readme: ---
  tags:
  - espnet
  - audio
  - audio-source-separation
  - audio-to-audio
  language: en
  datasets:
  - librimix
  license: cc-by-4.0
  inference: false
  ---
  
  # ESPnet2 ENH pretrained model
  
  ## `anogkongda/librimix_enh_train_raw_valid.si_snr.ave`
  
  ♻️ Imported from <https://zenodo.org/record/4480771#.YN70WJozZH4>
  This model was trained by anogkongda using librimix recipe in [espnet](https://github.com/espnet/espnet/).
  
  ### Demo: How to use in ESPnet2
  
  ```python
  # coming soon
  ```
  
  ### Citing ESPnet
  
  ```BibTex
  @inproceedings{watanabe2018espnet,
    author={Shinji Watanabe and Takaaki Hori and Shigeki Karita and Tomoki Hayashi and Jiro Nishitoba and Yuya Unno and Nelson {Enrique Yalta Soplin} and Jahn Heymann and Matthew Wiesner and Nanxin Chen and Adithya Renduchintala and Tsubasa Ochiai},
    title={{ESPnet}: End-to-End Speech Processing Toolkit},
    year={2018},
    booktitle={Proceedings of Interspeech},
    pages={2207--2211},
    doi={10.21437/Interspeech.2018-1456},
    url={http://dx.doi.org/10.21437/Interspeech.2018-1456}
  }
  @inproceedings{hayashi2020espnet,
    title={{Espnet-TTS}: Unified, reproducible, and integratable open source end-to-end text-to-speech toolkit},
    author={Hayashi, Tomoki and Yamamoto, Ryuichi and Inoue, Katsuki and Yoshimura, Takenori and Watanabe, Shinji and Toda, Tomoki and Takeda, Kazuya and Zhang, Yu and Tan, Xu},
    booktitle={Proceedings of IEEE International Conference on Acoustics, Speech and Signal Processing (ICASSP)},
    pages={7654--7658},
    year={2020},
    organization={IEEE}
  }
  ```
  
  or arXiv:
  
  ```bibtex
  @misc{watanabe2018espnet,
        title={ESPnet: End-to-End Speech Processing Toolkit}, 
        author={Shinji Watanabe and Takaaki Hori and Shigeki Karita and Tomoki Hayashi and Jiro Nishitoba and Yuya Unno and Nelson Enrique Yalta Soplin and Jahn Heymann and Matthew Wiesner and Nanxin Chen and Adithya Renduchintala and Tsubasa Ochiai},
        year={2018},
        eprint={1804.00015},
        archivePrefix={arXiv},
        primaryClass={cs.CL}
  }
  ```
  
  ### Training config
  
  See full config in [`config.yaml`](./config.yaml)
  
  ```yaml
  config: conf/tuning/train_conformer_fastspeech2.yaml
  print_config: false
  log_level: INFO
  dry_run: false
  iterator_type: sequence
  output_dir: exp/tts_train_conformer_fastspeech2_raw_phn_jaconv_pyopenjtalk
  ngpu: 1
  seed: 0
  num_workers: 1
  num_att_plot: 3
  dist_backend: nccl
  dist_init_method: env://
  dist_world_size: null
  dist_rank: null
  local_rank: 0
  dist_master_addr: null
  dist_master_port: null
  dist_launcher: null
  multiprocessing_distributed: false
  cudnn_enabled: true
  cudnn_benchmark: false
  cudnn_deterministic: true
  ```
  