Audio-Text-to-Text
Transformers
Safetensors
English
Korean
fastslm
feature-extraction
audio
text-generation
custom_code
Eval Results
Instructions to use okestro-ai-lab/FastSLM with libraries, inference providers, notebooks, and local apps. Follow these links to get started.
- Libraries
- Transformers
How to use okestro-ai-lab/FastSLM with Transformers:
# pip install -U transformers accelerate # Load model directly from transformers import AutoModel model = AutoModel.from_pretrained("okestro-ai-lab/FastSLM", trust_remote_code=True, device_map="auto") - Notebooks
- Google Colab
- Kaggle
Update README.md
Browse files
README.md
CHANGED
|
@@ -123,10 +123,9 @@ base_model:
|
|
| 123 |
# - name: Test WER
|
| 124 |
# type: wer
|
| 125 |
# value: 2.33
|
| 126 |
-
# new_version: okestro-ai-lab/SYMPHONY-ASR
|
| 127 |
---
|
| 128 |
|
| 129 |
-
*
|
| 130 |
|
| 131 |
|
| 132 |
<!-- * 🔊 **HFQ-Former**: Hierarchically compresses high-frame-rate audio features while preserving the audio's local and global contextual information. -->
|
|
@@ -135,7 +134,7 @@ base_model:
|
|
| 135 |
|
| 136 |
## 📖 Model Architecture
|
| 137 |
|
| 138 |
-
🚀 **
|
| 139 |
|
| 140 |
## 📌 Key Features (Safe Version)
|
| 141 |
|
|
@@ -178,7 +177,7 @@ import torch
|
|
| 178 |
from transformers import AutoModelForCausalLM, AutoTokenizer, GenerationConfig
|
| 179 |
|
| 180 |
# ⬅️ Enter your Hugging Face repository ID here.
|
| 181 |
-
repo_id = "okestro-ai-lab/
|
| 182 |
|
| 183 |
model = AutoModelForCausalLM.from_pretrained(
|
| 184 |
repo_id,
|
|
@@ -204,7 +203,7 @@ import librosa
|
|
| 204 |
wav_path = "sample_audio/English_audio.wav"
|
| 205 |
wav,sample_rate = librosa.load(wav_path)
|
| 206 |
|
| 207 |
-
#
|
| 208 |
if sample_rate != 16000:
|
| 209 |
audio = librosa.resample(wav,orig_sr=sample_rate,target_sr=16000)
|
| 210 |
else:
|
|
@@ -261,7 +260,7 @@ You can perform different tasks by using the following special tokens in your pr
|
|
| 261 |
|
| 262 |
## ⚡ GPU Requirements
|
| 263 |
|
| 264 |
-
|
| 265 |
|
| 266 |
| Task | Recommended GPU | Minimum VRAM |
|
| 267 |
| --------- | ---------------- | --------- |
|
|
|
|
| 123 |
# - name: Test WER
|
| 124 |
# type: wer
|
| 125 |
# value: 2.33
|
|
|
|
| 126 |
---
|
| 127 |
|
| 128 |
+
* FastSLM is an designed for efficient and accurate speech-to-text transcription.
|
| 129 |
|
| 130 |
|
| 131 |
<!-- * 🔊 **HFQ-Former**: Hierarchically compresses high-frame-rate audio features while preserving the audio's local and global contextual information. -->
|
|
|
|
| 134 |
|
| 135 |
## 📖 Model Architecture
|
| 136 |
|
| 137 |
+
🚀 **FastSLM** is a designed for efficient speech-to-text transcription.
|
| 138 |
|
| 139 |
## 📌 Key Features (Safe Version)
|
| 140 |
|
|
|
|
| 177 |
from transformers import AutoModelForCausalLM, AutoTokenizer, GenerationConfig
|
| 178 |
|
| 179 |
# ⬅️ Enter your Hugging Face repository ID here.
|
| 180 |
+
repo_id = "okestro-ai-lab/FastSLM"
|
| 181 |
|
| 182 |
model = AutoModelForCausalLM.from_pretrained(
|
| 183 |
repo_id,
|
|
|
|
| 203 |
wav_path = "sample_audio/English_audio.wav"
|
| 204 |
wav,sample_rate = librosa.load(wav_path)
|
| 205 |
|
| 206 |
+
# FastSLM requires 16kHz audio.
|
| 207 |
if sample_rate != 16000:
|
| 208 |
audio = librosa.resample(wav,orig_sr=sample_rate,target_sr=16000)
|
| 209 |
else:
|
|
|
|
| 260 |
|
| 261 |
## ⚡ GPU Requirements
|
| 262 |
|
| 263 |
+
FastSLM inference requires a GPU with sufficient memory.
|
| 264 |
|
| 265 |
| Task | Recommended GPU | Minimum VRAM |
|
| 266 |
| --------- | ---------------- | --------- |
|