w2v-model

.gitattributes

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 *.zip filter=lfs diff=lfs merge=lfs -text
 *.zst filter=lfs diff=lfs merge=lfs -text
 *tfevents* filter=lfs diff=lfs merge=lfs -text
+asr.ckpt filter=lfs diff=lfs merge=lfs -text
+tokenizer.ckpt filter=lfs diff=lfs merge=lfs -text
+wav2vec2.ckpt filter=lfs diff=lfs merge=lfs -text

README.md

@@ -1,3 +1,126 @@
 ---
+language:
+- en
+thumbnail: null
+tags:
+- automatic-speech-recognition
+- CTC
+- Attention
+- pytorch
+- speechbrain
 license: apache-2.0
+datasets:
+- switchboard
+metrics:
+- wer
+- cer
+
 ---
+
+<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>
+<br/><br/>
+
+# wav2vec 2.0 with CTC/Attention trained on Switchboard (No LM)
+
+This repository provides all the necessary tools to perform automatic speech recognition from an end-to-end system pretrained on the Switchboard (EN) corpus within SpeechBrain. 
+For a better experience, we encourage you to learn more about [SpeechBrain](https://speechbrain.github.io).
+
+The performance of the model is the following:
+
+| Release  | Swbd CER | Callhome CER | Eval2000 CER | Swbd WER | Callhome WER | Eval2000 WER | GPUs        |
+|:--------:|:--------:|:------------:|:------------:|:--------:|:------------:|:------------:|:-----------:|
+| 17-09-22 |  5.24   |  9.69       |   7.44      |  8 .76   |  14.67       |     11.78    | 4xA100 40GB |
+
+## Pipeline Description
+
+This ASR system is composed of 2 different but linked blocks:
+- Tokenizer (unigram) that transforms words into subword units trained on the Switchboard training transcriptions and the Fisher corpus. 
+- Acoustic model (wav2vec2.0 + CTC). A pretrained wav2vec 2.0 model ([facebook/wav2vec2-large-lv60](https://huggingface.co/facebook/wav2vec2-large-lv60)) is combined with a feature encoder consisting of three DNN layers and finetuned on Switchboard. The obtained final acoustic representation is given to a greedy CTC decoder.
+
+The system is trained with recordings sampled at 16kHz (single channel).   
+The code will automatically normalize your audio (i.e., resampling + mono channel selection) when calling `transcribe_file` if needed.
+
+## Install SpeechBrain
+
+First of all, please install tranformers and SpeechBrain with the following command:
+
+```
+pip install speechbrain transformers
+```
+
+Please notice that we encourage you to read our tutorials and learn more about
+[SpeechBrain](https://speechbrain.github.io).
+
+## Transcribing Your Own Audio Files
+
+```python
+from speechbrain.pretrained import EncoderASR
+
+asr_model = EncoderASR.from_hparams(source="speechbrain/asr-wav2vec2-switchboard", savedir="pretrained_models/asr-wav2vec2-switchboard")
+asr_model.transcribe_file('speechbrain/asr-wav2vec2-switchboard/example.wav')
+
+```
+
+## Inference on GPU
+
+To perform inference on the GPU, add  `run_opts={"device":"cuda"}`  when calling the `from_hparams` method.
+
+## Training
+
+The model was trained with SpeechBrain (commit hash: `70904d0`).
+To train it from scratch follow these steps:
+
+1. Clone SpeechBrain:
+```bash
+git clone https://github.com/speechbrain/speechbrain/
+```
+
+2. Install it:
+```bash
+cd speechbrain
+pip install -r requirements.txt
+pip install -e .
+```
+
+3. Run Training:
+```bash
+cd recipes/Switchboard/ASR/CTC
+python train_with_wav2vec.py hparams/train_with_wav2vec.yaml --data_folder=your_data_folder
+```
+
+## Limitations
+
+The SpeechBrain team does not provide any warranty on the performance achieved by this model when used on other datasets. 
+
+## Credits
+
+This model was trained with resources provided by the [THN Center for AI](https://www.th-nuernberg.de/en/kiz). 
+
+# About SpeechBrain
+
+SpeechBrain is an open-source and all-in-one speech toolkit. It is designed to be simple, extremely flexible, and user-friendly. 
+Competitive or state-of-the-art performance is obtained in various domains.
+
+- Website: https://speechbrain.github.io/
+- GitHub: https://github.com/speechbrain/speechbrain/
+- HuggingFace: https://huggingface.co/speechbrain/
+
+# Citing SpeechBrain
+
+Please cite SpeechBrain if you use it for your research or business.
+
+```bibtex
+@misc{speechbrain,
+  title={{SpeechBrain}: A General-Purpose Speech Toolkit},
+  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},
+  year={2021},
+  eprint={2106.04624},
+  archivePrefix={arXiv},
+  primaryClass={eess.AS},
+  note={arXiv:2106.04624}
+}
+```
+
+
+
+

asr.ckpt

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config.json

@@ -0,0 +1,76 @@
+{
+  "speechbrain_interface": "EncoderASR",
+  "activation_dropout": 0.0,
+  "apply_spec_augment": true,
+  "architectures": [
+    "Wav2Vec2Model"
+  ],
+  "attention_dropout": 0.1,
+  "bos_token_id": 1,
+  "conv_bias": true,
+  "conv_dim": [
+    512,
+    512,
+    512,
+    512,
+    512,
+    512,
+    512
+  ],
+  "conv_kernel": [
+    10,
+    3,
+    3,
+    3,
+    3,
+    2,
+    2
+  ],
+  "conv_stride": [
+    5,
+    2,
+    2,
+    2,
+    2,
+    2,
+    2
+  ],
+  "ctc_loss_reduction": "sum",
+  "ctc_zero_infinity": false,
+  "do_stable_layer_norm": true,
+  "eos_token_id": 2,
+  "feat_extract_activation": "gelu",
+  "feat_extract_dropout": 0.0,
+  "feat_extract_norm": "layer",
+  "feat_proj_dropout": 0.1,
+  "final_dropout": 0.0,
+  "gradient_checkpointing": false,
+  "hidden_act": "gelu",
+  "hidden_dropout": 0.1,
+  "hidden_size": 1024,
+  "initializer_range": 0.02,
+  "intermediate_size": 4096,
+  "layer_norm_eps": 1e-05,
+  "layerdrop": 0.1,
+  "mask_channel_length": 10,
+  "mask_channel_min_space": 1,
+  "mask_channel_other": 0.0,
+  "mask_channel_prob": 0.0,
+  "mask_channel_selection": "static",
+  "mask_feature_length": 10,
+  "mask_feature_prob": 0.0,
+  "mask_time_length": 10,
+  "mask_time_min_space": 1,
+  "mask_time_other": 0.0,
+  "mask_time_prob": 0.075,
+  "mask_time_selection": "static",
+  "model_type": "wav2vec2",
+  "num_attention_heads": 16,
+  "num_conv_pos_embedding_groups": 16,
+  "num_conv_pos_embeddings": 128,
+  "num_feat_extract_layers": 7,
+  "num_hidden_layers": 24,
+  "pad_token_id": 0,
+  "transformers_version": "4.5.1",
+  "vocab_size": 32
+}

hyperparams.yaml

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+# ################################
+# Model: wav2vec2 + DNN + CTC
+# Augmentation: SpecAugment
+# Authors: Titouan Parcollet 2021, Dominik Wagner 2022
+# ################################
+
+wav2vec2_hub: facebook/wav2vec2-large-lv60
+sample_rate: 16000
+
+# BPE parameters
+token_type: unigram  # ["unigram", "bpe", "char"]
+character_coverage: 1.0
+
+# Model parameters
+wav2vec_output_dim: 1024
+dnn_neurons: 1024
+freeze_wav2vec: False
+dropout: 0.15
+
+# Outputs
+output_neurons: 1000  # BPE size, index(blank/eos/bos) = 0
+
+# Decoding parameters
+# Be sure that the bos and eos index match with the BPEs ones
+blank_index: 0
+bos_index: 1
+eos_index: 2
+
+enc: !new:speechbrain.nnet.containers.Sequential
+  input_shape: [null, null, !ref <wav2vec_output_dim>]
+  linear1: !name:speechbrain.nnet.linear.Linear
+    n_neurons: !ref <dnn_neurons>
+    bias: True
+  bn1: !name:speechbrain.nnet.normalization.BatchNorm1d
+  activation: !new:torch.nn.LeakyReLU
+  drop: !new:torch.nn.Dropout
+    p: !ref <dropout>
+  linear2: !name:speechbrain.nnet.linear.Linear
+    n_neurons: !ref <dnn_neurons>
+    bias: True
+  bn2: !name:speechbrain.nnet.normalization.BatchNorm1d
+  activation2: !new:torch.nn.LeakyReLU
+  drop2: !new:torch.nn.Dropout
+    p: !ref <dropout>
+  linear3: !name:speechbrain.nnet.linear.Linear
+    n_neurons: !ref <dnn_neurons>
+    bias: True
+  bn3: !name:speechbrain.nnet.normalization.BatchNorm1d
+  activation3: !new:torch.nn.LeakyReLU
+
+wav2vec2: !new:speechbrain.lobes.models.huggingface_wav2vec.HuggingFaceWav2Vec2
+  source: !ref <wav2vec2_hub>
+  output_norm: True
+  freeze: !ref <freeze_wav2vec>
+  save_path: wav2vec2_checkpoint
+
+ctc_lin: !new:speechbrain.nnet.linear.Linear
+  input_size: !ref <dnn_neurons>
+  n_neurons: !ref <output_neurons>
+
+log_softmax: !new:speechbrain.nnet.activations.Softmax
+  apply_log: True
+
+ctc_cost: !name:speechbrain.nnet.losses.ctc_loss
+  blank_index: !ref <blank_index>
+
+asr_model: !new:torch.nn.ModuleList
+    - [!ref <enc>, !ref <ctc_lin>]
+
+tokenizer: !new:sentencepiece.SentencePieceProcessor
+
+encoder: !new:speechbrain.nnet.containers.LengthsCapableSequential
+    wav2vec2: !ref <wav2vec2>
+    enc: !ref <enc>
+    ctc_lin: !ref <ctc_lin>
+
+modules:
+  encoder: !ref <encoder>
+
+decoding_function: !name:speechbrain.decoders.ctc_greedy_decode
+    blank_id: !ref <blank_index>
+
+pretrainer: !new:speechbrain.utils.parameter_transfer.Pretrainer
+   loadables:
+      wav2vec2: !ref <wav2vec2>
+      asr: !ref <asr_model>
+      tokenizer: !ref <tokenizer>
+

preprocessor_config.json

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+{
+  "do_normalize": true,
+  "feature_size": 1,
+  "padding_side": "right",
+  "padding_value": 0.0,
+  "return_attention_mask": true,
+  "sampling_rate": 16000
+}

tokenizer.ckpt

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wav2vec2.ckpt

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