Owsm Ctc V3.1 1B

🚀 OWSM-CTC語音基礎模型

OWSM-CTC是一個基於分層多任務自條件CTC的僅編碼器語音基礎模型，可用於多語言語音識別、任意到任意的語音翻譯和語言識別。它在180k小時的公共音頻數據上進行訓練，遵循Open Whisper-style Speech Model (OWSM)項目的設計。

🚀 快速開始

要使用預訓練模型，請安裝espnet和espnet_model_zoo，所需依賴如下：

librosa
torch
espnet
espnet_model_zoo

該模型的配方可在ESPnet中找到： https://github.com/espnet/espnet/tree/master/egs2/owsm_ctc_v3.1/s2t1

✨ 主要特性

• 基於分層多任務自條件CTC的僅編碼器架構。
• 在180k小時的公共音頻數據上進行訓練，支持多語言語音識別、任意到任意的語音翻譯和語言識別。
• 提供統一的批量推理方法，支持短音頻和長音頻的處理。

📦 安裝指南

安裝所需的庫，依賴列表如下：

librosa
torch
espnet
espnet_model_zoo

💻 使用示例

基礎用法

批量推理示例腳本

Speech2TextGreedySearch現在提供了統一的批量推理方法batch_decode，它可以對一批短音頻或長音頻進行CTC貪心解碼。如果音頻短於30秒，將被填充到30秒；否則，將被分割成重疊的片段。

from espnet2.bin.s2t_inference_ctc import Speech2TextGreedySearch

s2t = Speech2TextGreedySearch.from_pretrained(
    "espnet/owsm_ctc_v3.1_1B",
    device="cuda",
    use_flash_attn=False,   # set to True for better efficiency if flash attn is installed and dtype is float16 or bfloat16
    lang_sym='<eng>',
    task_sym='<asr>',
)

res = s2t.batch_decode(
    "audio.wav",    # a single audio (path or 1-D array/tensor) as input
    batch_size=16,
    context_len_in_secs=4,
)   # res is a single str, i.e., the predicted text without special tokens

res = s2t.batch_decode(
    ["audio1.wav", "audio2.wav", "audio3.wav"], # a list of audios as input
    batch_size=16,
    context_len_in_secs=4,
)   # res is a list of str

# Please check the code of `batch_decode` for all supported inputs

短音頻語音識別/翻譯/語言識別示例腳本

我們的模型在16kHz、固定時長為30秒的音頻上進行訓練。使用預訓練模型時，請確保輸入語音為16kHz，並將其填充或截斷為30秒。

import librosa
from espnet2.bin.s2t_inference_ctc import Speech2TextGreedySearch

s2t = Speech2TextGreedySearch.from_pretrained(
    "espnet/owsm_ctc_v3.1_1B",
    device="cuda",
    generate_interctc_outputs=False,
    lang_sym='<eng>',
    task_sym='<asr>',
)

# NOTE: OWSM-CTC is trained on 16kHz audio with a fixed 30s duration. Please ensure your input has the correct sample rate; otherwise resample it to 16k before feeding it to the model
speech, rate = librosa.load("xxx.wav", sr=16000)
speech = librosa.util.fix_length(speech, size=(16000 * 30))

res = s2t(speech)[0]
print(res)

長音頻語音識別/翻譯示例腳本

import soundfile as sf
import torch
from espnet2.bin.s2t_inference_ctc import Speech2TextGreedySearch

context_len_in_secs = 4   # left and right context when doing buffered inference
batch_size = 32   # depends on the GPU memory
s2t = Speech2TextGreedySearch.from_pretrained(
    "espnet/owsm_ctc_v3.1_1B",
    device='cuda' if torch.cuda.is_available() else 'cpu',
    generate_interctc_outputs=False,
    lang_sym='<eng>',
    task_sym='<asr>',
)

speech, rate = sf.read(
    "xxx.wav"
)

text = s2t.decode_long_batched_buffered(
    speech,
    batch_size=batch_size,
    context_len_in_secs=context_len_in_secs,
)
print(text)

使用ctc-segmentation進行CTC強制對齊示例

CTC分割可以高效地應用於任意長度的音頻。

import soundfile as sf
from espnet2.bin.s2t_ctc_align import CTCSegmentation
from espnet_model_zoo.downloader import ModelDownloader

# Download model first
d = ModelDownloader()
downloaded = d.download_and_unpack("espnet/owsm_ctc_v3.1_1B")

aligner = CTCSegmentation(
    **downloaded,
    fs=16000,
    ngpu=1,
    batch_size=32,    # batched parallel decoding; reduce it if your GPU memory is smaller
    kaldi_style_text=True,
    time_stamps="auto",     # "auto" can be more accurate than "fixed" when converting token index to timestamp
    lang_sym="<eng>",
    task_sym="<asr>",
    context_len_in_secs=2,  # left and right context in buffered decoding
)

speech, rate = sf.read(
    "./test_utils/ctc_align_test.wav"
)
print(f"speech duration: {len(speech) / rate : .2f} seconds")
text = """
utt1 THE SALE OF THE HOTELS
utt2 IS PART OF HOLIDAY'S STRATEGY
utt3 TO SELL OFF ASSETS
utt4 AND CONCENTRATE ON PROPERTY MANAGEMENT
"""

segments = aligner(speech, text)
print(segments)

📚 詳細文檔

模型信息

引用信息

OWSM-CTC

@inproceedings{owsm-ctc,
    title = "{OWSM}-{CTC}: An Open Encoder-Only Speech Foundation Model for Speech Recognition, Translation, and Language Identification",
    author = "Peng, Yifan  and
      Sudo, Yui  and
      Shakeel, Muhammad  and
      Watanabe, Shinji",
    booktitle = "Proceedings of the Annual Meeting of the Association for Computational Linguistics (ACL)",
    year = "2024",
    month= {8},
    url = "https://aclanthology.org/2024.acl-long.549",
}

OWSM v3.1 and v3.2

@inproceedings{owsm-v32,
  title={On the Effects of Heterogeneous Data Sources on Speech-to-Text Foundation Models},
  author={Jinchuan Tian and Yifan Peng and William Chen and Kwanghee Choi and Karen Livescu and Shinji Watanabe},
  booktitle={Proceedings of the Annual Conference of the International Speech Communication Association (INTERSPEECH)},
  year={2024},
  month={9},
  pdf="https://arxiv.org/pdf/2406.09282"
}
@inproceedings{owsm-v31,
  title={{OWSM v3.1: Better and Faster Open Whisper-Style Speech Models based on E-Branchformer}},
  author={Yifan Peng and Jinchuan Tian and William Chen and Siddhant Arora and Brian Yan and Yui Sudo and Muhammad Shakeel and Kwanghee Choi and Jiatong Shi and Xuankai Chang and Jee-weon Jung and Shinji Watanabe},
  booktitle={Proceedings of the Annual Conference of the International Speech Communication Association (INTERSPEECH)},
  year={2024},
  month={9},
  pdf="https://arxiv.org/pdf/2401.16658",
}

Initial OWSM (v1, v2, v3)

@inproceedings{owsm,
  title={Reproducing Whisper-Style Training Using An Open-Source Toolkit And Publicly Available Data},
  author={Yifan Peng and Jinchuan Tian and Brian Yan and Dan Berrebbi and Xuankai Chang and Xinjian Li and Jiatong Shi and Siddhant Arora and William Chen and Roshan Sharma and Wangyou Zhang and Yui Sudo and Muhammad Shakeel and Jee-weon Jung and Soumi Maiti and Shinji Watanabe},
  booktitle={Proceedings of the IEEE Automatic Speech Recognition and Understanding Workshop (ASRU)},
  year={2023},
  month={12},
  pdf="https://arxiv.org/pdf/2309.13876",
}

📄 許可證

本項目採用CC BY 4.0許可證。詳情請見：https://creativecommons.org/licenses/by/4.0/