Upload 7 files
Browse filesASR-API version 1.0.0
- ASR_model/infer.py +41 -0
- ASR_model/inference/asr.ckpt +3 -0
- ASR_model/inference/hyperparams.yaml +139 -0
- ASR_model/inference/lm.ckpt +3 -0
- ASR_model/inference/normalizer.ckpt +3 -0
- ASR_model/inference/tokenizer.ckpt +3 -0
- ASR_model/test.wav +0 -0
ASR_model/infer.py
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from typing import List
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import torch
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import argparse
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import shutil
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import tempfile
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from speechbrain.pretrained import EncoderDecoderASR
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def asr_model_inference(model: EncoderDecoderASR, audios: List[str]) -> List[str]:
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"""
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convert input audio to words and return the result
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"""
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tmp_dir = tempfile.mkdtemp()
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results = [process_audio(model, audio, tmp_dir) for audio in audios]
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shutil.rmtree(tmp_dir)
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return results
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def process_audio(model: EncoderDecoderASR, audio: str, savedir:str) -> str:
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"""
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convert input audio to words and return the result
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"""
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waveform = model.load_audio(audio, savedir=savedir)
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# Fake a batch:
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batch = waveform.unsqueeze(0)
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rel_length = torch.tensor([1.0])
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predicted_words, predicted_tokens = model.transcribe_batch(
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batch, rel_length
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)
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return predicted_words[0]
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if __name__ == "__main__":
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parser = argparse.ArgumentParser()
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parser.add_argument("-I", dest="audio_file", required=True)
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args = parser.parse_args()
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asr_model = EncoderDecoderASR.from_hparams(
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source="./inference", hparams_file="hyperparams.yaml", savedir="inference", run_opts={"device": "cpu"})
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print(asr_model_inference(asr_model, [args.audio_file]))
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ASR_model/inference/asr.ckpt
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version https://git-lfs.github.com/spec/v1
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oid sha256:5533d036f8c2922e4e0246d4543b1936f9b1d80df1e09f624e927f5609e8f75f
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size 126714188
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ASR_model/inference/hyperparams.yaml
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tokenizer: !new:sentencepiece.SentencePieceProcessor
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pretrainer: !new:speechbrain.utils.parameter_transfer.Pretrainer
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loadables:
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lm: !ref <lm_model>
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tokenizer: !ref <tokenizer>
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normalizer: !ref <normalizer>
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asr: !ref <asr_model>
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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: 80
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hop_length: 20
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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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hop_length: !ref <hop_length>
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####################### Model parameters ###########################
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# Transformer
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d_model: 256
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nhead: 4
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num_encoder_layers: 12
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num_decoder_layers: 6
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d_ffn: 2048
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transformer_dropout: 0.1
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activation: !name:torch.nn.GELU
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output_neurons: 5000
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vocab_size: 5000
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# Outputs
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blank_index: 0
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label_smoothing: 0.1
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pad_index: 0
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bos_index: 1
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eos_index: 2
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unk_index: 0
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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_search_interval: 10
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valid_beam_size: 10
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test_beam_size: 10
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ctc_weight_decode: 0.3
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lm_weight: 0.2
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############################## models ################################
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CNN: !new:speechbrain.lobes.models.convolution.ConvolutionFrontEnd
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input_shape: !!python/tuple [8, 10, 8]
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num_blocks: 2
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num_layers_per_block: 1
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out_channels: !!python/tuple [256, 256]
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kernel_sizes: !!python/tuple [3, 3]
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strides: !!python/tuple [2, 2]
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residuals: !!python/tuple [False, False]
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Transformer:
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!new:speechbrain.lobes.models.transformer.TransformerASR.TransformerASR # yamllint disable-line rule:line-length
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input_size: 5120
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tgt_vocab: !ref <output_neurons>
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d_model: !ref <d_model>
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nhead: !ref <nhead>
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num_encoder_layers: !ref <num_encoder_layers>
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num_decoder_layers: !ref <num_decoder_layers>
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d_ffn: !ref <d_ffn>
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dropout: !ref <transformer_dropout>
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activation: !ref <activation>
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normalize_before: True
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lm_model:
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!new:speechbrain.lobes.models.transformer.TransformerLM.TransformerLM # yamllint disable-line rule:line-length
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vocab: !ref <output_neurons>
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d_model: 576
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nhead: 6
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num_encoder_layers: 6
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num_decoder_layers: 0
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d_ffn: 1538
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dropout: 0.2
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activation: !name:torch.nn.GELU
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normalize_before: False
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ctc_lin: !new:speechbrain.nnet.linear.Linear
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input_size: !ref <d_model>
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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 <d_model>
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n_neurons: !ref <output_neurons>
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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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cnn: !ref <CNN>
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transformer_encoder: !ref <Tencoder>
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asr_model: !new:torch.nn.ModuleList
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- [!ref <CNN>, !ref <Transformer>, !ref <seq_lin>, !ref <ctc_lin>]
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decoder: !new:speechbrain.decoders.S2STransformerBeamSearch
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modules: [!ref <Transformer>, !ref <seq_lin>, !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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ctc_weight: !ref <ctc_weight_decode>
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lm_weight: !ref <lm_weight>
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lm_modules: !ref <lm_model>
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temperature: 1.15
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temperature_lm: 1.15
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using_eos_threshold: False
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length_normalization: True
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Tencoder:
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!new:speechbrain.lobes.models.transformer.TransformerASR.EncoderWrapper
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transformer: !ref <Transformer>
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normalizer: !new:speechbrain.processing.features.InputNormalization
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norm_type: global
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update_until_epoch: 4
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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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# define two optimizers here for two-stage training
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log_softmax: !new:torch.nn.LogSoftmax
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dim: -1
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ASR_model/inference/lm.ckpt
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version https://git-lfs.github.com/spec/v1
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oid sha256:44e9e1dbc2935debdd681293515d27a3bb93578e8f4e3b7e01ee1d87c47bb10c
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size 104725990
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ASR_model/inference/normalizer.ckpt
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version https://git-lfs.github.com/spec/v1
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oid sha256:99a037e5582c33311d233f5465436f79de0331230e5ec433b9534a928325de69
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size 1783
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ASR_model/inference/tokenizer.ckpt
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version https://git-lfs.github.com/spec/v1
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oid sha256:448a66ac83788337506bff24c5be0f0ee97d59491af4f92e2341eca40a3b832c
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size 288715
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ASR_model/test.wav
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Binary file (263 kB). View file
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