primeline/whisper-large-v3-turbo-german

Summary

This model map provides information about a model based on Whisper Large v3 that has been fine-tuned for speech recognition in German. Whisper is a powerful speech recognition platform developed by OpenAI. This model has been specially optimized for processing and recognizing German speech.

Applications

This model can be used in various application areas, including

• Transcription of spoken German language
• Voice commands and voice control
• Automatic subtitling for German videos
• Voice-based search queries in German
• Dictation functions in word processing programs

Model family

Model	Parameters	link
Whisper large v3 german	1.54B	link
Whisper large v3 turbo german	809M	link
Distil-whisper large v3 german	756M	link
tiny whisper	37.8M	link

Evaluations - Word error rate

Dataset	openai-whisper-large-v3-turbo	openai-whisper-large-v3	primeline-whisper-large-v3-german	nyrahealth-CrisperWhisper (large)	primeline-whisper-large-v3-turbo-german
Tuda-De	8.300	7.884	7.711	5.148	6.441
common_voice_19_0	3.849	3.484	3.215	1.927	3.200
multilingual librispeech	3.203	2.832	2.129	2.815	2.070
All	3.649	3.279	2.734	2.662	2.628

The data and code for evaluations are available here

Training data

The training data for this model includes a large amount of spoken German from various sources. The data was carefully selected and processed to optimize recognition performance.

Training process

The training of the model was performed with the following hyperparameters

• Batch size: 12288
• Epochs: 3
• Learning rate: 1e-6
• Data augmentation: No
• Optimizer: Ademamix

How to use

import torch
from transformers import AutoModelForSpeechSeq2Seq, AutoProcessor, pipeline
from datasets import load_dataset
device = "cuda:0" if torch.cuda.is_available() else "cpu"
torch_dtype = torch.float16 if torch.cuda.is_available() else torch.float32
model_id = "primeline/whisper-large-v3-turbo-german"
model = AutoModelForSpeechSeq2Seq.from_pretrained(
    model_id, torch_dtype=torch_dtype, low_cpu_mem_usage=True, use_safetensors=True
)
model.to(device)
processor = AutoProcessor.from_pretrained(model_id)
pipe = pipeline(
    "automatic-speech-recognition",
    model=model,
    tokenizer=processor.tokenizer,
    feature_extractor=processor.feature_extractor,
    max_new_tokens=128,
    chunk_length_s=30,
    batch_size=16,
    return_timestamps=True,
    torch_dtype=torch_dtype,
    device=device,
)
dataset = load_dataset("distil-whisper/librispeech_long", "clean", split="validation")
sample = dataset[0]["audio"]
result = pipe(sample)
print(result["text"])

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Model author: Florian Zimmermeister