模型:
nvidia/stt_en_fastconformer_hybrid_large_pc
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This model transcribes speech in upper and lower case English alphabet along with spaces, periods, commas, and question marks. It is a "large" version of FastConformer Transducer-CTC (around 115M parameters) model. This is a hybrid model trained on two losses: Transducer (default) and CTC. See the model architecture section and NeMo documentation for complete architecture details.
To train, fine-tune or play with the model you will need to install NVIDIA NeMo . We recommend you install it after you've installed latest Pytorch version.
pip install nemo_toolkit['all']
The model is available for use in the NeMo toolkit [3], and can be used as a pre-trained checkpoint for inference or for fine-tuning on another dataset.
import nemo.collections.asr as nemo_asr asr_model = nemo_asr.models.EncDecHybridRNNTCTCBPEModel.from_pretrained(model_name="nvidia/stt_en_fastconformer_hybrid_large_pc")
First, let's get a sample
wget https://dldata-public.s3.us-east-2.amazonaws.com/2086-149220-0033.wav
Then simply do:
asr_model.transcribe(['2086-149220-0033.wav'])
Using Transducer mode inference:
python [NEMO_GIT_FOLDER]/examples/asr/transcribe_speech.py pretrained_name="nvidia/stt_en_fastconformer_hybrid_large_pc" audio_dir="<DIRECTORY CONTAINING AUDIO FILES>"
Using CTC mode inference:
python [NEMO_GIT_FOLDER]/examples/asr/transcribe_speech.py pretrained_name="nvidia/stt_en_fastconformer_hybrid_large_pc" audio_dir="<DIRECTORY CONTAINING AUDIO FILES>" decoder_type="ctc"
This model accepts 16000 Hz Mono-channel Audio (wav files) as input.
This model provides transcribed speech as a string for a given audio sample.
FastConformer [1] is an optimized version of the Conformer model with 8x depthwise-separable convolutional downsampling. The model is trained in a multitask setup with joint Transducer and CTC decoder loss. You may find more information on the details of FastConformer here: Fast-Conformer Model and about Hybrid Transducer-CTC training here: Hybrid Transducer-CTC .
The NeMo toolkit [3] was used for training the models for over several hundred epochs. These model are trained with this example script and this base config .
The tokenizers for these models were built using the text transcripts of the train set with this script .
The model in this collection is trained on a composite dataset (NeMo ASRSet En PC) comprising several thousand hours of English speech:
The performance of Automatic Speech Recognition models is measuring using Word Error Rate. Since this dataset is trained on multiple domains and a much larger corpus, it will generally perform better at transcribing audio in general.
The following tables summarizes the performance of the available models in this collection with the Transducer decoder. Performances of the ASR models are reported in terms of Word Error Rate (WER%) with greedy decoding.
a) On data without Punctuation and Capitalization with Transducer decoder
Version | Tokenizer | Vocabulary Size | MCV11 DEV | MCV11 TEST | MLS DEV | MLS TEST | VOXPOPULI DEV | VOXPOPULI TEST | EUROPARL DEV | EUROPARL TEST | FISHER DEV | FISHER TEST | SPGI DEV | SPGI TEST | LIBRISPEECH DEV CLEAN | LIBRISPEECH TEST CLEAN | LIBRISPEECH DEV OTHER | LIBRISPEECH TEST OTHER | NSC DEV | NSC TEST |
---|---|---|---|---|---|---|---|---|---|---|---|---|---|---|---|---|---|---|---|---|
1.18.0 | SentencePiece Unigram | 1024 | 7.39 | 8.23 | 4.48 | 4.53 | 4.22 | 4.54 | 9.69 | 8.02 | 10.53 | 10.34 | 2.32 | 2.26 | 1.74 | 2.03 | 4.02 | 4.07 | 4.71 | 4.6 |
b) On data with Punctuation and Capitalization with Transducer decoder
Version | Tokenizer | Vocabulary Size | MCV11 DEV | MCV11 TEST | MLS DEV | MLS TEST | VOXPOPULI DEV | VOXPOPULI TEST | EUROPARL DEV | EUROPARL TEST | FISHER DEV | FISHER TEST | SPGI DEV | SPGI TEST | LIBRISPEECH DEV CLEAN | LIBRISPEECH TEST CLEAN | LIBRISPEECH DEV OTHER | LIBRISPEECH TEST OTHER | NSC DEV | NSC TEST |
---|---|---|---|---|---|---|---|---|---|---|---|---|---|---|---|---|---|---|---|---|
1.18.0 | SentencePiece Unigram | 1024 | 9.32 | 10.1 | 9.73 | 12.65 | 6.72 | 6.73 | 14.55 | 12.52 | 19.14 | 19.02 | 5.25 | 5.06 | 6.74 | 7.35 | 8.98 | 9.16 | 9.77 | 7.19 |
Since this model was trained on publically available speech datasets, the performance of this model might degrade for speech which includes technical terms, or vernacular that the model has not been trained on. The model might also perform worse for accented speech. The model only outputs the punctuations: '.', ',', '?' and hence might not do well in scenarios where other punctuations are also expected.
NVIDIA Riva , is an accelerated speech AI SDK deployable on-prem, in all clouds, multi-cloud, hybrid, on edge, and embedded. Additionally, Riva provides:
Although this model isn’t supported yet by Riva, the list of supported models is here . Check out Riva live demo .
[1] Fast Conformer with Linearly Scalable Attention for Efficient Speech Recognition
[2] Google Sentencepiece Tokenizer
License to use this model is covered by the CC-BY-4.0 . By downloading the public and release version of the model, you accept the terms and conditions of the CC-BY-4.0 license.
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