whisper-small-tibetan

Model · Speech recognition

whisper-small

OpenAI

Whisper is a pre-trained model for automatic speech recognition (ASR) and speech translation. Trained on 680k hours of labelled data, Whisper models demonstrate a strong ability to generalise to many datasets and domains without the need for fine-tuning. Whisper was proposed in the paper Robust Speech Recognition via Large-Scale Weak Supervision by Alec Radford et al from OpenAI. The original code repository can be found here. Disclaimer: Content for this model card has partly been written by the Hugging Face team, and parts of it were copied and pasted from the original model card. Whisper is a Transformer based encoder-decoder model, also referred to as a sequence-to-sequence model. It…

Model · Speech recognition

whisper-small-malayalam

Sajil C K

Fine-tuned version of openai/whisper-small on a multi-corpus Malayalam speech dataset. - CPU speed (Transformers, FP32, 4 vCPU): RTF 1.96, i.e. slower than real time. For CPU deployment use the whisper.cpp builds The model was trained on an aggregated corpus of 5 Malayalam speech datasets, combined and published as sajilck/malayalam-asr-corpus. Total: ~86,000 samples across TTS-recorded, read speech, and crowdsourced domains. - 3× smaller model than Malwhisper-v1-medium, better WER - 5 corpora vs 1 — better speaker and domain diversity - Multi-domain training — TTS, read speech, and crowdsourced audio The CommonVoice figure above covers one domain. To see how the model does across all…

This is the model card of a transformers model that has been pushed on the Hub. Users (both direct and downstream) should be made aware of the risks, biases and limitations of the model. Use the code below to get started with the model. Carbon emissions can be estimated using the Machine Learning Impact calculator presented in Lacoste et al. (2019).

This model is a fine-tuned version of openai/whisper-small on an unknown dataset. The following hyperparameters were used during training: - learningrate: 1e-05 - trainbatchsize: 8 - evalbatchsize: 8 - lrschedulertype: linear - numepochs: 3 - mixedprecisiontraining: Native AMP - Transformers 5.18.0 - Pytorch 2.11.0+cu130 - Datasets 4.8.5 - Tokenizers 0.23.2

Model · Speech recognition

mms-300m-1130-forced-aligner

Mahmoud Ashraf

This Python package provides an efficient way to perform forced alignment between text and audio using Hugging Face's pretrained models. it also features an improved implementation to use much less memory than TorchAudio forced alignment API. The model checkpoint uploaded here is a conversion from torchaudio to HF Transformers for the MMS-300M checkpoint trained on forced alignment dataset

Model · Speech recognition

wav2vec2-xls-r-300m-hebrew

Vladimir Gurevich

This model is a fine-tuned version of facebook/wav2vec2-xls-r-300m on the private datasets in 2 stages - firstly was fine-tuned on a small dataset with good samples Then the obtained model was fine-tuned on a large dataset with the small good dataset, with various samples from different sources, and with an unlabeled dataset that was weakly labeled using a previously trained model. (weakly labeled data wasn't used in validation set) on small dataset on large dataset on small dataset on large dataset The following hyperparameters were used during training: - learningrate: 0.0003 - trainbatchsize: 8 - evalbatchsize: 8 - distributedtype: multi-GPU - numdevices: 2 - gradientaccumulationsteps: 4…