From c1d4bd297a4418ef44882079c4845cfe64ed0b21 Mon Sep 17 00:00:00 2001
From: 游雁 <zhifu.gzf@alibaba-inc.com>
Date: 星期四, 27 四月 2023 19:36:32 +0800
Subject: [PATCH] docs
---
egs_modelscope/asr/uniasr/speech_UniASR_asr_2pass-he-16k-common-vocab1085-pytorch/infer.py | 4
egs_modelscope/asr/uniasr/speech_UniASR_asr_2pass-ko-16k-common-vocab6400-tensorflow1-online/infer.py | 4
egs_modelscope/asr/uniasr/speech_UniASR_asr_2pass-zh-cn-16k-common-vocab8358-tensorflow1-offline/infer.py | 4
egs_modelscope/asr/uniasr/speech_UniASR_asr_2pass-cn-dialect-16k-vocab8358-tensorflow1-offline/infer.py | 4
egs_modelscope/asr/uniasr/speech_UniASR_asr_2pass-pt-16k-common-vocab1617-tensorflow1-online/infer.py | 4
egs_modelscope/asr/uniasr/speech_UniASR_asr_2pass-fa-16k-common-vocab1257-pytorch-online/infer.py | 4
egs_modelscope/asr/uniasr/speech_UniASR_asr_2pass-vi-16k-common-vocab1001-pytorch-online/infer.py | 4
egs_modelscope/asr/paraformer/speech_paraformer_asr_nat-zh-cn-16k-aishell2-vocab5212-pytorch/demo.py | 4
egs_modelscope/asr/uniasr/speech_UniASR_asr_2pass-my-16k-common-vocab696-pytorch/infer.py | 4
egs_modelscope/asr/uniasr/speech_UniASR_asr_2pass-ko-16k-common-vocab6400-tensorflow1-offline/infer.py | 4
egs_modelscope/tp/speech_timestamp_prediction-v1-16k-offline/demo.py | 4
egs_modelscope/vad/speech_fsmn_vad_zh-cn-8k-common/demo.py | 4
egs_modelscope/asr/uniasr/speech_UniASR_asr_2pass-ja-16k-common-vocab93-tensorflow1-offline/infer.py | 4
egs_modelscope/asr/uniasr/speech_UniASR_asr_2pass-pt-16k-common-vocab1617-tensorflow1-offline/infer.py | 4
egs_modelscope/asr/uniasr/speech_UniASR_asr_2pass-de-16k-common-vocab3690-tensorflow1-offline/infer.py | 4
egs_modelscope/asr/uniasr/speech_UniASR_asr_2pass-de-16k-common-vocab3690-tensorflow1-online/infer.py | 4
egs_modelscope/asr/uniasr/speech_UniASR_asr_2pass-ur-16k-common-vocab877-pytorch/infer.py | 4
egs_modelscope/vad/speech_fsmn_vad_zh-cn-16k-common/demo_online.py | 4
egs_modelscope/asr/uniasr/speech_UniASR_asr_2pass-ru-16k-common-vocab1664-tensorflow1-online/infer.py | 4
egs_modelscope/asr/uniasr/speech_UniASR_asr_2pass-en-16k-common-vocab1080-tensorflow1-online/infer.py | 4
egs_modelscope/asr/uniasr/speech_UniASR_asr_2pass-id-16k-common-vocab1067-tensorflow1-online/infer.py | 4
egs_modelscope/asr/uniasr/speech_UniASR_asr_2pass-ja-16k-common-vocab93-tensorflow1-online/infer.py | 4
egs_modelscope/asr/uniasr/speech_UniASR_asr_2pass-zh-cn-8k-common-vocab8358-tensorflow1-offline/infer.py | 4
egs_modelscope/asr/uniasr/speech_UniASR_asr_2pass-zh-cn-8k-common-vocab8358-tensorflow1-online/infer.py | 4
egs_modelscope/asr/uniasr/speech_UniASR_asr_2pass-fr-16k-common-vocab3472-tensorflow1-online/infer.py | 4
egs_modelscope/asr/uniasr/speech_UniASR_asr_2pass-zh-cn-8k-common-vocab3445-pytorch-online/infer.py | 4
egs_modelscope/asr/uniasr/speech_UniASR_asr_2pass-zh-cn-16k-common-vocab8358-tensorflow1-online/infer.py | 4
egs_modelscope/asr/uniasr/speech_UniASR_asr_2pass-cantonese-CHS-16k-common-vocab1468-tensorflow1-online/infer.py | 4
egs_modelscope/tp/TEMPLATE/README.md | 4
egs_modelscope/asr/data2vec/speech_data2vec_pretrain-paraformer-zh-cn-aishell2-16k/infer.py | 4
egs_modelscope/asr/paraformer/speech_paraformer-tiny-commandword_asr_nat-zh-cn-16k-vocab544-pytorch/infer.py | 4
egs_modelscope/asr/uniasr/speech_UniASR_asr_2pass-fr-16k-common-vocab3472-tensorflow1-offline/infer.py | 4
egs_modelscope/vad/speech_fsmn_vad_zh-cn-16k-common/demo.py | 4
egs_modelscope/asr/conformer/speech_conformer_asr_nat-zh-cn-16k-aishell1-vocab4234-pytorch/demo.py | 4
egs_modelscope/asr/uniasr/speech_UniASR_asr_2pass-cantonese-CHS-16k-common-vocab1468-tensorflow1-offline/infer.py | 4
egs_modelscope/asr/uniasr/speech_UniASR_asr_2pass-es-16k-common-vocab3445-tensorflow1-offline/infer.py | 4
egs_modelscope/asr/uniasr/speech_UniASR_asr_2pass-en-16k-common-vocab1080-tensorflow1-offline/infer.py | 4
egs_modelscope/asr/uniasr/speech_UniASR_asr_2pass-es-16k-common-vocab3445-tensorflow1-online/infer.py | 4
egs_modelscope/asr/paraformer/speech_paraformer_asr_nat-zh-cn-16k-aishell1-vocab4234-pytorch/demo.py | 4
egs_modelscope/asr/uniasr/speech_UniASR_asr_2pass-vi-16k-common-vocab1001-pytorch-offline/infer.py | 4
egs_modelscope/asr/paraformerbert/speech_paraformerbert_asr_nat-zh-cn-16k-aishell2-vocab5212-pytorch/infer.py | 4
egs_modelscope/vad/speech_fsmn_vad_zh-cn-8k-common/demo_online.py | 4
egs_modelscope/asr/uniasr/speech_UniASR_asr_2pass-cn-dialect-16k-vocab8358-tensorflow1-online/infer.py | 4
egs_modelscope/punctuation/TEMPLATE/README.md | 8 +-
egs_modelscope/asr/uniasr/speech_UniASR_asr_2pass-zh-cn-8k-common-vocab3445-pytorch-offline/infer.py | 4
egs_modelscope/asr/data2vec/speech_data2vec_pretrain-zh-cn-aishell2-16k-pytorch/infer.py | 4
egs_modelscope/asr/uniasr/speech_UniASR_asr_2pass-id-16k-common-vocab1067-tensorflow1-offline/infer.py | 4
egs_modelscope/asr/paraformerbert/speech_paraformerbert_asr_nat-zh-cn-16k-aishell1-vocab4234-pytorch/infer.py | 4
egs_modelscope/asr/uniasr/speech_UniASR_asr_2pass-fa-16k-common-vocab1257-pytorch-offline/infer.py | 4
egs_modelscope/punctuation/punc_ct-transformer_zh-cn-common-vocab272727-pytorch/demo.py | 4
egs_modelscope/asr/uniasr/speech_UniASR_asr_2pass-ru-16k-common-vocab1664-tensorflow1-offline/infer.py | 4
egs_modelscope/asr/conformer/speech_conformer_asr_nat-zh-cn-16k-aishell2-vocab5212-pytorch/demo.py | 4
egs_modelscope/lm/speech_transformer_lm_zh-cn-common-vocab8404-pytorch/infer.py | 4
53 files changed, 108 insertions(+), 108 deletions(-)
diff --git a/egs_modelscope/asr/conformer/speech_conformer_asr_nat-zh-cn-16k-aishell1-vocab4234-pytorch/demo.py b/egs_modelscope/asr/conformer/speech_conformer_asr_nat-zh-cn-16k-aishell1-vocab4234-pytorch/demo.py
index 3594815..87bb652 100644
--- a/egs_modelscope/asr/conformer/speech_conformer_asr_nat-zh-cn-16k-aishell1-vocab4234-pytorch/demo.py
+++ b/egs_modelscope/asr/conformer/speech_conformer_asr_nat-zh-cn-16k-aishell1-vocab4234-pytorch/demo.py
@@ -4,11 +4,11 @@
if __name__ == '__main__':
audio_in = 'https://isv-data.oss-cn-hangzhou.aliyuncs.com/ics/MaaS/ASR/test_audio/asr_example_zh.wav'
output_dir = None
- inference_pipline = pipeline(
+ inference_pipeline = pipeline(
task=Tasks.auto_speech_recognition,
model="damo/speech_conformer_asr_nat-zh-cn-16k-aishell1-vocab4234-pytorch",
output_dir=output_dir,
)
- rec_result = inference_pipline(audio_in=audio_in)
+ rec_result = inference_pipeline(audio_in=audio_in)
print(rec_result)
diff --git a/egs_modelscope/asr/conformer/speech_conformer_asr_nat-zh-cn-16k-aishell2-vocab5212-pytorch/demo.py b/egs_modelscope/asr/conformer/speech_conformer_asr_nat-zh-cn-16k-aishell2-vocab5212-pytorch/demo.py
index b55b59f..3b0164a 100644
--- a/egs_modelscope/asr/conformer/speech_conformer_asr_nat-zh-cn-16k-aishell2-vocab5212-pytorch/demo.py
+++ b/egs_modelscope/asr/conformer/speech_conformer_asr_nat-zh-cn-16k-aishell2-vocab5212-pytorch/demo.py
@@ -4,10 +4,10 @@
if __name__ == "__main__":
audio_in = "https://modelscope.oss-cn-beijing.aliyuncs.com/test/audios/asr_example.wav"
output_dir = "./results"
- inference_pipline = pipeline(
+ inference_pipeline = pipeline(
task=Tasks.auto_speech_recognition,
model="damo/speech_conformer_asr_nat-zh-cn-16k-aishell2-vocab5212-pytorch",
output_dir=output_dir,
)
- rec_result = inference_pipline(audio_in=audio_in)
+ rec_result = inference_pipeline(audio_in=audio_in)
print(rec_result)
diff --git a/egs_modelscope/asr/data2vec/speech_data2vec_pretrain-paraformer-zh-cn-aishell2-16k/infer.py b/egs_modelscope/asr/data2vec/speech_data2vec_pretrain-paraformer-zh-cn-aishell2-16k/infer.py
index 77b2cbd..7a6b750 100644
--- a/egs_modelscope/asr/data2vec/speech_data2vec_pretrain-paraformer-zh-cn-aishell2-16k/infer.py
+++ b/egs_modelscope/asr/data2vec/speech_data2vec_pretrain-paraformer-zh-cn-aishell2-16k/infer.py
@@ -16,13 +16,13 @@
os.environ['CUDA_VISIBLE_DEVICES'] = str(gpu_list[gpu_id])
else:
os.environ['CUDA_VISIBLE_DEVICES'] = str(gpu_id)
- inference_pipline = pipeline(
+ inference_pipeline = pipeline(
task=Tasks.auto_speech_recognition,
model="damo/speech_data2vec_pretrain-paraformer-zh-cn-aishell2-16k",
output_dir=output_dir_job,
)
audio_in = os.path.join(split_dir, "wav.{}.scp".format(idx))
- inference_pipline(audio_in=audio_in)
+ inference_pipeline(audio_in=audio_in)
def modelscope_infer(params):
diff --git a/egs_modelscope/asr/data2vec/speech_data2vec_pretrain-zh-cn-aishell2-16k-pytorch/infer.py b/egs_modelscope/asr/data2vec/speech_data2vec_pretrain-zh-cn-aishell2-16k-pytorch/infer.py
index 0d06377..f07f308 100644
--- a/egs_modelscope/asr/data2vec/speech_data2vec_pretrain-zh-cn-aishell2-16k-pytorch/infer.py
+++ b/egs_modelscope/asr/data2vec/speech_data2vec_pretrain-zh-cn-aishell2-16k-pytorch/infer.py
@@ -16,13 +16,13 @@
os.environ['CUDA_VISIBLE_DEVICES'] = str(gpu_list[gpu_id])
else:
os.environ['CUDA_VISIBLE_DEVICES'] = str(gpu_id)
- inference_pipline = pipeline(
+ inference_pipeline = pipeline(
task=Tasks.auto_speech_recognition,
model="damo/speech_data2vec_pretrain-zh-cn-aishell2-16k-pytorch",
output_dir=output_dir_job,
)
audio_in = os.path.join(split_dir, "wav.{}.scp".format(idx))
- inference_pipline(audio_in=audio_in)
+ inference_pipeline(audio_in=audio_in)
def modelscope_infer(params):
diff --git a/egs_modelscope/asr/paraformer/speech_paraformer-tiny-commandword_asr_nat-zh-cn-16k-vocab544-pytorch/infer.py b/egs_modelscope/asr/paraformer/speech_paraformer-tiny-commandword_asr_nat-zh-cn-16k-vocab544-pytorch/infer.py
index d1fbca2..00be793 100644
--- a/egs_modelscope/asr/paraformer/speech_paraformer-tiny-commandword_asr_nat-zh-cn-16k-vocab544-pytorch/infer.py
+++ b/egs_modelscope/asr/paraformer/speech_paraformer-tiny-commandword_asr_nat-zh-cn-16k-vocab544-pytorch/infer.py
@@ -16,14 +16,14 @@
os.environ['CUDA_VISIBLE_DEVICES'] = str(gpu_list[gpu_id])
else:
os.environ['CUDA_VISIBLE_DEVICES'] = str(gpu_id)
- inference_pipline = pipeline(
+ inference_pipeline = pipeline(
task=Tasks.auto_speech_recognition,
model="damo/speech_paraformer-tiny-commandword_asr_nat-zh-cn-16k-vocab544-pytorch",
output_dir=output_dir_job,
batch_size=64
)
audio_in = os.path.join(split_dir, "wav.{}.scp".format(idx))
- inference_pipline(audio_in=audio_in)
+ inference_pipeline(audio_in=audio_in)
def modelscope_infer(params):
diff --git a/egs_modelscope/asr/paraformer/speech_paraformer_asr_nat-zh-cn-16k-aishell1-vocab4234-pytorch/demo.py b/egs_modelscope/asr/paraformer/speech_paraformer_asr_nat-zh-cn-16k-aishell1-vocab4234-pytorch/demo.py
index 4125a57..2863c1a 100644
--- a/egs_modelscope/asr/paraformer/speech_paraformer_asr_nat-zh-cn-16k-aishell1-vocab4234-pytorch/demo.py
+++ b/egs_modelscope/asr/paraformer/speech_paraformer_asr_nat-zh-cn-16k-aishell1-vocab4234-pytorch/demo.py
@@ -4,12 +4,12 @@
if __name__ == '__main__':
audio_in = 'https://isv-data.oss-cn-hangzhou.aliyuncs.com/ics/MaaS/ASR/test_audio/asr_example_zh.wav'
output_dir = None
- inference_pipline = pipeline(
+ inference_pipeline = pipeline(
task=Tasks.auto_speech_recognition,
model="damo/speech_paraformer_asr_nat-zh-cn-16k-aishell1-vocab4234-pytorch",
output_dir=output_dir,
batch_size=1,
)
- rec_result = inference_pipline(audio_in=audio_in)
+ rec_result = inference_pipeline(audio_in=audio_in)
print(rec_result)
diff --git a/egs_modelscope/asr/paraformer/speech_paraformer_asr_nat-zh-cn-16k-aishell2-vocab5212-pytorch/demo.py b/egs_modelscope/asr/paraformer/speech_paraformer_asr_nat-zh-cn-16k-aishell2-vocab5212-pytorch/demo.py
index dec7de0..f2db74e 100644
--- a/egs_modelscope/asr/paraformer/speech_paraformer_asr_nat-zh-cn-16k-aishell2-vocab5212-pytorch/demo.py
+++ b/egs_modelscope/asr/paraformer/speech_paraformer_asr_nat-zh-cn-16k-aishell2-vocab5212-pytorch/demo.py
@@ -4,10 +4,10 @@
if __name__ == "__main__":
audio_in = "https://modelscope.oss-cn-beijing.aliyuncs.com/test/audios/asr_example.wav"
output_dir = "./results"
- inference_pipline = pipeline(
+ inference_pipeline = pipeline(
task=Tasks.auto_speech_recognition,
model="damo/speech_paraformer_asr_nat-zh-cn-16k-aishell2-vocab5212-pytorch",
output_dir=output_dir,
)
- rec_result = inference_pipline(audio_in=audio_in)
+ rec_result = inference_pipeline(audio_in=audio_in)
print(rec_result)
diff --git a/egs_modelscope/asr/paraformerbert/speech_paraformerbert_asr_nat-zh-cn-16k-aishell1-vocab4234-pytorch/infer.py b/egs_modelscope/asr/paraformerbert/speech_paraformerbert_asr_nat-zh-cn-16k-aishell1-vocab4234-pytorch/infer.py
index df18903..f4c4fc2 100644
--- a/egs_modelscope/asr/paraformerbert/speech_paraformerbert_asr_nat-zh-cn-16k-aishell1-vocab4234-pytorch/infer.py
+++ b/egs_modelscope/asr/paraformerbert/speech_paraformerbert_asr_nat-zh-cn-16k-aishell1-vocab4234-pytorch/infer.py
@@ -4,11 +4,11 @@
if __name__ == '__main__':
audio_in = 'https://isv-data.oss-cn-hangzhou.aliyuncs.com/ics/MaaS/ASR/test_audio/asr_example_zh.wav'
output_dir = None
- inference_pipline = pipeline(
+ inference_pipeline = pipeline(
task=Tasks.auto_speech_recognition,
model="damo/speech_paraformerbert_asr_nat-zh-cn-16k-aishell1-vocab4234-pytorch",
output_dir=output_dir,
)
- rec_result = inference_pipline(audio_in=audio_in)
+ rec_result = inference_pipeline(audio_in=audio_in)
print(rec_result)
diff --git a/egs_modelscope/asr/paraformerbert/speech_paraformerbert_asr_nat-zh-cn-16k-aishell2-vocab5212-pytorch/infer.py b/egs_modelscope/asr/paraformerbert/speech_paraformerbert_asr_nat-zh-cn-16k-aishell2-vocab5212-pytorch/infer.py
index 83d6805..63bed40 100644
--- a/egs_modelscope/asr/paraformerbert/speech_paraformerbert_asr_nat-zh-cn-16k-aishell2-vocab5212-pytorch/infer.py
+++ b/egs_modelscope/asr/paraformerbert/speech_paraformerbert_asr_nat-zh-cn-16k-aishell2-vocab5212-pytorch/infer.py
@@ -4,10 +4,10 @@
if __name__ == "__main__":
audio_in = "https://modelscope.oss-cn-beijing.aliyuncs.com/test/audios/asr_example.wav"
output_dir = "./results"
- inference_pipline = pipeline(
+ inference_pipeline = pipeline(
task=Tasks.auto_speech_recognition,
model="damo/speech_paraformerbert_asr_nat-zh-cn-16k-aishell2-vocab5212-pytorch",
output_dir=output_dir,
)
- rec_result = inference_pipline(audio_in=audio_in)
+ rec_result = inference_pipeline(audio_in=audio_in)
print(rec_result)
diff --git a/egs_modelscope/asr/uniasr/speech_UniASR_asr_2pass-cantonese-CHS-16k-common-vocab1468-tensorflow1-offline/infer.py b/egs_modelscope/asr/uniasr/speech_UniASR_asr_2pass-cantonese-CHS-16k-common-vocab1468-tensorflow1-offline/infer.py
index c151149..862f881 100644
--- a/egs_modelscope/asr/uniasr/speech_UniASR_asr_2pass-cantonese-CHS-16k-common-vocab1468-tensorflow1-offline/infer.py
+++ b/egs_modelscope/asr/uniasr/speech_UniASR_asr_2pass-cantonese-CHS-16k-common-vocab1468-tensorflow1-offline/infer.py
@@ -4,10 +4,10 @@
if __name__ == "__main__":
audio_in = "https://isv-data.oss-cn-hangzhou.aliyuncs.com/ics/MaaS/ASR/test_audio/asr_example_cantonese-CHS.wav"
output_dir = "./results"
- inference_pipline = pipeline(
+ inference_pipeline = pipeline(
task=Tasks.auto_speech_recognition,
model="damo/speech_UniASR_asr_2pass-cantonese-CHS-16k-common-vocab1468-tensorflow1-online",
output_dir=output_dir,
)
- rec_result = inference_pipline(audio_in=audio_in, param_dict={"decoding_model":"offline"})
+ rec_result = inference_pipeline(audio_in=audio_in, param_dict={"decoding_model":"offline"})
print(rec_result)
diff --git a/egs_modelscope/asr/uniasr/speech_UniASR_asr_2pass-cantonese-CHS-16k-common-vocab1468-tensorflow1-online/infer.py b/egs_modelscope/asr/uniasr/speech_UniASR_asr_2pass-cantonese-CHS-16k-common-vocab1468-tensorflow1-online/infer.py
index ac73adf..d4f8d76 100644
--- a/egs_modelscope/asr/uniasr/speech_UniASR_asr_2pass-cantonese-CHS-16k-common-vocab1468-tensorflow1-online/infer.py
+++ b/egs_modelscope/asr/uniasr/speech_UniASR_asr_2pass-cantonese-CHS-16k-common-vocab1468-tensorflow1-online/infer.py
@@ -4,10 +4,10 @@
if __name__ == "__main__":
audio_in = "https://isv-data.oss-cn-hangzhou.aliyuncs.com/ics/MaaS/ASR/test_audio/asr_example_cantonese-CHS.wav"
output_dir = "./results"
- inference_pipline = pipeline(
+ inference_pipeline = pipeline(
task=Tasks.auto_speech_recognition,
model="damo/speech_UniASR_asr_2pass-cantonese-CHS-16k-common-vocab1468-tensorflow1-online",
output_dir=output_dir,
)
- rec_result = inference_pipline(audio_in=audio_in, param_dict={"decoding_model":"normal"})
+ rec_result = inference_pipeline(audio_in=audio_in, param_dict={"decoding_model":"normal"})
print(rec_result)
diff --git a/egs_modelscope/asr/uniasr/speech_UniASR_asr_2pass-cn-dialect-16k-vocab8358-tensorflow1-offline/infer.py b/egs_modelscope/asr/uniasr/speech_UniASR_asr_2pass-cn-dialect-16k-vocab8358-tensorflow1-offline/infer.py
index 227f4bf..347d316 100644
--- a/egs_modelscope/asr/uniasr/speech_UniASR_asr_2pass-cn-dialect-16k-vocab8358-tensorflow1-offline/infer.py
+++ b/egs_modelscope/asr/uniasr/speech_UniASR_asr_2pass-cn-dialect-16k-vocab8358-tensorflow1-offline/infer.py
@@ -4,11 +4,11 @@
if __name__ == '__main__':
audio_in = 'https://isv-data.oss-cn-hangzhou.aliyuncs.com/ics/MaaS/ASR/test_audio/asr_example_zh.wav'
output_dir = None
- inference_pipline = pipeline(
+ inference_pipeline = pipeline(
task=Tasks.auto_speech_recognition,
model="damo/speech_UniASR_asr_2pass-cn-dialect-16k-vocab8358-tensorflow1-offline",
output_dir=output_dir,
)
- rec_result = inference_pipline(audio_in=audio_in)
+ rec_result = inference_pipeline(audio_in=audio_in)
print(rec_result)
diff --git a/egs_modelscope/asr/uniasr/speech_UniASR_asr_2pass-cn-dialect-16k-vocab8358-tensorflow1-online/infer.py b/egs_modelscope/asr/uniasr/speech_UniASR_asr_2pass-cn-dialect-16k-vocab8358-tensorflow1-online/infer.py
index 74d9764..936d6d7 100644
--- a/egs_modelscope/asr/uniasr/speech_UniASR_asr_2pass-cn-dialect-16k-vocab8358-tensorflow1-online/infer.py
+++ b/egs_modelscope/asr/uniasr/speech_UniASR_asr_2pass-cn-dialect-16k-vocab8358-tensorflow1-online/infer.py
@@ -4,11 +4,11 @@
if __name__ == '__main__':
audio_in = 'https://isv-data.oss-cn-hangzhou.aliyuncs.com/ics/MaaS/ASR/test_audio/asr_example_zh.wav'
output_dir = None
- inference_pipline = pipeline(
+ inference_pipeline = pipeline(
task=Tasks.auto_speech_recognition,
model="damo/speech_UniASR_asr_2pass-cn-dialect-16k-vocab8358-tensorflow1-online",
output_dir=output_dir,
)
- rec_result = inference_pipline(audio_in=audio_in)
+ rec_result = inference_pipeline(audio_in=audio_in)
print(rec_result)
diff --git a/egs_modelscope/asr/uniasr/speech_UniASR_asr_2pass-de-16k-common-vocab3690-tensorflow1-offline/infer.py b/egs_modelscope/asr/uniasr/speech_UniASR_asr_2pass-de-16k-common-vocab3690-tensorflow1-offline/infer.py
index 5ace7e4..f82c1f4 100644
--- a/egs_modelscope/asr/uniasr/speech_UniASR_asr_2pass-de-16k-common-vocab3690-tensorflow1-offline/infer.py
+++ b/egs_modelscope/asr/uniasr/speech_UniASR_asr_2pass-de-16k-common-vocab3690-tensorflow1-offline/infer.py
@@ -4,10 +4,10 @@
if __name__ == "__main__":
audio_in = "https://isv-data.oss-cn-hangzhou.aliyuncs.com/ics/MaaS/ASR/test_audio/asr_example_de.wav"
output_dir = "./results"
- inference_pipline = pipeline(
+ inference_pipeline = pipeline(
task=Tasks.auto_speech_recognition,
model="damo/speech_UniASR_asr_2pass-de-16k-common-vocab3690-tensorflow1-offline",
output_dir=output_dir,
)
- rec_result = inference_pipline(audio_in=audio_in, param_dict={"decoding_model":"offline"})
+ rec_result = inference_pipeline(audio_in=audio_in, param_dict={"decoding_model":"offline"})
print(rec_result)
diff --git a/egs_modelscope/asr/uniasr/speech_UniASR_asr_2pass-de-16k-common-vocab3690-tensorflow1-online/infer.py b/egs_modelscope/asr/uniasr/speech_UniASR_asr_2pass-de-16k-common-vocab3690-tensorflow1-online/infer.py
index f8d91b8..48b4807 100644
--- a/egs_modelscope/asr/uniasr/speech_UniASR_asr_2pass-de-16k-common-vocab3690-tensorflow1-online/infer.py
+++ b/egs_modelscope/asr/uniasr/speech_UniASR_asr_2pass-de-16k-common-vocab3690-tensorflow1-online/infer.py
@@ -4,10 +4,10 @@
if __name__ == "__main__":
audio_in = "https://isv-data.oss-cn-hangzhou.aliyuncs.com/ics/MaaS/ASR/test_audio/asr_example_de.wav"
output_dir = "./results"
- inference_pipline = pipeline(
+ inference_pipeline = pipeline(
task=Tasks.auto_speech_recognition,
model="damo/speech_UniASR_asr_2pass-de-16k-common-vocab3690-tensorflow1-online",
output_dir=output_dir,
)
- rec_result = inference_pipline(audio_in=audio_in, param_dict={"decoding_model":"normal"})
+ rec_result = inference_pipeline(audio_in=audio_in, param_dict={"decoding_model":"normal"})
print(rec_result)
diff --git a/egs_modelscope/asr/uniasr/speech_UniASR_asr_2pass-en-16k-common-vocab1080-tensorflow1-offline/infer.py b/egs_modelscope/asr/uniasr/speech_UniASR_asr_2pass-en-16k-common-vocab1080-tensorflow1-offline/infer.py
index 49b884b..98f31b6 100644
--- a/egs_modelscope/asr/uniasr/speech_UniASR_asr_2pass-en-16k-common-vocab1080-tensorflow1-offline/infer.py
+++ b/egs_modelscope/asr/uniasr/speech_UniASR_asr_2pass-en-16k-common-vocab1080-tensorflow1-offline/infer.py
@@ -4,10 +4,10 @@
if __name__ == "__main__":
audio_in = "https://isv-data.oss-cn-hangzhou.aliyuncs.com/ics/MaaS/ASR/test_audio/asr_example_en.wav"
output_dir = "./results"
- inference_pipline = pipeline(
+ inference_pipeline = pipeline(
task=Tasks.auto_speech_recognition,
model="damo/speech_UniASR_asr_2pass-en-16k-common-vocab1080-tensorflow1-offline",
output_dir=output_dir,
)
- rec_result = inference_pipline(audio_in=audio_in, param_dict={"decoding_model":"offline"})
+ rec_result = inference_pipeline(audio_in=audio_in, param_dict={"decoding_model":"offline"})
print(rec_result)
diff --git a/egs_modelscope/asr/uniasr/speech_UniASR_asr_2pass-en-16k-common-vocab1080-tensorflow1-online/infer.py b/egs_modelscope/asr/uniasr/speech_UniASR_asr_2pass-en-16k-common-vocab1080-tensorflow1-online/infer.py
index 57a3afd..423c503 100644
--- a/egs_modelscope/asr/uniasr/speech_UniASR_asr_2pass-en-16k-common-vocab1080-tensorflow1-online/infer.py
+++ b/egs_modelscope/asr/uniasr/speech_UniASR_asr_2pass-en-16k-common-vocab1080-tensorflow1-online/infer.py
@@ -4,10 +4,10 @@
if __name__ == "__main__":
audio_in = "https://isv-data.oss-cn-hangzhou.aliyuncs.com/ics/MaaS/ASR/test_audio/asr_example_en.wav"
output_dir = "./results"
- inference_pipline = pipeline(
+ inference_pipeline = pipeline(
task=Tasks.auto_speech_recognition,
model="damo/speech_UniASR_asr_2pass-en-16k-common-vocab1080-tensorflow1-online",
output_dir=output_dir,
)
- rec_result = inference_pipline(audio_in=audio_in, param_dict={"decoding_model":"normal"})
+ rec_result = inference_pipeline(audio_in=audio_in, param_dict={"decoding_model":"normal"})
print(rec_result)
diff --git a/egs_modelscope/asr/uniasr/speech_UniASR_asr_2pass-es-16k-common-vocab3445-tensorflow1-offline/infer.py b/egs_modelscope/asr/uniasr/speech_UniASR_asr_2pass-es-16k-common-vocab3445-tensorflow1-offline/infer.py
index 510f008..75e22a0 100644
--- a/egs_modelscope/asr/uniasr/speech_UniASR_asr_2pass-es-16k-common-vocab3445-tensorflow1-offline/infer.py
+++ b/egs_modelscope/asr/uniasr/speech_UniASR_asr_2pass-es-16k-common-vocab3445-tensorflow1-offline/infer.py
@@ -4,10 +4,10 @@
if __name__ == "__main__":
audio_in = "https://isv-data.oss-cn-hangzhou.aliyuncs.com/ics/MaaS/ASR/test_audio/asr_example_es.wav"
output_dir = "./results"
- inference_pipline = pipeline(
+ inference_pipeline = pipeline(
task=Tasks.auto_speech_recognition,
model="damo/speech_UniASR_asr_2pass-es-16k-common-vocab3445-tensorflow1-offline",
output_dir=output_dir,
)
- rec_result = inference_pipline(audio_in=audio_in, param_dict={"decoding_model":"offline"})
+ rec_result = inference_pipeline(audio_in=audio_in, param_dict={"decoding_model":"offline"})
print(rec_result)
diff --git a/egs_modelscope/asr/uniasr/speech_UniASR_asr_2pass-es-16k-common-vocab3445-tensorflow1-online/infer.py b/egs_modelscope/asr/uniasr/speech_UniASR_asr_2pass-es-16k-common-vocab3445-tensorflow1-online/infer.py
index 2ec5940..cb1b4fa 100644
--- a/egs_modelscope/asr/uniasr/speech_UniASR_asr_2pass-es-16k-common-vocab3445-tensorflow1-online/infer.py
+++ b/egs_modelscope/asr/uniasr/speech_UniASR_asr_2pass-es-16k-common-vocab3445-tensorflow1-online/infer.py
@@ -4,10 +4,10 @@
if __name__ == "__main__":
audio_in = "https://isv-data.oss-cn-hangzhou.aliyuncs.com/ics/MaaS/ASR/test_audio/asr_example_es.wav"
output_dir = "./results"
- inference_pipline = pipeline(
+ inference_pipeline = pipeline(
task=Tasks.auto_speech_recognition,
model="damo/speech_UniASR_asr_2pass-es-16k-common-vocab3445-tensorflow1-online",
output_dir=output_dir,
)
- rec_result = inference_pipline(audio_in=audio_in, param_dict={"decoding_model":"normal"})
+ rec_result = inference_pipeline(audio_in=audio_in, param_dict={"decoding_model":"normal"})
print(rec_result)
diff --git a/egs_modelscope/asr/uniasr/speech_UniASR_asr_2pass-fa-16k-common-vocab1257-pytorch-offline/infer.py b/egs_modelscope/asr/uniasr/speech_UniASR_asr_2pass-fa-16k-common-vocab1257-pytorch-offline/infer.py
index 040265d..e6c39c2 100644
--- a/egs_modelscope/asr/uniasr/speech_UniASR_asr_2pass-fa-16k-common-vocab1257-pytorch-offline/infer.py
+++ b/egs_modelscope/asr/uniasr/speech_UniASR_asr_2pass-fa-16k-common-vocab1257-pytorch-offline/infer.py
@@ -16,14 +16,14 @@
os.environ['CUDA_VISIBLE_DEVICES'] = str(gpu_list[gpu_id])
else:
os.environ['CUDA_VISIBLE_DEVICES'] = str(gpu_id)
- inference_pipline = pipeline(
+ inference_pipeline = pipeline(
task=Tasks.auto_speech_recognition,
model="damo/speech_UniASR_asr_2pass-fa-16k-common-vocab1257-pytorch-offline",
output_dir=output_dir_job,
batch_size=1
)
audio_in = os.path.join(split_dir, "wav.{}.scp".format(idx))
- inference_pipline(audio_in=audio_in, param_dict={"decoding_model":"offline"})
+ inference_pipeline(audio_in=audio_in, param_dict={"decoding_model":"offline"})
def modelscope_infer(params):
diff --git a/egs_modelscope/asr/uniasr/speech_UniASR_asr_2pass-fa-16k-common-vocab1257-pytorch-online/infer.py b/egs_modelscope/asr/uniasr/speech_UniASR_asr_2pass-fa-16k-common-vocab1257-pytorch-online/infer.py
index 055e4eb..124d5ed 100644
--- a/egs_modelscope/asr/uniasr/speech_UniASR_asr_2pass-fa-16k-common-vocab1257-pytorch-online/infer.py
+++ b/egs_modelscope/asr/uniasr/speech_UniASR_asr_2pass-fa-16k-common-vocab1257-pytorch-online/infer.py
@@ -16,14 +16,14 @@
os.environ['CUDA_VISIBLE_DEVICES'] = str(gpu_list[gpu_id])
else:
os.environ['CUDA_VISIBLE_DEVICES'] = str(gpu_id)
- inference_pipline = pipeline(
+ inference_pipeline = pipeline(
task=Tasks.auto_speech_recognition,
model="damo/speech_UniASR_asr_2pass-fa-16k-common-vocab1257-pytorch-online",
output_dir=output_dir_job,
batch_size=1
)
audio_in = os.path.join(split_dir, "wav.{}.scp".format(idx))
- inference_pipline(audio_in=audio_in, param_dict={"decoding_model":"normal"})
+ inference_pipeline(audio_in=audio_in, param_dict={"decoding_model":"normal"})
def modelscope_infer(params):
diff --git a/egs_modelscope/asr/uniasr/speech_UniASR_asr_2pass-fr-16k-common-vocab3472-tensorflow1-offline/infer.py b/egs_modelscope/asr/uniasr/speech_UniASR_asr_2pass-fr-16k-common-vocab3472-tensorflow1-offline/infer.py
index 6aedeea..627d132 100644
--- a/egs_modelscope/asr/uniasr/speech_UniASR_asr_2pass-fr-16k-common-vocab3472-tensorflow1-offline/infer.py
+++ b/egs_modelscope/asr/uniasr/speech_UniASR_asr_2pass-fr-16k-common-vocab3472-tensorflow1-offline/infer.py
@@ -4,10 +4,10 @@
if __name__ == "__main__":
audio_in = "https://isv-data.oss-cn-hangzhou.aliyuncs.com/ics/MaaS/ASR/test_audio/asr_example_fr.wav"
output_dir = "./results"
- inference_pipline = pipeline(
+ inference_pipeline = pipeline(
task=Tasks.auto_speech_recognition,
model="damo/speech_UniASR_asr_2pass-fr-16k-common-vocab3472-tensorflow1-offline",
output_dir=output_dir,
)
- rec_result = inference_pipline(audio_in=audio_in, param_dict={"decoding_model":"offline"})
+ rec_result = inference_pipeline(audio_in=audio_in, param_dict={"decoding_model":"offline"})
print(rec_result)
diff --git a/egs_modelscope/asr/uniasr/speech_UniASR_asr_2pass-fr-16k-common-vocab3472-tensorflow1-online/infer.py b/egs_modelscope/asr/uniasr/speech_UniASR_asr_2pass-fr-16k-common-vocab3472-tensorflow1-online/infer.py
index 2f3e833..305d990 100644
--- a/egs_modelscope/asr/uniasr/speech_UniASR_asr_2pass-fr-16k-common-vocab3472-tensorflow1-online/infer.py
+++ b/egs_modelscope/asr/uniasr/speech_UniASR_asr_2pass-fr-16k-common-vocab3472-tensorflow1-online/infer.py
@@ -4,10 +4,10 @@
if __name__ == "__main__":
audio_in = "https://isv-data.oss-cn-hangzhou.aliyuncs.com/ics/MaaS/ASR/test_audio/asr_example_fr.wav"
output_dir = "./results"
- inference_pipline = pipeline(
+ inference_pipeline = pipeline(
task=Tasks.auto_speech_recognition,
model="damo/speech_UniASR_asr_2pass-fr-16k-common-vocab3472-tensorflow1-online",
output_dir=output_dir,
)
- rec_result = inference_pipline(audio_in=audio_in, param_dict={"decoding_model":"normal"})
+ rec_result = inference_pipeline(audio_in=audio_in, param_dict={"decoding_model":"normal"})
print(rec_result)
diff --git a/egs_modelscope/asr/uniasr/speech_UniASR_asr_2pass-he-16k-common-vocab1085-pytorch/infer.py b/egs_modelscope/asr/uniasr/speech_UniASR_asr_2pass-he-16k-common-vocab1085-pytorch/infer.py
index c54ab8c..e0d1a4d 100644
--- a/egs_modelscope/asr/uniasr/speech_UniASR_asr_2pass-he-16k-common-vocab1085-pytorch/infer.py
+++ b/egs_modelscope/asr/uniasr/speech_UniASR_asr_2pass-he-16k-common-vocab1085-pytorch/infer.py
@@ -4,10 +4,10 @@
if __name__ == "__main__":
audio_in = "https://isv-data.oss-cn-hangzhou.aliyuncs.com/ics/MaaS/ASR/test_audio/asr_example_he.wav"
output_dir = "./results"
- inference_pipline = pipeline(
+ inference_pipeline = pipeline(
task=Tasks.auto_speech_recognition,
model="damo/speech_UniASR_asr_2pass-he-16k-common-vocab1085-pytorch",
output_dir=output_dir,
)
- rec_result = inference_pipline(audio_in=audio_in, param_dict={"decoding_model":"offline"})
+ rec_result = inference_pipeline(audio_in=audio_in, param_dict={"decoding_model":"offline"})
print(rec_result)
diff --git a/egs_modelscope/asr/uniasr/speech_UniASR_asr_2pass-id-16k-common-vocab1067-tensorflow1-offline/infer.py b/egs_modelscope/asr/uniasr/speech_UniASR_asr_2pass-id-16k-common-vocab1067-tensorflow1-offline/infer.py
index 219c9ec..e53c37e 100644
--- a/egs_modelscope/asr/uniasr/speech_UniASR_asr_2pass-id-16k-common-vocab1067-tensorflow1-offline/infer.py
+++ b/egs_modelscope/asr/uniasr/speech_UniASR_asr_2pass-id-16k-common-vocab1067-tensorflow1-offline/infer.py
@@ -4,10 +4,10 @@
if __name__ == "__main__":
audio_in = "https://isv-data.oss-cn-hangzhou.aliyuncs.com/ics/MaaS/ASR/test_audio/asr_example_id.wav"
output_dir = "./results"
- inference_pipline = pipeline(
+ inference_pipeline = pipeline(
task=Tasks.auto_speech_recognition,
model="damo/speech_UniASR_asr_2pass-id-16k-common-vocab1067-tensorflow1-online",
output_dir=output_dir,
)
- rec_result = inference_pipline(audio_in=audio_in, param_dict={"decoding_model":"offline"})
+ rec_result = inference_pipeline(audio_in=audio_in, param_dict={"decoding_model":"offline"})
print(rec_result)
diff --git a/egs_modelscope/asr/uniasr/speech_UniASR_asr_2pass-id-16k-common-vocab1067-tensorflow1-online/infer.py b/egs_modelscope/asr/uniasr/speech_UniASR_asr_2pass-id-16k-common-vocab1067-tensorflow1-online/infer.py
index ad2671a..75ec783 100644
--- a/egs_modelscope/asr/uniasr/speech_UniASR_asr_2pass-id-16k-common-vocab1067-tensorflow1-online/infer.py
+++ b/egs_modelscope/asr/uniasr/speech_UniASR_asr_2pass-id-16k-common-vocab1067-tensorflow1-online/infer.py
@@ -4,10 +4,10 @@
if __name__ == "__main__":
audio_in = "https://isv-data.oss-cn-hangzhou.aliyuncs.com/ics/MaaS/ASR/test_audio/asr_example_id.wav"
output_dir = "./results"
- inference_pipline = pipeline(
+ inference_pipeline = pipeline(
task=Tasks.auto_speech_recognition,
model="damo/speech_UniASR_asr_2pass-id-16k-common-vocab1067-tensorflow1-online",
output_dir=output_dir,
)
- rec_result = inference_pipline(audio_in=audio_in, param_dict={"decoding_model":"normal"})
+ rec_result = inference_pipeline(audio_in=audio_in, param_dict={"decoding_model":"normal"})
print(rec_result)
diff --git a/egs_modelscope/asr/uniasr/speech_UniASR_asr_2pass-ja-16k-common-vocab93-tensorflow1-offline/infer.py b/egs_modelscope/asr/uniasr/speech_UniASR_asr_2pass-ja-16k-common-vocab93-tensorflow1-offline/infer.py
index 1a174bb..68cc41d 100644
--- a/egs_modelscope/asr/uniasr/speech_UniASR_asr_2pass-ja-16k-common-vocab93-tensorflow1-offline/infer.py
+++ b/egs_modelscope/asr/uniasr/speech_UniASR_asr_2pass-ja-16k-common-vocab93-tensorflow1-offline/infer.py
@@ -4,10 +4,10 @@
if __name__ == "__main__":
audio_in = "https://isv-data.oss-cn-hangzhou.aliyuncs.com/ics/MaaS/ASR/test_audio/asr_example_ja.wav"
output_dir = "./results"
- inference_pipline = pipeline(
+ inference_pipeline = pipeline(
task=Tasks.auto_speech_recognition,
model="damo/speech_UniASR_asr_2pass-ja-16k-common-vocab93-tensorflow1-offline",
output_dir=output_dir,
)
- rec_result = inference_pipline(audio_in=audio_in, param_dict={"decoding_model":"offline"})
+ rec_result = inference_pipeline(audio_in=audio_in, param_dict={"decoding_model":"offline"})
print(rec_result)
diff --git a/egs_modelscope/asr/uniasr/speech_UniASR_asr_2pass-ja-16k-common-vocab93-tensorflow1-online/infer.py b/egs_modelscope/asr/uniasr/speech_UniASR_asr_2pass-ja-16k-common-vocab93-tensorflow1-online/infer.py
index f15bc2d..a741e18 100644
--- a/egs_modelscope/asr/uniasr/speech_UniASR_asr_2pass-ja-16k-common-vocab93-tensorflow1-online/infer.py
+++ b/egs_modelscope/asr/uniasr/speech_UniASR_asr_2pass-ja-16k-common-vocab93-tensorflow1-online/infer.py
@@ -4,10 +4,10 @@
if __name__ == "__main__":
audio_in = "https://isv-data.oss-cn-hangzhou.aliyuncs.com/ics/MaaS/ASR/test_audio/asr_example_ja.wav"
output_dir = "./results"
- inference_pipline = pipeline(
+ inference_pipeline = pipeline(
task=Tasks.auto_speech_recognition,
model="damo/speech_UniASR_asr_2pass-ja-16k-common-vocab93-tensorflow1-online",
output_dir=output_dir,
)
- rec_result = inference_pipline(audio_in=audio_in, param_dict={"decoding_model":"normal"})
+ rec_result = inference_pipeline(audio_in=audio_in, param_dict={"decoding_model":"normal"})
print(rec_result)
diff --git a/egs_modelscope/asr/uniasr/speech_UniASR_asr_2pass-ko-16k-common-vocab6400-tensorflow1-offline/infer.py b/egs_modelscope/asr/uniasr/speech_UniASR_asr_2pass-ko-16k-common-vocab6400-tensorflow1-offline/infer.py
index 618b3f6..b87bcbb 100644
--- a/egs_modelscope/asr/uniasr/speech_UniASR_asr_2pass-ko-16k-common-vocab6400-tensorflow1-offline/infer.py
+++ b/egs_modelscope/asr/uniasr/speech_UniASR_asr_2pass-ko-16k-common-vocab6400-tensorflow1-offline/infer.py
@@ -4,10 +4,10 @@
if __name__ == "__main__":
audio_in = "https://isv-data.oss-cn-hangzhou.aliyuncs.com/ics/MaaS/ASR/test_audio/asr_example_ko.wav"
output_dir = "./results"
- inference_pipline = pipeline(
+ inference_pipeline = pipeline(
task=Tasks.auto_speech_recognition,
model="damo/speech_UniASR_asr_2pass-ko-16k-common-vocab6400-tensorflow1-offline",
output_dir=output_dir,
)
- rec_result = inference_pipline(audio_in=audio_in, param_dict={"decoding_model":"offline"})
+ rec_result = inference_pipeline(audio_in=audio_in, param_dict={"decoding_model":"offline"})
print(rec_result)
diff --git a/egs_modelscope/asr/uniasr/speech_UniASR_asr_2pass-ko-16k-common-vocab6400-tensorflow1-online/infer.py b/egs_modelscope/asr/uniasr/speech_UniASR_asr_2pass-ko-16k-common-vocab6400-tensorflow1-online/infer.py
index 135e8f8..9be791c 100644
--- a/egs_modelscope/asr/uniasr/speech_UniASR_asr_2pass-ko-16k-common-vocab6400-tensorflow1-online/infer.py
+++ b/egs_modelscope/asr/uniasr/speech_UniASR_asr_2pass-ko-16k-common-vocab6400-tensorflow1-online/infer.py
@@ -4,10 +4,10 @@
if __name__ == "__main__":
audio_in = "https://isv-data.oss-cn-hangzhou.aliyuncs.com/ics/MaaS/ASR/test_audio/asr_example_ko.wav"
output_dir = "./results"
- inference_pipline = pipeline(
+ inference_pipeline = pipeline(
task=Tasks.auto_speech_recognition,
model="damo/speech_UniASR_asr_2pass-ko-16k-common-vocab6400-tensorflow1-online",
output_dir=output_dir,
)
- rec_result = inference_pipline(audio_in=audio_in, param_dict={"decoding_model":"normal"})
+ rec_result = inference_pipeline(audio_in=audio_in, param_dict={"decoding_model":"normal"})
print(rec_result)
diff --git a/egs_modelscope/asr/uniasr/speech_UniASR_asr_2pass-my-16k-common-vocab696-pytorch/infer.py b/egs_modelscope/asr/uniasr/speech_UniASR_asr_2pass-my-16k-common-vocab696-pytorch/infer.py
index cfd869f..b3a9058 100644
--- a/egs_modelscope/asr/uniasr/speech_UniASR_asr_2pass-my-16k-common-vocab696-pytorch/infer.py
+++ b/egs_modelscope/asr/uniasr/speech_UniASR_asr_2pass-my-16k-common-vocab696-pytorch/infer.py
@@ -4,10 +4,10 @@
if __name__ == "__main__":
audio_in = "https://isv-data.oss-cn-hangzhou.aliyuncs.com/ics/MaaS/ASR/test_audio/asr_example_my.wav"
output_dir = "./results"
- inference_pipline = pipeline(
+ inference_pipeline = pipeline(
task=Tasks.auto_speech_recognition,
model="damo/speech_UniASR_asr_2pass-my-16k-common-vocab696-pytorch",
output_dir=output_dir,
)
- rec_result = inference_pipline(audio_in=audio_in, param_dict={"decoding_model":"offline"})
+ rec_result = inference_pipeline(audio_in=audio_in, param_dict={"decoding_model":"offline"})
print(rec_result)
diff --git a/egs_modelscope/asr/uniasr/speech_UniASR_asr_2pass-pt-16k-common-vocab1617-tensorflow1-offline/infer.py b/egs_modelscope/asr/uniasr/speech_UniASR_asr_2pass-pt-16k-common-vocab1617-tensorflow1-offline/infer.py
index 2dcb663..4a43e7c 100644
--- a/egs_modelscope/asr/uniasr/speech_UniASR_asr_2pass-pt-16k-common-vocab1617-tensorflow1-offline/infer.py
+++ b/egs_modelscope/asr/uniasr/speech_UniASR_asr_2pass-pt-16k-common-vocab1617-tensorflow1-offline/infer.py
@@ -4,10 +4,10 @@
if __name__ == "__main__":
audio_in = "https://isv-data.oss-cn-hangzhou.aliyuncs.com/ics/MaaS/ASR/test_audio/asr_example_pt.wav"
output_dir = "./results"
- inference_pipline = pipeline(
+ inference_pipeline = pipeline(
task=Tasks.auto_speech_recognition,
model="damo/speech_UniASR_asr_2pass-pt-16k-common-vocab1617-tensorflow1-offline",
output_dir=output_dir,
)
- rec_result = inference_pipline(audio_in=audio_in, param_dict={"decoding_model":"offline"})
+ rec_result = inference_pipeline(audio_in=audio_in, param_dict={"decoding_model":"offline"})
print(rec_result)
diff --git a/egs_modelscope/asr/uniasr/speech_UniASR_asr_2pass-pt-16k-common-vocab1617-tensorflow1-online/infer.py b/egs_modelscope/asr/uniasr/speech_UniASR_asr_2pass-pt-16k-common-vocab1617-tensorflow1-online/infer.py
index aff2a9a..7029fd9 100644
--- a/egs_modelscope/asr/uniasr/speech_UniASR_asr_2pass-pt-16k-common-vocab1617-tensorflow1-online/infer.py
+++ b/egs_modelscope/asr/uniasr/speech_UniASR_asr_2pass-pt-16k-common-vocab1617-tensorflow1-online/infer.py
@@ -4,10 +4,10 @@
if __name__ == "__main__":
audio_in = "https://isv-data.oss-cn-hangzhou.aliyuncs.com/ics/MaaS/ASR/test_audio/asr_example_pt.wav"
output_dir = "./results"
- inference_pipline = pipeline(
+ inference_pipeline = pipeline(
task=Tasks.auto_speech_recognition,
model="damo/speech_UniASR_asr_2pass-pt-16k-common-vocab1617-tensorflow1-online",
output_dir=output_dir,
)
- rec_result = inference_pipline(audio_in=audio_in, param_dict={"decoding_model":"normal"})
+ rec_result = inference_pipeline(audio_in=audio_in, param_dict={"decoding_model":"normal"})
print(rec_result)
diff --git a/egs_modelscope/asr/uniasr/speech_UniASR_asr_2pass-ru-16k-common-vocab1664-tensorflow1-offline/infer.py b/egs_modelscope/asr/uniasr/speech_UniASR_asr_2pass-ru-16k-common-vocab1664-tensorflow1-offline/infer.py
index 95f447d..3c9d364 100644
--- a/egs_modelscope/asr/uniasr/speech_UniASR_asr_2pass-ru-16k-common-vocab1664-tensorflow1-offline/infer.py
+++ b/egs_modelscope/asr/uniasr/speech_UniASR_asr_2pass-ru-16k-common-vocab1664-tensorflow1-offline/infer.py
@@ -4,10 +4,10 @@
if __name__ == "__main__":
audio_in = "https://isv-data.oss-cn-hangzhou.aliyuncs.com/ics/MaaS/ASR/test_audio/asr_example_ru.wav"
output_dir = "./results"
- inference_pipline = pipeline(
+ inference_pipeline = pipeline(
task=Tasks.auto_speech_recognition,
model="damo/speech_UniASR_asr_2pass-ru-16k-common-vocab1664-tensorflow1-offline",
output_dir=output_dir,
)
- rec_result = inference_pipline(audio_in=audio_in, param_dict={"decoding_model":"offline"})
+ rec_result = inference_pipeline(audio_in=audio_in, param_dict={"decoding_model":"offline"})
print(rec_result)
diff --git a/egs_modelscope/asr/uniasr/speech_UniASR_asr_2pass-ru-16k-common-vocab1664-tensorflow1-online/infer.py b/egs_modelscope/asr/uniasr/speech_UniASR_asr_2pass-ru-16k-common-vocab1664-tensorflow1-online/infer.py
index 88c06b4..95da479 100644
--- a/egs_modelscope/asr/uniasr/speech_UniASR_asr_2pass-ru-16k-common-vocab1664-tensorflow1-online/infer.py
+++ b/egs_modelscope/asr/uniasr/speech_UniASR_asr_2pass-ru-16k-common-vocab1664-tensorflow1-online/infer.py
@@ -4,10 +4,10 @@
if __name__ == "__main__":
audio_in = "https://isv-data.oss-cn-hangzhou.aliyuncs.com/ics/MaaS/ASR/test_audio/asr_example_ru.wav"
output_dir = "./results"
- inference_pipline = pipeline(
+ inference_pipeline = pipeline(
task=Tasks.auto_speech_recognition,
model="damo/speech_UniASR_asr_2pass-ru-16k-common-vocab1664-tensorflow1-online",
output_dir=output_dir,
)
- rec_result = inference_pipline(audio_in=audio_in, param_dict={"decoding_model":"normal"})
+ rec_result = inference_pipeline(audio_in=audio_in, param_dict={"decoding_model":"normal"})
print(rec_result)
diff --git a/egs_modelscope/asr/uniasr/speech_UniASR_asr_2pass-ur-16k-common-vocab877-pytorch/infer.py b/egs_modelscope/asr/uniasr/speech_UniASR_asr_2pass-ur-16k-common-vocab877-pytorch/infer.py
index e8c5524..04b02fe 100644
--- a/egs_modelscope/asr/uniasr/speech_UniASR_asr_2pass-ur-16k-common-vocab877-pytorch/infer.py
+++ b/egs_modelscope/asr/uniasr/speech_UniASR_asr_2pass-ur-16k-common-vocab877-pytorch/infer.py
@@ -4,10 +4,10 @@
if __name__ == "__main__":
audio_in = "https://isv-data.oss-cn-hangzhou.aliyuncs.com/ics/MaaS/ASR/test_audio/asr_example_ur.wav"
output_dir = "./results"
- inference_pipline = pipeline(
+ inference_pipeline = pipeline(
task=Tasks.auto_speech_recognition,
model="damo/speech_UniASR_asr_2pass-ur-16k-common-vocab877-pytorch",
output_dir=output_dir,
)
- rec_result = inference_pipline(audio_in=audio_in, param_dict={"decoding_model":"offline"})
+ rec_result = inference_pipeline(audio_in=audio_in, param_dict={"decoding_model":"offline"})
print(rec_result)
diff --git a/egs_modelscope/asr/uniasr/speech_UniASR_asr_2pass-vi-16k-common-vocab1001-pytorch-offline/infer.py b/egs_modelscope/asr/uniasr/speech_UniASR_asr_2pass-vi-16k-common-vocab1001-pytorch-offline/infer.py
index 9472104..4218f3d 100644
--- a/egs_modelscope/asr/uniasr/speech_UniASR_asr_2pass-vi-16k-common-vocab1001-pytorch-offline/infer.py
+++ b/egs_modelscope/asr/uniasr/speech_UniASR_asr_2pass-vi-16k-common-vocab1001-pytorch-offline/infer.py
@@ -4,10 +4,10 @@
if __name__ == "__main__":
audio_in = "https://isv-data.oss-cn-hangzhou.aliyuncs.com/ics/MaaS/ASR/test_audio/asr_example_vi.wav"
output_dir = "./results"
- inference_pipline = pipeline(
+ inference_pipeline = pipeline(
task=Tasks.auto_speech_recognition,
model="damo/speech_UniASR_asr_2pass-vi-16k-common-vocab1001-pytorch-offline",
output_dir=output_dir,
)
- rec_result = inference_pipline(audio_in=audio_in, param_dict={"decoding_model":"offline"})
+ rec_result = inference_pipeline(audio_in=audio_in, param_dict={"decoding_model":"offline"})
print(rec_result)
diff --git a/egs_modelscope/asr/uniasr/speech_UniASR_asr_2pass-vi-16k-common-vocab1001-pytorch-online/infer.py b/egs_modelscope/asr/uniasr/speech_UniASR_asr_2pass-vi-16k-common-vocab1001-pytorch-online/infer.py
index 4a844fc..355e412 100644
--- a/egs_modelscope/asr/uniasr/speech_UniASR_asr_2pass-vi-16k-common-vocab1001-pytorch-online/infer.py
+++ b/egs_modelscope/asr/uniasr/speech_UniASR_asr_2pass-vi-16k-common-vocab1001-pytorch-online/infer.py
@@ -4,10 +4,10 @@
if __name__ == "__main__":
audio_in = "https://isv-data.oss-cn-hangzhou.aliyuncs.com/ics/MaaS/ASR/test_audio/asr_example_vi.wav"
output_dir = "./results"
- inference_pipline = pipeline(
+ inference_pipeline = pipeline(
task=Tasks.auto_speech_recognition,
model="damo/speech_UniASR_asr_2pass-vi-16k-common-vocab1001-pytorch-online",
output_dir=output_dir,
)
- rec_result = inference_pipline(audio_in=audio_in, param_dict={"decoding_model":"normal"})
+ rec_result = inference_pipeline(audio_in=audio_in, param_dict={"decoding_model":"normal"})
print(rec_result)
diff --git a/egs_modelscope/asr/uniasr/speech_UniASR_asr_2pass-zh-cn-16k-common-vocab8358-tensorflow1-offline/infer.py b/egs_modelscope/asr/uniasr/speech_UniASR_asr_2pass-zh-cn-16k-common-vocab8358-tensorflow1-offline/infer.py
index 40686ac..3520989 100644
--- a/egs_modelscope/asr/uniasr/speech_UniASR_asr_2pass-zh-cn-16k-common-vocab8358-tensorflow1-offline/infer.py
+++ b/egs_modelscope/asr/uniasr/speech_UniASR_asr_2pass-zh-cn-16k-common-vocab8358-tensorflow1-offline/infer.py
@@ -4,11 +4,11 @@
if __name__ == '__main__':
audio_in = 'https://isv-data.oss-cn-hangzhou.aliyuncs.com/ics/MaaS/ASR/test_audio/asr_example_zh.wav'
output_dir = None
- inference_pipline = pipeline(
+ inference_pipeline = pipeline(
task=Tasks.auto_speech_recognition,
model="damo/speech_UniASR_asr_2pass-zh-cn-16k-common-vocab8358-tensorflow1-offline",
output_dir=output_dir,
)
- rec_result = inference_pipline(audio_in=audio_in)
+ rec_result = inference_pipeline(audio_in=audio_in)
print(rec_result)
diff --git a/egs_modelscope/asr/uniasr/speech_UniASR_asr_2pass-zh-cn-16k-common-vocab8358-tensorflow1-online/infer.py b/egs_modelscope/asr/uniasr/speech_UniASR_asr_2pass-zh-cn-16k-common-vocab8358-tensorflow1-online/infer.py
index dfe934d..a3e2a00 100644
--- a/egs_modelscope/asr/uniasr/speech_UniASR_asr_2pass-zh-cn-16k-common-vocab8358-tensorflow1-online/infer.py
+++ b/egs_modelscope/asr/uniasr/speech_UniASR_asr_2pass-zh-cn-16k-common-vocab8358-tensorflow1-online/infer.py
@@ -4,11 +4,11 @@
if __name__ == '__main__':
audio_in = 'https://isv-data.oss-cn-hangzhou.aliyuncs.com/ics/MaaS/ASR/test_audio/asr_example_zh.wav'
output_dir = None
- inference_pipline = pipeline(
+ inference_pipeline = pipeline(
task=Tasks.auto_speech_recognition,
model="damo/speech_UniASR_asr_2pass-zh-cn-16k-common-vocab8358-tensorflow1-online",
output_dir=output_dir,
)
- rec_result = inference_pipline(audio_in=audio_in)
+ rec_result = inference_pipeline(audio_in=audio_in)
print(rec_result)
diff --git a/egs_modelscope/asr/uniasr/speech_UniASR_asr_2pass-zh-cn-8k-common-vocab3445-pytorch-offline/infer.py b/egs_modelscope/asr/uniasr/speech_UniASR_asr_2pass-zh-cn-8k-common-vocab3445-pytorch-offline/infer.py
index ce8988e..13d2a2e 100644
--- a/egs_modelscope/asr/uniasr/speech_UniASR_asr_2pass-zh-cn-8k-common-vocab3445-pytorch-offline/infer.py
+++ b/egs_modelscope/asr/uniasr/speech_UniASR_asr_2pass-zh-cn-8k-common-vocab3445-pytorch-offline/infer.py
@@ -16,14 +16,14 @@
os.environ['CUDA_VISIBLE_DEVICES'] = str(gpu_list[gpu_id])
else:
os.environ['CUDA_VISIBLE_DEVICES'] = str(gpu_id)
- inference_pipline = pipeline(
+ inference_pipeline = pipeline(
task=Tasks.auto_speech_recognition,
model="damo/speech_UniASR_asr_2pass-zh-cn-8k-common-vocab3445-pytorch-offline",
output_dir=output_dir_job,
batch_size=1
)
audio_in = os.path.join(split_dir, "wav.{}.scp".format(idx))
- inference_pipline(audio_in=audio_in)
+ inference_pipeline(audio_in=audio_in)
def modelscope_infer(params):
# prepare for multi-GPU decoding
diff --git a/egs_modelscope/asr/uniasr/speech_UniASR_asr_2pass-zh-cn-8k-common-vocab3445-pytorch-online/infer.py b/egs_modelscope/asr/uniasr/speech_UniASR_asr_2pass-zh-cn-8k-common-vocab3445-pytorch-online/infer.py
index 8b4a04d..876d51c 100644
--- a/egs_modelscope/asr/uniasr/speech_UniASR_asr_2pass-zh-cn-8k-common-vocab3445-pytorch-online/infer.py
+++ b/egs_modelscope/asr/uniasr/speech_UniASR_asr_2pass-zh-cn-8k-common-vocab3445-pytorch-online/infer.py
@@ -16,14 +16,14 @@
os.environ['CUDA_VISIBLE_DEVICES'] = str(gpu_list[gpu_id])
else:
os.environ['CUDA_VISIBLE_DEVICES'] = str(gpu_id)
- inference_pipline = pipeline(
+ inference_pipeline = pipeline(
task=Tasks.auto_speech_recognition,
model="damo/speech_UniASR_asr_2pass-zh-cn-8k-common-vocab3445-pytorch-online",
output_dir=output_dir_job,
batch_size=1
)
audio_in = os.path.join(split_dir, "wav.{}.scp".format(idx))
- inference_pipline(audio_in=audio_in, param_dict={"decoding_model": "normal"})
+ inference_pipeline(audio_in=audio_in, param_dict={"decoding_model": "normal"})
def modelscope_infer(params):
diff --git a/egs_modelscope/asr/uniasr/speech_UniASR_asr_2pass-zh-cn-8k-common-vocab8358-tensorflow1-offline/infer.py b/egs_modelscope/asr/uniasr/speech_UniASR_asr_2pass-zh-cn-8k-common-vocab8358-tensorflow1-offline/infer.py
index 1c1e303..8ec4288 100644
--- a/egs_modelscope/asr/uniasr/speech_UniASR_asr_2pass-zh-cn-8k-common-vocab8358-tensorflow1-offline/infer.py
+++ b/egs_modelscope/asr/uniasr/speech_UniASR_asr_2pass-zh-cn-8k-common-vocab8358-tensorflow1-offline/infer.py
@@ -4,11 +4,11 @@
if __name__ == '__main__':
audio_in = 'https://isv-data.oss-cn-hangzhou.aliyuncs.com/ics/MaaS/ASR/test_audio/asr_example_zh.wav'
output_dir = None
- inference_pipline = pipeline(
+ inference_pipeline = pipeline(
task=Tasks.auto_speech_recognition,
model="damo/speech_UniASR_asr_2pass-zh-cn-8k-common-vocab8358-tensorflow1-offline",
output_dir=output_dir,
)
- rec_result = inference_pipline(audio_in=audio_in)
+ rec_result = inference_pipeline(audio_in=audio_in)
print(rec_result)
diff --git a/egs_modelscope/asr/uniasr/speech_UniASR_asr_2pass-zh-cn-8k-common-vocab8358-tensorflow1-online/infer.py b/egs_modelscope/asr/uniasr/speech_UniASR_asr_2pass-zh-cn-8k-common-vocab8358-tensorflow1-online/infer.py
index 94c1b68..3ab16ea 100644
--- a/egs_modelscope/asr/uniasr/speech_UniASR_asr_2pass-zh-cn-8k-common-vocab8358-tensorflow1-online/infer.py
+++ b/egs_modelscope/asr/uniasr/speech_UniASR_asr_2pass-zh-cn-8k-common-vocab8358-tensorflow1-online/infer.py
@@ -4,11 +4,11 @@
if __name__ == '__main__':
audio_in = 'https://isv-data.oss-cn-hangzhou.aliyuncs.com/ics/MaaS/ASR/test_audio/asr_example_zh.wav'
output_dir = None
- inference_pipline = pipeline(
+ inference_pipeline = pipeline(
task=Tasks.auto_speech_recognition,
model="damo/speech_UniASR_asr_2pass-zh-cn-8k-common-vocab8358-tensorflow1-online",
output_dir=output_dir,
)
- rec_result = inference_pipline(audio_in=audio_in)
+ rec_result = inference_pipeline(audio_in=audio_in)
print(rec_result)
diff --git a/egs_modelscope/lm/speech_transformer_lm_zh-cn-common-vocab8404-pytorch/infer.py b/egs_modelscope/lm/speech_transformer_lm_zh-cn-common-vocab8404-pytorch/infer.py
index ec309b2..628cdd8 100644
--- a/egs_modelscope/lm/speech_transformer_lm_zh-cn-common-vocab8404-pytorch/infer.py
+++ b/egs_modelscope/lm/speech_transformer_lm_zh-cn-common-vocab8404-pytorch/infer.py
@@ -6,12 +6,12 @@
from modelscope.pipelines import pipeline
from modelscope.utils.constant import Tasks
-inference_pipline = pipeline(
+inference_pipeline = pipeline(
task=Tasks.language_score_prediction,
model='damo/speech_transformer_lm_zh-cn-common-vocab8404-pytorch',
output_dir="./tmp/"
)
-rec_result = inference_pipline(text_in=inputs)
+rec_result = inference_pipeline(text_in=inputs)
print(rec_result)
diff --git a/egs_modelscope/punctuation/TEMPLATE/README.md b/egs_modelscope/punctuation/TEMPLATE/README.md
index 7cbca05..19600d3 100644
--- a/egs_modelscope/punctuation/TEMPLATE/README.md
+++ b/egs_modelscope/punctuation/TEMPLATE/README.md
@@ -11,21 +11,21 @@
from modelscope.pipelines import pipeline
from modelscope.utils.constant import Tasks
-inference_pipline = pipeline(
+inference_pipeline = pipeline(
task=Tasks.punctuation,
model='damo/punc_ct-transformer_zh-cn-common-vocab272727-pytorch',
model_revision=None)
-rec_result = inference_pipline(text_in='example/punc_example.txt')
+rec_result = inference_pipeline(text_in='example/punc_example.txt')
print(rec_result)
```
- text浜岃繘鍒舵暟鎹紝渚嬪锛氱敤鎴风洿鎺ヤ粠鏂囦欢閲岃鍑篵ytes鏁版嵁
```python
-rec_result = inference_pipline(text_in='鎴戜滑閮芥槸鏈ㄥご浜轰笉浼氳璇濅笉浼氬姩')
+rec_result = inference_pipeline(text_in='鎴戜滑閮芥槸鏈ㄥご浜轰笉浼氳璇濅笉浼氬姩')
```
- text鏂囦欢url锛屼緥濡傦細https://isv-data.oss-cn-hangzhou.aliyuncs.com/ics/MaaS/ASR/test_text/punc_example.txt
```python
-rec_result = inference_pipline(text_in='https://isv-data.oss-cn-hangzhou.aliyuncs.com/ics/MaaS/ASR/test_text/punc_example.txt')
+rec_result = inference_pipeline(text_in='https://isv-data.oss-cn-hangzhou.aliyuncs.com/ics/MaaS/ASR/test_text/punc_example.txt')
```
#### [CT-Transformer Realtime model](https://www.modelscope.cn/models/damo/punc_ct-transformer_zh-cn-common-vad_realtime-vocab272727/summary)
diff --git a/egs_modelscope/punctuation/punc_ct-transformer_zh-cn-common-vocab272727-pytorch/demo.py b/egs_modelscope/punctuation/punc_ct-transformer_zh-cn-common-vocab272727-pytorch/demo.py
index 0da8d25..20994d3 100644
--- a/egs_modelscope/punctuation/punc_ct-transformer_zh-cn-common-vocab272727-pytorch/demo.py
+++ b/egs_modelscope/punctuation/punc_ct-transformer_zh-cn-common-vocab272727-pytorch/demo.py
@@ -12,12 +12,12 @@
from modelscope.pipelines import pipeline
from modelscope.utils.constant import Tasks
-inference_pipline = pipeline(
+inference_pipeline = pipeline(
task=Tasks.punctuation,
model='damo/punc_ct-transformer_zh-cn-common-vocab272727-pytorch',
model_revision="v1.1.7",
output_dir="./tmp/"
)
-rec_result = inference_pipline(text_in=inputs)
+rec_result = inference_pipeline(text_in=inputs)
print(rec_result)
diff --git a/egs_modelscope/tp/TEMPLATE/README.md b/egs_modelscope/tp/TEMPLATE/README.md
index 8d75581..745249f 100644
--- a/egs_modelscope/tp/TEMPLATE/README.md
+++ b/egs_modelscope/tp/TEMPLATE/README.md
@@ -8,12 +8,12 @@
from modelscope.pipelines import pipeline
from modelscope.utils.constant import Tasks
-inference_pipline = pipeline(
+inference_pipeline = pipeline(
task=Tasks.speech_timestamp,
model='damo/speech_timestamp_prediction-v1-16k-offline',
output_dir=None)
-rec_result = inference_pipline(
+rec_result = inference_pipeline(
audio_in='https://isv-data.oss-cn-hangzhou.aliyuncs.com/ics/MaaS/ASR/test_audio/asr_example_timestamps.wav',
text_in='涓� 涓� 涓� 澶� 骞� 娲� 鍥� 瀹� 涓� 浠� 涔� 璺� 鍒� 瑗� 澶� 骞� 娲� 鏉� 浜� 鍛�',)
print(rec_result)
diff --git a/egs_modelscope/tp/speech_timestamp_prediction-v1-16k-offline/demo.py b/egs_modelscope/tp/speech_timestamp_prediction-v1-16k-offline/demo.py
index 2e6f92f..bcc5128 100644
--- a/egs_modelscope/tp/speech_timestamp_prediction-v1-16k-offline/demo.py
+++ b/egs_modelscope/tp/speech_timestamp_prediction-v1-16k-offline/demo.py
@@ -1,12 +1,12 @@
from modelscope.pipelines import pipeline
from modelscope.utils.constant import Tasks
-inference_pipline = pipeline(
+inference_pipeline = pipeline(
task=Tasks.speech_timestamp,
model='damo/speech_timestamp_prediction-v1-16k-offline',
output_dir=None)
-rec_result = inference_pipline(
+rec_result = inference_pipeline(
audio_in='https://isv-data.oss-cn-hangzhou.aliyuncs.com/ics/MaaS/ASR/test_audio/asr_example_timestamps.wav',
text_in='涓� 涓� 涓� 澶� 骞� 娲� 鍥� 瀹� 涓� 浠� 涔� 璺� 鍒� 瑗� 澶� 骞� 娲� 鏉� 浜� 鍛�',)
print(rec_result)
\ No newline at end of file
diff --git a/egs_modelscope/vad/speech_fsmn_vad_zh-cn-16k-common/demo.py b/egs_modelscope/vad/speech_fsmn_vad_zh-cn-16k-common/demo.py
index 2bf3251..bbc16c5 100644
--- a/egs_modelscope/vad/speech_fsmn_vad_zh-cn-16k-common/demo.py
+++ b/egs_modelscope/vad/speech_fsmn_vad_zh-cn-16k-common/demo.py
@@ -4,12 +4,12 @@
if __name__ == '__main__':
audio_in = 'https://isv-data.oss-cn-hangzhou.aliyuncs.com/ics/MaaS/ASR/test_audio/vad_example.wav'
output_dir = None
- inference_pipline = pipeline(
+ inference_pipeline = pipeline(
task=Tasks.voice_activity_detection,
model="damo/speech_fsmn_vad_zh-cn-16k-common-pytorch",
model_revision='v1.2.0',
output_dir=output_dir,
batch_size=1,
)
- segments_result = inference_pipline(audio_in=audio_in)
+ segments_result = inference_pipeline(audio_in=audio_in)
print(segments_result)
diff --git a/egs_modelscope/vad/speech_fsmn_vad_zh-cn-16k-common/demo_online.py b/egs_modelscope/vad/speech_fsmn_vad_zh-cn-16k-common/demo_online.py
index 02e919d..65693b5 100644
--- a/egs_modelscope/vad/speech_fsmn_vad_zh-cn-16k-common/demo_online.py
+++ b/egs_modelscope/vad/speech_fsmn_vad_zh-cn-16k-common/demo_online.py
@@ -8,7 +8,7 @@
if __name__ == '__main__':
output_dir = None
- inference_pipline = pipeline(
+ inference_pipeline = pipeline(
task=Tasks.voice_activity_detection,
model="damo/speech_fsmn_vad_zh-cn-16k-common-pytorch",
model_revision='v1.2.0',
@@ -30,7 +30,7 @@
else:
is_final = False
param_dict['is_final'] = is_final
- segments_result = inference_pipline(audio_in=speech[sample_offset: sample_offset + step],
+ segments_result = inference_pipeline(audio_in=speech[sample_offset: sample_offset + step],
param_dict=param_dict)
print(segments_result)
diff --git a/egs_modelscope/vad/speech_fsmn_vad_zh-cn-8k-common/demo.py b/egs_modelscope/vad/speech_fsmn_vad_zh-cn-8k-common/demo.py
index 2e50275..84863d0 100644
--- a/egs_modelscope/vad/speech_fsmn_vad_zh-cn-8k-common/demo.py
+++ b/egs_modelscope/vad/speech_fsmn_vad_zh-cn-8k-common/demo.py
@@ -4,12 +4,12 @@
if __name__ == '__main__':
audio_in = 'https://isv-data.oss-cn-hangzhou.aliyuncs.com/ics/MaaS/ASR/test_audio/vad_example_8k.wav'
output_dir = None
- inference_pipline = pipeline(
+ inference_pipeline = pipeline(
task=Tasks.voice_activity_detection,
model="damo/speech_fsmn_vad_zh-cn-8k-common",
model_revision='v1.2.0',
output_dir=output_dir,
batch_size=1,
)
- segments_result = inference_pipline(audio_in=audio_in)
+ segments_result = inference_pipeline(audio_in=audio_in)
print(segments_result)
diff --git a/egs_modelscope/vad/speech_fsmn_vad_zh-cn-8k-common/demo_online.py b/egs_modelscope/vad/speech_fsmn_vad_zh-cn-8k-common/demo_online.py
index a8cc912..5b67da7 100644
--- a/egs_modelscope/vad/speech_fsmn_vad_zh-cn-8k-common/demo_online.py
+++ b/egs_modelscope/vad/speech_fsmn_vad_zh-cn-8k-common/demo_online.py
@@ -8,7 +8,7 @@
if __name__ == '__main__':
output_dir = None
- inference_pipline = pipeline(
+ inference_pipeline = pipeline(
task=Tasks.voice_activity_detection,
model="damo/speech_fsmn_vad_zh-cn-8k-common",
model_revision='v1.2.0',
@@ -30,7 +30,7 @@
else:
is_final = False
param_dict['is_final'] = is_final
- segments_result = inference_pipline(audio_in=speech[sample_offset: sample_offset + step],
+ segments_result = inference_pipeline(audio_in=speech[sample_offset: sample_offset + step],
param_dict=param_dict)
print(segments_result)
--
Gitblit v1.9.1