--- language: - en license: cc-by-nc-4.0 library_name: k2 pipeline_tag: automatic-speech-recognition datasets: - esb/datasets metrics: - wer tags: - automatic-speech-recognition - speech - audio - zipformer - transducer - cr-ctc - k2-fsa - offline-asr --- # Zipformer-cr-ctc-transducer-XL-290M Offline English ASR model based on the Icefall/K2 pruned transducer Zipformer-CR-CTC recipe. The model was trained with a transducer objective and CR-CTC (Consistency-Regularized CTC) auxiliary loss, and supports Icefall transducer and CTC decoding. ## Files - `model.pt`: Zipformer CR-CTC transducer checkpoint - `bpe.model`: SentencePiece BPE model - `tokens.txt`: Icefall token table exported from `bpe.model` - `config.yaml`: model architecture, feature extraction, tokenizer, Zipformer model, decoding settings, and Hugging Face Hub download-metrics query file ## Evaluation Open ASR Leaderboard English short-form result, decoded with `modified_beam_search` and beam size 6: | Metric | Value | | --- | ---: | | Average WER | 5.64 | | Parameters | 288M | Dataset WERs: | Dataset | WER | | --- | ---: | | AMI | 11.80 | | Earnings22 | 7.73 | | GigaSpeech | 8.35 | | LibriSpeech Clean | 1.29 | | LibriSpeech Other | 3.03 | | SPGISpeech | 1.65 | | VoxPopuli | 5.64 | | **Average** | **5.64** | ## Training Data The model was trained on a combined English training mixture built from the training portions of the datasets below. | Dataset | Train Hours | Source | | --- | ---: | --- | | LibriSpeech | 960.0 | ESB datasets | | Earnings-22 | 105.0 | ESB datasets | | AMI Meeting Corpus | 78.0 | ESB datasets | | Common Voice Scripted Speech 25.0 - English | ~1,679.0 | Mozilla Data Collective | | Common Voice Spontaneous Speech 3.0 - English | ~3.6 | Mozilla Data Collective | | GigaSpeech XL | 10,000.0 | ESB datasets | | SPGISpeech | 4,900.0 | ESB datasets | | TED-LIUM Release 3 | 454.0 | ESB datasets | | VoxPopuli | 523.0 | ESB datasets | | Total | ~18,700 | ESB datasets and Mozilla Data Collective | Common Voice Scripted Speech 25.0 - English and Common Voice Spontaneous Speech 3.0 - English are the English scripted and spontaneous Common Voice releases checked on May 12, 2026. For data normalization, the training transcripts were processed with direct LLM normalization using a self-hosted GLM-4.7 setup, plus custom agentic workflows built in collaboration with Claude Code 4.7 and Codex 5.4. ## Training The model was trained with the k2-fsa/Icefall framework, PyTorch 2.11, and CUDA 13.0 on 8 NVIDIA B300 GPUs using bf16 automatic mixed precision. ## Usage With Icefall From an Icefall checkout, download this model repo locally: ```bash huggingface-cli download soundsgoodai/Zipformer-cr-ctc-transducer-XL-290M \ --local-dir Zipformer-cr-ctc-transducer-XL-290M ``` Run offline decoding through Icefall. Input audio must already be 16 kHz: ```bash cd icefall/egs/librispeech/ASR PYTHONPATH=../../.. python zipformer/pretrained.py \ --checkpoint /path/to/Zipformer-cr-ctc-transducer-XL-290M/model.pt \ --tokens /path/to/Zipformer-cr-ctc-transducer-XL-290M/tokens.txt \ --method modified_beam_search \ --beam-size 6 \ --num-encoder-layers "2,2,4,5,4,2" \ --downsampling-factor "1,2,4,8,4,2" \ --feedforward-dim "512,1024,2048,3072,2048,1024" \ --num-heads "4,4,4,8,4,4" \ --encoder-dim "192,384,768,1024,768,384" \ --encoder-unmasked-dim "192,192,320,384,320,192" \ --query-head-dim "32" \ --value-head-dim "12" \ --pos-head-dim "4" \ --pos-dim 48 \ --cnn-module-kernel "31,31,15,15,15,31" \ --decoder-dim 512 \ --joiner-dim 512 \ --context-size 2 \ --causal false \ --chunk-size "16,32,64,-1" \ --left-context-frames "64,128,256,-1" \ --use-transducer true \ --use-ctc true \ --use-attention-decoder false \ --use-cr-ctc true \ /path/to/audio_1.wav \ /path/to/audio_2.wav ``` For CTC decoding, use Icefall's CTC entrypoint with the same architecture settings: ```bash cd icefall/egs/librispeech/ASR PYTHONPATH=../../.. python zipformer/pretrained_ctc.py \ --checkpoint /path/to/Zipformer-cr-ctc-transducer-XL-290M/model.pt \ --tokens /path/to/Zipformer-cr-ctc-transducer-XL-290M/tokens.txt \ --method ctc-decoding \ --sample-rate 16000 \ --num-encoder-layers "2,2,4,5,4,2" \ --downsampling-factor "1,2,4,8,4,2" \ --feedforward-dim "512,1024,2048,3072,2048,1024" \ --num-heads "4,4,4,8,4,4" \ --encoder-dim "192,384,768,1024,768,384" \ --encoder-unmasked-dim "192,192,320,384,320,192" \ --query-head-dim "32" \ --value-head-dim "12" \ --pos-head-dim "4" \ --pos-dim 48 \ --cnn-module-kernel "31,31,15,15,15,31" \ --decoder-dim 512 \ --joiner-dim 512 \ --context-size 2 \ --causal false \ --chunk-size "16,32,64,-1" \ --left-context-frames "64,128,256,-1" \ --use-transducer true \ --use-ctc true \ --use-attention-decoder false \ --use-cr-ctc true \ /path/to/audio_1.wav \ /path/to/audio_2.wav ``` The architecture and decoding values above are also recorded in `config.yaml`. ## Decoding Methods The model supports the following Icefall decoding methods: Transducer: - `greedy_search` - `modified_beam_search` - `fast_beam_search` CTC: - `ctc-decoding` The reported Open ASR Leaderboard result uses `modified_beam_search` with beam size 6. CTC decoding is supported, but the table above does not report CTC WERs. ## Feature Extraction And Resampling For this model, use Kaldi-style fbank features. `kaldifeat` and `kaldi-native-fbank` are the recommended feature extraction backends. Audio should be mono 16 kHz before feature extraction. For sample-rate conversion, `audioop.ratecv` is recommended because it matches the resampling path used for evaluation. On Python 3.13 and newer, use an `audioop`-compatible package such as `audioop-lts`. ## Output Formatting The model emits normalized English text with punctuation and capitalization. It does not automatically capitalize the first word of every sentence unless that word is normally capitalized, such as a proper noun, honorific, acronym, or similar named expression. The model also normalizes common written forms, including numbers, dates, and currency, to digit-based forms. ## Open ASR Leaderboard Evaluation The `open_asr_leaderboard/soundsgoodai` runner downloads this model repo, resamples audio to 16 kHz with `audioop`, computes fbank features with `kaldi-native-fbank`, and decodes with Icefall `modified_beam_search`. ```bash cd open_asr_leaderboard/soundsgoodai bash run_zipformer.sh ``` ## Notes - The model expects 16 kHz mono audio. - The packaged leaderboard decoding configuration uses transducer `modified_beam_search` with beam size 6. - The checkpoint is an offline ASR model and is not intended for streaming decoding. - The checkpoint includes both transducer and CR-CTC heads, so Icefall loading should use `--use-ctc true` and `--use-cr-ctc true`. ## References - Zipformer paper: Zengwei Yao, Liyong Guo, Xiaoyu Yang, Wei Kang, Fangjun Kuang, Yifan Yang, Zengrui Jin, Long Lin, and Daniel Povey. "Zipformer: A faster and better encoder for automatic speech recognition." https://arxiv.org/abs/2310.11230 - CR-CTC paper: Zengwei Yao, Wei Kang, Xiaoyu Yang, Fangjun Kuang, Liyong Guo, Han Zhu, Zengrui Jin, Zhaoqing Li, Long Lin, and Daniel Povey. "CR-CTC: Consistency regularization on CTC for improved speech recognition." https://arxiv.org/abs/2410.05101 - Icefall: https://github.com/k2-fsa/icefall - k2: https://github.com/k2-fsa/k2 - kaldifeat: https://csukuangfj.github.io/kaldifeat/intro.html - kaldi-native-fbank: https://github.com/csukuangfj/kaldi-native-fbank - Python `audioop.ratecv`: https://docs.python.org/3.11/library/audioop.html#audioop.ratecv - audioop-lts for Python 3.13+: https://pypi.org/project/audioop-lts/ - ESB datasets: https://huggingface.co/datasets/esb/datasets - LibriSpeech: https://www.openslr.org/12/ - Earnings22: https://github.com/revdotcom/speech-datasets/tree/main/earnings22 - AMI Meeting Corpus: https://www.idiap.ch/webarchives/sites/www.amiproject.org/ami-scientific-portal/meeting-corpus/ - Common Voice Scripted Speech 25.0 - English: https://mozilladatacollective.com/datasets/cmndapwry02jnmh07dyo46mot - Common Voice Spontaneous Speech 3.0 - English: https://datacollective.mozillafoundation.org/datasets/cmn1pv5hi00uto1072y1074y7 - GigaSpeech: https://github.com/SpeechColab/GigaSpeech - SPGISpeech: https://huggingface.co/datasets/kensho/spgispeech - TED-LIUM Release 3: https://lium.univ-lemans.fr/en/ted-lium3/ - VoxPopuli: https://github.com/facebookresearch/voxpopuli