sangeet2020 commited on
Commit
3fd6072
1 Parent(s): 5df46bf

pusing model files

Browse files
Files changed (7) hide show
  1. README.md +105 -1
  2. brain.ckpt +3 -0
  3. config.json +3 -0
  4. decoder.ckpt +3 -0
  5. encoder.ckpt +3 -0
  6. example_rescuespeech16k.wav +0 -0
  7. hyperparams.yaml +195 -0
README.md CHANGED
@@ -1,3 +1,107 @@
1
  ---
2
- license: apache-2.0
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
3
  ---
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
1
  ---
2
+ language: "de"
3
+ thumbnail:
4
+ tags:
5
+ - audio-to-audio
6
+ - Speech Enhancement
7
+ - RescueSpeech
8
+ - SepFormer
9
+ - Transformer
10
+ - pytorch
11
+ - speechbrain
12
+ license: "apache-2.0"
13
+ datasets:
14
+ - RescueSpeech
15
+ metrics:
16
+ - SI-SNR
17
+ - PESQ
18
+ - SDR
19
+
20
  ---
21
+
22
+ <iframe src="https://ghbtns.com/github-btn.html?user=speechbrain&repo=speechbrain&type=star&count=true&size=large&v=2" frameborder="0" scrolling="0" width="170" height="30" title="GitHub"></iframe>
23
+ <br/><br/>
24
+
25
+ # SepFormer trained on RescueSpeech for speech enhancement (16k sampling frequency)
26
+ This repository provides all the necessary tools to perform speech enhancement (denoising) with a [SepFormer](https://arxiv.org/abs/2010.13154v2) model, implemented with SpeechBrain. The model was first trained on Microsoft-DNS 4 dataset and further fine-tuned on RescueSpeech dataset 16k sampling frequency. For a better experience we encourage you to learn more about [SpeechBrain](https://speechbrain.github.io). Given below is model performance on RescueSpeech test set.
27
+
28
+
29
+ | Release | Test-Set SI-SNRi | Test-Set SI-SDRi |Test-Set PESQ |
30
+ |:-------------:|:--------------:|:--------------:|:--------------:|
31
+ | 07-01-23 | 7.849 | 8.414 | 2.24 |
32
+
33
+ where SI-SNRi and SI-SDRi indicates the improvement in SI-SNR and SI-SDR metric.
34
+
35
+ ## Install SpeechBrain
36
+
37
+ First of all, please install SpeechBrain with the following command:
38
+
39
+ ```
40
+ pip install speechbrain
41
+ ```
42
+
43
+ Please notice that we encourage you to read our tutorials and learn more about [SpeechBrain](https://speechbrain.github.io).
44
+
45
+ ### Perform speech enhancement on your own audio file
46
+
47
+ ```python
48
+ from speechbrain.pretrained import SepformerSeparation as separator
49
+ import torchaudio
50
+
51
+ model = separator.from_hparams(source="speechbrain/RescueSpeech_Sepformer", savedir='pretrained_models/RescueSpeech_Sepformer')
52
+
53
+ # for custom file, change path
54
+ est_sources = model.separate_file(path='speechbrain/RescueSpeech_Sepformer/example_rescuespeech16k.wav')
55
+
56
+ torchaudio.save("enhanced_rescuespeech16k.wav.wav", est_sources[:, :, 0].detach().cpu(), 16000)
57
+
58
+ ```
59
+
60
+ ### Inference on GPU
61
+ To perform inference on the GPU, add `run_opts={"device":"cuda"}` when calling the `from_hparams` method.
62
+
63
+ ### Limitations
64
+ The SpeechBrain team does not provide any warranty on the performance achieved by this model when used on other datasets.
65
+
66
+ #### Referencing SpeechBrain
67
+
68
+ ```bibtex
69
+ @misc{speechbrain,
70
+ title={{SpeechBrain}: A General-Purpose Speech Toolkit},
71
+ author={Mirco Ravanelli and Titouan Parcollet and Peter Plantinga and Aku Rouhe and Samuele Cornell and Loren Lugosch and Cem Subakan and Nauman Dawalatabad and Abdelwahab Heba and Jianyuan Zhong and Ju-Chieh Chou and Sung-Lin Yeh and Szu-Wei Fu and Chien-Feng Liao and Elena Rastorgueva and François Grondin and William Aris and Hwidong Na and Yan Gao and Renato De Mori and Yoshua Bengio},
72
+ year={2021},
73
+ eprint={2106.04624},
74
+ archivePrefix={arXiv},
75
+ primaryClass={eess.AS},
76
+ note={arXiv:2106.04624}
77
+ }
78
+ ```
79
+
80
+
81
+ #### Referencing SepFormer
82
+ ```bibtex
83
+ @inproceedings{subakan2021attention,
84
+ title={Attention is All You Need in Speech Separation},
85
+ author={Cem Subakan and Mirco Ravanelli and Samuele Cornell and Mirko Bronzi and Jianyuan Zhong},
86
+ year={2021},
87
+ booktitle={ICASSP 2021}
88
+ }
89
+ ```
90
+
91
+ ### Referencing RescueSpeech
92
+ ```bibtex
93
+ @misc{sagar2023rescuespeech,
94
+ title={RescueSpeech: A German Corpus for Speech Recognition in Search and Rescue Domain},
95
+ author={Sangeet Sagar and Mirco Ravanelli and Bernd Kiefer and Ivana Kruijff Korbayova and Josef van Genabith},
96
+ year={2023},
97
+ eprint={2306.04054},
98
+ archivePrefix={arXiv},
99
+ primaryClass={eess.AS}
100
+ }
101
+ ```
102
+
103
+
104
+ # **About SpeechBrain**
105
+ - Website: https://speechbrain.github.io/
106
+ - Code: https://github.com/speechbrain/speechbrain/
107
+ - HuggingFace: https://huggingface.co/speechbrain/
brain.ckpt ADDED
@@ -0,0 +1,3 @@
 
 
 
 
1
+ version https://git-lfs.github.com/spec/v1
2
+ oid sha256:33809a026a2c1febce7b03c8aafaee4ddfc851b2c70f180f8c06bf1017f4df5c
3
+ size 46
config.json ADDED
@@ -0,0 +1,3 @@
 
 
 
 
1
+ {
2
+ "speechbrain_interface": "SepformerSeparation"
3
+ }
decoder.ckpt ADDED
@@ -0,0 +1,3 @@
 
 
 
 
1
+ version https://git-lfs.github.com/spec/v1
2
+ oid sha256:00d272f965100f627a4a43d45dd919a7caf867372035139a91b8ece174c8b5f1
3
+ size 17195
encoder.ckpt ADDED
@@ -0,0 +1,3 @@
 
 
 
 
1
+ version https://git-lfs.github.com/spec/v1
2
+ oid sha256:a04882b82c623a27af12376cc05dc0eb6cc1d59fb483bfb463d89d7d6072ac4d
3
+ size 17259
example_rescuespeech16k.wav ADDED
Binary file (445 kB). View file
 
hyperparams.yaml ADDED
@@ -0,0 +1,195 @@
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
1
+ # Generated 2023-06-20 from:
2
+ # /netscratch/sagar/thesis/speechbrain/recipes/RescueSpeech/Enhancement/fine-tuning/hparams/sepformer_16k.yaml
3
+ # yamllint disable
4
+ # ################################
5
+ # Model: SepFormer for source separation
6
+ # https://arxiv.org/abs/2010.13154
7
+ #
8
+ # Author: Sangeet Sagar 2022
9
+ # Dataset : RescueSpeech
10
+ # ################################
11
+
12
+ # Basic parameters
13
+ # Seed needs to be set at top of yaml, before objects with parameters are made
14
+ seed: 8201
15
+ __set_seed: !apply:torch.manual_seed [8201]
16
+ experiment_name: sepformer-enhancement
17
+ output_folder: results/sepformer-enhancement/8201
18
+ train_log: results/sepformer-enhancement/8201/train_log.txt
19
+ save_folder: results/sepformer-enhancement/8201/save
20
+
21
+ # Dataset prep parameters
22
+ data_folder: ../../dataset/audio_sythesis/Task_enhancement/ # !PLACEHOLDER
23
+ csv_dir: csv_files
24
+ train_csv: csv_files/train.csv
25
+ valid_csv: csv_files/dev.csv
26
+ test_csv: csv_files/test.csv
27
+ skip_prep: false
28
+ sample_rate: 16000
29
+ task: enhance
30
+
31
+ dereverberate: false
32
+ shuffle_train_data: true
33
+
34
+ # Pretrained models
35
+ pretrained_model_path:
36
+ /netscratch/sagar/thesis/speechbrain/recipes/RescueSpeech/pre-trained/sepformer_dns_16k # !PLACEHOLDER # sepformer_dns_16k model
37
+
38
+ # Basic parameters
39
+ use_tensorboard: false
40
+ tensorboard_logs: results/sepformer-enhancement/8201/logs/
41
+
42
+ # Experiment params
43
+ auto_mix_prec: true # Set it to True for mixed precision
44
+ test_only: false
45
+ num_spks: 1
46
+ noprogressbar: false
47
+ save_audio: true # Save estimated sources on disk
48
+ downsample: false
49
+ n_audio_to_save: 500
50
+
51
+ # Training parameters
52
+ N_epochs: 150
53
+ batch_size: 1
54
+ batch_size_test: 1
55
+ lr: 0.00015
56
+ clip_grad_norm: 5
57
+ loss_upper_lim: 999999 # this is the upper limit for an acceptable loss
58
+ # if True, the training sequences are cut to a specified length
59
+ limit_training_signal_len: false
60
+ # this is the length of sequences if we choose to limit
61
+ # the signal length of training sequences
62
+ training_signal_len: 32000
63
+ ckpt_interval_minutes: 60
64
+
65
+ # Parameters for data augmentation
66
+ use_wavedrop: false
67
+ use_speedperturb: true
68
+ use_rand_shift: false
69
+ min_shift: -8000
70
+ max_shift: 8000
71
+
72
+ speedperturb: !new:speechbrain.lobes.augment.TimeDomainSpecAugment
73
+ perturb_prob: 1.0
74
+ drop_freq_prob: 0.0
75
+ drop_chunk_prob: 0.0
76
+ sample_rate: 16000
77
+ speeds: [95, 100, 105]
78
+
79
+ wavedrop: !new:speechbrain.lobes.augment.TimeDomainSpecAugment
80
+ perturb_prob: 0.0
81
+ drop_freq_prob: 1.0
82
+ drop_chunk_prob: 1.0
83
+ sample_rate: 16000
84
+
85
+ # loss thresholding -- this thresholds the training loss
86
+ threshold_byloss: true
87
+ threshold: -30
88
+
89
+ # Encoder parameters
90
+ N_encoder_out: 256
91
+ out_channels: 256
92
+ kernel_size: 16
93
+ kernel_stride: 8
94
+
95
+ # Dataloader options
96
+ dataloader_opts:
97
+ batch_size: 1
98
+ num_workers: 3
99
+
100
+ dataloader_opts_valid:
101
+ batch_size: 1
102
+ num_workers: 3
103
+
104
+ dataloader_opts_test:
105
+ batch_size: 1
106
+ num_workers: 3
107
+
108
+ # Specifying the network
109
+ Encoder: &id003 !new:speechbrain.lobes.models.dual_path.Encoder
110
+ kernel_size: 16
111
+ out_channels: 256
112
+
113
+ SBtfintra: &id001 !new:speechbrain.lobes.models.dual_path.SBTransformerBlock
114
+ num_layers: 8
115
+ d_model: 256
116
+ nhead: 8
117
+ d_ffn: 1024
118
+ dropout: 0
119
+ use_positional_encoding: true
120
+ norm_before: true
121
+
122
+ SBtfinter: &id002 !new:speechbrain.lobes.models.dual_path.SBTransformerBlock
123
+ num_layers: 8
124
+ d_model: 256
125
+ nhead: 8
126
+ d_ffn: 1024
127
+ dropout: 0
128
+ use_positional_encoding: true
129
+ norm_before: true
130
+
131
+ MaskNet: &id005 !new:speechbrain.lobes.models.dual_path.Dual_Path_Model
132
+
133
+ num_spks: 1
134
+ in_channels: 256
135
+ out_channels: 256
136
+ num_layers: 2
137
+ K: 250
138
+ intra_model: *id001
139
+ inter_model: *id002
140
+ norm: ln
141
+ linear_layer_after_inter_intra: false
142
+ skip_around_intra: true
143
+
144
+ Decoder: &id004 !new:speechbrain.lobes.models.dual_path.Decoder
145
+ in_channels: 256
146
+ out_channels: 1
147
+ kernel_size: 16
148
+ stride: 8
149
+ bias: false
150
+
151
+ optimizer: !name:torch.optim.Adam
152
+ lr: 0.00015
153
+ weight_decay: 0
154
+
155
+ loss: !name:speechbrain.nnet.losses.get_si_snr_with_pitwrapper
156
+
157
+ lr_scheduler: &id007 !new:speechbrain.nnet.schedulers.ReduceLROnPlateau
158
+
159
+ factor: 0.5
160
+ patience: 2
161
+ dont_halve_until_epoch: 85
162
+
163
+ epoch_counter: &id006 !new:speechbrain.utils.epoch_loop.EpochCounter
164
+ limit: 150
165
+
166
+ modules:
167
+ encoder: *id003
168
+ decoder: *id004
169
+ masknet: *id005
170
+ save_all_checkpoints: false
171
+ checkpointer: !new:speechbrain.utils.checkpoints.Checkpointer
172
+ checkpoints_dir: results/sepformer-enhancement/8201/save
173
+ recoverables:
174
+ encoder: *id003
175
+ decoder: *id004
176
+ masknet: *id005
177
+ counter: *id006
178
+ lr_scheduler: *id007
179
+ train_logger: !new:speechbrain.utils.train_logger.FileTrainLogger
180
+ save_file: results/sepformer-enhancement/8201/train_log.txt
181
+
182
+ ## Uncomment if you wish to fine-tune a pre-trained model.
183
+ pretrained_enhancement: !new:speechbrain.utils.parameter_transfer.Pretrainer
184
+ collect_in: results/sepformer-enhancement/8201/save
185
+ loadables:
186
+ encoder: *id003
187
+ decoder: *id004
188
+ masknet: *id005
189
+ paths:
190
+ encoder:
191
+ /netscratch/sagar/thesis/speechbrain/recipes/RescueSpeech/pre-trained/sepformer_dns_16k/encoder.ckpt
192
+ decoder:
193
+ /netscratch/sagar/thesis/speechbrain/recipes/RescueSpeech/pre-trained/sepformer_dns_16k/decoder.ckpt
194
+ masknet:
195
+ /netscratch/sagar/thesis/speechbrain/recipes/RescueSpeech/pre-trained/sepformer_dns_16k/masknet.ckpt