Hugging Face's logo

wyz
/
vctk_bsrnn_medium_causal

Audio-to-Audio
ESPnet
English
audio
Model card Files Community
Edit model card

ESPnet2 ENH model

wyz/vctk_bsrnn_medium_causal

This model was trained by wyz based on the universal_se_v1 recipe in espnet. More information can be found at https://github.com/Emrys365/se-scaling.

Demo: How to use in ESPnet2

Follow the ESPnet installation instructions if you haven't done that already.

To use the model in the Python interface, you could use the following code:

import soundfile as sf
from espnet2.bin.enh_inference import SeparateSpeech

# For model downloading + loading
model = SeparateSpeech.from_pretrained(
    model_tag="wyz/vctk_bsrnn_medium_causal",
    normalize_output_wav=True,
    device="cuda",
)
# For loading a downloaded model
# model = SeparateSpeech(
#     train_config="exp_vctk/enh_train_enh_bsrnn_medium_raw/config.yaml",
#     model_file="exp_vctk/enh_train_enh_bsrnn_medium_raw/xxxx.pth",
#     normalize_output_wav=True,
#     device="cuda",
# )

audio, fs = sf.read("/path/to/noisy/utt1.flac")
enhanced = model(audio[None, :], fs=fs)[0]

RESULTS

Environments

  • date: Wed Feb 28 12:07:14 EST 2024
  • python version: 3.8.16 (default, Mar 2 2023, 03:21:46) [GCC 11.2.0]
  • espnet version: espnet 202304
  • pytorch version: pytorch 2.0.1+cu118
  • Git hash: 443028662106472c60fe8bd892cb277e5b488651
    • Commit date: Thu May 11 03:32:59 2023 +0000

enhanced_test_16k

dataset PESQ_WB STOI SAR SDR SIR SI_SNR OVRL SIG BAK P808_MOS
chime4_et05_real_isolated_6ch_track 1.10 46.19 -3.90 -3.90 0.00 -31.52 2.39 2.75 3.48 3.17
chime4_et05_simu_isolated_6ch_track 1.15 63.97 4.12 4.12 0.00 -1.88 2.19 2.49 3.53 2.81
dns20_tt_synthetic_no_reverb 2.13 92.72 13.14 13.14 0.00 12.22 3.08 3.47 3.77 3.75
reverb_et_real_8ch_multich 1.07 54.20 2.78 2.78 0.00 -0.85 2.17 2.48 3.49 3.14
reverb_et_simu_8ch_multich 1.46 79.48 7.91 7.91 0.00 -10.95 2.71 3.07 3.69 3.51
whamr_tt_mix_single_reverb_max_16k 1.21 72.11 4.64 4.64 0.00 0.37 2.29 2.58 3.64 3.18

enhanced_test_48k

dataset STOI SAR SDR SIR SI_SNR OVRL SIG BAK P808_MOS
vctk_noisy_tt_2spk 94.58 19.73 19.73 0.00 18.70 3.09 3.42 3.92 3.47

ENH config

expand 
config: conf/tuning/train_enh_bsrnn_medium.yaml
print_config: false
log_level: INFO
dry_run: false
iterator_type: chunk
output_dir: exp_vctk/enh_train_enh_bsrnn_medium_raw
ngpu: 1
seed: 0
num_workers: 4
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
unused_parameters: true
sharded_ddp: false
cudnn_enabled: true
cudnn_benchmark: false
cudnn_deterministic: true
collect_stats: false
write_collected_feats: false
max_epoch: 100
patience: 40
val_scheduler_criterion:
- valid
- loss
early_stopping_criterion:
- valid
- loss
- min
best_model_criterion:
-   - valid
    - loss
    - min
keep_nbest_models: 1
nbest_averaging_interval: 0
grad_clip: 5.0
grad_clip_type: 2.0
grad_noise: false
accum_grad: 1
no_forward_run: false
resume: true
save_interval: 1000
train_dtype: float32
use_amp: false
log_interval: null
use_matplotlib: true
use_tensorboard: true
create_graph_in_tensorboard: false
use_wandb: false
wandb_project: null
wandb_id: null
wandb_entity: null
wandb_name: null
wandb_model_log_interval: -1
detect_anomaly: false
pretrain_path: null
init_param: []
ignore_init_mismatch: false
freeze_param: []
num_iters_per_epoch: 8000
num_iters_valid: null
batch_size: 4
valid_batch_size: null
batch_bins: 1000000
valid_batch_bins: null
train_shape_file:
- exp_vctk/enh_stats_16k/train/speech_mix_shape
- exp_vctk/enh_stats_16k/train/speech_ref1_shape
- exp_vctk/enh_stats_16k/train/dereverb_ref1_shape
valid_shape_file:
- exp_vctk/enh_stats_16k/valid/speech_mix_shape
- exp_vctk/enh_stats_16k/valid/speech_ref1_shape
- exp_vctk/enh_stats_16k/valid/dereverb_ref1_shape
batch_type: folded
valid_batch_type: null
fold_length:
- 80000
- 80000
- 80000
sort_in_batch: descending
sort_batch: descending
multiple_iterator: false
chunk_length: 32000
chunk_shift_ratio: 0.5
num_cache_chunks: 1024
chunk_excluded_key_prefixes: []
chunk_discard_short_samples: false
train_data_path_and_name_and_type:
-   - dump/raw/vctk_noisy_tr_26spk/wav.scp
    - speech_mix
    - sound
-   - dump/raw/vctk_noisy_tr_26spk/spk1.scp
    - speech_ref1
    - sound
-   - dump/raw/vctk_noisy_tr_26spk/dereverb1.scp
    - dereverb_ref1
    - sound
-   - dump/raw/vctk_noisy_tr_26spk/utt2category
    - category
    - text
-   - dump/raw/vctk_noisy_tr_26spk/utt2fs
    - fs
    - text_int
valid_data_path_and_name_and_type:
-   - dump/raw/vctk_noisy_cv_2spk/wav.scp
    - speech_mix
    - sound
-   - dump/raw/vctk_noisy_cv_2spk/spk1.scp
    - speech_ref1
    - sound
-   - dump/raw/vctk_noisy_cv_2spk/dereverb1.scp
    - dereverb_ref1
    - sound
-   - dump/raw/vctk_noisy_cv_2spk/utt2category
    - category
    - text
-   - dump/raw/vctk_noisy_cv_2spk/utt2fs
    - fs
    - text_int
allow_variable_data_keys: false
max_cache_size: 0.0
max_cache_fd: 32
allow_multi_rates: true
valid_max_cache_size: null
exclude_weight_decay: false
exclude_weight_decay_conf: {}
optim: adam
optim_conf:
    lr: 0.001
    eps: 1.0e-08
    weight_decay: 1.0e-05
scheduler: steplr
scheduler_conf:
    step_size: 2
    gamma: 0.99
init: null
model_conf:
    normalize_variance_per_ch: true
    categories:
    - 1ch_8k
    - 1ch_8k_r
    - 1ch_16k_r
    - 1ch_48k
    - 1ch_24k
    - 1ch_16k
    - 2ch_8k
    - 2ch_8k_r
    - 2ch_16k
    - 2ch_16k_r
    - 5ch_8k
    - 5ch_16k
    - 8ch_8k_r
    - 8ch_16k_r
criterions:
-   name: mr_l1_tfd
    conf:
        window_sz:
        - 256
        - 512
        - 768
        - 1024
        hop_sz: null
        eps: 1.0e-08
        time_domain_weight: 0.5
        normalize_variance: true
    wrapper: fixed_order
    wrapper_conf:
        weight: 1.0
-   name: si_snr
    conf:
        eps: 1.0e-07
    wrapper: fixed_order
    wrapper_conf:
        weight: 0.0
speech_volume_normalize: null
rir_scp: null
rir_apply_prob: 1.0
noise_scp: null
noise_apply_prob: 1.0
noise_db_range: '13_15'
short_noise_thres: 0.5
use_reverberant_ref: false
num_spk: 1
num_noise_type: 1
sample_rate: 8000
force_single_channel: true
channel_reordering: true
categories:
- 1ch_8k
- 1ch_8k_r
- 1ch_16k_r
- 1ch_48k
- 1ch_24k
- 1ch_16k
- 2ch_8k
- 2ch_8k_r
- 2ch_16k
- 2ch_16k_r
- 5ch_8k
- 5ch_16k
- 8ch_8k_r
- 8ch_16k_r
speech_segment: null
avoid_allzero_segment: true
flexible_numspk: false
dynamic_mixing: false
utt2spk: null
dynamic_mixing_gain_db: 0.0
encoder: stft
encoder_conf:
    n_fft: 960
    hop_length: 480
    use_builtin_complex: true
    default_fs: 48000
separator: bsrnn
separator_conf:
    num_spk: 1
    num_channels: 128
    num_layers: 6
    target_fs: 48000
    ref_channel: 0
decoder: stft
decoder_conf:
    n_fft: 960
    hop_length: 480
    default_fs: 48000
mask_module: multi_mask
mask_module_conf: {}
preprocessor: enh
preprocessor_conf: {}
required:
- output_dir
version: '202304'
distributed: false

Citing ESPnet 

@inproceedings{watanabe2018espnet,
  author={Shinji Watanabe and Takaaki Hori and Shigeki Karita and Tomoki Hayashi and Jiro Nishitoba and Yuya Unno and Nelson Yalta 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{ESPnet-SE,
  author = {Chenda Li and Jing Shi and Wangyou Zhang and Aswin Shanmugam Subramanian and Xuankai Chang and
  Naoyuki Kamo and Moto Hira and Tomoki Hayashi and Christoph B{"{o}}ddeker and Zhuo Chen and Shinji Watanabe},
  title = {ESPnet-SE: End-To-End Speech Enhancement and Separation Toolkit Designed for {ASR} Integration},
  booktitle = {{IEEE} Spoken Language Technology Workshop, {SLT} 2021, Shenzhen, China, January 19-22, 2021},
  pages = {785--792},
  publisher = {{IEEE}},
  year = {2021},
  url = {https://doi.org/10.1109/SLT48900.2021.9383615},
  doi = {10.1109/SLT48900.2021.9383615},
  timestamp = {Mon, 12 Apr 2021 17:08:59 +0200},
  biburl = {https://dblp.org/rec/conf/slt/Li0ZSCKHHBC021.bib},
  bibsource = {dblp computer science bibliography, https://dblp.org}
}

or arXiv:

@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 Yalta 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}
}
Downloads last month
4
Inference Examples
Audio-to-Audio
Inference API (serverless) does not yet support espnet models for this pipeline type.