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Copy HL decoding script to HLG decoding script
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egs/librispeech/ASR/conformer_ctc/jit_pretrained_decode_with_HLG.py
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232
egs/librispeech/ASR/conformer_ctc/jit_pretrained_decode_with_HLG.py
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#!/usr/bin/env python3
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# Copyright 2023 Xiaomi Corp. (authors: Fangjun Kuang)
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"""
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This file shows how to use a torchscript model for decoding with H
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on CPU using OpenFST and decoders from kaldi.
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Usage:
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./conformer_ctc/jit_pretrained_decode_with_H.py \
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--nn-model ./conformer_ctc/exp/cpu_jit.pt \
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--HL ./data/lang_bpe_500/HL.fst \
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--words ./data/lang_bpe_500/words.txt \
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./download/LibriSpeech/test-clean/1089/134686/1089-134686-0002.flac \
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./download/LibriSpeech/test-clean/1221/135766/1221-135766-0001.flac
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Note that to generate ./conformer_ctc/exp/cpu_jit.pt,
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you can use ./export.py --jit 1
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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 Dict, List
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import kaldi_hmm_gmm
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import kaldifeat
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import kaldifst
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import torch
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import torchaudio
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from kaldi_hmm_gmm import DecodableCtc, FasterDecoder, FasterDecoderOptions
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from torch.nn.utils.rnn import pad_sequence
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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",
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type=str,
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required=True,
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help="""Path to the torchscript model.
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You can use ./conformer_ctc/export.py --jit 1
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to obtain it
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""",
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)
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parser.add_argument(
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"--words",
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type=str,
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required=True,
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help="Path to words.txt",
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)
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parser.add_argument("--HL", type=str, required=True, help="Path to HL.fst")
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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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)
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return parser
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def read_words(words_txt: str) -> Dict[int, str]:
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id2word = dict()
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with open(words_txt, encoding="utf-8") as f:
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for line in f:
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word, idx = line.strip().split()
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id2word[int(idx)] = word
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return id2word
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def read_sound_files(
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filenames: List[str], expected_sample_rate: float
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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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if sample_rate != expected_sample_rate:
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wave = torchaudio.functional.resample(
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wave,
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orig_freq=sample_rate,
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new_freq=expected_sample_rate,
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)
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# We use only the first channel
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ans.append(wave[0].contiguous())
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return ans
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def decode(
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filename: str,
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nnet_output: torch.Tensor,
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HL: kaldifst,
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id2word: Dict[int, str],
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) -> List[str]:
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"""
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Args:
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filename:
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Path to the filename for decoding. Used for debugging.
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nnet_output:
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A 2-D float32 tensor of shape (num_frames, vocab_size). It
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contains output from log_softmax.
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HL:
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The HL graph.
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word2token:
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A map mapping token ID to word string.
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Returns:
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Return a list of decoded words.
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"""
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logging.info(f"{filename}, {nnet_output.shape}")
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decodable = DecodableCtc(nnet_output.cpu())
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decoder_opts = FasterDecoderOptions(max_active=3000)
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decoder = FasterDecoder(HL, decoder_opts)
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decoder.decode(decodable)
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if not decoder.reached_final():
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print(f"failed to decode {filename}")
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return [""]
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ok, best_path = decoder.get_best_path()
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(
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ok,
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isymbols_out,
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osymbols_out,
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total_weight,
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) = kaldifst.get_linear_symbol_sequence(best_path)
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if not ok:
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print(f"failed to get linear symbol sequence for {filename}")
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return [""]
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# are shifted by 1 during graph construction
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hyps = [id2word[i] for i in osymbols_out]
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return hyps
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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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device = torch.device("cpu")
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logging.info(f"device: {device}")
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logging.info("Loading torchscript model")
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model = torch.jit.load(args.nn_model)
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model.eval()
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model.to(device)
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logging.info(f"Loading HL from {args.HL}")
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HL = kaldifst.StdVectorFst.read(args.HL)
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sample_rate = 16000
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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 = sample_rate
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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, expected_sample_rate=sample_rate
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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.shape[0] for f in features]
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feature_lengths = torch.tensor(feature_lengths)
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supervisions = dict()
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supervisions["sequence_idx"] = torch.arange(len(features))
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supervisions["start_frame"] = torch.zeros(len(features))
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supervisions["num_frames"] = feature_lengths
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features = pad_sequence(features, batch_first=True, padding_value=math.log(1e-10))
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nnet_output, _, _ = model(features, supervisions)
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feature_lengths = ((feature_lengths - 1) // 2 - 1) // 2
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id2word = read_words(args.words)
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hyps = []
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for i in range(nnet_output.shape[0]):
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hyp = decode(
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filename=args.sound_files[i],
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nnet_output=nnet_output[i, : feature_lengths[i]],
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HL=HL,
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id2word=id2word,
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)
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hyps.append(hyp)
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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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