new conformer

README.md

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+---
+language:
+- en
+thumbnail: null
+tags:
+- automatic-speech-recognition
+- CTC
+- Attention
+- Transformer
+- Conformer
+- pytorch
+- speechbrain
+- hf-asr-leaderboard
+license: apache-2.0
+datasets:
+- librispeech
+metrics:
+- wer
+- cer
+model-index:
+- name: Transformer+TransformerLM by SpeechBrain
+  results:
+  - task:
+      name: Automatic Speech Recognition
+      type: automatic-speech-recognition
+    dataset:
+      name: LibriSpeech (clean)
+      type: librispeech_asr
+      config: clean
+      split: test
+      args: 
+        language: en
+    metrics:
+    - name: Test WER
+      type: wer
+      value: 2.0
+  - task:
+      name: Automatic Speech Recognition
+      type: automatic-speech-recognition
+    dataset:
+      name: LibriSpeech (other)
+      type: librispeech_asr
+      config: other
+      split: test
+      args: 
+        language: en
+    metrics:
+    - name: Test WER
+      type: wer
+      value: 4.5
+---
+
+<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/>
+
+# Transformer for LibriSpeech (with Transformer LM)
+
+This repository provides all the necessary tools to perform automatic speech
+recognition from an end-to-end system pretrained on LibriSpeech (EN) 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 | Test clean WER | Test other WER | GPUs |
+|:-------------:|:--------------:|:--------------:|:--------:|
+| 21-06-23 | 2.01 | 4.52 | 4xA100 80GB |
+
+## Pipeline description
+
+This ASR system is composed of 3 different but linked blocks:
+- Tokenizer (unigram) that transforms words into subword units and trained with
+the train transcriptions of LibriSpeech.
+- Neural language model (Transformer LM) trained on the full 10M words dataset.
+- Acoustic model made of a conformer encoder and a joint decoder with CTC +
+transformer. Hence, the decoding also incorporates the CTC probabilities.
+
+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 SpeechBrain with the following command:
+
+```
+pip install speechbrain
+```
+
+Please notice that we encourage you to read our tutorials and learn more about
+[SpeechBrain](https://speechbrain.github.io).
+
+### Transcribing your own audio files (in English)
+
+```python
+from speechbrain.pretrained import EncoderDecoderASR
+
+asr_model = EncoderDecoderASR.from_hparams(source="speechbrain/asr-conformer-transformerlm-librispeech", savedir="pretrained_models/asr-transformer-transformerlm-librispeech")
+asr_model.transcribe_file("speechbrain/asr-conformer-transformerlm-librispeech/example.wav")
+
+```
+### Inference on GPU
+To perform inference on the GPU, add  `run_opts={"device":"cuda"}`  when calling the `from_hparams` method.
+
+## Parallel Inference on a Batch
+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.
+
+### Training
+The model was trained with SpeechBrain (Commit hash: 'f73fcc35').
+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/LibriSpeech/ASR/transformer
+python train.py hparams/transformer.yaml --data_folder=your_data_folder
+```
+
+You can find our training results (models, logs, etc) [here](https://drive.google.com/drive/folders/1Nv1OLbHLqVeShyZ8LY9gjhYGE1DBFzFf?usp=sharing).
+
+### Limitations
+The SpeechBrain team does not provide any warranty on the performance achieved by this model when used on other datasets.
+
+# **About SpeechBrain**
+- Website: https://speechbrain.github.io/
+- Code: 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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+version https://git-lfs.github.com/spec/v1
+oid sha256:a738d0f06f29e248ee05331faa7db7e54564f65deb3c0d3e36d854334126b10a
+size 441842873

config.json

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+{
+  "speechbrain_interface": "EncoderDecoderASR"
+}

hyperparams.yaml

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+# ############################################################################
+# Model: E2E ASR with Transformer
+# Encoder: Transformer Encoder
+# Decoder: Transformer Decoder + (CTC/ATT joint) beamsearch + TransformerLM
+# Tokens: unigram
+# losses: CTC + KLdiv (Label Smoothing loss)
+# Training: Librispeech 960h
+# Authors:  Jianyuan Zhong, Titouan Parcollet 2021
+# ############################################################################
+
+# Feature parameters
+sample_rate: 16000
+n_fft: 400
+n_mels: 80
+
+####################### Model parameters ###########################
+# Transformer
+d_model: 512
+nhead: 4
+num_encoder_layers: 12
+num_decoder_layers: 6
+d_ffn: 2048
+transformer_dropout: 0.1
+activation: !name:torch.nn.GELU
+output_neurons: 5000
+vocab_size: 5000
+
+# Outputs
+blank_index: 0
+label_smoothing: 0.0
+pad_index: 0
+bos_index: 1
+eos_index: 2
+
+# Decoding parameters
+min_decode_ratio: 0.0
+max_decode_ratio: 1.0
+valid_search_interval: 10
+valid_beam_size: 10
+test_beam_size: 66
+lm_weight: 0.60
+ctc_weight_decode: 0.40
+
+############################## models ################################
+
+CNN: !new:speechbrain.lobes.models.convolution.ConvolutionFrontEnd
+    input_shape: (8, 10, 80)
+    num_blocks: 3
+    num_layers_per_block: 1
+    out_channels: (64, 64, 64)
+    kernel_sizes: (5, 5, 1)
+    strides: (2, 2, 1)
+    residuals: (False, False, True)
+    norm: !name:speechbrain.nnet.normalization.LayerNorm
+
+Transformer: !new:speechbrain.lobes.models.transformer.TransformerASR.TransformerASR
+    input_size: 1280
+    tgt_vocab: !ref <output_neurons>
+    d_model: !ref <d_model>
+    nhead: !ref <nhead>
+    num_encoder_layers: !ref <num_encoder_layers>
+    num_decoder_layers: !ref <num_decoder_layers>
+    d_ffn: !ref <d_ffn>
+    dropout: !ref <transformer_dropout>
+    activation: !ref <activation>
+    encoder_module: transformer
+    attention_type: regularMHA
+    normalize_before: True
+    causal: False
+
+ctc_lin: !new:speechbrain.nnet.linear.Linear
+    input_size: !ref <d_model>
+    n_neurons: !ref <output_neurons>
+
+seq_lin: !new:speechbrain.nnet.linear.Linear
+    input_size: !ref <d_model>
+    n_neurons: !ref <output_neurons>
+
+decoder: !new:speechbrain.decoders.S2STransformerBeamSearch
+    modules: [!ref <Transformer>, !ref <seq_lin>, !ref <ctc_lin>]
+    bos_index: !ref <bos_index>
+    eos_index: !ref <eos_index>
+    blank_index: !ref <blank_index>
+    min_decode_ratio: !ref <min_decode_ratio>
+    max_decode_ratio: !ref <max_decode_ratio>
+    beam_size: !ref <test_beam_size>
+    ctc_weight: !ref <ctc_weight_decode>
+    lm_weight: !ref <lm_weight>
+    lm_modules: !ref <lm_model>
+    temperature: 1.15
+    temperature_lm: 1.15
+    using_eos_threshold: False
+    length_normalization: True
+
+log_softmax: !new:torch.nn.LogSoftmax
+    dim: -1
+
+normalizer: !new:speechbrain.processing.features.InputNormalization
+    norm_type: global
+
+compute_features: !new:speechbrain.lobes.features.Fbank
+    sample_rate: !ref <sample_rate>
+    n_fft: !ref <n_fft>
+    n_mels: !ref <n_mels>
+
+# This is the Transformer LM that is used according to the Huggingface repository
+# Visit the HuggingFace model corresponding to the pretrained_lm_tokenizer_path
+# For more details about the model!
+# NB: It has to match the pre-trained TransformerLM!!
+lm_model: !new:speechbrain.lobes.models.transformer.TransformerLM.TransformerLM
+    vocab: 5000
+    d_model: 768
+    nhead: 12
+    num_encoder_layers: 12
+    num_decoder_layers: 0
+    d_ffn: 3072
+    dropout: 0.0
+    activation: !name:torch.nn.GELU
+    normalize_before: False
+
+tokenizer: !new:sentencepiece.SentencePieceProcessor
+
+Tencoder: !new:speechbrain.lobes.models.transformer.TransformerASR.EncoderWrapper
+    transformer: !ref <Transformer>
+
+encoder: !new:speechbrain.nnet.containers.LengthsCapableSequential
+    input_shape: [null, null, !ref <n_mels>]
+    compute_features: !ref <compute_features>
+    normalize: !ref <normalizer>
+    cnn: !ref <CNN>
+    transformer_encoder: !ref <Tencoder>
+
+# Models
+asr_model: !new:torch.nn.ModuleList
+    - [!ref <CNN>, !ref <Transformer>, !ref <seq_lin>, !ref <ctc_lin>]
+
+modules:
+   compute_features: !ref <compute_features>
+   normalizer: !ref <normalizer>
+   pre_transformer: !ref <CNN>
+   transformer: !ref <Transformer>
+   asr_model: !ref <asr_model>
+   lm_model: !ref <lm_model>
+   encoder: !ref <encoder>
+   decoder: !ref <decoder>
+
+# The pretrainer allows a mapping between pretrained files and instances that
+# are declared in the yaml.
+pretrainer: !new:speechbrain.utils.parameter_transfer.Pretrainer
+   loadables:
+      normalizer: !ref <normalizer>
+      asr: !ref <asr_model>
+      lm: !ref <lm_model>
+      tokenizer: !ref <tokenizer>

lm.ckpt

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