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* Add pruned_transducer_stateless7 for Aishell * update README.md * update comments and small fixes
279 lines
7.5 KiB
Python
Executable File
279 lines
7.5 KiB
Python
Executable File
#!/usr/bin/env python3
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# Copyright 2022 Xiaomi Corp. (authors: Fangjun Kuang)
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#
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# See ../../../../LICENSE for clarification regarding multiple authors
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#
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# Licensed under the Apache License, Version 2.0 (the "License");
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# you may not use this file except in compliance with the License.
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# You may obtain a copy of the License at
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#
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# http://www.apache.org/licenses/LICENSE-2.0
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#
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# Unless required by applicable law or agreed to in writing, software
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# distributed under the License is distributed on an "AS IS" BASIS,
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# WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
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# See the License for the specific language governing permissions and
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# limitations under the License.
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"""
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This script loads torchscript models, exported by `torch.jit.script()`
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and uses them to decode waves.
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You can use the following command to get the exported models:
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./pruned_transducer_stateless7/export.py \
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--exp-dir ./pruned_transducer_stateless7/exp \
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--lang-dir ./data/lang_char \
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--epoch 20 \
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--avg 10 \
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--jit 1
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Usage of this script:
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./pruned_transducer_stateless7/jit_pretrained.py \
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--nn-model-filename ./pruned_transducer_stateless7/exp/cpu_jit.pt \
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--lang-dir ./data/lang_char \
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/path/to/foo.wav \
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/path/to/bar.wav
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"""
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import argparse
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import logging
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import math
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from typing import List
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import kaldifeat
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import sentencepiece as spm
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import torch
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import torchaudio
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from torch.nn.utils.rnn import pad_sequence
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from icefall.lexicon import Lexicon
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def get_parser():
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parser = argparse.ArgumentParser(
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formatter_class=argparse.ArgumentDefaultsHelpFormatter
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)
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parser.add_argument(
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"--nn-model-filename",
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type=str,
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required=True,
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help="Path to the torchscript model cpu_jit.pt",
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)
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parser.add_argument(
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"--lang-dir",
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type=str,
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help="""The lang dir
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It contains language related input files such as
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"lexicon.txt"
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""",
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)
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parser.add_argument(
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"sound_files",
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type=str,
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nargs="+",
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help="The input sound file(s) to transcribe. "
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"Supported formats are those supported by torchaudio.load(). "
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"For example, wav and flac are supported. "
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"The sample rate has to be 16kHz.",
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)
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return parser
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def read_sound_files(
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filenames: List[str], expected_sample_rate: float = 16000
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) -> List[torch.Tensor]:
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"""Read a list of sound files into a list 1-D float32 torch tensors.
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Args:
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filenames:
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A list of sound filenames.
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expected_sample_rate:
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The expected sample rate of the sound files.
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Returns:
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Return a list of 1-D float32 torch tensors.
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"""
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ans = []
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for f in filenames:
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wave, sample_rate = torchaudio.load(f)
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assert (
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sample_rate == expected_sample_rate
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), f"expected sample rate: {expected_sample_rate}. Given: {sample_rate}"
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# We use only the first channel
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ans.append(wave[0])
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return ans
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def greedy_search(
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model: torch.jit.ScriptModule,
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encoder_out: torch.Tensor,
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encoder_out_lens: torch.Tensor,
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) -> List[List[int]]:
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"""Greedy search in batch mode. It hardcodes --max-sym-per-frame=1.
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Args:
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model:
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The transducer model.
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encoder_out:
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A 3-D tensor of shape (N, T, C)
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encoder_out_lens:
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A 1-D tensor of shape (N,).
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Returns:
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Return the decoded results for each utterance.
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"""
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assert encoder_out.ndim == 3
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assert encoder_out.size(0) >= 1, encoder_out.size(0)
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packed_encoder_out = torch.nn.utils.rnn.pack_padded_sequence(
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input=encoder_out,
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lengths=encoder_out_lens.cpu(),
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batch_first=True,
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enforce_sorted=False,
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)
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device = encoder_out.device
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blank_id = 0 # hard-code to 0
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batch_size_list = packed_encoder_out.batch_sizes.tolist()
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N = encoder_out.size(0)
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assert torch.all(encoder_out_lens > 0), encoder_out_lens
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assert N == batch_size_list[0], (N, batch_size_list)
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context_size = model.decoder.context_size
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hyps = [[blank_id] * context_size for _ in range(N)]
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decoder_input = torch.tensor(
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hyps,
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device=device,
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dtype=torch.int64,
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) # (N, context_size)
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decoder_out = model.decoder(
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decoder_input,
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need_pad=torch.tensor([False]),
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).squeeze(1)
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offset = 0
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for batch_size in batch_size_list:
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start = offset
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end = offset + batch_size
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current_encoder_out = packed_encoder_out.data[start:end]
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current_encoder_out = current_encoder_out
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# current_encoder_out's shape: (batch_size, encoder_out_dim)
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offset = end
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decoder_out = decoder_out[:batch_size]
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logits = model.joiner(
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current_encoder_out,
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decoder_out,
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)
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# logits'shape (batch_size, vocab_size)
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assert logits.ndim == 2, logits.shape
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y = logits.argmax(dim=1).tolist()
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emitted = False
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for i, v in enumerate(y):
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if v != blank_id:
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hyps[i].append(v)
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emitted = True
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if emitted:
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# update decoder output
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decoder_input = [h[-context_size:] for h in hyps[:batch_size]]
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decoder_input = torch.tensor(
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decoder_input,
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device=device,
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dtype=torch.int64,
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)
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decoder_out = model.decoder(
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decoder_input,
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need_pad=torch.tensor([False]),
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)
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decoder_out = decoder_out.squeeze(1)
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sorted_ans = [h[context_size:] for h in hyps]
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ans = []
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unsorted_indices = packed_encoder_out.unsorted_indices.tolist()
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for i in range(N):
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ans.append(sorted_ans[unsorted_indices[i]])
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return ans
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@torch.no_grad()
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def main():
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parser = get_parser()
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args = parser.parse_args()
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logging.info(vars(args))
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device = torch.device("cpu")
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if torch.cuda.is_available():
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device = torch.device("cuda", 0)
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logging.info(f"device: {device}")
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model = torch.jit.load(args.nn_model_filename)
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model.eval()
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model.to(device)
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lexicon = Lexicon(args.lang_dir)
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token_table = lexicon.token_table
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logging.info("Constructing Fbank computer")
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opts = kaldifeat.FbankOptions()
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opts.device = device
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opts.frame_opts.dither = 0
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opts.frame_opts.snip_edges = False
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opts.frame_opts.samp_freq = 16000
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opts.mel_opts.num_bins = 80
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fbank = kaldifeat.Fbank(opts)
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logging.info(f"Reading sound files: {args.sound_files}")
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waves = read_sound_files(
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filenames=args.sound_files,
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)
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waves = [w.to(device) for w in waves]
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logging.info("Decoding started")
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features = fbank(waves)
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feature_lengths = [f.size(0) for f in features]
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features = pad_sequence(
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features,
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batch_first=True,
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padding_value=math.log(1e-10),
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)
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feature_lengths = torch.tensor(feature_lengths, device=device)
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encoder_out, encoder_out_lens = model.encoder(
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x=features,
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x_lens=feature_lengths,
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)
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hyps = greedy_search(
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model=model,
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encoder_out=encoder_out,
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encoder_out_lens=encoder_out_lens,
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)
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hyps = [[token_table[t] for t in tokens] for tokens in hyps]
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s = "\n"
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for filename, hyp in zip(args.sound_files, hyps):
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words = " ".join(hyp)
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s += f"{filename}:\n{words}\n\n"
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logging.info(s)
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logging.info("Decoding Done")
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if __name__ == "__main__":
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formatter = "%(asctime)s %(levelname)s [%(filename)s:%(lineno)d] %(message)s"
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logging.basicConfig(format=formatter, level=logging.INFO)
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main()
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