Add decoding with H and HL for LibriSpeech

.github/scripts/run-pre-trained-conformer-ctc.sh

@@ -44,3 +44,46 @@ log "HLG decoding"
   $repo/test_wavs/1089-134686-0001.flac \
   $repo/test_wavs/1221-135766-0001.flac \
   $repo/test_wavs/1221-135766-0002.flac
+
+log "CTC decoding on CPU with kaldi decoders using OpenFst"
+
+log "Exporting model with torchscript"
+
+pushd $repo/exp
+ln -s pretrained.pt epoch-99.pt
+popd
+
+./conformer_ctc/export.py \
+  --epoch 99 \
+  --avg 1 \
+  --exp-dir $repo/exp \
+  --tokens $repo/data/lang_bpe_500/tokens.txt \
+  --jit 1
+
+ls -lh $repo/exp
+
+
+log "Generating H.fst, HL.fst"
+
+./local/prepare_lang_fst.py  --lang-dir $repo/data/lang_bpe_500
+ls -lh $repo/data/lang_bpe_500
+
+log "Decoding with H on CPU with OpenFst"
+
+./conformer_ctc/jit_pretrained_decode_with_H.py \
+  --nn-model $repo/exp/cpu_jit.pt \
+  --H $repo/data/lang_bpe_500/H.fst \
+  --tokens $repo/data/lang_bpe_500/tokens.txt \
+  $repo/test_wavs/1089-134686-0001.flac \
+  $repo/test_wavs/1221-135766-0001.flac \
+  $repo/test_wavs/1221-135766-0002.flac
+
+log "Decoding with HL on CPU with OpenFst"
+
+./conformer_ctc/jit_pretrained_decode_with_H.py \
+  --nn-model $repo/exp/cpu_jit.pt \
+  --HL $repo/data/lang_bpe_500/HL.fst \
+  --words $repo/data/lang_bpe_500/words.txt \
+  $repo/test_wavs/1089-134686-0001.flac \
+  $repo/test_wavs/1221-135766-0001.flac \
+  $repo/test_wavs/1221-135766-0002.flac

.github/workflows/run-pretrained-conformer-ctc.yml

@@ -29,7 +29,7 @@ concurrency:
 
 jobs:
   run_pre_trained_conformer_ctc:
-    if: github.event.label.name == 'ready' || github.event_name == 'push'
+    if: github.event.label.name == 'ready' || github.event_name == 'push' || github.event.label.name == 'ctc'
     runs-on: ${{ matrix.os }}
     strategy:
       matrix:

egs/librispeech/ASR/conformer_ctc/jit_pretrained_decode_with_H.py

@@ -0,0 +1,221 @@
+#!/usr/bin/env python3
+# Copyright      2023  Xiaomi Corp.        (authors: Fangjun Kuang)
+
+"""
+This file shows how to use a torchscript model for decoding with H
+on CPU using OpenFST and decoders from kaldi.
+
+Usage:
+
+    ./conformer_ctc/jit_pretrained_decode_with_H.py \
+      --nn-model ./cpu_jit.pt \
+      --H ./data/lang_bpe_500/H.fst \
+      --tokens ./data/lang_bpe_500/tokens.txt \
+      ./download/LibriSpeech/test-clean/1089/134686/1089-134686-0002.flac \
+      ./download/LibriSpeech/test-clean/1221/135766/1221-135766-0001.flac
+
+Note that to generate ./tdnn/exp/cpu_jit.pt,
+you can use ./export.py --jit 1
+"""
+
+import argparse
+import logging
+import math
+from typing import Dict, List
+
+import kaldi_hmm_gmm
+import kaldifeat
+import kaldifst
+import torch
+import torchaudio
+from kaldi_hmm_gmm import DecodableCtc, FasterDecoder, FasterDecoderOptions
+from torch.nn.utils.rnn import pad_sequence
+
+
+def get_parser():
+    parser = argparse.ArgumentParser(
+        formatter_class=argparse.ArgumentDefaultsHelpFormatter
+    )
+
+    parser.add_argument(
+        "--nn-model",
+        type=str,
+        required=True,
+        help="""Path to the torchscript model.
+        You can use ./tdnn/export.py --jit 1
+        to obtain it
+        """,
+    )
+
+    parser.add_argument(
+        "--tokens",
+        type=str,
+        required=True,
+        help="Path to tokens.txt",
+    )
+
+    parser.add_argument("--H", type=str, required=True, help="Path to H.fst")
+
+    parser.add_argument(
+        "sound_files",
+        type=str,
+        nargs="+",
+        help="The input sound file(s) to transcribe. "
+        "Supported formats are those supported by torchaudio.load(). "
+        "For example, wav and flac are supported. ",
+    )
+
+    return parser
+
+
+def read_tokens(tokens_txt: str) -> Dict[int, str]:
+    id2token = dict()
+    with open(tokens_txt, encoding="utf-8") as f:
+        for line in f:
+            token, idx = line.strip().split()
+            id2token[int(idx)] = token
+
+    return id2token
+
+
+def read_sound_files(
+    filenames: List[str], expected_sample_rate: float
+) -> List[torch.Tensor]:
+    """Read a list of sound files into a list 1-D float32 torch tensors.
+    Args:
+      filenames:
+        A list of sound filenames.
+      expected_sample_rate:
+        The expected sample rate of the sound files.
+    Returns:
+      Return a list of 1-D float32 torch tensors.
+    """
+    ans = []
+    for f in filenames:
+        wave, sample_rate = torchaudio.load(f)
+        if sample_rate != expected_sample_rate:
+            wave = torchaudio.functional.resample(
+                wave,
+                orig_freq=sample_rate,
+                new_freq=expected_sample_rate,
+            )
+
+        # We use only the first channel
+        ans.append(wave[0].contiguous())
+    return ans
+
+
+def decode(
+    filename: str,
+    nnet_output: torch.Tensor,
+    H: kaldifst,
+    id2token: Dict[int, str],
+) -> List[str]:
+    logging.info(f"{filename}, {nnet_output.shape}")
+    decodable = DecodableCtc(nnet_output)
+
+    decoder_opts = FasterDecoderOptions(max_active=3000)
+    decoder = FasterDecoder(H, decoder_opts)
+    decoder.decode(decodable)
+
+    if not decoder.reached_final():
+        print(f"failed to decode {filename}")
+        return ""
+
+    ok, best_path = decoder.get_best_path()
+
+    (
+        ok,
+        isymbols_out,
+        osymbols_out,
+        total_weight,
+    ) = kaldifst.get_linear_symbol_sequence(best_path)
+    if not ok:
+        print(f"failed to get linear symbol sequence for {filename}")
+        return ""
+
+    # tokens are incremented during graph construction
+    # so they need to be decremented
+    hyps = [id2token[i - 1] for i in osymbols_out]
+    #  hyps = "".join(hyps).split("▁")
+    hyps = "".join(hyps).split("\u2581")  # unicode codepoint of ▁
+
+    return hyps
+
+
+@torch.no_grad()
+def main():
+    parser = get_parser()
+    args = parser.parse_args()
+
+    device = torch.device("cpu")
+
+    logging.info(f"device: {device}")
+
+    logging.info("Loading torchscript model")
+    model = torch.jit.load(args.nn_model)
+    model.eval()
+    model.to(device)
+
+    logging.info(f"Loading H from {args.H}")
+    H = kaldifst.StdVectorFst.read(args.H)
+
+    sample_rate = 16000
+
+    logging.info("Constructing Fbank computer")
+    opts = kaldifeat.FbankOptions()
+    opts.device = device
+    opts.frame_opts.dither = 0
+    opts.frame_opts.snip_edges = False
+    opts.frame_opts.samp_freq = sample_rate
+    opts.mel_opts.num_bins = 80
+
+    fbank = kaldifeat.Fbank(opts)
+
+    logging.info(f"Reading sound files: {args.sound_files}")
+    waves = read_sound_files(
+        filenames=args.sound_files, expected_sample_rate=sample_rate
+    )
+    waves = [w.to(device) for w in waves]
+
+    logging.info("Decoding started")
+    features = fbank(waves)
+    feature_lengths = [f.shape[0] for f in features]
+    feature_lengths = torch.tensor(feature_lengths)
+
+    supervisions = dict()
+    supervisions["sequence_idx"] = torch.arange(len(features))
+    supervisions["start_frame"] = torch.zeros(len(features))
+    supervisions["num_frames"] = feature_lengths
+
+    features = pad_sequence(features, batch_first=True, padding_value=math.log(1e-10))
+
+    nnet_output, _, _ = model(features, supervisions)
+    feature_lengths = ((feature_lengths - 1) // 2 - 1) // 2
+
+    id2token = read_tokens(args.tokens)
+
+    hyps = []
+    for i in range(nnet_output.shape[0]):
+        hyp = decode(
+            filename=args.sound_files[i],
+            nnet_output=nnet_output[i, : feature_lengths[i]],
+            H=H,
+            id2token=id2token,
+        )
+        hyps.append(hyp)
+
+    s = "\n"
+    for filename, hyp in zip(args.sound_files, hyps):
+        words = " ".join(hyp)
+        s += f"{filename}:\n{words}\n\n"
+    logging.info(s)
+
+    logging.info("Decoding Done")
+
+
+if __name__ == "__main__":
+    formatter = "%(asctime)s %(levelname)s [%(filename)s:%(lineno)d] %(message)s"
+
+    logging.basicConfig(format=formatter, level=logging.INFO)
+    main()

egs/librispeech/ASR/conformer_ctc/jit_pretrained_decode_with_HL.py

@@ -0,0 +1,218 @@
+#!/usr/bin/env python3
+# Copyright      2023  Xiaomi Corp.        (authors: Fangjun Kuang)
+
+"""
+This file shows how to use a torchscript model for decoding with H
+on CPU using OpenFST and decoders from kaldi.
+
+Usage:
+
+    ./conformer_ctc/jit_pretrained_decode_with_H.py \
+      --nn-model ./cpu_jit.pt \
+      --HL ./data/lang_bpe_500/HL.fst \
+      --words ./data/lang_bpe_500/words.txt \
+      ./download/LibriSpeech/test-clean/1089/134686/1089-134686-0002.flac \
+      ./download/LibriSpeech/test-clean/1221/135766/1221-135766-0001.flac
+
+Note that to generate ./tdnn/exp/cpu_jit.pt,
+you can use ./export.py --jit 1
+"""
+
+import argparse
+import logging
+import math
+from typing import Dict, List
+
+import kaldi_hmm_gmm
+import kaldifeat
+import kaldifst
+import torch
+import torchaudio
+from kaldi_hmm_gmm import DecodableCtc, FasterDecoder, FasterDecoderOptions
+from torch.nn.utils.rnn import pad_sequence
+
+
+def get_parser():
+    parser = argparse.ArgumentParser(
+        formatter_class=argparse.ArgumentDefaultsHelpFormatter
+    )
+
+    parser.add_argument(
+        "--nn-model",
+        type=str,
+        required=True,
+        help="""Path to the torchscript model.
+        You can use ./tdnn/export.py --jit 1
+        to obtain it
+        """,
+    )
+
+    parser.add_argument(
+        "--words",
+        type=str,
+        required=True,
+        help="Path to words.txt",
+    )
+
+    parser.add_argument("--HL", type=str, required=True, help="Path to HL.fst")
+
+    parser.add_argument(
+        "sound_files",
+        type=str,
+        nargs="+",
+        help="The input sound file(s) to transcribe. "
+        "Supported formats are those supported by torchaudio.load(). "
+        "For example, wav and flac are supported. ",
+    )
+
+    return parser
+
+
+def read_words(words_txt: str) -> Dict[int, str]:
+    id2word = dict()
+    with open(words_txt, encoding="utf-8") as f:
+        for line in f:
+            word, idx = line.strip().split()
+            id2word[int(idx)] = word
+
+    return id2word
+
+
+def read_sound_files(
+    filenames: List[str], expected_sample_rate: float
+) -> List[torch.Tensor]:
+    """Read a list of sound files into a list 1-D float32 torch tensors.
+    Args:
+      filenames:
+        A list of sound filenames.
+      expected_sample_rate:
+        The expected sample rate of the sound files.
+    Returns:
+      Return a list of 1-D float32 torch tensors.
+    """
+    ans = []
+    for f in filenames:
+        wave, sample_rate = torchaudio.load(f)
+        if sample_rate != expected_sample_rate:
+            wave = torchaudio.functional.resample(
+                wave,
+                orig_freq=sample_rate,
+                new_freq=expected_sample_rate,
+            )
+
+        # We use only the first channel
+        ans.append(wave[0].contiguous())
+    return ans
+
+
+def decode(
+    filename: str,
+    nnet_output: torch.Tensor,
+    HL: kaldifst,
+    id2word: Dict[int, str],
+) -> List[str]:
+    logging.info(f"{filename}, {nnet_output.shape}")
+    decodable = DecodableCtc(nnet_output)
+
+    decoder_opts = FasterDecoderOptions(max_active=3000)
+    decoder = FasterDecoder(HL, decoder_opts)
+    decoder.decode(decodable)
+
+    if not decoder.reached_final():
+        print(f"failed to decode {filename}")
+        return ""
+
+    ok, best_path = decoder.get_best_path()
+
+    (
+        ok,
+        isymbols_out,
+        osymbols_out,
+        total_weight,
+    ) = kaldifst.get_linear_symbol_sequence(best_path)
+    if not ok:
+        print(f"failed to get linear symbol sequence for {filename}")
+        return ""
+
+    # are shifted by 1 during graph construction
+    hyps = [id2word[i] for i in osymbols_out]
+
+    return hyps
+
+
+@torch.no_grad()
+def main():
+    parser = get_parser()
+    args = parser.parse_args()
+
+    device = torch.device("cpu")
+
+    logging.info(f"device: {device}")
+
+    logging.info("Loading torchscript model")
+    model = torch.jit.load(args.nn_model)
+    model.eval()
+    model.to(device)
+
+    logging.info(f"Loading HL from {args.HL}")
+    HL = kaldifst.StdVectorFst.read(args.HL)
+
+    sample_rate = 16000
+
+    logging.info("Constructing Fbank computer")
+    opts = kaldifeat.FbankOptions()
+    opts.device = device
+    opts.frame_opts.dither = 0
+    opts.frame_opts.snip_edges = False
+    opts.frame_opts.samp_freq = sample_rate
+    opts.mel_opts.num_bins = 80
+
+    fbank = kaldifeat.Fbank(opts)
+
+    logging.info(f"Reading sound files: {args.sound_files}")
+    waves = read_sound_files(
+        filenames=args.sound_files, expected_sample_rate=sample_rate
+    )
+    waves = [w.to(device) for w in waves]
+
+    logging.info("Decoding started")
+    features = fbank(waves)
+    feature_lengths = [f.shape[0] for f in features]
+    feature_lengths = torch.tensor(feature_lengths)
+
+    supervisions = dict()
+    supervisions["sequence_idx"] = torch.arange(len(features))
+    supervisions["start_frame"] = torch.zeros(len(features))
+    supervisions["num_frames"] = feature_lengths
+
+    features = pad_sequence(features, batch_first=True, padding_value=math.log(1e-10))
+
+    nnet_output, _, _ = model(features, supervisions)
+    feature_lengths = ((feature_lengths - 1) // 2 - 1) // 2
+
+    id2word = read_words(args.words)
+
+    hyps = []
+    for i in range(nnet_output.shape[0]):
+        hyp = decode(
+            filename=args.sound_files[i],
+            nnet_output=nnet_output[i, : feature_lengths[i]],
+            HL=HL,
+            id2word=id2word,
+        )
+        hyps.append(hyp)
+
+    s = "\n"
+    for filename, hyp in zip(args.sound_files, hyps):
+        words = " ".join(hyp)
+        s += f"{filename}:\n{words}\n\n"
+    logging.info(s)
+
+    logging.info("Decoding Done")
+
+
+if __name__ == "__main__":
+    formatter = "%(asctime)s %(levelname)s [%(filename)s:%(lineno)d] %(message)s"
+
+    logging.basicConfig(format=formatter, level=logging.INFO)
+    main()

egs/librispeech/ASR/local/prepare_lang_fst.py

@@ -0,0 +1,127 @@
+#!/usr/bin/env python3
+
+# Copyright (c)  2023  Xiaomi Corporation (authors: Fangjun Kuang)
+
+"""
+This script takes as input lang_dir containing lexicon_disambig.txt,
+tokens.txt, and words.txt and generates the following files:
+
+    - H.fst
+    - HL.fst
+
+Note that saved files are in OpenFst binary format.
+
+Usage:
+
+./local/prepare_lang_fst.py \
+  --lang-dir ./data/lang_phone \
+  --has-silence 1
+
+Or
+
+./local/prepare_lang_fst.py \
+  --lang-dir ./data/lang_bpe_500
+"""
+
+import argparse
+import logging
+from pathlib import Path
+
+import kaldifst
+
+from icefall.ctc import (
+    Lexicon,
+    add_disambig_self_loops,
+    add_one,
+    build_standard_ctc_topo,
+    make_lexicon_fst_no_silence,
+    make_lexicon_fst_with_silence,
+)
+from icefall.utils import str2bool
+
+
+def get_args():
+    parser = argparse.ArgumentParser()
+    parser.add_argument(
+        "--lang-dir",
+        type=str,
+        help="""Input and output directory.
+        """,
+    )
+
+    parser.add_argument(
+        "--has-silence",
+        type=str2bool,
+        default=False,
+        help="True if the lexicon has silence.",
+    )
+
+    return parser.parse_args()
+
+
+def main():
+    args = get_args()
+    lang_dir = args.lang_dir
+
+    lexicon = Lexicon(lang_dir)
+
+    logging.info("Building standard CTC topology")
+    max_token_id = max(lexicon.tokens)
+    H = build_standard_ctc_topo(max_token_id=max_token_id)
+
+    # We need to add one to all tokens since we want to use ID 0
+    # for epsilon
+    add_one(H, treat_ilabel_zero_specially=False, update_olabel=True)
+    H.write(f"{lang_dir}/H.fst")
+
+    logging.info("Building L")
+    # Now for HL
+
+    if args.has_silence:
+        L = make_lexicon_fst_with_silence(lexicon, attach_symbol_table=False)
+    else:
+        L = make_lexicon_fst_no_silence(lexicon, attach_symbol_table=False)
+
+    if args.has_silence:
+        # We also need to change the input labels of L
+        add_one(L, treat_ilabel_zero_specially=True, update_olabel=False)
+    else:
+        add_one(L, treat_ilabel_zero_specially=False, update_olabel=False)
+
+    # Invoke add_disambig_self_loops() so that it eats the disambig symbols
+    # from L after composition
+    add_disambig_self_loops(
+        H,
+        start=lexicon.token2id["#0"] + 1,
+        end=lexicon.max_disambig_id + 1,
+    )
+    with open("H_1.fst.txt", "w") as f:
+        print(H, file=f)
+
+    kaldifst.arcsort(H, sort_type="olabel")
+    kaldifst.arcsort(L, sort_type="ilabel")
+
+    logging.info("Building HL")
+    HL = kaldifst.compose(H, L)
+    kaldifst.determinize_star(HL)
+
+    disambig0 = lexicon.token2id["#0"] + 1
+    max_disambig = lexicon.max_disambig_id + 1
+    for state in kaldifst.StateIterator(HL):
+        for arc in kaldifst.ArcIterator(HL, state):
+            # If treat_ilabel_zero_specially is False, we always change it
+            # Otherwise, we only change non-zero input labels
+            if disambig0 <= arc.ilabel <= max_disambig:
+                arc.ilabel = 0
+
+    # Note: We are not composing L with G, so there is no need to add
+    # self-loops to L to handle #0
+
+    HL.write(f"{lang_dir}/HL.fst")
+
+
+if __name__ == "__main__":
+    formatter = "%(asctime)s %(levelname)s [%(filename)s:%(lineno)d] %(message)s"
+
+    logging.basicConfig(format=formatter, level=logging.INFO)
+    main()

egs/librispeech/ASR/prepare.sh

@@ -242,6 +242,10 @@ if [ $stage -le 6 ] && [ $stop_stage -ge 6 ]; then
         $lang_dir/L_disambig.pt \
         $lang_dir/L_disambig.fst
     fi
+
+    if [ ! -f $lang_dir/HL.fst ]; then
+      ./local/prepare_lang_fst.py  --lang-dir $lang_dir
+    fi
   done
 fi
 

egs/yesno/ASR/local/prepare_lang_fst.py

@@ -1,67 +0,0 @@
-#!/usr/bin/env python3
-
-# Copyright (c)  2023  Xiaomi Corporation (authors: Fangjun Kuang)
-
-"""
-This script takes as input data/lang_phone containing lexicon_disambig.txt,
-tokens.txt, and words.txt and generates the following files:
-
-    - H.fst
-    - HL.fst
-
-TODO(fangjun): Generate HLG.fst
-
-Note that saved files are in OpenFst binary format.
-"""
-
-from pathlib import Path
-
-import kaldifst
-
-from icefall.ctc import (
-    Lexicon,
-    add_disambig_self_loops,
-    add_one,
-    build_standard_ctc_topo,
-    make_lexicon_fst_with_silence,
-)
-
-
-def main():
-    lang_dir = Path("data/lang_phone")
-    lexicon = Lexicon(lang_dir)
-
-    max_token_id = max(lexicon.tokens)
-    H = build_standard_ctc_topo(max_token_id=max_token_id)
-
-    # We need to add one to all tokens since we want to use ID 0
-    # for epsilon
-    add_one(H, treat_ilabel_zero_specially=False, update_olabel=True)
-    H.write(f"{lang_dir}/H.fst")
-
-    # Now for HL
-    L = make_lexicon_fst_with_silence(lexicon, attach_symbol_table=False)
-
-    # We also need to change the input labels of L
-    add_one(L, treat_ilabel_zero_specially=True, update_olabel=False)
-
-    # Invoke add_disambig_self_loops() so that it eats the disambig symbols
-    # from L after composition
-    add_disambig_self_loops(
-        H,
-        start=lexicon.token2id["#0"] + 1,
-        end=lexicon.max_disambig_id,
-    )
-
-    kaldifst.arcsort(H, sort_type="olabel")
-    kaldifst.arcsort(L, sort_type="ilabel")
-    HL = kaldifst.compose(H, L)
-
-    # Note: We are not composing L with G, so there is no need to add
-    # self-loops to L to handle #0
-
-    HL.write(f"{lang_dir}/HL.fst")
-
-
-if __name__ == "__main__":
-    main()

egs/yesno/ASR/local/prepare_lang_fst.py

@@ -0,0 +1 @@
+../../../librispeech/ASR/local/prepare_lang_fst.py

egs/yesno/ASR/prepare.sh

@@ -60,7 +60,7 @@ if [ $stage -le 3 ] && [ $stop_stage -ge 3 ]; then
   ) > $lang_dir/lexicon.txt
 
   ./local/prepare_lang.py
-  ./local/prepare_lang_fst.py
+  ./local/prepare_lang_fst.py --lang-dir ./data/lang_phone --has-slience 1
 fi
 
 if [ $stage -le 4 ] && [ $stop_stage -ge 4 ]; then

egs/yesno/ASR/tdnn/jit_pretrained_decode_with_H.py

@@ -28,11 +28,9 @@ import kaldifeat
 import kaldifst
 import torch
 import torchaudio
-from kaldi_hmm_gmm import FasterDecoder, FasterDecoderOptions
+from kaldi_hmm_gmm import DecodableCtc, FasterDecoder, FasterDecoderOptions
 from torch.nn.utils.rnn import pad_sequence
 
-from icefall.ctc import CtcDecodable
-
 
 def get_parser():
     parser = argparse.ArgumentParser(
@@ -113,8 +111,8 @@ def decode(
     H: kaldifst,
     id2token: Dict[int, str],
 ) -> List[str]:
-    decodable = CtcDecodable(nnet_output)
+    decodable = DecodableCtc(nnet_output)
-    decoder_opts = FasterDecoderOptions()
+    decoder_opts = FasterDecoderOptions(max_active=3000)
     decoder = FasterDecoder(H, decoder_opts)
     decoder.decode(decodable)
 

egs/yesno/ASR/tdnn/jit_pretrained_decode_with_HL.py

@@ -28,11 +28,9 @@ import kaldifeat
 import kaldifst
 import torch
 import torchaudio
-from kaldi_hmm_gmm import FasterDecoder, FasterDecoderOptions
+from kaldi_hmm_gmm import DecodableCtc, FasterDecoder, FasterDecoderOptions
 from torch.nn.utils.rnn import pad_sequence
 
-from icefall.ctc import CtcDecodable
-
 
 def get_parser():
     parser = argparse.ArgumentParser(
@@ -113,8 +111,8 @@ def decode(
     HL: kaldifst,
     id2word: Dict[int, str],
 ) -> List[str]:
-    decodable = CtcDecodable(nnet_output)
+    decodable = DecodableCtc(nnet_output)
-    decoder_opts = FasterDecoderOptions()
+    decoder_opts = FasterDecoderOptions(max_active=3000)
     decoder = FasterDecoder(HL, decoder_opts)
     decoder.decode(decodable)
 

icefall/ctc/__init__.py

@@ -1,4 +1,3 @@
-from .decodable import CtcDecodable
 from .prepare_lang import (
     Lexicon,
     make_lexicon_fst_no_silence,

icefall/ctc/decodable.py

@@ -1,30 +0,0 @@
-# Copyright    2023  Xiaomi Corp.        (author: Fangjun Kuang)
-
-import torch
-from kaldi_hmm_gmm import DecodableInterface
-
-
-class CtcDecodable(DecodableInterface):
-    """This class implements the interface
-    https://github.com/kaldi-asr/kaldi/blob/master/src/itf/decodable-itf.h
-    """
-
-    def __init__(self, nnet_output: torch.Tensor):
-        DecodableInterface.__init__(self)
-        assert nnet_output.ndim == 2, nnet_output.shape
-        self.nnet_output = nnet_output
-
-    def log_likelihood(self, frame: int, index: int) -> float:
-        # Note: We need to use index - 1 here since
-        # all the input labels of the H are incremented during graph
-        # construction
-        return self.nnet_output[frame][index - 1].item()
-
-    def is_last_frame(self, frame: int) -> bool:
-        return frame == self.nnet_output.shape[0] - 1
-
-    def num_frames_ready(self) -> int:
-        return self.nnet_output.shape[0]
-
-    def num_indices(self) -> int:
-        return self.nnet_output.shape[1]

icefall/ctc/test_ctc_topo.py

@@ -5,7 +5,12 @@ from pathlib import Path
 
 import graphviz
 import kaldifst
-from prepare_lang import Lexicon, make_lexicon_fst_with_silence
+import sentencepiece as spm
+from prepare_lang import (
+    Lexicon,
+    make_lexicon_fst_no_silence,
+    make_lexicon_fst_with_silence,
+)
 from topo import add_disambig_self_loops, add_one, build_standard_ctc_topo
 
 
@@ -85,8 +90,50 @@ def test_yesno():
     source.render(outfile="HL_yesno.pdf")
 
 
+def test_librispeech():
+    lang_dir = (
+        "/star-fj/fangjun/open-source/icefall-2/egs/librispeech/ASR/data/lang_bpe_500"
+    )
+
+    if not Path(lang_dir).is_dir():
+        print(f"{lang_dir} does not exist! Skip testing")
+        return
+
+    lexicon = Lexicon(lang_dir)
+    HL = kaldifst.StdVectorFst.read(lang_dir + "/HL.fst")
+
+    sp = spm.SentencePieceProcessor()
+    sp.load(lang_dir + "/bpe.model")
+
+    i = lexicon.word2id["HELLOA"]
+    k = lexicon.word2id["WORLD"]
+    print(i, k)
+    s = f"""
+        0 1 {i} {i}
+        1 2 {k} {k}
+        2
+    """
+    fst = kaldifst.compile(
+        s=s,
+        acceptor=False,
+    )
+
+    L = make_lexicon_fst_no_silence(lexicon, attach_symbol_table=False)
+    kaldifst.arcsort(L, sort_type="olabel")
+    with open("L.fst.txt", "w") as f:
+        print(L, file=f)
+
+    fst = kaldifst.compose(L, fst)
+    print(fst)
+    fst_dot = kaldifst.draw(fst, acceptor=False, portrait=True)
+    source = graphviz.Source(fst_dot)
+    source.render(outfile="a.pdf")
+    print(sp.encode(["HELLOA", "WORLD"]))
+
+
 def main():
     test_yesno()
+    test_librispeech()
 
 
 if __name__ == "__main__":

icefall/ctc/topo.py

@@ -107,9 +107,8 @@ def add_one(
 def add_disambig_self_loops(fst: kaldifst.StdVectorFst, start: int, end: int):
     """Add self-loops to each state.
 
-    For each disambig symbol, we add a self-loop with input label 0 and output
+    For each disambig symbol, we add a self-loop with input label disambig_id
-    label diambig_id of that disambig symbol. Note that input label 0 here
+    and output label diambig_id of that disambig symbol.
-    represents an epsilon.
 
     Args:
       fst:
@@ -119,14 +118,14 @@ def add_disambig_self_loops(fst: kaldifst.StdVectorFst, start: int, end: int):
       end:
         The ID of the last disambig symbol. For instance if there are 3
         disambig symbols ``#0``, ``#1``, and ``#2``, then ``end`` is the ID
-        of ``#3``.
+        of ``#2``.
     """
     for state in kaldifst.StateIterator(fst):
         for i in range(start, end + 1):
             fst.add_arc(
                 state=state,
                 arc=kaldifst.StdArc(
-                    ilabel=0,
+                    ilabel=i,
                     olabel=i,
                     weight=0,
                     nextstate=state,