update processor and readme
Browse files- README.md +7 -5
- config.json +3 -2
- preprocessor_config.json +3 -3
- pytorch_model.bin +2 -2
README.md
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@@ -35,16 +35,18 @@ transcripts by passing the speech features to the model.
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*Note: The `Speech2TextProcessor` object uses [torchaudio](https://github.com/pytorch/audio) to extract the
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filter bank features. Make sure to install the `torchaudio` package before running this example.*
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```python
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import torch
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from transformers import Speech2TextProcessor,
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from datasets import load_dataset
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import soundfile as sf
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model =
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processor = Speech2Textprocessor.from_pretrained("facebook/s2t-large-librispeech-asr")
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def map_to_array(batch):
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@@ -76,13 +78,13 @@ The following script shows how to evaluate this model on the [LibriSpeech](https
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```python
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from datasets import load_dataset, load_metric
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from transformers import
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import soundfile as sf
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librispeech_eval = load_dataset("librispeech_asr", "clean", split="test") # change to "other" for other test dataset
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wer = load_metric("wer")
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model =
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processor = Speech2TextProcessor.from_pretrained("facebook/s2t-large-librispeech-asr", do_upper_case=True)
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def map_to_array(batch):
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*Note: The `Speech2TextProcessor` object uses [torchaudio](https://github.com/pytorch/audio) to extract the
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filter bank features. Make sure to install the `torchaudio` package before running this example.*
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You could either install those as extra speech dependancies with
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`pip install transformers"[speech, sentencepiece]"` or install the packages seperatly
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with `pip install torchaudio sentencepiece`.
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```python
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import torch
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from transformers import Speech2TextProcessor, Speech2TextForConditionalGeneration
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from datasets import load_dataset
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import soundfile as sf
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model = Speech2TextForConditionalGeneration.from_pretrained("facebook/s2t-large-librispeech-asr")
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processor = Speech2Textprocessor.from_pretrained("facebook/s2t-large-librispeech-asr")
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def map_to_array(batch):
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```python
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from datasets import load_dataset, load_metric
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from transformers import Speech2TextForConditionalGeneration, Speech2TextProcessor
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import soundfile as sf
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librispeech_eval = load_dataset("librispeech_asr", "clean", split="test") # change to "other" for other test dataset
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wer = load_metric("wer")
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model = Speech2TextForConditionalGeneration.from_pretrained("facebook/s2t-large-librispeech-asr").to("cuda")
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processor = Speech2TextProcessor.from_pretrained("facebook/s2t-large-librispeech-asr", do_upper_case=True)
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def map_to_array(batch):
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config.json
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{
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"activation_dropout": 0.2,
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"activation_function": "relu",
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"architectures": [
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"
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],
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"attention_dropout": 0.2,
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"bos_token_id": 0,
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"max_length": 200,
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"max_source_positions": 6000,
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"max_target_positions": 1024,
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"model_type": "
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"num_beams": 5,
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"num_conv_layers": 2,
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"num_hidden_layers": 12,
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{
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"_name_or_path": "hf_models_fb/s2t-large-librispeech-asr/",
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"activation_dropout": 0.2,
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"activation_function": "relu",
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"architectures": [
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"Speech2TextForConditionalGeneration"
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],
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"attention_dropout": 0.2,
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"bos_token_id": 0,
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"max_length": 200,
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"max_source_positions": 6000,
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"max_target_positions": 1024,
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"model_type": "speech_to_text",
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"num_beams": 5,
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"num_conv_layers": 2,
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"num_hidden_layers": 12,
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preprocessor_config.json
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{
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"feature_size": 80,
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"num_mel_bins": 80,
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"padding_side": "right",
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"padding_value": 0.0,
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{
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"do_ceptral_normalize": true,
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"feature_size": 80,
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"normalize_means": true,
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"normalize_vars": true,
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"num_mel_bins": 80,
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"padding_side": "right",
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"padding_value": 0.0,
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pytorch_model.bin
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version https://git-lfs.github.com/spec/v1
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size
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version https://git-lfs.github.com/spec/v1
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size 1071459644
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