhifu.gzf/init_model/Whisper-large-v3" #iic/Whisper-large-v3
audio_encoder_conf:
hub: ms
freeze: true
llm: Vicuna
llm_conf:
hub: hf
init_param_path: "/nfs/maziyang.mzy/models/vicuna-7b-v1.5"
freeze: true
audio_adaptor: Linear
audio_adaptor_conf:
downsample_rate: 5
llm_dim: 4096
encoder_dim: 512
# frontend related
frontend: WhisperFrontend
frontend_conf:
fs: 16000
whisper_model: large-v3
do_pad_trim: true
permute: true # true: [bs, frames, dims]; false: [bs, dims, frames]
specaug: SpecAugLFR
specaug_conf:
apply_time_warp: false
time_warp_window: 5
time_warp_mode: bicubic
apply_freq_mask: true
freq_mask_width_range:
- 0
- 30
lfr_rate: 6
num_freq_mask: 1
apply_time_mask: true
time_mask_width_range:
- 0
- 12
num_time_mask: 1
train_conf:
accum_grad: 1
grad_clip: 5
max_epoch: 15
keep_nbest_models: 10
log_interval: 10
optim: adamw
optim_conf:
lr: 0.0001
weight_decay: 0
scheduler: warmuplr
scheduler_conf:
warmup_steps: 1500
dataset: AudioLLMVicunaDataset
dataset_conf:
index_ds: IndexDSJsonl
batch_sampler: CustomDistributedBatchSampler
batch_type: example # example or length
batch_size: 4 # if batch_type is example, batch_size is the numbers of samples; if length, batch_size is source_token_len+target_token_len;
max_token_length: 3000 # filter samples if source_token_len+target_token_len > max_token_length,
shuffle: True
num_workers: 4
# preprocessor_text: TextPreprocessRemovePunctuation
audio_adaptor_downsample_rate: ${audio_adaptor_conf.downsample_rate}
audio_encoder_downsample_rate: 2
tokenizer: HuggingfaceTokenizer
tokenizer_conf:
unk_symbol: <unk>
init_param_path: "/nfs/maziyang.mzy/models/vicuna-7b-v1.5"
examples/industrial_data_pretraining/llm_asr/demo_infer.sh
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# Copyright FunASR (https://github.com/alibaba-damo-academy/FunASR). All Rights Reserved.
# MIT License (https://opensource.org/licenses/MIT)
python -m funasr.bin.inference \
--config-path="/root/FunASR/examples/aishell/llm_asr_nar/conf" \
--config-name="template.yaml" \
++init_param="/mnt/workspace/FunASR/examples/aishell/paraformer/exp/baseline_paraformer_conformer_12e_6d_2048_256_zh_char_exp3/model.pt.ep38" \
++input="/nfs/beinian.lzr/workspace/datasets/data/16k/opendata/aishell1/dev/wav/S0724/BAC009S0724W0121.wav" \
++scope_map="encoder.model,audio_encoder,encoder_projector,adaptor" \
++output_dir="./outputs/debug" \
++device="cpu" \
examples/industrial_data_pretraining/llm_asr/demo_train_or_finetune.sh
+46
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# Copyright FunASR (https://github.com/alibaba-damo-academy/FunASR). All Rights Reserved.
# MIT License (https://opensource.org/licenses/MIT)
# which gpu to train or finetune
export CUDA_VISIBLE_DEVICES="0"
gpu_num=$(echo $CUDA_VISIBLE_DEVICES | awk -F "," '{print NF}')
# data dir, which contains: train.json, val.json, tokens.jsonl/tokens.txt, am.mvn
#data_dir="/Users/zhifu/funasr1.0/data/list"
## generate jsonl from wav.scp and text.txt
#python -m funasr.datasets.audio_datasets.scp2jsonl \
#++scp_file_list='["/Users/zhifu/funasr1.0/test_local/wav.scp", "/Users/zhifu/funasr1.0/test_local/text.txt"]' \
#++data_type_list='["source", "target"]' \
#++jsonl_file_out=/Users/zhifu/funasr1.0/test_local/audio_datasets.jsonl
train_data="/nfs/maziyang.mzy/data/librispeech/librispeech_train_960h.jsonl"
val_data="/nfs/maziyang.mzy/data/librispeech/librispeech_dev_other_filtered.jsonl"
# exp output dir
output_dir="/nfs/zhifu.gzf/ckpt/exp/llm_asr_whisper_vicuna_exp1"
log_file="${output_dir}/log.txt"
workspace=`pwd`
config="whisper_vicuna_linear.yaml"
init_param="${output_dir}/model.pt"
mkdir -p ${output_dir}
echo "log_file: ${log_file}"
torchrun \
--nnodes 1 \
--nproc_per_node ${gpu_num} \
../../../funasr/bin/train.py \
--config-path "${workspace}/conf" \
--config-name "${config}" \
++train_data_set_list="${train_data}" \
++valid_data_set_list="${val_data}" \
++dataset_conf.batch_size=4 \
++dataset_conf.num_workers=4 \
++train_conf.max_epoch=15 \
++optim_conf.lr=0.0001 \
++init_param="${init_param}" \
++output_dir="${output_dir}" &> ${log_file} &
examples/industrial_data_pretraining/monotonic_aligner/README_zh.md
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(简体中文|[English](./README.md))
# 语音识别
> **注意**:
> pipeline 支持 [modelscope模型仓库](https://alibaba-damo-academy.github.io/FunASR/en/model_zoo/modelscope_models.html#pretrained-models-on-modelscope) 中的所有模型进行推理和微调。这里我们以典型模型作为示例来演示使用方法。
## 推理
### 快速使用
#### [Paraformer 模型](https://www.modelscope.cn/models/damo/speech_paraformer-large_asr_nat-zh-cn-16k-common-vocab8404-pytorch/summary)
```python
from funasr import AutoModel
model = AutoModel(model="/Users/zhifu/Downloads/modelscope_models/speech_paraformer-large_asr_nat-zh-cn-16k-common-vocab8404-pytorch")
res = model(input="/Users/zhifu/Downloads/modelscope_models/speech_paraformer-large_asr_nat-zh-cn-16k-common-vocab8404-pytorch/example/asr_example.wav")
print(res)
```
### API接口说明
#### AutoModel 定义
- `model`: [模型仓库](https://alibaba-damo-academy.github.io/FunASR/en/model_zoo/modelscope_models.html#pretrained-models-on-modelscope) 中的模型名称，或本地磁盘中的模型路径
- `device`: `cuda`（默认），使用 GPU 进行推理。如果为`cpu`，则使用 CPU 进行推理
- `ncpu`: `None` （默认），设置用于 CPU 内部操作并行性的线程数
- `output_dir`: `None` （默认），如果设置，输出结果的输出路径
- `batch_size`: `1` （默认），解码时的批处理大小
#### AutoModel 推理
- `input`: 要解码的输入，可以是：
- wav文件路径, 例如: asr_example.wav
- pcm文件路径, 例如: asr_example.pcm，此时需要指定音频采样率fs（默认为16000）
- 音频字节数流，例如：麦克风的字节数数据
- wav.scp，kaldi 样式的 wav 列表 (`wav_id \t wav_path`), 例如:
```text
asr_example1 ./audios/asr_example1.wav
asr_example2 ./audios/asr_example2.wav
```
在这种输入 `wav.scp` 的情况下，必须设置 `output_dir` 以保存输出结果
- 音频采样点，例如：`audio, rate = soundfile.read("asr_example_zh.wav")`, 数据类型为 numpy.ndarray。支持batch输入，类型为list：
```[audio_sample1, audio_sample2, ..., audio_sampleN]```
- fbank输入，支持组batch。shape为[batch, frames, dim]，类型为torch.Tensor，例如
- `output_dir`: None （默认），如果设置，输出结果的输出路径
examples/industrial_data_pretraining/monotonic_aligner/demo.py
+18
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#!/usr/bin/env python3
# -*- encoding: utf-8 -*-
# Copyright FunASR (https://github.com/alibaba-damo-academy/FunASR). All Rights Reserved.
# MIT License (https://opensource.org/licenses/MIT)
from funasr import AutoModel
model = AutoModel(model="iic/speech_timestamp_prediction-v1-16k-offline")
res = model.generate(
input=(
"https://isv-data.oss-cn-hangzhou.aliyuncs.com/ics/MaaS/ASR/test_audio/asr_example_zh.wav",
"欢迎大家来到魔搭社区进行体验",
),
data_type=("sound", "text"),
batch_size=2,
)
print(res)
examples/industrial_data_pretraining/monotonic_aligner/demo.sh
+11
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@@ -0,0 +1,11 @@
model="iic/speech_timestamp_prediction-v1-16k-offline"
python funasr/bin/inference.py \
+model=${model} \
+input='["https://isv-data.oss-cn-hangzhou.aliyuncs.com/ics/MaaS/ASR/test_audio/asr_example_zh.wav", "欢迎大家来到魔搭社区进行体验"]' \
+data_type='["sound", "text"]' \
+output_dir="../outputs/debug" \
+device="cpu" \
+batch_size=2
examples/industrial_data_pretraining/paraformer-zh-spk/README_zh.md
+436
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(简体中文|[English](./README.md))
FunASR开源了大量在工业数据上预训练模型，您可以在 [模型许可协议](https://github.com/alibaba-damo-academy/FunASR/blob/main/MODEL_LICENSE)下自由使用、复制、修改和分享FunASR模型，下面列举代表性的模型，更多模型请参考 [模型仓库](https://github.com/alibaba-damo-academy/FunASR/tree/main/model_zoo)。
<div align="center">
<h4>
<a href="#模型推理"> 模型推理 </a>
｜<a href="#模型训练与测试"> 模型训练与测试 </a>
｜<a href="#模型导出与测试"> 模型导出与测试 </a>
</h4>
</div>
<a name="模型推理"></a>
## 模型推理
### 快速使用
命令行方式调用：
```shell
funasr ++model=paraformer-zh ++vad_model="fsmn-vad" ++punc_model="ct-punc" ++input=asr_example_zh.wav
```
python代码调用（推荐）
```python
from funasr import AutoModel
model = AutoModel(model="paraformer-zh")
res = model.generate(input="https://isv-data.oss-cn-hangzhou.aliyuncs.com/ics/MaaS/ASR/test_audio/vad_example.wav")
print(res)
```
### 接口说明
#### AutoModel 定义
```python
model = AutoModel(model=[str], device=[str], ncpu=[int], output_dir=[str], batch_size=[int], hub=[str], **kwargs)
```
- `model`(str): [模型仓库](https://github.com/alibaba-damo-academy/FunASR/tree/main/model_zoo) 中的模型名称，或本地磁盘中的模型路径
- `device`(str): `cuda:0`（默认gpu0），使用 GPU 进行推理，指定。如果为`cpu`，则使用 CPU 进行推理
- `ncpu`(int): `4` （默认），设置用于 CPU 内部操作并行性的线程数
- `output_dir`(str): `None` （默认），如果设置，输出结果的输出路径
- `batch_size`(int): `1` （默认），解码时的批处理，样本个数
- `hub`(str)：`ms`（默认），从modelscope下载模型。如果为`hf`，从huggingface下载模型。
- `**kwargs`(dict): 所有在`config.yaml`中参数，均可以直接在此处指定，例如，vad模型中最大切割长度 `max_single_segment_time=6000` （毫秒）。
#### AutoModel 推理
```python
res = model.generate(input=[str], output_dir=[str])
```
- `input`: 要解码的输入，可以是：
- wav文件路径, 例如: asr_example.wav
- pcm文件路径, 例如: asr_example.pcm，此时需要指定音频采样率fs（默认为16000）
- 音频字节数流，例如：麦克风的字节数数据
- wav.scp，kaldi 样式的 wav 列表 (`wav_id \t wav_path`), 例如:
```text
asr_example1 ./audios/asr_example1.wav
asr_example2 ./audios/asr_example2.wav
```
在这种输入 `wav.scp` 的情况下，必须设置 `output_dir` 以保存输出结果
- 音频采样点，例如：`audio, rate = soundfile.read("asr_example_zh.wav")`, 数据类型为 numpy.ndarray。支持batch输入，类型为list：
```[audio_sample1, audio_sample2, ..., audio_sampleN]```
- fbank输入，支持组batch。shape为[batch, frames, dim]，类型为torch.Tensor，例如
- `output_dir`: None （默认），如果设置，输出结果的输出路径
- `**kwargs`(dict): 与模型相关的推理参数，例如，`beam_size=10`，`decoding_ctc_weight=0.1`。
### 更多用法介绍
#### 非实时语音识别
```python
from funasr import AutoModel
# paraformer-zh is a multi-functional asr model
# use vad, punc, spk or not as you need
model = AutoModel(model="paraformer-zh",
vad_model="fsmn-vad",
vad_kwargs={"max_single_segment_time": 60000},
punc_model="ct-punc",
# spk_model="cam++"
)
wav_file = f"{model.model_path}/example/asr_example.wav"
res = model.generate(input=wav_file, batch_size_s=300, batch_size_threshold_s=60, hotword='魔搭')
print(res)
```
注意：
- 通常模型输入限制时长30s以下，组合`vad_model`后，支持任意时长音频输入，不局限于paraformer模型，所有音频输入模型均可以。
- `model`相关的参数可以直接在`AutoModel`定义中直接指定；与`vad_model`相关参数可以通过`vad_kwargs`来指定，类型为dict；类似的有`punc_kwargs`，`spk_kwargs`；
- `max_single_segment_time`: 表示`vad_model`最大切割音频时长, 单位是毫秒ms.
- `batch_size_s` 表示采用动态batch，batch中总音频时长，单位为秒s。
- `batch_size_threshold_s`: 表示`vad_model`切割后音频片段时长超过 `batch_size_threshold_s`阈值时，将batch_size数设置为1, 单位为秒s.
建议：当您输入为长音频，遇到OOM问题时，因为显存占用与音频时长呈平方关系增加，分为3种情况：
- a)推理起始阶段，显存主要取决于`batch_size_s`，适当减小该值，可以减少显存占用；
- b)推理中间阶段，遇到VAD切割的长音频片段，总token数小于`batch_size_s`，仍然出现OOM，可以适当减小`batch_size_threshold_s`，超过阈值，强制batch为1;
- c)推理快结束阶段，遇到VAD切割的长音频片段，总token数小于`batch_size_s`，且超过阈值`batch_size_threshold_s`，强制batch为1，仍然出现OOM，可以适当减小`max_single_segment_time`，使得VAD切割音频时长变短。
#### 实时语音识别
```python
from funasr import AutoModel
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")
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.generate(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)
```
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。
#### 语音端点检测（非实时）
```python
from funasr import AutoModel
model = AutoModel(model="fsmn-vad")
wav_file = f"{model.model_path}/example/vad_example.wav"
res = model.generate(input=wav_file)
print(res)
```
注：VAD模型输出格式为：`[[beg1, end1], [beg2, end2], .., [begN, endN]]`，其中`begN/endN`表示第`N`个有效音频片段的起始点/结束点，
单位为毫秒。
#### 语音端点检测（实时）
```python
from funasr import AutoModel
chunk_size = 200 # ms
model = AutoModel(model="fsmn-vad")
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.generate(input=speech_chunk, cache=cache, is_final=is_final, chunk_size=chunk_size)
if len(res[0]["value"]):
print(res)
```
注：流式VAD模型输出格式为4种情况：
- `[[beg1, end1], [beg2, end2], .., [begN, endN]]`：同上离线VAD输出结果。
- `[[beg, -1]]`：表示只检测到起始点。
- `[[-1, end]]`：表示只检测到结束点。
- `[]`：表示既没有检测到起始点，也没有检测到结束点
输出结果单位为毫秒，从起始点开始的绝对时间。
#### 标点恢复
```python
from funasr import AutoModel
model = AutoModel(model="ct-punc")
res = model.generate(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
model = AutoModel(model="fa-zh")
wav_file = f"{model.model_path}/example/asr_example.wav"
text_file = f"{model.model_path}/example/text.txt"
res = model.generate(input=(wav_file, text_file), data_type=("sound", "text"))
print(res)
```
更多（[示例](https://github.com/alibaba-damo-academy/FunASR/tree/main/examples/industrial_data_pretraining)）
<a name="核心功能"></a>
## 模型训练与测试
### 快速开始
命令行执行（用于快速测试，不推荐）：
```shell
funasr-train ++model=paraformer-zh ++train_data_set_list=data/list/train.jsonl ++valid_data_set_list=data/list/val.jsonl ++output_dir="./outputs" &> log.txt &
```
python代码执行（可以多机多卡，推荐）
```shell
cd examples/industrial_data_pretraining/paraformer
bash finetune.sh
# "log_file: ./outputs/log.txt"
```
详细完整的脚本参考 [finetune.sh](https://github.com/alibaba-damo-academy/FunASR/blob/main/examples/industrial_data_pretraining/paraformer/finetune.sh)
### 详细参数介绍
```shell
funasr/bin/train.py \
++model="${model_name_or_model_dir}" \
++train_data_set_list="${train_data}" \
++valid_data_set_list="${val_data}" \
++dataset_conf.batch_size=20000 \
++dataset_conf.batch_type="token" \
++dataset_conf.num_workers=4 \
++train_conf.max_epoch=50 \
++train_conf.log_interval=1 \
++train_conf.resume=false \
++train_conf.validate_interval=2000 \
++train_conf.save_checkpoint_interval=2000 \
++train_conf.keep_nbest_models=20 \
++train_conf.avg_nbest_model=10 \
++optim_conf.lr=0.0002 \
++output_dir="${output_dir}" &> ${log_file}
```
- `model`（str）：模型名字（模型仓库中的ID），此时脚本会自动下载模型到本读；或者本地已经下载好的模型路径。
- `train_data_set_list`（str）：训练数据路径，默认为jsonl格式，具体参考（[例子](https://github.com/alibaba-damo-academy/FunASR/blob/main/data/list)）。
- `valid_data_set_list`（str）：验证数据路径，默认为jsonl格式，具体参考（[例子](https://github.com/alibaba-damo-academy/FunASR/blob/main/data/list)）。
- `dataset_conf.batch_type`（str）：`example`（默认），batch的类型。`example`表示按照固定数目batch_size个样本组batch；`length` or `token` 表示动态组batch，batch总长度或者token数为batch_size。
- `dataset_conf.batch_size`（int）：与 `batch_type` 搭配使用，当 `batch_type=example` 时，表示样本个数；当 `batch_type=length` 时，表示样本中长度，单位为fbank帧数（1帧10ms）或者文字token个数。
- `train_conf.max_epoch`（int）：`100`（默认），训练总epoch数。
- `train_conf.log_interval`（int）：`50`（默认），打印日志间隔step数。
- `train_conf.resume`（int）：`True`（默认），是否开启断点重训。
- `train_conf.validate_interval`（int）：`5000`（默认），训练中做验证测试的间隔step数。
- `train_conf.save_checkpoint_interval`（int）：`5000`（默认），训练中模型保存间隔step数。
- `train_conf.avg_keep_nbest_models_type`（str）：`acc`（默认），保留nbest的标准为acc（越大越好）。`loss`表示，保留nbest的标准为loss（越小越好）。
- `train_conf.keep_nbest_models`（int）：`500`（默认），保留最大多少个模型参数，配合 `avg_keep_nbest_models_type` 按照验证集 acc/loss 保留最佳的n个模型，其他删除，节约存储空间。
- `train_conf.avg_nbest_model`（int）：`10`（默认），保留最大多少个模型参数，配合 `avg_keep_nbest_models_type` 按照验证集 acc/loss 对最佳的n个模型平均。
- `train_conf.accum_grad`（int）：`1`（默认），梯度累积功能。
- `train_conf.grad_clip`（float）：`10.0`（默认），梯度截断功能。
- `train_conf.use_fp16`（bool）：`False`（默认），开启fp16训练，加快训练速度。
- `optim_conf.lr`（float）：学习率。
- `output_dir`（str）：模型保存路径。
- `**kwargs`(dict): 所有在`config.yaml`中参数，均可以直接在此处指定，例如，过滤20s以上长音频：`dataset_conf.max_token_length=2000`，单位为音频fbank帧数（1帧10ms）或者文字token个数。
#### 多gpu训练
##### 单机多gpu训练
```shell
export CUDA_VISIBLE_DEVICES="0,1"
gpu_num=$(echo $CUDA_VISIBLE_DEVICES | awk -F "," '{print NF}')
torchrun --nnodes 1 --nproc_per_node ${gpu_num} \
../../../funasr/bin/train.py ${train_args}
```
--nnodes 表示参与的节点总数，--nproc_per_node 表示每个节点上运行的进程数
##### 多机多gpu训练
在主节点上，假设IP为192.168.1.1，端口为12345，使用的是2个GPU，则运行如下命令：
```shell
export CUDA_VISIBLE_DEVICES="0,1"
gpu_num=$(echo $CUDA_VISIBLE_DEVICES | awk -F "," '{print NF}')
torchrun --nnodes 2 --node_rank 0 --nproc_per_node ${gpu_num} --master_addr 192.168.1.1 --master_port 12345 \
../../../funasr/bin/train.py ${train_args}
```
在从节点上（假设IP为192.168.1.2），你需要确保MASTER_ADDR和MASTER_PORT环境变量与主节点设置的一致，并运行同样的命令：
```shell
export CUDA_VISIBLE_DEVICES="0,1"
gpu_num=$(echo $CUDA_VISIBLE_DEVICES | awk -F "," '{print NF}')
torchrun --nnodes 2 --node_rank 1 --nproc_per_node ${gpu_num} --master_addr 192.168.1.1 --master_port 12345 \
../../../funasr/bin/train.py ${train_args}
```
--nnodes 表示参与的节点总数，--node_rank 表示当前节点id，--nproc_per_node 表示每个节点上运行的进程数（通常为gpu个数）
#### 准备数据
`jsonl`格式可以参考（[例子](https://github.com/alibaba-damo-academy/FunASR/blob/main/data/list)）。
可以用指令 `scp2jsonl` 从wav.scp与text.txt生成。wav.scp与text.txt准备过程如下：
`train_text.txt`
左边为数据唯一ID，需与`train_wav.scp`中的`ID`一一对应
右边为音频文件标注文本，格式如下：
```bash
ID0012W0013 当客户风险承受能力评估依据发生变化时
ID0012W0014 所有只要处理 data 不管你是做 machine learning 做 deep learning
ID0012W0015 he tried to think how it could be
```
`train_wav.scp`
左边为数据唯一ID，需与`train_text.txt`中的`ID`一一对应
右边为音频文件的路径，格式如下
```bash
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
ID0012W0015 https://isv-data.oss-cn-hangzhou.aliyuncs.com/ics/MaaS/ASR/test_audio/asr_example_cn_en.wav
```
`生成指令`
```shell
# generate train.jsonl and val.jsonl from wav.scp and text.txt
scp2jsonl \
++scp_file_list='["../../../data/list/train_wav.scp", "../../../data/list/train_text.txt"]' \
++data_type_list='["source", "target"]' \
++jsonl_file_out="../../../data/list/train.jsonl"
```
（可选，非必需）如果需要从jsonl解析成wav.scp与text.txt，可以使用指令：
```shell
# generate wav.scp and text.txt from train.jsonl and val.jsonl
jsonl2scp \
++scp_file_list='["../../../data/list/train_wav.scp", "../../../data/list/train_text.txt"]' \
++data_type_list='["source", "target"]' \
++jsonl_file_in="../../../data/list/train.jsonl"
```
#### 查看训练日志
##### 查看实验log
```shell
tail log.txt
[2024-03-21 15:55:52,137][root][INFO] - train, rank: 3, epoch: 0/50, step: 6990/1, total step: 6990, (loss_avg_rank: 0.327), (loss_avg_epoch: 0.409), (ppl_avg_epoch: 1.506), (acc_avg_epoch: 0.795), (lr: 1.165e-04), [('loss_att', 0.259), ('acc', 0.825), ('loss_pre', 0.04), ('loss', 0.299), ('batch_size', 40)], {'data_load': '0.000', 'forward_time': '0.315', 'backward_time': '0.555', 'optim_time': '0.076', 'total_time': '0.947'}, GPU, memory: usage: 3.830 GB, peak: 18.357 GB, cache: 20.910 GB, cache_peak: 20.910 GB
[2024-03-21 15:55:52,139][root][INFO] - train, rank: 1, epoch: 0/50, step: 6990/1, total step: 6990, (loss_avg_rank: 0.334), (loss_avg_epoch: 0.409), (ppl_avg_epoch: 1.506), (acc_avg_epoch: 0.795), (lr: 1.165e-04), [('loss_att', 0.285), ('acc', 0.823), ('loss_pre', 0.046), ('loss', 0.331), ('batch_size', 36)], {'data_load': '0.000', 'forward_time': '0.334', 'backward_time': '0.536', 'optim_time': '0.077', 'total_time': '0.948'}, GPU, memory: usage: 3.943 GB, peak: 18.291 GB, cache: 19.619 GB, cache_peak: 19.619 GB
```
指标解释：
- `rank`：表示gpu id。
- `epoch`,`step`,`total step`：表示当前epoch，step，总step。
- `loss_avg_rank`：表示当前step，所有gpu平均loss。
- `loss/ppl/acc_avg_epoch`：表示当前epoch周期，截止当前step数时，总平均loss/ppl/acc。epoch结束时的最后一个step表示epoch总平均loss/ppl/acc，推荐使用acc指标。
- `lr`：当前step的学习率。
- `[('loss_att', 0.259), ('acc', 0.825), ('loss_pre', 0.04), ('loss', 0.299), ('batch_size', 40)]`：表示当前gpu id的具体数据。
- `total_time`：表示单个step总耗时。
- `GPU, memory`：分别表示，模型使用/峰值显存，模型+缓存使用/峰值显存。
##### tensorboard可视化
```bash
tensorboard --logdir /xxxx/FunASR/examples/industrial_data_pretraining/paraformer/outputs/log/tensorboard
```
浏览器中打开：http://localhost:6006/
### 训练后模型测试
#### 有configuration.json
假定，训练模型路径为：./model_dir，如果改目录下有生成configuration.json，只需要将 [上述模型推理方法](https://github.com/alibaba-damo-academy/FunASR/blob/main/examples/README_zh.md#%E6%A8%A1%E5%9E%8B%E6%8E%A8%E7%90%86) 中模型名字修改为模型路径即可
例如：
从shell推理
```shell
python -m funasr.bin.inference ++model="./model_dir" ++input=="${input}" ++output_dir="${output_dir}"
```
从python推理
```python
from funasr import AutoModel
model = AutoModel(model="./model_dir")
res = model.generate(input=wav_file)
print(res)
```
#### 无configuration.json时
如果模型路径中无configuration.json时，需要手动指定具体配置文件路径与模型路径
```shell
python -m funasr.bin.inference \
--config-path "${local_path}" \
--config-name "${config}" \
++init_param="${init_param}" \
++tokenizer_conf.token_list="${tokens}" \
++frontend_conf.cmvn_file="${cmvn_file}" \
++input="${input}" \
++output_dir="${output_dir}" \
++device="${device}"
```
参数介绍
- `config-path`：为实验中保存的 `config.yaml`，可以从实验输出目录中查找。
- `config-name`：配置文件名，一般为 `config.yaml`，支持yaml格式与json格式，例如 `config.json`
- `init_param`：需要测试的模型参数，一般为`model.pt`，可以自己选择具体的模型文件
- `tokenizer_conf.token_list`：词表文件路径，一般在 `config.yaml` 有指定，无需再手动指定，当 `config.yaml` 中路径不正确时，需要在此处手动指定。
- `frontend_conf.cmvn_file`：wav提取fbank中用到的cmvn文件，一般在 `config.yaml` 有指定，无需再手动指定，当 `config.yaml` 中路径不正确时，需要在此处手动指定。
其他参数同上，完整 [示例](https://github.com/alibaba-damo-academy/FunASR/blob/main/examples/industrial_data_pretraining/paraformer/infer_from_local.sh)
<a name="模型导出与测试"></a>
## 模型导出与测试
### 从命令行导出
```shell
funasr-export ++model=paraformer ++quantize=false
```
### 从Python导出
```python
from funasr import AutoModel
model = AutoModel(model="paraformer")
res = model.export(quantize=False)
```
### 测试ONNX
```python
# pip3 install -U funasr-onnx
from funasr_onnx import Paraformer
model_dir = "damo/speech_paraformer-large_asr_nat-zh-cn-16k-common-vocab8404-pytorch"
model = Paraformer(model_dir, batch_size=1, quantize=True)
wav_path = ['~/.cache/modelscope/hub/damo/speech_paraformer-large_asr_nat-zh-cn-16k-common-vocab8404-pytorch/example/asr_example.wav']
result = model(wav_path)
print(result)
```
更多例子请参考 [样例](https://github.com/alibaba-damo-academy/FunASR/tree/main/runtime/python/onnxruntime)
examples/industrial_data_pretraining/paraformer-zh-spk/__init__.py
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#!/usr/bin/env python3
# -*- encoding: utf-8 -*-
# Copyright FunASR (https://github.com/alibaba-damo-academy/FunASR). All Rights Reserved.
# MIT License (https://opensource.org/licenses/MIT)
from funasr import AutoModel
model = AutoModel(
model="iic/speech_seaco_paraformer_large_asr_nat-zh-cn-16k-common-vocab8404-pytorch",
vad_model="iic/speech_fsmn_vad_zh-cn-16k-common-pytorch",
punc_model="iic/punc_ct-transformer_zh-cn-common-vocab272727-pytorch",
spk_model="iic/speech_campplus_sv_zh-cn_16k-common",
)
res = model.generate(
input="https://isv-data.oss-cn-hangzhou.aliyuncs.com/ics/MaaS/ASR/test_audio/asr_example_zh.wav",
hotword="达摩院 磨搭",
)
print(res)
examples/industrial_data_pretraining/paraformer-zh-spk/demo.sh
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model="iic/speech_seaco_paraformer_large_asr_nat-zh-cn-16k-common-vocab8404-pytorch"
vad_model="iic/speech_fsmn_vad_zh-cn-16k-common-pytorch"
#punc_model="iic/punc_ct-transformer_zh-cn-common-vocab272727-pytorch"
punc_model="iic/punc_ct-transformer_cn-en-common-vocab471067-large"
spk_model="iic/speech_campplus_sv_zh-cn_16k-common"
python funasr/bin/inference.py \
++model=${model} \
++vad_model=${vad_model} \
++punc_model=${punc_model} \
++spk_model=${spk_model} \
++input="https://isv-data.oss-cn-hangzhou.aliyuncs.com/ics/MaaS/ASR/test_audio/asr_example_zh.wav" \
++output_dir="./outputs/debug" \
++device="cpu" \
++"hotword='达摩院 魔搭'"
examples/industrial_data_pretraining/paraformer/README.md
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([简体中文](./README_zh.md)|English)
FunASR has open-sourced a large number of pre-trained models on industrial data. You are free to use, copy, modify, and share FunASR models under the [Model License Agreement](https://github.com/alibaba-damo-academy/FunASR/blob/main/MODEL_LICENSE). Below, we list some representative models. For a comprehensive list, please refer to our [Model Zoo](https://github.com/alibaba-damo-academy/FunASR/tree/main/model_zoo).
<div align="center">
<h4>
<a href="#Inference"> Model Inference </a>
｜<a href="#Training"> Model Training and Testing </a>
｜<a href="#Export"> Model Export and Testing </a>
</h4>
</div>
<a name="Inference"></a>
## Model Inference
### Quick Start
For command-line invocation:
```shell
funasr ++model=paraformer-zh ++vad_model="fsmn-vad" ++punc_model="ct-punc" ++input=asr_example_zh.wav
```
For python code invocation (recommended):
```python
from funasr import AutoModel
model = AutoModel(model="paraformer-zh")
res = model.generate(input="https://isv-data.oss-cn-hangzhou.aliyuncs.com/ics/MaaS/ASR/test_audio/vad_example.wav")
print(res)
```
### API Description
#### AutoModel Definition
```python
model = AutoModel(model=[str], device=[str], ncpu=[int], output_dir=[str], batch_size=[int], hub=[str], **kwargs)
```
- `model`(str): model name in the [Model Repository](https://github.com/alibaba-damo-academy/FunASR/tree/main/model_zoo), or a model path on local disk.
- `device`(str): `cuda:0` (default gpu0) for using GPU for inference, specify `cpu` for using CPU.
- `ncpu`(int): `4` (default), sets the number of threads for CPU internal operations.
- `output_dir`(str): `None` (default), set this to specify the output path for the results.
- `batch_size`(int): `1` (default), the number of samples per batch during decoding.
- `hub`(str)：`ms` (default) to download models from ModelScope. Use `hf` to download models from Hugging Face.
- `**kwargs`(dict): Any parameters found in config.yaml can be directly specified here, for instance, the maximum segmentation length in the vad model max_single_segment_time=6000 (milliseconds).
#### AutoModel Inference
```python
res = model.generate(input=[str], output_dir=[str])
```
- `input`: The input to be decoded, which could be:
- A wav file path, e.g., asr_example.wav
- A pcm file path, e.g., asr_example.pcm, in this case, specify the audio sampling rate fs (default is 16000)
- An audio byte stream, e.g., byte data from a microphone
- A wav.scp, a Kaldi-style wav list (wav_id \t wav_path), for example:
```text
asr_example1 ./audios/asr_example1.wav
asr_example2 ./audios/asr_example2.wav
```
When using wav.scp as input, you must set output_dir to save the output results.
- Audio samples, `e.g.`: `audio, rate = soundfile.read("asr_example_zh.wav")`, data type is numpy.ndarray. Supports batch inputs, type is list：
```[audio_sample1, audio_sample2, ..., audio_sampleN]```
- fbank input, supports batch grouping. Shape is [batch, frames, dim], type is torch.Tensor.
- `output_dir`: None (default), if set, specifies the output path for the results.
- `**kwargs`(dict): Inference parameters related to the model, for example,`beam_size=10`，`decoding_ctc_weight=0.1`.
### More Usage Introduction
#### Speech Recognition (Non-streaming)
```python
from funasr import AutoModel
# paraformer-zh is a multi-functional asr model
# use vad, punc, spk or not as you need
model = AutoModel(model="paraformer-zh",
vad_model="fsmn-vad",
vad_kwargs={"max_single_segment_time": 60000},
punc_model="ct-punc",
# spk_model="cam++"
)
wav_file = f"{model.model_path}/example/asr_example.wav"
res = model.generate(input=wav_file, batch_size_s=300, batch_size_threshold_s=60, hotword='魔搭')
print(res)
```
Notes:
- Typically, the input duration for models is limited to under 30 seconds. However, when combined with `vad_model`, support for audio input of any length is enabled, not limited to the paraformer model—any audio input model can be used.
- Parameters related to model can be directly specified in the definition of AutoModel; parameters related to `vad_model` can be set through `vad_kwargs`, which is a dict; similar parameters include `punc_kwargs` and `spk_kwargs`.
- `max_single_segment_time`: Denotes the maximum audio segmentation length for `vad_model`, measured in milliseconds (ms).
- `batch_size_s` represents the use of dynamic batching, where the total audio duration within a batch is measured in seconds (s).
- `batch_size_threshold_s`: Indicates that when the duration of an audio segment post-VAD segmentation exceeds the batch_size_threshold_s threshold, the batch size is set to 1, measured in seconds (s).
Recommendations:
When you input long audio and encounter Out Of Memory (OOM) issues, since memory usage tends to increase quadratically with audio length, consider the following three scenarios:
a) At the beginning of inference, memory usage primarily depends on `batch_size_s`. Appropriately reducing this value can decrease memory usage.
b) During the middle of inference, when encountering long audio segments cut by VAD and the total token count is less than `batch_size_s`, yet still facing OOM, you can appropriately reduce `batch_size_threshold_s`. If the threshold is exceeded, the batch size is forced to 1.
c) Towards the end of inference, if long audio segments cut by VAD have a total token count less than `batch_size_s` and exceed the `threshold` batch_size_threshold_s, forcing the batch size to 1 and still facing OOM, you may reduce `max_single_segment_time` to shorten the VAD audio segment length.
#### Speech Recognition (Streaming)
```python
from funasr import AutoModel
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")
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.generate(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.
#### Voice Activity Detection (Non-Streaming)
```python
from funasr import AutoModel
model = AutoModel(model="fsmn-vad")
wav_file = f"{model.model_path}/example/vad_example.wav"
res = model.generate(input=wav_file)
print(res)
```
Note: The output format of the VAD model is: `[[beg1, end1], [beg2, end2], ..., [begN, endN]]`, where `begN/endN` indicates the starting/ending point of the `N-th` valid audio segment, measured in milliseconds.
#### Voice Activity Detection (Streaming)
```python
from funasr import AutoModel
chunk_size = 200 # ms
model = AutoModel(model="fsmn-vad")
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.generate(input=speech_chunk, cache=cache, is_final=is_final, chunk_size=chunk_size)
if len(res[0]["value"]):
print(res)
```
Note: The output format for the streaming VAD model can be one of four scenarios:
- `[[beg1, end1], [beg2, end2], .., [begN, endN]]`：The same as the offline VAD output result mentioned above.
- `[[beg, -1]]`：Indicates that only a starting point has been detected.
- `[[-1, end]]`：Indicates that only an ending point has been detected.
- `[]`：Indicates that neither a starting point nor an ending point has been detected.
The output is measured in milliseconds and represents the absolute time from the starting point.
#### Punctuation Restoration
```python
from funasr import AutoModel
model = AutoModel(model="ct-punc")
res = model.generate(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")
wav_file = f"{model.model_path}/example/asr_example.wav"
text_file = f"{model.model_path}/example/text.txt"
res = model.generate(input=(wav_file, text_file), data_type=("sound", "text"))
print(res)
```
More examples ref to [docs](https://github.com/alibaba-damo-academy/FunASR/tree/main/examples/industrial_data_pretraining)
<a name="Training"></a>
## Model Training and Testing
### Quick Start
Execute via command line (for quick testing, not recommended):
```shell
funasr-train ++model=paraformer-zh ++train_data_set_list=data/list/train.jsonl ++valid_data_set_list=data/list/val.jsonl ++output_dir="./outputs" &> log.txt &
```
Execute with Python code (supports multi-node and multi-GPU, recommended):
```shell
cd examples/industrial_data_pretraining/paraformer
bash finetune.sh
# "log_file: ./outputs/log.txt"
```
Full code ref to [finetune.sh](https://github.com/alibaba-damo-academy/FunASR/blob/main/examples/industrial_data_pretraining/paraformer/finetune.sh)
### Detailed Parameter Description:
```shell
funasr/bin/train.py \
++model="${model_name_or_model_dir}" \
++train_data_set_list="${train_data}" \
++valid_data_set_list="${val_data}" \
++dataset_conf.batch_size=20000 \
++dataset_conf.batch_type="token" \
++dataset_conf.num_workers=4 \
++train_conf.max_epoch=50 \
++train_conf.log_interval=1 \
++train_conf.resume=false \
++train_conf.validate_interval=2000 \
++train_conf.save_checkpoint_interval=2000 \
++train_conf.keep_nbest_models=20 \
++train_conf.avg_nbest_model=10 \
++optim_conf.lr=0.0002 \
++output_dir="${output_dir}" &> ${log_file}
```
- `model`（str）: The name of the model (the ID in the model repository), at which point the script will automatically download the model to local storage; alternatively, the path to a model already downloaded locally.
- `train_data_set_list`（str）: The path to the training data, typically in jsonl format, for specific details refer to [examples](https://github.com/alibaba-damo-academy/FunASR/blob/main/data/list).
- `valid_data_set_list`（str）：The path to the validation data, also generally in jsonl format, for specific details refer to examples](https://github.com/alibaba-damo-academy/FunASR/blob/main/data/list).
- `dataset_conf.batch_type`（str）：example (default), the type of batch. example means batches are formed with a fixed number of batch_size samples; length or token means dynamic batching, with total length or number of tokens of the batch equalling batch_size.
- `dataset_conf.batch_size`（int）：Used in conjunction with batch_type. When batch_type=example, it represents the number of samples; when batch_type=length, it represents the length of the samples, measured in fbank frames (1 frame = 10 ms) or the number of text tokens.
- `train_conf.max_epoch`（int）：The total number of epochs for training.
- `train_conf.log_interval`（int）：The number of steps between logging.
- `train_conf.resume`（int）：Whether to enable checkpoint resuming for training.
- `train_conf.validate_interval`（int）：The interval in steps to run validation tests during training.
- `train_conf.save_checkpoint_interval`（int）：The interval in steps for saving the model during training.
- `train_conf.keep_nbest_models`（int）：The maximum number of model parameters to retain, sorted by validation set accuracy, from highest to lowest.
- `train_conf.avg_nbest_model`（int）：Average over the top n models with the highest accuracy.
- `optim_conf.lr`（float）：The learning rate.
- `output_dir`（str）：The path for saving the model.
- `**kwargs`(dict): Any parameters in config.yaml can be specified directly here, for example, to filter out audio longer than 20s: dataset_conf.max_token_length=2000, measured in fbank frames (1 frame = 10 ms) or the number of text tokens.
#### Multi-GPU Training
##### Single-Machine Multi-GPU Training
```shell
export CUDA_VISIBLE_DEVICES="0,1"
gpu_num=$(echo $CUDA_VISIBLE_DEVICES | awk -F "," '{print NF}')
torchrun --nnodes 1 --nproc_per_node ${gpu_num} \
../../../funasr/bin/train.py ${train_args}
```
--nnodes represents the total number of participating nodes, while --nproc_per_node indicates the number of processes running on each node.
##### Multi-Machine Multi-GPU Training
On the master node, assuming the IP is 192.168.1.1 and the port is 12345, and you're using 2 GPUs, you would run the following command:
```shell
export CUDA_VISIBLE_DEVICES="0,1"
gpu_num=$(echo $CUDA_VISIBLE_DEVICES | awk -F "," '{print NF}')
torchrun --nnodes 2 --node_rank 0 --nproc_per_node ${gpu_num} --master_addr=192.168.1.1 --master_port=12345 \
../../../funasr/bin/train.py ${train_args}
```
On the worker node (assuming the IP is 192.168.1.2), you need to ensure that the MASTER_ADDR and MASTER_PORT environment variables are set to match those of the master node, and then run the same command:
```shell
export CUDA_VISIBLE_DEVICES="0,1"
gpu_num=$(echo $CUDA_VISIBLE_DEVICES | awk -F "," '{print NF}')
torchrun --nnodes 2 --node_rank 1 --nproc_per_node ${gpu_num} --master_addr=192.168.1.1 --master_port=12345 \
../../../funasr/bin/train.py ${train_args}
```
--nnodes indicates the total number of nodes participating in the training, --node_rank represents the ID of the current node, and --nproc_per_node specifies the number of processes running on each node (usually corresponds to the number of GPUs).
#### Data prepare
`jsonl` ref to（[demo](https://github.com/alibaba-damo-academy/FunASR/blob/main/data/list)）.
The instruction scp2jsonl can be used to generate from wav.scp and text.txt. The preparation process for wav.scp and text.txt is as follows:
`train_text.txt`
```bash
ID0012W0013 当客户风险承受能力评估依据发生变化时
ID0012W0014 所有只要处理 data 不管你是做 machine learning 做 deep learning
ID0012W0015 he tried to think how it could be
```
`train_wav.scp`
```bash
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
ID0012W0015 https://isv-data.oss-cn-hangzhou.aliyuncs.com/ics/MaaS/ASR/test_audio/asr_example_cn_en.wav
```
`Command`
```shell
# generate train.jsonl and val.jsonl from wav.scp and text.txt
scp2jsonl \
++scp_file_list='["../../../data/list/train_wav.scp", "../../../data/list/train_text.txt"]' \
++data_type_list='["source", "target"]' \
++jsonl_file_out="../../../data/list/train.jsonl"
```
(Optional, not required) If you need to parse from jsonl back to wav.scp and text.txt, you can use the following command:
```shell
# generate wav.scp and text.txt from train.jsonl and val.jsonl
jsonl2scp \
++scp_file_list='["../../../data/list/train_wav.scp", "../../../data/list/train_text.txt"]' \
++data_type_list='["source", "target"]' \
++jsonl_file_in="../../../data/list/train.jsonl"
```
#### Training log
##### log.txt
```shell
tail log.txt
[2024-03-21 15:55:52,137][root][INFO] - train, rank: 3, epoch: 0/50, step: 6990/1, total step: 6990, (loss_avg_rank: 0.327), (loss_avg_epoch: 0.409), (ppl_avg_epoch: 1.506), (acc_avg_epoch: 0.795), (lr: 1.165e-04), [('loss_att', 0.259), ('acc', 0.825), ('loss_pre', 0.04), ('loss', 0.299), ('batch_size', 40)], {'data_load': '0.000', 'forward_time': '0.315', 'backward_time': '0.555', 'optim_time': '0.076', 'total_time': '0.947'}, GPU, memory: usage: 3.830 GB, peak: 18.357 GB, cache: 20.910 GB, cache_peak: 20.910 GB
[2024-03-21 15:55:52,139][root][INFO] - train, rank: 1, epoch: 0/50, step: 6990/1, total step: 6990, (loss_avg_rank: 0.334), (loss_avg_epoch: 0.409), (ppl_avg_epoch: 1.506), (acc_avg_epoch: 0.795), (lr: 1.165e-04), [('loss_att', 0.285), ('acc', 0.823), ('loss_pre', 0.046), ('loss', 0.331), ('batch_size', 36)], {'data_load': '0.000', 'forward_time': '0.334', 'backward_time': '0.536', 'optim_time': '0.077', 'total_time': '0.948'}, GPU, memory: usage: 3.943 GB, peak: 18.291 GB, cache: 19.619 GB, cache_peak: 19.619 GB
```
- `rank`：gpu id。
- `epoch`,`step`,`total step`：the current epoch, step, and total steps.
- `loss_avg_rank`：the average loss across all GPUs for the current step.
- `loss/ppl/acc_avg_epoch`：the overall average loss/perplexity/accuracy for the current epoch, up to the current step count. The last step of the epoch when it ends represents the total average loss/perplexity/accuracy for that epoch; it is recommended to use the accuracy metric.
- `lr`：the learning rate for the current step.
- `[('loss_att', 0.259), ('acc', 0.825), ('loss_pre', 0.04), ('loss', 0.299), ('batch_size', 40)]`：the specific data for the current GPU ID.
- `total_time`：the total time taken for a single step.
- `GPU, memory`：the model-used/peak memory and the model+cache-used/peak memory.
##### tensorboard
```bash
tensorboard --logdir /xxxx/FunASR/examples/industrial_data_pretraining/paraformer/outputs/log/tensorboard
```
http://localhost:6006/
### 训练后模型测试
#### With `configuration.json` file
Assuming the training model path is: ./model_dir, if a configuration.json file has been generated in this directory, you only need to change the model name to the model path in the above model inference method.
For example, for shell inference:
```shell
python -m funasr.bin.inference ++model="./model_dir" ++input=="${input}" ++output_dir="${output_dir}"
```
Python inference
```python
from funasr import AutoModel
model = AutoModel(model="./model_dir")
res = model.generate(input=wav_file)
print(res)
```
#### Without `configuration.json` file
If there is no configuration.json in the model path, you need to manually specify the exact configuration file path and the model path.
```shell
python -m funasr.bin.inference \
--config-path "${local_path}" \
--config-name "${config}" \
++init_param="${init_param}" \
++tokenizer_conf.token_list="${tokens}" \
++frontend_conf.cmvn_file="${cmvn_file}" \
++input="${input}" \
++output_dir="${output_dir}" \
++device="${device}"
```
Parameter Introduction
- `config-path`：This is the path to the config.yaml saved during the experiment, which can be found in the experiment's output directory.
- `config-name`：The name of the configuration file, usually config.yaml. It supports both YAML and JSON formats, for example config.json.
- `init_param`：The model parameters that need to be tested, usually model.pt. You can choose a specific model file as needed.
- `tokenizer_conf.token_list`：The path to the vocabulary file, which is normally specified in config.yaml. There is no need to manually specify it again unless the path in config.yaml is incorrect, in which case the correct path must be manually specified here.
- `frontend_conf.cmvn_file`：The CMVN (Cepstral Mean and Variance Normalization) file used when extracting fbank features from WAV files, which is usually specified in config.yaml. There is no need to manually specify it again unless the path in config.yaml is incorrect, in which case the correct path must be manually specified here.
Other parameters are the same as mentioned above. A complete [example](https://github.com/alibaba-damo-academy/FunASR/blob/main/examples/industrial_data_pretraining/paraformer/infer_from_local.sh) can be found here.
<a name="Export"></a>
## Export ONNX
### Command-line usage
```shell
funasr-export ++model=paraformer ++quantize=false ++device=cpu
```
### Python
```python
from funasr import AutoModel
model = AutoModel(model="paraformer", device="cpu")
res = model.export(quantize=False)
```
### Test ONNX
```python
# pip3 install -U funasr-onnx
from funasr_onnx import Paraformer
model_dir = "damo/speech_paraformer-large_asr_nat-zh-cn-16k-common-vocab8404-pytorch"
model = Paraformer(model_dir, batch_size=1, quantize=True)
wav_path = ['~/.cache/modelscope/hub/damo/speech_paraformer-large_asr_nat-zh-cn-16k-common-vocab8404-pytorch/example/asr_example.wav']
result = model(wav_path)
print(result)
```
More examples ref to [demo](https://github.com/alibaba-damo-academy/FunASR/tree/main/runtime/python/onnxruntime)
examples/industrial_data_pretraining/paraformer/README_zh.md
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(简体中文|[English](./README.md))
FunASR开源了大量在工业数据上预训练模型，您可以在 [模型许可协议](https://github.com/alibaba-damo-academy/FunASR/blob/main/MODEL_LICENSE)下自由使用、复制、修改和分享FunASR模型，下面列举代表性的模型，更多模型请参考 [模型仓库](https://github.com/alibaba-damo-academy/FunASR/tree/main/model_zoo)。
<div align="center">
<h4>
<a href="#模型推理"> 模型推理 </a>
｜<a href="#模型训练与测试"> 模型训练与测试 </a>
｜<a href="#模型导出与测试"> 模型导出与测试 </a>
</h4>
</div>
<a name="模型推理"></a>
## 模型推理
### 快速使用
命令行方式调用：
```shell
funasr ++model=paraformer-zh ++vad_model="fsmn-vad" ++punc_model="ct-punc" ++input=asr_example_zh.wav
```
python代码调用（推荐）
```python
from funasr import AutoModel
model = AutoModel(model="paraformer-zh")
res = model.generate(input="https://isv-data.oss-cn-hangzhou.aliyuncs.com/ics/MaaS/ASR/test_audio/vad_example.wav")
print(res)
```
### 接口说明
#### AutoModel 定义
```python
model = AutoModel(model=[str], device=[str], ncpu=[int], output_dir=[str], batch_size=[int], hub=[str], **kwargs)
```
- `model`(str): [模型仓库](https://github.com/alibaba-damo-academy/FunASR/tree/main/model_zoo) 中的模型名称，或本地磁盘中的模型路径
- `device`(str): `cuda:0`（默认gpu0），使用 GPU 进行推理，指定。如果为`cpu`，则使用 CPU 进行推理
- `ncpu`(int): `4` （默认），设置用于 CPU 内部操作并行性的线程数
- `output_dir`(str): `None` （默认），如果设置，输出结果的输出路径
- `batch_size`(int): `1` （默认），解码时的批处理，样本个数
- `hub`(str)：`ms`（默认），从modelscope下载模型。如果为`hf`，从huggingface下载模型。
- `**kwargs`(dict): 所有在`config.yaml`中参数，均可以直接在此处指定，例如，vad模型中最大切割长度 `max_single_segment_time=6000` （毫秒）。
#### AutoModel 推理
```python
res = model.generate(input=[str], output_dir=[str])
```
- `input`: 要解码的输入，可以是：
- wav文件路径, 例如: asr_example.wav
- pcm文件路径, 例如: asr_example.pcm，此时需要指定音频采样率fs（默认为16000）
- 音频字节数流，例如：麦克风的字节数数据
- wav.scp，kaldi 样式的 wav 列表 (`wav_id \t wav_path`), 例如:
```text
asr_example1 ./audios/asr_example1.wav
asr_example2 ./audios/asr_example2.wav
```
在这种输入 `wav.scp` 的情况下，必须设置 `output_dir` 以保存输出结果
- 音频采样点，例如：`audio, rate = soundfile.read("asr_example_zh.wav")`, 数据类型为 numpy.ndarray。支持batch输入，类型为list：
```[audio_sample1, audio_sample2, ..., audio_sampleN]```
- fbank输入，支持组batch。shape为[batch, frames, dim]，类型为torch.Tensor，例如
- `output_dir`: None （默认），如果设置，输出结果的输出路径
- `**kwargs`(dict): 与模型相关的推理参数，例如，`beam_size=10`，`decoding_ctc_weight=0.1`。
### 更多用法介绍
#### 非实时语音识别
```python
from funasr import AutoModel
# paraformer-zh is a multi-functional asr model
# use vad, punc, spk or not as you need
model = AutoModel(model="paraformer-zh",
vad_model="fsmn-vad",
vad_kwargs={"max_single_segment_time": 60000},
punc_model="ct-punc",
# spk_model="cam++"
)
wav_file = f"{model.model_path}/example/asr_example.wav"
res = model.generate(input=wav_file, batch_size_s=300, batch_size_threshold_s=60, hotword='魔搭')
print(res)
```
注意：
- 通常模型输入限制时长30s以下，组合`vad_model`后，支持任意时长音频输入，不局限于paraformer模型，所有音频输入模型均可以。
- `model`相关的参数可以直接在`AutoModel`定义中直接指定；与`vad_model`相关参数可以通过`vad_kwargs`来指定，类型为dict；类似的有`punc_kwargs`，`spk_kwargs`；
- `max_single_segment_time`: 表示`vad_model`最大切割音频时长, 单位是毫秒ms.
- `batch_size_s` 表示采用动态batch，batch中总音频时长，单位为秒s。
- `batch_size_threshold_s`: 表示`vad_model`切割后音频片段时长超过 `batch_size_threshold_s`阈值时，将batch_size数设置为1, 单位为秒s.
建议：当您输入为长音频，遇到OOM问题时，因为显存占用与音频时长呈平方关系增加，分为3种情况：
- a)推理起始阶段，显存主要取决于`batch_size_s`，适当减小该值，可以减少显存占用；
- b)推理中间阶段，遇到VAD切割的长音频片段，总token数小于`batch_size_s`，仍然出现OOM，可以适当减小`batch_size_threshold_s`，超过阈值，强制batch为1;
- c)推理快结束阶段，遇到VAD切割的长音频片段，总token数小于`batch_size_s`，且超过阈值`batch_size_threshold_s`，强制batch为1，仍然出现OOM，可以适当减小`max_single_segment_time`，使得VAD切割音频时长变短。
#### 实时语音识别
```python
from funasr import AutoModel
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")
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.generate(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)
```
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。
#### 语音端点检测（非实时）
```python
from funasr import AutoModel
model = AutoModel(model="fsmn-vad")
wav_file = f"{model.model_path}/example/vad_example.wav"
res = model.generate(input=wav_file)
print(res)
```
注：VAD模型输出格式为：`[[beg1, end1], [beg2, end2], .., [begN, endN]]`，其中`begN/endN`表示第`N`个有效音频片段的起始点/结束点，
单位为毫秒。
#### 语音端点检测（实时）
```python
from funasr import AutoModel
chunk_size = 200 # ms
model = AutoModel(model="fsmn-vad")
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.generate(input=speech_chunk, cache=cache, is_final=is_final, chunk_size=chunk_size)
if len(res[0]["value"]):
print(res)
```
注：流式VAD模型输出格式为4种情况：
- `[[beg1, end1], [beg2, end2], .., [begN, endN]]`：同上离线VAD输出结果。
- `[[beg, -1]]`：表示只检测到起始点。
- `[[-1, end]]`：表示只检测到结束点。
- `[]`：表示既没有检测到起始点，也没有检测到结束点
输出结果单位为毫秒，从起始点开始的绝对时间。
#### 标点恢复
```python
from funasr import AutoModel
model = AutoModel(model="ct-punc")
res = model.generate(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
model = AutoModel(model="fa-zh")
wav_file = f"{model.model_path}/example/asr_example.wav"
text_file = f"{model.model_path}/example/text.txt"
res = model.generate(input=(wav_file, text_file), data_type=("sound", "text"))
print(res)
```
更多（[示例](https://github.com/alibaba-damo-academy/FunASR/tree/main/examples/industrial_data_pretraining)）
<a name="核心功能"></a>
## 模型训练与测试
### 快速开始
命令行执行（用于快速测试，不推荐）：
```shell
funasr-train ++model=paraformer-zh ++train_data_set_list=data/list/train.jsonl ++valid_data_set_list=data/list/val.jsonl ++output_dir="./outputs" &> log.txt &
```
python代码执行（可以多机多卡，推荐）
```shell
cd examples/industrial_data_pretraining/paraformer
bash finetune.sh
# "log_file: ./outputs/log.txt"
```
详细完整的脚本参考 [finetune.sh](https://github.com/alibaba-damo-academy/FunASR/blob/main/examples/industrial_data_pretraining/paraformer/finetune.sh)
### 详细参数介绍
```shell
funasr/bin/train.py \
++model="${model_name_or_model_dir}" \
++train_data_set_list="${train_data}" \
++valid_data_set_list="${val_data}" \
++dataset_conf.batch_size=20000 \
++dataset_conf.batch_type="token" \
++dataset_conf.num_workers=4 \
++train_conf.max_epoch=50 \
++train_conf.log_interval=1 \
++train_conf.resume=false \
++train_conf.validate_interval=2000 \
++train_conf.save_checkpoint_interval=2000 \
++train_conf.keep_nbest_models=20 \
++train_conf.avg_nbest_model=10 \
++optim_conf.lr=0.0002 \
++output_dir="${output_dir}" &> ${log_file}
```
- `model`（str）：模型名字（模型仓库中的ID），此时脚本会自动下载模型到本读；或者本地已经下载好的模型路径。
- `train_data_set_list`（str）：训练数据路径，默认为jsonl格式，具体参考（[例子](https://github.com/alibaba-damo-academy/FunASR/blob/main/data/list)）。
- `valid_data_set_list`（str）：验证数据路径，默认为jsonl格式，具体参考（[例子](https://github.com/alibaba-damo-academy/FunASR/blob/main/data/list)）。
- `dataset_conf.batch_type`（str）：`example`（默认），batch的类型。`example`表示按照固定数目batch_size个样本组batch；`length` or `token` 表示动态组batch，batch总长度或者token数为batch_size。
- `dataset_conf.batch_size`（int）：与 `batch_type` 搭配使用，当 `batch_type=example` 时，表示样本个数；当 `batch_type=length` 时，表示样本中长度，单位为fbank帧数（1帧10ms）或者文字token个数。
- `train_conf.max_epoch`（int）：`100`（默认），训练总epoch数。
- `train_conf.log_interval`（int）：`50`（默认），打印日志间隔step数。
- `train_conf.resume`（int）：`True`（默认），是否开启断点重训。
- `train_conf.validate_interval`（int）：`5000`（默认），训练中做验证测试的间隔step数。
- `train_conf.save_checkpoint_interval`（int）：`5000`（默认），训练中模型保存间隔step数。
- `train_conf.avg_keep_nbest_models_type`（str）：`acc`（默认），保留nbest的标准为acc（越大越好）。`loss`表示，保留nbest的标准为loss（越小越好）。
- `train_conf.keep_nbest_models`（int）：`500`（默认），保留最大多少个模型参数，配合 `avg_keep_nbest_models_type` 按照验证集 acc/loss 保留最佳的n个模型，其他删除，节约存储空间。
- `train_conf.avg_nbest_model`（int）：`10`（默认），保留最大多少个模型参数，配合 `avg_keep_nbest_models_type` 按照验证集 acc/loss 对最佳的n个模型平均。
- `train_conf.accum_grad`（int）：`1`（默认），梯度累积功能。
- `train_conf.grad_clip`（float）：`10.0`（默认），梯度截断功能。
- `train_conf.use_fp16`（bool）：`False`（默认），开启fp16训练，加快训练速度。
- `optim_conf.lr`（float）：学习率。
- `output_dir`（str）：模型保存路径。
- `**kwargs`(dict): 所有在`config.yaml`中参数，均可以直接在此处指定，例如，过滤20s以上长音频：`dataset_conf.max_token_length=2000`，单位为音频fbank帧数（1帧10ms）或者文字token个数。
#### 多gpu训练
##### 单机多gpu训练
```shell
export CUDA_VISIBLE_DEVICES="0,1"
gpu_num=$(echo $CUDA_VISIBLE_DEVICES | awk -F "," '{print NF}')
torchrun --nnodes 1 --nproc_per_node ${gpu_num} \
../../../funasr/bin/train.py ${train_args}
```
--nnodes 表示参与的节点总数，--nproc_per_node 表示每个节点上运行的进程数
##### 多机多gpu训练
在主节点上，假设IP为192.168.1.1，端口为12345，使用的是2个GPU，则运行如下命令：
```shell
export CUDA_VISIBLE_DEVICES="0,1"
gpu_num=$(echo $CUDA_VISIBLE_DEVICES | awk -F "," '{print NF}')
torchrun --nnodes 2 --node_rank 0 --nproc_per_node ${gpu_num} --master_addr 192.168.1.1 --master_port 12345 \
../../../funasr/bin/train.py ${train_args}
```
在从节点上（假设IP为192.168.1.2），你需要确保MASTER_ADDR和MASTER_PORT环境变量与主节点设置的一致，并运行同样的命令：
```shell
export CUDA_VISIBLE_DEVICES="0,1"
gpu_num=$(echo $CUDA_VISIBLE_DEVICES | awk -F "," '{print NF}')
torchrun --nnodes 2 --node_rank 1 --nproc_per_node ${gpu_num} --master_addr 192.168.1.1 --master_port 12345 \
../../../funasr/bin/train.py ${train_args}
```
--nnodes 表示参与的节点总数，--node_rank 表示当前节点id，--nproc_per_node 表示每个节点上运行的进程数（通常为gpu个数）
#### 准备数据
`jsonl`格式可以参考（[例子](https://github.com/alibaba-damo-academy/FunASR/blob/main/data/list)）。
可以用指令 `scp2jsonl` 从wav.scp与text.txt生成。wav.scp与text.txt准备过程如下：
`train_text.txt`
左边为数据唯一ID，需与`train_wav.scp`中的`ID`一一对应
右边为音频文件标注文本，格式如下：
```bash
ID0012W0013 当客户风险承受能力评估依据发生变化时
ID0012W0014 所有只要处理 data 不管你是做 machine learning 做 deep learning
ID0012W0015 he tried to think how it could be
```
`train_wav.scp`
左边为数据唯一ID，需与`train_text.txt`中的`ID`一一对应
右边为音频文件的路径，格式如下
```bash
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
ID0012W0015 https://isv-data.oss-cn-hangzhou.aliyuncs.com/ics/MaaS/ASR/test_audio/asr_example_cn_en.wav
```
`生成指令`
```shell
# generate train.jsonl and val.jsonl from wav.scp and text.txt
scp2jsonl \
++scp_file_list='["../../../data/list/train_wav.scp", "../../../data/list/train_text.txt"]' \
++data_type_list='["source", "target"]' \
++jsonl_file_out="../../../data/list/train.jsonl"
```
（可选，非必需）如果需要从jsonl解析成wav.scp与text.txt，可以使用指令：
```shell
# generate wav.scp and text.txt from train.jsonl and val.jsonl
jsonl2scp \
++scp_file_list='["../../../data/list/train_wav.scp", "../../../data/list/train_text.txt"]' \
++data_type_list='["source", "target"]' \
++jsonl_file_in="../../../data/list/train.jsonl"
```
#### 查看训练日志
##### 查看实验log
```shell
tail log.txt
[2024-03-21 15:55:52,137][root][INFO] - train, rank: 3, epoch: 0/50, step: 6990/1, total step: 6990, (loss_avg_rank: 0.327), (loss_avg_epoch: 0.409), (ppl_avg_epoch: 1.506), (acc_avg_epoch: 0.795), (lr: 1.165e-04), [('loss_att', 0.259), ('acc', 0.825), ('loss_pre', 0.04), ('loss', 0.299), ('batch_size', 40)], {'data_load': '0.000', 'forward_time': '0.315', 'backward_time': '0.555', 'optim_time': '0.076', 'total_time': '0.947'}, GPU, memory: usage: 3.830 GB, peak: 18.357 GB, cache: 20.910 GB, cache_peak: 20.910 GB
[2024-03-21 15:55:52,139][root][INFO] - train, rank: 1, epoch: 0/50, step: 6990/1, total step: 6990, (loss_avg_rank: 0.334), (loss_avg_epoch: 0.409), (ppl_avg_epoch: 1.506), (acc_avg_epoch: 0.795), (lr: 1.165e-04), [('loss_att', 0.285), ('acc', 0.823), ('loss_pre', 0.046), ('loss', 0.331), ('batch_size', 36)], {'data_load': '0.000', 'forward_time': '0.334', 'backward_time': '0.536', 'optim_time': '0.077', 'total_time': '0.948'}, GPU, memory: usage: 3.943 GB, peak: 18.291 GB, cache: 19.619 GB, cache_peak: 19.619 GB
```
指标解释：
- `rank`：表示gpu id。
- `epoch`,`step`,`total step`：表示当前epoch，step，总step。
- `loss_avg_rank`：表示当前step，所有gpu平均loss。
- `loss/ppl/acc_avg_epoch`：表示当前epoch周期，截止当前step数时，总平均loss/ppl/acc。epoch结束时的最后一个step表示epoch总平均loss/ppl/acc，推荐使用acc指标。
- `lr`：当前step的学习率。
- `[('loss_att', 0.259), ('acc', 0.825), ('loss_pre', 0.04), ('loss', 0.299), ('batch_size', 40)]`：表示当前gpu id的具体数据。
- `total_time`：表示单个step总耗时。
- `GPU, memory`：分别表示，模型使用/峰值显存，模型+缓存使用/峰值显存。
##### tensorboard可视化
```bash
tensorboard --logdir /xxxx/FunASR/examples/industrial_data_pretraining/paraformer/outputs/log/tensorboard
```
浏览器中打开：http://localhost:6006/
### 训练后模型测试
#### 有configuration.json
假定，训练模型路径为：./model_dir，如果改目录下有生成configuration.json，只需要将 [上述模型推理方法](https://github.com/alibaba-damo-academy/FunASR/blob/main/examples/README_zh.md#%E6%A8%A1%E5%9E%8B%E6%8E%A8%E7%90%86) 中模型名字修改为模型路径即可
例如：
从shell推理
```shell
python -m funasr.bin.inference ++model="./model_dir" ++input=="${input}" ++output_dir="${output_dir}"
```
从python推理
```python
from funasr import AutoModel
model = AutoModel(model="./model_dir")
res = model.generate(input=wav_file)
print(res)
```
#### 无configuration.json时
如果模型路径中无configuration.json时，需要手动指定具体配置文件路径与模型路径
```shell
python -m funasr.bin.inference \
--config-path "${local_path}" \
--config-name "${config}" \
++init_param="${init_param}" \
++tokenizer_conf.token_list="${tokens}" \
++frontend_conf.cmvn_file="${cmvn_file}" \
++input="${input}" \
++output_dir="${output_dir}" \
++device="${device}"
```
参数介绍
- `config-path`：为实验中保存的 `config.yaml`，可以从实验输出目录中查找。
- `config-name`：配置文件名，一般为 `config.yaml`，支持yaml格式与json格式，例如 `config.json`
- `init_param`：需要测试的模型参数，一般为`model.pt`，可以自己选择具体的模型文件
- `tokenizer_conf.token_list`：词表文件路径，一般在 `config.yaml` 有指定，无需再手动指定，当 `config.yaml` 中路径不正确时，需要在此处手动指定。
- `frontend_conf.cmvn_file`：wav提取fbank中用到的cmvn文件，一般在 `config.yaml` 有指定，无需再手动指定，当 `config.yaml` 中路径不正确时，需要在此处手动指定。
其他参数同上，完整 [示例](https://github.com/alibaba-damo-academy/FunASR/blob/main/examples/industrial_data_pretraining/paraformer/infer_from_local.sh)
<a name="模型导出与测试"></a>
## 模型导出与测试
### 从命令行导出
```shell
funasr-export ++model=paraformer ++quantize=false
```
### 从Python导出
```python
from funasr import AutoModel
model = AutoModel(model="paraformer")
res = model.export(quantize=False)
```
### 测试ONNX
```python
# pip3 install -U funasr-onnx
from funasr_onnx import Paraformer
model_dir = "damo/speech_paraformer-large_asr_nat-zh-cn-16k-common-vocab8404-pytorch"
model = Paraformer(model_dir, batch_size=1, quantize=True)
wav_path = ['~/.cache/modelscope/hub/damo/speech_paraformer-large_asr_nat-zh-cn-16k-common-vocab8404-pytorch/example/asr_example.wav']
result = model(wav_path)
print(result)
```
更多例子请参考 [样例](https://github.com/alibaba-damo-academy/FunASR/tree/main/runtime/python/onnxruntime)
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#!/usr/bin/env python3
# -*- encoding: utf-8 -*-
# Copyright FunASR (https://github.com/alibaba-damo-academy/FunASR). All Rights Reserved.
# MIT License (https://opensource.org/licenses/MIT)
from funasr import AutoModel
model = AutoModel(
model="iic/speech_paraformer-large-vad-punc_asr_nat-zh-cn-16k-common-vocab8404-pytorch",
vad_model="iic/speech_fsmn_vad_zh-cn-16k-common-pytorch",
vad_kwargs={"max_single_segment_time": 60000},
punc_model="iic/punc_ct-transformer_zh-cn-common-vocab272727-pytorch",
# spk_model="iic/speech_campplus_sv_zh-cn_16k-common",
)
res = model.generate(
input="https://isv-data.oss-cn-hangzhou.aliyuncs.com/ics/MaaS/ASR/test_audio/asr_example_zh.wav",
cache={},
)
print(res)
""" can not use currently
from funasr import AutoFrontend
frontend = AutoFrontend(model="iic/speech_paraformer-large_asr_nat-zh-cn-16k-common-vocab8404-pytorch")
fbanks = frontend(input="https://isv-data.oss-cn-hangzhou.aliyuncs.com/ics/MaaS/ASR/test_audio/asr_example_zh.wav", batch_size=2)
for batch_idx, fbank_dict in enumerate(fbanks):
res = model.generate(**fbank_dict)
print(res)
"""
examples/industrial_data_pretraining/paraformer/export.py
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#!/usr/bin/env python3
# -*- encoding: utf-8 -*-
# Copyright FunASR (https://github.com/alibaba-damo-academy/FunASR). All Rights Reserved.
# MIT License (https://opensource.org/licenses/MIT)
# method1, inference from model hub
from funasr import AutoModel
model = AutoModel(
model="iic/speech_paraformer-large_asr_nat-zh-cn-16k-common-vocab8404-pytorch",
)
res = model.export(type="onnx", quantize=False)
print(res)
# method2, inference from local path
from funasr import AutoModel
model = AutoModel(
model="/Users/zhifu/.cache/modelscope/hub/iic/speech_paraformer-large_asr_nat-zh-cn-16k-common-vocab8404-pytorch"
)
res = model.export(type="onnx", quantize=False)
print(res)
examples/industrial_data_pretraining/paraformer/export.sh
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# Copyright FunASR (https://github.com/alibaba-damo-academy/FunASR). All Rights Reserved.
# MIT License (https://opensource.org/licenses/MIT)
# method1, inference from model hub
export HYDRA_FULL_ERROR=1
model="iic/speech_paraformer-large_asr_nat-zh-cn-16k-common-vocab8404-pytorch"
python -m funasr.bin.export \
++model=${model} \
++type="onnx" \
++quantize=false
# method2, inference from local path
model="/Users/zhifu/.cache/modelscope/hub/iic/speech_paraformer-large_asr_nat-zh-cn-16k-common-vocab8404-pytorch"
python -m funasr.bin.export \
++model=${model} \
++type="onnx" \
++quantize=false
examples/industrial_data_pretraining/paraformer/finetune.sh
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# Copyright FunASR (https://github.com/alibaba-damo-academy/FunASR). All Rights Reserved.
# MIT License (https://opensource.org/licenses/MIT)
# which gpu to train or finetune
export CUDA_VISIBLE_DEVICES="0,1"
gpu_num=$(echo $CUDA_VISIBLE_DEVICES | awk -F "," '{print NF}')
# model_name from model_hub, or model_dir in local path
## option 1, download model automatically
model_name_or_model_dir="iic/speech_paraformer-large_asr_nat-zh-cn-16k-common-vocab8404-pytorch"
## option 2, download model by git
#local_path_root=${workspace}/modelscope_models
#mkdir -p ${local_path_root}/${model_name_or_model_dir}
#git clone https://www.modelscope.cn/${model_name_or_model_dir}.git ${local_path_root}/${model_name_or_model_dir}
#model_name_or_model_dir=${local_path_root}/${model_name_or_model_dir}
# data dir, which contains: train.json, val.json
data_dir="../../../data/list"
train_data="${data_dir}/train.jsonl"
val_data="${data_dir}/val.jsonl"
# generate train.jsonl and val.jsonl from wav.scp and text.txt
scp2jsonl \
++scp_file_list='["../../../data/list/train_wav.scp", "../../../data/list/train_text.txt"]' \
++data_type_list='["source", "target"]' \
++jsonl_file_out="${train_data}"
scp2jsonl \
++scp_file_list='["../../../data/list/val_wav.scp", "../../../data/list/val_text.txt"]' \
++data_type_list='["source", "target"]' \
++jsonl_file_out="${val_data}"
# exp output dir
output_dir="./outputs"
log_file="${output_dir}/log.txt"
mkdir -p ${output_dir}
echo "log_file: ${log_file}"
torchrun \
--nnodes 1 \
--node_rank 0 \
--nproc_per_node ${gpu_num} \
../../../funasr/bin/train.py \
++model="${model_name_or_model_dir}" \
++train_data_set_list="${train_data}" \
++valid_data_set_list="${val_data}" \
++dataset_conf.batch_size=20000 \
++dataset_conf.batch_type="token" \
++dataset_conf.num_workers=4 \
++train_conf.max_epoch=50 \
++train_conf.log_interval=1 \
++train_conf.resume=false \
++train_conf.validate_interval=2000 \
++train_conf.save_checkpoint_interval=2000 \
++train_conf.keep_nbest_models=20 \
++train_conf.avg_nbest_model=10 \
++optim_conf.lr=0.0002 \
++output_dir="${output_dir}" &> ${log_file}
examples/industrial_data_pretraining/paraformer/infer.sh
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# Copyright FunASR (https://github.com/alibaba-damo-academy/FunASR). All Rights Reserved.
# MIT License (https://opensource.org/licenses/MIT)
# method1, inference from model hub
# for more input type, please ref to readme.md
input="https://isv-data.oss-cn-hangzhou.aliyuncs.com/ics/MaaS/ASR/test_audio/asr_example_zh.wav"
output_dir="./outputs/debug"
model="iic/speech_paraformer-large_asr_nat-zh-cn-16k-common-vocab8404-pytorch"
device="cuda:0" # "cuda:0" for gpu0, "cuda:1" for gpu1, "cpu"
python -m funasr.bin.inference \
++model=${model} \
++input="${input}" \
++output_dir="${output_dir}" \
++device="${device}" \
examples/industrial_data_pretraining/paraformer/infer_from_local.sh
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# Copyright FunASR (https://github.com/alibaba-damo-academy/FunASR). All Rights Reserved.
# MIT License (https://opensource.org/licenses/MIT)
# method2, inference from local model
# for more input type, please ref to readme.md
input="https://isv-data.oss-cn-hangzhou.aliyuncs.com/ics/MaaS/ASR/test_audio/asr_example_zh.wav"
output_dir="./outputs/debug"
workspace=`pwd`
# download model
local_path_root=${workspace}/modelscope_models
mkdir -p ${local_path_root}
local_path=${local_path_root}/speech_paraformer-large_asr_nat-zh-cn-16k-common-vocab8404-pytorch
git clone https://www.modelscope.cn/damo/speech_paraformer-large_asr_nat-zh-cn-16k-common-vocab8404-pytorch.git ${local_path}
device="cuda:0" # "cuda:0" for gpu0, "cuda:1" for gpu1, "cpu"
tokens="${local_path}/tokens.json"
cmvn_file="${local_path}/am.mvn"
config="config.yaml"
init_param="${local_path}/model.pt"
python -m funasr.bin.inference \
--config-path "${local_path}" \
--config-name "${config}" \
++init_param="${init_param}" \
++tokenizer_conf.token_list="${tokens}" \
++frontend_conf.cmvn_file="${cmvn_file}" \
++input="${input}" \
++output_dir="${output_dir}" \
++device="${device}" \
examples/industrial_data_pretraining/paraformer/train_from_local.sh
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# Copyright FunASR (https://github.com/alibaba-damo-academy/FunASR). All Rights Reserved.
# MIT License (https://opensource.org/licenses/MIT)
# method2, finetune from local model
workspace=`pwd`
echo "current path: ${workspace}" # /xxxx/funasr/examples/industrial_data_pretraining/paraformer
# download model
local_path_root=${workspace}/modelscope_models
mkdir -p ${local_path_root}
local_path=${local_path_root}/speech_paraformer-large_asr_nat-zh-cn-16k-common-vocab8404-pytorch
git clone https://www.modelscope.cn/damo/speech_paraformer-large_asr_nat-zh-cn-16k-common-vocab8404-pytorch.git ${local_path}
# which gpu to train or finetune
export CUDA_VISIBLE_DEVICES="0,1"
gpu_num=$(echo $CUDA_VISIBLE_DEVICES | awk -F "," '{print NF}')
# data dir, which contains: train.json, val.json
data_dir="../../../data/list"
train_data="${data_dir}/train.jsonl"
val_data="${data_dir}/val.jsonl"
# generate train.jsonl and val.jsonl from wav.scp and text.txt
python -m funasr.datasets.audio_datasets.scp2jsonl \
++scp_file_list='["../../../data/list/train_wav.scp", "../../../data/list/train_text.txt"]' \
++data_type_list='["source", "target"]' \
++jsonl_file_out="${train_data}"
python -m funasr.datasets.audio_datasets.scp2jsonl \
++scp_file_list='["../../../data/list/val_wav.scp", "../../../data/list/val_text.txt"]' \
++data_type_list='["source", "target"]' \
++jsonl_file_out="${val_data}"
tokens="${local_path}/tokens.json"
cmvn_file="${local_path}/am.mvn"
# output dir
output_dir="./outputs"
log_file="${output_dir}/log.txt"
config_name="config.yaml"
init_param="${local_path}/model.pt"
mkdir -p ${output_dir}
echo "log_file: ${log_file}"
torchrun \
--nnodes 1 \
--nproc_per_node ${gpu_num} \
../../../funasr/bin/train.py \
--config-path "${local_path}" \
--config-name "${config_name}" \
++train_data_set_list="${train_data}" \
++valid_data_set_list="${val_data}" \
++dataset_conf.batch_size=20000 \
++dataset_conf.batch_type="token" \
++dataset_conf.num_workers=4 \
++train_conf.max_epoch=50 \
++train_conf.log_interval=10 \
++train_conf.resume=false \
++train_conf.validate_interval=15 \
++train_conf.save_checkpoint_interval=15 \
++train_conf.keep_nbest_models=50 \
++optim_conf.lr=0.0002 \
++init_param="${init_param}" \
++tokenizer_conf.token_list="${tokens}" \
++frontend_conf.cmvn_file="${cmvn_file}" \
++output_dir="${output_dir}" &> ${log_file}
examples/industrial_data_pretraining/paraformer_streaming/README_zh.md
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(简体中文|[English](./README.md))
FunASR开源了大量在工业数据上预训练模型，您可以在 [模型许可协议](https://github.com/alibaba-damo-academy/FunASR/blob/main/MODEL_LICENSE)下自由使用、复制、修改和分享FunASR模型，下面列举代表性的模型，更多模型请参考 [模型仓库](https://github.com/alibaba-damo-academy/FunASR/tree/main/model_zoo)。
<div align="center">
<h4>
<a href="#模型推理"> 模型推理 </a>
｜<a href="#模型训练与测试"> 模型训练与测试 </a>
｜<a href="#模型导出与测试"> 模型导出与测试 </a>
</h4>
</div>
<a name="模型推理"></a>
## 模型推理
### 快速使用
命令行方式调用：
```shell
funasr ++model=paraformer-zh ++vad_model="fsmn-vad" ++punc_model="ct-punc" ++input=asr_example_zh.wav
```
python代码调用（推荐）
```python
from funasr import AutoModel
model = AutoModel(model="paraformer-zh")
res = model.generate(input="https://isv-data.oss-cn-hangzhou.aliyuncs.com/ics/MaaS/ASR/test_audio/vad_example.wav")
print(res)
```
### 接口说明
#### AutoModel 定义
```python
model = AutoModel(model=[str], device=[str], ncpu=[int], output_dir=[str], batch_size=[int], hub=[str], **kwargs)
```
- `model`(str): [模型仓库](https://github.com/alibaba-damo-academy/FunASR/tree/main/model_zoo) 中的模型名称，或本地磁盘中的模型路径
- `device`(str): `cuda:0`（默认gpu0），使用 GPU 进行推理，指定。如果为`cpu`，则使用 CPU 进行推理
- `ncpu`(int): `4` （默认），设置用于 CPU 内部操作并行性的线程数
- `output_dir`(str): `None` （默认），如果设置，输出结果的输出路径
- `batch_size`(int): `1` （默认），解码时的批处理，样本个数
- `hub`(str)：`ms`（默认），从modelscope下载模型。如果为`hf`，从huggingface下载模型。
- `**kwargs`(dict): 所有在`config.yaml`中参数，均可以直接在此处指定，例如，vad模型中最大切割长度 `max_single_segment_time=6000` （毫秒）。
#### AutoModel 推理
```python
res = model.generate(input=[str], output_dir=[str])
```
- `input`: 要解码的输入，可以是：
- wav文件路径, 例如: asr_example.wav
- pcm文件路径, 例如: asr_example.pcm，此时需要指定音频采样率fs（默认为16000）
- 音频字节数流，例如：麦克风的字节数数据
- wav.scp，kaldi 样式的 wav 列表 (`wav_id \t wav_path`), 例如:
```text
asr_example1 ./audios/asr_example1.wav
asr_example2 ./audios/asr_example2.wav
```
在这种输入 `wav.scp` 的情况下，必须设置 `output_dir` 以保存输出结果
- 音频采样点，例如：`audio, rate = soundfile.read("asr_example_zh.wav")`, 数据类型为 numpy.ndarray。支持batch输入，类型为list：
```[audio_sample1, audio_sample2, ..., audio_sampleN]```
- fbank输入，支持组batch。shape为[batch, frames, dim]，类型为torch.Tensor，例如
- `output_dir`: None （默认），如果设置，输出结果的输出路径
- `**kwargs`(dict): 与模型相关的推理参数，例如，`beam_size=10`，`decoding_ctc_weight=0.1`。
### 更多用法介绍
#### 非实时语音识别
```python
from funasr import AutoModel
# paraformer-zh is a multi-functional asr model
# use vad, punc, spk or not as you need
model = AutoModel(model="paraformer-zh",
vad_model="fsmn-vad",
vad_kwargs={"max_single_segment_time": 60000},
punc_model="ct-punc",
# spk_model="cam++"
)
wav_file = f"{model.model_path}/example/asr_example.wav"
res = model.generate(input=wav_file, batch_size_s=300, batch_size_threshold_s=60, hotword='魔搭')
print(res)
```
注意：
- 通常模型输入限制时长30s以下，组合`vad_model`后，支持任意时长音频输入，不局限于paraformer模型，所有音频输入模型均可以。
- `model`相关的参数可以直接在`AutoModel`定义中直接指定；与`vad_model`相关参数可以通过`vad_kwargs`来指定，类型为dict；类似的有`punc_kwargs`，`spk_kwargs`；
- `max_single_segment_time`: 表示`vad_model`最大切割音频时长, 单位是毫秒ms.
- `batch_size_s` 表示采用动态batch，batch中总音频时长，单位为秒s。
- `batch_size_threshold_s`: 表示`vad_model`切割后音频片段时长超过 `batch_size_threshold_s`阈值时，将batch_size数设置为1, 单位为秒s.
建议：当您输入为长音频，遇到OOM问题时，因为显存占用与音频时长呈平方关系增加，分为3种情况：
- a)推理起始阶段，显存主要取决于`batch_size_s`，适当减小该值，可以减少显存占用；
- b)推理中间阶段，遇到VAD切割的长音频片段，总token数小于`batch_size_s`，仍然出现OOM，可以适当减小`batch_size_threshold_s`，超过阈值，强制batch为1;
- c)推理快结束阶段，遇到VAD切割的长音频片段，总token数小于`batch_size_s`，且超过阈值`batch_size_threshold_s`，强制batch为1，仍然出现OOM，可以适当减小`max_single_segment_time`，使得VAD切割音频时长变短。
#### 实时语音识别
```python
from funasr import AutoModel
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")
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.generate(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)
```
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。
#### 语音端点检测（非实时）
```python
from funasr import AutoModel
model = AutoModel(model="fsmn-vad")
wav_file = f"{model.model_path}/example/vad_example.wav"
res = model.generate(input=wav_file)
print(res)
```
注：VAD模型输出格式为：`[[beg1, end1], [beg2, end2], .., [begN, endN]]`，其中`begN/endN`表示第`N`个有效音频片段的起始点/结束点，
单位为毫秒。
#### 语音端点检测（实时）
```python
from funasr import AutoModel
chunk_size = 200 # ms
model = AutoModel(model="fsmn-vad")
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.generate(input=speech_chunk, cache=cache, is_final=is_final, chunk_size=chunk_size)
if len(res[0]["value"]):
print(res)
```
注：流式VAD模型输出格式为4种情况：
- `[[beg1, end1], [beg2, end2], .., [begN, endN]]`：同上离线VAD输出结果。
- `[[beg, -1]]`：表示只检测到起始点。
- `[[-1, end]]`：表示只检测到结束点。
- `[]`：表示既没有检测到起始点，也没有检测到结束点
输出结果单位为毫秒，从起始点开始的绝对时间。
#### 标点恢复
```python
from funasr import AutoModel
model = AutoModel(model="ct-punc")
res = model.generate(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
model = AutoModel(model="fa-zh")
wav_file = f"{model.model_path}/example/asr_example.wav"
text_file = f"{model.model_path}/example/text.txt"
res = model.generate(input=(wav_file, text_file), data_type=("sound", "text"))
print(res)
```
更多（[示例](https://github.com/alibaba-damo-academy/FunASR/tree/main/examples/industrial_data_pretraining)）
<a name="核心功能"></a>
## 模型训练与测试
### 快速开始
命令行执行（用于快速测试，不推荐）：
```shell
funasr-train ++model=paraformer-zh ++train_data_set_list=data/list/train.jsonl ++valid_data_set_list=data/list/val.jsonl ++output_dir="./outputs" &> log.txt &
```
python代码执行（可以多机多卡，推荐）
```shell
cd examples/industrial_data_pretraining/paraformer
bash finetune.sh
# "log_file: ./outputs/log.txt"
```
详细完整的脚本参考 [finetune.sh](https://github.com/alibaba-damo-academy/FunASR/blob/main/examples/industrial_data_pretraining/paraformer/finetune.sh)
### 详细参数介绍
```shell
funasr/bin/train.py \
++model="${model_name_or_model_dir}" \
++train_data_set_list="${train_data}" \
++valid_data_set_list="${val_data}" \
++dataset_conf.batch_size=20000 \
++dataset_conf.batch_type="token" \
++dataset_conf.num_workers=4 \
++train_conf.max_epoch=50 \
++train_conf.log_interval=1 \
++train_conf.resume=false \
++train_conf.validate_interval=2000 \
++train_conf.save_checkpoint_interval=2000 \
++train_conf.keep_nbest_models=20 \
++train_conf.avg_nbest_model=10 \
++optim_conf.lr=0.0002 \
++output_dir="${output_dir}" &> ${log_file}
```
- `model`（str）：模型名字（模型仓库中的ID），此时脚本会自动下载模型到本读；或者本地已经下载好的模型路径。
- `train_data_set_list`（str）：训练数据路径，默认为jsonl格式，具体参考（[例子](https://github.com/alibaba-damo-academy/FunASR/blob/main/data/list)）。
- `valid_data_set_list`（str）：验证数据路径，默认为jsonl格式，具体参考（[例子](https://github.com/alibaba-damo-academy/FunASR/blob/main/data/list)）。
- `dataset_conf.batch_type`（str）：`example`（默认），batch的类型。`example`表示按照固定数目batch_size个样本组batch；`length` or `token` 表示动态组batch，batch总长度或者token数为batch_size。
- `dataset_conf.batch_size`（int）：与 `batch_type` 搭配使用，当 `batch_type=example` 时，表示样本个数；当 `batch_type=length` 时，表示样本中长度，单位为fbank帧数（1帧10ms）或者文字token个数。
- `train_conf.max_epoch`（int）：`100`（默认），训练总epoch数。
- `train_conf.log_interval`（int）：`50`（默认），打印日志间隔step数。
- `train_conf.resume`（int）：`True`（默认），是否开启断点重训。
- `train_conf.validate_interval`（int）：`5000`（默认），训练中做验证测试的间隔step数。
- `train_conf.save_checkpoint_interval`（int）：`5000`（默认），训练中模型保存间隔step数。
- `train_conf.avg_keep_nbest_models_type`（str）：`acc`（默认），保留nbest的标准为acc（越大越好）。`loss`表示，保留nbest的标准为loss（越小越好）。
- `train_conf.keep_nbest_models`（int）：`500`（默认），保留最大多少个模型参数，配合 `avg_keep_nbest_models_type` 按照验证集 acc/loss 保留最佳的n个模型，其他删除，节约存储空间。
- `train_conf.avg_nbest_model`（int）：`10`（默认），保留最大多少个模型参数，配合 `avg_keep_nbest_models_type` 按照验证集 acc/loss 对最佳的n个模型平均。
- `train_conf.accum_grad`（int）：`1`（默认），梯度累积功能。
- `train_conf.grad_clip`（float）：`10.0`（默认），梯度截断功能。
- `train_conf.use_fp16`（bool）：`False`（默认），开启fp16训练，加快训练速度。
- `optim_conf.lr`（float）：学习率。
- `output_dir`（str）：模型保存路径。
- `**kwargs`(dict): 所有在`config.yaml`中参数，均可以直接在此处指定，例如，过滤20s以上长音频：`dataset_conf.max_token_length=2000`，单位为音频fbank帧数（1帧10ms）或者文字token个数。
#### 多gpu训练
##### 单机多gpu训练
```shell
export CUDA_VISIBLE_DEVICES="0,1"
gpu_num=$(echo $CUDA_VISIBLE_DEVICES | awk -F "," '{print NF}')
torchrun --nnodes 1 --nproc_per_node ${gpu_num} \
../../../funasr/bin/train.py ${train_args}
```
--nnodes 表示参与的节点总数，--nproc_per_node 表示每个节点上运行的进程数
##### 多机多gpu训练
在主节点上，假设IP为192.168.1.1，端口为12345，使用的是2个GPU，则运行如下命令：
```shell
export CUDA_VISIBLE_DEVICES="0,1"
gpu_num=$(echo $CUDA_VISIBLE_DEVICES | awk -F "," '{print NF}')
torchrun --nnodes 2 --node_rank 0 --nproc_per_node ${gpu_num} --master_addr 192.168.1.1 --master_port 12345 \
../../../funasr/bin/train.py ${train_args}
```
在从节点上（假设IP为192.168.1.2），你需要确保MASTER_ADDR和MASTER_PORT环境变量与主节点设置的一致，并运行同样的命令：
```shell
export CUDA_VISIBLE_DEVICES="0,1"
gpu_num=$(echo $CUDA_VISIBLE_DEVICES | awk -F "," '{print NF}')
torchrun --nnodes 2 --node_rank 1 --nproc_per_node ${gpu_num} --master_addr 192.168.1.1 --master_port 12345 \
../../../funasr/bin/train.py ${train_args}
```
--nnodes 表示参与的节点总数，--node_rank 表示当前节点id，--nproc_per_node 表示每个节点上运行的进程数（通常为gpu个数）
#### 准备数据
`jsonl`格式可以参考（[例子](https://github.com/alibaba-damo-academy/FunASR/blob/main/data/list)）。
可以用指令 `scp2jsonl` 从wav.scp与text.txt生成。wav.scp与text.txt准备过程如下：
`train_text.txt`
左边为数据唯一ID，需与`train_wav.scp`中的`ID`一一对应
右边为音频文件标注文本，格式如下：
```bash
ID0012W0013 当客户风险承受能力评估依据发生变化时
ID0012W0014 所有只要处理 data 不管你是做 machine learning 做 deep learning
ID0012W0015 he tried to think how it could be
```
`train_wav.scp`
左边为数据唯一ID，需与`train_text.txt`中的`ID`一一对应
右边为音频文件的路径，格式如下
```bash
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
ID0012W0015 https://isv-data.oss-cn-hangzhou.aliyuncs.com/ics/MaaS/ASR/test_audio/asr_example_cn_en.wav
```
`生成指令`
```shell
# generate train.jsonl and val.jsonl from wav.scp and text.txt
scp2jsonl \
++scp_file_list='["../../../data/list/train_wav.scp", "../../../data/list/train_text.txt"]' \
++data_type_list='["source", "target"]' \
++jsonl_file_out="../../../data/list/train.jsonl"
```
（可选，非必需）如果需要从jsonl解析成wav.scp与text.txt，可以使用指令：
```shell
# generate wav.scp and text.txt from train.jsonl and val.jsonl
jsonl2scp \
++scp_file_list='["../../../data/list/train_wav.scp", "../../../data/list/train_text.txt"]' \
++data_type_list='["source", "target"]' \
++jsonl_file_in="../../../data/list/train.jsonl"
```
#### 查看训练日志
##### 查看实验log
```shell
tail log.txt
[2024-03-21 15:55:52,137][root][INFO] - train, rank: 3, epoch: 0/50, step: 6990/1, total step: 6990, (loss_avg_rank: 0.327), (loss_avg_epoch: 0.409), (ppl_avg_epoch: 1.506), (acc_avg_epoch: 0.795), (lr: 1.165e-04), [('loss_att', 0.259), ('acc', 0.825), ('loss_pre', 0.04), ('loss', 0.299), ('batch_size', 40)], {'data_load': '0.000', 'forward_time': '0.315', 'backward_time': '0.555', 'optim_time': '0.076', 'total_time': '0.947'}, GPU, memory: usage: 3.830 GB, peak: 18.357 GB, cache: 20.910 GB, cache_peak: 20.910 GB
[2024-03-21 15:55:52,139][root][INFO] - train, rank: 1, epoch: 0/50, step: 6990/1, total step: 6990, (loss_avg_rank: 0.334), (loss_avg_epoch: 0.409), (ppl_avg_epoch: 1.506), (acc_avg_epoch: 0.795), (lr: 1.165e-04), [('loss_att', 0.285), ('acc', 0.823), ('loss_pre', 0.046), ('loss', 0.331), ('batch_size', 36)], {'data_load': '0.000', 'forward_time': '0.334', 'backward_time': '0.536', 'optim_time': '0.077', 'total_time': '0.948'}, GPU, memory: usage: 3.943 GB, peak: 18.291 GB, cache: 19.619 GB, cache_peak: 19.619 GB
```
指标解释：
- `rank`：表示gpu id。
- `epoch`,`step`,`total step`：表示当前epoch，step，总step。
- `loss_avg_rank`：表示当前step，所有gpu平均loss。
- `loss/ppl/acc_avg_epoch`：表示当前epoch周期，截止当前step数时，总平均loss/ppl/acc。epoch结束时的最后一个step表示epoch总平均loss/ppl/acc，推荐使用acc指标。
- `lr`：当前step的学习率。
- `[('loss_att', 0.259), ('acc', 0.825), ('loss_pre', 0.04), ('loss', 0.299), ('batch_size', 40)]`：表示当前gpu id的具体数据。
- `total_time`：表示单个step总耗时。
- `GPU, memory`：分别表示，模型使用/峰值显存，模型+缓存使用/峰值显存。
##### tensorboard可视化
```bash
tensorboard --logdir /xxxx/FunASR/examples/industrial_data_pretraining/paraformer/outputs/log/tensorboard
```
浏览器中打开：http://localhost:6006/
### 训练后模型测试
#### 有configuration.json
假定，训练模型路径为：./model_dir，如果改目录下有生成configuration.json，只需要将 [上述模型推理方法](https://github.com/alibaba-damo-academy/FunASR/blob/main/examples/README_zh.md#%E6%A8%A1%E5%9E%8B%E6%8E%A8%E7%90%86) 中模型名字修改为模型路径即可
例如：
从shell推理
```shell
python -m funasr.bin.inference ++model="./model_dir" ++input=="${input}" ++output_dir="${output_dir}"
```
从python推理
```python
from funasr import AutoModel
model = AutoModel(model="./model_dir")
res = model.generate(input=wav_file)
print(res)
```
#### 无configuration.json时
如果模型路径中无configuration.json时，需要手动指定具体配置文件路径与模型路径
```shell
python -m funasr.bin.inference \
--config-path "${local_path}" \
--config-name "${config}" \
++init_param="${init_param}" \
++tokenizer_conf.token_list="${tokens}" \
++frontend_conf.cmvn_file="${cmvn_file}" \
++input="${input}" \
++output_dir="${output_dir}" \
++device="${device}"
```
参数介绍
- `config-path`：为实验中保存的 `config.yaml`，可以从实验输出目录中查找。
- `config-name`：配置文件名，一般为 `config.yaml`，支持yaml格式与json格式，例如 `config.json`
- `init_param`：需要测试的模型参数，一般为`model.pt`，可以自己选择具体的模型文件
- `tokenizer_conf.token_list`：词表文件路径，一般在 `config.yaml` 有指定，无需再手动指定，当 `config.yaml` 中路径不正确时，需要在此处手动指定。
- `frontend_conf.cmvn_file`：wav提取fbank中用到的cmvn文件，一般在 `config.yaml` 有指定，无需再手动指定，当 `config.yaml` 中路径不正确时，需要在此处手动指定。
其他参数同上，完整 [示例](https://github.com/alibaba-damo-academy/FunASR/blob/main/examples/industrial_data_pretraining/paraformer/infer_from_local.sh)
<a name="模型导出与测试"></a>
## 模型导出与测试
### 从命令行导出
```shell
funasr-export ++model=paraformer ++quantize=false
```
### 从Python导出
```python
from funasr import AutoModel
model = AutoModel(model="paraformer")
res = model.export(quantize=False)
```
### 测试ONNX
```python
# pip3 install -U funasr-onnx
from funasr_onnx import Paraformer
model_dir = "damo/speech_paraformer-large_asr_nat-zh-cn-16k-common-vocab8404-pytorch"
model = Paraformer(model_dir, batch_size=1, quantize=True)
wav_path = ['~/.cache/modelscope/hub/damo/speech_paraformer-large_asr_nat-zh-cn-16k-common-vocab8404-pytorch/example/asr_example.wav']
result = model(wav_path)
print(result)
```
更多例子请参考 [样例](https://github.com/alibaba-damo-academy/FunASR/tree/main/runtime/python/onnxruntime)
examples/industrial_data_pretraining/paraformer_streaming/demo.py
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#!/usr/bin/env python3
# -*- encoding: utf-8 -*-
# Copyright FunASR (https://github.com/alibaba-damo-academy/FunASR). All Rights Reserved.
# MIT License (https://opensource.org/licenses/MIT)
import os
from funasr import AutoModel
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="iic/speech_paraformer_asr_nat-zh-cn-16k-common-vocab8404-online")
wav_file = os.path.join(model.model_path, "example/asr_example.wav")
res = model.generate(
input=wav_file,
chunk_size=chunk_size,
encoder_chunk_look_back=encoder_chunk_look_back,
decoder_chunk_look_back=decoder_chunk_look_back,
)
print(res)
import soundfile
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、480ms
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.generate(
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)
examples/industrial_data_pretraining/paraformer_streaming/demo.sh
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model="iic/speech_paraformer-large_asr_nat-zh-cn-16k-common-vocab8404-online"
python funasr/bin/inference.py \
+model=${model} \
+input="https://isv-data.oss-cn-hangzhou.aliyuncs.com/ics/MaaS/ASR/test_audio/asr_example_zh.wav" \
+output_dir="./outputs/debug" \
+device="cpu" \
examples/industrial_data_pretraining/paraformer_streaming/export.py
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@@ -0,0 +1,29 @@
#!/usr/bin/env python3
# -*- encoding: utf-8 -*-
# Copyright FunASR (https://github.com/alibaba-damo-academy/FunASR). All Rights Reserved.
# MIT License (https://opensource.org/licenses/MIT)
# method1, inference from model hub
from funasr import AutoModel
model = AutoModel(
model="iic/speech_paraformer_asr_nat-zh-cn-16k-common-vocab8404-online",
)
res = model.export(type="onnx", quantize=False)
print(res)
# method2, inference from local path
from funasr import AutoModel
model = AutoModel(
model="/Users/zhifu/.cache/modelscope/hub/iic/speech_paraformer_asr_nat-zh-cn-16k-common-vocab8404-online"
)
res = model.export(type="onnx", quantize=False)
print(res)
examples/industrial_data_pretraining/paraformer_streaming/export.sh
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# Copyright FunASR (https://github.com/alibaba-damo-academy/FunASR). All Rights Reserved.
# MIT License (https://opensource.org/licenses/MIT)
# method1, inference from model hub
export HYDRA_FULL_ERROR=1
model="iic/speech_paraformer_asr_nat-zh-cn-16k-common-vocab8404-online"
python -m funasr.bin.export \
++model=${model} \
++type="onnx" \
++quantize=false \
++device="cpu"
# method2, inference from local path
model="/Users/zhifu/.cache/modelscope/hub/iic/speech_paraformer_asr_nat-zh-cn-16k-common-vocab8404-online"
python -m funasr.bin.export \
++model=${model} \
++type="onnx" \
++quantize=false \
++device="cpu" \
++debug=false
examples/industrial_data_pretraining/paraformer_streaming/finetune.sh
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# Copyright FunASR (https://github.com/alibaba-damo-academy/FunASR). All Rights Reserved.
# MIT License (https://opensource.org/licenses/MIT)
# which gpu to train or finetune
export CUDA_VISIBLE_DEVICES="0,1"
gpu_num=$(echo $CUDA_VISIBLE_DEVICES | awk -F "," '{print NF}')
# model_name from model_hub, or model_dir in local path
## option 1, download model automatically
model_name_or_model_dir="iic/speech_paraformer_asr_nat-zh-cn-16k-common-vocab8404-online"
## option 2, download model by git
#local_path_root=${workspace}/modelscope_models
#mkdir -p ${local_path_root}/${model_name_or_model_dir}
#git clone https://www.modelscope.cn/${model_name_or_model_dir}.git ${local_path_root}/${model_name_or_model_dir}
#model_name_or_model_dir=${local_path_root}/${model_name_or_model_dir}
# data dir, which contains: train.json, val.json
data_dir="../../../data/list"
train_data="${data_dir}/train.jsonl"
val_data="${data_dir}/val.jsonl"
# generate train.jsonl and val.jsonl from wav.scp and text.txt
scp2jsonl \
++scp_file_list='["../../../data/list/train_wav.scp", "../../../data/list/train_text.txt"]' \
++data_type_list='["source", "target"]' \
++jsonl_file_out="${train_data}"
scp2jsonl \
++scp_file_list='["../../../data/list/val_wav.scp", "../../../data/list/val_text.txt"]' \
++data_type_list='["source", "target"]' \
++jsonl_file_out="${val_data}"
# exp output dir
output_dir="./outputs"
log_file="${output_dir}/log.txt"
mkdir -p ${output_dir}
echo "log_file: ${log_file}"
torchrun \
--nnodes 1 \
--nproc_per_node ${gpu_num} \
../../../funasr/bin/train.py \
++model="${model_name_or_model_dir}" \
++train_data_set_list="${train_data}" \
++valid_data_set_list="${val_data}" \
++dataset_conf.batch_size=20000 \
++dataset_conf.batch_type="token" \
++dataset_conf.num_workers=4 \
++train_conf.max_epoch=50 \
++train_conf.log_interval=1 \
++train_conf.resume=false \
++train_conf.validate_interval=2000 \
++train_conf.save_checkpoint_interval=2000 \
++train_conf.keep_nbest_models=20 \
++optim_conf.lr=0.0002 \
++output_dir="${output_dir}" &> ${log_file}
examples/industrial_data_pretraining/qwen_audio/demo.py
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#!/usr/bin/env python3
# -*- encoding: utf-8 -*-
# Copyright FunASR (https://github.com/alibaba-damo-academy/FunASR). All Rights Reserved.
# MIT License (https://opensource.org/licenses/MIT)
# To install requirements: pip3 install -U "funasr[llm]"
from funasr import AutoModel
model = AutoModel(model="Qwen-Audio")
audio_in = "https://github.com/QwenLM/Qwen-Audio/raw/main/assets/audio/1272-128104-0000.flac"
prompt = "<|startoftranscription|><|en|><|transcribe|><|en|><|notimestamps|><|wo_itn|>"
res = model.generate(input=audio_in, prompt=prompt)
print(res)
examples/industrial_data_pretraining/qwen_audio/demo_chat.py
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#!/usr/bin/env python3
# -*- encoding: utf-8 -*-
# Copyright FunASR (https://github.com/alibaba-damo-academy/FunASR). All Rights Reserved.
# MIT License (https://opensource.org/licenses/MIT)
# To install requirements: pip3 install -U "funasr[llm]"
from funasr import AutoModel
model = AutoModel(model="Qwen-Audio-Chat")
audio_in = "https://github.com/QwenLM/Qwen-Audio/raw/main/assets/audio/1272-128104-0000.flac"
# 1st dialogue turn
prompt = "what does the person say?"
cache = {"history": None}
res = model.generate(input=audio_in, prompt=prompt, cache=cache)
print(res)
# 2nd dialogue turn
prompt = 'Find the start time and end time of the word "middle classes"'
res = model.generate(input=None, prompt=prompt, cache=cache)
print(res)
examples/industrial_data_pretraining/qwen_audio/demo_chat_from_local.py
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#!/usr/bin/env python3
# -*- encoding: utf-8 -*-
# Copyright FunASR (https://github.com/alibaba-damo-academy/FunASR). All Rights Reserved.
# MIT License (https://opensource.org/licenses/MIT)
# To install requirements: pip3 install -U "funasr[llm]"
from funasr import AutoModel
model = AutoModel(
model="Qwen-Audio-Chat",
model_path="/nfs/zhifu.gzf/init_model/qwen/Qwen-Audio-Chat",
)
audio_in = (
"https://isv-data.oss-cn-hangzhou.aliyuncs.com/ics/MaaS/ASR/test_audio/asr_example_zh.wav"
)
# 1st dialogue turn
prompt = "what does the person say?"
cache = {"history": None}
res = model.generate(input=audio_in, prompt=prompt, cache=cache)
print(res)
# 2nd dialogue turn
prompt = 'Find the start time and end time of the word "middle classes"'
res = model.generate(input=None, prompt=prompt, cache=cache)
print(res)
examples/industrial_data_pretraining/qwen_audio/demo_from_local.py
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#!/usr/bin/env python3
# -*- encoding: utf-8 -*-
# Copyright FunASR (https://github.com/alibaba-damo-academy/FunASR). All Rights Reserved.
# MIT License (https://opensource.org/licenses/MIT)
# To install requirements: pip3 install -U "funasr[llm]"
from funasr import AutoModel
model = AutoModel(
model="Qwen-Audio",
model_path="/nfs/zhifu.gzf/init_model/qwen/Qwen-Audio",
)
audio_in = "https://github.com/QwenLM/Qwen-Audio/raw/main/assets/audio/1272-128104-0000.flac"
prompt = "<|startoftranscription|><|en|><|transcribe|><|en|><|notimestamps|><|wo_itn|>"
res = model.generate(input=audio_in, prompt=prompt)
print(res)
examples/industrial_data_pretraining/scama/demo.py
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#!/usr/bin/env python3
# -*- encoding: utf-8 -*-
# Copyright FunASR (https://github.com/alibaba-damo-academy/FunASR). All Rights Reserved.
# MIT License (https://opensource.org/licenses/MIT)
from funasr import AutoModel
chunk_size = [5, 10, 5] # [0, 10, 5] 600ms, [0, 8, 4] 480ms
encoder_chunk_look_back = 0 # number of chunks to lookback for encoder self-attention
decoder_chunk_look_back = 0 # number of encoder chunks to lookback for decoder cross-attention
model = AutoModel(
model="/Users/zhifu/Downloads/modelscope_models/speech_SCAMA_asr-zh-cn-16k-common-vocab8358-streaming"
)
cache = {}
res = model.generate(
input="https://isv-data.oss-cn-hangzhou.aliyuncs.com/ics/MaaS/ASR/test_audio/asr_example_zh.wav",
chunk_size=chunk_size,
encoder_chunk_look_back=encoder_chunk_look_back,
decoder_chunk_look_back=decoder_chunk_look_back,
)
print(res)
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、480ms
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.generate(
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)
examples/industrial_data_pretraining/scama/demo.sh
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model="iic/speech_paraformer-large_asr_nat-zh-cn-16k-common-vocab8404-online"
python funasr/bin/inference.py \
+model=${model} \
+input="https://isv-data.oss-cn-hangzhou.aliyuncs.com/ics/MaaS/ASR/test_audio/asr_example_zh.wav" \
+output_dir="./outputs/debug" \
+device="cpu" \
examples/industrial_data_pretraining/seaco_paraformer/demo.py
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#!/usr/bin/env python3
# -*- encoding: utf-8 -*-
# Copyright FunASR (https://github.com/alibaba-damo-academy/FunASR). All Rights Reserved.
# MIT License (https://opensource.org/licenses/MIT)
from funasr import AutoModel
model = AutoModel(
model="iic/speech_seaco_paraformer_large_asr_nat-zh-cn-16k-common-vocab8404-pytorch",
# vad_model="iic/speech_fsmn_vad_zh-cn-16k-common-pytorch",
# punc_model="iic/punc_ct-transformer_zh-cn-common-vocab272727-pytorch",
# spk_model="iic/speech_campplus_sv_zh-cn_16k-common",
)
# example1
res = model.generate(
input="https://isv-data.oss-cn-hangzhou.aliyuncs.com/ics/MaaS/ASR/test_audio/asr_example_zh.wav",
hotword="达摩院 魔搭",
# return_raw_text=True, # return raw text recognition results splited by space of equal length with timestamp
# preset_spk_num=2, # preset speaker num for speaker cluster model
# sentence_timestamp=True, # return sentence level information when spk_model is not given
)
print(res)
"""
# tensor or numpy as input
# example2
import torchaudio
import os
wav_file = os.path.join(model.model_path, "example/asr_example.wav")
input_tensor, sample_rate = torchaudio.load(wav_file)
input_tensor = input_tensor.mean(0)
res = model.generate(input=[input_tensor], batch_size_s=300, is_final=True)
# example3
import soundfile
wav_file = os.path.join(model.model_path, "example/asr_example.wav")
speech, sample_rate = soundfile.read(wav_file)
res = model.generate(input=[speech], batch_size_s=300, is_final=True)
"""
examples/industrial_data_pretraining/seaco_paraformer/demo.sh
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model="iic/speech_seaco_paraformer_large_asr_nat-zh-cn-16k-common-vocab8404-pytorch"
vad_model="iic/speech_fsmn_vad_zh-cn-16k-common-pytorch"
vad_model_revision="master"
punc_model="iic/punc_ct-transformer_zh-cn-common-vocab272727-pytorch"
punc_model_revision="master"
python funasr/bin/inference.py \
+model=${model} \
+vad_model=${vad_model} \
+vad_model_revision=${vad_model_revision} \
+punc_model=${punc_model} \
+punc_model_revision=${punc_model_revision} \
+input="https://isv-data.oss-cn-hangzhou.aliyuncs.com/ics/MaaS/ASR/test_audio/asr_example_zh.wav" \
+output_dir="./outputs/debug" \
+device="cpu" \
+"hotword='达摩院 魔搭'"
examples/industrial_data_pretraining/seaco_paraformer/finetune.sh
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# Copyright FunASR (https://github.com/alibaba-damo-academy/FunASR). All Rights Reserved.
# MIT License (https://opensource.org/licenses/MIT)
# which gpu to train or finetune
export CUDA_VISIBLE_DEVICES="0,1"
gpu_num=$(echo $CUDA_VISIBLE_DEVICES | awk -F "," '{print NF}')
# model_name from model_hub, or model_dir in local path
## option 1, download model automatically
model_name_or_model_dir="iic/speech_seaco_paraformer_large_asr_nat-zh-cn-16k-common-vocab8404-pytorch"
## option 2, download model by git
#local_path_root=${workspace}/modelscope_models
#mkdir -p ${local_path_root}/${model_name_or_model_dir}
#git clone https://www.modelscope.cn/${model_name_or_model_dir}.git ${local_path_root}/${model_name_or_model_dir}
#model_name_or_model_dir=${local_path_root}/${model_name_or_model_dir}
# data dir, which contains: train.json, val.json
data_dir="../../../data/list"
train_data="${data_dir}/train.jsonl"
val_data="${data_dir}/val.jsonl"
# generate train.jsonl and val.jsonl from wav.scp and text.txt
scp2jsonl \
++scp_file_list='["../../../data/list/train_wav.scp", "../../../data/list/train_text.txt"]' \
++data_type_list='["source", "target"]' \
++jsonl_file_out="${train_data}"
scp2jsonl \
++scp_file_list='["../../../data/list/val_wav.scp", "../../../data/list/val_text.txt"]' \
++data_type_list='["source", "target"]' \
++jsonl_file_out="${val_data}"
# exp output dir
output_dir="./outputs"
log_file="${output_dir}/log.txt"
mkdir -p ${output_dir}
echo "log_file: ${log_file}"
torchrun \
--nnodes 1 \
--nproc_per_node ${gpu_num} \
../../../funasr/bin/train.py \
++model="${model_name_or_model_dir}" \
++train_data_set_list="${train_data}" \
++valid_data_set_list="${val_data}" \
++dataset_conf.batch_size=20000 \
++dataset_conf.batch_type="token" \
++dataset_conf.num_workers=4 \
++train_conf.max_epoch=50 \
++train_conf.log_interval=1 \
++train_conf.resume=false \
++train_conf.validate_interval=2000 \
++train_conf.save_checkpoint_interval=2000 \
++train_conf.avg_keep_nbest_models_type='loss' \
++train_conf.keep_nbest_models=20 \
++optim_conf.lr=0.0002 \
++output_dir="${output_dir}" &> ${log_file}
examples/industrial_data_pretraining/sense_voice/demo.py
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#!/usr/bin/env python3
# -*- encoding: utf-8 -*-
# Copyright FunASR (https://github.com/alibaba-damo-academy/FunASR). All Rights Reserved.
# MIT License (https://opensource.org/licenses/MIT)
from funasr import AutoModel
model = AutoModel(
model="/Users/zhifu/Downloads/modelscope_models/SenseVoiceModelscope",
# vad_model="iic/speech_fsmn_vad_zh-cn-16k-common-pytorch",
# vad_kwargs={"max_single_segment_time": 30000},
)
input_wav = (
"https://isv-data.oss-cn-hangzhou.aliyuncs.com/ics/MaaS/ASR/test_audio/asr_example_zh.wav"
)
DecodingOptions = {
"task": ("ASR", "AED", "SER"),
"language": "auto",
"fp16": True,
"gain_event": True,
"beam_size": 5,
}
res = model.generate(input=input_wav, batch_size_s=0, DecodingOptions=DecodingOptions)
print(res)
examples/industrial_data_pretraining/sense_voice/demo_fsmn.py
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#!/usr/bin/env python3
# -*- encoding: utf-8 -*-
# Copyright FunASR (https://github.com/alibaba-damo-academy/FunASR). All Rights Reserved.
# MIT License (https://opensource.org/licenses/MIT)
from funasr import AutoModel
model = AutoModel(
model="/Users/zhifu/Downloads/modelscope_models/SenseVoiceModelscopeFSMN",
vad_model="iic/speech_fsmn_vad_zh-cn-16k-common-pytorch",
vad_kwargs={"max_single_segment_time": 30000},
)
input_wav = (
"https://isv-data.oss-cn-hangzhou.aliyuncs.com/ics/MaaS/ASR/test_audio/asr_example_zh.wav"
)
DecodingOptions = {
"task": ("ASR", "AED", "SER"),
"language": "auto",
"fp16": True,
"gain_event": True,
"beam_size": 5,
}
res = model.generate(input=input_wav, batch_size_s=0, DecodingOptions=DecodingOptions, beam_size=5)
print(res)
examples/industrial_data_pretraining/sense_voice/finetune.sh
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# Copyright FunASR (https://github.com/alibaba-damo-academy/FunASR). All Rights Reserved.
# MIT License (https://opensource.org/licenses/MIT)
# which gpu to train or finetune
export CUDA_VISIBLE_DEVICES="0"
gpu_num=$(echo $CUDA_VISIBLE_DEVICES | awk -F "," '{print NF}')
# model_name from model_hub, or model_dir in local path
## option 1, download model automatically
model_name_or_model_dir="iic/speech_paraformer-large_asr_nat-zh-cn-16k-common-vocab8404-pytorch"
model_name_or_model_dir="/Users/zhifu/Downloads/modelscope_models/SenseVoiceModelscope"
## option 2, download model by git
#local_path_root=${workspace}/modelscope_models
#mkdir -p ${local_path_root}/${model_name_or_model_dir}
#git clone https://www.modelscope.cn/${model_name_or_model_dir}.git ${local_path_root}/${model_name_or_model_dir}
#model_name_or_model_dir=${local_path_root}/${model_name_or_model_dir}
# data dir, which contains: train.json, val.json
data_dir="../../../data/list"
train_data="${data_dir}/train.jsonl"
val_data="${data_dir}/val.jsonl"
# generate train.jsonl and val.jsonl from wav.scp and text.txt
scp2jsonl \
++scp_file_list='["../../../data/list/train_wav.scp", "../../../data/list/train_text.txt"]' \
++data_type_list='["source", "target"]' \
++jsonl_file_out="${train_data}"
scp2jsonl \
++scp_file_list='["../../../data/list/val_wav.scp", "../../../data/list/val_text.txt"]' \
++data_type_list='["source", "target"]' \
++jsonl_file_out="${val_data}"
# exp output dir
output_dir="./outputs"
log_file="${output_dir}/log.txt"
mkdir -p ${output_dir}
echo "log_file: ${log_file}"
#torchrun \
#--nnodes 1 \
#--node_rank 0 \
#--nproc_per_node ${gpu_num} \
python \
../../../funasr/bin/train.py \
++model="${model_name_or_model_dir}" \
++train_data_set_list="${train_data}" \
++valid_data_set_list="${val_data}" \
++dataset_conf.batch_size=500 \
++dataset_conf.batch_type="token" \
++dataset_conf.num_workers=0 \
++train_conf.max_epoch=50 \
++train_conf.log_interval=1 \
++train_conf.resume=false \
++train_conf.validate_interval=2000 \
++train_conf.save_checkpoint_interval=2000 \
++train_conf.keep_nbest_models=20 \
++train_conf.avg_nbest_model=10 \
++optim_conf.lr=0.0002 \
++debug=true \
++device="cpu" \
++output_dir="${output_dir}" #&> ${log_file}
examples/industrial_data_pretraining/transducer/demo.py
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#!/usr/bin/env python3
# -*- encoding: utf-8 -*-
# Copyright FunASR (https://github.com/alibaba-damo-academy/FunASR). All Rights Reserved.
# MIT License (https://opensource.org/licenses/MIT)
from funasr import AutoModel
# Transducer, BAT and RWKV_BAT models are just same to use, use the correct model_revision
# https://modelscope.cn/models?name=transducer&page=1&tasks=auto-speech-recognition&type=audio
model = AutoModel(
model="iic/speech_bat_asr-zh-cn-16k-aishell1-vocab4234-pytorch",
)
res = model.generate(
input="https://isv-data.oss-cn-hangzhou.aliyuncs.com/ics/MaaS/ASR/test_audio/asr_example_zh.wav"
)
print(res)
examples/industrial_data_pretraining/uniasr/demo.py
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#!/usr/bin/env python3
# -*- encoding: utf-8 -*-
# Copyright FunASR (https://github.com/alibaba-damo-academy/FunASR). All Rights Reserved.
# MIT License (https://opensource.org/licenses/MIT)
from funasr import AutoModel
model = AutoModel(
model="iic/speech_UniASR-large_asr_2pass-zh-cn-16k-common-vocab8358-tensorflow1-offline",
)
res = model.generate(
input="https://isv-data.oss-cn-hangzhou.aliyuncs.com/ics/MaaS/ASR/test_audio/asr_example_zh.wav"
)
print(res)
""" can not use currently
from funasr import AutoFrontend
frontend = AutoFrontend(model="iic/speech_paraformer-large_asr_nat-zh-cn-16k-common-vocab8404-pytorch")
fbanks = frontend(input="https://isv-data.oss-cn-hangzhou.aliyuncs.com/ics/MaaS/ASR/test_audio/asr_example_zh.wav", batch_size=2)
for batch_idx, fbank_dict in enumerate(fbanks):
res = model.generate(**fbank_dict)
print(res)
"""
examples/industrial_data_pretraining/uniasr/demo.sh
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model="iic/speech_paraformer-large_asr_nat-zh-cn-16k-common-vocab8404-pytorch"
python funasr/bin/inference.py \
+model=${model} \
+input="https://isv-data.oss-cn-hangzhou.aliyuncs.com/ics/MaaS/ASR/test_audio/asr_example_zh.wav" \
+output_dir="./outputs/debug" \
+device="cpu" \
examples/industrial_data_pretraining/whisper/demo.py
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#!/usr/bin/env python3
# -*- encoding: utf-8 -*-
# Copyright FunASR (https://github.com/alibaba-damo-academy/FunASR). All Rights Reserved.
# MIT License (https://opensource.org/licenses/MIT)
# To install requirements: pip3 install -U openai-whisper
from funasr import AutoModel
model = AutoModel(
model="iic/Whisper-large-v3",
vad_model="iic/speech_fsmn_vad_zh-cn-16k-common-pytorch",
vad_kwargs={"max_single_segment_time": 30000},
)
DecodingOptions = {
"task": "transcribe",
"language": None,
"beam_size": None,
"fp16": True,
"without_timestamps": False,
"prompt": None,
}
res = model.generate(
DecodingOptions=DecodingOptions,
batch_size_s=0,
input="https://isv-data.oss-cn-hangzhou.aliyuncs.com/ics/MaaS/ASR/test_audio/asr_example_zh.wav",
)
print(res)
examples/industrial_data_pretraining/whisper/demo_from_openai.py
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#!/usr/bin/env python3
# -*- encoding: utf-8 -*-
# Copyright FunASR (https://github.com/alibaba-damo-academy/FunASR). All Rights Reserved.
# MIT License (https://opensource.org/licenses/MIT)
# To install requirements: pip3 install -U openai-whisper
from funasr import AutoModel
# model = AutoModel(model="Whisper-small", hub="openai")
# model = AutoModel(model="Whisper-medium", hub="openai")
# model = AutoModel(model="Whisper-large-v2", hub="openai")
model = AutoModel(
model="Whisper-large-v3",
vad_model="iic/speech_fsmn_vad_zh-cn-16k-common-pytorch",
vad_kwargs={"max_single_segment_time": 30000},
hub="openai",
)
DecodingOptions = {
"task": "transcribe",
"language": None,
"beam_size": None,
"fp16": True,
"without_timestamps": False,
"prompt": None,
}
res = model.generate(
DecodingOptions=DecodingOptions,
batch_size_s=0,
input="https://isv-data.oss-cn-hangzhou.aliyuncs.com/ics/MaaS/ASR/test_audio/asr_example_zh.wav",
)
print(res)
examples/industrial_data_pretraining/whisper/infer.sh
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# Copyright FunASR (https://github.com/alibaba-damo-academy/FunASR). All Rights Reserved.
# MIT License (https://opensource.org/licenses/MIT)
# To install requirements: pip3 install -U openai-whisper
# method1, inference from model hub
# for more input type, please ref to readme.md
input="https://isv-data.oss-cn-hangzhou.aliyuncs.com/ics/MaaS/ASR/test_audio/asr_example_zh.wav"
output_dir="./outputs/debug"
model="iic/speech_whisper-large_asr_multilingual"
device="cuda:0" # "cuda:0" for gpu0, "cuda:1" for gpu1, "cpu"
python -m funasr.bin.inference \
++model=${model} \
++input="${input}" \
++output_dir="${output_dir}" \
++device="${device}" \
examples/industrial_data_pretraining/whisper/infer_from_local.sh
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# Copyright FunASR (https://github.com/alibaba-damo-academy/FunASR). All Rights Reserved.
# MIT License (https://opensource.org/licenses/MIT)
# To install requirements: pip3 install -U openai-whisper
# method2, inference from local model
# for more input type, please ref to readme.md
input="https://isv-data.oss-cn-hangzhou.aliyuncs.com/ics/MaaS/ASR/test_audio/asr_example_zh.wav"
output_dir="./outputs/debug"
workspace=`pwd`
# download model
local_path_root=${workspace}/modelscope_models
mkdir -p ${local_path_root}
#Whisper-large-v2
#local_path=${local_path_root}/speech_whisper-large_asr_multilingual
#git clone https://www.modelscope.cn/iic/speech_whisper-large_asr_multilingual.git ${local_path}
#init_param="${local_path}/large-v2.pt"
#Whisper-large-v3
local_path=${local_path_root}/Whisper-large-v3
git clone https://www.modelscope.cn/iic/Whisper-large-v3.git ${local_path}
init_param="${local_path}/large-v3.pt"
device="cuda:0" # "cuda:0" for gpu0, "cuda:1" for gpu1, "cpu"
config="config.yaml"
python -m funasr.bin.inference \
--config-path "${local_path}" \
--config-name "${config}" \
++init_param="${init_param}" \
++input="${input}" \
++output_dir="${output_dir}" \
++device="${device}" \
examples/industrial_data_pretraining/whisper/infer_from_openai.sh
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# Copyright FunASR (https://github.com/alibaba-damo-academy/FunASR). All Rights Reserved.
# MIT License (https://opensource.org/licenses/MIT)
# To install requirements: pip3 install -U openai-whisper
# method1, inference from model hub
# for more input type, please ref to readme.md
input="https://isv-data.oss-cn-hangzhou.aliyuncs.com/ics/MaaS/ASR/test_audio/asr_example_zh.wav"
output_dir="./outputs/debug"
#model="Whisper-small"
#model="Whisper-medium"
model="Whisper-large-v2"
#model="Whisper-large-v3"
hub="openai"
device="cuda:0" # "cuda:0" for gpu0, "cuda:1" for gpu1, "cpu"
python -m funasr.bin.inference \
++model=${model} \
++hub=${hub} \
++input="${input}" \
++output_dir="${output_dir}" \
++device="${device}" \