pyf98/owsm_ctc_v3.2_ft_1B

OWSM-CTC (Peng et al., ACL 2024) is an encoder-only speech foundation model based on hierarchical multi-task self-conditioned CTC. It is trained on 180k hours of public audio data for multilingual speech recognition, any-to-any speech translation, and language identification, which follows the design of the project, Open Whisper-style Speech Model (OWSM).

This model is initialized with OWSM-CTC v3.1 and then fine-tuned on v3.2 data for 225k steps.

Currently, the code for OWSM-CTC has not been merged into ESPnet main branch. Instead, it is available as follows:

• PR in ESPnet: https://github.com/espnet/espnet/pull/5933
• Code in my repo: https://github.com/pyf98/espnet/tree/owsm-ctc
• Current model on HF: https://huggingface.co/pyf98/owsm_ctc_v3.2_ft_1B

To use the pre-trained model, you need to install espnet and espnet_model_zoo. The requirements are:

librosa
torch
espnet @ git+https://github.com/pyf98/espnet@owsm-ctc
espnet_model_zoo

We use FlashAttention during training, but we do not need it during inference. Please install it as follows:

pip install flash-attn --no-build-isolation

Example script for short-form ASR/ST

import soundfile as sf
import numpy as np
import librosa
import kaldiio
from espnet2.bin.s2t_inference_ctc import Speech2TextGreedySearch


s2t = Speech2TextGreedySearch.from_pretrained(
    "pyf98/owsm_ctc_v3.2_ft_1B",
    device="cuda",
    generate_interctc_outputs=False,
    lang_sym='<eng>',
    task_sym='<asr>',
)

speech, rate = sf.read(
    "xxx.wav"
)
speech = librosa.util.fix_length(speech, size=(16000 * 30))

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

Example script for long-form ASR/ST

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


if __name__ == "__main__":
    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(
        "pyf98/owsm_ctc_v3.2_ft_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,
        frames_per_sec=12.5,        # 80ms shift, model-dependent, don't change
    )
    print(text)

Example for CTC forced alignment using ctc-segmentation

It can be efficiently applied to audio of an arbitrary length. For model downloading, please refer to https://github.com/espnet/espnet?tab=readme-ov-file#ctc-segmentation-demo

import soundfile as sf
from espnet2.bin.s2t_ctc_align import CTCSegmentation


if __name__ == "__main__":
    ## Please download model first
    aligner = CTCSegmentation(
        s2t_model_file="exp/s2t_train_s2t_multitask-ctc_ebf27_conv2d8_size1024_raw_bpe50000/valid.total_count.ave_5best.till45epoch.pth",
        fs=16000,
        ngpu=1,
        batch_size=16,    # batched parallel decoding; reduce it if your GPU memory is smaller
        kaldi_style_text=True,
        time_stamps="fixed",
        samples_to_frames_ratio=1280,   # 80ms time shift; don't change as it depends on the pre-trained model
        lang_sym="<eng>",
        task_sym="<asr>",
        context_len_in_secs=2,  # left and right context in buffered decoding
        frames_per_sec=12.5,    # 80ms time shift; don't change as it depends on the pre-trained model
    )

    speech, rate = sf.read(
        "example.wav"
    )
    print(f"speech duration: {len(speech) / rate : .2f} seconds")
    text = '''
utt1 hello there
utt2 welcome to this repo
'''

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