haoneng.lhn
2023-05-19 dcc4f728cdb83a48250825288bbb92b7a0d2848b
update paraformer online text postprocess
2个文件已修改
1个文件已添加
52 ■■■■■ 已修改文件
egs_modelscope/asr/paraformer/speech_paraformer-large_asr_nat-zh-cn-16k-common-vocab8404-online/demo.py 2 ●●● 补丁 | 查看 | 原始文档 | blame | 历史
egs_modelscope/asr/paraformer/speech_paraformer_asr_nat-zh-cn-16k-common-vocab8404-online/demo.py 39 ●●●●● 补丁 | 查看 | 原始文档 | blame | 历史
funasr/bin/asr_infer.py 11 ●●●● 补丁 | 查看 | 原始文档 | blame | 历史
egs_modelscope/asr/paraformer/speech_paraformer-large_asr_nat-zh-cn-16k-common-vocab8404-online/demo.py
@@ -34,6 +34,6 @@
    rec_result = inference_pipeline(audio_in=speech[sample_offset: sample_offset + stride_size],
                                    param_dict=param_dict)
    if len(rec_result) != 0:
        final_result += rec_result['text'] + " "
        final_result += rec_result['text']
        print(rec_result)
print(final_result)
egs_modelscope/asr/paraformer/speech_paraformer_asr_nat-zh-cn-16k-common-vocab8404-online/demo.py
New file
@@ -0,0 +1,39 @@
import os
import logging
import torch
import soundfile
from modelscope.pipelines import pipeline
from modelscope.utils.constant import Tasks
from modelscope.utils.logger import get_logger
logger = get_logger(log_level=logging.CRITICAL)
logger.setLevel(logging.CRITICAL)
os.environ["MODELSCOPE_CACHE"] = "./"
inference_pipeline = pipeline(
    task=Tasks.auto_speech_recognition,
    model='damo/speech_paraformer_asr_nat-zh-cn-16k-common-vocab8404-online',
    model_revision='v1.0.4'
)
model_dir = os.path.join(os.environ["MODELSCOPE_CACHE"], "damo/speech_paraformer_asr_nat-zh-cn-16k-common-vocab8404-online")
speech, sample_rate = soundfile.read(os.path.join(model_dir, "example/asr_example.wav"))
speech_length = speech.shape[0]
sample_offset = 0
chunk_size = [8, 8, 4] #[5, 10, 5] 600ms, [8, 8, 4] 480ms
stride_size =  chunk_size[1] * 960
param_dict = {"cache": dict(), "is_final": False, "chunk_size": chunk_size}
final_result = ""
for sample_offset in range(0, speech_length, min(stride_size, speech_length - sample_offset)):
    if sample_offset + stride_size >= speech_length - 1:
        stride_size = speech_length - sample_offset
        param_dict["is_final"] = True
    rec_result = inference_pipeline(audio_in=speech[sample_offset: sample_offset + stride_size],
                                    param_dict=param_dict)
    if len(rec_result) != 0:
        final_result += rec_result['text']
        print(rec_result)
print(final_result.strip())
funasr/bin/asr_infer.py
@@ -828,9 +828,16 @@
                # Change integer-ids to tokens
                token = self.converter.ids2tokens(token_int)
                token = " ".join(token)
                postprocessed_result = ""
                for item in token:
                    if item.endswith('@@'):
                        postprocessed_result += item[:-2]
                    elif re.match('^[a-zA-Z]+$', item):
                        postprocessed_result += item + " "
                    else:
                        postprocessed_result += item
                results.append(token)
                results.append(postprocessed_result)
        # assert check_return_type(results)
        return results