add crdnn model
Browse files- .gitattributes +3 -0
- README.md +119 -2
- asr.ckpt +3 -0
- config.json +3 -0
- hyperparams.yaml +145 -0
- normalizer.ckpt +3 -0
- tokenizer.ckpt +3 -0
.gitattributes
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*.zip filter=lfs diff=lfs merge=lfs -text
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*.zst filter=lfs diff=lfs merge=lfs -text
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*.zip filter=lfs diff=lfs merge=lfs -text
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*.zst filter=lfs diff=lfs merge=lfs -text
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*tfevents* filter=lfs diff=lfs merge=lfs -text
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asr.ckpt filter=lfs diff=lfs merge=lfs -text
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normalizer.ckpt filter=lfs diff=lfs merge=lfs -text
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tokenizer.ckpt filter=lfs diff=lfs merge=lfs -text
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README.md
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---
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-
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nguage: "en"
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thumbnail:
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tags:
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- automatic-speech-recognition
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- CTC
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- Attention
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- pytorch
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- speechbrain
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license: "apache-2.0"
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datasets:
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- switchboard
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metrics:
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- wer
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- ser
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---
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<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>
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<br/><br/>
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# CRDNN with CTC/Attention trained on Switchboard
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This repository provides all the necessary tools to perform automatic speech
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recognition from an end-to-end system pretrained on Switchboard (EN) within
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SpeechBrain. For a better experience we encourage you to learn more about
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[SpeechBrain](https://speechbrain.github.io).
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The performance of the model is the following:
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| Release | Swbd SER | Callhome SER | Eval2000 SER | Swbd WER | Callhome WER | Eval2000 WER | GPUs |
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|:--------:|:--------:|:------------:|:------------:|:--------:|:------------:|:------------:|:-----------:|
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| 17-09-22 | 61.93 | 65.89 | 64.44 | 16.01 | 25.12 | 20.71 | 1xA100 40GB |
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## Pipeline description
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This ASR system is composed with 2 different but linked blocks:
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- Tokenizer (unigram) that transforms words into subword units and trained with
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the train transcriptions of Switchboard.
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- Acoustic model (CRDNN + CTC/Attention). The CRDNN architecture is made of
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N blocks of convolutional neural networks with normalisation and pooling on the
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frequency domain. Then, a bidirectional LSTM is connected to a final DNN to obtain
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the final acoustic representation that is given to the CTC and attention decoders.
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The system is trained with recordings sampled at 16kHz (single channel).
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The code will automatically normalize your audio (i.e., resampling + mono channel selection) when calling *transcribe_file* if needed.
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## Install SpeechBrain
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First of all, please install SpeechBrain with the following command:
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```
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pip install speechbrain
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```
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Please notice that we encourage you to read our tutorials and learn more about
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[SpeechBrain](https://speechbrain.github.io).
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### Transcribing your own audio files (in English)
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```python
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from speechbrain.pretrained import EncoderDecoderASR
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asr_model = EncoderDecoderASR.from_hparams(source="speechbrain/asr-crdnn-switchboard", savedir="pretrained_models/speechbrain/asr-crdnn-switchboard")
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asr_model.transcribe_file('path/to/your/audiofile')
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```
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### Inference on GPU
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To perform inference on the GPU, add `run_opts={"device":"cuda"}` when calling the `from_hparams` method.
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## Parallel Inference on a Batch
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Please, [see this Colab notebook](https://colab.research.google.com/drive/1hX5ZI9S4jHIjahFCZnhwwQmFoGAi3tmu?usp=sharing) to figure out how to transcribe in parallel a batch of input sentences using a pre-trained model.
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### Training
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The model was trained with SpeechBrain (Commit hash: '2abd9f01').
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To train it from scratch follow these steps:
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1. Clone SpeechBrain:
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```bash
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git clone https://github.com/speechbrain/speechbrain/
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```
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2. Install it:
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```bash
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cd speechbrain
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pip install -r requirements.txt
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pip install -e .
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```
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3. Run Training:
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```bash
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cd recipes/Switchboard/ASR/seq2seq/
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python train.py hparams/train_BPE_1000.yaml --data_folder=your_data_folder
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```
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You can find our training results (models, logs, etc) [here](https://drive.google.com/drive/folders/1SAndjcThdkO-YQF8kvwPOXlQ6LMT71vt?usp=sharing).
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### Limitations
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The SpeechBrain team does not provide any warranty on the performance achieved by this model when used on other datasets.
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# **About SpeechBrain**
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- Website: https://speechbrain.github.io/
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- Code: https://github.com/speechbrain/speechbrain/
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- HuggingFace: https://huggingface.co/speechbrain/
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# **Citing SpeechBrain**
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Please, cite SpeechBrain if you use it for your research or business.
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```bibtex
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@misc{speechbrain,
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title={{SpeechBrain}: A General-Purpose Speech Toolkit},
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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},
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year={2021},
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eprint={2106.04624},
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archivePrefix={arXiv},
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primaryClass={eess.AS},
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note={arXiv:2106.04624}
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}
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```
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asr.ckpt
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version https://git-lfs.github.com/spec/v1
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oid sha256:693dd171d58f2c1ddc477b957dd4c696530b6d4f0de1b96691578967973f782f
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size 486220553
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config.json
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{
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"speechbrain_interface": "EncoderDecoderASR"
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}
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hyperparams.yaml
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# ############################################################################
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# Model: E2E ASR with attention-based ASR
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# Encoder: CRDNN model
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# Decoder: GRU + beamsearch
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# Tokens: BPE with unigram
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# Losses: CTC + NLL
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# Training: Switchboard
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# Authors: Ju-Chieh Chou, Mirco Ravanelli, Abdel Heba, Peter Plantinga,
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# Samuele Cornell 2020, Dominik Wagner 2022
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# ############################################################################
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# Feature parameters
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sample_rate: 16000
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n_fft: 400
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n_mels: 40
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# Model parameters
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activation: !name:torch.nn.LeakyReLU
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dropout: 0.15
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cnn_blocks: 2
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cnn_channels: (128, 256)
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inter_layer_pooling_size: (2, 2)
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cnn_kernelsize: (3, 3)
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time_pooling_size: 4
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rnn_class: !name:speechbrain.nnet.RNN.LSTM
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rnn_layers: 4
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rnn_neurons: 1024
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rnn_bidirectional: True
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dnn_blocks: 2
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dnn_neurons: 512
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emb_size: 128
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dec_neurons: 1024
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output_neurons: 2000 # Number of tokens used for tokenizer
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blank_index: 0
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bos_index: 1
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eos_index: 2
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# Decoding parameters
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min_decode_ratio: 0.0
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max_decode_ratio: 1.0
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valid_beam_size: 60
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test_beam_size: 80
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eos_threshold: 1.5
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using_max_attn_shift: True
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max_attn_shift: 240
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ctc_weight_decode: 0.3
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coverage_penalty: 1.8
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temperature: 1.25
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normalizer: !new:speechbrain.processing.features.InputNormalization
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norm_type: global
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compute_features: !new:speechbrain.lobes.features.Fbank
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sample_rate: !ref <sample_rate>
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n_fft: !ref <n_fft>
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n_mels: !ref <n_mels>
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enc: !new:speechbrain.lobes.models.CRDNN.CRDNN
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input_shape: [null, null, !ref <n_mels>]
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activation: !ref <activation>
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dropout: !ref <dropout>
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cnn_blocks: !ref <cnn_blocks>
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cnn_channels: !ref <cnn_channels>
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cnn_kernelsize: !ref <cnn_kernelsize>
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inter_layer_pooling_size: !ref <inter_layer_pooling_size>
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time_pooling: True
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using_2d_pooling: False
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time_pooling_size: !ref <time_pooling_size>
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rnn_class: !ref <rnn_class>
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rnn_layers: !ref <rnn_layers>
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rnn_neurons: !ref <rnn_neurons>
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rnn_bidirectional: !ref <rnn_bidirectional>
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rnn_re_init: True
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dnn_blocks: !ref <dnn_blocks>
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dnn_neurons: !ref <dnn_neurons>
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use_rnnp: False
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emb: !new:speechbrain.nnet.embedding.Embedding
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num_embeddings: !ref <output_neurons>
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embedding_dim: !ref <emb_size>
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dec: !new:speechbrain.nnet.RNN.AttentionalRNNDecoder
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enc_dim: !ref <dnn_neurons>
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input_size: !ref <emb_size>
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rnn_type: gru
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attn_type: location
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hidden_size: !ref <dec_neurons>
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attn_dim: 1024
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num_layers: 1
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scaling: 1.0
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channels: 10
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kernel_size: 100
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re_init: True
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dropout: !ref <dropout>
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ctc_lin: !new:speechbrain.nnet.linear.Linear
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input_size: !ref <dnn_neurons>
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n_neurons: !ref <output_neurons>
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seq_lin: !new:speechbrain.nnet.linear.Linear
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input_size: !ref <dec_neurons>
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n_neurons: !ref <output_neurons>
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log_softmax: !new:speechbrain.nnet.activations.Softmax
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apply_log: True
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tokenizer: !new:sentencepiece.SentencePieceProcessor
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asr_model: !new:torch.nn.ModuleList
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- [!ref <enc>, !ref <emb>, !ref <dec>, !ref <ctc_lin>, !ref <seq_lin>]
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encoder: !new:speechbrain.nnet.containers.LengthsCapableSequential
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input_shape: [null, null, !ref <n_mels>]
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compute_features: !ref <compute_features>
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normalize: !ref <normalizer>
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model: !ref <enc>
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decoder: !new:speechbrain.decoders.S2SRNNBeamSearcher
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embedding: !ref <emb>
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decoder: !ref <dec>
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linear: !ref <seq_lin>
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ctc_linear: !ref <ctc_lin>
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bos_index: !ref <bos_index>
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eos_index: !ref <eos_index>
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blank_index: !ref <blank_index>
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min_decode_ratio: !ref <min_decode_ratio>
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max_decode_ratio: !ref <max_decode_ratio>
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beam_size: !ref <test_beam_size>
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eos_threshold: !ref <eos_threshold>
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using_max_attn_shift: !ref <using_max_attn_shift>
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max_attn_shift: !ref <max_attn_shift>
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coverage_penalty: !ref <coverage_penalty>
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ctc_weight: !ref <ctc_weight_decode>
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temperature: !ref <temperature>
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modules:
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normalizer: !ref <normalizer>
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encoder: !ref <encoder>
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decoder: !ref <decoder>
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pretrainer: !new:speechbrain.utils.parameter_transfer.Pretrainer
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loadables:
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normalizer: !ref <normalizer>
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asr: !ref <asr_model>
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tokenizer: !ref <tokenizer>
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normalizer.ckpt
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version https://git-lfs.github.com/spec/v1
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oid sha256:5023bda59574203adc674d85efd143c263a5bf5739c1cdac39751345d1a377af
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size 1383
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tokenizer.ckpt
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