ESPnet2 Codec model
espnet/amuse_dac_16k
This model was trained by ftshijt using amuse recipe in espnet.
Demo: How to use in ESPnet2
Follow the ESPnet installation instructions if you haven't done that already.
cd espnet
git checkout 5201685018b0e8fb9826bc51a710623140a06627
pip install -e .
cd egs2/amuse/codec1
./run.sh --skip_data_prep false --skip_train true --download_model espnet/amuse_dac_16k
Codec config
expand
config: conf/train_dac_fs16000.yaml
print_config: false
log_level: INFO
drop_last_iter: false
dry_run: false
iterator_type: chunk
valid_iterator_type: null
output_dir: exp_16k/codec_train_dac_fs16000_raw_fs16000
ngpu: 1
seed: 777
num_workers: 1
num_att_plot: 0
dist_backend: nccl
dist_init_method: env://
dist_world_size: 4
dist_rank: 0
local_rank: 0
dist_master_addr: localhost
dist_master_port: 50493
dist_launcher: null
multiprocessing_distributed: true
unused_parameters: true
sharded_ddp: false
cudnn_enabled: true
cudnn_benchmark: false
cudnn_deterministic: false
use_tf32: true
collect_stats: false
write_collected_feats: false
max_epoch: 120
patience: null
val_scheduler_criterion:
- valid
- loss
early_stopping_criterion:
- valid
- loss
- min
best_model_criterion:
- - valid
- mel_loss
- min
- - train
- mel_loss
- min
- - train
- total_count
- max
keep_nbest_models: 5
nbest_averaging_interval: 0
grad_clip: -1
grad_clip_type: 2.0
grad_noise: false
accum_grad: 1
no_forward_run: false
resume: true
train_dtype: float32
use_amp: false
log_interval: 1000
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
use_adapter: false
adapter: lora
save_strategy: all
adapter_conf: {}
pretrain_path: null
init_param: []
ignore_init_mismatch: false
freeze_param: []
num_iters_per_epoch: 5000
batch_size: 64
valid_batch_size: null
batch_bins: 1000000
valid_batch_bins: null
train_shape_file:
- exp_16k/codec_stats_raw/train/audio_shape
valid_shape_file:
- exp_16k/codec_stats_raw/valid/audio_shape
batch_type: unsorted
valid_batch_type: null
fold_length:
- 256000
sort_in_batch: descending
shuffle_within_batch: false
sort_batch: descending
multiple_iterator: false
chunk_length: 32000
chunk_shift_ratio: 0.5
num_cache_chunks: 128
chunk_excluded_key_prefixes: []
chunk_default_fs: null
train_data_path_and_name_and_type:
- - dump_16k/raw/train/wav.scp
- audio
- kaldi_ark
valid_data_path_and_name_and_type:
- - dump_16k/raw/dev-small/wav.scp
- audio
- kaldi_ark
multi_task_dataset: false
allow_variable_data_keys: false
max_cache_size: 0.0
max_cache_fd: 32
allow_multi_rates: false
valid_max_cache_size: null
exclude_weight_decay: false
exclude_weight_decay_conf: {}
optim: adam
optim_conf:
lr: 0.0002
betas:
- 0.5
- 0.9
eps: 1.0e-09
weight_decay: 0.0
scheduler: exponentiallr
scheduler_conf:
gamma: 0.999875
optim2: adam
optim2_conf:
lr: 0.0002
betas:
- 0.5
- 0.9
eps: 1.0e-09
weight_decay: 0.0
scheduler2: exponentiallr
scheduler2_conf:
gamma: 0.999875
generator_first: true
skip_discriminator_prob: 0.0
model_conf: {}
use_preprocessor: true
codec: dac
codec_conf:
sampling_rate: 16000
generator_params:
hidden_dim: 512
codebook_dim: 512
encdec_channels: 1
encdec_n_filters: 32
encdec_n_residual_layers: 3
encdec_ratios:
- 8
- 5
- 4
- 2
encdec_activation: Snake
encdec_norm: weight_norm
encdec_kernel_size: 7
encdec_residual_kernel_size: 7
encdec_last_kernel_size: 7
encdec_dilation_base: 2
encdec_causal: false
encdec_pad_mode: reflect
encdec_true_skip: false
encdec_compress: 2
encdec_lstm: 2
decoder_trim_right_ratio: 1.0
decoder_final_activation: null
decoder_final_activation_params: null
quantizer_n_q: 32
quantizer_bins: 1024
quantizer_decay: 0.99
quantizer_kmeans_init: true
quantizer_kmeans_iters: 50
quantizer_threshold_ema_dead_code: 2
quantizer_target_bandwidth:
- 2
- 4
- 8
- 16
- 32
quantizer_dropout: true
sample_rate: 16000
discriminator_params:
scales: 3
scale_downsample_pooling: AvgPool1d
scale_downsample_pooling_params:
kernel_size: 4
stride: 2
padding: 2
scale_discriminator_params:
in_channels: 1
out_channels: 1
kernel_sizes:
- 15
- 41
- 5
- 3
channels: 128
max_downsample_channels: 1024
max_groups: 16
bias: true
downsample_scales:
- 2
- 2
- 4
- 4
- 1
nonlinear_activation: LeakyReLU
nonlinear_activation_params:
negative_slope: 0.1
scale_follow_official_norm: false
msmpmb_discriminator_params:
rates: []
sample_rate: 16000
fft_sizes:
- 2048
- 1024
- 512
periods:
- 2
- 3
- 5
- 7
- 11
period_discriminator_params:
in_channels: 1
out_channels: 1
kernel_sizes:
- 5
- 3
channels: 32
downsample_scales:
- 3
- 3
- 3
- 3
- 1
max_downsample_channels: 1024
bias: true
nonlinear_activation: LeakyReLU
nonlinear_activation_params:
negative_slope: 0.1
use_weight_norm: true
use_spectral_norm: false
band_discriminator_params:
hop_factor: 0.25
sample_rate: 16000
bands:
- - 0.0
- 0.1
- - 0.1
- 0.25
- - 0.25
- 0.5
- - 0.5
- 0.75
- - 0.75
- 1.0
channel: 32
generator_adv_loss_params:
average_by_discriminators: false
loss_type: mse
discriminator_adv_loss_params:
average_by_discriminators: false
loss_type: mse
use_feat_match_loss: true
feat_match_loss_params:
average_by_discriminators: false
average_by_layers: false
include_final_outputs: true
use_mel_loss: true
mel_loss_params:
range_start: 6
range_end: 11
window: hann
n_mels: 80
fmin: 0
fmax: null
log_base: null
fs: 16000
lambda_quantization: 0.25
lambda_commit: 1.0
lambda_reconstruct: 1.0
lambda_adv: 1.0
lambda_mel: 45.0
lambda_feat_match: 2.0
cache_generator_outputs: true
required:
- output_dir
version: '202402'
distributed: true
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}
}
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}
}
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