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RobotsMali/soloba-tdt-0.6b-v1.5
soloba-tdt-0.6b-v1.5 is a automatic speech recognition model from RobotsMali. Use it when you need speech turned into text. It is set up for nemo. The card lists the license as cc-by-4.0.
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From the Hugging Face model README
soloba-tdt-0.6b-v1.5 is a fine tuned version of RobotsMali/soloba-tdt-0.6b-v0.5 on RobotsMali/kunkado. This model does not consistently produce Capitalizations and Punctuations and it cannot produce acoustic event tags like those found in Kunkado its transcriptions. It was fine-tuned using NVIDIA NeMo.
This model, along with its associated resources, is part of an ongoing research effort, improvements and refinements are expected in future versions. Users should be aware that:
To 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['asr']
Note that this model has been released for research purposes primarily.
import nemo.collections.asr as nemo_asr
asr_model = nemo_asr.models.ASRModel.from_pretrained(model_name="RobotsMali/soloba-tdt-0.6b-v1.5")
This checkpoint was created with NeMo 2.5.0. Loading it with newer NeMo versions (observed with 2.7.x) can fail because the strict decoding schema expects key_phrase_items_list; see NVIDIA-NeMo/Speech#15658. This workaround was tested with Python 3.12:
from pathlib import Path
from nemo.collections.asr.models import ASRModel
from omegaconf import OmegaConf
model_name = "RobotsMali/soloba-tdt-0.6b-v1.5"
cfg = ASRModel.from_pretrained(model_name, return_config=True)
OmegaConf.set_struct(cfg, False)
for decoder in ("greedy", "beam"):
boosting_tree = OmegaConf.select(cfg, f"decoding.{decoder}.boosting_tree")
if boosting_tree is not None:
boosting_tree.key_phrase_items_list = None
config_path = Path("patched_config.yaml").resolve()
OmegaConf.save(cfg, config_path)
asr_model = ASRModel.from_pretrained(
model_name=model_name,
override_config_path=str(config_path),
strict=False,
)
config_path.unlink()
asr_model.eval()
# Assuming you have a test audio file named sample_audio.wav
asr_model.transcribe(['sample_audio.wav'])
This model accepts any mono-channel audio (wav files) as input and resamples them to 16 kHz sample rate before performing the forward pass
This model provides transcribed speech as an hypothesis object with a text attribute containing the transcription string for a given speech sample. (nemo>=2.3)
This model uses a FastConformer encoder and an autoregressive Token-and-Duration Transducer decoder, a variant of RNN-T that predicts jointly learn to predict a token and its duration. FastConformer is an optimized version of the Conformer model with 8x depthwise-separable convolutional downsampling. You may find more information on the details of FastConformer here: Fast-Conformer Model.
The NeMo toolkit was used for finetuning this model for 40,000 steps over RobotsMali/soloba-tdt-0.6b-v0.5 model with batch size 32. The finetuning codes and configurations can be found at RobotsMali-AI/bambara-asr.
The tokenizer for this model was trained on the text transcripts of the train set of RobotsMali/kunkado using this script.
This model was fine-tuned on the kunkado dataset, the human-reviewed subset, which consists of ~40 hours of transcribed Bambara speech data. The text was normalized with the bambara-normalizer prior to training, normalizing numbers, removing punctuations and removing tags.
We report the Word Error Rate (WER) and Character Error Rate (CER) for this model:
| Benchmark | Decoding | WER (%) ↓ | CER (%) ↓ |
|---|---|---|---|
| Kunkado | TDT | 39.78 | 23.21 |
| Nyana Eval | TDT | 39.81 | 22.90 |
This model is released under the CC-BY-4.0 license. By using this model, you agree to the terms of the license.
Feel free to open a discussion on Hugging Face or file an issue on GitHub for help or contributions.