--- tags: - espnet - audio - automatic-speech-recognition - speech-translation - language-identification language: multilingual datasets: - owsm_v3.1_ctc license: cc-by-4.0 --- [OWSM-CTC](https://aclanthology.org/2024.acl-long.549/) (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)](https://arxiv.org/abs/2401.16658). Due to time constraint, the model used in the paper was trained for 40 "epochs". The new model trained for 45 "epochs" (approximately three entire passes on the full data) is also added in this repo in order to match the setup of encoder-decoder OWSM. It can have better performance than the old one in many test sets. 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_v3.1_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: ```bash pip install flash-attn --no-build-isolation ``` ### Example script for short-form ASR/ST ```python 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_v3.1_1B", device="cuda", generate_interctc_outputs=False, lang_sym='', task_sym='', ) 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 ```python 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( "pyf98/owsm_ctc_v3.1_1B", device='cuda' if torch.cuda.is_available() else 'cpu', generate_interctc_outputs=False, lang_sym='', task_sym='', ) 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 ```python import soundfile as sf from espnet2.bin.s2t_ctc_align import CTCSegmentation ## 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="", task_sym="", 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) ```